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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1,	ice-happy-eyeballs and ICE bis
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Date: Thu, 1 Aug 2013 08:19:47 +0000
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Cc: Jonathan Lennox <jonathan@vidyo.com>, =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Hi,

I want to revisit a discussion from an earlier version of the draft.=20

An earlier version of the draft suggested to do IPv4 and IPv6 checks in par=
allel. I like that idea because:

1. It really speeds up the connectivity check phase
2. Should have no impact on the NAT port allocation problem. (Or is there a=
 way for an IPv6 conn check to have a negative impact on the IPv4 NAT?)
3. Easy to implement. I would just change the way I traverse the the ordere=
d checklist. Send first available IPv6 test, then search the list for the f=
irst available IPv4 check. To avoid timing issues a ice-option might be use=
d to say that the checklist is done in parallel if it contains different ad=
dress types. (Do we want to support IPv7.. :-) )

.-.
P=E5l-Erik

On Jul 31, 2013, at 18:16 PM, Martin Thomson <martin.thomson@gmail.com>
 wrote:

> On 31 July 2013 17:56, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote:
>> overly complex
>=20
> Yes.  Given that this is a very small part of a spec that already fits
> that description, we need to be careful.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From ietf@meetecho.com  Thu Aug  1 02:07:29 2013
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Date: Thu, 1 Aug 2013 11:06:42 +0200 (CEST)
From: Meetecho Team <ietf@meetecho.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] MMUSIC session recording available
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Dear all,

the full recording (synchronized video, audio, slides and jabber room) of the 
MMUSIC WG session at IETF 87 is available at the following URL:
http://ietf87.conf.meetecho.com/index.php/Recorded_Sessions#MMUSIC

For the chair(s): please feel free to put the link to the recording in the minutes,
if you think this might be useful.

Cheers,
the Meetecho Team


This email has been automatically generated by The Meetecho Conferencing System


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From martin.thomson@gmail.com  Thu Aug  1 04:27:24 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 1 August 2013 10:19, Pal Martinsen (palmarti) <palmarti@cisco.com> wrote=
:
>
> An earlier version of the draft suggested to do IPv4 and IPv6 checks in p=
arallel. I like that idea because:

I don't think that that is necessary.

> 1. It really speeds up the connectivity check phase
> 2. Should have no impact on the NAT port allocation problem. (Or is there=
 a way for an IPv6 conn check to have a negative impact on the IPv4 NAT?)

I can think of a way that this would be bad, but I'm only speculating
about NAT implementation in doing so.

> 3. Easy to implement. I would just change the way I traverse the the orde=
red checklist. Send first available IPv6 test, then search the list for the=
 first available IPv4 check. To avoid timing issues a ice-option might be u=
sed to say that the checklist is done in parallel if it contains different =
address types. (Do we want to support IPv7.. :-) )

I don't believe that this is any easier to implement.  In fact, the
easiest thing to implement is a change to the priority calculation.
Besides, a priority change will actually cause the checklist ordering
to be the same without new signaling, anything else won't.

From bernard_aboba@hotmail.com  Thu Aug  1 04:41:04 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "Tirumaleswar	Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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> An earlier version of the draft suggested to do IPv4 and IPv6 checks in p=
arallel. I like that idea because:
>=20
> 1. It really speeds up the connectivity check phase
[BA] Not if the constraints on sending of connectivity checks are still ope=
rative.=20
> 2. Should have no impact on the NAT port allocation problem. (Or is there=
 a way for an IPv6 conn check to have a negative impact on the IPv4 NAT?)
[BA] There are (pathological) cases where a host with an IPv6 address may f=
ind that it has an IPv4 reflexive address (NAT64).=20
> 3. Easy to implement.=20
[BA] Given that there are now tens of millions of deployed hosts running IC=
E based on RFC 5245=2C "ease of implementation" of non-backward compatible =
extensions should not be a criteria.  A more relevant question is "can this=
 problem be solved in a backward compatible way?"  In this case=2C I believ=
e that the answer is Yes.   Given that=2C there is neither a need nor a com=
pelling case for breaking backward compatibility.  		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'><span style=3D"font-size: 12pt=
=3B">&gt=3B An earlier version of the draft suggested to do IPv4 and IPv6 c=
hecks in parallel. I like that idea because:</span><br><div>&gt=3B <br>&gt=
=3B 1. It really speeds up the connectivity check phase</div><div><br></div=
><div>[BA]&nbsp=3B<span style=3D"font-size: 12pt=3B">Not if the constraints=
 on sending of connectivity checks are still operative.&nbsp=3B</span></div=
><div><br>&gt=3B 2. Should have no impact on the NAT port allocation proble=
m. (Or is there a way for an IPv6 conn check to have a negative impact on t=
he IPv4 NAT?)</div><div><br></div><div>[BA] There are (pathological) cases =
where a host with an IPv6 address may find that it has an IPv4 reflexive ad=
dress (NAT64).&nbsp=3B</div><div><br>&gt=3B 3. Easy to implement.&nbsp=3B</=
div><div><br></div><div>[BA] Given that there are now tens of millions of d=
eployed hosts running ICE based on RFC 5245=2C "ease of implementation" of =
non-backward compatible extensions should not be a criteria. &nbsp=3BA more=
 relevant question is "can this problem be solved in a backward compatible =
way?" &nbsp=3BIn this case=2C I believe that the answer is Yes. &nbsp=3B Gi=
ven that=2C there is neither a need nor a compelling case for breaking back=
ward compatibility.&nbsp=3B</div> 		 	   		  </div></body>
</html>=

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From jonathan@vidyo.com  Thu Aug  1 04:50:46 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Martin Thomson <martin.thomson@gmail.com>
Date: Thu, 1 Aug 2013 07:50:35 -0400
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Aug 1, 2013, at 1:26 PM, Martin Thomson <martin.thomson@gmail.com> wrote=
:

> On 1 August 2013 10:19, Pal Martinsen (palmarti) <palmarti@cisco.com> wro=
te:
>> 1. It really speeds up the connectivity check phase
>> 2. Should have no impact on the NAT port allocation problem. (Or is ther=
e a way for an IPv6 conn check to have a negative impact on the IPv4 NAT?)
>=20
> I can think of a way that this would be bad, but I'm only speculating
> about NAT implementation in doing so.

Presumably a check on a Teredo candidate impacts your IPv4 NAT.

--
Jonathan Lennox
jonathan@vidyo.com



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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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> -----Original Message-----
> From: Ari Ker=E4nen [mailto:ari.keranen@ericsson.com]
> Sent: Wednesday, July 31, 2013 9:26 PM
> To: Jonathan Lennox
> Cc: Tirumaleswar Reddy (tireddy); mmusic@ietf.org; Martin Thomson
> Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and IC=
E bis
>=20
> On 7/31/13 4:45 PM, Jonathan Lennox wrote:
> >
> > On Jul 31, 2013, at 4:13 PM, "Tirumaleswar Reddy (tireddy)"
> <tireddy@cisco.com> wrote:
> >>
> >> Using the formula in RFC 5245
> >>
> >>    priority =3D (2^24)*(type preference) +
> >>               (2^8)*(local preference) +
> >>               (2^0)*(256 - component ID)
> >>
> >> Type preference of
> >>   Host candidates > Server-reflexive candidates > Relayed candidates
> >>
> >> (For example host candidate is 126, server-reflexive 105)
> >>
> >> that causes the problem that when candidate pair priority is calculate=
d
> (priority =3D 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0))
> >> The candidate pair IPv4 (host, server-reflexive) will only be tested o=
nly
> after all IPv6 host candidate-pairs, IPv4 host candidate pairs.
> >>
> >> Even if policy table RFC 6724 is changed so that IPv4 addresses have h=
igher
> precedence that does not solve the problem, It will still result in IPv4
> (host, server-reflexive) to be tested only after all IPv4 host candidate
> pairs, IPv6 host candidate-pairs.
> >>
> >> Hence during the WG session, I had explained that by tweaking the cand=
idate
> priority will not solve the problem, but the candidate pair priority has =
to be
> adjusted after offer/exchange such that after NNN number of preferred IPv=
6
> candidate pairs the highest-priority IPv4 candidate-pair is tested and th=
e
> above adjustment continued until all candidate pairs of the preferred add=
ress
> family are exhausted.
> >
> > The first formula you quote is the one that RFC 5245 defines as recomme=
nded,
> not mandatory.  What we're suggesting be changed is not the "local prefer=
ence"
> field, but actually the calculated 32-bit "priority" field for the candid=
ate.
> You can set your priorities such that some server-reflexive candidates ha=
ve
> higher priority than some host candidates, and the calculation of the pai=
r's
> 64-bit priority will still work.
> >
> > We need to be pretty careful when defining a new formula for how these
> priorities are set, though.
>=20
> If it's only about updating that formula, this indeed looks like a good
> way forward.
>=20
> Just need to think how to get all we want to express (and what do we
> actually want to express) in a priority formula that isn't overly
> complex. Expressing things like "some IPv4 host-srvrflx should be
> prioritized over (some) IPv6 host-host but v4 host-host below (some) v6
> host-host" etc may get a bit messy.

How about using the existing recommended formula in RFC 5245 and after the =
local candidates are prioritized do the following additional steps :
=20
a.  If the first 'N' candidates are of the same IP address family, then the=
 highest-priority candidate of the other address family is promoted to posi=
tion 'N' in the list.

b.  Step b is repeated for rest of candidates. This is continued until all =
candidates with the preferred address family are exhausted.

This would work even when policy table is modified such that IPv4 addresses=
 have higher precedence than IPv6 using DHCP http://tools.ietf.org/html/dra=
ft-ietf-6man-addr-select-opt-10.=20

--Tiru.
>=20
>=20
> Cheers,
> Ari

From martin.thomson@gmail.com  Fri Aug  2 23:48:03 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 2 August 2013 17:11, Tirumaleswar Reddy (tireddy) <tireddy@cisco.com> wrote:
> a.  If the first 'N' candidates are of the same IP address family, then the highest-priority candidate of the other address family is promoted to position 'N' in the list.
>
> b.  Step b is repeated for rest of candidates. This is continued until all candidates with the preferred address family are exhausted.
>
> This would work even when policy table is modified such that IPv4 addresses have higher precedence than IPv6 using DHCP http://tools.ietf.org/html/draft-ietf-6man-addr-select-opt-10.

I'm not sure that your specific formulation of the algorithm makes
sense (reread step b), but I think that the general principle you use
in the draft (as you explained in the WG meeting) is certainly a
viable option.

From tireddy@cisco.com  Sat Aug  3 12:56:27 2013
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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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From christer.holmberg@ericsson.com  Mon Aug  5 07:06:52 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Subject: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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--_000_7594FB04B1934943A5C02806D1A2204B1C41D191ESESSMB209erics_
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Hi,

Within SDP Offers and Answers, BUNDLE (-04) does currently not allow usage =
of port zero in bundled m- lines.

However, due to the Berlin agreement to move forward with the UP (unified p=
lan), we need to change that for SDP Offers, as port zero is used in bundle=
d "bundle-only" m- lines.

Now, as the agreed context for the UP was WEBRTC only, we need to determine=
 whether similar restrictions are needed in the generic BUNDLE spec. We als=
o need to decide whether port zero is allowed for "bundle-only" m- lines on=
ly, or for any bundled m- line.

So, a couple of questions:

Q10: Assuming we define the SDP bundle-only attribute in the BUNDLE spec, d=
o we need to limit the scope to WEBRTC (or whatever the scope also in the B=
UNDLE spec?

Q11: Do we, in SDP Offers, allow port zero ONLY in "bundle-only" m- lines, =
or in ANY bundled m- line (obviously, the semantics will vary)?

Note that the UP does NOT require bundled port zero m- lines in SDP Answers=
, and I am NOT suggesting a change to that.

Regards,

Christer




Sent from Windows using TouchDown (www.nitrodesk.com)

--_000_7594FB04B1934943A5C02806D1A2204B1C41D191ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"font-family:Calibri, Arial, Helvetica, sans-serif;font-size:=
11pt;color:black">
<div>&nbsp;</div>
<div><font face=3D"Calibri">Hi,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Within SDP Offers and Answers, BUNDLE (-04)&nbs=
p;does currently not allow usage of port zero in bundled m- lines.</font></=
div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">However,&nbsp;due to the Berlin&nbsp;agreement =
to move forward with the UP (unified plan), we need to change that for SDP =
Offers, as port zero is used&nbsp;in bundled&nbsp;&quot;bundle-only&quot; m=
- lines.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Now, as the agreed context for the UP was WEBRT=
C only, we need to determine whether similar restrictions are needed in the=
 generic BUNDLE spec. We also need to decide&nbsp;whether port zero is allo=
wed for&nbsp;&quot;bundle-only&quot; m- lines only, or
 for any bundled m- line.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">So, a couple of questions:</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Q10: Assuming we define the SDP bundle-only att=
ribute in the BUNDLE spec, do we need to limit the scope to WEBRTC (or what=
ever the scope also in the BUNDLE spec?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Q11: Do we, in SDP Offers, allow port zero ONLY=
 in &quot;bundle-only&quot; m- lines, or in ANY bundled m- line (obviously,=
 the semantics&nbsp;will vary)?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Note that the UP does NOT require bundled port =
zero m- lines in SDP Answers, and I am NOT suggesting a change to that.</fo=
nt></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<div>&nbsp;</div>
<span style=3D"color: black; font-family: Calibri, Arial, Helvetica, sans-s=
erif; font-size: 11pt;">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color: blue;">TouchDown</=
b> (www.nitrodesk.com)</div>
</span>
<div></div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C41D191ESESSMB209erics_--

From hadriel.kaplan@oracle.com  Mon Aug  5 07:51:33 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On Jul 31, 2013, at 12:42 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> =
wrote:

> I don't understand how sendonly and recvonly reduce glare.
> If I offer a sendonly m-line, it still must be answered. And =
subsequently an offer for that m-line can come from either side, =
possibly producing glare.
>=20
> ISTM that you should offer a sendrecv m-line iff you intend a =
bidirectional flow, where the two sides are related, will have common =
properties and lifetimes.
>=20
> There is still a problem if both sides want a bidirectional flow, but =
aren't agreed on who should initiate it. In that case they both might =
offer and glare. And there there remains the question of whether they =
recognize this as glare, rather than thinking they need both flows.

My understanding is the concept isn't such that there isn't an offer =
going in both directions - there is an offer going in both directions.  =
But if the two offers don't "collide" in the things they change, then =
it's not a glare, and instead they can just accept the offer they each =
receive as an answer.

So in the direction attribute case, if you send a=3Dsendonly and the =
other side sent a=3Drecvonly, there is no collision - each side can just =
treat the offer as an answer.  Likewise if you both send "a=3Dsendrecv", =
or both send "a=3Dinactive".

We can even define rock-paper-scissor rules, as was done for when a =
stream is removed and changed at the same time (in that case removal =
trumps changing).  For example, we can make it so that an "a=3Dinactive" =
trumps all other directions; so regardless of what direction attribute =
you sent, if you receive an "a=3Dinactive", that's the new state.

We could even define a lookup table for it, like this:

Sent Offer  Rcvd Offer      Local state
--------------------------------------
sendrecv    sendrecv        sendrecv
sendrecv    sendonly        recvonly
sendrecv    recvonly        sendonly
sendrecv    inactive        inactive
sendonly    sendrecv        sendonly
sendonly    sendonly        inactive
sendonly    recvonly        sendonly
sendonly    inactive        inactive
recvonly    sendrecv        recvonly
recvonly    sendonly        recvonly
recvonly    recvonly        inactive
recvonly    inactive        inactive
inactive    sendrecv        inactive
inactive    sendonly        inactive
inactive    recvonly        inactive
inactive    inactive        inactive

-hadriel


From christer.holmberg@ericsson.com  Mon Aug  5 08:01:44 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] unified plan glareless atomicity improvement
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Date: Mon, 5 Aug 2013 15:01:35 +0000
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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So, you are basically saying that both endpoints can send partial offers fo=
r the SAME m- line, WITHOUT "triggering" glare, if there is no "state colli=
sion"???

My understanding of the UP (unified plan) was that you would still get glar=
e if both endpoints simultaneously send a partial offer for the same m- lin=
e.

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Paul Kyzivat [pkyzivat@alum.mit.edu]
CC: mmusic@ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement

On Jul 31, 2013, at 12:42 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I don't understand how sendonly and recvonly reduce glare.
> If I offer a sendonly m-line, it still must be answered. And subsequently=
 an offer for that m-line can come from either side, possibly producing gla=
re.
>
> ISTM that you should offer a sendrecv m-line iff you intend a bidirection=
al flow, where the two sides are related, will have common properties and l=
ifetimes.
>
> There is still a problem if both sides want a bidirectional flow, but are=
n't agreed on who should initiate it. In that case they both might offer an=
d glare. And there there remains the question of whether they recognize thi=
s as glare, rather than thinking they need both flows.

My understanding is the concept isn't such that there isn't an offer going =
in both directions - there is an offer going in both directions.  But if th=
e two offers don't "collide" in the things they change, then it's not a gla=
re, and instead they can just accept the offer they each receive as an answ=
er.

So in the direction attribute case, if you send a=3Dsendonly and the other =
side sent a=3Drecvonly, there is no collision - each side can just treat th=
e offer as an answer.  Likewise if you both send "a=3Dsendrecv", or both se=
nd "a=3Dinactive".

We can even define rock-paper-scissor rules, as was done for when a stream =
is removed and changed at the same time (in that case removal trumps changi=
ng).  For example, we can make it so that an "a=3Dinactive" trumps all othe=
r directions; so regardless of what direction attribute you sent, if you re=
ceive an "a=3Dinactive", that's the new state.

We could even define a lookup table for it, like this:

Sent Offer  Rcvd Offer      Local state
--------------------------------------
sendrecv    sendrecv        sendrecv
sendrecv    sendonly        recvonly
sendrecv    recvonly        sendonly
sendrecv    inactive        inactive
sendonly    sendrecv        sendonly
sendonly    sendonly        inactive
sendonly    recvonly        sendonly
sendonly    inactive        inactive
recvonly    sendrecv        recvonly
recvonly    sendonly        recvonly
recvonly    recvonly        inactive
recvonly    inactive        inactive
inactive    sendrecv        inactive
inactive    sendonly        inactive
inactive    recvonly        inactive
inactive    inactive        inactive

-hadriel

_______________________________________________
mmusic mailing list
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<div>&nbsp;</div>
<div><font face=3D"Calibri">So, you are basically saying that both endpoint=
s can send partial&nbsp;offers for the SAME m- line, WITHOUT &quot;triggeri=
ng&quot; glare, if there is no &quot;state collision&quot;???</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">My understanding of the UP (unified plan) was t=
hat you would still get glare if both endpoints simultaneously send a parti=
al offer for the same m- line.</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>CC:</b> mmusic@ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] unified plan glareless atomicity improvement</=
span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Jul 31, 2013, at 12:42 PM, Paul Kyzivat &lt;pkyzivat@alum.mit.edu&gt; wr=
ote:<br>
<br>
&gt; I don't understand how sendonly and recvonly reduce glare.<br>
&gt; If I offer a sendonly m-line, it still must be answered. And subsequen=
tly an offer for that m-line can come from either side, possibly producing =
glare.<br>
&gt; <br>
&gt; ISTM that you should offer a sendrecv m-line iff you intend a bidirect=
ional flow, where the two sides are related, will have common properties an=
d lifetimes.<br>
&gt; <br>
&gt; There is still a problem if both sides want a bidirectional flow, but =
aren't agreed on who should initiate it. In that case they both might offer=
 and glare. And there there remains the question of whether they recognize =
this as glare, rather than thinking
 they need both flows.<br>
<br>
My understanding is the concept isn't such that there isn't an offer going =
in both directions - there is an offer going in both directions.&nbsp; But =
if the two offers don't &quot;collide&quot; in the things they change, then=
 it's not a glare, and instead they can just accept
 the offer they each receive as an answer.<br>
<br>
So in the direction attribute case, if you send a=3Dsendonly and the other =
side sent a=3Drecvonly, there is no collision - each side can just treat th=
e offer as an answer.&nbsp; Likewise if you both send &quot;a=3Dsendrecv&qu=
ot;, or both send &quot;a=3Dinactive&quot;.<br>
<br>
We can even define rock-paper-scissor rules, as was done for when a stream =
is removed and changed at the same time (in that case removal trumps changi=
ng).&nbsp; For example, we can make it so that an &quot;a=3Dinactive&quot; =
trumps all other directions; so regardless of what
 direction attribute you sent, if you receive an &quot;a=3Dinactive&quot;, =
that's the new state.<br>
<br>
We could even define a lookup table for it, like this:<br>
<br>
Sent Offer&nbsp; Rcvd Offer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Local state<br>
--------------------------------------<br>
sendrecv&nbsp;&nbsp;&nbsp; sendrecv&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; sendrecv<br>
sendrecv&nbsp;&nbsp;&nbsp; sendonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; recvonly<br>
sendrecv&nbsp;&nbsp;&nbsp; recvonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; sendonly<br>
sendrecv&nbsp;&nbsp;&nbsp; inactive&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
sendonly&nbsp;&nbsp;&nbsp; sendrecv&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; sendonly<br>
sendonly&nbsp;&nbsp;&nbsp; sendonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
sendonly&nbsp;&nbsp;&nbsp; recvonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; sendonly<br>
sendonly&nbsp;&nbsp;&nbsp; inactive&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
recvonly&nbsp;&nbsp;&nbsp; sendrecv&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; recvonly<br>
recvonly&nbsp;&nbsp;&nbsp; sendonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; recvonly<br>
recvonly&nbsp;&nbsp;&nbsp; recvonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
recvonly&nbsp;&nbsp;&nbsp; inactive&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
inactive&nbsp;&nbsp;&nbsp; sendrecv&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
inactive&nbsp;&nbsp;&nbsp; sendonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
inactive&nbsp;&nbsp;&nbsp; recvonly&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
inactive&nbsp;&nbsp;&nbsp; inactive&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; inactive<br>
<br>
-hadriel<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: Can a partial offer be a full offer?
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Date: Mon, 5 Aug 2013 15:10:42 +0000
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Subject: [MMUSIC] UP: Can a partial offer be a full offer?
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Hi,

I realize we are still far from having a working partial O/A solution, and =
MMUSIC may not even be the place to discuss it, but I think we need to warm=
 up our brains a little, so I'll throw out a ball related to the semantics =
:)

When reading the UP draft, I can't find any restrictions on how many m- lin=
es a partial offer can contain.

So, I assume that means I can send a partial offer which contains ALL m- li=
nes of my session SDP, i.e. it would look like a vanilla 3264 offer?

Now, WHY would I do that? Well, if we are going to use something more "ligh=
tweight" than re-INVITE/UPDATE to send partial offers, that may be a good r=
eason.

Regards,

Christer


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erif; font-size: 11pt;">
<div><br>
Hi,</div>
<div>&nbsp;</div>
<div>I realize we are still far from having a working partial O/A solution,=
 and MMUSIC may not even be the place to discuss it, but I think we need to=
 warm up our brains a little, so I'll throw out a ball related to the seman=
tics :)</div>
<div>&nbsp;</div>
<div>When reading the UP draft, I can't find any restrictions on how many m=
- lines a partial offer can contain.</div>
<div>&nbsp;</div>
<div>So, I assume that means I can send a partial offer which contains ALL =
m- lines of my session SDP, i.e. it would look like a vanilla 3264 offer?</=
div>
<div>&nbsp;</div>
<div>Now,&nbsp;WHY would I do that? Well, if we are going to use something =
more &quot;lightweight&quot; than re-INVITE/UPDATE to send partial offers, =
that may be a good reason.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color: blue;">TouchDown</=
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] UP: Can a partial offer be a full offer?
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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Note that my re-INVITE/UPDATE example is based on the assumption that one c=
an use partial offers/answers with SIP, on the wire. Not sure if that was p=
art of the agreed scope for the UP.

But, even without that, I think the question on including all m- lines on a=
 partial offer still applies :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Christer Holmberg [christer.holmberg@ericsson.com]
To: mmusic [mmusic@ietf.org]
Subject: [MMUSIC] UP: Can a partial offer be a full offer?

Hi,

I realize we are still far from having a working partial O/A solution, and =
MMUSIC may not even be the place to discuss it, but I think we need to warm=
 up our brains a little, so I'll throw out a ball related to the semantics =
:)

When reading the UP draft, I can't find any restrictions on how many m- lin=
es a partial offer can contain.

So, I assume that means I can send a partial offer which contains ALL m- li=
nes of my session SDP, i.e. it would look like a vanilla 3264 offer?

Now, WHY would I do that? Well, if we are going to use something more "ligh=
tweight" than re-INVITE/UPDATE to send partial offers, that may be a good r=
eason.

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

--_000_7594FB04B1934943A5C02806D1A2204B1C41D9A9ESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11=
pt; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Note that my re-INVITE/UPDATE example is based on the=
 assumption that one can use partial offers/answers with SIP, on the wire. =
Not sure if that was part of the agreed
 scope for the UP.</span></div>
<div>&nbsp;</div>
<div>But, even without that, I think the question on including all m- lines=
 on a partial offer still applies :)</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>To:</b> mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> [MMUSIC] UP: Can a partial offer be a full offer?</span>
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">
<div><br>
Hi,</div>
<div>&nbsp;</div>
<div>I realize we are still far from having a working partial O/A solution,=
 and MMUSIC may not even be the place to discuss it, but I think we need to=
 warm up our brains a little, so I'll throw out a ball related to the seman=
tics :)</div>
<div>&nbsp;</div>
<div>When reading the UP draft, I can't find any restrictions on how many m=
- lines a partial offer can contain.</div>
<div>&nbsp;</div>
<div>So, I assume that means I can send a partial offer which contains ALL =
m- lines of my session SDP, i.e. it would look like a vanilla 3264 offer?</=
div>
<div>&nbsp;</div>
<div>Now,&nbsp;WHY would I do that? Well, if we are going to use something =
more &quot;lightweight&quot; than re-INVITE/UPDATE to send partial offers, =
that may be a good reason.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
</div>
</body>
</html>

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From adam@nostrum.com  Mon Aug  5 08:21:12 2013
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On 8/5/13 10:17, Christer Holmberg wrote:
> Note that my re-INVITE/UPDATE example is based on the assumption that 
> one can use partial offers/answers with SIP, on the wire. Not sure if 
> that was part of the agreed scope for the UP.

 From the phrasing the chairs used when asking the question, my 
understanding is that it's not. That doesn't mean that SIPCORE won't 
decide to come along and do it at some point in the future, but it's a 
different question for a different time.

> But, even without that, I think the question on including all m- lines 
> on a partial offer still applies :)

Yeah, I'm pondering this. I'll try to respond when I have my thoughts 
gathered.

/a

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    <div class="moz-cite-prefix">On 8/5/13 10:17, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41D9A9@ESESSMB209.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div><span style="color:black;
          font-family:Calibri,Arial,Helvetica,sans-serif;
          font-size:11pt">Note that my re-INVITE/UPDATE example is based
          on the assumption that one can use partial offers/answers with
          SIP, on the wire. Not sure if that was part of the agreed
          scope for the UP.</span></div>
    </blockquote>
    <br>
    From the phrasing the chairs used when asking the question, my
    understanding is that it's not. That doesn't mean that SIPCORE won't
    decide to come along and do it at some point in the future, but it's
    a different question for a different time.<br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41D9A9@ESESSMB209.ericsson.se"
      type="cite">
      <div>But, even without that, I think the question on including all
        m- lines on a partial offer still applies :)</div>
    </blockquote>
    <br>
    Yeah, I'm pondering this. I'll try to respond when I have my
    thoughts gathered.<br>
    <br>
    /a<br>
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--------------050202080801060805020102--

From hadriel.kaplan@oracle.com  Mon Aug  5 08:21:33 2013
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 5, 2013, at 10:06 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Within SDP Offers and Answers, BUNDLE (-04) does currently not allow =
usage of port zero in bundled m- lines.
> However, due to the Berlin agreement to move forward with the UP =
(unified plan), we need to change that for SDP Offers, as port zero is =
used in bundled "bundle-only" m- lines.
> =20
> Now, as the agreed context for the UP was WEBRTC only, we need to =
determine whether similar restrictions are needed in the generic BUNDLE =
spec.

My understanding of the consensus call was that the UP as a single, =
collective whole was scoped for WebRTC for now.  That doesn't mean =
individual parts can't be generically applicable, nor that we can't =
decide all the parts could be generically applicable in the end.


> We also need to decide whether port zero is allowed for "bundle-only" =
m- lines only, or for any bundled m- line.
> So, a couple of questions:
> =20
> Q10: Assuming we define the SDP bundle-only attribute in the BUNDLE =
spec, do we need to limit the scope to WEBRTC (or whatever the scope =
also in the BUNDLE spec?

If the mechanism is backwards-compatible with not only existing RFCs, =
but also existing deployed systems, I can't see a reason not to make the =
mechanism generically applicable.  It was my understanding that the =
bundle-only port-0 concept requires and will always perform a second =
offer/answer round, with real port numbers.  Yes or no?


> =20
> Q11: Do we, in SDP Offers, allow port zero ONLY in "bundle-only" m- =
lines, or in ANY bundled m- line (obviously, the semantics will vary)?

If I understand it correctly, a "bundle-only" m-line is one that you =
only want to use if and only if the other end supports BUNDLE; a =
"bundled" m-line is one you want to use regardless of whether the =
far-end supports BUNDLE, but would want to bundle together if the =
far-end does support BUDNLE.  Correct?  If so, then let's call the =
"bundled" one a "bundle-up" for now, for "bundle upgrade".

If the above is correct, then ISTM the goal of a "bundle-up" m-line is =
(1) backwards compatibility, and (2) to successfully use the media =
stream regardless of the far-end's BUNDLE support.  For (1) I think it's =
safer not to use a port of 0.  For (2) I don't see a purpose in setting =
an m-line to port 0.  So why do it for bundle-up?

-hadriel


From adam@nostrum.com  Mon Aug  5 08:31:39 2013
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On 8/5/13 10:21, Hadriel Kaplan wrote:
> It was my understanding that the bundle-only port-0 concept requires and will always perform a second offer/answer round, with real port numbers.  Yes or no?

The port-0 concept does *not* require a second exchange.

There are a lot of circumstances in which a second exchange is likely to 
take place (cf. "Handshaking"), but none of these are influenced one way 
or another by the use of zero-ports on bundle-only initial offers.

I presume you're concerned that the far end won't know which port to 
use? There's really not an issue here. These lines are interpreted by a 
bundle-aware implementation as having the same port number as the 
non-bundle-only lines that they're bundled together with. To be clear: 
you can't have a bundle that consists exclusively of bundle-only lines. 
At least one of them has to be non-bundle-only and have a real port 
number associated with it.

/a

From hadriel.kaplan@oracle.com  Mon Aug  5 08:34:34 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On Aug 5, 2013, at 11:17 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Note that my re-INVITE/UPDATE example is based on the assumption that =
one can use partial offers/answers with SIP, on the wire. Not sure if =
that was part of the agreed scope for the UP.

Definitely NOT agreed on.  Doesn't mean it won't happen in the future, =
but the scope of UP was limited to WebRTC for now.  We can debate which =
parts of UP are generically applicable, and which not.  For SIP, this =
isn't the right WG for that debate.


> But, even without that, I think the question on including all m- lines =
on a partial offer still applies :)

I don't see why WebRTC would restrict itself in any way regarding the =
number of m-lines in a partial offer/answer.  Nothing "breaks" if all =
m-lines are in it, does it?  The real question is if it can be greater =
than *one* m-line.  If it can, and I think it can, then I can't see how =
the exact quantity of them matters.  The only downside is you increase =
the chance of glare, I assume.

Afaik, the 'mid' attribute is there to identity which m-lines in the =
partial offer correspond to which m-lines in the full ordered set. (the =
'mid' could have even been an index number, except it's also used to =
avoid glare when adding m-lines, which a numeric index wouldn't have =
done)  So what's the issue with one or multiple m-lines?

-hadriel


From hadriel.kaplan@oracle.com  Mon Aug  5 08:40:09 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 5, 2013, at 11:31 AM, Adam Roach <adam@nostrum.com> wrote:

> On 8/5/13 10:21, Hadriel Kaplan wrote:
>> It was my understanding that the bundle-only port-0 concept requires =
and will always perform a second offer/answer round, with real port =
numbers.  Yes or no?
>=20
> The port-0 concept does *not* require a second exchange.
> There are a lot of circumstances in which a second exchange is likely =
to take place (cf. "Handshaking"), but none of these are influenced one =
way or another by the use of zero-ports on bundle-only initial offers.
>=20
> I presume you're concerned that the far end won't know which port to =
use? There's really not an issue here. These lines are interpreted by a =
bundle-aware implementation as having the same port number as the =
non-bundle-only lines that they're bundled together with. To be clear: =
you can't have a bundle that consists exclusively of bundle-only lines. =
At least one of them has to be non-bundle-only and have a real port =
number associated with it.

No, I'm concerned with usages other than WebRTC.  The question was on =
whether the bundle-only port-0 thing should be generically applicable, =
or scoped to WebRTC.  So I'm thinking of what its impact is on SIP usage =
in the real world, including on things that go bump in the night.  All =
kinds of crap happens in the middle these days in SIP networks. (and no, =
not only with SBCs)

-hadriel


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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On 8/5/13 10:39, Hadriel Kaplan wrote:
> No, I'm concerned with usages other than WebRTC. The question was on 
> whether the bundle-only port-0 thing should be generically applicable, 
> or scoped to WebRTC. So I'm thinking of what its impact is on SIP 
> usage in the real world, including on things that go bump in the 
> night. All kinds of crap happens in the middle these days in SIP 
> networks. (and no, not only with SBCs)

I don't think I'd expect bundle-only lines to work through 
extra-specification bumps in the wire. If the bumps are upgraded to work 
with bundle, great, but I know that this won't happen overnight (and, 
for many corners of the network, ever). But given that these bumps in 
some sense serve as the "other end" of the connection, a failure to get 
bundle-only lines through them is merely the design working per its 
requirements.

Remember, we're designing these lines to be fragile on purpose. That's 
the whole point of using a zero port: if we don't have end-to-end 
understanding of bundle-only, we need them to break.

/a

From christer.holmberg@ericsson.com  Mon Aug  5 09:35:37 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] UP: Can a partial offer be a full offer?
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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--_000_7594FB04B1934943A5C02806D1A2204B1C41DA7AESESSMB209erics_
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Hi,

I see no issue with multiple m- lines - just wanted to verify that it would=
 be allowed :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?

On Aug 5, 2013, at 11:17 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Note that my re-INVITE/UPDATE example is based on the assumption that one=
 can use partial offers/answers with SIP, on the wire. Not sure if that was=
 part of the agreed scope for the UP.

Definitely NOT agreed on.  Doesn't mean it won't happen in the future, but =
the scope of UP was limited to WebRTC for now.  We can debate which parts o=
f UP are generically applicable, and which not.  For SIP, this isn't the ri=
ght WG for that debate.


> But, even without that, I think the question on including all m- lines on=
 a partial offer still applies :)

I don't see why WebRTC would restrict itself in any way regarding the numbe=
r of m-lines in a partial offer/answer.  Nothing "breaks" if all m-lines ar=
e in it, does it?  The real question is if it can be greater than *one* m-l=
ine.  If it can, and I think it can, then I can't see how the exact quantit=
y of them matters.  The only downside is you increase the chance of glare, =
I assume.

Afaik, the 'mid' attribute is there to identity which m-lines in the partia=
l offer correspond to which m-lines in the full ordered set. (the 'mid' cou=
ld have even been an index number, except it's also used to avoid glare whe=
n adding m-lines, which a numeric index wouldn't have done)  So what's the =
issue with one or multiple m-lines?

-hadriel


--_000_7594FB04B1934943A5C02806D1A2204B1C41DA7AESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I see no issue with multiple m- lines - just wa=
nted to&nbsp;verify that it would be allowed :)</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
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<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] UP: Can a partial offer be a full offer?</span=
></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 5, 2013, at 11:17 AM, Christer Holmberg &lt;christer.holmberg@ericss=
on.com&gt; wrote:<br>
<br>
&gt; Note that my re-INVITE/UPDATE example is based on the assumption that =
one can use partial offers/answers with SIP, on the wire. Not sure if that =
was part of the agreed scope for the UP.<br>
<br>
Definitely NOT agreed on.&nbsp; Doesn't mean it won't happen in the future,=
 but the scope of UP was limited to WebRTC for now.&nbsp; We can debate whi=
ch parts of UP are generically applicable, and which not.&nbsp; For SIP, th=
is isn't the right WG for that debate.<br>
<br>
<br>
&gt; But, even without that, I think the question on including all m- lines=
 on a partial offer still applies :)<br>
<br>
I don't see why WebRTC would restrict itself in any way regarding the numbe=
r of m-lines in a partial offer/answer.&nbsp; Nothing &quot;breaks&quot; if=
 all m-lines are in it, does it?&nbsp; The real question is if it can be gr=
eater than *one* m-line.&nbsp; If it can, and I think it
 can, then I can't see how the exact quantity of them matters.&nbsp; The on=
ly downside is you increase the chance of glare, I assume.<br>
<br>
Afaik, the 'mid' attribute is there to identity which m-lines in the partia=
l offer correspond to which m-lines in the full ordered set. (the 'mid' cou=
ld have even been an index number, except it's also used to avoid glare whe=
n adding m-lines, which a numeric
 index wouldn't have done)&nbsp; So what's the issue with one or multiple m=
-lines?<br>
<br>
-hadriel<br>
<br>
</div>
</span></font>
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From christer.holmberg@ericsson.com  Mon Aug  5 09:48:27 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Hi,

As Adam indicated, the need to send a second offer isn't introduced by port=
 zero. Currently the second offer is always sent after the initial offer (s=
ome comments were raised regarding that, but it is a separate discussion).

Having said that, if a second offer IS sent (for whatever reason), I assume=
 that the actual bundle port value would be assigned also to the bundle-onl=
y m- lines (assuming the answerer supports bundle-only, that is)?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Adam Roach [adam@nostrum.com]
To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic@ietf=
.org]
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in=
 bundled m- lines (SDP Offers only)
On 8/5/13 10:21, Hadriel Kaplan wrote:
> It was my understanding that the bundle-only port-0 concept requires and =
will always perform a second offer/answer round, with real port numbers.  Y=
es or no?

The port-0 concept does *not* require a second exchange.

There are a lot of circumstances in which a second exchange is likely to
take place (cf. "Handshaking"), but none of these are influenced one way
or another by the use of zero-ports on bundle-only initial offers.

I presume you're concerned that the far end won't know which port to
use? There's really not an issue here. These lines are interpreted by a
bundle-aware implementation as having the same port number as the
non-bundle-only lines that they're bundled together with. To be clear:
you can't have a bundle that consists exclusively of bundle-only lines.
At least one of them has to be non-bundle-only and have a real port
number associated with it.

/a

--_000_7594FB04B1934943A5C02806D1A2204B1C41DADFESESSMB209erics_
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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">As Adam indicated,&nbsp;the need to send a seco=
nd offer isn't introduced by port zero. Currently the second offer is alway=
s sent after the initial offer (some comments were raised regarding that, b=
ut it is a separate discussion).</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Having said that, if a second offer IS sent (fo=
r whatever reason), I assume that the actual bundle port value would be ass=
igned also to the bundle-only m- lines (assuming the answerer supports bund=
le-only, that is)?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Adam Roach [adam@nostrum.com]<br>
<b>To:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmus=
ic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero u=
sage in bundled m- lines (SDP Offers only)</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 8/5/13 10:21, Hadriel Kaplan wrote:<br>
&gt; It was my understanding that the bundle-only port-0 concept requires a=
nd will always perform a second offer/answer round, with real port numbers.=
&nbsp; Yes or no?<br>
<br>
The port-0 concept does *not* require a second exchange.<br>
<br>
There are a lot of circumstances in which a second exchange is likely to <b=
r>
take place (cf. &quot;Handshaking&quot;), but none of these are influenced =
one way <br>
or another by the use of zero-ports on bundle-only initial offers.<br>
<br>
I presume you're concerned that the far end won't know which port to <br>
use? There's really not an issue here. These lines are interpreted by a <br=
>
bundle-aware implementation as having the same port number as the <br>
non-bundle-only lines that they're bundled together with. To be clear: <br>
you can't have a bundle that consists exclusively of bundle-only lines. <br=
>
At least one of them has to be non-bundle-only and have a real port <br>
number associated with it.<br>
<br>
/a<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C41DADFESESSMB209erics_--

From hadriel.kaplan@oracle.com  Mon Aug  5 09:48:59 2013
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 5, 2013, at 11:59 AM, Adam Roach <adam@nostrum.com> wrote:

> On 8/5/13 10:39, Hadriel Kaplan wrote:
>> No, I'm concerned with usages other than WebRTC. The question was on =
whether the bundle-only port-0 thing should be generically applicable, =
or scoped to WebRTC. So I'm thinking of what its impact is on SIP usage =
in the real world, including on things that go bump in the night. All =
kinds of crap happens in the middle these days in SIP networks. (and no, =
not only with SBCs)
>=20
> I don't think I'd expect bundle-only lines to work through =
extra-specification bumps in the wire. If the bumps are upgraded to work =
with bundle, great, but I know that this won't happen overnight (and, =
for many corners of the network, ever). But given that these bumps in =
some sense serve as the "other end" of the connection, a failure to get =
bundle-only lines through them is merely the design working per its =
requirements.
>=20
> Remember, we're designing these lines to be fragile on purpose. That's =
the whole point of using a zero port: if we don't have end-to-end =
understanding of bundle-only, we need them to break.

Sure, but the problem I'm worried about is when they break in weird ways =
- like they fail in the answer phase rather than offer phase; or they =
appear to succeed in SDP, but there's no media; or not enough bandwidth =
is allocated in controlled pipes; or the call gets torn down some time =
later by those bumps-in-the-wire.

-hadriel


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: Meaning of "scope is WEBRTC"
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Hi,

While cleaning up my MMUSIC notes today, I was listening to the audio recor=
ding, in order to get a clarification regarding what "scope is WEBRTC" real=
ly means.  However, unless I missed something, I didn't get a clear answer.

Does it only cover the browser API (JSEP) or does it also cover something t=
hat can be sent out on the wire (as long as it is not SIP) by a browser?

Feel free to tell me that it is a stupid question, and that the answer is v=
ery clear :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

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11pt;color:black">
<div><span style=3D"color: black; font-family: Calibri, Arial, Helvetica, s=
ans-serif; font-size: 11pt;">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">While cleaning up my MMUSIC notes today, I was =
listening to the audio recording, in order to get a clarification regarding=
 what &quot;scope is WEBRTC&quot; really means.&nbsp; However, unless I mis=
sed something, I didn't get a clear answer.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Does it only cover the browser API (JSEP)&nbsp;=
or does it also cover something that can be sent out on the wire (as long a=
s it is not SIP) by a browser?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Feel free to tell me that it is a stupid questi=
on, and that the answer is very clear :)</font></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color: black; font-family: Calibri, Arial, Helvetica, sans-s=
erif; font-size: 11pt;">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color: blue;">TouchDown</=
b> (www.nitrodesk.com)</div>
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From christer.holmberg@ericsson.com  Mon Aug  5 10:05:33 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
Thread-Index: AQHOkfuxQP+jLN773kOpD9GU/Ld9gJmG1/eK
Date: Mon, 5 Aug 2013 17:05:20 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C41D191@ESESSMB209.ericsson.se> <262469F4-1F64-4FA0-91A3-102A937E0E7D@oracle.com> <51FFC541.1060406@nostrum.com> <74024C75-90B0-421C-A1A5-5A67FB02AE1C@oracle.com> <51FFCBE8.9030601@nostrum.com>, <ACDBA9A4-BD39-49C7-A37C-8E8DFC3053B4@oracle.com>
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in	bundled m- lines (SDP Offers only)
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In SIP, unless you have control of each node between the endpoints, the onl=
y way to guarantee a "consistent break" is by using Proxy-Require.
But, as we know, that sucks, because it would mandate entities that could n=
ot care less about SDP (yes, such entities DO still exist :) to support bun=
dle-only.

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Adam Roach [adam@nostrum.com]
CC: mmusic (E-mail) [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in=
 bundled m- lines (SDP Offers only)

On Aug 5, 2013, at 11:59 AM, Adam Roach <adam@nostrum.com> wrote:

> On 8/5/13 10:39, Hadriel Kaplan wrote:
>> No, I'm concerned with usages other than WebRTC. The question was on whe=
ther the bundle-only port-0 thing should be generically applicable, or scop=
ed to WebRTC. So I'm thinking of what its impact is on SIP usage in the rea=
l world, including on things that go bump in the night. All kinds of crap h=
appens in the middle these days in SIP networks. (and no, not only with SBC=
s)
>
> I don't think I'd expect bundle-only lines to work through extra-specific=
ation bumps in the wire. If the bumps are upgraded to work with bundle, gre=
at, but I know that this won't happen overnight (and, for many corners of t=
he network, ever). But given that these bumps in some sense serve as the "o=
ther end" of the connection, a failure to get bundle-only lines through the=
m is merely the design working per its requirements.
>
> Remember, we're designing these lines to be fragile on purpose. That's th=
e whole point of using a zero port: if we don't have end-to-end understandi=
ng of bundle-only, we need them to break.

Sure, but the problem I'm worried about is when they break in weird ways - =
like they fail in the answer phase rather than offer phase; or they appear =
to succeed in SDP, but there's no media; or not enough bandwidth is allocat=
ed in controlled pipes; or the call gets torn down some time later by those=
 bumps-in-the-wire.

-hadriel

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">In SIP, unless you have control of each node between =
the endpoints, the only way to guarantee a &quot;consistent break&quot; is =
by using Proxy-Require.<br>
</span></div>
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">But, as we know, that sucks, because it would&nbsp;ma=
ndate entities that could not care less about SDP (yes, such entities DO st=
ill exist :) to support bundle-only.</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Adam Roach [adam@nostrum.com]<br>
<b>CC:</b> mmusic (E-mail) [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero u=
sage in bundled m- lines (SDP Offers only)</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 5, 2013, at 11:59 AM, Adam Roach &lt;adam@nostrum.com&gt; wrote:<br>
<br>
&gt; On 8/5/13 10:39, Hadriel Kaplan wrote:<br>
&gt;&gt; No, I'm concerned with usages other than WebRTC. The question was =
on whether the bundle-only port-0 thing should be generically applicable, o=
r scoped to WebRTC. So I'm thinking of what its impact is on SIP usage in t=
he real world, including on things that
 go bump in the night. All kinds of crap happens in the middle these days i=
n SIP networks. (and no, not only with SBCs)<br>
&gt; <br>
&gt; I don't think I'd expect bundle-only lines to work through extra-speci=
fication bumps in the wire. If the bumps are upgraded to work with bundle, =
great, but I know that this won't happen overnight (and, for many corners o=
f the network, ever). But given that
 these bumps in some sense serve as the &quot;other end&quot; of the connec=
tion, a failure to get bundle-only lines through them is merely the design =
working per its requirements.<br>
&gt; <br>
&gt; Remember, we're designing these lines to be fragile on purpose. That's=
 the whole point of using a zero port: if we don't have end-to-end understa=
nding of bundle-only, we need them to break.<br>
<br>
Sure, but the problem I'm worried about is when they break in weird ways - =
like they fail in the answer phase rather than offer phase; or they appear =
to succeed in SDP, but there's no media; or not enough bandwidth is allocat=
ed in controlled pipes; or the call
 gets torn down some time later by those bumps-in-the-wire.<br>
<br>
-hadriel<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
</div>
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From matthew@matthew.at  Mon Aug  5 12:42:40 2013
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On 8/5/2013 9:59 AM, Christer Holmberg wrote:
> Hi,
> While cleaning up my MMUSIC notes today, I was listening to the audio 
> recording, in order to get a clarification regarding what "scope is 
> WEBRTC" really means. However, unless I missed something, I didn't get 
> a clear answer.
> Does it only cover the browser API (JSEP) or does it also cover 
> something that can be sent out on the wire (as long as it is not SIP) 
> by a browser?
>

And what if it passes through a gateway that doesn't inspect the SDP and 
blindly passes it to a SIP device. What happens then?

RTCWEB used SDP so it would be compatible with SIP legacy devices, so 
certainly this will work... or something about all this has gone off the 
rails.

Matthew Kaufman

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    <div class="moz-cite-prefix">On 8/5/2013 9:59 AM, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41DB40@ESESSMB209.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div><span style="color: black; font-family: Calibri, Arial,
          Helvetica, sans-serif; font-size: 11pt;">Hi,</span></div>
      <div>&nbsp;</div>
      <div><font face="Calibri">While cleaning up my MMUSIC notes today,
          I was listening to the audio recording, in order to get a
          clarification regarding what "scope is WEBRTC" really means.&nbsp;
          However, unless I missed something, I didn't get a clear
          answer.</font></div>
      <div>&nbsp;</div>
      <div><font face="Calibri">Does it only cover the browser API
          (JSEP)&nbsp;or does it also cover something that can be sent out on
          the wire (as long as it is not SIP) by a browser?</font></div>
      &nbsp;
      <div><br>
      </div>
    </blockquote>
    <br>
    And what if it passes through a gateway that doesn't inspect the SDP
    and blindly passes it to a SIP device. What happens then?<br>
    <br>
    RTCWEB used SDP so it would be compatible with SIP legacy devices,
    so certainly this will work... or something about all this has gone
    off the rails.<br>
    <br>
    Matthew Kaufman<br>
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--------------030409010803090200040800--

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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 8/5/13 4:51 PM, Hadriel Kaplan wrote:
>
> On Jul 31, 2013, at 12:42 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> I don't understand how sendonly and recvonly reduce glare.
>> If I offer a sendonly m-line, it still must be answered. And subsequently an offer for that m-line can come from either side, possibly producing glare.
>>
>> ISTM that you should offer a sendrecv m-line iff you intend a bidirectional flow, where the two sides are related, will have common properties and lifetimes.
>>
>> There is still a problem if both sides want a bidirectional flow, but aren't agreed on who should initiate it. In that case they both might offer and glare. And there there remains the question of whether they recognize this as glare, rather than thinking they need both flows.
>
> My understanding is the concept isn't such that there isn't an offer going in both directions - there is an offer going in both directions.  But if the two offers don't "collide" in the things they change, then it's not a glare, and instead they can just accept the offer they each receive as an answer.

That isn't my understanding. My understanding was that the intent was to 
reduce the probability that there will be concurrent offers from both 
sides for the same subset of the SDP.

> So in the direction attribute case, if you send a=sendonly and the other side sent a=recvonly, there is no collision - each side can just treat the offer as an answer.  Likewise if you both send "a=sendrecv", or both send "a=inactive".

Assume that we did an initial full O/A, and m-line#1 sendonly for X and 
recvonly for Y.

Subsequently, one side, or the other, or both, may wish to make a change 
to the media section headed by m-line#1. For instance, one side may wish 
to change the codec list - perhaps drop one of the PTs from the initial 
O/A, or add a new one.

If both sides attempt this then there is still glare. One can't assume 
that one side won't ever have need to do this.

> We can even define rock-paper-scissor rules, as was done for when a stream is removed and changed at the same time (in that case removal trumps changing).  For example, we can make it so that an "a=inactive" trumps all other directions; so regardless of what direction attribute you sent, if you receive an "a=inactive", that's the new state.

Having one side change the directionality is just one of the multitude 
of changes that might come up. It probably isn't the most significant 
one. The likelihood if it will be reduced if sendrecv is never used.

> We could even define a lookup table for it, like this:
>
> Sent Offer  Rcvd Offer      Local state
> --------------------------------------
> sendrecv    sendrecv        sendrecv
> sendrecv    sendonly        recvonly
> sendrecv    recvonly        sendonly
> sendrecv    inactive        inactive
> sendonly    sendrecv        sendonly
> sendonly    sendonly        inactive
> sendonly    recvonly        sendonly
> sendonly    inactive        inactive
> recvonly    sendrecv        recvonly
> recvonly    sendonly        recvonly
> recvonly    recvonly        inactive
> recvonly    inactive        inactive
> inactive    sendrecv        inactive
> inactive    sendonly        inactive
> inactive    recvonly        inactive
> inactive    inactive        inactive

If nothing else was changed, then there could be some advantage to this 
in reducing glare. But it doesn't help with changing anything else about 
the media section. And while similar rules might be worked out for some 
of those things, I think it will be hard. (Similar in scale and 
complexity to draft-nandakumar-mmusic-sdp-mux-attributes.)

I think it is probably true that using sendonly and recvonly when adding 
*new* m-lines may work out better than using sendrecv and counting on 
the other side to pair the reverse direction. But even then it will only 
matter with certain usage scenarios.

IMO the key to using sendrecv is not to avoid it entirely, but rather to 
figure out when it makes sense. I think that is when you *want* "fate 
sharing" between the paired streams.

	Thanks,
	Paul


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On Aug 5, 2013, at 5:25 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

>> My understanding is the concept isn't such that there isn't an offer =
going in both directions - there is an offer going in both directions.  =
But if the two offers don't "collide" in the things they change, then =
it's not a glare, and instead they can just accept the offer they each =
receive as an answer.
>=20
> That isn't my understanding. My understanding was that the intent was =
to reduce the probability that there will be concurrent offers from both =
sides for the same subset of the SDP.

Huh.  Yup you could be right.  Heck if I know - I'm only going by what I =
thought Martin said at the mic in the meeting.  I thought he said it =
would be useful to reduce glare if all that changed was direction =
attribute on both sides. (like you both hit "hold" or "un-hold" =
whatever)  But I could easily be misunderstanding what he said.

-hadriel


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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On Aug 5, 2013, at 11:01 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> =20
> So, you are basically saying that both endpoints can send partial =
offers for the SAME m- line, WITHOUT "triggering" glare, if there is no =
"state collision"???
> =20
> My understanding of the UP (unified plan) was that you would still get =
glare if both endpoints simultaneously send a partial offer for the same =
m- line.=20

Currently in the UP draft, that's true.  I thought Martin's point at the =
mic was we could do a bit better for specific attribute changes, such as =
direction, of the same m-line.  Or I could be reading something into his =
words that he didn't intend.

I eagerly await his email to enlighten us. :)

-hadriel


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On Aug 5, 2013, at 5:25 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Having one side change the directionality is just one of the multitude =
of changes that might come up. It probably isn't the most significant =
one. The likelihood if it will be reduced if sendrecv is never used.

Afaict in *SIP* the most common causes of glare are (1) transport =
addr:port changes and (2) directionality attribute changes.  Usually =
early in calls, like in the first few seconds; sometimes in the middle =
of calls, if both sides press a hold button at the same time for =
example.[1]

But yeah since the use-case that drove UP is basically hangouts, Martin =
probably did mean something completely unrelated to what I was thinking.

-hadriel
[1] It's hard to generalize them into "most common causes", because =
there're plenty of other cases too; and probably different people see =
different cases more/less frequently.


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To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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Hi,

If you use separate m- lines per media direction, there MAY be a smaller ri=
sk that both endpoints try to modify the same m- line. Maybe that is what h=
e was talking about :)

Regards,

Christer

-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Hadriel Kaplan
L=E4hetetty: 6. elokuuta 2013 1:45
Vastaanottaja: Paul Kyzivat
Kopio: mmusic@ietf.org
Aihe: Re: [MMUSIC] unified plan glareless atomicity improvement


On Aug 5, 2013, at 5:25 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Having one side change the directionality is just one of the multitude of=
 changes that might come up. It probably isn't the most significant one. Th=
e likelihood if it will be reduced if sendrecv is never used.

Afaict in *SIP* the most common causes of glare are (1) transport addr:port=
 changes and (2) directionality attribute changes.  Usually early in calls,=
 like in the first few seconds; sometimes in the middle of calls, if both s=
ides press a hold button at the same time for example.[1]

But yeah since the use-case that drove UP is basically hangouts, Martin pro=
bably did mean something completely unrelated to what I was thinking.

-hadriel
[1] It's hard to generalize them into "most common causes", because there'r=
e plenty of other cases too; and probably different people see different ca=
ses more/less frequently.

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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I don't really understand the "only for WebRTC" agreement.
I do see how defining extensions to O/A might be done only for WebRTC 
and not for 3264. (I don't *agree*, but I do *understand* that.)

But BUNDLE is a change to SDP, and SDP extensions are required to 
specify how they interact with O/A. I don't think it makes any sense to 
say that such changes only apply to WebRTC. They should apply to 
anything that chooses to claim support to the BUNDLE draft.

	Thanks,
	Paul

On 8/5/13 5:39 PM, Hadriel Kaplan wrote:
>
> On Aug 5, 2013, at 11:31 AM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/5/13 10:21, Hadriel Kaplan wrote:
>>> It was my understanding that the bundle-only port-0 concept requires and will always perform a second offer/answer round, with real port numbers.  Yes or no?
>>
>> The port-0 concept does *not* require a second exchange.
>> There are a lot of circumstances in which a second exchange is likely to take place (cf. "Handshaking"), but none of these are influenced one way or another by the use of zero-ports on bundle-only initial offers.
>>
>> I presume you're concerned that the far end won't know which port to use? There's really not an issue here. These lines are interpreted by a bundle-aware implementation as having the same port number as the non-bundle-only lines that they're bundled together with. To be clear: you can't have a bundle that consists exclusively of bundle-only lines. At least one of them has to be non-bundle-only and have a real port number associated with it.
>
> No, I'm concerned with usages other than WebRTC.  The question was on whether the bundle-only port-0 thing should be generically applicable, or scoped to WebRTC.  So I'm thinking of what its impact is on SIP usage in the real world, including on things that go bump in the night.  All kinds of crap happens in the middle these days in SIP networks. (and no, not only with SBCs)
>
> -hadriel
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From christer.holmberg@ericsson.com  Tue Aug  6 08:59:50 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, mmusic_ietf.org <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Hi Paul,

Personally, I intend to focus mainly on the technical parts - others will t=
hen have to tell me what/how to write in order to scope some feature, if ne=
eded :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: mmusic@ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in=
 bundled m- lines (SDP Offers only)
I don't really understand the "only for WebRTC" agreement.
I do see how defining extensions to O/A might be done only for WebRTC
and not for 3264. (I don't *agree*, but I do *understand* that.)

But BUNDLE is a change to SDP, and SDP extensions are required to
specify how they interact with O/A. I don't think it makes any sense to
say that such changes only apply to WebRTC. They should apply to
anything that chooses to claim support to the BUNDLE draft.

        Thanks,
        Paul

On 8/5/13 5:39 PM, Hadriel Kaplan wrote:
>
> On Aug 5, 2013, at 11:31 AM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/5/13 10:21, Hadriel Kaplan wrote:
>>> It was my understanding that the bundle-only port-0 concept requires an=
d will always perform a second offer/answer round, with real port numbers. =
 Yes or no?
>>
>> The port-0 concept does *not* require a second exchange.
>> There are a lot of circumstances in which a second exchange is likely to=
 take place (cf. "Handshaking"), but none of these are influenced one way o=
r another by the use of zero-ports on bundle-only initial offers.
>>
>> I presume you're concerned that the far end won't know which port to use=
? There's really not an issue here. These lines are interpreted by a bundle=
-aware implementation as having the same port number as the non-bundle-only=
 lines that they're bundled together with. To be clear: you can't have a bu=
ndle that consists exclusively of bundle-only lines. At least one of them h=
as to be non-bundle-only and have a real port number associated with it.
>
> No, I'm concerned with usages other than WebRTC.  The question was on whe=
ther the bundle-only port-0 thing should be generically applicable, or scop=
ed to WebRTC.  So I'm thinking of what its impact is on SIP usage in the re=
al world, including on things that go bump in the night.  All kinds of crap=
 happens in the middle these days in SIP networks. (and no, not only with S=
BCs)
>
> -hadriel
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Paul,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Personally, I&nbsp;intend to focus mainly on th=
e technical parts - others will then have to tell me what/how to write in o=
rder to scope some feature, if needed :)</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> mmusic@ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero u=
sage in bundled m- lines (SDP Offers only)</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">I don't really understand the &quot;only for WebRT=
C&quot; agreement.<br>
I do see how defining extensions to O/A might be done only for WebRTC <br>
and not for 3264. (I don't *agree*, but I do *understand* that.)<br>
<br>
But BUNDLE is a change to SDP, and SDP extensions are required to <br>
specify how they interact with O/A. I don't think it makes any sense to <br=
>
say that such changes only apply to WebRTC. They should apply to <br>
anything that chooses to claim support to the BUNDLE draft.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
On 8/5/13 5:39 PM, Hadriel Kaplan wrote:<br>
&gt;<br>
&gt; On Aug 5, 2013, at 11:31 AM, Adam Roach &lt;adam@nostrum.com&gt; wrote=
:<br>
&gt;<br>
&gt;&gt; On 8/5/13 10:21, Hadriel Kaplan wrote:<br>
&gt;&gt;&gt; It was my understanding that the bundle-only port-0 concept re=
quires and will always perform a second offer/answer round, with real port =
numbers.&nbsp; Yes or no?<br>
&gt;&gt;<br>
&gt;&gt; The port-0 concept does *not* require a second exchange.<br>
&gt;&gt; There are a lot of circumstances in which a second exchange is lik=
ely to take place (cf. &quot;Handshaking&quot;), but none of these are infl=
uenced one way or another by the use of zero-ports on bundle-only initial o=
ffers.<br>
&gt;&gt;<br>
&gt;&gt; I presume you're concerned that the far end won't know which port =
to use? There's really not an issue here. These lines are interpreted by a =
bundle-aware implementation as having the same port number as the non-bundl=
e-only lines that they're bundled together
 with. To be clear: you can't have a bundle that consists exclusively of bu=
ndle-only lines. At least one of them has to be non-bundle-only and have a =
real port number associated with it.<br>
&gt;<br>
&gt; No, I'm concerned with usages other than WebRTC.&nbsp; The question wa=
s on whether the bundle-only port-0 thing should be generically applicable,=
 or scoped to WebRTC.&nbsp; So I'm thinking of what its impact is on SIP us=
age in the real world, including on things that
 go bump in the night.&nbsp; All kinds of crap happens in the middle these =
days in SIP networks. (and no, not only with SBCs)<br>
&gt;<br>
&gt; -hadriel<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
</div>
</span></font>
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From christer.holmberg@ericsson.com  Tue Aug  6 09:14:36 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Hi,

It has been indicated that POFs (partial offers) and PANs (partial answers)=
 form a stand-alone mechanism, instead of being an extension of RFC3264.

Now, assuming that is true, we still need to describe the interaction betwe=
en POFs, PANs and 3264 Offer/Answers.

Because, if I send a POF, and then receive a 3264 Offer (including the same=
 m- line as the POF),  there is a glare and I need to reject the 3264 Offer=
.

In the same way, if I send a 3264 Offer, and receive a POF (including the s=
ame m- line as the 3264 Offer), there is a glare and I need to reject the P=
OF.

I ASSUME the above is clear to everyone, including the authors of the UP, b=
ut IF someone has a different opinion I'd like to hear about it :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"font-family:Calibri, Arial, Helvetica, sans-serif;font-size:=
11pt;color:black">
<span style=3D"color: black; font-family: Calibri, Arial, Helvetica, sans-s=
erif; font-size: 11pt;">
<div><br>
Hi,</div>
<div>&nbsp;</div>
<div>It has been indicated that POFs (partial offers) and PANs (partial ans=
wers)&nbsp;form a stand-alone mechanism,&nbsp;instead of being&nbsp;an exte=
nsion of RFC3264.</div>
<div><br>
Now, assuming that is true, we still need to describe the interaction betwe=
en POFs, PANs and 3264 Offer/Answers.</div>
<div>&nbsp;</div>
<div>Because, if I send&nbsp;a POF, and then receive a 3264 Offer (includin=
g the same m- line as the POF), &nbsp;there is a glare and I need to reject=
 the 3264 Offer.</div>
<div>&nbsp;</div>
<div>In the same way, if I send a 3264 Offer, and receive&nbsp;a POF (inclu=
ding the same m- line as the 3264 Offer), there is a glare and I need to re=
ject the POF.</div>
<div>&nbsp;</div>
<div>I ASSUME the above is clear to everyone, including the authors of the =
UP, but IF someone has a different opinion I'd like to hear about it :)</di=
v>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color: blue;">TouchDown</=
b> (www.nitrodesk.com)</div>
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From pkyzivat@alum.mit.edu  Tue Aug  6 09:31:25 2013
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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I agree with Christer.

I wonder if Adam is assuming that the Offers, Answers, POFs, and PANs 
used in WebRTC are all serialized over a reliable channel? That would 
simplify things because it will ensure how they relate to one another, 
though it doesn't prevent the glare Christer describes.

If this is totally discarding 3264, then I *guess* you could define that 
O/A is only used once, and after that changes are made exclusively with 
POFs and PANs. But I don't have the sense that is being proposed.

IMO it makes much more sense to analyze this as an extension to 3264.

I also think these POFs and PANs should have the o-line to confirm what 
prior SDP the new one updates. (It is more important than it is with 
regular O/A.)

	Thanks,
	Paul

On 8/6/13 6:14 PM, Christer Holmberg wrote:
>
> Hi,
> It has been indicated that POFs (partial offers) and PANs (partial
> answers) form a stand-alone mechanism, instead of being an extension of
> RFC3264.
>
> Now, assuming that is true, we still need to describe the interaction
> between POFs, PANs and 3264 Offer/Answers.
> Because, if I send a POF, and then receive a 3264 Offer (including the
> same m- line as the POF),  there is a glare and I need to reject the
> 3264 Offer.
> In the same way, if I send a 3264 Offer, and receive a POF (including
> the same m- line as the 3264 Offer), there is a glare and I need to
> reject the POF.
> I ASSUME the above is clear to everyone, including the authors of the
> UP, but IF someone has a different opinion I'd like to hear about it :)
> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From christer.holmberg@ericsson.com  Tue Aug  6 11:02:53 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, mmusic_ietf.org <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Of course, in some cases (e.g. in JSEP) there might not even be a need to d=
istinguish between POFs/PANs and 3264 Offer/Answers.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: mmusic@ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answer=
s
I agree with Christer.

I wonder if Adam is assuming that the Offers, Answers, POFs, and PANs
used in WebRTC are all serialized over a reliable channel? That would
simplify things because it will ensure how they relate to one another,
though it doesn't prevent the glare Christer describes.

If this is totally discarding 3264, then I *guess* you could define that
O/A is only used once, and after that changes are made exclusively with
POFs and PANs. But I don't have the sense that is being proposed.

IMO it makes much more sense to analyze this as an extension to 3264.

I also think these POFs and PANs should have the o-line to confirm what
prior SDP the new one updates. (It is more important than it is with
regular O/A.)

        Thanks,
        Paul

On 8/6/13 6:14 PM, Christer Holmberg wrote:
>
> Hi,
> It has been indicated that POFs (partial offers) and PANs (partial
> answers) form a stand-alone mechanism, instead of being an extension of
> RFC3264.
>
> Now, assuming that is true, we still need to describe the interaction
> between POFs, PANs and 3264 Offer/Answers.
> Because, if I send a POF, and then receive a 3264 Offer (including the
> same m- line as the POF),  there is a glare and I need to reject the
> 3264 Offer.
> In the same way, if I send a 3264 Offer, and receive a POF (including
> the same m- line as the 3264 Offer), there is a glare and I need to
> reject the POF.
> I ASSUME the above is clear to everyone, including the authors of the
> UP, but IF someone has a different opinion I'd like to hear about it :)
> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com<http://www.nit=
rodesk.com>)
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Of course, in some cases (e.g. in JSEP) there might n=
ot even be a need to distinguish between POFs/PANs and 3264 Offer/Answers.
</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> mmusic@ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer=
/Answers</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">I agree with Christer.<br>
<br>
I wonder if Adam is assuming that the Offers, Answers, POFs, and PANs <br>
used in WebRTC are all serialized over a reliable channel? That would <br>
simplify things because it will ensure how they relate to one another, <br>
though it doesn't prevent the glare Christer describes.<br>
<br>
If this is totally discarding 3264, then I *guess* you could define that <b=
r>
O/A is only used once, and after that changes are made exclusively with <br=
>
POFs and PANs. But I don't have the sense that is being proposed.<br>
<br>
IMO it makes much more sense to analyze this as an extension to 3264.<br>
<br>
I also think these POFs and PANs should have the o-line to confirm what <br=
>
prior SDP the new one updates. (It is more important than it is with <br>
regular O/A.)<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
On 8/6/13 6:14 PM, Christer Holmberg wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt; It has been indicated that POFs (partial offers) and PANs (partial<br>
&gt; answers) form a stand-alone mechanism, instead of being an extension o=
f<br>
&gt; RFC3264.<br>
&gt;<br>
&gt; Now, assuming that is true, we still need to describe the interaction<=
br>
&gt; between POFs, PANs and 3264 Offer/Answers.<br>
&gt; Because, if I send a POF, and then receive a 3264 Offer (including the=
<br>
&gt; same m- line as the POF),&nbsp; there is a glare and I need to reject =
the<br>
&gt; 3264 Offer.<br>
&gt; In the same way, if I send a 3264 Offer, and receive a POF (including<=
br>
&gt; the same m- line as the 3264 Offer), there is a glare and I need to<br=
>
&gt; reject the POF.<br>
&gt; I ASSUME the above is clear to everyone, including the authors of the<=
br>
&gt; UP, but IF someone has a different opinion I'd like to hear about it :=
)<br>
&gt; Regards,<br>
&gt; Christer<br>
&gt;<br>
&gt; Sent from */Windows/* using *TouchDown* (<a href=3D"http://www.nitrode=
sk.com">www.nitrodesk.com</a>)<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
</div>
</span></font>
</body>
</html>

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On 8/6/13 11:14, Christer Holmberg wrote:
> It has been indicated that POFs (partial offers) and PANs (partial 
> answers) form a stand-alone mechanism, instead of being an extension 
> of RFC3264.
>
> Now, assuming that is true, we still need to describe the interaction 
> between POFs, PANs and 3264 Offer/Answers.
> Because, if I send a POF, and then receive a 3264 Offer (including the 
> same m- line as the POF),  there is a glare and I need to reject the 
> 3264 Offer.
> In the same way, if I send a 3264 Offer, and receive a POF (including 
> the same m- line as the 3264 Offer), there is a glare and I need to 
> reject the POF.
> I ASSUME the above is clear to everyone, including the authors of the 
> UP, but IF someone has a different opinion I'd like to hear about it :)
>

That is certainly what I had been assuming. Of course, if the POF is a 
new m-line (that is, it doesn't appear in the offer coming the other 
way), then there is no glare condition.

/a

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    <div class="moz-cite-prefix">On 8/6/13 11:14, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41EC88@ESESSMB209.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <span style="color: black; font-family: Calibri, Arial, Helvetica,
        sans-serif; font-size: 11pt;">
        <div>&nbsp;</div>
        <div>It has been indicated that POFs (partial offers) and PANs
          (partial answers)&nbsp;form a stand-alone mechanism,&nbsp;instead of
          being&nbsp;an extension of RFC3264.</div>
        <div><br>
          Now, assuming that is true, we still need to describe the
          interaction between POFs, PANs and 3264 Offer/Answers.</div>
        <div>&nbsp;</div>
        <div>Because, if I send&nbsp;a POF, and then receive a 3264 Offer
          (including the same m- line as the POF), &nbsp;there is a glare and
          I need to reject the 3264 Offer.</div>
        <div>&nbsp;</div>
        <div>In the same way, if I send a 3264 Offer, and receive&nbsp;a POF
          (including the same m- line as the 3264 Offer), there is a
          glare and I need to reject the POF.</div>
        <div>&nbsp;</div>
        <div>I ASSUME the above is clear to everyone, including the
          authors of the UP, but IF someone has a different opinion I'd
          like to hear about it :)</div>
        &nbsp;
        <div><br>
        </div>
      </span></blockquote>
    <br>
    That is certainly what I had been assuming. Of course, if the POF is
    a new m-line (that is, it doesn't appear in the offer coming the
    other way), then there is no glare condition.<br>
    <br>
    /a<br>
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--------------050401090107020208060904--

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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Christer said:=20
=20
=0A=
"While cleaning up my MMUSIC notes today=2C I was listening to the audio re=
cording=2C in order to get a clarification regarding what "scope is WEBRTC"=
 really means.  However=2C unless I missed something=2C I didn't get a clea=
r answer.=0A=
 =0A=
Does it only cover the browser API (JSEP) or does it also cover something t=
hat can be sent out on the wire (as long as it is not SIP) by a browser?=0A=
 =0A=
Feel free to tell me that it is a stupid question=2C and that the answer is=
 very clear :)" [BA]  I actually think it is a good question :) One interpr=
etation of "scope is WebRTC"  is that the  SDP in "Unified Plan" applies to=
 use within  the JSEP API.    As a result=2C one might expect the SDP used =
in createOffer=2C setLocalDescription and setRemoteDescription to resemble =
that in the "Unified Plan" document.   Since signaling is out of scope for =
WebRTC=2C applications that use the JSEP API might or might not use the "Un=
ified Plan" SDP in signaling on the wire.   However=2C "Unified Plan" inclu=
des normative references to published SDP specifications that were develope=
d for general use=2C and the adoption of "Unified Plan" (or even an applica=
bility statement within the "Unified Plan" document) doesn't change the app=
licability of those documents.  Also=2C there may be some specifications th=
at are part of the "Unified Plan" workload that may have applicability beyo=
nd WebRTC (e.g. a=3Dgroup: SIMULCAST)  Ultimately=2C I am expecting the wor=
k involved in "Unified Plan" to span multiple documents=2C and to clarify t=
he applicability=2C  those individual documents will need to explain what t=
hey apply to (just WebRTC use or more general use).  		 	   		  =

--_7e5baefb-478a-4e6e-8571-420b7a7708af_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>Christer said: <BR>&nbsp=3B<BR><=
div>=0A=
<div><font face=3D"Calibri">"While cleaning up my MMUSIC notes today=2C I w=
as listening to the audio recording=2C in order to get a clarification rega=
rding what "scope is WEBRTC" really means.&nbsp=3B However=2C unless I miss=
ed something=2C I didn't get a clear answer.</font></div>=0A=
<div><font face=3D"Calibri"></font>&nbsp=3B</div>=0A=
<div><font face=3D"Calibri">Does it only cover the browser API (JSEP)&nbsp=
=3Bor does it also cover something that can be sent out on the wire (as lon=
g as it is not SIP) by a browser?</font></div>=0A=
<div><font face=3D"Calibri"></font>&nbsp=3B</div>=0A=
<div><font face=3D"Calibri">Feel free to tell me that it is a stupid questi=
on=2C and that the answer is very clear :)"</font></div><div>&nbsp=3B</div>=
<div>[BA]&nbsp=3B I actually think it is a good question :)</div><div>&nbsp=
=3B</div><div>One interpretation of&nbsp=3B"scope is WebRTC"&nbsp=3B&nbsp=
=3Bis that the &nbsp=3BSDP in "Unified Plan"&nbsp=3Bapplies to use within &=
nbsp=3Bthe JSEP API. &nbsp=3B&nbsp=3B&nbsp=3BAs a result=2C one might&nbsp=
=3Bexpect the SDP used in createOffer=2C setLocalDescription and setRemoteD=
escription to resemble that in the "Unified Plan" document.&nbsp=3B&nbsp=3B=
 Since signaling is out of scope for WebRTC=2C applications that use the JS=
EP API might or might not use the&nbsp=3B"Unified Plan" SDP in signaling on=
 the wire.&nbsp=3B </div><div>&nbsp=3B</div><div>However=2C "Unified Plan"&=
nbsp=3Bincludes normative references&nbsp=3Bto&nbsp=3Bpublished SDP specifi=
cations that were developed for general use=2C and the adoption of "Unified=
 Plan" (or even an applicability statement&nbsp=3Bwithin the&nbsp=3B"Unifie=
d Plan" document) doesn't change the applicability of those documents. </di=
v><div>&nbsp=3B</div><div>Also=2C there may be some specifications that are=
 part of the "Unified Plan" workload that may have applicability beyond Web=
RTC (e.g. a=3Dgroup: SIMULCAST) </div><div>&nbsp=3B</div><div>Ultimately=2C=
 I am expecting the work involved in "Unified Plan" to span multiple docume=
nts=2C and to clarify the applicability=2C&nbsp=3B those individual documen=
ts will need to explain what they apply to (just WebRTC use or more general=
 use).&nbsp=3B</div></div> 		 	   		  </div></body>
</html>=

--_7e5baefb-478a-4e6e-8571-420b7a7708af_--

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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 5, 2013, at 1:05 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> In SIP, unless you have control of each node between the endpoints, =
the only way to guarantee a "consistent break" is by using =
Proxy-Require.
> But, as we know, that sucks, because it would mandate entities that =
could not care less about SDP (yes, such entities DO still exist :) to =
support bundle-only.
> Regards,=20
> Christer

Technically, I think you'd have to put something in both Proxy-Require =
and Require.  But that would make it virtually DOA, from a practical =
perspective.  It would fail constantly.  I realize that's kinda the =
point, to fail early, but failing all the time probably isn't the point.

The other challenge is it's not just about SBCs, B2BUAs, app servers, =
etc.  It's also about the lovely "ALGs" in NATs, CGNs, and Firewalls, =
and the passive inspecting things made by certain popular router vendors =
- you know, all the things that ostensibly prevented us from adding two =
bytes for an RTP shim header to mux/demux, even though somehow we can =
now add more bytes as RTP extension headers instead - those things don't =
understand or involve themselves with Proxy-Require/Require, as far as I =
can tell.

-hadriel


> -----Original Message-----
> From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> To: Adam Roach [adam@nostrum.com]
> CC: mmusic (E-mail) [mmusic@ietf.org]
> Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero =
usage in bundled m- lines (SDP Offers only)
>=20
> On Aug 5, 2013, at 11:59 AM, Adam Roach <adam@nostrum.com> wrote:
>=20
> > On 8/5/13 10:39, Hadriel Kaplan wrote:
> >> No, I'm concerned with usages other than WebRTC. The question was =
on whether the bundle-only port-0 thing should be generically =
applicable, or scoped to WebRTC. So I'm thinking of what its impact is =
on SIP usage in the real world, including on things that go bump in the =
night. All kinds of crap happens in the middle these days in SIP =
networks. (and no, not only with SBCs)
> >=20
> > I don't think I'd expect bundle-only lines to work through =
extra-specification bumps in the wire. If the bumps are upgraded to work =
with bundle, great, but I know that this won't happen overnight (and, =
for many corners of the network, ever). But given that these bumps in =
some sense serve as the "other end" of the connection, a failure to get =
bundle-only lines through them is merely the design working per its =
requirements.
> >=20
> > Remember, we're designing these lines to be fragile on purpose. =
That's the whole point of using a zero port: if we don't have end-to-end =
understanding of bundle-only, we need them to break.
>=20
> Sure, but the problem I'm worried about is when they break in weird =
ways - like they fail in the answer phase rather than offer phase; or =
they appear to succeed in SDP, but there's no media; or not enough =
bandwidth is allocated in controlled pipes; or the call gets torn down =
some time later by those bumps-in-the-wire.
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From martin.thomson@gmail.com  Tue Aug  6 22:44:09 2013
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Date: Wed, 7 Aug 2013 07:44:07 +0200
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 6 August 2013 00:55, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> If you use separate m- lines per media direction, there MAY be a smaller risk that both endpoints try to modify the same m- line. Maybe that is what he was talking about :)

That's actually all I was suggesting.  But the wild speculation that
ensued was quite interesting nonetheless.

I believe that most of what has been said on this thread is also true.
 Yes, you can probably reduce glare further by defining some specific
rules (reject beats any other change, etc...).  Yes, you can devise
rules for other stuff.  Yes, it's probably not worth it for a lot of
those things anyway.  Yes, changing parameters that affect multiple
m-lines, like transport parameters, is probably going to cause
unavoidable wide-scale glare.  Etc..

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
Thread-Index: Ac6SwAaH+8B2wVyxQFSEGu6rKTxpcQALCkmAABMWT0A=
Date: Wed, 7 Aug 2013 06:39:34 +0000
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Hi,
=A0
>> It has been indicated that POFs (partial offers) and PANs (partial answe=
rs)=A0form a stand-alone mechanism,=A0instead of being=A0an extension of RF=
C3264.
>>
>> Now, assuming that is true, we still need to describe the interaction be=
tween POFs, PANs and 3264 Offer/Answers.
>>
>> Because, if I send=A0a POF, and then receive a 3264 Offer (including the=
 same m- line as the POF), =A0there is a glare and I need to reject the 326=
4 Offer.
>>
>> In the same way, if I send a 3264 Offer, and receive=A0a POF (including =
the same m- line as the 3264 Offer), there is a glare and I need to reject =
the POF.
>>
>> I ASSUME the above is clear to everyone, including the authors of the UP=
, but IF someone has a different opinion I'd like to hear about it :)
>=A0
> That is certainly what I had been assuming. Of course, if the POF is a ne=
w m-line (that is, it doesn't appear in the offer coming the other way), th=
en there is no glare condition.

Correct - assuming that the addition of the new m- lines does not indirectl=
y affect values etc associated with existing m- lines.

Regards,

Christer


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, Christer Holmberg <christer.holmberg@ericsson.com>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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--_000_949EF20990823C4C85C18D59AA11AD8B07D89DFR712WXCHMBA11zeu_
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As one of those that used the term, my intention was that the initial decis=
ion was to agree to define what RTCWEB needed.

As each dependent working group looks at the extensions they need to do, th=
en they need to make the decision as to whether a wider scope of possible, =
desirable, or dangerous.

So the BUNDLE status at the moment is that any changes for the unified plan=
 should work for RTCWEB usage, but not for general. It would probably make =
sense to make it more generally applicable, but that is the next step in th=
e decision process, and may need some other technical decisions before it h=
appens.

Keith

________________________________
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: 07 August 2013 00:50
To: Christer Holmberg; mmusic
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"

Christer said:

"While cleaning up my MMUSIC notes today, I was listening to the audio reco=
rding, in order to get a clarification regarding what "scope is WEBRTC" rea=
lly means.  However, unless I missed something, I didn't get a clear answer=
.

Does it only cover the browser API (JSEP) or does it also cover something t=
hat can be sent out on the wire (as long as it is not SIP) by a browser?

Feel free to tell me that it is a stupid question, and that the answer is v=
ery clear :)"

[BA]  I actually think it is a good question :)

One interpretation of "scope is WebRTC"  is that the  SDP in "Unified Plan"=
 applies to use within  the JSEP API.    As a result, one might expect the =
SDP used in createOffer, setLocalDescription and setRemoteDescription to re=
semble that in the "Unified Plan" document.   Since signaling is out of sco=
pe for WebRTC, applications that use the JSEP API might or might not use th=
e "Unified Plan" SDP in signaling on the wire.

However, "Unified Plan" includes normative references to published SDP spec=
ifications that were developed for general use, and the adoption of "Unifie=
d Plan" (or even an applicability statement within the "Unified Plan" docum=
ent) doesn't change the applicability of those documents.

Also, there may be some specifications that are part of the "Unified Plan" =
workload that may have applicability beyond WebRTC (e.g. a=3Dgroup: SIMULCA=
ST)

Ultimately, I am expecting the work involved in "Unified Plan" to span mult=
iple documents, and to clarify the applicability,  those individual documen=
ts will need to explain what they apply to (just WebRTC use or more general=
 use).

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<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">As one of those that used the term, my=
 intention was that the initial decision was to agree to define what RTCWEB=
 needed.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">As each dependent working group looks =
at the extensions they need to do, then they need to make the decision as t=
o whether a wider scope
 of possible, desirable, or dangerous.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">So the BUNDLE status at the moment is =
that any changes for the unified plan should work for RTCWEB usage, but not=
 for general. It would
 probably make sense to make it more generally applicable, but that is the =
next step in the decision process, and may need some other technical decisi=
ons before it happens.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Keith<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span lang=3D"EN-US" style=3D"font-siz=
e:12.0pt">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN=
-US" style=3D"font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</=
span></font></b><font size=3D"2" face=3D"Tahoma"><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;font-family:Tahoma"> mmusic-bounces@ietf.org
 [mailto:mmusic-bounces@ietf.org] <b><span style=3D"font-weight:bold">On Be=
half Of </span>
</b>Bernard Aboba<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> 07 August 2013 00:50<b=
r>
<b><span style=3D"font-weight:bold">To:</span></b> Christer Holmberg; mmusi=
c<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [MMUSIC] UP: Me=
aning of &quot;scope is WEBRTC&quot;</span></font><span lang=3D"EN-US"><o:p=
></o:p></span></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">Christer said:
<br>
&nbsp;<o:p></o:p></span></font></p>
<div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&quot;While cleaning up my MMUSIC notes today, I was l=
istening to the audio recording, in order to get a clarification regarding =
what &quot;scope is WEBRTC&quot; really
 means.&nbsp; However, unless I missed something, I didn't get a clear answ=
er.<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">Does it only cover the browser
<st1:stockticker w:st=3D"on">API</st1:stockticker> (JSEP)&nbsp;or does it a=
lso cover something that can be sent out on the wire (as long as it is not =
SIP) by a browser?<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">Feel free to tell me that it is a stupid question, and=
 that the answer is very clear :)&quot;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">[BA]&nbsp; I actually think it is a good question :)<o=
:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">One interpretation of&nbsp;&quot;scope is WebRTC&quot;=
&nbsp;&nbsp;is that the &nbsp;SDP in &quot;Unified Plan&quot;&nbsp;applies =
to use within &nbsp;the JSEP
<st1:stockticker w:st=3D"on">API</st1:stockticker>. &nbsp;&nbsp;&nbsp;As a =
result, one might&nbsp;expect the SDP used in createOffer, setLocalDescript=
ion and setRemoteDescription to resemble that in the &quot;Unified Plan&quo=
t; document.&nbsp;&nbsp; Since signaling is out of scope for WebRTC, applic=
ations
 that use the JSEP <st1:stockticker w:st=3D"on">API</st1:stockticker> might=
 or might not use the&nbsp;&quot;Unified Plan&quot; SDP in signaling on the=
 wire.&nbsp;
<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">However, &quot;Unified Plan&quot;&nbsp;includes normat=
ive references&nbsp;to&nbsp;published SDP specifications that were develope=
d for general use, and the adoption of &quot;Unified Plan&quot;
 (or even an applicability statement&nbsp;within the&nbsp;&quot;Unified Pla=
n&quot; document) doesn't change the applicability of those documents.
<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">Also, there may be some specifications that are part o=
f the &quot;Unified Plan&quot; workload that may have applicability beyond =
WebRTC (e.g. a=3Dgroup: SIMULCAST)
<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Calibri"><span style=3D"fon=
t-size:12.0pt;
font-family:Calibri">Ultimately, I am expecting the work involved in &quot;=
Unified Plan&quot; to span multiple documents, and to clarify the applicabi=
lity,&nbsp; those individual documents
 will need to explain what they apply to (just WebRTC use or more general u=
se).&nbsp;<o:p></o:p></span></font></p>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_949EF20990823C4C85C18D59AA11AD8B07D89DFR712WXCHMBA11zeu_--

From pkyzivat@alum.mit.edu  Wed Aug  7 06:27:29 2013
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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What about adding a new m-line into a bundle?
IIUC this is likely to be one of the most important use cases.
Can this all be done with POF and PAN?

	Thanks,
	Paul

On 8/7/13 8:39 AM, Christer Holmberg wrote:
> Hi,
>
>>> It has been indicated that POFs (partial offers) and PANs (partial answers) form a stand-alone mechanism, instead of being an extension of RFC3264.
>>>
>>> Now, assuming that is true, we still need to describe the interaction between POFs, PANs and 3264 Offer/Answers.
>>>
>>> Because, if I send a POF, and then receive a 3264 Offer (including the same m- line as the POF),  there is a glare and I need to reject the 3264 Offer.
>>>
>>> In the same way, if I send a 3264 Offer, and receive a POF (including the same m- line as the 3264 Offer), there is a glare and I need to reject the POF.
>>>
>>> I ASSUME the above is clear to everyone, including the authors of the UP, but IF someone has a different opinion I'd like to hear about it :)
>>
>> That is certainly what I had been assuming. Of course, if the POF is a new m-line (that is, it doesn't appear in the offer coming the other way), then there is no glare condition.
>
> Correct - assuming that the addition of the new m- lines does not indirectly affect values etc associated with existing m- lines.
>
> Regards,
>
> Christer
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
> As one of those that used the term, my intention was that the initial
> decision was to agree to define what RTCWEB needed.
>
> As each dependent working group looks at the extensions they need to do,
> then they need to make the decision as to whether a wider scope of
> possible, desirable, or dangerous.
>
> So the BUNDLE status at the moment is that any changes for the unified
> plan should work for RTCWEB usage, but not for general. It would
> probably make sense to make it more generally applicable, but that is
> the next step in the decision process, and may need some other technical
> decisions before it happens.

I don't know how you intend the above to be parsed. Specifically, when 
you say:
   "any changes for the unified plan should work for RTCWEB usage"

Do you mean:

1) that the changes should be designed solely with RTCWEB usage in mind? 
(But are still applicable in other environments that find them useful 
and choose to adopt them.)

Or do you mean:

2) that those changes should not be used in any environment other than 
RTCWEB?

I might be able to accept (1), but not (2).

	Thanks,
	Paul


> Keith
>
> ------------------------------------------------------------------------
>
> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
> Behalf Of *Bernard Aboba
> *Sent:* 07 August 2013 00:50
> *To:* Christer Holmberg; mmusic
> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>
> Christer said:
>
> "While cleaning up my MMUSIC notes today, I was listening to the audio
> recording, in order to get a clarification regarding what "scope is
> WEBRTC" really means.  However, unless I missed something, I didn't get
> a clear answer.
>
> Does it only cover the browser API (JSEP) or does it also cover
> something that can be sent out on the wire (as long as it is not SIP) by
> a browser?
>
> Feel free to tell me that it is a stupid question, and that the answer
> is very clear :)"
>
> [BA]  I actually think it is a good question :)
>
> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
> Plan" applies to use within  the JSEP API.    As a result, one
> might expect the SDP used in createOffer, setLocalDescription and
> setRemoteDescription to resemble that in the "Unified Plan" document.
> Since signaling is out of scope for WebRTC, applications that use the
> JSEP API might or might not use the "Unified Plan" SDP in signaling on
> the wire.
>
> However, "Unified Plan" includes normative references to published SDP
> specifications that were developed for general use, and the adoption of
> "Unified Plan" (or even an applicability statement within the "Unified
> Plan" document) doesn't change the applicability of those documents.
>
> Also, there may be some specifications that are part of the "Unified
> Plan" workload that may have applicability beyond WebRTC (e.g. a=group:
> SIMULCAST)
>
> Ultimately, I am expecting the work involved in "Unified Plan" to span
> multiple documents, and to clarify the applicability,  those individual
> documents will need to explain what they apply to (just WebRTC use or
> more general use).
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From mary.ietf.barnes@gmail.com  Wed Aug  7 07:56:06 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
>
>> As one of those that used the term, my intention was that the initial
>> decision was to agree to define what RTCWEB needed.
>>
>> As each dependent working group looks at the extensions they need to do,
>> then they need to make the decision as to whether a wider scope of
>> possible, desirable, or dangerous.
>>
>> So the BUNDLE status at the moment is that any changes for the unified
>> plan should work for RTCWEB usage, but not for general. It would
>> probably make sense to make it more generally applicable, but that is
>> the next step in the decision process, and may need some other technical
>> decisions before it happens.
>>
>
> I don't know how you intend the above to be parsed. Specifically, when you
> say:
>
>   "any changes for the unified plan should work for RTCWEB usage"
>
> Do you mean:
>
> 1) that the changes should be designed solely with RTCWEB usage in mind?
> (But are still applicable in other environments that find them useful and
> choose to adopt them.)
>
> Or do you mean:
>
> 2) that those changes should not be used in any environment other than
> RTCWEB?
>
> I might be able to accept (1), but not (2).
>
[MB] I would agree with your (1), however, not with the parenthetical part.
 I would suggest that be "But MAY still be applicable in other
environments... "   My concern is that at this point in time, you can't say
that the changes ARE applicable to other environments ....  [/MB]


>
>         Thanks,
>         Paul
>
>
>  Keith
>>
>> ------------------------------**------------------------------**
>> ------------
>>
>> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.**org<mmusic-bounces@ietf.org>]
>> *On
>> Behalf Of *Bernard Aboba
>> *Sent:* 07 August 2013 00:50
>> *To:* Christer Holmberg; mmusic
>> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>>
>>
>> Christer said:
>>
>> "While cleaning up my MMUSIC notes today, I was listening to the audio
>> recording, in order to get a clarification regarding what "scope is
>> WEBRTC" really means.  However, unless I missed something, I didn't get
>> a clear answer.
>>
>> Does it only cover the browser API (JSEP) or does it also cover
>> something that can be sent out on the wire (as long as it is not SIP) by
>> a browser?
>>
>> Feel free to tell me that it is a stupid question, and that the answer
>> is very clear :)"
>>
>> [BA]  I actually think it is a good question :)
>>
>> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
>> Plan" applies to use within  the JSEP API.    As a result, one
>> might expect the SDP used in createOffer, setLocalDescription and
>> setRemoteDescription to resemble that in the "Unified Plan" document.
>> Since signaling is out of scope for WebRTC, applications that use the
>> JSEP API might or might not use the "Unified Plan" SDP in signaling on
>> the wire.
>>
>> However, "Unified Plan" includes normative references to published SDP
>> specifications that were developed for general use, and the adoption of
>> "Unified Plan" (or even an applicability statement within the "Unified
>> Plan" document) doesn't change the applicability of those documents.
>>
>> Also, there may be some specifications that are part of the "Unified
>> Plan" workload that may have applicability beyond WebRTC (e.g. a=group:
>> SIMULCAST)
>>
>> Ultimately, I am expecting the work involved in "Unified Plan" to span
>> multiple documents, and to clarify the applicability,  those individual
>> documents will need to explain what they apply to (just WebRTC use or
>> more general use).
>>
>>
>>
>> ______________________________**_________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailman/listinfo/mmusic>
>>
>>
> ______________________________**_________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailman/listinfo/mmusic>
>

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Wed, Aug 7, 2013 at 8:40 AM,=
 Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu=
" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br><div cla=
ss=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 8/7/13 2:04 PM, DRAGE, =
Keith (Keith) wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
As one of those that used the term, my intention was that the initial<br>
decision was to agree to define what RTCWEB needed.<br>
<br>
As each dependent working group looks at the extensions they need to do,<br=
>
then they need to make the decision as to whether a wider scope of<br>
possible, desirable, or dangerous.<br>
<br>
So the BUNDLE status at the moment is that any changes for the unified<br>
plan should work for RTCWEB usage, but not for general. It would<br>
probably make sense to make it more generally applicable, but that is<br>
the next step in the decision process, and may need some other technical<br=
>
decisions before it happens.<br>
</blockquote>
<br></div>
I don&#39;t know how you intend the above to be parsed. Specifically, when =
you say:<div class=3D"im"><br>
=A0 &quot;any changes for the unified plan should work for RTCWEB usage&quo=
t;<br>
<br></div>
Do you mean:<br>
<br>
1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)<br>
<br>
Or do you mean:<br>
<br>
2) that those changes should not be used in any environment other than RTCW=
EB?<br>
<br>
I might be able to accept (1), but not (2).<br></blockquote><div>[MB] I wou=
ld agree with your (1), however, not with the parenthetical part. =A0I woul=
d suggest that be &quot;But MAY still be applicable in other environments..=
. &quot; =A0 My concern is that at this point in time, you can&#39;t say th=
at the changes ARE applicable to other environments .... =A0[/MB]</div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Keith<br>
<br>
------------------------------<u></u>------------------------------<u></u>-=
-----------<br>
<br>
*From:*<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" tar=
get=3D"_blank">mmusic-bounces@ietf.<u></u>org</a>] *On<br>
Behalf Of *Bernard Aboba<br>
*Sent:* 07 August 2013 00:50<br>
*To:* Christer Holmberg; mmusic<br>
*Subject:* Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<div><div=
 class=3D"h5"><br>
<br>
Christer said:<br>
<br>
&quot;While cleaning up my MMUSIC notes today, I was listening to the audio=
<br>
recording, in order to get a clarification regarding what &quot;scope is<br=
>
WEBRTC&quot; really means. =A0However, unless I missed something, I didn&#3=
9;t get<br>
a clear answer.<br>
<br>
Does it only cover the browser API (JSEP) or does it also cover<br>
something that can be sent out on the wire (as long as it is not SIP) by<br=
>
a browser?<br>
<br>
Feel free to tell me that it is a stupid question, and that the answer<br>
is very clear :)&quot;<br>
<br>
[BA] =A0I actually think it is a good question :)<br>
<br>
One interpretation of &quot;scope is WebRTC&quot; =A0is that the =A0SDP in =
&quot;Unified<br>
Plan&quot; applies to use within =A0the JSEP API. =A0 =A0As a result, one<b=
r>
might expect the SDP used in createOffer, setLocalDescription and<br>
setRemoteDescription to resemble that in the &quot;Unified Plan&quot; docum=
ent.<br>
Since signaling is out of scope for WebRTC, applications that use the<br>
JSEP API might or might not use the &quot;Unified Plan&quot; SDP in signali=
ng on<br>
the wire.<br>
<br>
However, &quot;Unified Plan&quot; includes normative references to publishe=
d SDP<br>
specifications that were developed for general use, and the adoption of<br>
&quot;Unified Plan&quot; (or even an applicability statement within the &qu=
ot;Unified<br>
Plan&quot; document) doesn&#39;t change the applicability of those document=
s.<br>
<br>
Also, there may be some specifications that are part of the &quot;Unified<b=
r>
Plan&quot; workload that may have applicability beyond WebRTC (e.g. a=3Dgro=
up:<br>
SIMULCAST)<br>
<br>
Ultimately, I am expecting the work involved in &quot;Unified Plan&quot; to=
 span<br>
multiple documents, and to clarify the applicability, =A0those individual<b=
r>
documents will need to explain what they apply to (just WebRTC use or<br>
more general use).<br>
<br>
<br>
<br></div></div>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
<br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</blockquote></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Hi,

Ok, so what is "RTCWEB usage" then? :)

For example, if I implement a JavaScript SIP app, and want to use the UP wi=
th SIP on the wire, would that be a "RTCWEB usage"? My understanding is tha=
t it would NOT be.

My personal understanding of the scope is:


1)      JSEP; and

2)      Browser-to-browser cases where we KNOW there are no intermediaries =
etc in between.

Regards,

Christer

L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Mary Barnes
L=E4hetetty: 7. elokuuta 2013 17:56
Vastaanottaja: Paul Kyzivat
Kopio: mmusic@ietf.org
Aihe: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"

On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:=
pkyzivat@alum.mit.edu>> wrote:
On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
As one of those that used the term, my intention was that the initial
decision was to agree to define what RTCWEB needed.

As each dependent working group looks at the extensions they need to do,
then they need to make the decision as to whether a wider scope of
possible, desirable, or dangerous.

So the BUNDLE status at the moment is that any changes for the unified
plan should work for RTCWEB usage, but not for general. It would
probably make sense to make it more generally applicable, but that is
the next step in the decision process, and may need some other technical
decisions before it happens.

I don't know how you intend the above to be parsed. Specifically, when you =
say:

  "any changes for the unified plan should work for RTCWEB usage"
Do you mean:

1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)

Or do you mean:

2) that those changes should not be used in any environment other than RTCW=
EB?

I might be able to accept (1), but not (2).
[MB] I would agree with your (1), however, not with the parenthetical part.=
  I would suggest that be "But MAY still be applicable in other environment=
s... "   My concern is that at this point in time, you can't say that the c=
hanges ARE applicable to other environments ....  [/MB]


        Thanks,
        Paul

Keith

------------------------------------------------------------------------

*From:*mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [mailto:mmus=
ic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org>] *On
Behalf Of *Bernard Aboba
*Sent:* 07 August 2013 00:50
*To:* Christer Holmberg; mmusic
*Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"


Christer said:

"While cleaning up my MMUSIC notes today, I was listening to the audio
recording, in order to get a clarification regarding what "scope is
WEBRTC" really means.  However, unless I missed something, I didn't get
a clear answer.

Does it only cover the browser API (JSEP) or does it also cover
something that can be sent out on the wire (as long as it is not SIP) by
a browser?

Feel free to tell me that it is a stupid question, and that the answer
is very clear :)"

[BA]  I actually think it is a good question :)

One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
Plan" applies to use within  the JSEP API.    As a result, one
might expect the SDP used in createOffer, setLocalDescription and
setRemoteDescription to resemble that in the "Unified Plan" document.
Since signaling is out of scope for WebRTC, applications that use the
JSEP API might or might not use the "Unified Plan" SDP in signaling on
the wire.

However, "Unified Plan" includes normative references to published SDP
specifications that were developed for general use, and the adoption of
"Unified Plan" (or even an applicability statement within the "Unified
Plan" document) doesn't change the applicability of those documents.

Also, there may be some specifications that are part of the "Unified
Plan" workload that may have applicability beyond WebRTC (e.g. a=3Dgroup:
SIMULCAST)

Ultimately, I am expecting the work involved in "Unified Plan" to span
multiple documents, and to clarify the applicability,  those individual
documents will need to explain what they apply to (just WebRTC use or
more general use).


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok, so wha=
t is &#8221;RTCWEB usage&#8221; then? :)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For exampl=
e, if I implement a JavaScript SIP app, and want to use the UP with SIP on =
the wire, would that be a &#8220;RTCWEB usage&#8221;? My understanding
 is that it would NOT be.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">My persona=
l understanding of the scope is:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:11.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><sp=
an style=3D"mso-list:Ignore">1)<span style=3D"font:7.0pt &quot;Times New Ro=
man&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US" style=3D"font-size:11.0=
pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">JS=
EP; and<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span lang=3D"EN-US" style=3D"font-size:11.0p=
t;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><sp=
an style=3D"mso-list:Ignore">2)<span style=3D"font:7.0pt &quot;Times New Ro=
man&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span lang=3D"EN-US" style=3D"font-size:11.0=
pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Br=
owser-to-browser cases where we KNOW there are no intermediaries etc in bet=
ween.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]
<b>Puolesta </b>Mary Barnes<br>
<b>L=E4hetetty:</b> 7. elokuuta 2013 17:56<br>
<b>Vastaanottaja:</b> Paul Kyzivat<br>
<b>Kopio:</b> mmusic@ietf.org<br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.e=
du</a>&gt; wrote:<o:p></o:p></p>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<p class=3D"MsoNormal">On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:<o:p><=
/o:p></p>
<p class=3D"MsoNormal">As one of those that used the term, my intention was=
 that the initial<br>
decision was to agree to define what RTCWEB needed.<br>
<br>
As each dependent working group looks at the extensions they need to do,<br=
>
then they need to make the decision as to whether a wider scope of<br>
possible, desirable, or dangerous.<br>
<br>
So the BUNDLE status at the moment is that any changes for the unified<br>
plan should work for RTCWEB usage, but not for general. It would<br>
probably make sense to make it more generally applicable, but that is<br>
the next step in the decision process, and may need some other technical<br=
>
decisions before it happens.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">I don't know how you intend the above to be parsed. =
Specifically, when you say:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
&nbsp; &quot;any changes for the unified plan should work for RTCWEB usage&=
quot;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal">Do you mean:<br>
<br>
1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)<br>
<br>
Or do you mean:<br>
<br>
2) that those changes should not be used in any environment other than RTCW=
EB?<br>
<br>
I might be able to accept (1), but not (2).<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal">[MB] I would agree with your (1), however, not with =
the parenthetical part. &nbsp;I would suggest that be &quot;But MAY still b=
e applicable in other environments... &quot; &nbsp; My concern is that at t=
his point in time, you can't say that the changes ARE
 applicable to other environments .... &nbsp;[/MB]<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
&nbsp; &nbsp; &nbsp; &nbsp; Thanks,<br>
&nbsp; &nbsp; &nbsp; &nbsp; Paul<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">Keith<br>
<br>
------------------------------------------------------------------------<br=
>
<br>
*From:*<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" tar=
get=3D"_blank">mmusic-bounces@ietf.org</a>] *On<br>
Behalf Of *Bernard Aboba<br>
*Sent:* 07 August 2013 00:50<br>
*To:* Christer Holmberg; mmusic<br>
*Subject:* Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<o:p></o:=
p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Christer said:<br>
<br>
&quot;While cleaning up my MMUSIC notes today, I was listening to the audio=
<br>
recording, in order to get a clarification regarding what &quot;scope is<br=
>
WEBRTC&quot; really means. &nbsp;However, unless I missed something, I didn=
't get<br>
a clear answer.<br>
<br>
Does it only cover the browser API (JSEP) or does it also cover<br>
something that can be sent out on the wire (as long as it is not SIP) by<br=
>
a browser?<br>
<br>
Feel free to tell me that it is a stupid question, and that the answer<br>
is very clear :)&quot;<br>
<br>
[BA] &nbsp;I actually think it is a good question :)<br>
<br>
One interpretation of &quot;scope is WebRTC&quot; &nbsp;is that the &nbsp;S=
DP in &quot;Unified<br>
Plan&quot; applies to use within &nbsp;the JSEP API. &nbsp; &nbsp;As a resu=
lt, one<br>
might expect the SDP used in createOffer, setLocalDescription and<br>
setRemoteDescription to resemble that in the &quot;Unified Plan&quot; docum=
ent.<br>
Since signaling is out of scope for WebRTC, applications that use the<br>
JSEP API might or might not use the &quot;Unified Plan&quot; SDP in signali=
ng on<br>
the wire.<br>
<br>
However, &quot;Unified Plan&quot; includes normative references to publishe=
d SDP<br>
specifications that were developed for general use, and the adoption of<br>
&quot;Unified Plan&quot; (or even an applicability statement within the &qu=
ot;Unified<br>
Plan&quot; document) doesn't change the applicability of those documents.<b=
r>
<br>
Also, there may be some specifications that are part of the &quot;Unified<b=
r>
Plan&quot; workload that may have applicability beyond WebRTC (e.g. a=3Dgro=
up:<br>
SIMULCAST)<br>
<br>
Ultimately, I am expecting the work involved in &quot;Unified Plan&quot; to=
 span<br>
multiple documents, and to clarify the applicability, &nbsp;those individua=
l<br>
documents will need to explain what they apply to (just WebRTC use or<br>
more general use).<br>
<br>
<br>
<o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">_____________________=
__________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Hi,

> What about adding a new m-line into a bundle?
> IIUC this is likely to be one of the most important use cases.
> Can this all be done with POF and PAN?

I have been thinking about that, because BUNDLE is one example where a chan=
ge to one m- line may have impacts on other m- lines.

Also, my understanding of the UP is that a PAN can only contain m- line tha=
t were present in the POF. So, if the POF triggers additional m- lines to b=
e changed on the remote side, those m- lines cannot be placed into the PAN.

Regards,

Christer

=09

On 8/7/13 8:39 AM, Christer Holmberg wrote:
> Hi,
>
>>> It has been indicated that POFs (partial offers) and PANs (partial answ=
ers) form a stand-alone mechanism, instead of being an extension of RFC3264=
.
>>>
>>> Now, assuming that is true, we still need to describe the interaction b=
etween POFs, PANs and 3264 Offer/Answers.
>>>
>>> Because, if I send a POF, and then receive a 3264 Offer (including the =
same m- line as the POF),  there is a glare and I need to reject the 3264 O=
ffer.
>>>
>>> In the same way, if I send a 3264 Offer, and receive a POF (including t=
he same m- line as the 3264 Offer), there is a glare and I need to reject t=
he POF.
>>>
>>> I ASSUME the above is clear to everyone, including the authors of the U=
P, but IF someone has a different opinion I'd like to hear about it :)
>>
>> That is certainly what I had been assuming. Of course, if the POF is a n=
ew m-line (that is, it doesn't appear in the offer coming the other way), t=
hen there is no glare condition.
>
> Correct - assuming that the addition of the new m- lines does not indirec=
tly affect values etc associated with existing m- lines.
>
> Regards,
>
> Christer
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
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Date: Wed, 7 Aug 2013 10:25:14 -0500
Message-ID: <CAHBDyN6F25u1FEB092arzOTx8Ecin8h0T9QYmUO9YSXMMLUwJw@mail.gmail.com>
From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Content-Type: multipart/alternative; boundary=089e011823a2e7f66804e35d25a3
Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,****
>
> ** **
>
> Ok, so what is =94RTCWEB usage=94 then? :)****
>
> ** **
>
> For example, if I implement a JavaScript SIP app, and want to use the UP
> with SIP on the wire, would that be a =93RTCWEB usage=94? My understandin=
g is
> that it would NOT be.****
>
> ** **
>
> My personal understanding of the scope is:
>
[MB] That is also my understanding. [/MB]

> ****
>
> ** **
>
> **1)      **JSEP; and****
>
> **2)      **Browser-to-browser cases where we KNOW there are no
> intermediaries etc in between.****
>
> ** **
>
> Regards,****
>
> ** **
>
> Christer****
>
> ** **
>
> *L=E4hett=E4j=E4:* mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.or=
g] *Puolesta
> *Mary Barnes
> *L=E4hetetty:* 7. elokuuta 2013 17:56
> *Vastaanottaja:* Paul Kyzivat
> *Kopio:* mmusic@ietf.org
> *Aihe:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"****
>
> ** **
>
> On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat <pkyzivat@alum.mit.edu>
> wrote:****
>
>  On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:****
>
> As one of those that used the term, my intention was that the initial
> decision was to agree to define what RTCWEB needed.
>
> As each dependent working group looks at the extensions they need to do,
> then they need to make the decision as to whether a wider scope of
> possible, desirable, or dangerous.
>
> So the BUNDLE status at the moment is that any changes for the unified
> plan should work for RTCWEB usage, but not for general. It would
> probably make sense to make it more generally applicable, but that is
> the next step in the decision process, and may need some other technical
> decisions before it happens.****
>
> ** **
>
> I don't know how you intend the above to be parsed. Specifically, when yo=
u
> say:****
>
>
>   "any changes for the unified plan should work for RTCWEB usage"****
>
> Do you mean:
>
> 1) that the changes should be designed solely with RTCWEB usage in mind?
> (But are still applicable in other environments that find them useful and
> choose to adopt them.)
>
> Or do you mean:
>
> 2) that those changes should not be used in any environment other than
> RTCWEB?
>
> I might be able to accept (1), but not (2).****
>
>  [MB] I would agree with your (1), however, not with the parenthetical
> part.  I would suggest that be "But MAY still be applicable in other
> environments... "   My concern is that at this point in time, you can't s=
ay
> that the changes ARE applicable to other environments ....  [/MB]****
>
>  ****
>
>
>         Thanks,
>         Paul
>
> ****
>
> Keith
>
> ------------------------------------------------------------------------
>
> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
> Behalf Of *Bernard Aboba
> *Sent:* 07 August 2013 00:50
> *To:* Christer Holmberg; mmusic
> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"****
>
>
>
> Christer said:
>
> "While cleaning up my MMUSIC notes today, I was listening to the audio
> recording, in order to get a clarification regarding what "scope is
> WEBRTC" really means.  However, unless I missed something, I didn't get
> a clear answer.
>
> Does it only cover the browser API (JSEP) or does it also cover
> something that can be sent out on the wire (as long as it is not SIP) by
> a browser?
>
> Feel free to tell me that it is a stupid question, and that the answer
> is very clear :)"
>
> [BA]  I actually think it is a good question :)
>
> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
> Plan" applies to use within  the JSEP API.    As a result, one
> might expect the SDP used in createOffer, setLocalDescription and
> setRemoteDescription to resemble that in the "Unified Plan" document.
> Since signaling is out of scope for WebRTC, applications that use the
> JSEP API might or might not use the "Unified Plan" SDP in signaling on
> the wire.
>
> However, "Unified Plan" includes normative references to published SDP
> specifications that were developed for general use, and the adoption of
> "Unified Plan" (or even an applicability statement within the "Unified
> Plan" document) doesn't change the applicability of those documents.
>
> Also, there may be some specifications that are part of the "Unified
> Plan" workload that may have applicability beyond WebRTC (e.g. a=3Dgroup:
> SIMULCAST)
>
> Ultimately, I am expecting the work involved in "Unified Plan" to span
> multiple documents, and to clarify the applicability,  those individual
> documents will need to explain what they apply to (just WebRTC use or
> more general use).
>
>
> ****
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic****
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic****
>
>  ** **
>

--089e011823a2e7f66804e35d25a3
Content-Type: text/html; charset=windows-1252
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Ok, so wha=
t is =94RTCWEB usage=94 then? :)<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">For exampl=
e, if I implement a JavaScript SIP app, and want to use the UP with SIP on =
the wire, would that be a =93RTCWEB usage=94? My understanding
 is that it would NOT be.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">My persona=
l understanding of the scope is:</span></p></div></div></blockquote><div>[M=
B] That is also my understanding. [/MB]=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"FI" link=3D"blue" vlink=3D"purp=
le"><div><p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.=
0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><=
u></u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p><u></u><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;=
Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><span>1)<span style=3D"=
font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=A0=A0
</span></span></span><u></u><span lang=3D"EN-US" style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">JSEP;=
 and<u></u><u></u></span></p>
<p><u></u><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;=
Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><span>2)<span style=3D"=
font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=A0=A0
</span></span></span><u></u><span lang=3D"EN-US" style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Brows=
er-to-browser cases where we KNOW there are no intermediaries etc in betwee=
n.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0=
<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> <a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-boun=
ces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" target=
=3D"_blank">mmusic-bounces@ietf.org</a>]
<b>Puolesta </b>Mary Barnes<br>
<b>L=E4hetetty:</b> 7. elokuuta 2013 17:56<br>
<b>Vastaanottaja:</b> Paul Kyzivat<br>
<b>Kopio:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@i=
etf.org</a><br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<u></u>=
<u></u></span></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.e=
du</a>&gt; wrote:<u></u><u></u></p>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<p class=3D"MsoNormal">On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:<u></u=
><u></u></p>
<p class=3D"MsoNormal">As one of those that used the term, my intention was=
 that the initial<br>
decision was to agree to define what RTCWEB needed.<br>
<br>
As each dependent working group looks at the extensions they need to do,<br=
>
then they need to make the decision as to whether a wider scope of<br>
possible, desirable, or dangerous.<br>
<br>
So the BUNDLE status at the moment is that any changes for the unified<br>
plan should work for RTCWEB usage, but not for general. It would<br>
probably make sense to make it more generally applicable, but that is<br>
the next step in the decision process, and may need some other technical<br=
>
decisions before it happens.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<p class=3D"MsoNormal">I don&#39;t know how you intend the above to be pars=
ed. Specifically, when you say:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
=A0 &quot;any changes for the unified plan should work for RTCWEB usage&quo=
t;<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">Do you mean:<br>
<br>
1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)<br>
<br>
Or do you mean:<br>
<br>
2) that those changes should not be used in any environment other than RTCW=
EB?<br>
<br>
I might be able to accept (1), but not (2).<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal">[MB] I would agree with your (1), however, not with =
the parenthetical part. =A0I would suggest that be &quot;But MAY still be a=
pplicable in other environments... &quot; =A0 My concern is that at this po=
int in time, you can&#39;t say that the changes ARE
 applicable to other environments .... =A0[/MB]<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
<br>
<u></u><u></u></p>
<p class=3D"MsoNormal">Keith<br>
<br>
------------------------------------------------------------------------<br=
>
<br>
*From:*<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" tar=
get=3D"_blank">mmusic-bounces@ietf.org</a>] *On<br>
Behalf Of *Bernard Aboba<br>
*Sent:* 07 August 2013 00:50<br>
*To:* Christer Holmberg; mmusic<br>
*Subject:* Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<u></u><u=
></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Christer said:<br>
<br>
&quot;While cleaning up my MMUSIC notes today, I was listening to the audio=
<br>
recording, in order to get a clarification regarding what &quot;scope is<br=
>
WEBRTC&quot; really means. =A0However, unless I missed something, I didn&#3=
9;t get<br>
a clear answer.<br>
<br>
Does it only cover the browser API (JSEP) or does it also cover<br>
something that can be sent out on the wire (as long as it is not SIP) by<br=
>
a browser?<br>
<br>
Feel free to tell me that it is a stupid question, and that the answer<br>
is very clear :)&quot;<br>
<br>
[BA] =A0I actually think it is a good question :)<br>
<br>
One interpretation of &quot;scope is WebRTC&quot; =A0is that the =A0SDP in =
&quot;Unified<br>
Plan&quot; applies to use within =A0the JSEP API. =A0 =A0As a result, one<b=
r>
might expect the SDP used in createOffer, setLocalDescription and<br>
setRemoteDescription to resemble that in the &quot;Unified Plan&quot; docum=
ent.<br>
Since signaling is out of scope for WebRTC, applications that use the<br>
JSEP API might or might not use the &quot;Unified Plan&quot; SDP in signali=
ng on<br>
the wire.<br>
<br>
However, &quot;Unified Plan&quot; includes normative references to publishe=
d SDP<br>
specifications that were developed for general use, and the adoption of<br>
&quot;Unified Plan&quot; (or even an applicability statement within the &qu=
ot;Unified<br>
Plan&quot; document) doesn&#39;t change the applicability of those document=
s.<br>
<br>
Also, there may be some specifications that are part of the &quot;Unified<b=
r>
Plan&quot; workload that may have applicability beyond WebRTC (e.g. a=3Dgro=
up:<br>
SIMULCAST)<br>
<br>
Ultimately, I am expecting the work involved in &quot;Unified Plan&quot; to=
 span<br>
multiple documents, and to clarify the applicability, =A0those individual<b=
r>
documents will need to explain what they apply to (just WebRTC use or<br>
more general use).<br>
<br>
<br>
<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">_____________________=
__________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><u></u><u></u></p>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
</div>
</div></div></div>
</div>

</blockquote></div><br></div></div>

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From christer.holmberg@ericsson.com  Wed Aug  7 08:27:45 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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If everyone has the same understanding, I would ask the chairs to make it v=
ery clear in the final minutes ;)

Regards,

Christer



L=E4hett=E4j=E4: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
L=E4hetetty: 7. elokuuta 2013 18:25
Vastaanottaja: Christer Holmberg
Kopio: Paul Kyzivat; mmusic@ietf.org
Aihe: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"



On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

Ok, so what is "RTCWEB usage" then? :)

For example, if I implement a JavaScript SIP app, and want to use the UP wi=
th SIP on the wire, would that be a "RTCWEB usage"? My understanding is tha=
t it would NOT be.

My personal understanding of the scope is:
[MB] That is also my understanding. [/MB]


1)      JSEP; and

2)      Browser-to-browser cases where we KNOW there are no intermediaries =
etc in between.

Regards,

Christer

L=E4hett=E4j=E4: mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [m=
ailto:mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org>] Puolesta Mar=
y Barnes
L=E4hetetty: 7. elokuuta 2013 17:56
Vastaanottaja: Paul Kyzivat
Kopio: mmusic@ietf.org<mailto:mmusic@ietf.org>
Aihe: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"

On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:=
pkyzivat@alum.mit.edu>> wrote:
On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
As one of those that used the term, my intention was that the initial
decision was to agree to define what RTCWEB needed.

As each dependent working group looks at the extensions they need to do,
then they need to make the decision as to whether a wider scope of
possible, desirable, or dangerous.

So the BUNDLE status at the moment is that any changes for the unified
plan should work for RTCWEB usage, but not for general. It would
probably make sense to make it more generally applicable, but that is
the next step in the decision process, and may need some other technical
decisions before it happens.

I don't know how you intend the above to be parsed. Specifically, when you =
say:

  "any changes for the unified plan should work for RTCWEB usage"
Do you mean:

1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)

Or do you mean:

2) that those changes should not be used in any environment other than RTCW=
EB?

I might be able to accept (1), but not (2).
[MB] I would agree with your (1), however, not with the parenthetical part.=
  I would suggest that be "But MAY still be applicable in other environment=
s... "   My concern is that at this point in time, you can't say that the c=
hanges ARE applicable to other environments ....  [/MB]


        Thanks,
        Paul
Keith

------------------------------------------------------------------------

*From:*mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [mailto:mmus=
ic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org>] *On
Behalf Of *Bernard Aboba
*Sent:* 07 August 2013 00:50
*To:* Christer Holmberg; mmusic
*Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"


Christer said:

"While cleaning up my MMUSIC notes today, I was listening to the audio
recording, in order to get a clarification regarding what "scope is
WEBRTC" really means.  However, unless I missed something, I didn't get
a clear answer.

Does it only cover the browser API (JSEP) or does it also cover
something that can be sent out on the wire (as long as it is not SIP) by
a browser?

Feel free to tell me that it is a stupid question, and that the answer
is very clear :)"

[BA]  I actually think it is a good question :)

One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
Plan" applies to use within  the JSEP API.    As a result, one
might expect the SDP used in createOffer, setLocalDescription and
setRemoteDescription to resemble that in the "Unified Plan" document.
Since signaling is out of scope for WebRTC, applications that use the
JSEP API might or might not use the "Unified Plan" SDP in signaling on
the wire.

However, "Unified Plan" includes normative references to published SDP
specifications that were developed for general use, and the adoption of
"Unified Plan" (or even an applicability statement within the "Unified
Plan" document) doesn't change the applicability of those documents.

Also, there may be some specifications that are part of the "Unified
Plan" workload that may have applicability beyond WebRTC (e.g. a=3Dgroup:
SIMULCAST)

Ultimately, I am expecting the work involved in "Unified Plan" to span
multiple documents, and to clarify the applicability,  those individual
documents will need to explain what they apply to (just WebRTC use or
more general use).

_______________________________________________
mmusic mailing list
mmusic@ietf.org<mailto:mmusic@ietf.org>
https://www.ietf.org/mailman/listinfo/mmusic

_______________________________________________
mmusic mailing list
mmusic@ietf.org<mailto:mmusic@ietf.org>
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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If everyon=
e has the same understanding, I would ask the chairs to make it
<b>very clear</b> in the final minutes ;)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
<br>
<b>L=E4hetetty:</b> 7. elokuuta 2013 18:25<br>
<b>Vastaanottaja:</b> Christer Holmberg<br>
<b>Kopio:</b> Paul Kyzivat; mmusic@ietf.org<br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok, so what is &#8221;RT=
CWEB usage&#8221; then? :)</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For example, if I implem=
ent a JavaScript SIP app, and want to use the UP with SIP on
 the wire, would that be a &#8220;RTCWEB usage&#8221;? My understanding is =
that it would NOT be.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">My personal understandin=
g of the scope is:</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">[MB] That is also my understanding. [/MB]&nbsp;<o:p>=
</o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri=
&quot;,&quot;sans-serif&quot;;color:#1F497D">1)</span><span lang=3D"EN-US" =
style=3D"font-size:7.0pt;color:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1F497D">JSEP; and</span><o:p></o:p=
></p>
<p><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri=
&quot;,&quot;sans-serif&quot;;color:#1F497D">2)</span><span lang=3D"EN-US" =
style=3D"font-size:7.0pt;color:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Browser-to-browser cases w=
here we KNOW there are no intermediaries etc in between.</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</span><o:p></o:=
p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</span><o:p></o:=
p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p>=
</p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,=
&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span style=3D"font-size=
:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"=
_blank">mmusic-bounces@ietf.org</a>]
<b>Puolesta </b>Mary Barnes<br>
<b>L=E4hetetty:</b> 7. elokuuta 2013 17:56<br>
<b>Vastaanottaja:</b> Paul Kyzivat<br>
<b>Kopio:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@i=
etf.org</a><br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;</span>=
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat &lt;<a href=3D"mailto=
:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt; wro=
te:<o:p></o:p></p>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">As one of those that used the term, my intention was that the init=
ial<br>
decision was to agree to define what RTCWEB needed.<br>
<br>
As each dependent working group looks at the extensions they need to do,<br=
>
then they need to make the decision as to whether a wider scope of<br>
possible, desirable, or dangerous.<br>
<br>
So the BUNDLE status at the moment is that any changes for the unified<br>
plan should work for RTCWEB usage, but not for general. It would<br>
probably make sense to make it more generally applicable, but that is<br>
the next step in the decision process, and may need some other technical<br=
>
decisions before it happens.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I don't know how you intend the above to be parsed. Specifically, =
when you say:<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><br>
&nbsp; &quot;any changes for the unified plan should work for RTCWEB usage&=
quot;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Do you mean:<br>
<br>
1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)<br>
<br>
Or do you mean:<br>
<br>
2) that those changes should not be used in any environment other than RTCW=
EB?<br>
<br>
I might be able to accept (1), but not (2).<o:p></o:p></p>
</blockquote>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">[MB] I would agree with your (1), however, not with the parentheti=
cal part. &nbsp;I would suggest that be &quot;But MAY still be applicable i=
n other environments... &quot; &nbsp; My concern is that
 at this point in time, you can't say that the changes ARE applicable to ot=
her environments .... &nbsp;[/MB]<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><br>
&nbsp; &nbsp; &nbsp; &nbsp; Thanks,<br>
&nbsp; &nbsp; &nbsp; &nbsp; Paul<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Keith<br>
<br>
------------------------------------------------------------------------<br=
>
<br>
*From:*<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" tar=
get=3D"_blank">mmusic-bounces@ietf.org</a>] *On<br>
Behalf Of *Bernard Aboba<br>
*Sent:* 07 August 2013 00:50<br>
*To:* Christer Holmberg; mmusic<br>
*Subject:* Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<o:p></o:=
p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t"><br>
<br>
Christer said:<br>
<br>
&quot;While cleaning up my MMUSIC notes today, I was listening to the audio=
<br>
recording, in order to get a clarification regarding what &quot;scope is<br=
>
WEBRTC&quot; really means. &nbsp;However, unless I missed something, I didn=
't get<br>
a clear answer.<br>
<br>
Does it only cover the browser API (JSEP) or does it also cover<br>
something that can be sent out on the wire (as long as it is not SIP) by<br=
>
a browser?<br>
<br>
Feel free to tell me that it is a stupid question, and that the answer<br>
is very clear :)&quot;<br>
<br>
[BA] &nbsp;I actually think it is a good question :)<br>
<br>
One interpretation of &quot;scope is WebRTC&quot; &nbsp;is that the &nbsp;S=
DP in &quot;Unified<br>
Plan&quot; applies to use within &nbsp;the JSEP API. &nbsp; &nbsp;As a resu=
lt, one<br>
might expect the SDP used in createOffer, setLocalDescription and<br>
setRemoteDescription to resemble that in the &quot;Unified Plan&quot; docum=
ent.<br>
Since signaling is out of scope for WebRTC, applications that use the<br>
JSEP API might or might not use the &quot;Unified Plan&quot; SDP in signali=
ng on<br>
the wire.<br>
<br>
However, &quot;Unified Plan&quot; includes normative references to publishe=
d SDP<br>
specifications that were developed for general use, and the adoption of<br>
&quot;Unified Plan&quot; (or even an applicability statement within the &qu=
ot;Unified<br>
Plan&quot; document) doesn't change the applicability of those documents.<b=
r>
<br>
Also, there may be some specifications that are part of the &quot;Unified<b=
r>
Plan&quot; workload that may have applicability beyond WebRTC (e.g. a=3Dgro=
up:<br>
SIMULCAST)<br>
<br>
Ultimately, I am expecting the work involved in &quot;Unified Plan&quot; to=
 span<br>
multiple documents, and to clarify the applicability, &nbsp;those individua=
l<br>
documents will need to explain what they apply to (just WebRTC use or<br>
more general use).<br>
<br>
<o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
</blockquote>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C42157DESESSMB209erics_--

From keith.drage@alcatel-lucent.com  Wed Aug  7 10:26:46 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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I said that all the involved groups need to make a second step with all the=
ir deliverables, which is to decide whether what they are doing is used gen=
erally, or solely in the context of WEBRTC.

Deciding whether it is 1) or 2) has not been defined in my understanding, a=
nd would really come under the scope of the second decision.

I for one am hoping as much of this as possible comes out as being generall=
y applicable, and obviously designed as such as a result, but the MMUSIC WG=
 did not appear to be ready to say that last week. So some steps to get the=
re.

Keith


> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat=20
> Sent: 07 August 2013 14:41
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>=20
> On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
> > As one of those that used the term, my intention was that the initial
> > decision was to agree to define what RTCWEB needed.
> >
> > As each dependent working group looks at the extensions they need to do=
,
> > then they need to make the decision as to whether a wider scope of
> > possible, desirable, or dangerous.
> >
> > So the BUNDLE status at the moment is that any changes for the unified
> > plan should work for RTCWEB usage, but not for general. It would
> > probably make sense to make it more generally applicable, but that is
> > the next step in the decision process, and may need some other technica=
l
> > decisions before it happens.
>=20
> I don't know how you intend the above to be parsed. Specifically, when
> you say:
>    "any changes for the unified plan should work for RTCWEB usage"
>=20
> Do you mean:
>=20
> 1) that the changes should be designed solely with RTCWEB usage in mind?
> (But are still applicable in other environments that find them useful
> and choose to adopt them.)
>=20
> Or do you mean:
>=20
> 2) that those changes should not be used in any environment other than
> RTCWEB?
>=20
> I might be able to accept (1), but not (2).
>=20
> 	Thanks,
> 	Paul
>=20
>=20
> > Keith
> >
> > -----------------------------------------------------------------------=
-
> >
> > *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
> > Behalf Of *Bernard Aboba
> > *Sent:* 07 August 2013 00:50
> > *To:* Christer Holmberg; mmusic
> > *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
> >
> > Christer said:
> >
> > "While cleaning up my MMUSIC notes today, I was listening to the audio
> > recording, in order to get a clarification regarding what "scope is
> > WEBRTC" really means.  However, unless I missed something, I didn't get
> > a clear answer.
> >
> > Does it only cover the browser API (JSEP) or does it also cover
> > something that can be sent out on the wire (as long as it is not SIP) b=
y
> > a browser?
> >
> > Feel free to tell me that it is a stupid question, and that the answer
> > is very clear :)"
> >
> > [BA]  I actually think it is a good question :)
> >
> > One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
> > Plan" applies to use within  the JSEP API.    As a result, one
> > might expect the SDP used in createOffer, setLocalDescription and
> > setRemoteDescription to resemble that in the "Unified Plan" document.
> > Since signaling is out of scope for WebRTC, applications that use the
> > JSEP API might or might not use the "Unified Plan" SDP in signaling on
> > the wire.
> >
> > However, "Unified Plan" includes normative references to published SDP
> > specifications that were developed for general use, and the adoption of
> > "Unified Plan" (or even an applicability statement within the "Unified
> > Plan" document) doesn't change the applicability of those documents.
> >
> > Also, there may be some specifications that are part of the "Unified
> > Plan" workload that may have applicability beyond WebRTC (e.g. a=3Dgrou=
p:
> > SIMULCAST)
> >
> > Ultimately, I am expecting the work involved in "Unified Plan" to span
> > multiple documents, and to clarify the applicability,  those individual
> > documents will need to explain what they apply to (just WebRTC use or
> > more general use).
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On Aug 7, 2013, at 11:21 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Ok, so what is =94RTCWEB usage=94 then? :)
> For example, if I implement a JavaScript SIP app, and want to use the =
UP with SIP on the wire, would that be a =93RTCWEB usage=94? My =
understanding is that it would NOT be.

No it wouldn't be, imo, but perhaps in a different way than you mean it. =
 If you implement "SIP", by definition you follow the SIP RFCs.  If you =
don't follow SIP RFCs, you're not really implementing SIP.  It's no =
different than if you wrote it in C++ or Java or whatever, as a native =
application.  The fact that you're running inside a browser and using =
javascript as a language doesn't change the SIP protocol or what it =
needs to do/not-do to interoperate with other SIP devices. =20

If you don't want to interoperate with other SIP devices and the =
deployed SIP domains, then do whatever you like and add whatever =
extensions you wish.  There's no protocol police, nor need for one, when =
two hosts want to do whatever they want to do with each other.  So long =
as they are running the same code (e.g., javascript), or privately agree =
on how things work between each other, things work in an interoperable =
fashion.

I think people want to think of WebRTC as SIPv3, but it isn't - there's =
no actual signaling protocol defined, nor does it make sense to define =
one.  Even the JSEP draft makes the point that the application can do =
whatever it likes on-the-wire. (right?)  The fact that WebRTC uses SDP =
for the "API" is an (unfortunate) coincidence - I don't mean it was done =
by accident, since clearly it wasn't - but I mean its usage in a WebRTC =
context is not defined as the on-the-wire encoding for host-to-host =
media session exchange/negotiation protocol.  We might use it that way =
in our javascript app, and draw it that way in powerpoint, but it's not =
one.  It's silly to constrain WebRTC applications to semantics/behaviors =
SIP or RTSP or MGCP support, even when using SDP for the Javascript API. =
 If JSEP, for example, wants to go do all kinds of new things for =
itself, it should be able to - without impacting anything else.


> My personal understanding of the scope is:
> =20
> 1)      JSEP; and
> 2)      Browser-to-browser cases where we KNOW there are no =
intermediaries etc in between.

My understanding of the "scope" is that it's applicable to the actual =
Javascript API as well as (1) JSEP.  Saying (2) is a bit weird, both =
because it can be between a browser and a native application or server =
or gateway or whatever, but also because there are in fact =
intermediaries (HTTP Proxies, for example).

-hadriel


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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On Aug 7, 2013, at 1:44 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 6 August 2013 00:55, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
>> If you use separate m- lines per media direction, there MAY be a =
smaller risk that both endpoints try to modify the same m- line. Maybe =
that is what he was talking about :)
>=20
> That's actually all I was suggesting.  But the wild speculation that
> ensued was quite interesting nonetheless.
>=20
> I believe that most of what has been said on this thread is also true.
> Yes, you can probably reduce glare further by defining some specific
> rules (reject beats any other change, etc...).  Yes, you can devise
> rules for other stuff.  Yes, it's probably not worth it for a lot of
> those things anyway.  Yes, changing parameters that affect multiple
> m-lines, like transport parameters, is probably going to cause
> unavoidable wide-scale glare.  Etc..

Huh.  Well my idea was much better... sure, it may not provide any =
value, benefit, or fix any problem... but it's a way cooler idea. ;)

-hadriel


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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 6, 2013, at 11:49 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I don't really understand the "only for WebRTC" agreement.
> I do see how defining extensions to O/A might be done only for WebRTC =
and not for 3264. (I don't *agree*, but I do *understand* that.)
>=20
> But BUNDLE is a change to SDP, and SDP extensions are required to =
specify how they interact with O/A.

Is that actually true?  I don't see that requirement in RFC 4566 IANA =
registration procedures, nor in IANA's SDP parameters registry.


> I don't think it makes any sense to say that such changes only apply =
to WebRTC. They should apply to anything that chooses to claim support =
to the BUNDLE draft.

OK, but in a SIP-based usage, I think it's dangerous to offer port=3D0 =
media lines and negotiate to use those media lines with the far end due =
to an answer containing a new bundle attribute, without sending another =
offer indicating you're using those media lines.

So I think if we want to use BUNDLE in SIP as well, we should really =
require it to do another offer/answer round.  In a WebRTC usage (ie, in =
Javascript API and JSEP), it can avoid the second round.

One way to describe this in a BUNDLE RFC, I think, is to say it MUST do =
another offer/answer exchange, except when used in [draft-jsep] or as an =
application programming interface. (or something like that)

-hadriel


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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 8/7/13 12:29, Hadriel Kaplan wrote:
> Huh.  Well my idea was much better... sure, it may not provide any value, benefit, or fix any problem... but it's a way cooler idea. ;)
>

Yeah, maybe -- but I think we want to keep this relatively simple. More 
directly: we need to be careful to evaluate what incremental gains added 
complexity gets us over a simpler mechanism. I think the scheme you 
outlined is out in the "diminishing returns" part of the solution space.

/a

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Date: Wed, 7 Aug 2013 13:02:00 -0500
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Wed, Aug 7, 2013 at 12:52 PM, Hadriel Kaplan
<hadriel.kaplan@oracle.com>wrote:

>
> On Aug 6, 2013, at 11:49 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
> > I don't really understand the "only for WebRTC" agreement.
> > I do see how defining extensions to O/A might be done only for WebRTC
> and not for 3264. (I don't *agree*, but I do *understand* that.)
> >
> > But BUNDLE is a change to SDP, and SDP extensions are required to
> specify how they interact with O/A.
>
> Is that actually true?  I don't see that requirement in RFC 4566 IANA
> registration procedures, nor in IANA's SDP parameters registry.
>
>
> > I don't think it makes any sense to say that such changes only apply to
> WebRTC. They should apply to anything that chooses to claim support to the
> BUNDLE draft.
>
> OK, but in a SIP-based usage, I think it's dangerous to offer port=0 media
> lines and negotiate to use those media lines with the far end due to an
> answer containing a new bundle attribute, without sending another offer
> indicating you're using those media lines.
>
> So I think if we want to use BUNDLE in SIP as well, we should really
> require it to do another offer/answer round.  In a WebRTC usage (ie, in
> Javascript API and JSEP), it can avoid the second round.
>
> One way to describe this in a BUNDLE RFC, I think, is to say it MUST do
> another offer/answer exchange, except when used in [draft-jsep] or as an
> application programming interface. (or something like that)
>
[MB] Exactly.  I think it's extremely important that the different usages
of BUNDLE/SDP are clearly delineated.  I think that's one of the reasons we
have gotten so bogged down with this whole topic.  There's no one to have a
one size fits all solution give the disparity in the usages. [/MB]

>
> -hadriel
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Aug 7, 2013 at 12:52 PM, Hadriel Kaplan <span dir=3D"ltr">&=
lt;<a href=3D"mailto:hadriel.kaplan@oracle.com" target=3D"_blank">hadriel.k=
aplan@oracle.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Aug 6, 2013, at 11:49 AM, Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@al=
um.mit.edu">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
<br>
&gt; I don&#39;t really understand the &quot;only for WebRTC&quot; agreemen=
t.<br>
&gt; I do see how defining extensions to O/A might be done only for WebRTC =
and not for 3264. (I don&#39;t *agree*, but I do *understand* that.)<br>
&gt;<br>
&gt; But BUNDLE is a change to SDP, and SDP extensions are required to spec=
ify how they interact with O/A.<br>
<br>
</div>Is that actually true? =A0I don&#39;t see that requirement in RFC 456=
6 IANA registration procedures, nor in IANA&#39;s SDP parameters registry.<=
br>
<div class=3D"im"><br>
<br>
&gt; I don&#39;t think it makes any sense to say that such changes only app=
ly to WebRTC. They should apply to anything that chooses to claim support t=
o the BUNDLE draft.<br>
<br>
</div>OK, but in a SIP-based usage, I think it&#39;s dangerous to offer por=
t=3D0 media lines and negotiate to use those media lines with the far end d=
ue to an answer containing a new bundle attribute, without sending another =
offer indicating you&#39;re using those media lines.<br>

<br>
So I think if we want to use BUNDLE in SIP as well, we should really requir=
e it to do another offer/answer round. =A0In a WebRTC usage (ie, in Javascr=
ipt API and JSEP), it can avoid the second round.<br>
<br>
One way to describe this in a BUNDLE RFC, I think, is to say it MUST do ano=
ther offer/answer exchange, except when used in [draft-jsep] or as an appli=
cation programming interface. (or something like that)<br></blockquote>
<div>[MB] Exactly. =A0I think it&#39;s extremely important that the differe=
nt usages of BUNDLE/SDP are clearly delineated. =A0I think that&#39;s one o=
f the reasons we have gotten so bogged down with this whole topic. =A0There=
&#39;s no one to have a one size fits all solution give the disparity in th=
e usages. [/MB]=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
-hadriel<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div></div>

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From christer.holmberg@ericsson.com  Wed Aug  7 11:05:33 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in	bundled m- lines (SDP Offers only)
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Hi,

>> I don't think it makes any sense to say that such changes only apply to =
WebRTC. They should apply to anything that chooses to claim support to the =
BUNDLE draft.
>
> OK, but in a SIP-based usage, I think it's dangerous to offer port=3D0 me=
dia lines and negotiate to use those media lines with the far end due to an=
 answer containing a new bundle attribute, without sending another offer in=
dicating you're using those media lines.

You have to send an offer which indicates the port you are going to use. So=
metimes that will be a "second offer" (because the "first offer" contained =
different ports).

> So I think if we want to use BUNDLE in SIP as well, we should really requ=
ire it to do another offer/answer round.  In a WebRTC usage (ie, in Javascr=
ipt API and JSEP), it can avoid the second round.

BUNDLE as such *CAN* be used with SIP. The question is whether specific par=
ts, e.g. the SDP bundle-only attribute, needs to be scoped to "WebRTC only"=
.

> One way to describe this in a BUNDLE RFC, I think, is to say it MUST do a=
nother offer/answer exchange, except when used in [draft-jsep] or as an app=
lication programming interface. (or something like that)

The draft already says that a second offer/answer must be sent, if the prev=
ious one did not contain the port values that are going to be used.

Regards,

Christer


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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On Aug 7, 2013, at 2:05 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

>> One way to describe this in a BUNDLE RFC, I think, is to say it MUST =
do another offer/answer exchange, except when used in [draft-jsep] or as =
an application programming interface. (or something like that)
>=20
> The draft already says that a second offer/answer must be sent, if the =
previous one did not contain the port values that are going to be used.

This is challenging to discuss because as far as I'm aware, the UP plan =
will make changes to the BUDNLE draft.  Correct?  If so, saying "but it =
already says X" doesn't give me the warm-fuzzies.

I believe (but could easily be wrong) the new UP plan would not require =
a second O/A.  Or at least that's what I interpreted Adam's email to =
mean here:
http://www.ietf.org/mail-archive/web/mmusic/current/msg12030.html

If I misunderstood his email, I apologize.

-hadriel


From christer.holmberg@ericsson.com  Wed Aug  7 11:28:42 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Hi,

>>> One way to describe this in a BUNDLE RFC, I think, is to say it MUST do=
 another offer/answer exchange, except when used in [draft-jsep] or as an a=
pplication programming interface. (or something like that)
>>=20
>> The draft already says that a second offer/answer must be sent, if the p=
revious one did not contain the port values that are going to be used.
>
> This is challenging to discuss because as far as I'm aware, the UP plan w=
ill make changes to the BUDNLE draft.  Correct?  If so, saying "but it alre=
ady says X" doesn't give me the warm-fuzzies.
>
> I believe (but could easily be wrong) the new UP plan would not require a=
 second O/A.  Or at least that's what I interpreted Adam's email to mean he=
re:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg12030.html
>
> If I misunderstood his email, I apologize.

You probably understood correct. Now, when I re-read it, I remembered that =
I didn't agree :) My opinion is that a second offer is always needed if the=
 previous offer did not contain the port values that you are actually going=
 to end up using.

So, if the first offer contains bundle-only/port=3D0, and the receiver acce=
pts it, the second offer would contain bundle-only/port=3Dbundle_port.

But, obviously this is something that we'll have to discuss as part of the =
adding-UP-to-BUNDLE procedure.

Regards,

Christer


From hadriel.kaplan@oracle.com  Wed Aug  7 13:14:41 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On Aug 7, 2013, at 1:55 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/7/13 12:29, Hadriel Kaplan wrote:
>> Huh.  Well my idea was much better... sure, it may not provide any =
value, benefit, or fix any problem... but it's a way cooler idea. ;)
>>=20
>=20
> Yeah, maybe -- but I think we want to keep this relatively simple. =
More directly: we need to be careful to evaluate what incremental gains =
added complexity gets us over a simpler mechanism. I think the scheme =
you outlined is out in the "diminishing returns" part of the solution =
space.

Dude, it was a joke.  Apparently not a funny one, though. :(

-hadriel


From christer.holmberg@ericsson.com  Wed Aug  7 13:38:30 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] unified plan glareless atomicity improvement
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Cc: "mmusic@ietf.org \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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Hadriel, your jokes work much better during f2f meetings, because part of t=
he joke telling is that you first laugh at your joke yourself, before anyon=
e else :)

Regards,

Christer

-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Hadriel Kaplan
L=E4hetetty: 7. elokuuta 2013 23:14
Vastaanottaja: Adam Roach
Kopio: mmusic@ietf.org (E-mail)
Aihe: Re: [MMUSIC] unified plan glareless atomicity improvement


On Aug 7, 2013, at 1:55 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/7/13 12:29, Hadriel Kaplan wrote:
>> Huh.  Well my idea was much better... sure, it may not provide any value=
, benefit, or fix any problem... but it's a way cooler idea. ;)
>>=20
>=20
> Yeah, maybe -- but I think we want to keep this relatively simple. More d=
irectly: we need to be careful to evaluate what incremental gains added com=
plexity gets us over a simpler mechanism. I think the scheme you outlined i=
s out in the "diminishing returns" part of the solution space.

Dude, it was a joke.  Apparently not a funny one, though. :(

-hadriel

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

From partha@parthasarathi.co.in  Wed Aug  7 14:50:45 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Hadriel Kaplan'" <hadriel.kaplan@oracle.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in	bundled m- lines (SDP Offers only)
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Hi all,

It is a concern area for UP plan in case it works only for WebRTC and not
for SIP/general. The changing current BUNDLE approach breaks interop with
SIP/legacy endpoints. 

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Wednesday, August 07, 2013 11:47 PM
> To: Christer Holmberg
> Cc: mmusic@ietf.org (E-mail)
> Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero
> usage in bundled m- lines (SDP Offers only)
> 
> 
> On Aug 7, 2013, at 2:05 PM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
> 
> >> One way to describe this in a BUNDLE RFC, I think, is to say it MUST
> do another offer/answer exchange, except when used in [draft-jsep] or
> as an application programming interface. (or something like that)
> >
> > The draft already says that a second offer/answer must be sent, if
> the previous one did not contain the port values that are going to be
> used.
> 
> This is challenging to discuss because as far as I'm aware, the UP plan
> will make changes to the BUDNLE draft.  Correct?  If so, saying "but it
> already says X" doesn't give me the warm-fuzzies.
> 
> I believe (but could easily be wrong) the new UP plan would not require
> a second O/A.  Or at least that's what I interpreted Adam's email to
> mean here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg12030.html
> 
> If I misunderstood his email, I apologize.
> 
> -hadriel
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Wed Aug  7 23:18:54 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

As has been indicated, bundle-only can only be used when one knows that the=
 receiver, and possible intermediaries etc, support it.

So, based on that assumption, my question is: why do we need port 0? Why no=
t assign the suggested bundle port to the bundle-only m- lines from the beg=
inning? As the bundle port is anyway going to be used (even if the receiver=
 does not want to use bundle), there is no need to reserve additional ports=
 for bundle-only m- lines.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

--_000_7594FB04B1934943A5C02806D1A2204B1C421C80ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"font-family:Calibri, Arial, Helvetica, sans-serif;font-size:=
11pt;color:black">
<div><span style=3D"color: black; font-family: Calibri, Arial, Helvetica, s=
ans-serif; font-size: 11pt;">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">As has been indicated, bundle-only can only be =
used when one knows that the receiver, and possible intermediaries etc, sup=
port it.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">So, based on that assumption, my question is: w=
hy do we need port 0? Why not&nbsp;assign the suggested bundle port to the =
bundle-only m- lines from the beginning? As the bundle port is anyway going=
 to be used (even if the receiver does
 not want to use bundle), there is no need to reserve additional ports for =
bundle-only m- lines.</font></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color: black; font-family: Calibri, Arial, Helvetica, sans-s=
erif; font-size: 11pt;">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color: blue;">TouchDown</=
b> (www.nitrodesk.com)</div>
</span>
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From hadriel.kaplan@oracle.com  Thu Aug  8 05:21:27 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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If I understand your question, you're asking why bundle-only m-lines =
would use a port number of 0 in the offer, instead of using the same =
port number as the regular bundle m-line they're going to =
transport-bundle with?

I believe the rationale was basically argued about before, either in =
IETF 86 Orlando or IETF 85 Atlanta.  It gets back to the original debate =
about whether a BUNDLE offer should use the same port numbers or =
different ones, for bundle m-lines (what was called "BUNDLE vs. CRUNDLE" =
if I recall correctly).

One WG participant argued against using the same port numbers for bundle =
lines.  His argument was that since it was never described in an RFC, =
and he knew of one implementation that didn't handle it well, that =
therefore we couldn't send an offer using the same port numbers until we =
found out if the far-end supported BUNDLE or not.

That's what drove the decision to use different port numbers even in =
regular bundle cases, afaik.

Therefore, even for bundle-only lines, we can't use the same non-0 port =
number for them, because the same theoretical issue will exist.

Also, from a purely protocol perspective, it seems quite logical to me =
to use a port number of 0 for an m-line you literally don't want to use =
if the far-end doesn't support BUNDLE - a port number of 0 effectively =
means "disabled" today, and that's basically what you want.  What I =
don't think is right, is (1) having the far-end send an SDP answer back =
that sets a real port number for those same "disabled" m-lines, and (2) =
not requiring another offer/answer exchange that then uses the correct =
(same) port numbers for the previously-disabled m-lines.

What I think should happen is the UAS should send an SDP answer using =
port=3D0 for those same m-lines, and the UAC should then send a new =
offer with those m-lines having the real port number of the bundle =
transport, and the far-end should answer that in the same fashion.

-hadriel


On Aug 8, 2013, at 2:18 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> As has been indicated, bundle-only can only be used when one knows =
that the receiver, and possible intermediaries etc, support it.
> =20
> So, based on that assumption, my question is: why do we need port 0? =
Why not assign the suggested bundle port to the bundle-only m- lines =
from the beginning? As the bundle port is anyway going to be used (even =
if the receiver does not want to use bundle), there is no need to =
reserve additional ports for bundle-only m- lines.
> =20
> Regards,
> =20
> Christer
> =20
>=20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Thu Aug  8 06:04:36 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
Thread-Index: AQHOlDHLQN7nRiDYV02hbFJeK66gpJmLRGsw
Date: Thu, 8 Aug 2013 13:04:28 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

> If I understand your question, you're asking why bundle-only m-lines woul=
d use a port number of 0 in the offer, instead of using the same port numbe=
r as the regular bundle m-line they're going to transport-bundle with?

Yes.

> I believe the rationale was basically argued about before, either in IETF=
 86 Orlando or IETF 85 Atlanta.  It gets back to the original debate about =
whether a BUNDLE offer should=20
> use the same port numbers or different ones, for bundle m-lines (what was=
 called "BUNDLE vs. CRUNDLE" if I recall correctly).
>
> One WG participant argued against using the same port numbers for bundle =
lines.  His argument was that since it was never described in an RFC, and h=
e knew of one implementation=20
> that didn't handle it well, that therefore we couldn't send an offer usin=
g the same port numbers until we found out if the far-end supported BUNDLE =
or not.
>
> That's what drove the decision to use different port numbers even in regu=
lar bundle cases, afaik.

Correct.=20

Another advantage of using different port numbers is that it provides a fal=
lback in case the remote endpoint does not support, or does not want to use=
 BUNDLE.

However, for bundle-only m- lines such fallback is not needed - they are ON=
LY going to be used IF the remote endpoint chooses to use BUNDLE.

...and, based on the agreed scope, the Offerer KNOWS that the remote endpoi=
nt supports BUNDLE, so the "one implementation that doesn't handle it" will=
 not be there :)=20

> Therefore, even for bundle-only lines, we can't use the same non-0 port n=
umber for them, because the same theoretical issue will exist.

Well, I think there will be some cases where we will have to allow the same=
 port number. Bundle-only is one case, trickle ICE another (where one can a=
ssign port 9 for multiple m- lines).

> Also, from a purely protocol perspective, it seems quite logical to me to=
 use a port number of 0 for an m-line you literally don't want to use if th=
e far-end doesn't support=20
> BUNDLE - a port number of 0 effectively means "disabled" today, and that'=
s basically what you want. =20

Well, one could also say that it means "Here are some m- lines  I want to b=
undle, with this port, but only if you (the remote endpoint) wants to use b=
undle." :)

> What I don't think is right, is (1) having the far-end send an SDP answer=
 back that sets a real port number for those same "disabled" m-lines, and (=
2) not requiring=20
> another offer/answer exchange that then uses the correct (same) port numb=
ers for the previously-disabled m-lines.

So far I agree with you on this one :)

> What I think should happen is the UAS should send an SDP answer using por=
t=3D0 for those same m-lines, and the UAC should then send a new offer with=
 those m-lines having the real port number of the bundle transport, and the=
 far-end should answer that in the same fashion.

The problem is that the grouping framework does not allow, in SDP Answers, =
usage of port=3D0 in grouped m- lines. It would require an update of the RF=
C, and I think we want to avoid that.

Also, if the Answerer chooses to use BUNDLE, as it DOES include its bundle =
port for all other m- lines, I think it should also do it for the bundle-on=
ly m- lines.

Regards,

Christer



On Aug 8, 2013, at 2:18 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Hi,
> =20
> As has been indicated, bundle-only can only be used when one knows that t=
he receiver, and possible intermediaries etc, support it.
> =20
> So, based on that assumption, my question is: why do we need port 0? Why =
not assign the suggested bundle port to the bundle-only m- lines from the b=
eginning? As the bundle port is anyway going to be used (even if the receiv=
er does not want to use bundle), there is no need to reserve additional por=
ts for bundle-only m- lines.
> =20
> Regards,
> =20
> Christer
> =20
>=20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From hadriel.kaplan@oracle.com  Thu Aug  8 07:15:59 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 9:04 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Another advantage of using different port numbers is that it provides =
a fallback in case the remote endpoint does not support, or does not =
want to use BUNDLE.

Yes, except as far as I can tell all devices (and even intermediaries) =
handle an offer with the same port number just fine.  But that's just a =
personal observation, and dangerous to assume ALL of them handle it =
fine; that's why I didn't argue for long that we should use the same =
port number in the offer - because it concerned me that someone knew of =
at least one implementation that didn't handle it well. (not that we can =
truly accommodate ALL implementations anyway, but it's got to be a heck =
of a lot higher than simple majority, and as close to 100% as we can =
get)


> However, for bundle-only m- lines such fallback is not needed - they =
are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> ...and, based on the agreed scope, the Offerer KNOWS that the remote =
endpoint supports BUNDLE, so the "one implementation that doesn't handle =
it" will not be there :)=20

For WebRTC sure - but we were talking about using BUNDLE generically.  =
Or at least that's what I thought you were asking about.  If we're only =
talking about using BUNDLE for WebRTC, then not only do you not need to =
use a port=3D0, but you also don't need a second offer/answer, etc.


>> Also, from a purely protocol perspective, it seems quite logical to =
me to use a port number of 0 for an m-line you literally don't want to =
use if the far-end doesn't support=20
>> BUNDLE - a port number of 0 effectively means "disabled" today, and =
that's basically what you want. =20
>=20
> Well, one could also say that it means "Here are some m- lines  I want =
to bundle, with this port, but only if you (the remote endpoint) wants =
to use bundle." :)

We can certainly change the semantics of SDP offers with port=3D0 for =
WebRTC, but I think it would be really dangerous to change its semantics =
for general use.


>> What I think should happen is the UAS should send an SDP answer using =
port=3D0 for those same m-lines, and the UAC should then send a new =
offer with those m-lines having the real port number of the bundle =
transport, and the far-end should answer that in the same fashion.
>=20
> The problem is that the grouping framework does not allow, in SDP =
Answers, usage of port=3D0 in grouped m- lines. It would require an =
update of the RFC, and I think we want to avoid that.

Considering the number of docs the Unified-Plan proposes to either =
create, change, update or whatever... methinks having BUNDLE update the =
grouping framework RFC is rather a minor nit. :)

-hadriel



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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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This is a multi-part message in MIME format.
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On 8/8/13 01:18, Christer Holmberg wrote:
> As has been indicated, bundle-only can only be used when one knows 
> that the receiver, and possible intermediaries etc, support it.

That's not quite right.

Bundle-only can be sent in any offer. Even if you don't know anything 
about the remote endpoint or other devices along the path.

The bundle-only lines are designed to succeed only when the remote 
endpoint supports bundle. They are designed to fail in a very graceful, 
predictable, backwards-compatible, and well-defined way under other 
circumstances.

> So, based on that assumption, my question is: why do we need port 0? 
> Why not assign the suggested bundle port to the bundle-only m- lines 
> from the beginning? As the bundle port is anyway going to be used 
> (even if the receiver does not want to use bundle), there is no need 
> to reserve additional ports for bundle-only m- lines.

The point of the port-0 mechanism is that it will cause that m-line line 
to fail if you hit a non-bundle endpoint. If we put a valid port there 
that matches other m-lines, then legacy implementations behave in odd, 
unpredictable, and (most importantly) widely varying ways. The reaction 
to port 0 is better defined, and more consistent.

/a

--------------080906070500040606030809
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 8/8/13 01:18, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <font face="Calibri">As has been indicated, bundle-only can only
        be used when one knows that the receiver, and possible
        intermediaries etc, support it.</font></blockquote>
    <br>
    That's not quite right.<br>
    <br>
    Bundle-only can be sent in any offer. Even if you don't know
    anything about the remote endpoint or other devices along the path.<br>
    <br>
    The bundle-only lines are designed to succeed only when the remote
    endpoint supports bundle. They are designed to fail in a very
    graceful, predictable, backwards-compatible, and well-defined way
    under other circumstances.<br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se"
      type="cite">
      <div>&nbsp;<font face="Calibri">So, based on that assumption, my
          question is: why do we need port 0? Why not&nbsp;assign the
          suggested bundle port to the bundle-only m- lines from the
          beginning? As the bundle port is anyway going to be used (even
          if the receiver does not want to use bundle), there is no need
          to reserve additional ports for bundle-only m- lines.</font></div>
    </blockquote>
    <br>
    The point of the port-0 mechanism is that it will cause that m-line
    line to fail if you hit a non-bundle endpoint. If we put a valid
    port there that matches other m-lines, then legacy implementations
    behave in odd, unpredictable, and (most importantly) widely varying
    ways. The reaction to port 0 is better defined, and more consistent.<br>
    <br>
    /a<br>
  </body>
</html>

--------------080906070500040606030809--

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Hadriel Kaplan'" <hadriel.kaplan@oracle.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Cc: 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi Hadriel/Christer,

My understanding is that the initial offer with same port breaks more than
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
indicates one of the possible interop issue with existing mobile networks.
So, I don't agree with Hadriel claim of 99% system works with same port. The
current BUNDLE draft mechanism interop in the better way.

In case it is known to offerer that answerer supports BUNDLE before sending
offer, it is fine to send the same port in the offer using bundle-only
attribute. I'm not seeing any issue there.

Thanks
Partha





> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Thursday, August 08, 2013 7:46 PM
> To: Christer Holmberg
> Cc: mmusic
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
> 
> 
> On Aug 8, 2013, at 9:04 AM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
> 
> > Another advantage of using different port numbers is that it provides
> a fallback in case the remote endpoint does not support, or does not
> want to use BUNDLE.
> 
> Yes, except as far as I can tell all devices (and even intermediaries)
> handle an offer with the same port number just fine.  But that's just a
> personal observation, and dangerous to assume ALL of them handle it
> fine; that's why I didn't argue for long that we should use the same
> port number in the offer - because it concerned me that someone knew of
> at least one implementation that didn't handle it well. (not that we
> can truly accommodate ALL implementations anyway, but it's got to be a
> heck of a lot higher than simple majority, and as close to 100% as we
> can get)
> 
> 
> > However, for bundle-only m- lines such fallback is not needed - they
> are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> > ...and, based on the agreed scope, the Offerer KNOWS that the remote
> endpoint supports BUNDLE, so the "one implementation that doesn't
> handle it" will not be there :)
> 
> For WebRTC sure - but we were talking about using BUNDLE generically.
> Or at least that's what I thought you were asking about.  If we're only
> talking about using BUNDLE for WebRTC, then not only do you not need to
> use a port=0, but you also don't need a second offer/answer, etc.
> 
> 
> >> Also, from a purely protocol perspective, it seems quite logical to
> me to use a port number of 0 for an m-line you literally don't want to
> use if the far-end doesn't support
> >> BUNDLE - a port number of 0 effectively means "disabled" today, and
> that's basically what you want.
> >
> > Well, one could also say that it means "Here are some m- lines  I
> want to bundle, with this port, but only if you (the remote endpoint)
> wants to use bundle." :)
> 
> We can certainly change the semantics of SDP offers with port=0 for
> WebRTC, but I think it would be really dangerous to change its
> semantics for general use.
> 
> 
> >> What I think should happen is the UAS should send an SDP answer
> using port=0 for those same m-lines, and the UAC should then send a new
> offer with those m-lines having the real port number of the bundle
> transport, and the far-end should answer that in the same fashion.
> >
> > The problem is that the grouping framework does not allow, in SDP
> Answers, usage of port=0 in grouped m- lines. It would require an
> update of the RFC, and I think we want to avoid that.
> 
> Considering the number of docs the Unified-Plan proposes to either
> create, change, update or whatever... methinks having BUNDLE update the
> grouping framework RFC is rather a minor nit. :)
> 
> -hadriel
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 07:21, Hadriel Kaplan wrote:
> What I think should happen is the UAS should send an SDP answer using port=0 for those same m-lines

Why? You're proposing that we remove the mechanism for rejecting media 
streams. Consider the example I put up during the meeting: you have a 
regular audio m-line and a bundle-only video m-line. The other end wants 
to decline the video. You've just taken away the tool for doing so.

There'd better be a heck of an upside to removing this critical protocol 
mechanism. So far, the upside you've described is "Hadriel thinks it's 
not aesthetically pleasing," which doesn't seem to be a very practical 
consideration. Is there a technically compelling argument for your position?

/a

From christer.holmberg@ericsson.com  Thu Aug  8 11:19:28 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, 'Hadriel Kaplan' <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi,

Some people suggested that it should be allowed to send the same ports in t=
he first offer, IF it is known that the answerer supports BUNDLE.

However, that is currently not described in the draft, and there were also =
objections to such approach.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: &apos;Hadriel Kaplan&apos; [hadriel.kaplan@oracle.com]; Christer Holmbe=
rg [christer.holmberg@ericsson.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
Hi Hadriel/Christer,

My understanding is that the initial offer with same port breaks more than
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
indicates one of the possible interop issue with existing mobile networks.
So, I don't agree with Hadriel claim of 99% system works with same port. Th=
e
current BUNDLE draft mechanism interop in the better way.

In case it is known to offerer that answerer supports BUNDLE before sending
offer, it is fine to send the same port in the offer using bundle-only
attribute. I'm not seeing any issue there.

Thanks
Partha





> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Thursday, August 08, 2013 7:46 PM
> To: Christer Holmberg
> Cc: mmusic
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
>
>
> On Aug 8, 2013, at 9:04 AM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
>
> > Another advantage of using different port numbers is that it provides
> a fallback in case the remote endpoint does not support, or does not
> want to use BUNDLE.
>
> Yes, except as far as I can tell all devices (and even intermediaries)
> handle an offer with the same port number just fine.  But that's just a
> personal observation, and dangerous to assume ALL of them handle it
> fine; that's why I didn't argue for long that we should use the same
> port number in the offer - because it concerned me that someone knew of
> at least one implementation that didn't handle it well. (not that we
> can truly accommodate ALL implementations anyway, but it's got to be a
> heck of a lot higher than simple majority, and as close to 100% as we
> can get)
>
>
> > However, for bundle-only m- lines such fallback is not needed - they
> are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> > ...and, based on the agreed scope, the Offerer KNOWS that the remote
> endpoint supports BUNDLE, so the "one implementation that doesn't
> handle it" will not be there :)
>
> For WebRTC sure - but we were talking about using BUNDLE generically.
> Or at least that's what I thought you were asking about.  If we're only
> talking about using BUNDLE for WebRTC, then not only do you not need to
> use a port=3D0, but you also don't need a second offer/answer, etc.
>
>
> >> Also, from a purely protocol perspective, it seems quite logical to
> me to use a port number of 0 for an m-line you literally don't want to
> use if the far-end doesn't support
> >> BUNDLE - a port number of 0 effectively means "disabled" today, and
> that's basically what you want.
> >
> > Well, one could also say that it means "Here are some m- lines  I
> want to bundle, with this port, but only if you (the remote endpoint)
> wants to use bundle." :)
>
> We can certainly change the semantics of SDP offers with port=3D0 for
> WebRTC, but I think it would be really dangerous to change its
> semantics for general use.
>
>
> >> What I think should happen is the UAS should send an SDP answer
> using port=3D0 for those same m-lines, and the UAC should then send a new
> offer with those m-lines having the real port number of the bundle
> transport, and the far-end should answer that in the same fashion.
> >
> > The problem is that the grouping framework does not allow, in SDP
> Answers, usage of port=3D0 in grouped m- lines. It would require an
> update of the RFC, and I think we want to avoid that.
>
> Considering the number of docs the Unified-Plan proposes to either
> create, change, update or whatever... methinks having BUNDLE update the
> grouping framework RFC is rather a minor nit. :)
>
> -hadriel
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Some people suggested that it should be allowed=
 to send the same ports in the first offer, IF it is known that the answere=
r supports BUNDLE.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">However, that is currently not described in the=
 draft, and there were also objections to such approach.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> &amp;apos;Hadriel Kaplan&amp;apos; [hadriel.kaplan@oracle.com]; =
Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Hi Hadriel/Christer,<br>
<br>
My understanding is that the initial offer with same port breaks more than<=
br>
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft<br>
indicates one of the possible interop issue with existing mobile networks.<=
br>
So, I don't agree with Hadriel claim of 99% system works with same port. Th=
e<br>
current BUNDLE draft mechanism interop in the better way.<br>
<br>
In case it is known to offerer that answerer supports BUNDLE before sending=
<br>
offer, it is fine to send the same port in the offer using bundle-only<br>
attribute. I'm not seeing any issue there.<br>
<br>
Thanks<br>
Partha<br>
<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounces@ietf.o=
rg">mailto:mmusic-bounces@ietf.org</a>] On<br>
&gt; Behalf Of Hadriel Kaplan<br>
&gt; Sent: Thursday, August 08, 2013 7:46 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: mmusic<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-=
<br>
&gt; only m- lines?<br>
&gt; <br>
&gt; <br>
&gt; On Aug 8, 2013, at 9:04 AM, Christer Holmberg<br>
&gt; &lt;christer.holmberg@ericsson.com&gt; wrote:<br>
&gt; <br>
&gt; &gt; Another advantage of using different port numbers is that it prov=
ides<br>
&gt; a fallback in case the remote endpoint does not support, or does not<b=
r>
&gt; want to use BUNDLE.<br>
&gt; <br>
&gt; Yes, except as far as I can tell all devices (and even intermediaries)=
<br>
&gt; handle an offer with the same port number just fine.&nbsp; But that's =
just a<br>
&gt; personal observation, and dangerous to assume ALL of them handle it<br=
>
&gt; fine; that's why I didn't argue for long that we should use the same<b=
r>
&gt; port number in the offer - because it concerned me that someone knew o=
f<br>
&gt; at least one implementation that didn't handle it well. (not that we<b=
r>
&gt; can truly accommodate ALL implementations anyway, but it's got to be a=
<br>
&gt; heck of a lot higher than simple majority, and as close to 100% as we<=
br>
&gt; can get)<br>
&gt; <br>
&gt; <br>
&gt; &gt; However, for bundle-only m- lines such fallback is not needed - t=
hey<br>
&gt; are ONLY going to be used IF the remote endpoint chooses to use BUNDLE=
.<br>
&gt; &gt; ...and, based on the agreed scope, the Offerer KNOWS that the rem=
ote<br>
&gt; endpoint supports BUNDLE, so the &quot;one implementation that doesn't=
<br>
&gt; handle it&quot; will not be there :)<br>
&gt; <br>
&gt; For WebRTC sure - but we were talking about using BUNDLE generically.<=
br>
&gt; Or at least that's what I thought you were asking about.&nbsp; If we'r=
e only<br>
&gt; talking about using BUNDLE for WebRTC, then not only do you not need t=
o<br>
&gt; use a port=3D0, but you also don't need a second offer/answer, etc.<br=
>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; Also, from a purely protocol perspective, it seems quite logi=
cal to<br>
&gt; me to use a port number of 0 for an m-line you literally don't want to=
<br>
&gt; use if the far-end doesn't support<br>
&gt; &gt;&gt; BUNDLE - a port number of 0 effectively means &quot;disabled&=
quot; today, and<br>
&gt; that's basically what you want.<br>
&gt; &gt;<br>
&gt; &gt; Well, one could also say that it means &quot;Here are some m- lin=
es&nbsp; I<br>
&gt; want to bundle, with this port, but only if you (the remote endpoint)<=
br>
&gt; wants to use bundle.&quot; :)<br>
&gt; <br>
&gt; We can certainly change the semantics of SDP offers with port=3D0 for<=
br>
&gt; WebRTC, but I think it would be really dangerous to change its<br>
&gt; semantics for general use.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; What I think should happen is the UAS should send an SDP answ=
er<br>
&gt; using port=3D0 for those same m-lines, and the UAC should then send a =
new<br>
&gt; offer with those m-lines having the real port number of the bundle<br>
&gt; transport, and the far-end should answer that in the same fashion.<br>
&gt; &gt;<br>
&gt; &gt; The problem is that the grouping framework does not allow, in SDP=
<br>
&gt; Answers, usage of port=3D0 in grouped m- lines. It would require an<br=
>
&gt; update of the RFC, and I think we want to avoid that.<br>
&gt; <br>
&gt; Considering the number of docs the Unified-Plan proposes to either<br>
&gt; create, change, update or whatever... methinks having BUNDLE update th=
e<br>
&gt; grouping framework RFC is rather a minor nit. :)<br>
&gt; <br>
&gt; -hadriel<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
<br>
</div>
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From hadriel.kaplan@oracle.com  Thu Aug  8 11:22:11 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 1:39 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 07:21, Hadriel Kaplan wrote:
>> What I think should happen is the UAS should send an SDP answer using =
port=3D0 for those same m-lines
>=20
> Why? You're proposing that we remove the mechanism for rejecting media =
streams. Consider the example I put up during the meeting: you have a =
regular audio m-line and a bundle-only video m-line. The other end wants =
to decline the video. You've just taken away the tool for doing so.
>=20
> There'd better be a heck of an upside to removing this critical =
protocol mechanism. So far, the upside you've described is "Hadriel =
thinks it's not aesthetically pleasing," which doesn't seem to be a very =
practical consideration. Is there a technically compelling argument for =
your position?

Huh?  You either misunderstood my email, or I'm not following your =
statement above.  I don't believe I'm proposing we remove the mechanism =
of rejecting media streams.  How/where/when did I propose that?  You =
lost me.

And I don't think what I'm suggesting is to be "aesthetically pleasing" =
either... although SDP-based ascii art might be kinda cool.

-hadriel


From christer.holmberg@ericsson.com  Thu Aug  8 11:24:11 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
Thread-Index: AQHOlDHLQN7nRiDYV02hbFJeK66gpJmLcqaAgAAt4U0=
Date: Thu, 8 Aug 2013 18:24:02 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

I agree with Adam on this one. Port zero is used in answers to reject a str=
eam, and I don't think we should change that.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Adam Roach [adam@nostrum.com]
To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic@ietf=
.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
On 8/8/13 07:21, Hadriel Kaplan wrote:
> What I think should happen is the UAS should send an SDP answer using por=
t=3D0 for those same m-lines

Why? You're proposing that we remove the mechanism for rejecting media
streams. Consider the example I put up during the meeting: you have a
regular audio m-line and a bundle-only video m-line. The other end wants
to decline the video. You've just taken away the tool for doing so.

There'd better be a heck of an upside to removing this critical protocol
mechanism. So far, the upside you've described is "Hadriel thinks it's
not aesthetically pleasing," which doesn't seem to be a very practical
consideration. Is there a technically compelling argument for your position=
?

/a

--_000_7594FB04B1934943A5C02806D1A2204B1C422522ESESSMB209erics_
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1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
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<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I agree with Adam on this one. Port zero is use=
d in answers to reject a stream, and I don't think we should change that.</=
font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Adam Roach [adam@nostrum.com]<br>
<b>To:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmus=
ic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 8/8/13 07:21, Hadriel Kaplan wrote:<br>
&gt; What I think should happen is the UAS should send an SDP answer using =
port=3D0 for those same m-lines<br>
<br>
Why? You're proposing that we remove the mechanism for rejecting media <br>
streams. Consider the example I put up during the meeting: you have a <br>
regular audio m-line and a bundle-only video m-line. The other end wants <b=
r>
to decline the video. You've just taken away the tool for doing so.<br>
<br>
There'd better be a heck of an upside to removing this critical protocol <b=
r>
mechanism. So far, the upside you've described is &quot;Hadriel thinks it's=
 <br>
not aesthetically pleasing,&quot; which doesn't seem to be a very practical=
 <br>
consideration. Is there a technically compelling argument for your position=
?<br>
<br>
/a<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422522ESESSMB209erics_--

From christer.holmberg@ericsson.com  Thu Aug  8 11:32:56 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
Thread-Index: AQHOlFprBnOgSvDKakySvPl07qCSvZmLoqjt
Date: Thu, 8 Aug 2013 18:32:48 +0000
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi Adam,

Ok, so if we make no assumptions about the receiver, I understand the reaso=
ning for port zero.

However, even if both endpoints support bundle-only, what if there are inte=
rmediaries that don't? They may pass the SDP, but unless you send a second =
offer, with the bundle port, I doubt they will forward any media associated=
 with the bundle-only m- lines towards the Offerer, as they only see a port=
 zero value.

Or, do you assume that intermediaries that don't support bundle-only will r=
eject the first offer?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Adam Roach [adam@nostrum.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
On 8/8/13 01:18, Christer Holmberg wrote:
As has been indicated, bundle-only can only be used when one knows that the=
 receiver, and possible intermediaries etc, support it.

That's not quite right.

Bundle-only can be sent in any offer. Even if you don't know anything about=
 the remote endpoint or other devices along the path.

The bundle-only lines are designed to succeed only when the remote endpoint=
 supports bundle. They are designed to fail in a very graceful, predictable=
, backwards-compatible, and well-defined way under other circumstances.

 So, based on that assumption, my question is: why do we need port 0? Why n=
ot assign the suggested bundle port to the bundle-only m- lines from the be=
ginning? As the bundle port is anyway going to be used (even if the receive=
r does not want to use bundle), there is no need to reserve additional port=
s for bundle-only m- lines.

The point of the port-0 mechanism is that it will cause that m-line line to=
 fail if you hit a non-bundle endpoint. If we put a valid port there that m=
atches other m-lines, then legacy implementations behave in odd, unpredicta=
ble, and (most importantly) widely varying ways. The reaction to port 0 is =
better defined, and more consistent.

/a

--_000_7594FB04B1934943A5C02806D1A2204B1C422564ESESSMB209erics_
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</head>
<body bgcolor=3D"#FFFFFF">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Adam,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Ok, so if we make no assumptions about the rece=
iver, I understand the reasoning for port zero.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">However, even if both endpoints support bundle-=
only, what if there are intermediaries that don't? They may pass the SDP, b=
ut unless you send a second offer, with the bundle port, I doubt they will =
forward any media&nbsp;associated with&nbsp;the
 bundle-only m- lines towards the Offerer, as they only see a port zero val=
ue.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Or, do you assume that intermediaries that don'=
t support bundle-only will reject the first offer?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Adam Roach [adam@nostrum.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span>
<div>
<div class=3D"moz-cite-prefix">On 8/8/13 01:18, Christer Holmberg wrote:<br=
>
</div>
<blockquote type=3D"cite"><font face=3D"Calibri">As has been indicated, bun=
dle-only can only be used when one knows that the receiver, and possible in=
termediaries etc, support it.</font></blockquote>
<br>
That's not quite right.<br>
<br>
Bundle-only can be sent in any offer. Even if you don't know anything about=
 the remote endpoint or other devices along the path.<br>
<br>
The bundle-only lines are designed to succeed only when the remote endpoint=
 supports bundle. They are designed to fail in a very graceful, predictable=
, backwards-compatible, and well-defined way under other circumstances.<br>
<br>
<blockquote type=3D"cite">
<div>&nbsp;<font face=3D"Calibri">So, based on that assumption, my question=
 is: why do we need port 0? Why not&nbsp;assign the suggested bundle port t=
o the bundle-only m- lines from the beginning? As the bundle port is anyway=
 going to be used (even if the receiver does
 not want to use bundle), there is no need to reserve additional ports for =
bundle-only m- lines.</font></div>
</blockquote>
<br>
The point of the port-0 mechanism is that it will cause that m-line line to=
 fail if you hit a non-bundle endpoint. If we put a valid port there that m=
atches other m-lines, then legacy implementations behave in odd, unpredicta=
ble, and (most importantly) widely
 varying ways. The reaction to port 0 is better defined, and more consisten=
t.<br>
<br>
/a<br>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422564ESESSMB209erics_--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Date: Thu, 8 Aug 2013 18:40:56 +0000
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi Hadriel,

My opinion is also that intermediaries will handle offers with the same por=
t value. But, there were other opinions, so we agreed on the two-O/A mechan=
ism.

Regarding the modified semantics of port zero in bundle-only m- lines, if w=
e want to change the semantics only for WebRTC, it means we can only allow =
bundle-only for WebRTC. Because, I don't think we want to have different se=
mantics for port zero bundle-only m- lines depending on whether they are us=
ed in WebRTC or not...

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

On Aug 8, 2013, at 9:04 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Another advantage of using different port numbers is that it provides a f=
allback in case the remote endpoint does not support, or does not want to u=
se BUNDLE.

Yes, except as far as I can tell all devices (and even intermediaries) hand=
le an offer with the same port number just fine.  But that's just a persona=
l observation, and dangerous to assume ALL of them handle it fine; that's w=
hy I didn't argue for long that we should use the same port number in the o=
ffer - because it concerned me that someone knew of at least one implementa=
tion that didn't handle it well. (not that we can truly accommodate ALL imp=
lementations anyway, but it's got to be a heck of a lot higher than simple =
majority, and as close to 100% as we can get)


> However, for bundle-only m- lines such fallback is not needed - they are =
ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> ...and, based on the agreed scope, the Offerer KNOWS that the remote endp=
oint supports BUNDLE, so the "one implementation that doesn't handle it" wi=
ll not be there :)

For WebRTC sure - but we were talking about using BUNDLE generically.  Or a=
t least that's what I thought you were asking about.  If we're only talking=
 about using BUNDLE for WebRTC, then not only do you not need to use a port=
=3D0, but you also don't need a second offer/answer, etc.


>> Also, from a purely protocol perspective, it seems quite logical to me t=
o use a port number of 0 for an m-line you literally don't want to use if t=
he far-end doesn't support
>> BUNDLE - a port number of 0 effectively means "disabled" today, and that=
's basically what you want.
>
> Well, one could also say that it means "Here are some m- lines  I want to=
 bundle, with this port, but only if you (the remote endpoint) wants to use=
 bundle." :)

We can certainly change the semantics of SDP offers with port=3D0 for WebRT=
C, but I think it would be really dangerous to change its semantics for gen=
eral use.


>> What I think should happen is the UAS should send an SDP answer using po=
rt=3D0 for those same m-lines, and the UAC should then send a new offer wit=
h those m-lines having the real port number of the bundle transport, and th=
e far-end should answer that in the same fashion.
>
> The problem is that the grouping framework does not allow, in SDP Answers=
, usage of port=3D0 in grouped m- lines. It would require an update of the =
RFC, and I think we want to avoid that.

Considering the number of docs the Unified-Plan proposes to either create, =
change, update or whatever... methinks having BUNDLE update the grouping fr=
amework RFC is rather a minor nit. :)

-hadriel



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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Hadriel,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">My opinion is also that intermediaries will han=
dle offers with the same port value. But, there were other opinions, so we =
agreed on the two-O/A mechanism.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regarding the modified&nbsp;semantics of port z=
ero in bundle-only m- lines, if we want to change the semantics only for We=
bRTC, it means we can only allow bundle-only for WebRTC. Because, I don't t=
hink we want to have different semantics
 for port zero&nbsp;bundle-only m- lines depending on whether they are used=
 in WebRTC or not...</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 8, 2013, at 9:04 AM, Christer Holmberg &lt;christer.holmberg@ericsso=
n.com&gt; wrote:<br>
<br>
&gt; Another advantage of using different port numbers is that it provides =
a fallback in case the remote endpoint does not support, or does not want t=
o use BUNDLE.<br>
<br>
Yes, except as far as I can tell all devices (and even intermediaries) hand=
le an offer with the same port number just fine.&nbsp; But that's just a pe=
rsonal observation, and dangerous to assume ALL of them handle it fine; tha=
t's why I didn't argue for long that
 we should use the same port number in the offer - because it concerned me =
that someone knew of at least one implementation that didn't handle it well=
. (not that we can truly accommodate ALL implementations anyway, but it's g=
ot to be a heck of a lot higher
 than simple majority, and as close to 100% as we can get)<br>
<br>
<br>
&gt; However, for bundle-only m- lines such fallback is not needed - they a=
re ONLY going to be used IF the remote endpoint chooses to use BUNDLE.<br>
&gt; ...and, based on the agreed scope, the Offerer KNOWS that the remote e=
ndpoint supports BUNDLE, so the &quot;one implementation that doesn't handl=
e it&quot; will not be there :)
<br>
<br>
For WebRTC sure - but we were talking about using BUNDLE generically.&nbsp;=
 Or at least that's what I thought you were asking about.&nbsp; If we're on=
ly talking about using BUNDLE for WebRTC, then not only do you not need to =
use a port=3D0, but you also don't need a second
 offer/answer, etc.<br>
<br>
<br>
&gt;&gt; Also, from a purely protocol perspective, it seems quite logical t=
o me to use a port number of 0 for an m-line you literally don't want to us=
e if the far-end doesn't support
<br>
&gt;&gt; BUNDLE - a port number of 0 effectively means &quot;disabled&quot;=
 today, and that's basically what you want.&nbsp;
<br>
&gt; <br>
&gt; Well, one could also say that it means &quot;Here are some m- lines&nb=
sp; I want to bundle, with this port, but only if you (the remote endpoint)=
 wants to use bundle.&quot; :)<br>
<br>
We can certainly change the semantics of SDP offers with port=3D0 for WebRT=
C, but I think it would be really dangerous to change its semantics for gen=
eral use.<br>
<br>
<br>
&gt;&gt; What I think should happen is the UAS should send an SDP answer us=
ing port=3D0 for those same m-lines, and the UAC should then send a new off=
er with those m-lines having the real port number of the bundle transport, =
and the far-end should answer that in the
 same fashion.<br>
&gt; <br>
&gt; The problem is that the grouping framework does not allow, in SDP Answ=
ers, usage of port=3D0 in grouped m- lines. It would require an update of t=
he RFC, and I think we want to avoid that.<br>
<br>
Considering the number of docs the Unified-Plan proposes to either create, =
change, update or whatever... methinks having BUNDLE update the grouping fr=
amework RFC is rather a minor nit. :)<br>
<br>
-hadriel<br>
<br>
<br>
</div>
</span></font>
</body>
</html>

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From hadriel.kaplan@oracle.com  Thu Aug  8 11:42:53 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Parthasarathi R <partha@parthasarathi.co.in>
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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On Aug 8, 2013, at 1:30 PM, Parthasarathi R <partha@parthasarathi.co.in> =
wrote:

> My understanding is that the initial offer with same port breaks more =
than
> one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
> indicates one of the possible interop issue with existing mobile =
networks.

I've only skimmed the draft, but I don't see the problem.  I know =
there's an issue with multiplexing all the RTP on one 5-tuple, but =
that's actually an orthogonal issue.  I was talking about whether an SDP =
offer with multiple m-lines using the same port number would have issues =
or not.  Am I missing something in the draft?


> So, I don't agree with Hadriel claim of 99% system works with same =
port. The
> current BUNDLE draft mechanism interop in the better way.

Please don't misrepresent what I said - I think if you read my =
paragraph, it's pretty clear I am not claiming 99% of them work with the =
same port; nor am I suggesting we go back to using the same port again.  =
It was in the context of the debates we had back when the issue was =
debated a couple IETF meetings ago, and I said as far as I knew they can =
handle an *SDP offer* with it... and I then *immediately* said I didn't =
argue it for long, because someone else stated they knew of a device =
which had a problem, and that concerned me.  Since most people generally =
try not to contradict themselves in the span of a dozen words, I think =
it's reasonable to assume I no longer believe all devices work.  I mean =
I may not be smart, but I'm not a complete idiot. (well... not generally =
an idiot anyway)

-hadriel


From christer.holmberg@ericsson.com  Thu Aug  8 11:45:34 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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When both Hadriel and Adam think pot zero in bundle-only m- lines is good, =
I'll rest my case :)

However, I do still think we need the second offer also for bundle-only m- =
lines.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

If I understand your question, you're asking why bundle-only m-lines would =
use a port number of 0 in the offer, instead of using the same port number =
as the regular bundle m-line they're going to transport-bundle with?

I believe the rationale was basically argued about before, either in IETF 8=
6 Orlando or IETF 85 Atlanta.  It gets back to the original debate about wh=
ether a BUNDLE offer should use the same port numbers or different ones, fo=
r bundle m-lines (what was called "BUNDLE vs. CRUNDLE" if I recall correctl=
y).

One WG participant argued against using the same port numbers for bundle li=
nes.  His argument was that since it was never described in an RFC, and he =
knew of one implementation that didn't handle it well, that therefore we co=
uldn't send an offer using the same port numbers until we found out if the =
far-end supported BUNDLE or not.

That's what drove the decision to use different port numbers even in regula=
r bundle cases, afaik.

Therefore, even for bundle-only lines, we can't use the same non-0 port num=
ber for them, because the same theoretical issue will exist.

Also, from a purely protocol perspective, it seems quite logical to me to u=
se a port number of 0 for an m-line you literally don't want to use if the =
far-end doesn't support BUNDLE - a port number of 0 effectively means "disa=
bled" today, and that's basically what you want.  What I don't think is rig=
ht, is (1) having the far-end send an SDP answer back that sets a real port=
 number for those same "disabled" m-lines, and (2) not requiring another of=
fer/answer exchange that then uses the correct (same) port numbers for the =
previously-disabled m-lines.

What I think should happen is the UAS should send an SDP answer using port=
=3D0 for those same m-lines, and the UAC should then send a new offer with =
those m-lines having the real port number of the bundle transport, and the =
far-end should answer that in the same fashion.

-hadriel


On Aug 8, 2013, at 2:18 AM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Hi,
>
> As has been indicated, bundle-only can only be used when one knows that t=
he receiver, and possible intermediaries etc, support it.
>
> So, based on that assumption, my question is: why do we need port 0? Why =
not assign the suggested bundle port to the bundle-only m- lines from the b=
eginning? As the bundle port is anyway going to be used (even if the receiv=
er does not want to use bundle), there is no need to reserve additional por=
ts for bundle-only m- lines.
>
> Regards,
>
> Christer
>
>
>
> Sent from Windows using TouchDown (www.nitrodesk.com<http://www.nitrodesk=
.com>)
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
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<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">When both Hadriel and Adam&nbsp;think pot zero in bun=
dle-only m- lines is good, I'll rest my case :)</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">However, I do still think we need the second of=
fer also for bundle-only m- lines.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
If I understand your question, you're asking why bundle-only m-lines would =
use a port number of 0 in the offer, instead of using the same port number =
as the regular bundle m-line they're going to transport-bundle with?<br>
<br>
I believe the rationale was basically argued about before, either in IETF 8=
6 Orlando or IETF 85 Atlanta.&nbsp; It gets back to the original debate abo=
ut whether a BUNDLE offer should use the same port numbers or different one=
s, for bundle m-lines (what was called
 &quot;BUNDLE vs. CRUNDLE&quot; if I recall correctly).<br>
<br>
One WG participant argued against using the same port numbers for bundle li=
nes.&nbsp; His argument was that since it was never described in an RFC, an=
d he knew of one implementation that didn't handle it well, that therefore =
we couldn't send an offer using the same
 port numbers until we found out if the far-end supported BUNDLE or not.<br=
>
<br>
That's what drove the decision to use different port numbers even in regula=
r bundle cases, afaik.<br>
<br>
Therefore, even for bundle-only lines, we can't use the same non-0 port num=
ber for them, because the same theoretical issue will exist.<br>
<br>
Also, from a purely protocol perspective, it seems quite logical to me to u=
se a port number of 0 for an m-line you literally don't want to use if the =
far-end doesn't support BUNDLE - a port number of 0 effectively means &quot=
;disabled&quot; today, and that's basically
 what you want.&nbsp; What I don't think is right, is (1) having the far-en=
d send an SDP answer back that sets a real port number for those same &quot=
;disabled&quot; m-lines, and (2) not requiring another offer/answer exchang=
e that then uses the correct (same) port numbers
 for the previously-disabled m-lines.<br>
<br>
What I think should happen is the UAS should send an SDP answer using port=
=3D0 for those same m-lines, and the UAC should then send a new offer with =
those m-lines having the real port number of the bundle transport, and the =
far-end should answer that in the same
 fashion.<br>
<br>
-hadriel<br>
<br>
<br>
On Aug 8, 2013, at 2:18 AM, Christer Holmberg &lt;christer.holmberg@ericsso=
n.com&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;&nbsp; <br>
&gt; As has been indicated, bundle-only can only be used when one knows tha=
t the receiver, and possible intermediaries etc, support it.<br>
&gt;&nbsp; <br>
&gt; So, based on that assumption, my question is: why do we need port 0? W=
hy not assign the suggested bundle port to the bundle-only m- lines from th=
e beginning? As the bundle port is anyway going to be used (even if the rec=
eiver does not want to use bundle),
 there is no need to reserve additional ports for bundle-only m- lines.<br>
&gt;&nbsp; <br>
&gt; Regards,<br>
&gt;&nbsp; <br>
&gt; Christer<br>
&gt;&nbsp; <br>
&gt; <br>
&gt; <br>
&gt; Sent from Windows using TouchDown (<a href=3D"http://www.nitrodesk.com=
">www.nitrodesk.com</a>)<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C4225D0ESESSMB209erics_--

From kase.saylor@swri.org  Thu Aug  8 11:50:16 2013
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From: "Saylor, Kase J." <kase.saylor@swri.org>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: RTSP for data recorder design
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Subject: [MMUSIC] RTSP for data recorder design
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To Whom it may concern,

I am helping the US Army develop standards for a network recorder on US Arm=
y vehicles. We like to use available standards when at all possible, and my=
 research seems to indicate that RTSP would be a great candidate standard. =
I've been poring over the RTSP specification (along with the SDP standard) =
and I have your douce code for LiveMedia. However, I still have a few quest=
ions concerning the RTSP standard and its use for our particular need. It s=
eems like there is plenty of examples on the Internet for the PLAY method, =
but not much for RECORD. I really hope that someone may be able to provide =
some insight that will get me moving in the right direction. On to the ques=
tions!

Our recorder requires the ability to "nominate" data streams for recording,=
 so I believe the correct RTSP method is ANNOUNCE. My first question is whe=
n I was to add a stream using ANNOUNCE, if I use the same Session number fo=
r multiple streams, does this basically create a presentation? For example:

C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
           CSeq: 3
           Date: 7 Aug 2013 15:35:06 GMT
           Session: 47112344
           Content-Type: application/sdp
           Content-Length: 332

           v=3D0
           o=3D- 2890844526 2890845468 IN IP4 192.168.0.123
           s=3DThreat Stream 1
           i=3DThreat stream from threat service on the Boomerang
           c=3DIN IP4 239.192.0.123/127
           t=3D0 0
           a=3Dunique_id:threat1
   a=3Ddata_type:threat
           m=3Dmessage 12345

(some response back from server)

C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
           CSeq: 4
           Date: 7 Aug 2013 15:35:07 GMT
           Session: 47112344
           Content-Type: application/sdp
           Content-Length: 332

           v=3D0
           o=3D- 2890844527 2890845471 IN IP4 192.168.0.123
           s=3DPosition Stream 1
           i=3DPosition stream from position service
           c=3DIN IP4 239.192.0.124/127
           t=3D0 0
           a=3Dunique_id:pos1
   a=3Ddata_type:position
           m=3Dmessage 12346
(some response back from server)

Based upon these two announce messages, is it safe to assume that the follo=
wing command would cause the recorder to start recording those two data str=
eams (assuming the current date/time is 8/4/2013 1530 GMT)?

     C->S: RECORD rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP=
/1.0
           CSeq: 5
           Content-Type: text/parameters
           Session: 47112344
   Range: clock=3D20130804T153000Z-

Furthermore, would the following command stop the recorder (assuming the cu=
rrent date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in the RFC for=
 stopping a recording.

     C->S: RECORD rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP=
/1.0
           CSeq: 6
           Content-Type: text/parameters
           Session: 47112344
   Range: clock=3D- 20130804T153200Z


I apologize but I have at least one or two more questions ;-) After tearing=
 down a session, how would one go back and playback recorded data? From my =
RECORD example I've assumed that providing a URI with a "path" based upon r=
ecord start time, could be one way to mark recording events. Do you think t=
his is an acceptable approach? Then the play command would something like:

     C->S: PLAY rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1=
.0
           CSeq: 6
           Content-Type: text/parameters
           Session: 1234567

I suppose I could specify unique parameters and use the GET_PARAMETER metho=
d to list the available recordings. Perhaps something like:

     C->S: SETUP rtsp://192.168.0.1/mission_recorder RTSP/1.0
           CSeq: 1
   Transport: unicast;client_port=3D4588-4589

     S->C: RTSP/1.0 200 OK
           CSeq: 1
   Date: 5 Aug 2013 11:15:07 GMT
           Session: 1234567
           Transport: unicast;client_port=3D4588-4589;server_port=3D6256-62=
57

     C->S: GET_PARAMETER rtsp://192.168.0.1/mission_recorder RTSP/1.0
           CSeq: 7
           Content-Type: text/parameters
           Session: 1234567
           Content-Length: 15

   list_recorded_data


     S->C: RTSP/1.0 200 OK
           CSeq: 7
   Date: 5 Aug 2013 11:15:10 GMT
           Content-Length: 46
           Content-Type: text/parameters

           list_recorded_data:20130804T153000Z,20130804T213000Z

     C->S: TEARDOWN rtsp://192.168.0.1/mission_recorder RTSP/1.0
           CSeq: 8
   Session: 1234567

     S->C: RTSP/1.0 200 OK
           CSeq: 8


Then we'd specify the proper way to interpret the response from the server =
and open up a new RTSP session to playback the recorded data. For example:

     C->S: SETUP rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/=
1.0
           CSeq: 1
   Transport: multicast;client_port=3D4588-4589

     S->C: RTSP/1.0 200 OK
           CSeq: 1
   Date: 6 Aug 2013 11:15:07 GMT
           Session: 7654321
           Transport: multicast;client_port=3D4588-4589;server_port=3D6256-=
6257

     C->S: PLAY rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1=
.0
           CSeq: 2
           Content-Type: text/parameters
           Session: 7654321

     S->C: RTSP/1.0 200 OK
           CSeq: 2
   Date: 6 Aug 2013 11:15:10 GMT

Okay, last question, would the DESCRIBE method for rtsp://192.168.0.1/missi=
on_recorder/20130804T153000Z be expected to describe the "streams" that wer=
e recorded? In other words could the following be expected?

C->S: DESCRIBE rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.=
0
           CSeq: 312
           Accept: application/sdp, application/rtsl, application/mheg

S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 7 Aug 2013 15:35:07 GMT
           Content-Type: application/sdp
           Content-Length: 332

           v=3D0
           o=3D- 2890844527 2890845471 IN IP4 192.168.0.123
           s=3DPosition Stream 1
           i=3DThreat stream from threat service on the Boomerang
           c=3DIN IP4 239.192.0.124/127
           t=3D0 0
           a=3Dunique_id:pos1
   a=3Ddata_type:position
           m=3Dmessage 12346
   s=3DPosition Stream 1
           i=3DPosition stream from position service
           c=3DIN IP4 239.192.0.124/127
           t=3D0 0
           a=3Dunique_id:pos1
   a=3Ddata_type:position
           m=3Dmessage 12346

I really have no idea how multiple streams would be described!

Thanks in advance for any help!
--
Regards,

Kase Saylor
Southwest Research Institute


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
To Whom it may concern,
<div><br>
</div>
<div>I am helping the US Army develop standards for a network recorder on U=
S Army vehicles. We like to use available standards when at all possible, a=
nd my research seems to indicate that RTSP would be a great candidate stand=
ard. I've been poring over the RTSP
 specification (along with the SDP standard) and I have your douce code for=
 LiveMedia. However, I still have a few questions concerning the RTSP stand=
ard and its use for our particular need. It seems like there is plenty of e=
xamples on the Internet for the
 PLAY method, but not much for RECORD. I really hope that someone may be ab=
le to provide some insight that will get me moving in the right direction. =
On to the questions!</div>
<div><br>
</div>
<div>Our recorder requires the ability to &quot;nominate&quot; data streams=
 for recording, so I believe the correct RTSP method is ANNOUNCE. My first =
question is when I was to add a stream using ANNOUNCE, if I use the same Se=
ssion number for multiple streams, does this
 basically create a presentation? For example:</div>
<div><br>
</div>
<div>C-&gt;S: ANNOUNCE&nbsp;<a href=3D"rtsp://192.168.0.1/mission_recorder"=
>rtsp://192.168.0.1/mission_recorder</a>&nbsp;RTSP/1.0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 3<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Date: 7 Aug 2013 15:35:06 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session: 47112344<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: application/sdp<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Length: 332<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;v=3D0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;o=3D- 2890844526 2890845468 IN IP4=
 192.168.0.123<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;s=3DThreat Stream 1<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;i=3DThreat stream from threat serv=
ice on the Boomerang<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;c=3DIN IP4 239.192.0.123/127<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;t=3D0 0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a=3Dunique_id:threat1<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;a=3Ddata_type:threat<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;m=3Dmessage 12345</div>
<div><br>
</div>
<div>(some response back from server)</div>
<div><br>
</div>
<div>C-&gt;S: ANNOUNCE&nbsp;<a href=3D"rtsp://192.168.0.1/mission_recorder"=
>rtsp://192.168.0.1/mission_recorder</a>&nbsp;RTSP/1.0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 4<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Date: 7 Aug 2013 15:35:07 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session: 47112344<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: application/sdp<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Length: 332<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;v=3D0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;o=3D- 2890844527 2890845471 IN IP4=
 192.168.0.123<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;s=3DPosition Stream 1<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;i=3DPosition stream from position =
service<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;c=3DIN IP4 239.192.0.124/127<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;t=3D0 0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a=3Dunique_id:pos1<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;a=3Ddata_type:position<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;m=3Dmessage 12346</div>
<div>(some response back from server)</div>
<div><br>
</div>
<div>Based upon these two announce messages, is it safe to assume that the =
following command would cause the recorder to start recording those two dat=
a streams (assuming the current date/time is 8/4/2013 1530 GMT)?</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: RECORD&nbsp;<a href=3D"rtsp://192.168.0.1=
/mission_recorder/">rtsp://192.168.0.1/mission_recorder/</a>20130804T153000=
Z&nbsp;RTSP/1.0</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 5<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session:&nbsp;47112344<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;Range: clock=3D20130804T153000Z-</div>
<div><br>
</div>
<div>Furthermore, would the following command stop the recorder&nbsp;(assum=
ing the current date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in t=
he RFC for stopping a recording.</div>
<div><br>
</div>
<div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: RECORD&nbsp;<a href=3D"rtsp://192.168.0.1=
/mission_recorder/">rtsp://192.168.0.1/mission_recorder/</a>20130804T153000=
Z&nbsp;RTSP/1.0</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 6<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session:&nbsp;47112344<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;Range: clock=3D- 20130804T153200Z</div>
<div><br>
</div>
<div><br>
</div>
<div>I apologize but I have at least one or two more questions ;-) After te=
aring down a session, how would one go back and playback recorded data? Fro=
m my RECORD example I've assumed that providing a URI with a &quot;path&quo=
t; based upon record start time, could be
 one way to mark recording events. Do you think this is an acceptable appro=
ach? Then the play command would something like:</div>
<div><br>
</div>
<div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: PLAY&nbsp;<a href=3D"rtsp://192.168.0.1/m=
ission_recorder/20130804T153000Z">rtsp://192.168.0.1/mission_recorder/20130=
804T153000Z</a>&nbsp;RTSP/1.0</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 6<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session:&nbsp;1234567<br>
</div>
</div>
<div><br>
</div>
<div>I suppose I could specify unique parameters and use the GET_PARAMETER =
method to list the available recordings. Perhaps something like:</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: SETUP&nbsp;<a href=3D"rtsp://192.168.0.1/=
mission_recorder">rtsp://192.168.0.1/mission_recorder</a>&nbsp;RTSP/1.0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 1<br>
</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Transport: unicast;client_port=3D4588-4589</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 1</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Date: 5 Aug 2013 11:15:07 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session: 1234567<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Transport: unicast;client_port=3D4=
588-4589;server_port=3D6256-6257</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: GET_PARAMETER&nbsp;<a href=3D"rtsp://192.=
168.0.1/mission_recorder">rtsp://192.168.0.1/mission_recorder</a>&nbsp;RTSP=
/1.0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 7<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session:&nbsp;1234567<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Length: 15</div>
<div><br>
</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;list_recorded_data</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 7</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Date: 5 Aug 2013 11:15:10 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Length: 46<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;list_recorded_data:20130804T153000=
Z,20130804T213000Z</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: TEARDOWN&nbsp;<a href=3D"rtsp://192.168.0=
.1/mission_recorder">rtsp://192.168.0.1/mission_recorder</a>&nbsp;RTSP/1.0<=
br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 8<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;Session: 1234567<br>
<br>
&nbsp; &nbsp; &nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 8<br>
<br>
</div>
<div><br>
</div>
<div>Then we'd specify the proper way to interpret the response from the se=
rver and open up a new RTSP session to playback the recorded data. For exam=
ple:</div>
<div><br>
</div>
<div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: SETUP&nbsp;<a href=3D"rtsp://192.168.0.1/=
mission_recorder/">rtsp://192.168.0.1/mission_recorder/</a>20130804T153000Z=
&nbsp;RTSP/1.0</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 1<br>
</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Transport: multicast;client_port=3D4588-4589</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 1</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Date: 6 Aug 2013 11:15:07 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session: 7654321<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Transport: multicast;client_port=
=3D4588-4589;server_port=3D6256-6257</div>
</div>
<div><br>
</div>
<div>&nbsp; &nbsp; &nbsp;C-&gt;S: PLAY&nbsp;<a href=3D"rtsp://192.168.0.1/m=
ission_recorder/20130804T153000Z">rtsp://192.168.0.1/mission_recorder/20130=
804T153000Z</a>&nbsp;RTSP/1.0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 2<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: text/parameters<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Session: 7654321<br>
<br>
</div>
<div>&nbsp; &nbsp; &nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 2<br>
</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;Date: 6 Aug 2013 11:15:10 GMT</div>
<div><br>
</div>
<div>Okay, last question, would the DESCRIBE method for&nbsp;<a href=3D"rts=
p://192.168.0.1/mission_recorder/20130804T153000Z">rtsp://192.168.0.1/missi=
on_recorder/20130804T153000Z</a>&nbsp;be expected to describe the &quot;str=
eams&quot; that were recorded? In other words could the
 following be expected?</div>
<div><br>
</div>
<div>
<div>C-&gt;S: DESCRIBE&nbsp;<a href=3D"rtsp://192.168.0.1/mission_recorder/=
">rtsp://192.168.0.1/mission_recorder/</a>20130804T153000Z&nbsp;RTSP/1.0</d=
iv>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 312</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Accept: application/sdp, appl=
ication/rtsl, application/mheg</div>
</div>
<div><br>
</div>
<div>
<div>S-&gt;C:&nbsp;<span style=3D"white-space: pre-wrap; ">RTSP/1.0 200 OK<=
/span></div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;CSeq: 312<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Date: 7 Aug 2013 15:35:07 GMT<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Type: application/sdp<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Content-Length: 332<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;v=3D0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;o=3D- 2890844527 2890845471 IN IP4=
 192.168.0.123<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;s=3DPosition Stream 1<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;i=3DThreat stream from threat serv=
ice on the Boomerang<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;c=3DIN IP4 239.192.0.124/127<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;t=3D0 0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a=3Dunique_id:pos1<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;a=3Ddata_type:position<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;m=3Dmessage 12346</div>
</div>
<div><span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nb=
sp; &nbsp;s=3DPosition Stream 1</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;i=3DPosition stream from posi=
tion service<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;c=3DIN IP4 239.192.0.124/127<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;t=3D0 0<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;a=3Dunique_id:pos1<br>
<span class=3D"Apple-tab-span" style=3D"white-space: pre; "></span>&nbsp; &=
nbsp;a=3Ddata_type:position<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;m=3Dmessage 12346</div>
<div><br>
</div>
<div>I really have no idea how multiple streams would be described!</div>
<div><br>
</div>
<div>Thanks in advance for any help!</div>
</div>
<div apple-content-edited=3D"true">
<div>--</div>
<div>Regards,</div>
<div><br>
</div>
<div>Kase Saylor</div>
<div>Southwest Research Institute</div>
</div>
<div apple-content-edited=3D"true"></div>
<br>
</body>
</html>

--_000_862F245D079548E18D5AFEA6A0792D44swriorg_--

From adam@nostrum.com  Thu Aug  8 11:54:00 2013
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 13:21, Hadriel Kaplan wrote:
> On Aug 8, 2013, at 1:39 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/8/13 07:21, Hadriel Kaplan wrote:
>>> What I think should happen is the UAS should send an SDP answer using port=0 for those same m-lines
>> Why? You're proposing that we remove the mechanism for rejecting media streams.
> Huh?  You either misunderstood my email, or I'm not following your statement above.  I don't believe I'm proposing we remove the mechanism of rejecting media streams.  How/where/when did I propose that?  You lost me.

You're proposing that "the UAS should send an SDP answer using port=0 
for [bundle-only] m-lines", even if it wants to accept those m-lines. In 
other words:

  * If the answerer wants to accept the m-lines, it does so by setting
    the port to 0
  * If the answerer wants to reject the m-lines, it does so by setting
    the port to 0


I'm having a hard time understanding how the recipient of the answer 
distinguishes between port=0 and port=0.

/a


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    <div class="moz-cite-prefix">On 8/8/13 13:21, Hadriel Kaplan wrote:<br>
    </div>
    <blockquote
      cite="mid:E3966F35-5B9A-40F3-BFEF-FE97B6B09B5A@oracle.com"
      type="cite">
      <pre wrap="">
On Aug 8, 2013, at 1:39 PM, Adam Roach <a class="moz-txt-link-rfc2396E" href="mailto:adam@nostrum.com">&lt;adam@nostrum.com&gt;</a> wrote:

</pre>
      <blockquote type="cite">
        <pre wrap="">On 8/8/13 07:21, Hadriel Kaplan wrote:
</pre>
        <blockquote type="cite">
          <pre wrap="">What I think should happen is the UAS should send an SDP answer using port=0 for those same m-lines
</pre>
        </blockquote>
        <pre wrap="">
Why? You're proposing that we remove the mechanism for rejecting media streams.
</pre>
      </blockquote>
      <pre wrap="">
Huh?  You either misunderstood my email, or I'm not following your statement above.  I don't believe I'm proposing we remove the mechanism of rejecting media streams.  How/where/when did I propose that?  You lost me.</pre>
    </blockquote>
    <br>
    You're proposing that "the UAS should send an SDP answer using
    port=0 for [bundle-only] m-lines", even if it wants to accept those
    m-lines. In other words:<br>
    <br>
    <ul>
      <li>If the answerer wants to accept the m-lines, it does so by
        setting the port to 0</li>
      <li>If the answerer wants to reject the m-lines, it does so by
        setting the port to 0</li>
    </ul>
    <p><br>
      I'm having a hard time understanding how the recipient of the
      answer distinguishes between port=0 and port=0.<br>
    </p>
    <p>/a<br>
    </p>
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Hadriel Kaplan'" <hadriel.kaplan@oracle.com>
References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se>	<9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>	<7594FB04B1934943A5C02806D1A2204B1C422263@ESESSMB209.ericsson.se> <E4D2C482-C7EF-4BCB-BC4B-BA218E9BFCE8@oracle.com>, <001601ce945c$ef7509d0$ce5f1d70$@co.in> <7594FB04B1934943A5C02806D1A2204B1C4224FF@ESESSMB209.ericsson.se>
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Cc: 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi Christer,

 

Thanks for the clarification that it is a new requirement for BUNDLE.

 

In case it should be allowed to send the BUNDLE in the first offer, the
mechanism for understanding answerer support for BUNDLE has to be
standardized as well like in Sec 9 of RFC 3264 (Indicating capabilities).

 

Thanks

Partha 

 

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] 
Sent: Thursday, August 08, 2013 11:49 PM
To: Parthasarathi R; 'Hadriel Kaplan'
Cc: 'mmusic'
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

 

Hi,

 

Some people suggested that it should be allowed to send the same ports in
the first offer, IF it is known that the answerer supports BUNDLE.

 

However, that is currently not described in the draft, and there were also
objections to such approach.

 

Regards,

 

Christer

 



Sent from Windows using TouchDown (www.nitrodesk.com)


-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: &apos;Hadriel Kaplan&apos; [hadriel.kaplan@oracle.com]; Christer
Holmberg [christer.holmberg@ericsson.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

Hi Hadriel/Christer,

My understanding is that the initial offer with same port breaks more than
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
indicates one of the possible interop issue with existing mobile networks.
So, I don't agree with Hadriel claim of 99% system works with same port. The
current BUNDLE draft mechanism interop in the better way.

In case it is known to offerer that answerer supports BUNDLE before sending
offer, it is fine to send the same port in the offer using bundle-only
attribute. I'm not seeing any issue there.

Thanks
Partha





> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Thursday, August 08, 2013 7:46 PM
> To: Christer Holmberg
> Cc: mmusic
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
> 
> 
> On Aug 8, 2013, at 9:04 AM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
> 
> > Another advantage of using different port numbers is that it provides
> a fallback in case the remote endpoint does not support, or does not
> want to use BUNDLE.
> 
> Yes, except as far as I can tell all devices (and even intermediaries)
> handle an offer with the same port number just fine.  But that's just a
> personal observation, and dangerous to assume ALL of them handle it
> fine; that's why I didn't argue for long that we should use the same
> port number in the offer - because it concerned me that someone knew of
> at least one implementation that didn't handle it well. (not that we
> can truly accommodate ALL implementations anyway, but it's got to be a
> heck of a lot higher than simple majority, and as close to 100% as we
> can get)
> 
> 
> > However, for bundle-only m- lines such fallback is not needed - they
> are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> > ...and, based on the agreed scope, the Offerer KNOWS that the remote
> endpoint supports BUNDLE, so the "one implementation that doesn't
> handle it" will not be there :)
> 
> For WebRTC sure - but we were talking about using BUNDLE generically.
> Or at least that's what I thought you were asking about.  If we're only
> talking about using BUNDLE for WebRTC, then not only do you not need to
> use a port=0, but you also don't need a second offer/answer, etc.
> 
> 
> >> Also, from a purely protocol perspective, it seems quite logical to
> me to use a port number of 0 for an m-line you literally don't want to
> use if the far-end doesn't support
> >> BUNDLE - a port number of 0 effectively means "disabled" today, and
> that's basically what you want.
> >
> > Well, one could also say that it means "Here are some m- lines  I
> want to bundle, with this port, but only if you (the remote endpoint)
> wants to use bundle." :)
> 
> We can certainly change the semantics of SDP offers with port=0 for
> WebRTC, but I think it would be really dangerous to change its
> semantics for general use.
> 
> 
> >> What I think should happen is the UAS should send an SDP answer
> using port=0 for those same m-lines, and the UAC should then send a new
> offer with those m-lines having the real port number of the bundle
> transport, and the far-end should answer that in the same fashion.
> >
> > The problem is that the grouping framework does not allow, in SDP
> Answers, usage of port=0 in grouped m- lines. It would require an
> update of the RFC, and I think we want to avoid that.
> 
> Considering the number of docs the Unified-Plan proposes to either
> create, change, update or whatever... methinks having BUNDLE update the
> grouping framework RFC is rather a minor nit. :)
> 
> -hadriel
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Christer,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks for the clarification that it is a new requirement =
for
BUNDLE.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In case it should be allowed to send the BUNDLE in the =
first
offer, the mechanism for understanding answerer support for BUNDLE has =
to be
standardized as well like in Sec 9 of RFC 3264 (Indicating =
capabilities).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Christer =
Holmberg
[mailto:christer.holmberg@ericsson.com] <br>
<b>Sent:</b> Thursday, August 08, 2013 11:49 PM<br>
<b>To:</b> Parthasarathi R; 'Hadriel Kaplan'<br>
<b>Cc:</b> 'mmusic'<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Hi,<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Some people suggested that it should be allowed to send the =
same
ports in the first offer, IF it is known that the answerer supports =
BUNDLE.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>However, that is currently not described in the draft, and =
there
were also objections to such approach.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Regards,<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Christer<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

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<div>

<p class=3DMsoNormal><span =
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color:black'><br>
<br>
Sent from <b><i>Windows</i></b> using </span><b><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:blue'>TouchDown</span></b><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>
(www.nitrodesk.com)<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'><br>
-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> &amp;apos;Hadriel Kaplan&amp;apos; =
[hadriel.kaplan@oracle.com];
Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt'>Hi
Hadriel/Christer,<br>
<br>
My understanding is that the initial offer with same port breaks more =
than<br>
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft<br>
indicates one of the possible interop issue with existing mobile =
networks.<br>
So, I don't agree with Hadriel claim of 99% system works with same port. =
The<br>
current BUNDLE draft mechanism interop in the better way.<br>
<br>
In case it is known to offerer that answerer supports BUNDLE before =
sending<br>
offer, it is fine to send the same port in the offer using =
bundle-only<br>
attribute. I'm not seeing any issue there.<br>
<br>
Thanks<br>
Partha<br>
<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic-bounces@ietf.org [<a =
href=3D"mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a=
>]
On<br>
&gt; Behalf Of Hadriel Kaplan<br>
&gt; Sent: Thursday, August 08, 2013 7:46 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: mmusic<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-<br>
&gt; only m- lines?<br>
&gt; <br>
&gt; <br>
&gt; On Aug 8, 2013, at 9:04 AM, Christer Holmberg<br>
&gt; &lt;christer.holmberg@ericsson.com&gt; wrote:<br>
&gt; <br>
&gt; &gt; Another advantage of using different port numbers is that it =
provides<br>
&gt; a fallback in case the remote endpoint does not support, or does =
not<br>
&gt; want to use BUNDLE.<br>
&gt; <br>
&gt; Yes, except as far as I can tell all devices (and even =
intermediaries)<br>
&gt; handle an offer with the same port number just fine.&nbsp; But =
that's just
a<br>
&gt; personal observation, and dangerous to assume ALL of them handle =
it<br>
&gt; fine; that's why I didn't argue for long that we should use the =
same<br>
&gt; port number in the offer - because it concerned me that someone =
knew of<br>
&gt; at least one implementation that didn't handle it well. (not that =
we<br>
&gt; can truly accommodate ALL implementations anyway, but it's got to =
be a<br>
&gt; heck of a lot higher than simple majority, and as close to 100% as =
we<br>
&gt; can get)<br>
&gt; <br>
&gt; <br>
&gt; &gt; However, for bundle-only m- lines such fallback is not needed =
- they<br>
&gt; are ONLY going to be used IF the remote endpoint chooses to use =
BUNDLE.<br>
&gt; &gt; ...and, based on the agreed scope, the Offerer KNOWS that the =
remote<br>
&gt; endpoint supports BUNDLE, so the &quot;one implementation that =
doesn't<br>
&gt; handle it&quot; will not be there :)<br>
&gt; <br>
&gt; For WebRTC sure - but we were talking about using BUNDLE =
generically.<br>
&gt; Or at least that's what I thought you were asking about.&nbsp; If =
we're
only<br>
&gt; talking about using BUNDLE for WebRTC, then not only do you not =
need to<br>
&gt; use a port=3D0, but you also don't need a second offer/answer, =
etc.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; Also, from a purely protocol perspective, it seems quite =
logical
to<br>
&gt; me to use a port number of 0 for an m-line you literally don't want =
to<br>
&gt; use if the far-end doesn't support<br>
&gt; &gt;&gt; BUNDLE - a port number of 0 effectively means
&quot;disabled&quot; today, and<br>
&gt; that's basically what you want.<br>
&gt; &gt;<br>
&gt; &gt; Well, one could also say that it means &quot;Here are some m-
lines&nbsp; I<br>
&gt; want to bundle, with this port, but only if you (the remote =
endpoint)<br>
&gt; wants to use bundle.&quot; :)<br>
&gt; <br>
&gt; We can certainly change the semantics of SDP offers with port=3D0 =
for<br>
&gt; WebRTC, but I think it would be really dangerous to change its<br>
&gt; semantics for general use.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; What I think should happen is the UAS should send an SDP =
answer<br>
&gt; using port=3D0 for those same m-lines, and the UAC should then send =
a new<br>
&gt; offer with those m-lines having the real port number of the =
bundle<br>
&gt; transport, and the far-end should answer that in the same =
fashion.<br>
&gt; &gt;<br>
&gt; &gt; The problem is that the grouping framework does not allow, in =
SDP<br>
&gt; Answers, usage of port=3D0 in grouped m- lines. It would require =
an<br>
&gt; update of the RFC, and I think we want to avoid that.<br>
&gt; <br>
&gt; Considering the number of docs the Unified-Plan proposes to =
either<br>
&gt; create, change, update or whatever... methinks having BUNDLE update =
the<br>
&gt; grouping framework RFC is rather a minor nit. :)<br>
&gt; <br>
&gt; -hadriel<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.or=
g/mailman/listinfo/mmusic</a><o:p></o:p></span></p>

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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 2:24 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> I agree with Adam on this one. Port zero is used in answers to reject =
a stream, and I don't think we should change that.=20

Ahhh - now I think I understand what Adam's angst was. (maybe)  You're =
thinking that if the UAS' answer keeps the port=3D0, then that =
contradicts using port=3D0 meaning it was rejected?

Well, hate to tell y'all, but a port number of zero means that in the =
offer TOO.

=46rom RFC 3264:
   A port number of zero in the offer indicates that the
   stream is offered but MUST NOT be used.

What I'm proposing is we don't break/change that whatsoever.  We have =
the UAC send port=3D0 and "a=3Dbundle-only" in the Offer, and UAS send =
an SDP answer with port=3D0 too, and ALSO "a=3Dbundle-only" for the =
m-lines it is ok with using.  Or "a=3Dbundle-only:use" or whatever.  The =
UAC sees that in the answer, and can send a new Offer with the bundle =
port number, likewise the UAS in its answer.

-hadriel


From pkyzivat@alum.mit.edu  Thu Aug  8 12:00:50 2013
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] RTSP for data recorder design
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Kase,

Its hard to discern your exact requirements. But you should at least 
look at the work in progress by the siprec workgroup. You can find the 
relevant drafts at:

http://datatracker.ietf.org/wg/siprec/

This may not be what you need, but is worth a look.

	Thanks,
	Paul

On 8/8/13 8:49 PM, Saylor, Kase J. wrote:
> To Whom it may concern,
>
> I am helping the US Army develop standards for a network recorder on US
> Army vehicles. We like to use available standards when at all possible,
> and my research seems to indicate that RTSP would be a great candidate
> standard. I've been poring over the RTSP specification (along with the
> SDP standard) and I have your douce code for LiveMedia. However, I still
> have a few questions concerning the RTSP standard and its use for our
> particular need. It seems like there is plenty of examples on the
> Internet for the PLAY method, but not much for RECORD. I really hope
> that someone may be able to provide some insight that will get me moving
> in the right direction. On to the questions!
>
> Our recorder requires the ability to "nominate" data streams for
> recording, so I believe the correct RTSP method is ANNOUNCE. My first
> question is when I was to add a stream using ANNOUNCE, if I use the same
> Session number for multiple streams, does this basically create a
> presentation? For example:
>
> C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
>             CSeq: 3
>             Date: 7 Aug 2013 15:35:06 GMT
>             Session: 47112344
>             Content-Type: application/sdp
>             Content-Length: 332
>
>             v=0
>             o=- 2890844526 2890845468 IN IP4 192.168.0.123
>             s=Threat Stream 1
>             i=Threat stream from threat service on the Boomerang
>             c=IN IP4 239.192.0.123/127
>             t=0 0
>             a=unique_id:threat1
>     a=data_type:threat
>             m=message 12345
>
> (some response back from server)
>
> C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
>             CSeq: 4
>             Date: 7 Aug 2013 15:35:07 GMT
>             Session: 47112344
>             Content-Type: application/sdp
>             Content-Length: 332
>
>             v=0
>             o=- 2890844527 2890845471 IN IP4 192.168.0.123
>             s=Position Stream 1
>             i=Position stream from position service
>             c=IN IP4 239.192.0.124/127
>             t=0 0
>             a=unique_id:pos1
>     a=data_type:position
>             m=message 12346
> (some response back from server)
>
> Based upon these two announce messages, is it safe to assume that the
> following command would cause the recorder to start recording those two
> data streams (assuming the current date/time is 8/4/2013 1530 GMT)?
>
>       C->S: RECORD
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 5
>             Content-Type: text/parameters
>             Session: 47112344
>     Range: clock=20130804T153000Z-
>
> Furthermore, would the following command stop the recorder (assuming the
> current date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in the
> RFC for stopping a recording.
>
>       C->S: RECORD
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 6
>             Content-Type: text/parameters
>             Session: 47112344
>     Range: clock=- 20130804T153200Z
>
>
> I apologize but I have at least one or two more questions ;-) After
> tearing down a session, how would one go back and playback recorded
> data? From my RECORD example I've assumed that providing a URI with a
> "path" based upon record start time, could be one way to mark recording
> events. Do you think this is an acceptable approach? Then the play
> command would something like:
>
>       C->S: PLAY
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 6
>             Content-Type: text/parameters
>             Session: 1234567
>
> I suppose I could specify unique parameters and use the GET_PARAMETER
> method to list the available recordings. Perhaps something like:
>
>       C->S: SETUP rtsp://192.168.0.1/mission_recorder RTSP/1.0
>             CSeq: 1
>     Transport: unicast;client_port=4588-4589
>
>       S->C: RTSP/1.0 200 OK
>             CSeq: 1
>     Date: 5 Aug 2013 11:15:07 GMT
>             Session: 1234567
>             Transport: unicast;client_port=4588-4589;server_port=6256-6257
>
>       C->S: GET_PARAMETER rtsp://192.168.0.1/mission_recorder RTSP/1.0
>             CSeq: 7
>             Content-Type: text/parameters
>             Session: 1234567
>             Content-Length: 15
>
>     list_recorded_data
>
>
>       S->C: RTSP/1.0 200 OK
>             CSeq: 7
>     Date: 5 Aug 2013 11:15:10 GMT
>             Content-Length: 46
>             Content-Type: text/parameters
>
>             list_recorded_data:20130804T153000Z,20130804T213000Z
>
>       C->S: TEARDOWN rtsp://192.168.0.1/mission_recorder RTSP/1.0
>             CSeq: 8
>     Session: 1234567
>
>       S->C: RTSP/1.0 200 OK
>             CSeq: 8
>
>
> Then we'd specify the proper way to interpret the response from the
> server and open up a new RTSP session to playback the recorded data. For
> example:
>
>       C->S: SETUP
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 1
>     Transport: multicast;client_port=4588-4589
>
>       S->C: RTSP/1.0 200 OK
>             CSeq: 1
>     Date: 6 Aug 2013 11:15:07 GMT
>             Session: 7654321
>             Transport: multicast;client_port=4588-4589;server_port=6256-6257
>
>       C->S: PLAY
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 2
>             Content-Type: text/parameters
>             Session: 7654321
>
>       S->C: RTSP/1.0 200 OK
>             CSeq: 2
>     Date: 6 Aug 2013 11:15:10 GMT
>
> Okay, last question, would the DESCRIBE method for
> rtsp://192.168.0.1/mission_recorder/20130804T153000Z be expected to
> describe the "streams" that were recorded? In other words could the
> following be expected?
>
> C->S: DESCRIBE rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>             CSeq: 312
>             Accept: application/sdp, application/rtsl, application/mheg
>
> S->C: RTSP/1.0 200 OK
>             CSeq: 312
>             Date: 7 Aug 2013 15:35:07 GMT
>             Content-Type: application/sdp
>             Content-Length: 332
>
>             v=0
>             o=- 2890844527 2890845471 IN IP4 192.168.0.123
>             s=Position Stream 1
>             i=Threat stream from threat service on the Boomerang
>             c=IN IP4 239.192.0.124/127
>             t=0 0
>             a=unique_id:pos1
>     a=data_type:position
>             m=message 12346
>     s=Position Stream 1
>             i=Position stream from position service
>             c=IN IP4 239.192.0.124/127
>             t=0 0
>             a=unique_id:pos1
>     a=data_type:position
>             m=message 12346
>
> I really have no idea how multiple streams would be described!
>
> Thanks in advance for any help!
> --
> Regards,
>
> Kase Saylor
> Southwest Research Institute
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From hadriel.kaplan@oracle.com  Thu Aug  8 12:01:12 2013
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 2:53 PM, Adam Roach <adam@nostrum.com> wrote:

> You're proposing that "the UAS should send an SDP answer using port=3D0 =
for [bundle-only] m-lines", even if it wants to accept those m-lines. In =
other words:
>=20
> 	=95 If the answerer wants to accept the m-lines, it does so by =
setting the port to 0
> 	=95 If the answerer wants to reject the m-lines, it does so by =
setting the port to 0
>=20
> I'm having a hard time understanding how the recipient of the answer =
distinguishes between port=3D0 and port=3D0.
> /a

Sorry, yeah I figured out what you meant a few minutes later after =
reading Christer's email.  I just responded to that in my reply to him.

-hadriel
p.s. isn't email a fun medium for real-time chat?  :)


From christer.holmberg@ericsson.com  Thu Aug  8 12:01:21 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, 'Hadriel Kaplan' <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
Thread-Index: AQHOlFzzTHFUEWnKxkeE8bnrSX+BlpmLntotgAAIhlCAAAMirQ==
Date: Thu, 8 Aug 2013 19:00:59 +0000
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Cc: 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi,

I don't think the answerer needs a feature tag to indicate BUNDLE support. =
If it doesn't support BUNDLE, it won't include the BUNDLE group in the answ=
er, nor will it assign a single port value to multiple m- lines.

Or, did I misunderstand your issue?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: Christer Holmberg [christer.holmberg@ericsson.com]; &apos;Hadriel Kapla=
n&apos; [hadriel.kaplan@oracle.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
Hi Christer,

Thanks for the clarification that it is a new requirement for BUNDLE.

In case it should be allowed to send the BUNDLE in the first offer, the mec=
hanism for understanding answerer support for BUNDLE has to be standardized=
 as well like in Sec 9 of RFC 3264 (Indicating capabilities).

Thanks
Partha

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Thursday, August 08, 2013 11:49 PM
To: Parthasarathi R; 'Hadriel Kaplan'
Cc: 'mmusic'
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

Hi,

Some people suggested that it should be allowed to send the same ports in t=
he first offer, IF it is known that the answerer supports BUNDLE.

However, that is currently not described in the draft, and there were also =
objections to such approach.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: &apos;Hadriel Kaplan&apos; [hadriel.kaplan@oracle.com]; Christer Holmbe=
rg [christer.holmberg@ericsson.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
Hi Hadriel/Christer,

My understanding is that the initial offer with same port breaks more than
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
indicates one of the possible interop issue with existing mobile networks.
So, I don't agree with Hadriel claim of 99% system works with same port. Th=
e
current BUNDLE draft mechanism interop in the better way.

In case it is known to offerer that answerer supports BUNDLE before sending
offer, it is fine to send the same port in the offer using bundle-only
attribute. I'm not seeing any issue there.

Thanks
Partha





> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Thursday, August 08, 2013 7:46 PM
> To: Christer Holmberg
> Cc: mmusic
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
>
>
> On Aug 8, 2013, at 9:04 AM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
>
> > Another advantage of using different port numbers is that it provides
> a fallback in case the remote endpoint does not support, or does not
> want to use BUNDLE.
>
> Yes, except as far as I can tell all devices (and even intermediaries)
> handle an offer with the same port number just fine.  But that's just a
> personal observation, and dangerous to assume ALL of them handle it
> fine; that's why I didn't argue for long that we should use the same
> port number in the offer - because it concerned me that someone knew of
> at least one implementation that didn't handle it well. (not that we
> can truly accommodate ALL implementations anyway, but it's got to be a
> heck of a lot higher than simple majority, and as close to 100% as we
> can get)
>
>
> > However, for bundle-only m- lines such fallback is not needed - they
> are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> > ...and, based on the agreed scope, the Offerer KNOWS that the remote
> endpoint supports BUNDLE, so the "one implementation that doesn't
> handle it" will not be there :)
>
> For WebRTC sure - but we were talking about using BUNDLE generically.
> Or at least that's what I thought you were asking about.  If we're only
> talking about using BUNDLE for WebRTC, then not only do you not need to
> use a port=3D0, but you also don't need a second offer/answer, etc.
>
>
> >> Also, from a purely protocol perspective, it seems quite logical to
> me to use a port number of 0 for an m-line you literally don't want to
> use if the far-end doesn't support
> >> BUNDLE - a port number of 0 effectively means "disabled" today, and
> that's basically what you want.
> >
> > Well, one could also say that it means "Here are some m- lines  I
> want to bundle, with this port, but only if you (the remote endpoint)
> wants to use bundle." :)
>
> We can certainly change the semantics of SDP offers with port=3D0 for
> WebRTC, but I think it would be really dangerous to change its
> semantics for general use.
>
>
> >> What I think should happen is the UAS should send an SDP answer
> using port=3D0 for those same m-lines, and the UAC should then send a new
> offer with those m-lines having the real port number of the bundle
> transport, and the far-end should answer that in the same fashion.
> >
> > The problem is that the grouping framework does not allow, in SDP
> Answers, usage of port=3D0 in grouped m- lines. It would require an
> update of the RFC, and I think we want to avoid that.
>
> Considering the number of docs the Unified-Plan proposes to either
> create, change, update or whatever... methinks having BUNDLE update the
> grouping framework RFC is rather a minor nit. :)
>
> -hadriel
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I don't think the answerer needs a feature tag =
to indicate BUNDLE support. If it doesn't support BUNDLE, it won't include =
the BUNDLE group in the answer, nor will it assign a single port value to m=
ultiple m- lines.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Or, did I misunderstand your issue?</font></div=
>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font>&nbsp;</div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]; &amp;apos;Ha=
driel Kaplan&amp;apos; [hadriel.kaplan@oracle.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span>
<div>
<div class=3D"Section1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Hi Christer,</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Thanks for the clarific=
ation that it is a new requirement for BUNDLE.</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">In case it should be al=
lowed to send the BUNDLE in the first offer, the mechanism for understandin=
g answerer support for BUNDLE has to be standardized as
 well like in Sec 9 of RFC 3264 (Indicating capabilities).</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Thanks</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">Partha
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:#1F497D">&nbsp;</span></p>
<div style=3D"border:none; border-left:solid blue 1.5pt; padding:0in 0in 0i=
n 4.0pt">
<div>
<div style=3D"border:none; border-top:solid #B5C4DF 1.0pt; padding:3.0pt 0i=
n 0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt; font-family:&quo=
t;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-=
size:10.0pt; font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Christ=
er Holmberg [mailto:christer.holmberg@ericsson.com]
<br>
<b>Sent:</b> Thursday, August 08, 2013 11:49 PM<br>
<b>To:</b> Parthasarathi R; 'Hadriel Kaplan'<br>
<b>Cc:</b> 'mmusic'<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;</p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">Hi,</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">Some people suggested tha=
t it should be allowed to send the same ports in the first offer, IF it is =
known that the answerer supports BUNDLE.</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">However, that is currentl=
y not described in the draft, and there were also objections to such approa=
ch.</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">Regards,</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">Christer</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black">&nbsp;</span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black"><br>
<br>
Sent from <b><i>Windows</i></b> using </span><b><span style=3D"font-size:11=
.0pt; font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;; color:blue">T=
ouchDown</span></b><span style=3D"font-size:11.0pt; font-family:&quot;Calib=
ri&quot;,&quot;sans-serif&quot;; color:black"> (www.nitrodesk.com)</span></=
p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt; font-family:&quot;C=
alibri&quot;,&quot;sans-serif&quot;; color:black"><br>
-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> &amp;apos;Hadriel Kaplan&amp;apos; [hadriel.kaplan@oracle.com]; =
Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt">Hi Hadriel/Christer,<br>
<br>
My understanding is that the initial offer with same port breaks more than<=
br>
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft<br>
indicates one of the possible interop issue with existing mobile networks.<=
br>
So, I don't agree with Hadriel claim of 99% system works with same port. Th=
e<br>
current BUNDLE draft mechanism interop in the better way.<br>
<br>
In case it is known to offerer that answerer supports BUNDLE before sending=
<br>
offer, it is fine to send the same port in the offer using bundle-only<br>
attribute. I'm not seeing any issue there.<br>
<br>
Thanks<br>
Partha<br>
<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounces@ietf.o=
rg">mailto:mmusic-bounces@ietf.org</a>] On<br>
&gt; Behalf Of Hadriel Kaplan<br>
&gt; Sent: Thursday, August 08, 2013 7:46 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: mmusic<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-=
<br>
&gt; only m- lines?<br>
&gt; <br>
&gt; <br>
&gt; On Aug 8, 2013, at 9:04 AM, Christer Holmberg<br>
&gt; &lt;christer.holmberg@ericsson.com&gt; wrote:<br>
&gt; <br>
&gt; &gt; Another advantage of using different port numbers is that it prov=
ides<br>
&gt; a fallback in case the remote endpoint does not support, or does not<b=
r>
&gt; want to use BUNDLE.<br>
&gt; <br>
&gt; Yes, except as far as I can tell all devices (and even intermediaries)=
<br>
&gt; handle an offer with the same port number just fine.&nbsp; But that's =
just a<br>
&gt; personal observation, and dangerous to assume ALL of them handle it<br=
>
&gt; fine; that's why I didn't argue for long that we should use the same<b=
r>
&gt; port number in the offer - because it concerned me that someone knew o=
f<br>
&gt; at least one implementation that didn't handle it well. (not that we<b=
r>
&gt; can truly accommodate ALL implementations anyway, but it's got to be a=
<br>
&gt; heck of a lot higher than simple majority, and as close to 100% as we<=
br>
&gt; can get)<br>
&gt; <br>
&gt; <br>
&gt; &gt; However, for bundle-only m- lines such fallback is not needed - t=
hey<br>
&gt; are ONLY going to be used IF the remote endpoint chooses to use BUNDLE=
.<br>
&gt; &gt; ...and, based on the agreed scope, the Offerer KNOWS that the rem=
ote<br>
&gt; endpoint supports BUNDLE, so the &quot;one implementation that doesn't=
<br>
&gt; handle it&quot; will not be there :)<br>
&gt; <br>
&gt; For WebRTC sure - but we were talking about using BUNDLE generically.<=
br>
&gt; Or at least that's what I thought you were asking about.&nbsp; If we'r=
e only<br>
&gt; talking about using BUNDLE for WebRTC, then not only do you not need t=
o<br>
&gt; use a port=3D0, but you also don't need a second offer/answer, etc.<br=
>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; Also, from a purely protocol perspective, it seems quite logi=
cal to<br>
&gt; me to use a port number of 0 for an m-line you literally don't want to=
<br>
&gt; use if the far-end doesn't support<br>
&gt; &gt;&gt; BUNDLE - a port number of 0 effectively means &quot;disabled&=
quot; today, and<br>
&gt; that's basically what you want.<br>
&gt; &gt;<br>
&gt; &gt; Well, one could also say that it means &quot;Here are some m- lin=
es&nbsp; I<br>
&gt; want to bundle, with this port, but only if you (the remote endpoint)<=
br>
&gt; wants to use bundle.&quot; :)<br>
&gt; <br>
&gt; We can certainly change the semantics of SDP offers with port=3D0 for<=
br>
&gt; WebRTC, but I think it would be really dangerous to change its<br>
&gt; semantics for general use.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; What I think should happen is the UAS should send an SDP answ=
er<br>
&gt; using port=3D0 for those same m-lines, and the UAC should then send a =
new<br>
&gt; offer with those m-lines having the real port number of the bundle<br>
&gt; transport, and the far-end should answer that in the same fashion.<br>
&gt; &gt;<br>
&gt; &gt; The problem is that the grouping framework does not allow, in SDP=
<br>
&gt; Answers, usage of port=3D0 in grouped m- lines. It would require an<br=
>
&gt; update of the RFC, and I think we want to avoid that.<br>
&gt; <br>
&gt; Considering the number of docs the Unified-Plan proposes to either<br>
&gt; create, change, update or whatever... methinks having BUNDLE update th=
e<br>
&gt; grouping framework RFC is rather a minor nit. :)<br>
&gt; <br>
&gt; -hadriel<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a></span></p>
</div>
</div>
</div>
</div>
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</html>

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From adam@nostrum.com  Thu Aug  8 12:03:40 2013
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 13:58, Hadriel Kaplan wrote:
> What I'm proposing is we don't break/change that whatsoever. We have 
> the UAC send port=0 and "a=bundle-only" in the Offer, and UAS send an 
> SDP answer with port=0 too, and ALSO "a=bundle-only" for the m-lines 
> it is ok with using. Or "a=bundle-only:use" or whatever.

But why? You're introducing additional mechanism (and hence complexity) 
for no technical advantage.

/a

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

So, you suggest that port zero means reject, and port-zero + a=3Dbundle-onl=
y means something else - in Offers AND Answers?

Even if that might be doable, my main issue is still that since the Answere=
r assign the bundle port to all other bundled m- lines, it should also assi=
gn it to the bundle-only m- lines.

Also, we need to study whether we need to indicate bundle-only in answers t=
o begin with, as the purpose of the indicator only applies to offers...

Regards,

Christer





Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: Adam Roach [adam@nostrum.com]; mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

On Aug 8, 2013, at 2:24 PM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Hi,
>
> I agree with Adam on this one. Port zero is used in answers to reject a s=
tream, and I don't think we should change that.

Ahhh - now I think I understand what Adam's angst was. (maybe)  You're thin=
king that if the UAS' answer keeps the port=3D0, then that contradicts usin=
g port=3D0 meaning it was rejected?

Well, hate to tell y'all, but a port number of zero means that in the offer=
 TOO.

>From RFC 3264:
   A port number of zero in the offer indicates that the
   stream is offered but MUST NOT be used.

What I'm proposing is we don't break/change that whatsoever.  We have the U=
AC send port=3D0 and "a=3Dbundle-only" in the Offer, and UAS send an SDP an=
swer with port=3D0 too, and ALSO "a=3Dbundle-only" for the m-lines it is ok=
 with using.  Or "a=3Dbundle-only:use" or whatever.  The UAC sees that in t=
he answer, and can send a new Offer with the bundle port number, likewise t=
he UAS in its answer.

-hadriel


--_000_7594FB04B1934943A5C02806D1A2204B1C422753ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">So, you suggest that port zero means reject, an=
d port-zero &#43; a=3Dbundle-only means something else - in Offers AND Answ=
ers?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Even if that might be doable, my main issue is =
still that since the Answerer assign the bundle port to all other bundled m=
- lines, it should also assign it to the bundle-only m- lines.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Also, we need to study whether we need to indic=
ate bundle-only in answers to begin with, as the purpose of the indicator o=
nly applies to offers...</font></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div>&nbsp;</div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> Adam Roach [adam@nostrum.com]; mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 8, 2013, at 2:24 PM, Christer Holmberg &lt;christer.holmberg@ericsso=
n.com&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;&nbsp; <br>
&gt; I agree with Adam on this one. Port zero is used in answers to reject =
a stream, and I don't think we should change that.
<br>
<br>
Ahhh - now I think I understand what Adam's angst was. (maybe)&nbsp; You're=
 thinking that if the UAS' answer keeps the port=3D0, then that contradicts=
 using port=3D0 meaning it was rejected?<br>
<br>
Well, hate to tell y'all, but a port number of zero means that in the offer=
 TOO.<br>
<br>
>From RFC 3264:<br>
&nbsp;&nbsp; A port number of zero in the offer indicates that the<br>
&nbsp;&nbsp; stream is offered but MUST NOT be used.<br>
<br>
What I'm proposing is we don't break/change that whatsoever.&nbsp; We have =
the UAC send port=3D0 and &quot;a=3Dbundle-only&quot; in the Offer, and UAS=
 send an SDP answer with port=3D0 too, and ALSO &quot;a=3Dbundle-only&quot;=
 for the m-lines it is ok with using.&nbsp; Or &quot;a=3Dbundle-only:use&qu=
ot; or
 whatever.&nbsp; The UAC sees that in the answer, and can send a new Offer =
with the bundle port number, likewise the UAS in its answer.<br>
<br>
-hadriel<br>
<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422753ESESSMB209erics_--

From hadriel.kaplan@oracle.com  Thu Aug  8 12:11:42 2013
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 3:03 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 13:58, Hadriel Kaplan wrote:
>> What I'm proposing is we don't break/change that whatsoever. We have =
the UAC send port=3D0 and "a=3Dbundle-only" in the Offer, and UAS send =
an SDP answer with port=3D0 too, and ALSO "a=3Dbundle-only" for the =
m-lines it is ok with using. Or "a=3Dbundle-only:use" or whatever.
>=20
> But why? You're introducing additional mechanism (and hence =
complexity) for no technical advantage.

1) To avoid confusing the things that go bump in the night.

2) To avoid updating RFC 3264.

3) Because it's the right thing to do.  If I send a new SDP Offer 10 =
minutes later during the session, and I set an existing m-line to port 0 =
in that Offer, does it mean I want to disable the media, or restart =
bundling, or what?  We're talking about SDP here, and a port=3D0 means =
"disabled", or "unwanted", or whatever word you want to use.  It means =
that in an offer, and it means that in an answer.  It doesn't matter =
when you got the offer, or when you got the answer.  It's stateless.  I =
don't see how that adds complexity - seems simpler to me.

-hadriel


From hadriel.kaplan@oracle.com  Thu Aug  8 12:32:16 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
In-Reply-To: <7594FB04B1934943A5C02806D1A2204B1C422753@ESESSMB209.ericsson.se>
Date: Thu, 8 Aug 2013 15:32:00 -0400
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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General question: Are we talking about this for WebRTC only, or for =
general SDP applications?  I'm worried I'm not in the same context as =
you guys.  I'm talking about this stuff in terms of using it for SIP.  =
If you're only talking about it for WebRTC, please let me know.


On Aug 8, 2013, at 3:09 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> So, you suggest that port zero means reject, and port-zero + =
a=3Dbundle-only means something else - in Offers AND Answers?

I'm suggesting that if we want to use this for SIP, instead of only =
WebRTC, that RFC 3264 be followed:
   A port number of zero in the offer indicates that the
   stream is offered but MUST NOT be used.  This has no useful semantics
   in an initial offer, but is allowed for reasons of completeness,
   since the answer can contain a zero port indicating a rejected stream
   (Section 6).  Furthermore, existing streams can be terminated by
   setting the port to zero (Section 8).  In general, a port number of
   zero indicates that the media stream is not wanted.

BUNDLE would have an attribute it defines of "a=3Dbundle-only" or =
whatever, which has a defined semantic of "I am willing to use this =
media, but only in a BUNDLE RFC multiplex transport thingamajig".  It =
has that meaning in SDP, period... what does it matter if it's in an =
offer or answer?  Why have different meanings for it? (I'm not following =
you on the concern - I must be missing something)


> Even if that might be doable, my main issue is still that since the =
Answerer assign the bundle port to all other bundled m- lines, it should =
also assign it to the bundle-only m- lines.

If we're talking about using this for SIP, I think the safest thing to =
do is have both the initial Offer and initial Answer keep to 3264, and =
have the Answer use port=3D0 for any m-lines it received as port=3D0 in =
an Offer.  Obviously it's going to take another O/A round, and I hate =
that... but that's life. (...for SIP - we can optimize WebRTC if we =
think its necessary)

-hadriel


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Cc: 'mmusic' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi Christer,

 

<snip>

Some people suggested that it should be allowed to send the same ports in
the first offer, IF it is known that the answerer supports BUNDLE.

</snip>

 

My question is how offerer will get to know about answerer capabilities in
the first offer?  In case it should be allowed to send the BUNDLE in the
first offer, the mechanism for understanding answerer's BUNDLE capabilities
has to be standardized as well like in Sec 9 of RFC 3264 (Indicating
capabilities). Let us discuss separately how the answerer will indicate
BUNDLE capabilities (BUNDLE group attributes missing,"bundle-only"
attribute) which is not a concern as of now. Hope this clarifies.

 

Thanks

Partha

 

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] 
Sent: Friday, August 09, 2013 12:31 AM
To: Parthasarathi R; 'Hadriel Kaplan'
Cc: 'mmusic'
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

 

Hi,

 

I don't think the answerer needs a feature tag to indicate BUNDLE support.

 

 If it doesn't support BUNDLE, it won't include the BUNDLE group in the
answer, nor will it assign a single port value to multiple m- lines.

 

Or, did I misunderstand your issue?

 

Regards,

 

Christer 

 



Sent from Windows using TouchDown (www.nitrodesk.com)


-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: Christer Holmberg [christer.holmberg@ericsson.com]; &apos;Hadriel
Kaplan&apos; [hadriel.kaplan@oracle.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines? 

Hi Christer,

 

Thanks for the clarification that it is a new requirement for BUNDLE.

 

In case it should be allowed to send the BUNDLE in the first offer, the
mechanism for understanding answerer support for BUNDLE has to be
standardized as well like in Sec 9 of RFC 3264 (Indicating capabilities).

 

Thanks

Partha 

 

From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] 
Sent: Thursday, August 08, 2013 11:49 PM
To: Parthasarathi R; 'Hadriel Kaplan'
Cc: 'mmusic'
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

 

Hi,

 

Some people suggested that it should be allowed to send the same ports in
the first offer, IF it is known that the answerer supports BUNDLE.

 

However, that is currently not described in the draft, and there were also
objections to such approach.

 

Regards,

 

Christer

 



Sent from Windows using TouchDown (www.nitrodesk.com)


-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: &apos;Hadriel Kaplan&apos; [hadriel.kaplan@oracle.com]; Christer
Holmberg [christer.holmberg@ericsson.com]
CC: &apos;mmusic&apos; [mmusic@ietf.org]
Subject: RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

Hi Hadriel/Christer,

My understanding is that the initial offer with same port breaks more than
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft
indicates one of the possible interop issue with existing mobile networks.
So, I don't agree with Hadriel claim of 99% system works with same port. The
current BUNDLE draft mechanism interop in the better way.

In case it is known to offerer that answerer supports BUNDLE before sending
offer, it is fine to send the same port in the offer using bundle-only
attribute. I'm not seeing any issue there.

Thanks
Partha





> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Thursday, August 08, 2013 7:46 PM
> To: Christer Holmberg
> Cc: mmusic
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
> 
> 
> On Aug 8, 2013, at 9:04 AM, Christer Holmberg
> <christer.holmberg@ericsson.com> wrote:
> 
> > Another advantage of using different port numbers is that it provides
> a fallback in case the remote endpoint does not support, or does not
> want to use BUNDLE.
> 
> Yes, except as far as I can tell all devices (and even intermediaries)
> handle an offer with the same port number just fine.  But that's just a
> personal observation, and dangerous to assume ALL of them handle it
> fine; that's why I didn't argue for long that we should use the same
> port number in the offer - because it concerned me that someone knew of
> at least one implementation that didn't handle it well. (not that we
> can truly accommodate ALL implementations anyway, but it's got to be a
> heck of a lot higher than simple majority, and as close to 100% as we
> can get)
> 
> 
> > However, for bundle-only m- lines such fallback is not needed - they
> are ONLY going to be used IF the remote endpoint chooses to use BUNDLE.
> > ...and, based on the agreed scope, the Offerer KNOWS that the remote
> endpoint supports BUNDLE, so the "one implementation that doesn't
> handle it" will not be there :)
> 
> For WebRTC sure - but we were talking about using BUNDLE generically.
> Or at least that's what I thought you were asking about.  If we're only
> talking about using BUNDLE for WebRTC, then not only do you not need to
> use a port=0, but you also don't need a second offer/answer, etc.
> 
> 
> >> Also, from a purely protocol perspective, it seems quite logical to
> me to use a port number of 0 for an m-line you literally don't want to
> use if the far-end doesn't support
> >> BUNDLE - a port number of 0 effectively means "disabled" today, and
> that's basically what you want.
> >
> > Well, one could also say that it means "Here are some m- lines  I
> want to bundle, with this port, but only if you (the remote endpoint)
> wants to use bundle." :)
> 
> We can certainly change the semantics of SDP offers with port=0 for
> WebRTC, but I think it would be really dangerous to change its
> semantics for general use.
> 
> 
> >> What I think should happen is the UAS should send an SDP answer
> using port=0 for those same m-lines, and the UAC should then send a new
> offer with those m-lines having the real port number of the bundle
> transport, and the far-end should answer that in the same fashion.
> >
> > The problem is that the grouping framework does not allow, in SDP
> Answers, usage of port=0 in grouped m- lines. It would require an
> update of the RFC, and I think we want to avoid that.
> 
> Considering the number of docs the Unified-Plan proposes to either
> create, change, update or whatever... methinks having BUNDLE update the
> grouping framework RFC is rather a minor nit. :)
> 
> -hadriel
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Christer,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&lt;snip&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Some people suggested that it should be allowed to send the =
same
ports in the first offer, IF it is known that the answerer supports =
BUNDLE.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&lt;/snip&gt;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>My question is how offerer will get to know about =
answerer
capabilities in the first offer?&nbsp; </span><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>In case it should be =
allowed
to send the BUNDLE in the first offer, the mechanism for understanding =
answerer&#8217;s
BUNDLE capabilities has to be standardized as well like in Sec 9 of RFC =
3264
(Indicating capabilities). Let us discuss separately how the answerer =
will
indicate BUNDLE capabilities (BUNDLE group attributes =
missing,&#8221;bundle-only&#8221;
attribute) which is not a concern as of now. Hope this =
clarifies.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
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color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Christer =
Holmberg
[mailto:christer.holmberg@ericsson.com] <br>
<b>Sent:</b> Friday, August 09, 2013 12:31 AM<br>
<b>To:</b> Parthasarathi R; 'Hadriel Kaplan'<br>
<b>Cc:</b> 'mmusic'<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Hi,</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>I don't
think the answerer needs a feature tag to indicate BUNDLE support.<span
style=3D'color:#1F497D'><o:p></o:p></span></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>&nbsp;If it
doesn't support BUNDLE, it won't include the BUNDLE group in the answer, =
nor
will it assign a single port value to multiple m- =
lines.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>Or, did I
misunderstand your issue?</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>Regards,</span><o:p></o:p></=
p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>Christer</span>&nbsp;<o:p></=
o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'><br>
<br>
Sent from <b><i>Windows</i></b> using </span><b><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:blue'>TouchDown</span></b><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>
(www.nitrodesk.com)<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><br>
<span style=3D'color:black'>-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]; =
&amp;apos;Hadriel
Kaplan&amp;apos; [hadriel.kaplan@oracle.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?</span> <o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Christer,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks for the clarification that it is a new requirement =
for BUNDLE.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In case it should be allowed to send the BUNDLE in the =
first
offer, the mechanism for understanding answerer support for BUNDLE has =
to be
standardized as well like in Sec 9 of RFC 3264 (Indicating =
capabilities).</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha </span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp;</span><o:p></o:p></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Christer =
Holmberg
[mailto:christer.holmberg@ericsson.com] <br>
<b>Sent:</b> Thursday, August 08, 2013 11:49 PM<br>
<b>To:</b> Parthasarathi R; 'Hadriel Kaplan'<br>
<b>Cc:</b> 'mmusic'<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?</span><o:p></o:p></p>

</div>

</div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Hi,</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Some people suggested that it should be allowed to send the =
same
ports in the first offer, IF it is known that the answerer supports =
BUNDLE.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>However, that is currently not described in the draft, and =
there
were also objections to such approach.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Regards,</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Christer</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'><br>
<br>
Sent from <b><i>Windows</i></b> using </span><b><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:blue'>TouchDown</span></b><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>
(www.nitrodesk.com)</span><o:p></o:p></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'><br>
-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> &amp;apos;Hadriel Kaplan&amp;apos; =
[hadriel.kaplan@oracle.com];
Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> &amp;apos;mmusic&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> RE: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt'>Hi
Hadriel/Christer,<br>
<br>
My understanding is that the initial offer with same port breaks more =
than<br>
one existing system. Sec 4.1.1 of draft-reddy-rtcweb-mobile-03 draft<br>
indicates one of the possible interop issue with existing mobile =
networks.<br>
So, I don't agree with Hadriel claim of 99% system works with same port. =
The<br>
current BUNDLE draft mechanism interop in the better way.<br>
<br>
In case it is known to offerer that answerer supports BUNDLE before =
sending<br>
offer, it is fine to send the same port in the offer using =
bundle-only<br>
attribute. I'm not seeing any issue there.<br>
<br>
Thanks<br>
Partha<br>
<br>
<br>
<br>
<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic-bounces@ietf.org [<a =
href=3D"mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a=
>]
On<br>
&gt; Behalf Of Hadriel Kaplan<br>
&gt; Sent: Thursday, August 08, 2013 7:46 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: mmusic<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-<br>
&gt; only m- lines?<br>
&gt; <br>
&gt; <br>
&gt; On Aug 8, 2013, at 9:04 AM, Christer Holmberg<br>
&gt; &lt;christer.holmberg@ericsson.com&gt; wrote:<br>
&gt; <br>
&gt; &gt; Another advantage of using different port numbers is that it =
provides<br>
&gt; a fallback in case the remote endpoint does not support, or does =
not<br>
&gt; want to use BUNDLE.<br>
&gt; <br>
&gt; Yes, except as far as I can tell all devices (and even =
intermediaries)<br>
&gt; handle an offer with the same port number just fine.&nbsp; But =
that's just
a<br>
&gt; personal observation, and dangerous to assume ALL of them handle =
it<br>
&gt; fine; that's why I didn't argue for long that we should use the =
same<br>
&gt; port number in the offer - because it concerned me that someone =
knew of<br>
&gt; at least one implementation that didn't handle it well. (not that =
we<br>
&gt; can truly accommodate ALL implementations anyway, but it's got to =
be a<br>
&gt; heck of a lot higher than simple majority, and as close to 100% as =
we<br>
&gt; can get)<br>
&gt; <br>
&gt; <br>
&gt; &gt; However, for bundle-only m- lines such fallback is not needed =
- they<br>
&gt; are ONLY going to be used IF the remote endpoint chooses to use =
BUNDLE.<br>
&gt; &gt; ...and, based on the agreed scope, the Offerer KNOWS that the =
remote<br>
&gt; endpoint supports BUNDLE, so the &quot;one implementation that =
doesn't<br>
&gt; handle it&quot; will not be there :)<br>
&gt; <br>
&gt; For WebRTC sure - but we were talking about using BUNDLE =
generically.<br>
&gt; Or at least that's what I thought you were asking about.&nbsp; If =
we're
only<br>
&gt; talking about using BUNDLE for WebRTC, then not only do you not =
need to<br>
&gt; use a port=3D0, but you also don't need a second offer/answer, =
etc.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; Also, from a purely protocol perspective, it seems quite =
logical
to<br>
&gt; me to use a port number of 0 for an m-line you literally don't want =
to<br>
&gt; use if the far-end doesn't support<br>
&gt; &gt;&gt; BUNDLE - a port number of 0 effectively means
&quot;disabled&quot; today, and<br>
&gt; that's basically what you want.<br>
&gt; &gt;<br>
&gt; &gt; Well, one could also say that it means &quot;Here are some m-
lines&nbsp; I<br>
&gt; want to bundle, with this port, but only if you (the remote =
endpoint)<br>
&gt; wants to use bundle.&quot; :)<br>
&gt; <br>
&gt; We can certainly change the semantics of SDP offers with port=3D0 =
for<br>
&gt; WebRTC, but I think it would be really dangerous to change its<br>
&gt; semantics for general use.<br>
&gt; <br>
&gt; <br>
&gt; &gt;&gt; What I think should happen is the UAS should send an SDP =
answer<br>
&gt; using port=3D0 for those same m-lines, and the UAC should then send =
a new<br>
&gt; offer with those m-lines having the real port number of the =
bundle<br>
&gt; transport, and the far-end should answer that in the same =
fashion.<br>
&gt; &gt;<br>
&gt; &gt; The problem is that the grouping framework does not allow, in =
SDP<br>
&gt; Answers, usage of port=3D0 in grouped m- lines. It would require =
an<br>
&gt; update of the RFC, and I think we want to avoid that.<br>
&gt; <br>
&gt; Considering the number of docs the Unified-Plan proposes to =
either<br>
&gt; create, change, update or whatever... methinks having BUNDLE update =
the<br>
&gt; grouping framework RFC is rather a minor nit. :)<br>
&gt; <br>
&gt; -hadriel<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.or=
g/mailman/listinfo/mmusic</a></span><o:p></o:p></p>

</div>

</div>

</div>

</div>

</div>

</div>

</body>

</html>

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From adam@nostrum.com  Thu Aug  8 12:37:40 2013
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To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se> <1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com> <5203EB64.70605@nostrum.com> <65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com>
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 14:11, Hadriel Kaplan wrote:
> If I send a new SDP Offer 10 minutes later during the session, and I set an existing m-line to port 0 in that Offer, does it mean I want to disable the media, or restart bundling, or what?

We already talked about that in Berlin: it means you want to disable the 
line. We're assigning special semantics to port=0 in the **FIRST** offer 
of a session. Only. Basically, it's a secret handshake that allows you 
to negotiate bundle at the same time as using it without confusing those 
implementations that have no idea what all this bundle stuff means. Once 
you've established that the other end can understand bundle, you don't 
need the port-0 indication.

/a

From hadriel.kaplan@oracle.com  Thu Aug  8 12:42:22 2013
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To: Adam Roach <adam@nostrum.com>
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 3:37 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 14:11, Hadriel Kaplan wrote:
>> If I send a new SDP Offer 10 minutes later during the session, and I =
set an existing m-line to port 0 in that Offer, does it mean I want to =
disable the media, or restart bundling, or what?
>=20
> We already talked about that in Berlin: it means you want to disable =
the line. We're assigning special semantics to port=3D0 in the **FIRST** =
offer of a session. Only. Basically, it's a secret handshake that allows =
you to negotiate bundle at the same time as using it without confusing =
those implementations that have no idea what all this bundle stuff =
means. Once you've established that the other end can understand bundle, =
you don't need the port-0 indication.

Figured... so you're treating it different. "But why? You're introducing =
additional mechanism (and hence complexity) for no technical advantage."
:)

-hadriel


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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 14:41, Hadriel Kaplan wrote:
> On Aug 8, 2013, at 3:37 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/8/13 14:11, Hadriel Kaplan wrote:
>>> If I send a new SDP Offer 10 minutes later during the session, and I set an existing m-line to port 0 in that Offer, does it mean I want to disable the media, or restart bundling, or what?
>> We already talked about that in Berlin: it means you want to disable the line. We're assigning special semantics to port=0 in the **FIRST** offer of a session. Only. Basically, it's a secret handshake that allows you to negotiate bundle at the same time as using it without confusing those implementations that have no idea what all this bundle stuff means. Once you've established that the other end can understand bundle, you don't need the port-0 indication.
> Figured... so you're treating it different. "But why? You're introducing additional mechanism (and hence complexity) for no technical advantage."
>

The technical advantage is exactly the property I just mentioned: it 
"allows you to negotiate bundle at the same time as using it without 
confusing those implementations that have no idea what all this bundle 
stuff means." That's a pretty important property.

/a

From kase.saylor@swri.org  Thu Aug  8 12:44:27 2013
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From: "Saylor, Kase J." <kase.saylor@swri.org>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] RTSP for data recorder design
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Subject: Re: [MMUSIC] RTSP for data recorder design
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--_000_5D319E1E0055408A8973E110DFBC9BBDswriorg_
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Paul,

Thanks for the link for siprec, unfortunately I am looking for a current IE=
TF RFC that could be used. As siprec is informational at this time, it's no=
t really an option. I think I'm pretty close with RTSP, but there's just a =
few things that I haven't sorted out. Perhaps I'll try to condense my confu=
sing email into something a little bit more concise. Here we go:

1. I have data sources that provide data primarily as "streams" of multicas=
t datagrams. We have specified the format of those datagrams, and we now ne=
ed a way to record them so that they can be retrieved at a later time.

2. We have created a SOAP-based (role you own) interface that would provide=
 the needed functionality, but if I could accomplish that functionality wit=
h RTSP, that would be better for the overall community.

3. I need the following functionality:
a. Nominate certain (or all) multicast streams for recording (I assume ANNO=
UNCE would be the way to do that)
b. Record said streams whenever needed
c. Allow a client to find out what streams are available
d. Allow the client to playback those streams

So my first questions was about "adding" streams to a session. After much r=
ambling and attempts to provide example RTSP communications between client =
and server, my last question was about how, using DESCRIBE, would I be able=
 to show all the streams that were in a single session (based upon the assu=
mption that multiple data streams could be group under one session).

Does this at all help clear up my original post? Thanks again for your time=
 and help.

--
Regards,

Kase Saylor
Southwest Research Institute

On Aug 8, 2013, at 2:28 PM, Kase Saylor <ksaylor@swri.org<mailto:ksaylor@sw=
ri.org>> wrote:

Paul,

Thanks for the link for siprec, unfortunately I am looking for a current IE=
TF RFC that could be used. As siprec is informational at this time, it's no=
t really an option. I think I'm pretty close with RTSP, but there's just a =
few things that I haven't sorted out. Perhaps I'll try to condense my confu=
sing email into something a little bit more concise. Here we go:

1. I have data sources that provide data primarily as "streams" of multicas=
t datagrams. We have specified the format of those datagrams, and we now ne=
ed a way to record them so that they can be retrieved at a later time.

2. We have created a SOAP-based (role you own) interface that would provide=
 the needed functionality, but if I could accomplish that functionality wit=
h RTSP, that would be better for the overall community.

3. I need the following functionality:
a. Nominate certain (or all) multicast streams for recording (I assume ANNO=
UNCE would be the way to do that)
b. Record said streams whenever needed
c. Allow a client to find out what streams are available
d. Allow the client to playback those streams

So my first questions was about "adding" streams to a session. After much r=
ambling and attempts to provide example RTSP communications between client =
and server, my last question was about how, using DESCRIBE, would I be able=
 to show all the streams that were in a single session (based upon the assu=
mption that multiple data streams could be group under one session).

Does this at all help clear up my original post? Thanks again for your time=
 and help.

--
Regards,

Kase Saylor
Southwest Research Institute


On Aug 8, 2013, at 2:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyz=
ivat@alum.mit.edu>>
wrote:

Kase,

Its hard to discern your exact requirements. But you should at least look a=
t the work in progress by the siprec workgroup. You can find the relevant d=
rafts at:

http://datatracker.ietf.org/wg/siprec/

This may not be what you need, but is worth a look.

Thanks,
Paul

On 8/8/13 8:49 PM, Saylor, Kase J. wrote:
To Whom it may concern,

I am helping the US Army develop standards for a network recorder on US
Army vehicles. We like to use available standards when at all possible,
and my research seems to indicate that RTSP would be a great candidate
standard. I've been poring over the RTSP specification (along with the
SDP standard) and I have your douce code for LiveMedia. However, I still
have a few questions concerning the RTSP standard and its use for our
particular need. It seems like there is plenty of examples on the
Internet for the PLAY method, but not much for RECORD. I really hope
that someone may be able to provide some insight that will get me moving
in the right direction. On to the questions!

Our recorder requires the ability to "nominate" data streams for
recording, so I believe the correct RTSP method is ANNOUNCE. My first
question is when I was to add a stream using ANNOUNCE, if I use the same
Session number for multiple streams, does this basically create a
presentation? For example:

C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
          CSeq: 3
          Date: 7 Aug 2013 15:35:06 GMT
          Session: 47112344
          Content-Type: application/sdp
          Content-Length: 332

          v=3D0
          o=3D- 2890844526 2890845468 IN IP4 192.168.0.123
          s=3DThreat Stream 1
          i=3DThreat stream from threat service on the Boomerang
          c=3DIN IP4 239.192.0.123/127
          t=3D0 0
          a=3Dunique_id:threat1
  a=3Ddata_type:threat
          m=3Dmessage 12345

(some response back from server)

C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
          CSeq: 4
          Date: 7 Aug 2013 15:35:07 GMT
          Session: 47112344
          Content-Type: application/sdp
          Content-Length: 332

          v=3D0
          o=3D- 2890844527 2890845471 IN IP4 192.168.0.123
          s=3DPosition Stream 1
          i=3DPosition stream from position service
          c=3DIN IP4 239.192.0.124/127
          t=3D0 0
          a=3Dunique_id:pos1
  a=3Ddata_type:position
          m=3Dmessage 12346
(some response back from server)

Based upon these two announce messages, is it safe to assume that the
following command would cause the recorder to start recording those two
data streams (assuming the current date/time is 8/4/2013 1530 GMT)?

    C->S: RECORD
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
          CSeq: 5
          Content-Type: text/parameters
          Session: 47112344
  Range: clock=3D20130804T153000Z-

Furthermore, would the following command stop the recorder (assuming the
current date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in the
RFC for stopping a recording.

    C->S: RECORD
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
          CSeq: 6
          Content-Type: text/parameters
          Session: 47112344
  Range: clock=3D- 20130804T153200Z


I apologize but I have at least one or two more questions ;-) After
tearing down a session, how would one go back and playback recorded
data? From my RECORD example I've assumed that providing a URI with a
"path" based upon record start time, could be one way to mark recording
events. Do you think this is an acceptable approach? Then the play
command would something like:

    C->S: PLAY
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
          CSeq: 6
          Content-Type: text/parameters
          Session: 1234567

I suppose I could specify unique parameters and use the GET_PARAMETER
method to list the available recordings. Perhaps something like:

    C->S: SETUP rtsp://192.168.0.1/mission_recorder RTSP/1.0
          CSeq: 1
  Transport: unicast;client_port=3D4588-4589

    S->C: RTSP/1.0 200 OK
          CSeq: 1
  Date: 5 Aug 2013 11:15:07 GMT
          Session: 1234567
          Transport: unicast;client_port=3D4588-4589;server_port=3D6256-625=
7

    C->S: GET_PARAMETER rtsp://192.168.0.1/mission_recorder RTSP/1.0
          CSeq: 7
          Content-Type: text/parameters
          Session: 1234567
          Content-Length: 15

  list_recorded_data


    S->C: RTSP/1.0 200 OK
          CSeq: 7
  Date: 5 Aug 2013 11:15:10 GMT
          Content-Length: 46
          Content-Type: text/parameters

          list_recorded_data:20130804T153000Z,20130804T213000Z

    C->S: TEARDOWN rtsp://192.168.0.1/mission_recorder RTSP/1.0
          CSeq: 8
  Session: 1234567

    S->C: RTSP/1.0 200 OK
          CSeq: 8


Then we'd specify the proper way to interpret the response from the
server and open up a new RTSP session to playback the recorded data. For
example:

    C->S: SETUP
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
          CSeq: 1
  Transport: multicast;client_port=3D4588-4589

    S->C: RTSP/1.0 200 OK
          CSeq: 1
  Date: 6 Aug 2013 11:15:07 GMT
          Session: 7654321
          Transport: multicast;client_port=3D4588-4589;server_port=3D6256-6=
257

    C->S: PLAY
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
          CSeq: 2
          Content-Type: text/parameters
          Session: 7654321

    S->C: RTSP/1.0 200 OK
          CSeq: 2
  Date: 6 Aug 2013 11:15:10 GMT

Okay, last question, would the DESCRIBE method for
rtsp://192.168.0.1/mission_recorder/20130804T153000Z be expected to
describe the "streams" that were recorded? In other words could the
following be expected?

C->S: DESCRIBE rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.=
0
          CSeq: 312
          Accept: application/sdp, application/rtsl, application/mheg

S->C: RTSP/1.0 200 OK
          CSeq: 312
          Date: 7 Aug 2013 15:35:07 GMT
          Content-Type: application/sdp
          Content-Length: 332

          v=3D0
          o=3D- 2890844527 2890845471 IN IP4 192.168.0.123
          s=3DPosition Stream 1
          i=3DThreat stream from threat service on the Boomerang
          c=3DIN IP4 239.192.0.124/127
          t=3D0 0
          a=3Dunique_id:pos1
  a=3Ddata_type:position
          m=3Dmessage 12346
  s=3DPosition Stream 1
          i=3DPosition stream from position service
          c=3DIN IP4 239.192.0.124/127
          t=3D0 0
          a=3Dunique_id:pos1
  a=3Ddata_type:position
          m=3Dmessage 12346

I really have no idea how multiple streams would be described!

Thanks in advance for any help!
--
Regards,

Kase Saylor
Southwest Research Institute



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<span style=3D"font-family: monospace; ">Paul,</span><br style=3D"font-fami=
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<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">Thanks for the link for siprec, un=
fortunately I am looking for a current IETF RFC that could be used. As sipr=
ec is informational at this time, it's not really an option. I think I'm pr=
etty close with RTSP, but there's
 just a few things that I haven't sorted out. Perhaps I'll try to condense =
my confusing email into something a little bit more concise. Here we go:</s=
pan><br style=3D"font-family: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">1. I have data sources that provid=
e data primarily as &quot;streams&quot; of multicast datagrams. We have spe=
cified the format of those datagrams, and we now need a way to record them =
so that they can be retrieved at a later time.</span><br style=3D"font-fami=
ly: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">2. We have created a SOAP-based (r=
ole you own) interface that would provide the needed functionality, but if =
I could accomplish that functionality with RTSP, that would be better for t=
he overall community.</span><br style=3D"font-family: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">3. I need the following functional=
ity:</span><br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">a. Nominate certain (or all) multi=
cast streams for recording (I assume ANNOUNCE would be the way to do that)<=
/span><br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">b. Record said streams whenever ne=
eded</span><br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">c. Allow a client to find out what=
 streams are available</span><br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">d. Allow the client to playback th=
ose streams</span><br style=3D"font-family: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">So my first questions was about &q=
uot;adding&quot; streams to a session. After much rambling and attempts to =
provide example RTSP communications between client and server, my last ques=
tion was about how, using DESCRIBE, would I
 be able to show all the streams that were in a single session (based upon =
the assumption that multiple data streams could be group under one session)=
.</span><br style=3D"font-family: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">Does this at all help clear up my =
original post? Thanks again for your time and help.</span><br style=3D"font=
-family: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">--</span><br style=3D"font-family:=
 monospace; ">
<span style=3D"font-family: monospace; ">Regards,</span><br style=3D"font-f=
amily: monospace; ">
<br style=3D"font-family: monospace; ">
<span style=3D"font-family: monospace; ">Kase Saylor</span><br style=3D"fon=
t-family: monospace; ">
<span style=3D"font-family: monospace; ">Southwest Research Institute</span=
><br>
<div apple-content-edited=3D"true"></div>
<br>
<div>
<div>On Aug 8, 2013, at 2:28 PM, Kase Saylor &lt;<a href=3D"mailto:ksaylor@=
swri.org">ksaylor@swri.org</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">Paul,<br>
<br>
Thanks for the link for siprec, unfortunately I am looking for a current IE=
TF RFC that could be used. As siprec is informational at this time, it's no=
t really an option. I think I'm pretty close with RTSP, but there's just a =
few things that I haven't sorted
 out. Perhaps I'll try to condense my confusing email into something a litt=
le bit more concise. Here we go:<br>
<br>
1. I have data sources that provide data primarily as &quot;streams&quot; o=
f multicast datagrams. We have specified the format of those datagrams, and=
 we now need a way to record them so that they can be retrieved at a later =
time.<br>
<br>
2. We have created a SOAP-based (role you own) interface that would provide=
 the needed functionality, but if I could accomplish that functionality wit=
h RTSP, that would be better for the overall community.<br>
<br>
3. I need the following functionality:<br>
a. Nominate certain (or all) multicast streams for recording (I assume ANNO=
UNCE would be the way to do that)<br>
b. Record said streams whenever needed<br>
c. Allow a client to find out what streams are available<br>
d. Allow the client to playback those streams<br>
<br>
So my first questions was about &quot;adding&quot; streams to a session. Af=
ter much rambling and attempts to provide example RTSP communications betwe=
en client and server, my last question was about how, using DESCRIBE, would=
 I be able to show all the streams that were
 in a single session (based upon the assumption that multiple data streams =
could be group under one session).<br>
<br>
Does this at all help clear up my original post? Thanks again for your time=
 and help.<br>
<br>
--<br>
Regards,<br>
<br>
Kase Saylor<br>
Southwest Research Institute<br>
<br>
<br>
On Aug 8, 2013, at 2:00 PM, Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alu=
m.mit.edu">pkyzivat@alum.mit.edu</a>&gt;<br>
wrote:<br>
<br>
<blockquote type=3D"cite">Kase,<br>
<br>
Its hard to discern your exact requirements. But you should at least look a=
t the work in progress by the siprec workgroup. You can find the relevant d=
rafts at:<br>
<br>
<a href=3D"http://datatracker.ietf.org/wg/siprec/">http://datatracker.ietf.=
org/wg/siprec/</a><br>
<br>
This may not be what you need, but is worth a look.<br>
<br>
<span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Thanks,<br>
<span class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Paul<br>
<br>
On 8/8/13 8:49 PM, Saylor, Kase J. wrote:<br>
<blockquote type=3D"cite">To Whom it may concern,<br>
<br>
I am helping the US Army develop standards for a network recorder on US<br>
Army vehicles. We like to use available standards when at all possible,<br>
and my research seems to indicate that RTSP would be a great candidate<br>
standard. I've been poring over the RTSP specification (along with the<br>
SDP standard) and I have your douce code for LiveMedia. However, I still<br=
>
have a few questions concerning the RTSP standard and its use for our<br>
particular need. It seems like there is plenty of examples on the<br>
Internet for the PLAY method, but not much for RECORD. I really hope<br>
that someone may be able to provide some insight that will get me moving<br=
>
in the right direction. On to the questions!<br>
<br>
Our recorder requires the ability to &quot;nominate&quot; data streams for<=
br>
recording, so I believe the correct RTSP method is ANNOUNCE. My first<br>
question is when I was to add a stream using ANNOUNCE, if I use the same<br=
>
Session number for multiple streams, does this basically create a<br>
presentation? For example:<br>
<br>
C-&gt;S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 3<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Date: 7 Aug 201=
3 15:35:06 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 471123=
44<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: a=
pplication/sdp<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Length:=
 332<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;v=3D0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;o=3D- 289084452=
6 2890845468 IN IP4 192.168.0.123<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;s=3DThreat Stre=
am 1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;i=3DThreat stre=
am from threat service on the Boomerang<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;c=3DIN IP4 239.=
192.0.123/127<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;t=3D0 0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;a=3Dunique_id:t=
hreat1<br>
&nbsp;&nbsp;a=3Ddata_type:threat<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;m=3Dmessage 123=
45<br>
<br>
(some response back from server)<br>
<br>
C-&gt;S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 4<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Date: 7 Aug 201=
3 15:35:07 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 471123=
44<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: a=
pplication/sdp<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Length:=
 332<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;v=3D0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;o=3D- 289084452=
7 2890845471 IN IP4 192.168.0.123<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;s=3DPosition St=
ream 1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;i=3DPosition st=
ream from position service<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;c=3DIN IP4 239.=
192.0.124/127<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;t=3D0 0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;a=3Dunique_id:p=
os1<br>
&nbsp;&nbsp;a=3Ddata_type:position<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;m=3Dmessage 123=
46<br>
(some response back from server)<br>
<br>
Based upon these two announce messages, is it safe to assume that the<br>
following command would cause the recorder to start recording those two<br>
data streams (assuming the current date/time is 8/4/2013 1530 GMT)?<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: RECORD<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 5<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 471123=
44<br>
&nbsp;&nbsp;Range: clock=3D20130804T153000Z-<br>
<br>
Furthermore, would the following command stop the recorder (assuming the<br=
>
current date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in the<br>
RFC for stopping a recording.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: RECORD<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 6<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 471123=
44<br>
&nbsp;&nbsp;Range: clock=3D- 20130804T153200Z<br>
<br>
<br>
I apologize but I have at least one or two more questions ;-) After<br>
tearing down a session, how would one go back and playback recorded<br>
data? From my RECORD example I've assumed that providing a URI with a<br>
&quot;path&quot; based upon record start time, could be one way to mark rec=
ording<br>
events. Do you think this is an acceptable approach? Then the play<br>
command would something like:<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: PLAY<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 6<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 123456=
7<br>
<br>
I suppose I could specify unique parameters and use the GET_PARAMETER<br>
method to list the available recordings. Perhaps something like:<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: SETUP rtsp://192.168.0.1/mission_recorder =
RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 1<br>
&nbsp;&nbsp;Transport: unicast;client_port=3D4588-4589<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 1<br>
&nbsp;&nbsp;Date: 5 Aug 2013 11:15:07 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 123456=
7<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Transport: unic=
ast;client_port=3D4588-4589;server_port=3D6256-6257<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: GET_PARAMETER rtsp://192.168.0.1/mission_r=
ecorder RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 7<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 123456=
7<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Length:=
 15<br>
<br>
&nbsp;&nbsp;list_recorded_data<br>
<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 7<br>
&nbsp;&nbsp;Date: 5 Aug 2013 11:15:10 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Length:=
 46<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;list_recorded_d=
ata:20130804T153000Z,20130804T213000Z<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: TEARDOWN rtsp://192.168.0.1/mission_record=
er RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 8<br>
&nbsp;&nbsp;Session: 1234567<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 8<br>
<br>
<br>
Then we'd specify the proper way to interpret the response from the<br>
server and open up a new RTSP session to playback the recorded data. For<br=
>
example:<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: SETUP<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 1<br>
&nbsp;&nbsp;Transport: multicast;client_port=3D4588-4589<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 1<br>
&nbsp;&nbsp;Date: 6 Aug 2013 11:15:07 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 765432=
1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Transport: mult=
icast;client_port=3D4588-4589;server_port=3D6256-6257<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;C-&gt;S: PLAY<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 2<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: t=
ext/parameters<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Session: 765432=
1<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 2<br>
&nbsp;&nbsp;Date: 6 Aug 2013 11:15:10 GMT<br>
<br>
Okay, last question, would the DESCRIBE method for<br>
rtsp://192.168.0.1/mission_recorder/20130804T153000Z be expected to<br>
describe the &quot;streams&quot; that were recorded? In other words could t=
he<br>
following be expected?<br>
<br>
C-&gt;S: DESCRIBE rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP=
/1.0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 312<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Accept: applica=
tion/sdp, application/rtsl, application/mheg<br>
<br>
S-&gt;C: RTSP/1.0 200 OK<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;CSeq: 312<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Date: 7 Aug 201=
3 15:35:07 GMT<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Type: a=
pplication/sdp<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Content-Length:=
 332<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;v=3D0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;o=3D- 289084452=
7 2890845471 IN IP4 192.168.0.123<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;s=3DPosition St=
ream 1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;i=3DThreat stre=
am from threat service on the Boomerang<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;c=3DIN IP4 239.=
192.0.124/127<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;t=3D0 0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;a=3Dunique_id:p=
os1<br>
&nbsp;&nbsp;a=3Ddata_type:position<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;m=3Dmessage 123=
46<br>
&nbsp;&nbsp;s=3DPosition Stream 1<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;i=3DPosition st=
ream from position service<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;c=3DIN IP4 239.=
192.0.124/127<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;t=3D0 0<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;a=3Dunique_id:p=
os1<br>
&nbsp;&nbsp;a=3Ddata_type:position<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;m=3Dmessage 123=
46<br>
<br>
I really have no idea how multiple streams would be described!<br>
<br>
Thanks in advance for any help!<br>
--<br>
Regards,<br>
<br>
Kase Saylor<br>
Southwest Research Institute<br>
<br>
<br>
<br>
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Subject: Re: [MMUSIC] RTSP for data recorder design
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Kase,

Based on the additional info below I don't think siprec would fit your 
needs anyway. I'm not involved with RTSP, so I can't comment on that.

	Thanks,
	Paul

On 8/8/13 9:44 PM, Saylor, Kase J. wrote:
> Paul,
>
> Thanks for the link for siprec, unfortunately I am looking for a current
> IETF RFC that could be used. As siprec is informational at this time,
> it's not really an option. I think I'm pretty close with RTSP, but
> there's just a few things that I haven't sorted out. Perhaps I'll try to
> condense my confusing email into something a little bit more concise.
> Here we go:
>
> 1. I have data sources that provide data primarily as "streams" of
> multicast datagrams. We have specified the format of those datagrams,
> and we now need a way to record them so that they can be retrieved at a
> later time.
>
> 2. We have created a SOAP-based (role you own) interface that would
> provide the needed functionality, but if I could accomplish that
> functionality with RTSP, that would be better for the overall community.
>
> 3. I need the following functionality:
> a. Nominate certain (or all) multicast streams for recording (I assume
> ANNOUNCE would be the way to do that)
> b. Record said streams whenever needed
> c. Allow a client to find out what streams are available
> d. Allow the client to playback those streams
>
> So my first questions was about "adding" streams to a session. After
> much rambling and attempts to provide example RTSP communications
> between client and server, my last question was about how, using
> DESCRIBE, would I be able to show all the streams that were in a single
> session (based upon the assumption that multiple data streams could be
> group under one session).
>
> Does this at all help clear up my original post? Thanks again for your
> time and help.
>
> --
> Regards,
>
> Kase Saylor
> Southwest Research Institute
>
> On Aug 8, 2013, at 2:28 PM, Kase Saylor <ksaylor@swri.org
> <mailto:ksaylor@swri.org>> wrote:
>
>> Paul,
>>
>> Thanks for the link for siprec, unfortunately I am looking for a
>> current IETF RFC that could be used. As siprec is informational at
>> this time, it's not really an option. I think I'm pretty close with
>> RTSP, but there's just a few things that I haven't sorted out. Perhaps
>> I'll try to condense my confusing email into something a little bit
>> more concise. Here we go:
>>
>> 1. I have data sources that provide data primarily as "streams" of
>> multicast datagrams. We have specified the format of those datagrams,
>> and we now need a way to record them so that they can be retrieved at
>> a later time.
>>
>> 2. We have created a SOAP-based (role you own) interface that would
>> provide the needed functionality, but if I could accomplish that
>> functionality with RTSP, that would be better for the overall community.
>>
>> 3. I need the following functionality:
>> a. Nominate certain (or all) multicast streams for recording (I assume
>> ANNOUNCE would be the way to do that)
>> b. Record said streams whenever needed
>> c. Allow a client to find out what streams are available
>> d. Allow the client to playback those streams
>>
>> So my first questions was about "adding" streams to a session. After
>> much rambling and attempts to provide example RTSP communications
>> between client and server, my last question was about how, using
>> DESCRIBE, would I be able to show all the streams that were in a
>> single session (based upon the assumption that multiple data streams
>> could be group under one session).
>>
>> Does this at all help clear up my original post? Thanks again for your
>> time and help.
>>
>> --
>> Regards,
>>
>> Kase Saylor
>> Southwest Research Institute
>>
>>
>> On Aug 8, 2013, at 2:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
>> <mailto:pkyzivat@alum.mit.edu>>
>> wrote:
>>
>>> Kase,
>>>
>>> Its hard to discern your exact requirements. But you should at least
>>> look at the work in progress by the siprec workgroup. You can find
>>> the relevant drafts at:
>>>
>>> http://datatracker.ietf.org/wg/siprec/
>>>
>>> This may not be what you need, but is worth a look.
>>>
>>> Thanks,
>>> Paul
>>>
>>> On 8/8/13 8:49 PM, Saylor, Kase J. wrote:
>>>> To Whom it may concern,
>>>>
>>>> I am helping the US Army develop standards for a network recorder on US
>>>> Army vehicles. We like to use available standards when at all possible,
>>>> and my research seems to indicate that RTSP would be a great candidate
>>>> standard. I've been poring over the RTSP specification (along with the
>>>> SDP standard) and I have your douce code for LiveMedia. However, I still
>>>> have a few questions concerning the RTSP standard and its use for our
>>>> particular need. It seems like there is plenty of examples on the
>>>> Internet for the PLAY method, but not much for RECORD. I really hope
>>>> that someone may be able to provide some insight that will get me moving
>>>> in the right direction. On to the questions!
>>>>
>>>> Our recorder requires the ability to "nominate" data streams for
>>>> recording, so I believe the correct RTSP method is ANNOUNCE. My first
>>>> question is when I was to add a stream using ANNOUNCE, if I use the same
>>>> Session number for multiple streams, does this basically create a
>>>> presentation? For example:
>>>>
>>>> C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
>>>>           CSeq: 3
>>>>           Date: 7 Aug 2013 15:35:06 GMT
>>>>           Session: 47112344
>>>>           Content-Type: application/sdp
>>>>           Content-Length: 332
>>>>
>>>>           v=0
>>>>           o=- 2890844526 2890845468 IN IP4 192.168.0.123
>>>>           s=Threat Stream 1
>>>>           i=Threat stream from threat service on the Boomerang
>>>>           c=IN IP4 239.192.0.123/127
>>>>           t=0 0
>>>>           a=unique_id:threat1
>>>>   a=data_type:threat
>>>>           m=message 12345
>>>>
>>>> (some response back from server)
>>>>
>>>> C->S: ANNOUNCE rtsp://192.168.0.1/mission_recorder RTSP/1.0
>>>>           CSeq: 4
>>>>           Date: 7 Aug 2013 15:35:07 GMT
>>>>           Session: 47112344
>>>>           Content-Type: application/sdp
>>>>           Content-Length: 332
>>>>
>>>>           v=0
>>>>           o=- 2890844527 2890845471 IN IP4 192.168.0.123
>>>>           s=Position Stream 1
>>>>           i=Position stream from position service
>>>>           c=IN IP4 239.192.0.124/127
>>>>           t=0 0
>>>>           a=unique_id:pos1
>>>>   a=data_type:position
>>>>           m=message 12346
>>>> (some response back from server)
>>>>
>>>> Based upon these two announce messages, is it safe to assume that the
>>>> following command would cause the recorder to start recording those two
>>>> data streams (assuming the current date/time is 8/4/2013 1530 GMT)?
>>>>
>>>>     C->S: RECORD
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>>>>           CSeq: 5
>>>>           Content-Type: text/parameters
>>>>           Session: 47112344
>>>>   Range: clock=20130804T153000Z-
>>>>
>>>> Furthermore, would the following command stop the recorder (assuming the
>>>> current date/time is 8/4/2013 1532 GMT)? I didn't see anywhere in the
>>>> RFC for stopping a recording.
>>>>
>>>>     C->S: RECORD
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>>>>           CSeq: 6
>>>>           Content-Type: text/parameters
>>>>           Session: 47112344
>>>>   Range: clock=- 20130804T153200Z
>>>>
>>>>
>>>> I apologize but I have at least one or two more questions ;-) After
>>>> tearing down a session, how would one go back and playback recorded
>>>> data? From my RECORD example I've assumed that providing a URI with a
>>>> "path" based upon record start time, could be one way to mark recording
>>>> events. Do you think this is an acceptable approach? Then the play
>>>> command would something like:
>>>>
>>>>     C->S: PLAY
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>>>>           CSeq: 6
>>>>           Content-Type: text/parameters
>>>>           Session: 1234567
>>>>
>>>> I suppose I could specify unique parameters and use the GET_PARAMETER
>>>> method to list the available recordings. Perhaps something like:
>>>>
>>>>     C->S: SETUP rtsp://192.168.0.1/mission_recorder RTSP/1.0
>>>>           CSeq: 1
>>>>   Transport: unicast;client_port=4588-4589
>>>>
>>>>     S->C: RTSP/1.0 200 OK
>>>>           CSeq: 1
>>>>   Date: 5 Aug 2013 11:15:07 GMT
>>>>           Session: 1234567
>>>>           Transport: unicast;client_port=4588-4589;server_port=6256-6257
>>>>
>>>>     C->S: GET_PARAMETER rtsp://192.168.0.1/mission_recorder RTSP/1.0
>>>>           CSeq: 7
>>>>           Content-Type: text/parameters
>>>>           Session: 1234567
>>>>           Content-Length: 15
>>>>
>>>>   list_recorded_data
>>>>
>>>>
>>>>     S->C: RTSP/1.0 200 OK
>>>>           CSeq: 7
>>>>   Date: 5 Aug 2013 11:15:10 GMT
>>>>           Content-Length: 46
>>>>           Content-Type: text/parameters
>>>>
>>>>           list_recorded_data:20130804T153000Z,20130804T213000Z
>>>>
>>>>     C->S: TEARDOWN rtsp://192.168.0.1/mission_recorder RTSP/1.0
>>>>           CSeq: 8
>>>>   Session: 1234567
>>>>
>>>>     S->C: RTSP/1.0 200 OK
>>>>           CSeq: 8
>>>>
>>>>
>>>> Then we'd specify the proper way to interpret the response from the
>>>> server and open up a new RTSP session to playback the recorded data. For
>>>> example:
>>>>
>>>>     C->S: SETUP
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>>>>           CSeq: 1
>>>>   Transport: multicast;client_port=4588-4589
>>>>
>>>>     S->C: RTSP/1.0 200 OK
>>>>           CSeq: 1
>>>>   Date: 6 Aug 2013 11:15:07 GMT
>>>>           Session: 7654321
>>>>           Transport:
>>>> multicast;client_port=4588-4589;server_port=6256-6257
>>>>
>>>>     C->S: PLAY
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z RTSP/1.0
>>>>           CSeq: 2
>>>>           Content-Type: text/parameters
>>>>           Session: 7654321
>>>>
>>>>     S->C: RTSP/1.0 200 OK
>>>>           CSeq: 2
>>>>   Date: 6 Aug 2013 11:15:10 GMT
>>>>
>>>> Okay, last question, would the DESCRIBE method for
>>>> rtsp://192.168.0.1/mission_recorder/20130804T153000Z be expected to
>>>> describe the "streams" that were recorded? In other words could the
>>>> following be expected?
>>>>
>>>> C->S: DESCRIBE rtsp://192.168.0.1/mission_recorder/20130804T153000Z
>>>> RTSP/1.0
>>>>           CSeq: 312
>>>>           Accept: application/sdp, application/rtsl, application/mheg
>>>>
>>>> S->C: RTSP/1.0 200 OK
>>>>           CSeq: 312
>>>>           Date: 7 Aug 2013 15:35:07 GMT
>>>>           Content-Type: application/sdp
>>>>           Content-Length: 332
>>>>
>>>>           v=0
>>>>           o=- 2890844527 2890845471 IN IP4 192.168.0.123
>>>>           s=Position Stream 1
>>>>           i=Threat stream from threat service on the Boomerang
>>>>           c=IN IP4 239.192.0.124/127
>>>>           t=0 0
>>>>           a=unique_id:pos1
>>>>   a=data_type:position
>>>>           m=message 12346
>>>>   s=Position Stream 1
>>>>           i=Position stream from position service
>>>>           c=IN IP4 239.192.0.124/127
>>>>           t=0 0
>>>>           a=unique_id:pos1
>>>>   a=data_type:position
>>>>           m=message 12346
>>>>
>>>> I really have no idea how multiple streams would be described!
>>>>
>>>> Thanks in advance for any help!
>>>> --
>>>> Regards,
>>>>
>>>> Kase Saylor
>>>> Southwest Research Institute
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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>


From hadriel.kaplan@oracle.com  Thu Aug  8 13:52:27 2013
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To: Adam Roach <adam@nostrum.com>
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 3:43 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 14:41, Hadriel Kaplan wrote:
>> On Aug 8, 2013, at 3:37 PM, Adam Roach <adam@nostrum.com> wrote:
>>=20
>>> On 8/8/13 14:11, Hadriel Kaplan wrote:
>>>> If I send a new SDP Offer 10 minutes later during the session, and =
I set an existing m-line to port 0 in that Offer, does it mean I want to =
disable the media, or restart bundling, or what?
>>> We already talked about that in Berlin: it means you want to disable =
the line. We're assigning special semantics to port=3D0 in the **FIRST** =
offer of a session. Only. Basically, it's a secret handshake that allows =
you to negotiate bundle at the same time as using it without confusing =
those implementations that have no idea what all this bundle stuff =
means. Once you've established that the other end can understand bundle, =
you don't need the port-0 indication.
>> Figured... so you're treating it different. "But why? You're =
introducing additional mechanism (and hence complexity) for no technical =
advantage."
>>=20
>=20
> The technical advantage is exactly the property I just mentioned: it =
"allows you to negotiate bundle at the same time as using it without =
confusing those implementations that have no idea what all this bundle =
stuff means." That's a pretty important property.

Yeah, I as being a wise-a$$.  But my point is it won't really have that =
property, in practice.  As much as we'd like it to be otherwise, there =
really are things in the middle that will cause this thing to screw up =
in weird ways.  It's just not as simple as "if they don't understand the =
new a=3Dbundle-only attribute, then they'll reject the request".[1]

It's not unlike the need to use different port numbers for normal bundle =
lines.

-hadriel
[1] BTW, I'm not speaking about my employer's SBCs in particular - I =
actually have no idea what we'll do with an UP bundle-only thing lying =
about a port=3D0.  We could handle it all correctly, or tear the call =
down later, or call WOPR and tell it to play Global Thermonuclear War.  =
If something's broken in our implementation, we'll fix it... eventually. =
 I'm more worried about other systems, from vendors who don't attend the =
IETF; and I'm worried about the people who have to run SIP networks, for =
whom this stuff is not a greenfield.



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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/8/13 15:51, Hadriel Kaplan wrote:
> BTW, I'm not speaking about my employer's SBCs in particular - I actually have no idea what we'll do with an UP bundle-only thing lying about a port=0.  We could handle it all correctly, or tear the call down later, or call WOPR and tell it to play Global Thermonuclear War.  If something's broken in our implementation, we'll fix it... eventually.  I'm more worried about other systems, from vendors who don't attend the IETF; and I'm worried about the people who have to run SIP networks, for whom this stuff is not a greenfield.

The thing is: no matter what we do here, it's not going to look 100% 
like the stuff that came before it. All we can do is look at the 
products we know about, look at the existing published semantics for 
protocol elements, and make educated guesses about what is likely to 
cause the fewest problems. We *know* that using the same port numbers 
for multiple m-lines causes bad behavior in more than a few endpoints, 
so that's out. On the other hand, the semantics of port=0 are pretty 
well defined, elicit the exact response we want for remote endpoints, 
and avoid egregiously large port allocations during call setup (which is 
an important goal).

I think that's a lot more solid set of reasons than the opposing 
rationale you proffer of "there may or may not be products somewhere in 
the world that don't really like this approach," not least of all 
because that statement is true of any approach we come up with.

/a

From christer.holmberg@ericsson.com  Thu Aug  8 14:35:13 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
Thread-Index: AQHOlDHLQN7nRiDYV02hbFJeK66gpJmLcqaAgAAt4U3//+gtgIAAAS4AgAACXICAAAdHAIAAAUGAgAAAYQCAABMsgIAAAuWAgAAqqc0=
Date: Thu, 8 Aug 2013 21:35:00 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se> <1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com> <5203EB64.70605@nostrum.com> <65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com> <5203F37A.8030204@nostrum.com> <33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com> <5203F4D8.9040904@nostrum.com> <9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com>, <5204075B.1080108@nostrum.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hadriel,

I'm a little confused. I thought you wanted port zero in the Answer, but th=
e discussion now seems to be around port zero in the Offer.

Or, have I missed something?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Adam Roach [adam@nostrum.com]
To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic@ietf=
.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?
On 8/8/13 15:51, Hadriel Kaplan wrote:
> BTW, I'm not speaking about my employer's SBCs in particular - I actually=
 have no idea what we'll do with an UP bundle-only thing lying about a port=
=3D0.  We could handle it all correctly, or tear the call down later, or ca=
ll WOPR and tell it to play Global Thermonuclear War.  If something's broke=
n in our implementation, we'll fix it... eventually.  I'm more worried abou=
t other systems, from vendors who don't attend the IETF; and I'm worried ab=
out the people who have to run SIP networks, for whom this stuff is not a g=
reenfield.

The thing is: no matter what we do here, it's not going to look 100%
like the stuff that came before it. All we can do is look at the
products we know about, look at the existing published semantics for
protocol elements, and make educated guesses about what is likely to
cause the fewest problems. We *know* that using the same port numbers
for multiple m-lines causes bad behavior in more than a few endpoints,
so that's out. On the other hand, the semantics of port=3D0 are pretty
well defined, elicit the exact response we want for remote endpoints,
and avoid egregiously large port allocations during call setup (which is
an important goal).

I think that's a lot more solid set of reasons than the opposing
rationale you proffer of "there may or may not be products somewhere in
the world that don't really like this approach," not least of all
because that statement is true of any approach we come up with.

/a

--_000_7594FB04B1934943A5C02806D1A2204B1C422A4FESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hadriel,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I'm a little confused. I thought you wanted por=
t zero in the Answer, but the discussion now seems to be around port zero i=
n the Offer.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Or, have I missed something?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Adam Roach [adam@nostrum.com]<br>
<b>To:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmus=
ic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt; BTW, I'm not speaking about my employer's SBCs in particular - I actua=
lly have no idea what we'll do with an UP bundle-only thing lying about a p=
ort=3D0.&nbsp; We could handle it all correctly, or tear the call down late=
r, or call WOPR and tell it to play Global
 Thermonuclear War.&nbsp; If something's broken in our implementation, we'l=
l fix it... eventually.&nbsp; I'm more worried about other systems, from ve=
ndors who don't attend the IETF; and I'm worried about the people who have =
to run SIP networks, for whom this stuff is
 not a greenfield.<br>
<br>
The thing is: no matter what we do here, it's not going to look 100% <br>
like the stuff that came before it. All we can do is look at the <br>
products we know about, look at the existing published semantics for <br>
protocol elements, and make educated guesses about what is likely to <br>
cause the fewest problems. We *know* that using the same port numbers <br>
for multiple m-lines causes bad behavior in more than a few endpoints, <br>
so that's out. On the other hand, the semantics of port=3D0 are pretty <br>
well defined, elicit the exact response we want for remote endpoints, <br>
and avoid egregiously large port allocations during call setup (which is <b=
r>
an important goal).<br>
<br>
I think that's a lot more solid set of reasons than the opposing <br>
rationale you proffer of &quot;there may or may not be products somewhere i=
n <br>
the world that don't really like this approach,&quot; not least of all <br>
because that statement is true of any approach we come up with.<br>
<br>
/a<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422A4FESESSMB209erics_--

From hadriel.kaplan@oracle.com  Thu Aug  8 14:40:42 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 5:35 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hadriel,
> I'm a little confused. I thought you wanted port zero in the Answer, =
but the discussion now seems to be around port zero in the Offer.
> Or, have I missed something?

Sorry for the confusion - the thing I'm arguing against is the Offer =
having port=3D0, while the Answer having port=3DN.

I'm fine with the Offer having port=3D0 and the Answer having port=3D0 =
(i.e., sticking to RFC 3264).

[yeeesh this topic is hard to discuss in email!]

-hadriel


> Sent from Windows using TouchDown (www.nitrodesk.com)
>=20
> -----Original Message-----
> From: Adam Roach [adam@nostrum.com]
> To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic =
[mmusic@ietf.org]
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-only m- lines?
> On 8/8/13 15:51, Hadriel Kaplan wrote:
> > BTW, I'm not speaking about my employer's SBCs in particular - I =
actually have no idea what we'll do with an UP bundle-only thing lying =
about a port=3D0.  We could handle it all correctly, or tear the call =
down later, or call WOPR and tell it to play Global Thermonuclear War.  =
If something's broken in our implementation, we'll fix it... eventually. =
 I'm more worried about other systems, from vendors who don't attend the =
IETF; and I'm worried about the people who have to run SIP networks, for =
whom this stuff is not a greenfield.
>=20
> The thing is: no matter what we do here, it's not going to look 100%=20=

> like the stuff that came before it. All we can do is look at the=20
> products we know about, look at the existing published semantics for=20=

> protocol elements, and make educated guesses about what is likely to=20=

> cause the fewest problems. We *know* that using the same port numbers=20=

> for multiple m-lines causes bad behavior in more than a few endpoints,=20=

> so that's out. On the other hand, the semantics of port=3D0 are pretty=20=

> well defined, elicit the exact response we want for remote endpoints,=20=

> and avoid egregiously large port allocations during call setup (which =
is=20
> an important goal).
>=20
> I think that's a lot more solid set of reasons than the opposing=20
> rationale you proffer of "there may or may not be products somewhere =
in=20
> the world that don't really like this approach," not least of all=20
> because that statement is true of any approach we come up with.
>=20
> /a


From christer.holmberg@ericsson.com  Thu Aug  8 14:52:26 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Date: Thu, 8 Aug 2013 21:51:32 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se> <1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com> <5203EB64.70605@nostrum.com> <65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com> <5203F37A.8030204@nostrum.com> <33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com> <5203F4D8.9040904@nostrum.com> <9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com>, <5204075B.1080108@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422A4F@ESESSMB209.ericsson.se>, <FABBA52E-66D3-42C9-B25C-956A248C0139@oracle.com>
In-Reply-To: <FABBA52E-66D3-42C9-B25C-956A248C0139@oracle.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Maybe we should create persistent chat rooms to discuss specific topics, or=
 something...

Anyway, so you are saying that a port-zero Offer + a non-port zero Answer w=
ill cause problems, and/or will require an update of 3264?

Regarding the possible problem, I believe (I haven't checked how the produc=
ts my company make behave, though) that intermediaries simply would not car=
e about the port value in the Answer, as it was zero in the Offer. They wou=
ld consider the associated media disabled/removed - which is why I believe =
a second offer is needed also for bundle-only...

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: Adam Roach [adam@nostrum.com]; mmusic [mmusic@ietf.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

On Aug 8, 2013, at 5:35 PM, Christer Holmberg <christer.holmberg@ericsson.c=
om> wrote:

> Hadriel,
> I'm a little confused. I thought you wanted port zero in the Answer, but =
the discussion now seems to be around port zero in the Offer.
> Or, have I missed something?

Sorry for the confusion - the thing I'm arguing against is the Offer having=
 port=3D0, while the Answer having port=3DN.

I'm fine with the Offer having port=3D0 and the Answer having port=3D0 (i.e=
., sticking to RFC 3264).

[yeeesh this topic is hard to discuss in email!]

-hadriel


> Sent from Windows using TouchDown (www.nitrodesk.com<http://www.nitrodesk=
.com>)
>
> -----Original Message-----
> From: Adam Roach [adam@nostrum.com]
> To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic@ie=
tf.org]
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-onl=
y m- lines?
> On 8/8/13 15:51, Hadriel Kaplan wrote:
> > BTW, I'm not speaking about my employer's SBCs in particular - I actual=
ly have no idea what we'll do with an UP bundle-only thing lying about a po=
rt=3D0.  We could handle it all correctly, or tear the call down later, or =
call WOPR and tell it to play Global Thermonuclear War.  If something's bro=
ken in our implementation, we'll fix it... eventually.  I'm more worried ab=
out other systems, from vendors who don't attend the IETF; and I'm worried =
about the people who have to run SIP networks, for whom this stuff is not a=
 greenfield.
>
> The thing is: no matter what we do here, it's not going to look 100%
> like the stuff that came before it. All we can do is look at the
> products we know about, look at the existing published semantics for
> protocol elements, and make educated guesses about what is likely to
> cause the fewest problems. We *know* that using the same port numbers
> for multiple m-lines causes bad behavior in more than a few endpoints,
> so that's out. On the other hand, the semantics of port=3D0 are pretty
> well defined, elicit the exact response we want for remote endpoints,
> and avoid egregiously large port allocations during call setup (which is
> an important goal).
>
> I think that's a lot more solid set of reasons than the opposing
> rationale you proffer of "there may or may not be products somewhere in
> the world that don't really like this approach," not least of all
> because that statement is true of any approach we come up with.
>
> /a


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1">
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ding-left: 4pt; border-left: #800000 2px solid; } --></style>
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<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div><br>
Maybe we should create persistent chat rooms to discuss specific topics,&nb=
sp;or something...</div>
<div>&nbsp;</div>
<div>Anyway, so you are saying that a port-zero Offer &#43; a non-port zero=
 Answer will cause problems, and/or will require an update of 3264?</div>
<div>&nbsp;</div>
<div>Regarding the possible problem, I believe (I haven't checked how the p=
roducts my company make behave, though) that intermediaries simply would no=
t care about the port value in the Answer, as it was zero in the Offer. The=
y would consider the associated
 media disabled/removed - which is why I believe a second offer is needed a=
lso for bundle-only...</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> Adam Roach [adam@nostrum.com]; mmusic [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 8, 2013, at 5:35 PM, Christer Holmberg &lt;christer.holmberg@ericsso=
n.com&gt; wrote:<br>
<br>
&gt; Hadriel,<br>
&gt; I'm a little confused. I thought you wanted port zero in the Answer, b=
ut the discussion now seems to be around port zero in the Offer.<br>
&gt; Or, have I missed something?<br>
<br>
Sorry for the confusion - the thing I'm arguing against is the Offer having=
 port=3D0, while the Answer having port=3DN.<br>
<br>
I'm fine with the Offer having port=3D0 and the Answer having port=3D0 (i.e=
., sticking to RFC 3264).<br>
<br>
[yeeesh this topic is hard to discuss in email!]<br>
<br>
-hadriel<br>
<br>
<br>
&gt; Sent from Windows using TouchDown (<a href=3D"http://www.nitrodesk.com=
">www.nitrodesk.com</a>)<br>
&gt; <br>
&gt; -----Original Message-----<br>
&gt; From: Adam Roach [adam@nostrum.com]<br>
&gt; To: Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
&gt; CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic=
@ietf.org]<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-=
only m- lines?<br>
&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt; &gt; BTW, I'm not speaking about my employer's SBCs in particular - I =
actually have no idea what we'll do with an UP bundle-only thing lying abou=
t a port=3D0.&nbsp; We could handle it all correctly, or tear the call down=
 later, or call WOPR and tell it to play Global
 Thermonuclear War.&nbsp; If something's broken in our implementation, we'l=
l fix it... eventually.&nbsp; I'm more worried about other systems, from ve=
ndors who don't attend the IETF; and I'm worried about the people who have =
to run SIP networks, for whom this stuff is
 not a greenfield.<br>
&gt; <br>
&gt; The thing is: no matter what we do here, it's not going to look 100% <=
br>
&gt; like the stuff that came before it. All we can do is look at the <br>
&gt; products we know about, look at the existing published semantics for <=
br>
&gt; protocol elements, and make educated guesses about what is likely to <=
br>
&gt; cause the fewest problems. We *know* that using the same port numbers =
<br>
&gt; for multiple m-lines causes bad behavior in more than a few endpoints,=
 <br>
&gt; so that's out. On the other hand, the semantics of port=3D0 are pretty=
 <br>
&gt; well defined, elicit the exact response we want for remote endpoints, =
<br>
&gt; and avoid egregiously large port allocations during call setup (which =
is <br>
&gt; an important goal).<br>
&gt; <br>
&gt; I think that's a lot more solid set of reasons than the opposing <br>
&gt; rationale you proffer of &quot;there may or may not be products somewh=
ere in <br>
&gt; the world that don't really like this approach,&quot; not least of all=
 <br>
&gt; because that statement is true of any approach we come up with.<br>
&gt; <br>
&gt; /a<br>
<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422B36ESESSMB209erics_--

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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com> <5203D7E7.4070207@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se> <1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com> <5203EB64.70605@nostrum.com> <65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com> <5203F37A.8030204@nostrum.com> <33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com> <5203F4D8.9040904@nostrum.com> <9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com> <5204075B.1080108@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422A4F@ESESSMB209.ericsson.se> <FABBA52E-66D3-42C9-B25C-956A248C0139@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C422B36@ESESSMB209.ericsson.se>
Date: Thu, 8 Aug 2013 16:59:01 -0500
Message-ID: <CAHBDyN747TSDakeqtocn7F-4qxszCBo4rkUTFBcWb4WcBBrWXA@mail.gmail.com>
From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Content-Type: multipart/alternative; boundary=047d7bd766de06502304e376c4c8
Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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There is already a chatroom configured for each of the WGs:
http://www.ietf.org/jabber/index.html
It's the same one you would join during a meeting.  You can access it via
the tools page:
http://tools.ietf.org/wg/mmusic/

Logs are permanent, etc.  And, of course subject to IETF IPR rules ;)

Mary.


On Thu, Aug 8, 2013 at 4:51 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>
> Maybe we should create persistent chat rooms to discuss specific
> topics, or something...
>
> Anyway, so you are saying that a port-zero Offer + a non-port zero Answer
> will cause problems, and/or will require an update of 3264?
>
> Regarding the possible problem, I believe (I haven't checked how the
> products my company make behave, though) that intermediaries simply would
> not care about the port value in the Answer, as it was zero in the Offer.
> They would consider the associated media disabled/removed - which is why I
> believe a second offer is needed also for bundle-only...
>
> Regards,
>
> Christer
>
>
> Sent from *Windows* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> *To:* Christer Holmberg [christer.holmberg@ericsson.com]
> *CC:* Adam Roach [adam@nostrum.com]; mmusic [mmusic@ietf.org]
> *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> bundle-only m- lines?
>
> On Aug 8, 2013, at 5:35 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> > Hadriel,
> > I'm a little confused. I thought you wanted port zero in the Answer, but
> the discussion now seems to be around port zero in the Offer.
> > Or, have I missed something?
>
> Sorry for the confusion - the thing I'm arguing against is the Offer
> having port=0, while the Answer having port=N.
>
> I'm fine with the Offer having port=0 and the Answer having port=0 (i.e.,
> sticking to RFC 3264).
>
> [yeeesh this topic is hard to discuss in email!]
>
> -hadriel
>
>
> > Sent from Windows using TouchDown (www.nitrodesk.com)
> >
> > -----Original Message-----
> > From: Adam Roach [adam@nostrum.com]
> > To: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> > CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [
> mmusic@ietf.org]
> > Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> bundle-only m- lines?
> > On 8/8/13 15:51, Hadriel Kaplan wrote:
> > > BTW, I'm not speaking about my employer's SBCs in particular - I
> actually have no idea what we'll do with an UP bundle-only thing lying
> about a port=0.  We could handle it all correctly, or tear the call down
> later, or call WOPR and tell it to play Global Thermonuclear War.  If
> something's broken in our implementation, we'll fix it... eventually.  I'm
> more worried about other systems, from vendors who don't attend the IETF;
> and I'm worried about the people who have to run SIP networks, for whom
> this stuff is not a greenfield.
> >
> > The thing is: no matter what we do here, it's not going to look 100%
> > like the stuff that came before it. All we can do is look at the
> > products we know about, look at the existing published semantics for
> > protocol elements, and make educated guesses about what is likely to
> > cause the fewest problems. We *know* that using the same port numbers
> > for multiple m-lines causes bad behavior in more than a few endpoints,
> > so that's out. On the other hand, the semantics of port=0 are pretty
> > well defined, elicit the exact response we want for remote endpoints,
> > and avoid egregiously large port allocations during call setup (which is
> > an important goal).
> >
> > I think that's a lot more solid set of reasons than the opposing
> > rationale you proffer of "there may or may not be products somewhere in
> > the world that don't really like this approach," not least of all
> > because that statement is true of any approach we come up with.
> >
> > /a
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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<div dir=3D"ltr">There is already a chatroom configured for each of the WGs=
:<div><a href=3D"http://www.ietf.org/jabber/index.html">http://www.ietf.org=
/jabber/index.html</a><br></div><div>It&#39;s the same one you would join d=
uring a meeting. =A0You can access it via the tools page:=A0</div>
<div><a href=3D"http://tools.ietf.org/wg/mmusic/">http://tools.ietf.org/wg/=
mmusic/</a></div><div><br></div><div>Logs are permanent, etc. =A0And, of co=
urse subject to IETF IPR rules ;)</div><div><br></div><div><div>Mary.</div>
</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n Thu, Aug 8, 2013 at 4:51 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.ho=
lmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div>
<div style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans-serif=
">
<span style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans-seri=
f">
<div><br>
Maybe we should create persistent chat rooms to discuss specific topics,=A0=
or something...</div>
<div>=A0</div>
<div>Anyway, so you are saying that a port-zero Offer + a non-port zero Ans=
wer will cause problems, and/or will require an update of 3264?</div>
<div>=A0</div>
<div>Regarding the possible problem, I believe (I haven&#39;t checked how t=
he products my company make behave, though) that intermediaries simply woul=
d not care about the port value in the Answer, as it was zero in the Offer.=
 They would consider the associated
 media disabled/removed - which is why I believe a second offer is needed a=
lso for bundle-only...</div>
<div>=A0</div>
<div>Regards,</div><span class=3D"HOEnZb"><font color=3D"#888888">
<div>=A0</div>
<div>Christer</div></font></span><div class=3D"im">
<div>=A0</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (<a href=3D"http://www.nitrodesk.com" target=3D"_blank">www.nitrodesk.com<=
/a>)</div>
</div></span>
<div></div>
<br>
<span style><div class=3D"im">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [<a href=3D"mailto:hadriel.kaplan@oracle.com" t=
arget=3D"_blank">hadriel.kaplan@oracle.com</a>]<br>
<b>To:</b> Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.=
com" target=3D"_blank">christer.holmberg@ericsson.com</a>]<br>
</div><div><div class=3D"h5"><b>CC:</b> Adam Roach [<a href=3D"mailto:adam@=
nostrum.com" target=3D"_blank">adam@nostrum.com</a>]; mmusic [<a href=3D"ma=
ilto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</div></div></span></div><div><div class=3D"h5">
<font><span style=3D"font-size:10pt">
<div><br>
On Aug 8, 2013, at 5:35 PM, Christer Holmberg &lt;<a href=3D"mailto:christe=
r.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</=
a>&gt; wrote:<br>
<br>
&gt; Hadriel,<br>
&gt; I&#39;m a little confused. I thought you wanted port zero in the Answe=
r, but the discussion now seems to be around port zero in the Offer.<br>
&gt; Or, have I missed something?<br>
<br>
Sorry for the confusion - the thing I&#39;m arguing against is the Offer ha=
ving port=3D0, while the Answer having port=3DN.<br>
<br>
I&#39;m fine with the Offer having port=3D0 and the Answer having port=3D0 =
(i.e., sticking to RFC 3264).<br>
<br>
[yeeesh this topic is hard to discuss in email!]<br>
<br>
-hadriel<br>
<br>
<br>
&gt; Sent from Windows using TouchDown (<a href=3D"http://www.nitrodesk.com=
" target=3D"_blank">www.nitrodesk.com</a>)<br>
&gt; <br>
&gt; -----Original Message-----<br>
&gt; From: Adam Roach [<a href=3D"mailto:adam@nostrum.com" target=3D"_blank=
">adam@nostrum.com</a>]<br>
&gt; To: Hadriel Kaplan [<a href=3D"mailto:hadriel.kaplan@oracle.com" targe=
t=3D"_blank">hadriel.kaplan@oracle.com</a>]<br>
&gt; CC: Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.co=
m" target=3D"_blank">christer.holmberg@ericsson.com</a>]; mmusic [<a href=
=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>]<br>
&gt; Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-=
only m- lines?<br>
&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt; &gt; BTW, I&#39;m not speaking about my employer&#39;s SBCs in particu=
lar - I actually have no idea what we&#39;ll do with an UP bundle-only thin=
g lying about a port=3D0.=A0 We could handle it all correctly, or tear the =
call down later, or call WOPR and tell it to play Global
 Thermonuclear War.=A0 If something&#39;s broken in our implementation, we&=
#39;ll fix it... eventually.=A0 I&#39;m more worried about other systems, f=
rom vendors who don&#39;t attend the IETF; and I&#39;m worried about the pe=
ople who have to run SIP networks, for whom this stuff is
 not a greenfield.<br>
&gt; <br>
&gt; The thing is: no matter what we do here, it&#39;s not going to look 10=
0% <br>
&gt; like the stuff that came before it. All we can do is look at the <br>
&gt; products we know about, look at the existing published semantics for <=
br>
&gt; protocol elements, and make educated guesses about what is likely to <=
br>
&gt; cause the fewest problems. We *know* that using the same port numbers =
<br>
&gt; for multiple m-lines causes bad behavior in more than a few endpoints,=
 <br>
&gt; so that&#39;s out. On the other hand, the semantics of port=3D0 are pr=
etty <br>
&gt; well defined, elicit the exact response we want for remote endpoints, =
<br>
&gt; and avoid egregiously large port allocations during call setup (which =
is <br>
&gt; an important goal).<br>
&gt; <br>
&gt; I think that&#39;s a lot more solid set of reasons than the opposing <=
br>
&gt; rationale you proffer of &quot;there may or may not be products somewh=
ere in <br>
&gt; the world that don&#39;t really like this approach,&quot; not least of=
 all <br>
&gt; because that statement is true of any approach we come up with.<br>
&gt; <br>
&gt; /a<br>
<br>
</div>
</span></font>
</div></div></div>

<br>_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br></blockquote></div><br></div>

--047d7bd766de06502304e376c4c8--

From hadriel.kaplan@oracle.com  Thu Aug  8 15:08:21 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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To: Adam Roach <adam@nostrum.com>
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Cc: mmusic <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 15:51, Hadriel Kaplan wrote:
>> BTW, I'm not speaking about my employer's SBCs in particular - I =
actually have no idea what we'll do with an UP bundle-only thing lying =
about a port=3D0.  We could handle it all correctly, or tear the call =
down later, or call WOPR and tell it to play Global Thermonuclear War.  =
If something's broken in our implementation, we'll fix it... eventually. =
 I'm more worried about other systems, from vendors who don't attend the =
IETF; and I'm worried about the people who have to run SIP networks, for =
whom this stuff is not a greenfield.
>=20
> The thing is: no matter what we do here, it's not going to look 100% =
like the stuff that came before it. All we can do is look at the =
products we know about, look at the existing published semantics for =
protocol elements, and make educated guesses about what is likely to =
cause the fewest problems. We *know* that using the same port numbers =
for multiple m-lines causes bad behavior in more than a few endpoints, =
so that's out. On the other hand, the semantics of port=3D0 are pretty =
well defined, elicit the exact response we want for remote endpoints, =
and avoid egregiously large port allocations during call setup (which is =
an important goal).

The semantics of using port=3D0 are well defined, as you say - those =
semantics are that you MUST NOT use the media.  You are changing those =
semantics, if the far-end supports BUNDLE.  You feel it's safe to change =
those semantics, because you feel that SDP is only being processed by =
the local and remote endpoints.  I'm saying that's a really bad =
assumption, because I know that in fact it is not the case.



> I think that's a lot more solid set of reasons than the opposing =
rationale you proffer of "there may or may not be products somewhere in =
the world that don't really like this approach," not least of all =
because that statement is true of any approach we come up with.

Do you honestly believe that using a port number of 0 in an offer, while =
the answer uses a port number of N, is equally impacting on =
interoperability as using an attribute for capability discovery and then =
a second O/A round of the real numbers?  C'mon man.  You've seen the =
stuff that's out there.  The SDP Offer and Answer's m-line port number =
is used by all kinds of whacky equipment, to determine the RTP/RTCP flow =
information.  It's used by passive monitoring systems, NATs, CGNs, =
Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and god knows what =
else.  Even just in my own employer, we've got five different types of =
equipment that look at the the m-line port numbers, only one of which is =
our SBCs (and we've got multiple flavors of them too, fwiw).

-hadriel


From matthew@matthew.at  Thu Aug  8 15:09:47 2013
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On 8/5/2013 12:42 PM, Matthew Kaufman wrote:
> On 8/5/2013 9:59 AM, Christer Holmberg wrote:
>> Hi,
>> While cleaning up my MMUSIC notes today, I was listening to the audio 
>> recording, in order to get a clarification regarding what "scope is 
>> WEBRTC" really means.  However, unless I missed something, I didn't 
>> get a clear answer.
>> Does it only cover the browser API (JSEP) or does it also cover 
>> something that can be sent out on the wire (as long as it is not SIP) 
>> by a browser?
>>
>
> And what if it passes through a gateway that doesn't inspect the SDP 
> and blindly passes it to a SIP device. What happens then?
>
> RTCWEB used SDP so it would be compatible with SIP legacy devices, so 
> certainly this will work... or something about all this has gone off 
> the rails.
>
> Matthew Kaufman


There's been some great discussion in this thread, but nobody has 
answered my question or provided feedback on my interpretation of what 
might has happened. Anyone?

Matthew Kaufman

--------------080904020604060504040707
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    <div class="moz-cite-prefix">On 8/5/2013 12:42 PM, Matthew Kaufman
      wrote:<br>
    </div>
    <blockquote cite="mid:5200000A.40102@matthew.at" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 8/5/2013 9:59 AM, Christer
        Holmberg wrote:<br>
      </div>
      <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41DB40@ESESSMB209.ericsson.se"
        type="cite">
        <meta http-equiv="Content-Type" content="text/html;
          charset=ISO-8859-1">
        <div><span style="color: black; font-family: Calibri, Arial,
            Helvetica, sans-serif; font-size: 11pt;">Hi,</span></div>
        <div>&nbsp;</div>
        <div><font face="Calibri">While cleaning up my MMUSIC notes
            today, I was listening to the audio recording, in order to
            get a clarification regarding what "scope is WEBRTC" really
            means.&nbsp; However, unless I missed something, I didn't get a
            clear answer.</font></div>
        <div>&nbsp;</div>
        <div><font face="Calibri">Does it only cover the browser API
            (JSEP)&nbsp;or does it also cover something that can be sent out
            on the wire (as long as it is not SIP) by a browser?</font></div>
        &nbsp;
        <div><br>
        </div>
      </blockquote>
      <br>
      And what if it passes through a gateway that doesn't inspect the
      SDP and blindly passes it to a SIP device. What happens then?<br>
      <br>
      RTCWEB used SDP so it would be compatible with SIP legacy devices,
      so certainly this will work... or something about all this has
      gone off the rails.<br>
      <br>
      Matthew Kaufman<br>
    </blockquote>
    <br>
    <br>
    There's been some great discussion in this thread, but nobody has
    answered my question or provided feedback on my interpretation of
    what might has happened. Anyone?<br>
    <br>
    Matthew Kaufman<br>
  </body>
</html>

--------------080904020604060504040707--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

An alternative way, using BUNDLE as currently specified, is the add the bun=
dle-only m- lines in a subsequent Offer, once BUNDLE usage has been negotia=
ted. Then one does not need port zero. Of course, that would a "third" Offe=
r (technically it could be done already in the "second" Offer, but BUNDLE r=
ecommends against that).

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
To: Adam Roach [adam@nostrum.com]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmusic@ietf=
.org]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?

On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:

> On 8/8/13 15:51, Hadriel Kaplan wrote:
>> BTW, I'm not speaking about my employer's SBCs in particular - I actuall=
y have no idea what we'll do with an UP bundle-only thing lying about a por=
t=3D0.  We could handle it all correctly, or tear the call down later, or c=
all WOPR and tell it to play Global Thermonuclear War.  If something's brok=
en in our implementation, we'll fix it... eventually.  I'm more worried abo=
ut other systems, from vendors who don't attend the IETF; and I'm worried a=
bout the people who have to run SIP networks, for whom this stuff is not a =
greenfield.
>
> The thing is: no matter what we do here, it's not going to look 100% like=
 the stuff that came before it. All we can do is look at the products we kn=
ow about, look at the existing published semantics for protocol elements, a=
nd make educated guesses about what is likely to cause the fewest problems.=
 We *know* that using the same port numbers for multiple m-lines causes bad=
 behavior in more than a few endpoints, so that's out. On the other hand, t=
he semantics of port=3D0 are pretty well defined, elicit the exact response=
 we want for remote endpoints, and avoid egregiously large port allocations=
 during call setup (which is an important goal).

The semantics of using port=3D0 are well defined, as you say - those semant=
ics are that you MUST NOT use the media.  You are changing those semantics,=
 if the far-end supports BUNDLE.  You feel it's safe to change those semant=
ics, because you feel that SDP is only being processed by the local and rem=
ote endpoints.  I'm saying that's a really bad assumption, because I know t=
hat in fact it is not the case.



> I think that's a lot more solid set of reasons than the opposing rational=
e you proffer of "there may or may not be products somewhere in the world t=
hat don't really like this approach," not least of all because that stateme=
nt is true of any approach we come up with.

Do you honestly believe that using a port number of 0 in an offer, while th=
e answer uses a port number of N, is equally impacting on interoperability =
as using an attribute for capability discovery and then a second O/A round =
of the real numbers?  C'mon man.  You've seen the stuff that's out there.  =
The SDP Offer and Answer's m-line port number is used by all kinds of whack=
y equipment, to determine the RTP/RTCP flow information.  It's used by pass=
ive monitoring systems, NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs=
, App servers, and god knows what else.  Even just in my own employer, we'v=
e got five different types of equipment that look at the the m-line port nu=
mbers, only one of which is our SBCs (and we've got multiple flavors of the=
m too, fwiw).

-hadriel


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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">An alternative way, using BUNDLE as currently s=
pecified, is the add the bundle-only m- lines in a subsequent Offer, once B=
UNDLE usage has been negotiated. Then one does not need port zero. Of cours=
e, that&nbsp;would a &quot;third&quot; Offer (technically
 it could be done already in the &quot;second&quot; Offer, but BUNDLE recom=
mends against that).</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [hadriel.kaplan@oracle.com]<br>
<b>To:</b> Adam Roach [adam@nostrum.com]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [mmus=
ic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Aug 8, 2013, at 5:02 PM, Adam Roach &lt;adam@nostrum.com&gt; wrote:<br>
<br>
&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt;&gt; BTW, I'm not speaking about my employer's SBCs in particular - I a=
ctually have no idea what we'll do with an UP bundle-only thing lying about=
 a port=3D0.&nbsp; We could handle it all correctly, or tear the call down =
later, or call WOPR and tell it to play Global
 Thermonuclear War.&nbsp; If something's broken in our implementation, we'l=
l fix it... eventually.&nbsp; I'm more worried about other systems, from ve=
ndors who don't attend the IETF; and I'm worried about the people who have =
to run SIP networks, for whom this stuff is
 not a greenfield.<br>
&gt; <br>
&gt; The thing is: no matter what we do here, it's not going to look 100% l=
ike the stuff that came before it. All we can do is look at the products we=
 know about, look at the existing published semantics for protocol elements=
, and make educated guesses about what
 is likely to cause the fewest problems. We *know* that using the same port=
 numbers for multiple m-lines causes bad behavior in more than a few endpoi=
nts, so that's out. On the other hand, the semantics of port=3D0 are pretty=
 well defined, elicit the exact response
 we want for remote endpoints, and avoid egregiously large port allocations=
 during call setup (which is an important goal).<br>
<br>
The semantics of using port=3D0 are well defined, as you say - those semant=
ics are that you MUST NOT use the media.&nbsp; You are changing those seman=
tics, if the far-end supports BUNDLE.&nbsp; You feel it's safe to change th=
ose semantics, because you feel that SDP is
 only being processed by the local and remote endpoints.&nbsp; I'm saying t=
hat's a really bad assumption, because I know that in fact it is not the ca=
se.<br>
<br>
<br>
<br>
&gt; I think that's a lot more solid set of reasons than the opposing ratio=
nale you proffer of &quot;there may or may not be products somewhere in the=
 world that don't really like this approach,&quot; not least of all because=
 that statement is true of any approach we come
 up with.<br>
<br>
Do you honestly believe that using a port number of 0 in an offer, while th=
e answer uses a port number of N, is equally impacting on interoperability =
as using an attribute for capability discovery and then a second O/A round =
of the real numbers?&nbsp; C'mon man.&nbsp;
 You've seen the stuff that's out there.&nbsp; The SDP Offer and Answer's m=
-line port number is used by all kinds of whacky equipment, to determine th=
e RTP/RTCP flow information.&nbsp; It's used by passive monitoring systems,=
 NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs,
 SBCs, App servers, and god knows what else.&nbsp; Even just in my own empl=
oyer, we've got five different types of equipment that look at the the m-li=
ne port numbers, only one of which is our SBCs (and we've got multiple flav=
ors of them too, fwiw).<br>
<br>
-hadriel<br>
<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C422D7DESESSMB209erics_--

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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 9, 2013, at 12:38 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> An alternative way, using BUNDLE as currently specified, is the add =
the bundle-only m- lines in a subsequent Offer, once BUNDLE usage has =
been negotiated. Then one does not need port zero.

Yup, that makes a lot of sense.


> Of course, that would a "third" Offer (technically it could be done =
already in the "second" Offer, but BUNDLE recommends against that).

Why does BUNDLE recommend against it, such that it needs a third round?

-hadriel



From christer.holmberg@ericsson.com  Fri Aug  9 06:21:09 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

>> An alternative way, using BUNDLE as currently specified, is the add the =
bundle-only m- lines in a subsequent Offer, once BUNDLE usage has been nego=
tiated. Then one does not need port zero.
>
> Yup, that makes a lot of sense.
>
>> Of course, that would a "third" Offer (technically it could be done alre=
ady in the "second" Offer, but BUNDLE recommends against that).
>
> Why does BUNDLE recommend against it, such that it needs a third round?

I don't have the exact wording in front of me, but as the second offer is m=
eant mainly for "port synchronizing", one should not put lots of extra stuf=
f to it, which raises the chance for the offer being rejected.

But, of course, you don't have to reject the whole offer if you don't like =
the new m- lines, as you can reject them with port zero...

Regards,

Christer




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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 9, 2013, at 9:20 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> An alternative way, using BUNDLE as currently specified, is the add =
the bundle-only m- lines in a subsequent Offer, once BUNDLE usage has =
been negotiated. Then one does not need port zero.
>>=20
>> Yup, that makes a lot of sense.
>>=20
>>> Of course, that would a "third" Offer (technically it could be done =
already in the "second" Offer, but BUNDLE recommends against that).
>>=20
>> Why does BUNDLE recommend against it, such that it needs a third =
round?
>=20
> I don't have the exact wording in front of me, but as the second offer =
is meant mainly for "port synchronizing", one should not put lots of =
extra stuff to it, which raises the chance for the offer being rejected.
>=20
> But, of course, you don't have to reject the whole offer if you don't =
like the new m- lines, as you can reject them with port zero=85

Yup.  And the advantage with this whole idea is there is no =
"a=3Dbundle-only" attribute needed whatsoever, no change in port=3D0 =
behavior, etc.  It just goes back to the semantics we have today, which =
is arguably the safest thing to do.  I like your idea a lot.

The concept of "change as little as possible in the second offer" makes =
some pragmatic sense, but a UAC could be more aggressive/optimistic by =
having the second offer do it all, and if that Offer gets rejected =
entirely then do it piecemeal.  In other words, have the initial Offer =
advertise the current bundle stuff (without bundle-only m-lines), and if =
the Answer says bundle, then a second offer sends the original m-lines =
with corrected port numbers as well as new bundled m-lines - the ones it =
was only willing to use if the other end supported BUNDLE.  And if that =
Offer fails completely, then send an Offer of only the original bundle =
lines with corrected port numbers.

Or we could even just leave that type of decision up to local policy.  =
We don't usually say what to do if a whole Offer gets rejected.  Of =
course we should remind the implementor that it might happen, and how =
they could handle it, but it doesn't have to be normative at all.

-hadriel


From hadriel.kaplan@oracle.com  Fri Aug  9 07:24:24 2013
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On Aug 8, 2013, at 6:08 PM, Matthew Kaufman <matthew@matthew.at> wrote:

>> And what if it passes through a gateway that doesn't inspect the SDP =
and blindly passes it to a SIP device. What happens then?
>>=20
>> RTCWEB used SDP so it would be compatible with SIP legacy devices, so =
certainly this will work... or something about all this has gone off the =
rails.

Are we talking about theory, or reality?

I've been told that one vendor thinks they'll be able to pass the SDP =
from WebRTC-land to SIP-land unchanged.  My belief is that this would =
require all the SIP devices in their SIP-land to be manufactured by =
them, or at least known to support their particular features.  But for =
their general/common customer environment, that's likely to be the case =
so it's not an issue for them. (unless of course the requests get =
forwarded outside their domain, but for that they sell SBCs to normalize =
it all)

For most vendors and deployment scenarios, however, I think blindly =
passing SDP between WebRTC and SIP is a pipe dream.  At the very least, =
there'll be a lot of media-plane interworking for some WbeRTC-SIP =
interworking gateway to perform, so it can't be SDP agnostic/blind.  =
Things like ICE termination, SRTP interworking or termination, audio =
transcoding, and perhaps data-channel interworking for certain =
applications.  It might also want to do BUNDLE rtp-mux termination, =
although the complexities of that in the media-plane likely outweigh the =
benefits.

As far as I can recall, however, the main benefit of WebRTC using SDP =
was NOT the ability to interwork to SIP easily.  That was more of a =
frosting-on-the-cake type thing.  The main benefit its proponents =
espoused was that it would let simple WebRTC use-cases be done easily, =
with few lines of JS code.  Advanced use-cases would be possible, but =
not easy.  I.e., they would require SDP munging in JS, or worse.

Or at least that is my impression of what the main argument for SDP was. =
 Of course this was like two years ago, so my memory could be faulty.  =
Maybe it was not the main argument, but just the one that I personally =
felt was actually valid and laudable. :)

-hadriel


From christer.holmberg@ericsson.com  Fri Aug  9 08:10:01 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

>>>> An alternative way, using BUNDLE as currently specified, is the add th=
e bundle-only m- lines in a subsequent Offer, once BUNDLE usage has been ne=
gotiated. Then one does not need port zero.
>>>=20
>>> Yup, that makes a lot of sense.
>>>=20
>>>> Of course, that would a "third" Offer (technically it could be done al=
ready in the "second" Offer, but BUNDLE recommends against that).
>>>=20
>>> Why does BUNDLE recommend against it, such that it needs a third round?
>>=20
>> I don't have the exact wording in front of me, but as the second offer i=
s meant mainly for "port synchronizing", one should not put lots of extra s=
tuff to it, which raises the chance for the offer being rejected.
>>=20
>> But, of course, you don't have to reject the whole offer if you don't li=
ke the new m- lines, as you can reject them with port zero...
>
> Yup.  And the advantage with this whole idea is there is no "a=3Dbundle-o=
nly" attribute needed whatsoever, no change in port=3D0 behavior, etc.  It =
just goes back to the semantics we have today, which is arguably the safest=
 thing to do.  I like your idea a lot.

But, that doesn't mean you will put my picture on one of your future meetin=
g slides, does it? ;)

> The concept of "change as little as possible in the second offer" makes s=
ome pragmatic sense, but a UAC could be more aggressive/optimistic by havin=
g the second offer do it all, and if that Offer gets rejected entirely then=
 do it piecemeal.  In other=20
> words, have the initial Offer advertise the current bundle > stuff (witho=
ut bundle-only m-lines), and if the Answer says bundle, then a second offer=
 sends the original m-lines with corrected port numbers as well as new bund=
led m-lines - the ones it=20
> was only willing to use if the other end supported BUNDLE.  And if that O=
ffer fails completely, then send an Offer of only the original bundle lines=
 with corrected port numbers.

Yes.

> Or we could even just leave that type of decision up to local policy.  We=
 don't usually say what to do if a whole Offer gets rejected.  Of course we=
 should remind the implementor that it might happen, and how they could han=
dle it, but it doesn't have to be normative at all.

Currently, the only thing the standard (RFC 3264) says is that, if the Offe=
r gets rejected, the previous SDP state apply. However, in the case of BUND=
LE, as the second offer is very much "bound" to the first one, since it com=
pletes the BUNDLE negotiation, we may need to say something (the draft may =
already say something) on how to behave.

Regards,

Christer

From ekr@rtfm.com  Fri Aug  9 08:35:18 2013
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Uh, just say no to this.

We can't design a system under the constraint that it doesn't annoy
any middlebox in the world.

-Ekr



On Thu, Aug 8, 2013 at 9:38 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
> An alternative way, using BUNDLE as currently specified, is the add the
> bundle-only m- lines in a subsequent Offer, once BUNDLE usage has been
> negotiated. Then one does not need port zero. Of course, that would a
> "third" Offer (technically it could be done already in the "second" Offer,
> but BUNDLE recommends against that).
>
> Regards,
>
> Christer
>
>
>
> Sent from *Windows* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> *To:* Adam Roach [adam@nostrum.com]
> *CC:* Christer Holmberg [christer.holmberg@ericsson.com]; mmusic [
> mmusic@ietf.org]
> *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> bundle-only m- lines?
>
> On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
>
> > On 8/8/13 15:51, Hadriel Kaplan wrote:
> >> BTW, I'm not speaking about my employer's SBCs in particular - I
> actually have no idea what we'll do with an UP bundle-only thing lying
> about a port=0.  We could handle it all correctly, or tear the call down
> later, or call WOPR and tell it to play Global Thermonuclear War.  If
> something's broken in our implementation, we'll fix it... eventually.  I'm
> more worried about other systems, from vendors who don't attend the IETF;
> and I'm worried about the people who have to run SIP networks, for whom
> this stuff is not a greenfield.
> >
> > The thing is: no matter what we do here, it's not going to look 100%
> like the stuff that came before it. All we can do is look at the products
> we know about, look at the existing published semantics for protocol
> elements, and make educated guesses about what is likely to cause the
> fewest problems. We *know* that using the same port numbers for multiple
> m-lines causes bad behavior in more than a few endpoints, so that's out. On
> the other hand, the semantics of port=0 are pretty well defined, elicit the
> exact response we want for remote endpoints, and avoid egregiously large
> port allocations during call setup (which is an important goal).
>
> The semantics of using port=0 are well defined, as you say - those
> semantics are that you MUST NOT use the media.  You are changing those
> semantics, if the far-end supports BUNDLE.  You feel it's safe to change
> those semantics, because you feel that SDP is only being processed by the
> local and remote endpoints.  I'm saying that's a really bad assumption,
> because I know that in fact it is not the case.
>
>
>
> > I think that's a lot more solid set of reasons than the opposing
> rationale you proffer of "there may or may not be products somewhere in the
> world that don't really like this approach," not least of all because that
> statement is true of any approach we come up with.
>
> Do you honestly believe that using a port number of 0 in an offer, while
> the answer uses a port number of N, is equally impacting on
> interoperability as using an attribute for capability discovery and then a
> second O/A round of the real numbers?  C'mon man.  You've seen the stuff
> that's out there.  The SDP Offer and Answer's m-line port number is used by
> all kinds of whacky equipment, to determine the RTP/RTCP flow information.
> It's used by passive monitoring systems, NATs, CGNs, Firewalls, Routers,
> PDGs, x-CSCFs, SBCs, App servers, and god knows what else.  Even just in my
> own employer, we've got five different types of equipment that look at the
> the m-line port numbers, only one of which is our SBCs (and we've got
> multiple flavors of them too, fwiw).
>
> -hadriel
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

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<div dir=3D"ltr">Uh, just say no to this.<div><br></div><div><div>We can&#3=
9;t design a system under the constraint that it doesn&#39;t annoy</div><di=
v>any middlebox in the world.</div><div><br></div><div>-Ekr</div><div><br>


</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n Thu, Aug 8, 2013 at 9:38 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.ho=
lmberg@ericsson.com</a>&gt;</span> wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div>
<div style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans-serif=
">
<div><span style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans=
-serif">Hi,</span></div>
<div>=A0</div>
<div><font face=3D"Calibri">An alternative way, using BUNDLE as currently s=
pecified, is the add the bundle-only m- lines in a subsequent Offer, once B=
UNDLE usage has been negotiated. Then one does not need port zero. Of cours=
e, that=A0would a &quot;third&quot; Offer (technically
 it could be done already in the &quot;second&quot; Offer, but BUNDLE recom=
mends against that).</font></div><div>
<div><font face=3D"Calibri"></font>=A0</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>=A0</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans-seri=
f">
<div>=A0</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (<a href=3D"http://www.nitrodesk.com" target=3D"_blank">www.nitrodesk.com<=
/a>)</div>
</span>
<div></div>
<br>
</div><div><span>-----Original Message-----<br>
<b>From:</b> Hadriel Kaplan [<a href=3D"mailto:hadriel.kaplan@oracle.com" t=
arget=3D"_blank">hadriel.kaplan@oracle.com</a>]<br>
<b>To:</b> Adam Roach [<a href=3D"mailto:adam@nostrum.com" target=3D"_blank=
">adam@nostrum.com</a>]<br>
<b>CC:</b> Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.=
com" target=3D"_blank">christer.holmberg@ericsson.com</a>]; mmusic [<a href=
=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span></div></div><div><div>
<font><span style=3D"font-size:10pt">
<div><br>
On Aug 8, 2013, at 5:02 PM, Adam Roach &lt;<a href=3D"mailto:adam@nostrum.c=
om" target=3D"_blank">adam@nostrum.com</a>&gt; wrote:<br>
<br>
&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt;&gt; BTW, I&#39;m not speaking about my employer&#39;s SBCs in particul=
ar - I actually have no idea what we&#39;ll do with an UP bundle-only thing=
 lying about a port=3D0.=A0 We could handle it all correctly, or tear the c=
all down later, or call WOPR and tell it to play Global
 Thermonuclear War.=A0 If something&#39;s broken in our implementation, we&=
#39;ll fix it... eventually.=A0 I&#39;m more worried about other systems, f=
rom vendors who don&#39;t attend the IETF; and I&#39;m worried about the pe=
ople who have to run SIP networks, for whom this stuff is
 not a greenfield.<br>
&gt; <br>
&gt; The thing is: no matter what we do here, it&#39;s not going to look 10=
0% like the stuff that came before it. All we can do is look at the product=
s we know about, look at the existing published semantics for protocol elem=
ents, and make educated guesses about what
 is likely to cause the fewest problems. We *know* that using the same port=
 numbers for multiple m-lines causes bad behavior in more than a few endpoi=
nts, so that&#39;s out. On the other hand, the semantics of port=3D0 are pr=
etty well defined, elicit the exact response
 we want for remote endpoints, and avoid egregiously large port allocations=
 during call setup (which is an important goal).<br>
<br>
The semantics of using port=3D0 are well defined, as you say - those semant=
ics are that you MUST NOT use the media.=A0 You are changing those semantic=
s, if the far-end supports BUNDLE.=A0 You feel it&#39;s safe to change thos=
e semantics, because you feel that SDP is
 only being processed by the local and remote endpoints.=A0 I&#39;m saying =
that&#39;s a really bad assumption, because I know that in fact it is not t=
he case.<br>
<br>
<br>
<br>
&gt; I think that&#39;s a lot more solid set of reasons than the opposing r=
ationale you proffer of &quot;there may or may not be products somewhere in=
 the world that don&#39;t really like this approach,&quot; not least of all=
 because that statement is true of any approach we come
 up with.<br>
<br>
Do you honestly believe that using a port number of 0 in an offer, while th=
e answer uses a port number of N, is equally impacting on interoperability =
as using an attribute for capability discovery and then a second O/A round =
of the real numbers?=A0 C&#39;mon man.=A0
 You&#39;ve seen the stuff that&#39;s out there.=A0 The SDP Offer and Answe=
r&#39;s m-line port number is used by all kinds of whacky equipment, to det=
ermine the RTP/RTCP flow information.=A0 It&#39;s used by passive monitorin=
g systems, NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs,
 SBCs, App servers, and god knows what else.=A0 Even just in my own employe=
r, we&#39;ve got five different types of equipment that look at the the m-l=
ine port numbers, only one of which is our SBCs (and we&#39;ve got multiple=
 flavors of them too, fwiw).<br>



<br>
-hadriel<br>
<br>
</div>
</span></font>
</div></div></div>

<br>_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br></blockquote></div><br></div></div>

--001a11c3e4208a5ed504e3856e31--

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Hadriel Kaplan'" <hadriel.kaplan@oracle.com>, "'Adam Roach'" <adam@nostrum.com>
References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se>	<9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com>	<7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se>	<1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com>	<5203EB64.70605@nostrum.com>	<65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com>	<5203F37A.8030204@nostrum.com>	<33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com>	<5203F4D8.9040904@nostrum.com>	<9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com>	<5204075B.1080108@nostrum.com> <A2E26DB5-CF47-44DF-8779-D470280977B6@oracle.com>
In-Reply-To: <A2E26DB5-CF47-44DF-8779-D470280977B6@oracle.com>
Date: Fri, 9 Aug 2013 21:46:32 +0530
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Cc: 'mmusic' <mmusic@ietf.org>, 'Christer Holmberg' <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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I agree with Hadriel that offer with port 0 and answer with non-zero 
Port is harmful and leads to interop issues. This approach also
Violates Sec 8.2 of RFC 3264 requirement

" A stream that is offered with a port of zero MUST be marked with port
   zero in the answer."

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: Friday, August 09, 2013 3:37 AM
> To: Adam Roach
> Cc: mmusic; Christer Holmberg
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
> 
> 
> On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
> 
> > On 8/8/13 15:51, Hadriel Kaplan wrote:
> >> BTW, I'm not speaking about my employer's SBCs in particular - I
> actually have no idea what we'll do with an UP bundle-only thing lying
> about a port=0.  We could handle it all correctly, or tear the call
> down later, or call WOPR and tell it to play Global Thermonuclear War.
> If something's broken in our implementation, we'll fix it...
> eventually.  I'm more worried about other systems, from vendors who
> don't attend the IETF; and I'm worried about the people who have to run
> SIP networks, for whom this stuff is not a greenfield.
> >
> > The thing is: no matter what we do here, it's not going to look 100%
> like the stuff that came before it. All we can do is look at the
> products we know about, look at the existing published semantics for
> protocol elements, and make educated guesses about what is likely to
> cause the fewest problems. We *know* that using the same port numbers
> for multiple m-lines causes bad behavior in more than a few endpoints,
> so that's out. On the other hand, the semantics of port=0 are pretty
> well defined, elicit the exact response we want for remote endpoints,
> and avoid egregiously large port allocations during call setup (which
> is an important goal).
> 
> The semantics of using port=0 are well defined, as you say - those
> semantics are that you MUST NOT use the media.  You are changing those
> semantics, if the far-end supports BUNDLE.  You feel it's safe to
> change those semantics, because you feel that SDP is only being
> processed by the local and remote endpoints.  I'm saying that's a
> really bad assumption, because I know that in fact it is not the case.
> 
> 
> 
> > I think that's a lot more solid set of reasons than the opposing
> rationale you proffer of "there may or may not be products somewhere in
> the world that don't really like this approach," not least of all
> because that statement is true of any approach we come up with.
> 
> Do you honestly believe that using a port number of 0 in an offer,
> while the answer uses a port number of N, is equally impacting on
> interoperability as using an attribute for capability discovery and
> then a second O/A round of the real numbers?  C'mon man.  You've seen
> the stuff that's out there.  The SDP Offer and Answer's m-line port
> number is used by all kinds of whacky equipment, to determine the
> RTP/RTCP flow information.  It's used by passive monitoring systems,
> NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and
> god knows what else.  Even just in my own employer, we've got five
> different types of equipment that look at the the m-line port numbers,
> only one of which is our SBCs (and we've got multiple flavors of them
> too, fwiw).
> 
> -hadriel
> 
> _______________________________________________
> mmusic mailing list
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Wow! I ignore this thread for a day and it goes wild!
I get the advantage of looking at the long discussion and only replying 
selectively.

On 8/8/13 9:37 PM, Adam Roach wrote:
> On 8/8/13 14:11, Hadriel Kaplan wrote:
>> If I send a new SDP Offer 10 minutes later during the session, and I
>> set an existing m-line to port 0 in that Offer, does it mean I want to
>> disable the media, or restart bundling, or what?
>
> We already talked about that in Berlin: it means you want to disable the
> line. We're assigning special semantics to port=0 in the **FIRST** offer
> of a session.

IMO this is the wrong way to think of this special case.
Rather, it is special to the first offer that proposes a new bundle.
It need not be the first offer of the session. It may be that you have 
many offers in the session that don't use bundle at all. And then 
eventually you get into a state where you need a bundle. (Yes, I know 
that is not the WebRTC use case.)

	Thanks,
	Paul

> Only. Basically, it's a secret handshake that allows you
> to negotiate bundle at the same time as using it without confusing those
> implementations that have no idea what all this bundle stuff means. Once
> you've established that the other end can understand bundle, you don't
> need the port-0 indication.
>
> /a
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/9/13 6:38 AM, Christer Holmberg wrote:
> Hi,
> An alternative way, using BUNDLE as currently specified, is the add the
> bundle-only m- lines in a subsequent Offer, once BUNDLE usage has been
> negotiated. Then one does not need port zero. Of course, that would a
> "third" Offer (technically it could be done already in the "second"
> Offer, but BUNDLE recommends against that).

I quite like this, but prefer that it be allowed in the 2nd O/A.
It is consistent with my desire to make many of these "rules" a bit more 
lenient.

If done in the 2nd O/A, which is required anyway, then that should 
eliminate much (not all) of the objections that are likely to come from 
RTCWEB.

	Thanks,
	Paul

> Regards,
> Christer
>
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> *To:* Adam Roach [adam@nostrum.com]
> *CC:* Christer Holmberg [christer.holmberg@ericsson.com]; mmusic
> [mmusic@ietf.org]
> *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> bundle-only m- lines?
>
> On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/8/13 15:51, Hadriel Kaplan wrote:
>>> BTW, I'm not speaking about my employer's SBCs in particular - I actually have no idea what we'll do with an UP bundle-only thing lying about a port=0.  We could handle it all correctly, or tear the call down later, or call WOPR and tell it to play Global  Thermonuclear War.  If something's broken in our implementation, we'll
> fix it... eventually.  I'm more worried about other systems, from
> vendors who don't attend the IETF; and I'm worried about the people who
> have to run SIP networks, for whom this stuff is not a greenfield.
>>
>> The thing is: no matter what we do here, it's not going to look 100% like the stuff that came before it. All we can do is look at the products we know about, look at the existing published semantics for protocol elements, and make educated guesses about what  is likely to cause the fewest problems. We *know* that using the same
> port numbers for multiple m-lines causes bad behavior in more than a few
> endpoints, so that's out. On the other hand, the semantics of port=0 are
> pretty well defined, elicit the exact response we want for remote
> endpoints, and avoid egregiously large port allocations during call
> setup (which is an important goal).
>
> The semantics of using port=0 are well defined, as you say - those
> semantics are that you MUST NOT use the media.  You are changing those
> semantics, if the far-end supports BUNDLE.  You feel it's safe to change
> those semantics, because you feel that SDP is only being processed by
> the local and remote endpoints.  I'm saying that's a really bad
> assumption, because I know that in fact it is not the case.
>
>
>
>> I think that's a lot more solid set of reasons than the opposing rationale you proffer of "there may or may not be products somewhere in the world that don't really like this approach," not least of all because that statement is true of any approach we come  up with.
>
> Do you honestly believe that using a port number of 0 in an offer, while
> the answer uses a port number of N, is equally impacting on
> interoperability as using an attribute for capability discovery and then
> a second O/A round of the real numbers?  C'mon man. You've seen the
> stuff that's out there.  The SDP Offer and Answer's m-line port number
> is used by all kinds of whacky equipment, to determine the RTP/RTCP flow
> information.  It's used by passive monitoring systems, NATs, CGNs,
> Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and god knows what
> else.  Even just in my own employer, we've got five different types of
> equipment that look at the the m-line port numbers, only one of which is
> our SBCs (and we've got multiple flavors of them too, fwiw).
>
> -hadriel
>
>
>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/9/13 11:42, Paul Kyzivat wrote:
> Wow! I ignore this thread for a day and it goes wild!
> I get the advantage of looking at the long discussion and only 
> replying selectively.
>
> On 8/8/13 9:37 PM, Adam Roach wrote:
>> On 8/8/13 14:11, Hadriel Kaplan wrote:
>>> If I send a new SDP Offer 10 minutes later during the session, and I
>>> set an existing m-line to port 0 in that Offer, does it mean I want to
>>> disable the media, or restart bundling, or what?
>>
>> We already talked about that in Berlin: it means you want to disable the
>> line. We're assigning special semantics to port=0 in the **FIRST** offer
>> of a session.
>
> IMO this is the wrong way to think of this special case.
> Rather, it is special to the first offer that proposes a new bundle.
> It need not be the first offer of the session. It may be that you have 
> many offers in the session that don't use bundle at all. And then 
> eventually you get into a state where you need a bundle. (Yes, I know 
> that is not the WebRTC use case.)

Yes, that's a good point. Basically, it's got to be the first message 
that tries to use bundle.

/a

From hadriel.kaplan@oracle.com  Fri Aug  9 10:03:20 2013
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 9, 2013, at 11:28 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> Uh, just say no to this.
>=20
> We can't design a system under the constraint that it doesn't annoy
> any middlebox in the world.

If we're talking about solely for WebRTC usage, I agree: we have a =
greenfield playground, in an early phase of life.

If we're talking about SIP, it's a widely deployed critical =
infrastructure - we can make changes, but we really do have to be super =
cautious about what the changes are and what they impact.  Not only have =
we previously done exactly that for exactly these reasons for very =
similar situations for SIP including SDP changes... but we do the same =
for other IETF protocols that are no longer in incubation phases but =
rather widely deployed and critical for users that they don't break: for =
example DNS, SMTP, and even HTTP.  The IETF recognizes that there are =
undefined middleboxes (both active and passive), in various deployment =
models, which constrain the types of changes that can be safely made to =
those protocols; and we recognize that breaking things isn't in the best =
interest of the users of the Internet, nor the admins which have to run =
it.

I'm not saying we can't ever make changes to SIP - of course we can, and =
we do - but we really do need to consider the implications of what we're =
changing on the deployed infrastructure, and in way more than a one =
month "let us know if we break stuff" review period.

-hadriel


From christer.holmberg@ericsson.com  Fri Aug  9 12:51:08 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi,

Also keep in mind that what I am suggesting *IS* already supported by BUNDL=
E. There is nothing we need to change - not in BUNDLE, not in 3264, not any=
where else :)

And, as the second offer is needed anyway, we are not adding any extra dela=
y etc either.

Regards,

Christer


-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Hadriel Kaplan
L=E4hetetty: 9. elokuuta 2013 19:57
Vastaanottaja: Eric Rescorla
Kopio: mmusic
Aihe: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- =
lines?


On Aug 9, 2013, at 11:28 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> Uh, just say no to this.
>=20
> We can't design a system under the constraint that it doesn't annoy=20
> any middlebox in the world.

If we're talking about solely for WebRTC usage, I agree: we have a greenfie=
ld playground, in an early phase of life.

If we're talking about SIP, it's a widely deployed critical infrastructure =
- we can make changes, but we really do have to be super cautious about wha=
t the changes are and what they impact.  Not only have we previously done e=
xactly that for exactly these reasons for very similar situations for SIP i=
ncluding SDP changes... but we do the same for other IETF protocols that ar=
e no longer in incubation phases but rather widely deployed and critical fo=
r users that they don't break: for example DNS, SMTP, and even HTTP.  The I=
ETF recognizes that there are undefined middleboxes (both active and passiv=
e), in various deployment models, which constrain the types of changes that=
 can be safely made to those protocols; and we recognize that breaking thin=
gs isn't in the best interest of the users of the Internet, nor the admins =
which have to run it.

I'm not saying we can't ever make changes to SIP - of course we can, and we=
 do - but we really do need to consider the implications of what we're chan=
ging on the deployed infrastructure, and in way more than a one month "let =
us know if we break stuff" review period.

-hadriel

_______________________________________________
mmusic mailing list
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,
,
>> An alternative way, using BUNDLE as currently specified, is the add=20
>> the bundle-only m- lines in a subsequent Offer, once BUNDLE usage has=20
>> been negotiated. Then one does not need port zero. Of course, that=20
>> would a "third" Offer (technically it could be done already in the "seco=
nd"
>> Offer, but BUNDLE recommends against that).
>
> I quite like this, but prefer that it be allowed in the 2nd O/A.
> It is consistent with my desire to make many of these "rules" a bit more =
lenient.
>
> If done in the 2nd O/A, which is required anyway, then that should elimin=
ate much (not all) of the objections that are likely to come from RTCWEB.

Yes.

And, the 2nd O/A should be fine, as the new bundle-only m- lines can be rej=
ected (using port zero in the Answer) without having to reject the whole Of=
fer (i.e. it won't affect the "port synchronization" of the other m- lines =
in the Offer).

Regards,

Christer






> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> *To:* Adam Roach [adam@nostrum.com]
> *CC:* Christer Holmberg [christer.holmberg@ericsson.com]; mmusic=20
> [mmusic@ietf.org]
> *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for=20
> bundle-only m- lines?
>
> On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 8/8/13 15:51, Hadriel Kaplan wrote:
>>> BTW, I'm not speaking about my employer's SBCs in particular - I=20
>>> actually have no idea what we'll do with an UP bundle-only thing=20
>>> lying about a port=3D0.  We could handle it all correctly, or tear the=
=20
>>> call down later, or call WOPR and tell it to play Global =20
>>> Thermonuclear War.  If something's broken in our implementation,=20
>>> we'll
> fix it... eventually.  I'm more worried about other systems, from=20
> vendors who don't attend the IETF; and I'm worried about the people=20
> who have to run SIP networks, for whom this stuff is not a greenfield.
>>
>> The thing is: no matter what we do here, it's not going to look 100%=20
>> like the stuff that came before it. All we can do is look at the=20
>> products we know about, look at the existing published semantics for=20
>> protocol elements, and make educated guesses about what  is likely to=20
>> cause the fewest problems. We *know* that using the same
> port numbers for multiple m-lines causes bad behavior in more than a=20
> few endpoints, so that's out. On the other hand, the semantics of=20
> port=3D0 are pretty well defined, elicit the exact response we want for=20
> remote endpoints, and avoid egregiously large port allocations during=20
> call setup (which is an important goal).
>
> The semantics of using port=3D0 are well defined, as you say - those=20
> semantics are that you MUST NOT use the media.  You are changing those=20
> semantics, if the far-end supports BUNDLE.  You feel it's safe to=20
> change those semantics, because you feel that SDP is only being=20
> processed by the local and remote endpoints.  I'm saying that's a=20
> really bad assumption, because I know that in fact it is not the case.
>
>
>
>> I think that's a lot more solid set of reasons than the opposing rationa=
le you proffer of "there may or may not be products somewhere in the world =
that don't really like this approach," not least of all because that statem=
ent is true of any approach we come  up with.
>
> Do you honestly believe that using a port number of 0 in an offer,=20
> while the answer uses a port number of N, is equally impacting on=20
> interoperability as using an attribute for capability discovery and=20
> then a second O/A round of the real numbers?  C'mon man. You've seen=20
> the stuff that's out there.  The SDP Offer and Answer's m-line port=20
> number is used by all kinds of whacky equipment, to determine the=20
> RTP/RTCP flow information.  It's used by passive monitoring systems,=20
> NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and=20
> god knows what else.  Even just in my own employer, we've got five=20
> different types of equipment that look at the the m-line port numbers,=20
> only one of which is our SBCs (and we've got multiple flavors of them too=
, fwiw).
>
> -hadriel
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

> Wow! I ignore this thread for a day and it goes wild!
> I get the advantage of looking at the long discussion and only replying s=
electively.

We should have someone "taking notes" of the e-mail discussions, and then p=
ost a week-everyone-who-cares-need-to-know summary once a week :)

>>> If I send a new SDP Offer 10 minutes later during the session, and I=20
>>> set an existing m-line to port 0 in that Offer, does it mean I want=20
>>> to disable the media, or restart bundling, or what?
>>
>> We already talked about that in Berlin: it means you want to disable=20
>> the line. We're assigning special semantics to port=3D0 in the **FIRST**=
=20
>> offer of a session.
>
> IMO this is the wrong way to think of this special case.
> Rather, it is special to the first offer that proposes a new bundle.
> It need not be the first offer of the session. It may be that you have ma=
ny offers in the session that don't use bundle at all. And then eventually =
you get into a state where you need a bundle. (Yes, I know that is not the =
WebRTC use case.)

Can't you have a WebRTC use case with only one m- line, which obviously wou=
ldn't require usage of BUNDLE? Then,  when you later add new m- lines, you =
also want to use BUNDLE...

Regards,

Christer



> Only. Basically, it's a secret handshake that allows you to negotiate=20
> bundle at the same time as using it without confusing those=20
> implementations that have no idea what all this bundle stuff means.=20
> Once you've established that the other end can understand bundle, you=20
> don't need the port-0 indication.
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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Subject: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)]
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Greetings MMUSIC

As you can see per the below, the IESG approved the Circuit-Switched 
Bearers draft for publication, however before doing so, they requested a 
couple of changes. The resulting approved document can be found at 
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21

One of the changes introduced resulted in all 3 correlation mechanisms 
now being mandatory-to-implement (previously they were not):
<quote>

    Implementations MUST support the three correlation mechanisms
    specified inSection 5.2.3.2  <http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.2>,Section 5.2.3.3  <http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.3>  andSection 5.2.3.4  <http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.4>,
    and SHOULD include all supported mechanisms the initial Offer.

</quote>
Since this is a non-trivial change that has not been discussed by the WG 
(and we know of at least one person having expressed concerns with it 
off-line), we are hereby solicting the WG participants for input on 
changes introduced as a result of the IESG review with a focus on the 
above. In doing so, please keep in mind that we are developing standards 
that work in the general Internet.


The following URL provides a diff-file showing the changes resulting 
from the IESG review:
http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&url2=draft-ietf-mmusic-sdp-cs-21.txt

Comments are due by August 23 (two weeks from today)

Thanks

-- Flemming (MMUSIC co-chair)



-------- Original Message --------
Subject: 	[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) 
Extension For Setting Audio and Video Media Streams Over 
Circuit-Switched Bearers In The Public Switched Telephone Network 
(PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
Date: 	Mon, 1 Jul 2013 11:56:23 -0700
From: 	The IESG <iesg-secretary@ietf.org>
To: 	IETF-Announce <ietf-announce@ietf.org>
CC: 	mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list 
<mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>



The IESG has approved the following document:
- 'Session Description Protocol (SDP) Extension For Setting Audio and
    Video Media Streams Over Circuit-Switched Bearers In The Public
    Switched Telephone Network (PSTN)'
   (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard

This document is the product of the Multiparty Multimedia Session Control
Working Group.

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/




*Technical Summary
*

The document describes use cases, requirements, and protocol extensions
for using the Session Description Protocol (SDP) Offer/Answer model for
establishing audio and video media streams over circuit-switched bearers
in the Publich Switched Telephone Network (PSTN)

*Working Group Summary
*

The WG had some discussion around the format to use for E.164 numbers
and whether to align this with the existing definition in RFC 3108. The
RFC 3108 definition was seen as deficient and the WG agreed it was
better to align with relevant parts of the tel URI format defined in RFC
3966, not least since SDP address types are defined in the context of a
particular network type, and hence RFC 3108 compatibility is not a
concern (the implication is that the "E164" address type may differ
between network types in SDP).

*Document Quality
*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

The document has received good overall review in the WG, some of which
resulted in changes to the detailed specification. The document has been
reviewed in detail several times, including of the last few versions.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.


*Personnel
*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic
.




--------------020204010109030904060700
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    Greetings MMUSIC<br>
    <br>
    As you can see per the below, the IESG approved the Circuit-Switched
    Bearers draft for publication, however before doing so, they
    requested a couple of changes. The resulting approved document can
    be found at <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21">http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21</a><br>
    <br>
    One of the changes introduced resulted in all 3 correlation
    mechanisms now being mandatory-to-implement (previously they were
    not):<br>
    &lt;quote&gt;<br>
    <meta charset="utf-8">
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">   Implementations MUST support the three correlation mechanisms
   specified in <a href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.2">Section 5.2.3.2</a>, <a href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.3">Section 5.2.3.3</a> and <a href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.4">Section 5.2.3.4</a>,
   and SHOULD include all supported mechanisms the initial Offer.</pre>
    &lt;/quote&gt;<br>
    Since this is a non-trivial change that has not been discussed by
    the WG (and we know of at least one person having expressed concerns
    with it off-line), we are hereby solicting the WG participants for
    input on changes introduced as a result of the IESG review with a
    focus on the above. In doing so, please keep in mind that we are
    developing standards that work in the general Internet. <br>
    <br>
    <br>
    The following URL provides a diff-file showing the changes resulting
    from the IESG review:<br>
    &nbsp;&nbsp;&nbsp; <a
href="http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&amp;url2=draft-ietf-mmusic-sdp-cs-21.txt"
      target="_blank">http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&amp;url2=draft-ietf-mmusic-sdp-cs-21.txt</a><br>
    <br>
    Comments are due by August 23 (two weeks from today)<br>
    <br>
    Thanks <br>
    <br>
    -- Flemming (MMUSIC co-chair)<br>
    <br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>[MMUSIC] Protocol Action: 'Session Description Protocol
              (SDP) Extension For Setting Audio and Video Media Streams
              Over Circuit-Switched Bearers In The Public Switched
              Telephone Network (PSTN)' to Proposed Standard
              (draft-ietf-mmusic-sdp-cs-21.txt)</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Mon, 1 Jul 2013 11:56:23 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td>The IESG <a class="moz-txt-link-rfc2396E" href="mailto:iesg-secretary@ietf.org">&lt;iesg-secretary@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>IETF-Announce <a class="moz-txt-link-rfc2396E" href="mailto:ietf-announce@ietf.org">&lt;ietf-announce@ietf.org&gt;</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
            <td>mmusic chair <a class="moz-txt-link-rfc2396E" href="mailto:mmusic-chairs@tools.ietf.org">&lt;mmusic-chairs@tools.ietf.org&gt;</a>,
              mmusic mailing list <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a>, RFC Editor
              <a class="moz-txt-link-rfc2396E" href="mailto:rfc-editor@rfc-editor.org">&lt;rfc-editor@rfc-editor.org&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>The IESG has approved the following document:
- 'Session Description Protocol (SDP) Extension For Setting Audio and
   Video Media Streams Over Circuit-Switched Bearers In The Public
   Switched Telephone Network (PSTN)'
  (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard

This document is the product of the Multiparty Multimedia Session Control
Working Group.

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:
<a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/">http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/</a>




*Technical Summary
*

The document describes use cases, requirements, and protocol extensions
for using the Session Description Protocol (SDP) Offer/Answer model for
establishing audio and video media streams over circuit-switched bearers
in the Publich Switched Telephone Network (PSTN)

*Working Group Summary
*

The WG had some discussion around the format to use for E.164 numbers
and whether to align this with the existing definition in RFC 3108. The
RFC 3108 definition was seen as deficient and the WG agreed it was
better to align with relevant parts of the tel URI format defined in RFC
3966, not least since SDP address types are defined in the context of a
particular network type, and hence RFC 3108 compatibility is not a
concern (the implication is that the "E164" address type may differ
between network types in SDP).

*Document Quality
*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

The document has received good overall review in the WG, some of which
resulted in changes to the detailed specification. The document has been
reviewed in detail several times, including of the last few versions.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.


*Personnel
*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo 
_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
.

</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------020204010109030904060700--

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Date: Fri, 9 Aug 2013 15:35:00 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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That is not my understanding. If we don't care about intermediaries, we
could have made BUNDLE and UP a lot simpler.

I think the expectation should be that Matthew's proposed case will work,
modulo any implementation bugs on the remote device.


On Wed, Aug 7, 2013 at 8:27 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  If everyone has the same understanding, I would ask the chairs to make
> it *very clear* in the final minutes ;)****
>
> ** **
>
> Regards,****
>
> ** **
>
> Christer****
>
> ** **
>
> ** **
>
> ** **
>
> *L=C3=A4hett=C3=A4j=C3=A4:* Mary Barnes [mailto:mary.ietf.barnes@gmail.co=
m]
> *L=C3=A4hetetty:* 7. elokuuta 2013 18:25
> *Vastaanottaja:* Christer Holmberg
> *Kopio:* Paul Kyzivat; mmusic@ietf.org
>
> *Aihe:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"****
>
> ** **
>
> ** **
>
> ** **
>
> On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:****
>
> Hi,****
>
>  ****
>
> Ok, so what is =E2=80=9DRTCWEB usage=E2=80=9D then? :)****
>
>  ****
>
> For example, if I implement a JavaScript SIP app, and want to use the UP
> with SIP on the wire, would that be a =E2=80=9CRTCWEB usage=E2=80=9D? My =
understanding is
> that it would NOT be.****
>
>  ****
>
> My personal understanding of the scope is:****
>
> [MB] That is also my understanding. [/MB] ****
>
>   ****
>
> 1)      JSEP; and****
>
> 2)      Browser-to-browser cases where we KNOW there are no
> intermediaries etc in between.****
>
>  ****
>
> Regards,****
>
>  ****
>
> Christer****
>
>  ****
>
> *L=C3=A4hett=C3=A4j=C3=A4:* mmusic-bounces@ietf.org [mailto:mmusic-bounce=
s@ietf.org] *Puolesta
> *Mary Barnes
> *L=C3=A4hetetty:* 7. elokuuta 2013 17:56
> *Vastaanottaja:* Paul Kyzivat
> *Kopio:* mmusic@ietf.org
> *Aihe:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"****
>
>  ****
>
> On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat <pkyzivat@alum.mit.edu>
> wrote:****
>
>  On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:****
>
> As one of those that used the term, my intention was that the initial
> decision was to agree to define what RTCWEB needed.
>
> As each dependent working group looks at the extensions they need to do,
> then they need to make the decision as to whether a wider scope of
> possible, desirable, or dangerous.
>
> So the BUNDLE status at the moment is that any changes for the unified
> plan should work for RTCWEB usage, but not for general. It would
> probably make sense to make it more generally applicable, but that is
> the next step in the decision process, and may need some other technical
> decisions before it happens.****
>
>  ****
>
> I don't know how you intend the above to be parsed. Specifically, when yo=
u
> say:****
>
>
>   "any changes for the unified plan should work for RTCWEB usage"****
>
> Do you mean:
>
> 1) that the changes should be designed solely with RTCWEB usage in mind?
> (But are still applicable in other environments that find them useful and
> choose to adopt them.)
>
> Or do you mean:
>
> 2) that those changes should not be used in any environment other than
> RTCWEB?
>
> I might be able to accept (1), but not (2).****
>
>  [MB] I would agree with your (1), however, not with the parenthetical
> part.  I would suggest that be "But MAY still be applicable in other
> environments... "   My concern is that at this point in time, you can't s=
ay
> that the changes ARE applicable to other environments ....  [/MB]****
>
>  ****
>
>
>         Thanks,
>         Paul****
>
> Keith
>
> ------------------------------------------------------------------------
>
> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
> Behalf Of *Bernard Aboba
> *Sent:* 07 August 2013 00:50
> *To:* Christer Holmberg; mmusic
> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"****
>
>
>
> Christer said:
>
> "While cleaning up my MMUSIC notes today, I was listening to the audio
> recording, in order to get a clarification regarding what "scope is
> WEBRTC" really means.  However, unless I missed something, I didn't get
> a clear answer.
>
> Does it only cover the browser API (JSEP) or does it also cover
> something that can be sent out on the wire (as long as it is not SIP) by
> a browser?
>
> Feel free to tell me that it is a stupid question, and that the answer
> is very clear :)"
>
> [BA]  I actually think it is a good question :)
>
> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
> Plan" applies to use within  the JSEP API.    As a result, one
> might expect the SDP used in createOffer, setLocalDescription and
> setRemoteDescription to resemble that in the "Unified Plan" document.
> Since signaling is out of scope for WebRTC, applications that use the
> JSEP API might or might not use the "Unified Plan" SDP in signaling on
> the wire.
>
> However, "Unified Plan" includes normative references to published SDP
> specifications that were developed for general use, and the adoption of
> "Unified Plan" (or even an applicability statement within the "Unified
> Plan" document) doesn't change the applicability of those documents.
>
> Also, there may be some specifications that are part of the "Unified
> Plan" workload that may have applicability beyond WebRTC (e.g. a=3Dgroup:
> SIMULCAST)
>
> Ultimately, I am expecting the work involved in "Unified Plan" to span
> multiple documents, and to clarify the applicability,  those individual
> documents will need to explain what they apply to (just WebRTC use or
> more general use).
>
> ****
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic****
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic****
>
>   ****
>
>  ** **
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--047d7bdc0ddac8d6ce04e38b63bf
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<div dir=3D"ltr">That is not my understanding. If we don&#39;t care about i=
ntermediaries, we could have made BUNDLE and UP a lot simpler.<div><br></di=
v><div>I think the expectation should be that Matthew&#39;s proposed case w=
ill work, modulo any implementation bugs on the remote device.</div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed,=
 Aug 7, 2013 at 8:27 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D=
"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg=
@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">If everyon=
e has the same understanding, I would ask the chairs to make it
<b>very clear</b> in the final minutes ;)<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=
=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=
=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u=
></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=
=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=
=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=
=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=C3=A4hett=C3=A4j=C3=A4:</span></b><=
span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-se=
rif&quot;"> Mary Barnes [mailto:<a href=3D"mailto:mary.ietf.barnes@gmail.co=
m" target=3D"_blank">mary.ietf.barnes@gmail.com</a>]
<br>
<b>L=C3=A4hetetty:</b> 7. elokuuta 2013 18:25<br>
<b>Vastaanottaja:</b> Christer Holmberg<br>
<b>Kopio:</b> Paul Kyzivat; <a href=3D"mailto:mmusic@ietf.org" target=3D"_b=
lank">mmusic@ietf.org</a></span></p><div><div class=3D"h5"><br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<u></u>=
<u></u></div></div><p></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<p class=3D"MsoNormal">On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt; wrote:<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,</span><u></u><u></u><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Ok, so wha=
t is =E2=80=9DRTCWEB usage=E2=80=9D then? :)</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">For exampl=
e, if I implement a JavaScript SIP app, and want to use the UP with SIP on
 the wire, would that be a =E2=80=9CRTCWEB usage=E2=80=9D? My understanding=
 is that it would NOT be.</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">My persona=
l understanding of the scope is:</span><u></u><u></u></p>
</div>
</div>
<div>
<p class=3D"MsoNormal">[MB] That is also my understanding. [/MB]=C2=A0<u></=
u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri=
&quot;,&quot;sans-serif&quot;;color:#1f497d">1)</span><span lang=3D"EN-US" =
style=3D"font-size:7.0pt;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1f497d">JSEP; and</span><u></u><u>=
</u></p>
<p><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri=
&quot;,&quot;sans-serif&quot;;color:#1f497d">2)</span><span lang=3D"EN-US" =
style=3D"font-size:7.0pt;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Browser-to-browser cases w=
here we KNOW there are no intermediaries etc in between.</span><u></u><u></=
u></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,</=
span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer</=
span><u></u><u></u></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</sp=
an><u></u><u></u></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=C3=A4hett=C3=A4j=C3=A4:</span></b><=
span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-se=
rif&quot;">
<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"=
_blank">mmusic-bounces@ietf.org</a>]
<b>Puolesta </b>Mary Barnes<br>
<b>L=C3=A4hetetty:</b> 7. elokuuta 2013 17:56<br>
<b>Vastaanottaja:</b> Paul Kyzivat<br>
<b>Kopio:</b> <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@i=
etf.org</a><br>
<b>Aihe:</b> Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;</span>=
<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<div>
<div>
<p class=3D"MsoNormal">On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.e=
du</a>&gt; wrote:<u></u><u></u></p>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<p class=3D"MsoNormal">On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:<u></u=
><u></u></p>
<p class=3D"MsoNormal">As one of those that used the term, my intention was=
 that the initial<br>
decision was to agree to define what RTCWEB needed.<br>
<br>
As each dependent working group looks at the extensions they need to do,<br=
>
then they need to make the decision as to whether a wider scope of<br>
possible, desirable, or dangerous.<br>
<br>
So the BUNDLE status at the moment is that any changes for the unified<br>
plan should work for RTCWEB usage, but not for general. It would<br>
probably make sense to make it more generally applicable, but that is<br>
the next step in the decision process, and may need some other technical<br=
>
decisions before it happens.<u></u><u></u></p>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">I don&#39;t know how you intend the above to be pars=
ed. Specifically, when you say:<u></u><u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
=C2=A0 &quot;any changes for the unified plan should work for RTCWEB usage&=
quot;<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">Do you mean:<br>
<br>
1) that the changes should be designed solely with RTCWEB usage in mind? (B=
ut are still applicable in other environments that find them useful and cho=
ose to adopt them.)<br>
<br>
Or do you mean:<br>
<br>
2) that those changes should not be used in any environment other than RTCW=
EB?<br>
<br>
I might be able to accept (1), but not (2).<u></u><u></u></p>
</blockquote>
<div>
<p class=3D"MsoNormal">[MB] I would agree with your (1), however, not with =
the parenthetical part. =C2=A0I would suggest that be &quot;But MAY still b=
e applicable in other environments... &quot; =C2=A0 My concern is that
 at this point in time, you can&#39;t say that the changes ARE applicable t=
o other environments .... =C2=A0[/MB]<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<u></u><u></u></p>
<p class=3D"MsoNormal">Keith<br>
<br>
------------------------------------------------------------------------<br=
>
<br>
*From:*<a href=3D"mailto:mmusic-bounces@ietf.org" target=3D"_blank">mmusic-=
bounces@ietf.org</a> [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org" tar=
get=3D"_blank">mmusic-bounces@ietf.org</a>] *On<br>
Behalf Of *Bernard Aboba<br>
*Sent:* 07 August 2013 00:50<br>
*To:* Christer Holmberg; mmusic<br>
*Subject:* Re: [MMUSIC] UP: Meaning of &quot;scope is WEBRTC&quot;<u></u><u=
></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
Christer said:<br>
<br>
&quot;While cleaning up my MMUSIC notes today, I was listening to the audio=
<br>
recording, in order to get a clarification regarding what &quot;scope is<br=
>
WEBRTC&quot; really means. =C2=A0However, unless I missed something, I didn=
&#39;t get<br>
a clear answer.<br>
<br>
Does it only cover the browser API (JSEP) or does it also cover<br>
something that can be sent out on the wire (as long as it is not SIP) by<br=
>
a browser?<br>
<br>
Feel free to tell me that it is a stupid question, and that the answer<br>
is very clear :)&quot;<br>
<br>
[BA] =C2=A0I actually think it is a good question :)<br>
<br>
One interpretation of &quot;scope is WebRTC&quot; =C2=A0is that the =C2=A0S=
DP in &quot;Unified<br>
Plan&quot; applies to use within =C2=A0the JSEP API. =C2=A0 =C2=A0As a resu=
lt, one<br>
might expect the SDP used in createOffer, setLocalDescription and<br>
setRemoteDescription to resemble that in the &quot;Unified Plan&quot; docum=
ent.<br>
Since signaling is out of scope for WebRTC, applications that use the<br>
JSEP API might or might not use the &quot;Unified Plan&quot; SDP in signali=
ng on<br>
the wire.<br>
<br>
However, &quot;Unified Plan&quot; includes normative references to publishe=
d SDP<br>
specifications that were developed for general use, and the adoption of<br>
&quot;Unified Plan&quot; (or even an applicability statement within the &qu=
ot;Unified<br>
Plan&quot; document) doesn&#39;t change the applicability of those document=
s.<br>
<br>
Also, there may be some specifications that are part of the &quot;Unified<b=
r>
Plan&quot; workload that may have applicability beyond WebRTC (e.g. a=3Dgro=
up:<br>
SIMULCAST)<br>
<br>
Ultimately, I am expecting the work involved in &quot;Unified Plan&quot; to=
 span<br>
multiple documents, and to clarify the applicability, =C2=A0those individua=
l<br>
documents will need to explain what they apply to (just WebRTC use or<br>
more general use).<br>
<br>
<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">_____________________=
__________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><u></u><u></u></p>
<p class=3D"MsoNormal"><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><u></u><u></u></p>
</blockquote>
</div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div></div></div>
</div>

<br>_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
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<br></blockquote></div><br></div>

--047d7bdc0ddac8d6ce04e38b63bf--

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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On 8/9/2013 3:35 PM, Justin Uberti wrote:
> That is not my understanding. If we don't care about intermediaries, 
> we could have made BUNDLE and UP a lot simpler.
>
> I think the expectation should be that Matthew's proposed case will 
> work, modulo any implementation bugs on the remote device.
>


I agree. Otherwise we're doing this "standards" thing wrong OR shouldn't 
be reusing *this* standard for *that* application.

Matthew Kaufman

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From: Justin Uberti <juberti@google.com>
Date: Fri, 9 Aug 2013 16:20:37 -0700
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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--047d7bdc0ddae75c3204e38c06c9
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Requiring a second offer to add the bundle-only m-lines defeats the purpose
of bundle-only.
a) The delay to do an O/A will often exceed the time to allocate
candidates, for small numbers of streams; bundle-only was meant to avoid
this candidate latency.
b) Once you have the initial answer, you know whether the other side
supports BUNDLE, so you can take the appropriate action (and so there is no
need for bundle-only).

If SIP can't deal with the bundle-only concept, then WebRTC apps that care
about latency will have to decide whether they are running in WebRTC-only
mode or compatibility mode. In WebRTC-only mode, streams will be bundled in
the initial offer; in compatibility mode, 2 offer-answers will be needed.

This would be a somewhat unfortunate outcome, so I'd like to see
bundle-only (or something like it) work.






On Fri, Aug 9, 2013 at 12:49 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
> ,
> >> An alternative way, using BUNDLE as currently specified, is the add
> >> the bundle-only m- lines in a subsequent Offer, once BUNDLE usage has
> >> been negotiated. Then one does not need port zero. Of course, that
> >> would a "third" Offer (technically it could be done already in the
> "second"
> >> Offer, but BUNDLE recommends against that).
> >
> > I quite like this, but prefer that it be allowed in the 2nd O/A.
> > It is consistent with my desire to make many of these "rules" a bit more
> lenient.
> >
> > If done in the 2nd O/A, which is required anyway, then that should
> eliminate much (not all) of the objections that are likely to come from
> RTCWEB.
>
> Yes.
>
> And, the 2nd O/A should be fine, as the new bundle-only m- lines can be
> rejected (using port zero in the Answer) without having to reject the whole
> Offer (i.e. it won't affect the "port synchronization" of the other m-
> lines in the Offer).
>
> Regards,
>
> Christer
>
>
>
>
>
>
> > Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
> >
> > -----Original Message-----
> > *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> > *To:* Adam Roach [adam@nostrum.com]
> > *CC:* Christer Holmberg [christer.holmberg@ericsson.com]; mmusic
> > [mmusic@ietf.org]
> > *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> > bundle-only m- lines?
> >
> > On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
> >
> >> On 8/8/13 15:51, Hadriel Kaplan wrote:
> >>> BTW, I'm not speaking about my employer's SBCs in particular - I
> >>> actually have no idea what we'll do with an UP bundle-only thing
> >>> lying about a port=0.  We could handle it all correctly, or tear the
> >>> call down later, or call WOPR and tell it to play Global
> >>> Thermonuclear War.  If something's broken in our implementation,
> >>> we'll
> > fix it... eventually.  I'm more worried about other systems, from
> > vendors who don't attend the IETF; and I'm worried about the people
> > who have to run SIP networks, for whom this stuff is not a greenfield.
> >>
> >> The thing is: no matter what we do here, it's not going to look 100%
> >> like the stuff that came before it. All we can do is look at the
> >> products we know about, look at the existing published semantics for
> >> protocol elements, and make educated guesses about what  is likely to
> >> cause the fewest problems. We *know* that using the same
> > port numbers for multiple m-lines causes bad behavior in more than a
> > few endpoints, so that's out. On the other hand, the semantics of
> > port=0 are pretty well defined, elicit the exact response we want for
> > remote endpoints, and avoid egregiously large port allocations during
> > call setup (which is an important goal).
> >
> > The semantics of using port=0 are well defined, as you say - those
> > semantics are that you MUST NOT use the media.  You are changing those
> > semantics, if the far-end supports BUNDLE.  You feel it's safe to
> > change those semantics, because you feel that SDP is only being
> > processed by the local and remote endpoints.  I'm saying that's a
> > really bad assumption, because I know that in fact it is not the case.
> >
> >
> >
> >> I think that's a lot more solid set of reasons than the opposing
> rationale you proffer of "there may or may not be products somewhere in the
> world that don't really like this approach," not least of all because that
> statement is true of any approach we come  up with.
> >
> > Do you honestly believe that using a port number of 0 in an offer,
> > while the answer uses a port number of N, is equally impacting on
> > interoperability as using an attribute for capability discovery and
> > then a second O/A round of the real numbers?  C'mon man. You've seen
> > the stuff that's out there.  The SDP Offer and Answer's m-line port
> > number is used by all kinds of whacky equipment, to determine the
> > RTP/RTCP flow information.  It's used by passive monitoring systems,
> > NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and
> > god knows what else.  Even just in my own employer, we've got five
> > different types of equipment that look at the the m-line port numbers,
> > only one of which is our SBCs (and we've got multiple flavors of them
> too, fwiw).
> >
> > -hadriel
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

--047d7bdc0ddae75c3204e38c06c9
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Requiring a second offer to add the bundle-only m-lines de=
feats the purpose of bundle-only.<div>a) The delay to do an O/A will often =
exceed the time to allocate candidates, for small numbers of streams; bundl=
e-only was meant to avoid this candidate latency.</div>

<div>b) Once you have the initial answer, you know whether the other side s=
upports BUNDLE, so you can take the appropriate action (and so there is no =
need for bundle-only).</div><div><br></div><div>If SIP can&#39;t deal with =
the bundle-only concept, then WebRTC apps that care about latency will have=
 to decide whether they are running in WebRTC-only mode or compatibility mo=
de. In WebRTC-only mode, streams will be bundled in the initial offer; in c=
ompatibility mode, 2 offer-answers will be needed.</div>

<div><br></div><div>This would be a somewhat unfortunate outcome, so I&#39;=
d like to see bundle-only (or something like it) work.</div><div><div><br><=
/div><div><br><div><br></div><div><br></div></div></div></div><div class=3D=
"gmail_extra">

<br><br><div class=3D"gmail_quote">On Fri, Aug 9, 2013 at 12:49 PM, Christe=
r Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericss=
on.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wro=
te:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div class=3D"im">,<br>
&gt;&gt; An alternative way, using BUNDLE as currently specified, is the ad=
d<br>
&gt;&gt; the bundle-only m- lines in a subsequent Offer, once BUNDLE usage =
has<br>
&gt;&gt; been negotiated. Then one does not need port zero. Of course, that=
<br>
&gt;&gt; would a &quot;third&quot; Offer (technically it could be done alre=
ady in the &quot;second&quot;<br>
&gt;&gt; Offer, but BUNDLE recommends against that).<br>
&gt;<br>
&gt; I quite like this, but prefer that it be allowed in the 2nd O/A.<br>
&gt; It is consistent with my desire to make many of these &quot;rules&quot=
; a bit more lenient.<br>
&gt;<br>
&gt; If done in the 2nd O/A, which is required anyway, then that should eli=
minate much (not all) of the objections that are likely to come from RTCWEB=
.<br>
<br>
</div>Yes.<br>
<br>
And, the 2nd O/A should be fine, as the new bundle-only m- lines can be rej=
ected (using port zero in the Answer) without having to reject the whole Of=
fer (i.e. it won&#39;t affect the &quot;port synchronization&quot; of the o=
ther m- lines in the Offer).<br>


<div class=3D"HOEnZb"><div class=3D"h5"><br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
<br>
<br>
<br>
<br>
&gt; Sent from */Windows/* using *TouchDown* (<a href=3D"http://www.nitrode=
sk.com" target=3D"_blank">www.nitrodesk.com</a>)<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; *From:* Hadriel Kaplan [<a href=3D"mailto:hadriel.kaplan@oracle.com">h=
adriel.kaplan@oracle.com</a>]<br>
&gt; *To:* Adam Roach [<a href=3D"mailto:adam@nostrum.com">adam@nostrum.com=
</a>]<br>
&gt; *CC:* Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.=
com">christer.holmberg@ericsson.com</a>]; mmusic<br>
&gt; [<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>]<br>
&gt; *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for<br>
&gt; bundle-only m- lines?<br>
&gt;<br>
&gt; On Aug 8, 2013, at 5:02 PM, Adam Roach &lt;<a href=3D"mailto:adam@nost=
rum.com">adam@nostrum.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt;&gt;&gt; BTW, I&#39;m not speaking about my employer&#39;s SBCs in part=
icular - I<br>
&gt;&gt;&gt; actually have no idea what we&#39;ll do with an UP bundle-only=
 thing<br>
&gt;&gt;&gt; lying about a port=3D0. =C2=A0We could handle it all correctly=
, or tear the<br>
&gt;&gt;&gt; call down later, or call WOPR and tell it to play Global<br>
&gt;&gt;&gt; Thermonuclear War. =C2=A0If something&#39;s broken in our impl=
ementation,<br>
&gt;&gt;&gt; we&#39;ll<br>
&gt; fix it... eventually. =C2=A0I&#39;m more worried about other systems, =
from<br>
&gt; vendors who don&#39;t attend the IETF; and I&#39;m worried about the p=
eople<br>
&gt; who have to run SIP networks, for whom this stuff is not a greenfield.=
<br>
&gt;&gt;<br>
&gt;&gt; The thing is: no matter what we do here, it&#39;s not going to loo=
k 100%<br>
&gt;&gt; like the stuff that came before it. All we can do is look at the<b=
r>
&gt;&gt; products we know about, look at the existing published semantics f=
or<br>
&gt;&gt; protocol elements, and make educated guesses about what =C2=A0is l=
ikely to<br>
&gt;&gt; cause the fewest problems. We *know* that using the same<br>
&gt; port numbers for multiple m-lines causes bad behavior in more than a<b=
r>
&gt; few endpoints, so that&#39;s out. On the other hand, the semantics of<=
br>
&gt; port=3D0 are pretty well defined, elicit the exact response we want fo=
r<br>
&gt; remote endpoints, and avoid egregiously large port allocations during<=
br>
&gt; call setup (which is an important goal).<br>
&gt;<br>
&gt; The semantics of using port=3D0 are well defined, as you say - those<b=
r>
&gt; semantics are that you MUST NOT use the media. =C2=A0You are changing =
those<br>
&gt; semantics, if the far-end supports BUNDLE. =C2=A0You feel it&#39;s saf=
e to<br>
&gt; change those semantics, because you feel that SDP is only being<br>
&gt; processed by the local and remote endpoints. =C2=A0I&#39;m saying that=
&#39;s a<br>
&gt; really bad assumption, because I know that in fact it is not the case.=
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt; I think that&#39;s a lot more solid set of reasons than the opposi=
ng rationale you proffer of &quot;there may or may not be products somewher=
e in the world that don&#39;t really like this approach,&quot; not least of=
 all because that statement is true of any approach we come =C2=A0up with.<=
br>


&gt;<br>
&gt; Do you honestly believe that using a port number of 0 in an offer,<br>
&gt; while the answer uses a port number of N, is equally impacting on<br>
&gt; interoperability as using an attribute for capability discovery and<br=
>
&gt; then a second O/A round of the real numbers? =C2=A0C&#39;mon man. You&=
#39;ve seen<br>
&gt; the stuff that&#39;s out there. =C2=A0The SDP Offer and Answer&#39;s m=
-line port<br>
&gt; number is used by all kinds of whacky equipment, to determine the<br>
&gt; RTP/RTCP flow information. =C2=A0It&#39;s used by passive monitoring s=
ystems,<br>
&gt; NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and<=
br>
&gt; god knows what else. =C2=A0Even just in my own employer, we&#39;ve got=
 five<br>
&gt; different types of equipment that look at the the m-line port numbers,=
<br>
&gt; only one of which is our SBCs (and we&#39;ve got multiple flavors of t=
hem too, fwiw).<br>
&gt;<br>
&gt; -hadriel<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div>

--047d7bdc0ddae75c3204e38c06c9--

From hadriel.kaplan@oracle.com  Fri Aug  9 16:33:07 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)]
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Wow.  I can kinda understand why they'd say that (interop reasons, =
presumably), but that's a big deal.  The 3 mechanisms are very =
different, with very different architectural and deployment =
implications.  I have a very hard time believing any of us would truly =
implement all three.  In fact, I don't truly believe anyone would.

So... I hate to do this, but can the authors summarize the history of =
how we got 3 mechanisms to begin with, and who would really implement =
them all?

Or to put it another way, instead of specifying all 3, would it be =
impossible to take a step back and reconsider the real motivating =
use-case (which as far as I know is 3GPP), and only specify the one they =
need/use?

-hadriel


On Aug 9, 2013, at 6:10 PM, Flemming Andreasen <fandreas@cisco.com> =
wrote:

> Greetings MMUSIC
>=20
> As you can see per the below, the IESG approved the Circuit-Switched =
Bearers draft for publication, however before doing so, they requested a =
couple of changes. The resulting approved document can be found at =
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21
>=20
> One of the changes introduced resulted in all 3 correlation mechanisms =
now being mandatory-to-implement (previously they were not):
> <quote>
>    Implementations MUST support the three correlation mechanisms
>    specified in=20
> Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
> ,
>    and SHOULD include all supported mechanisms the initial Offer.
>=20
> </quote>
> Since this is a non-trivial change that has not been discussed by the =
WG (and we know of at least one person having expressed concerns with it =
off-line), we are hereby solicting the WG participants for input on =
changes introduced as a result of the IESG review with a focus on the =
above. In doing so, please keep in mind that we are developing standards =
that work in the general Internet.=20
>=20
>=20
> The following URL provides a diff-file showing the changes resulting =
from the IESG review:
>     =
http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&url2=3D=
draft-ietf-mmusic-sdp-cs-21.txt
>=20
> Comments are due by August 23 (two weeks from today)
>=20
> Thanks=20
>=20
> -- Flemming (MMUSIC co-chair)
>=20
>=20
>=20
> -------- Original Message --------
> Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol =
(SDP) Extension For Setting Audio and Video Media Streams Over =
Circuit-Switched Bearers In The Public Switched Telephone Network =
(PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
> Date:	Mon, 1 Jul 2013 11:56:23 -0700
> From:	The IESG <iesg-secretary@ietf.org>
> To:	IETF-Announce <ietf-announce@ietf.org>
> CC:	mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list =
<mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
>=20
> The IESG has approved the following document:
> - 'Session Description Protocol (SDP) Extension For Setting Audio and
>    Video Media Streams Over Circuit-Switched Bearers In The Public
>    Switched Telephone Network (PSTN)'
>   (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard
>=20
> This document is the product of the Multiparty Multimedia Session =
Control
> Working Group.
>=20
> The IESG contact persons are Gonzalo Camarillo and Richard Barnes.
>=20
> A URL of this Internet Draft is:
>=20
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>=20
>=20
>=20
>=20
>=20
> *Technical Summary
> *
>=20
> The document describes use cases, requirements, and protocol =
extensions
> for using the Session Description Protocol (SDP) Offer/Answer model =
for
> establishing audio and video media streams over circuit-switched =
bearers
> in the Publich Switched Telephone Network (PSTN)
>=20
> *Working Group Summary
> *
>=20
> The WG had some discussion around the format to use for E.164 numbers
> and whether to align this with the existing definition in RFC 3108. =
The
> RFC 3108 definition was seen as deficient and the WG agreed it was
> better to align with relevant parts of the tel URI format defined in =
RFC
> 3966, not least since SDP address types are defined in the context of =
a
> particular network type, and hence RFC 3108 compatibility is not a
> concern (the implication is that the "E164" address type may differ
> between network types in SDP).
>=20
> *Document Quality
> *
>=20
> There are currently no known implementations of the draft, however the
> draft is a dependency for 3GPP, so future implementations are =
expected.
>=20
> The document has received good overall review in the WG, some of which
> resulted in changes to the detailed specification. The document has =
been
> reviewed in detail several times, including of the last few versions.
> The major contributors to these as well as earlier discussions are
> listed in the Acknowledgements section of the document.
>=20
>=20
> *Personnel
> *
>=20
> Document Shepherd: Flemming Andreasen
> Responsible AD: Gonzalo Camarillo=20
> _______________________________________________
> mmusic mailing list
>=20
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20
> .
>=20
>=20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Fri Aug  9 16:42:37 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se> <9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com> <5203D7E7.4070207@nostrum.com> <7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se> <1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com> <5203EB64.70605@nostrum.com> <65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com> <5203F37A.8030204@nostrum.com> <33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com> <5203F4D8.9040904@nostrum.com> <9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com> <5204075B.1080108@nostrum.com> <A2E26DB5-CF47-44DF-8779-D470280977B6@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C422D7D@ESESSMB209.ericsson.se> <52051D0C.2070106@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B1C42339E@ESESSMB209.ericsson.se> <CAOJ7v-2LpF6HGApc5V-AA-D=D=EOtFmErmPSrRPKVdM8kQYHtw@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1C427679@ESESSMB208.ericsson.se>
From: Justin Uberti <juberti@google.com>
Date: Fri, 9 Aug 2013 17:02:46 -0700
Message-ID: <CAOJ7v-3Az7skFCy7KbNDWoMRUP+MPrVMK9Fj8uVAT_Z-dExJ2w@mail.gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Fri, Aug 9, 2013 at 4:36 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi Justin,
>
> > Requiring a second offer to add the bundle-only m-lines defeats the
> purpose of bundle-only.
>
> We already require a second offer. I know that some people, including
> yourself, want to change that, but that is nothing we have agreed on.
>
> > a) The delay to do an O/A will often exceed the time to allocate
> candidates, for small numbers of streams; bundle-only was meant to avoid
> this candidate latency.
>
> You do not have to allocate ports for the bundle-only m- lines.


> > b) Once you have the initial answer, you know whether the other side
> supports BUNDLE, so you can take the appropriate action (and so there is no
> need for bundle-only).
> >
> > If SIP can't deal with the bundle-only concept, then WebRTC apps that
> care about latency will have to decide whether they are running in
> WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will
> be bundled in the initial offer; in compatibility mode, 2 offer-answers
> will be needed.
>
> There are no "modes" in BUNDLE today :)
>

Of course there are. The handling in a reinvite case is distinct from the
initial offer, so that is modal behavior.

>
> > This would be a somewhat unfortunate outcome, so I'd like to see
> bundle-only (or something like it) work.
>
> I don't think the second offer is going to cause any latency. Also, keep
> in mind that you only need the 2nd Offer/Answer during the session
> establishment. As agreed in Berlin, in mid-session changes you do NOT need
> 2 offer/answers.
>

Please provide support for this statement. It seems clear that a 2nd offer
adds _at least_ 1 RTT latency.

>
> Regards,
>
> Christer
>
>
>
>
>
> On Fri, Aug 9, 2013 at 12:49 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> Hi,
> ,
> >> An alternative way, using BUNDLE as currently specified, is the add
> >> the bundle-only m- lines in a subsequent Offer, once BUNDLE usage has
> >> been negotiated. Then one does not need port zero. Of course, that
> >> would a "third" Offer (technically it could be done already in the
> "second"
> >> Offer, but BUNDLE recommends against that).
> >
> > I quite like this, but prefer that it be allowed in the 2nd O/A.
> > It is consistent with my desire to make many of these "rules" a bit more
> lenient.
> >
> > If done in the 2nd O/A, which is required anyway, then that should
> eliminate much (not all) of the objections that are likely to come from
> RTCWEB.
> Yes.
>
> And, the 2nd O/A should be fine, as the new bundle-only m- lines can be
> rejected (using port zero in the Answer) without having to reject the whole
> Offer (i.e. it won't affect the "port synchronization" of the other m-
> lines in the Offer).
>
> Regards,
>
> Christer
>
>
>
>
>
>
> > Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
> >
> > -----Original Message-----
> > *From:* Hadriel Kaplan [hadriel.kaplan@oracle.com]
> > *To:* Adam Roach [adam@nostrum.com]
> > *CC:* Christer Holmberg [christer.holmberg@ericsson.com]; mmusic
> > [mmusic@ietf.org]
> > *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> > bundle-only m- lines?
> >
> > On Aug 8, 2013, at 5:02 PM, Adam Roach <adam@nostrum.com> wrote:
> >
> >> On 8/8/13 15:51, Hadriel Kaplan wrote:
> >>> BTW, I'm not speaking about my employer's SBCs in particular - I
> >>> actually have no idea what we'll do with an UP bundle-only thing
> >>> lying about a port=0.  We could handle it all correctly, or tear the
> >>> call down later, or call WOPR and tell it to play Global
> >>> Thermonuclear War.  If something's broken in our implementation,
> >>> we'll
> > fix it... eventually.  I'm more worried about other systems, from
> > vendors who don't attend the IETF; and I'm worried about the people
> > who have to run SIP networks, for whom this stuff is not a greenfield.
> >>
> >> The thing is: no matter what we do here, it's not going to look 100%
> >> like the stuff that came before it. All we can do is look at the
> >> products we know about, look at the existing published semantics for
> >> protocol elements, and make educated guesses about what  is likely to
> >> cause the fewest problems. We *know* that using the same
> > port numbers for multiple m-lines causes bad behavior in more than a
> > few endpoints, so that's out. On the other hand, the semantics of
> > port=0 are pretty well defined, elicit the exact response we want for
> > remote endpoints, and avoid egregiously large port allocations during
> > call setup (which is an important goal).
> >
> > The semantics of using port=0 are well defined, as you say - those
> > semantics are that you MUST NOT use the media.  You are changing those
> > semantics, if the far-end supports BUNDLE.  You feel it's safe to
> > change those semantics, because you feel that SDP is only being
> > processed by the local and remote endpoints.  I'm saying that's a
> > really bad assumption, because I know that in fact it is not the case.
> >
> >
> >
> >> I think that's a lot more solid set of reasons than the opposing
> rationale you proffer of "there may or may not be products somewhere in the
> world that don't really like this approach," not least of all because that
> statement is true of any approach we come  up with.
> >
> > Do you honestly believe that using a port number of 0 in an offer,
> > while the answer uses a port number of N, is equally impacting on
> > interoperability as using an attribute for capability discovery and
> > then a second O/A round of the real numbers?  C'mon man. You've seen
> > the stuff that's out there.  The SDP Offer and Answer's m-line port
> > number is used by all kinds of whacky equipment, to determine the
> > RTP/RTCP flow information.  It's used by passive monitoring systems,
> > NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and
> > god knows what else.  Even just in my own employer, we've got five
> > different types of equipment that look at the the m-line port numbers,
> > only one of which is our SBCs (and we've got multiple flavors of them
> too, fwiw).
> >
> > -hadriel
> >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--089e013a0270a06c0204e38c9d3a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Aug 9, 2013 at 4:36 PM, Christer Holmberg <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">ch=
rister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Justin,<br>
<div class=3D"im"><br>
&gt; Requiring a second offer to add the bundle-only m-lines defeats the pu=
rpose of bundle-only.<br>
<br>
</div>We already require a second offer. I know that some people, including=
 yourself, want to change that, but that is nothing we have agreed on.<br>
<div class=3D"im"><br>
&gt; a) The delay to do an O/A will often exceed the time to allocate candi=
dates, for small numbers of streams; bundle-only was meant to avoid this ca=
ndidate latency.<br>
<br>
</div>You do not have to allocate ports for the bundle-only m- lines.=C2=A0=
</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
&gt; b) Once you have the initial answer, you know whether the other side s=
upports BUNDLE, so you can take the appropriate action (and so there is no =
need for bundle-only).<br>
&gt;<br>
&gt; If SIP can&#39;t deal with the bundle-only concept, then WebRTC apps t=
hat care about latency will have to decide whether they are running in WebR=
TC-only mode or compatibility mode. In WebRTC-only mode, streams will be bu=
ndled in the initial offer; in compatibility mode, 2 offer-answers will be =
needed.<br>


<br>
</div>There are no &quot;modes&quot; in BUNDLE today :)<br></blockquote><di=
v><br></div><div>Of course there are. The handling in a reinvite case is di=
stinct from the initial offer, so that is modal behavior.=C2=A0</div><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">


<div class=3D"im"><br>
&gt; This would be a somewhat unfortunate outcome, so I&#39;d like to see b=
undle-only (or something like it) work.<br>
<br>
</div>I don&#39;t think the second offer is going to cause any latency. Als=
o, keep in mind that you only need the 2nd Offer/Answer during the session =
establishment. As agreed in Berlin, in mid-session changes you do NOT need =
2 offer/answers.<br>

</blockquote><div><br></div><div>Please provide support for this statement.=
 It seems clear that a 2nd offer adds _at least_ 1 RTT latency.=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">


<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
<br>
On Fri, Aug 9, 2013 at 12:49 PM, Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:=
<br>
Hi,<br>
,<br>
&gt;&gt; An alternative way, using BUNDLE as currently specified, is the ad=
d<br>
&gt;&gt; the bundle-only m- lines in a subsequent Offer, once BUNDLE usage =
has<br>
&gt;&gt; been negotiated. Then one does not need port zero. Of course, that=
<br>
&gt;&gt; would a &quot;third&quot; Offer (technically it could be done alre=
ady in the &quot;second&quot;<br>
&gt;&gt; Offer, but BUNDLE recommends against that).<br>
&gt;<br>
&gt; I quite like this, but prefer that it be allowed in the 2nd O/A.<br>
&gt; It is consistent with my desire to make many of these &quot;rules&quot=
; a bit more lenient.<br>
&gt;<br>
&gt; If done in the 2nd O/A, which is required anyway, then that should eli=
minate much (not all) of the objections that are likely to come from RTCWEB=
.<br>
Yes.<br>
<br>
And, the 2nd O/A should be fine, as the new bundle-only m- lines can be rej=
ected (using port zero in the Answer) without having to reject the whole Of=
fer (i.e. it won&#39;t affect the &quot;port synchronization&quot; of the o=
ther m- lines in the Offer).<br>


<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
<br>
<br>
<br>
<br>
&gt; Sent from */Windows/* using *TouchDown* (<a href=3D"http://www.nitrode=
sk.com" target=3D"_blank">www.nitrodesk.com</a>)<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; *From:* Hadriel Kaplan [<a href=3D"mailto:hadriel.kaplan@oracle.com">h=
adriel.kaplan@oracle.com</a>]<br>
&gt; *To:* Adam Roach [<a href=3D"mailto:adam@nostrum.com">adam@nostrum.com=
</a>]<br>
&gt; *CC:* Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.=
com">christer.holmberg@ericsson.com</a>]; mmusic<br>
&gt; [<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>]<br>
&gt; *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for<br>
&gt; bundle-only m- lines?<br>
&gt;<br>
&gt; On Aug 8, 2013, at 5:02 PM, Adam Roach &lt;<a href=3D"mailto:adam@nost=
rum.com">adam@nostrum.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt; On 8/8/13 15:51, Hadriel Kaplan wrote:<br>
&gt;&gt;&gt; BTW, I&#39;m not speaking about my employer&#39;s SBCs in part=
icular - I<br>
&gt;&gt;&gt; actually have no idea what we&#39;ll do with an UP bundle-only=
 thing<br>
&gt;&gt;&gt; lying about a port=3D0. =C2=A0We could handle it all correctly=
, or tear the<br>
&gt;&gt;&gt; call down later, or call WOPR and tell it to play Global<br>
&gt;&gt;&gt; Thermonuclear War. =C2=A0If something&#39;s broken in our impl=
ementation,<br>
&gt;&gt;&gt; we&#39;ll<br>
&gt; fix it... eventually. =C2=A0I&#39;m more worried about other systems, =
from<br>
&gt; vendors who don&#39;t attend the IETF; and I&#39;m worried about the p=
eople<br>
&gt; who have to run SIP networks, for whom this stuff is not a greenfield.=
<br>
&gt;&gt;<br>
&gt;&gt; The thing is: no matter what we do here, it&#39;s not going to loo=
k 100%<br>
&gt;&gt; like the stuff that came before it. All we can do is look at the<b=
r>
&gt;&gt; products we know about, look at the existing published semantics f=
or<br>
&gt;&gt; protocol elements, and make educated guesses about what =C2=A0is l=
ikely to<br>
&gt;&gt; cause the fewest problems. We *know* that using the same<br>
&gt; port numbers for multiple m-lines causes bad behavior in more than a<b=
r>
&gt; few endpoints, so that&#39;s out. On the other hand, the semantics of<=
br>
&gt; port=3D0 are pretty well defined, elicit the exact response we want fo=
r<br>
&gt; remote endpoints, and avoid egregiously large port allocations during<=
br>
&gt; call setup (which is an important goal).<br>
&gt;<br>
&gt; The semantics of using port=3D0 are well defined, as you say - those<b=
r>
&gt; semantics are that you MUST NOT use the media. =C2=A0You are changing =
those<br>
&gt; semantics, if the far-end supports BUNDLE. =C2=A0You feel it&#39;s saf=
e to<br>
&gt; change those semantics, because you feel that SDP is only being<br>
&gt; processed by the local and remote endpoints. =C2=A0I&#39;m saying that=
&#39;s a<br>
&gt; really bad assumption, because I know that in fact it is not the case.=
<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt; I think that&#39;s a lot more solid set of reasons than the opposi=
ng rationale you proffer of &quot;there may or may not be products somewher=
e in the world that don&#39;t really like this approach,&quot; not least of=
 all because that statement is true of any approach we come =C2=A0up with.<=
br>


&gt;<br>
&gt; Do you honestly believe that using a port number of 0 in an offer,<br>
&gt; while the answer uses a port number of N, is equally impacting on<br>
&gt; interoperability as using an attribute for capability discovery and<br=
>
&gt; then a second O/A round of the real numbers? =C2=A0C&#39;mon man. You&=
#39;ve seen<br>
&gt; the stuff that&#39;s out there. =C2=A0The SDP Offer and Answer&#39;s m=
-line port<br>
&gt; number is used by all kinds of whacky equipment, to determine the<br>
&gt; RTP/RTCP flow information. =C2=A0It&#39;s used by passive monitoring s=
ystems,<br>
&gt; NATs, CGNs, Firewalls, Routers, PDGs, x-CSCFs, SBCs, App servers, and<=
br>
&gt; god knows what else. =C2=A0Even just in my own employer, we&#39;ve got=
 five<br>
&gt; different types of equipment that look at the the m-line port numbers,=
<br>
&gt; only one of which is our SBCs (and we&#39;ve got multiple flavors of t=
hem too, fwiw).<br>
&gt;<br>
&gt; -hadriel<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br>
</div></div></blockquote></div><br></div></div>

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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
In-Reply-To: <CAOJ7v-2LpF6HGApc5V-AA-D=D=EOtFmErmPSrRPKVdM8kQYHtw@mail.gmail.com>
Date: Fri, 9 Aug 2013 20:13:07 -0400
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To: Justin Uberti <juberti@google.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 9, 2013, at 7:20 PM, Justin Uberti <juberti@google.com> wrote:

> Requiring a second offer to add the bundle-only m-lines defeats the =
purpose of bundle-only.
> a) The delay to do an O/A will often exceed the time to allocate =
candidates, for small numbers of streams; bundle-only was meant to avoid =
this candidate latency.

You lost me - I thought the "bundle-only" lines do not use their own =
transport, and instead require a regular "bundle" line to use its =
transport info.  Afaict, that means the bundle-only have no candidates.  =
No?  (or I could be misunderstanding your point)

We're really only talking about the bundle-only lines in this =
discussion, I think.


> b) Once you have the initial answer, you know whether the other side =
supports BUNDLE, so you can take the appropriate action (and so there is =
no need for bundle-only).
>=20

Christer's proposal is to send the regular m-lines, including regular =
bundle m-lines, in the initial Offer.  The Answer comes back for those, =
and tells you if the far-end does bundle.  You then send another Offer, =
with more *bundle* m-lines.  There is no "bundle-only" needed anymore.


> If SIP can't deal with the bundle-only concept, then WebRTC apps that =
care about latency will have to decide whether they are running in =
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams =
will be bundled in the initial offer; in compatibility mode, 2 =
offer-answers will be needed.

Sure, that's WebRTC's decision.  Certainly for a JS<->Browser API, doing =
2 offer-answers seems silly to me.

I don't know what you mean by "compatibility" mode.  What is a =
JS<->Browser API that's *not* "WebRTC-only" mode?  Shouldn't a Browser =
just always be in "WebRTC" mode?  If some specific *application* needs =
to do something else, I would think that's what JavaScript is for.

Wasn't the premise supposed to be WebRTC does the basic things by =
default in SDP to make simple things easy to write in JS; but lets the =
advanced applications be possible by munging SDP and doing other things =
in JS?  Surely an application developer who wishes to interface to =
SIP-land for more than just the basic audio+video calls can figure out =
how to write some Javascript to do a double offer/answer, no?

-hadriel


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From: Justin Uberti <juberti@google.com>
Date: Fri, 9 Aug 2013 17:42:03 -0700
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To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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--089e013a02701ae87404e38d2abc
Content-Type: text/plain; charset=UTF-8

On Fri, Aug 9, 2013 at 5:13 PM, Hadriel Kaplan <hadriel.kaplan@oracle.com>wrote:

>
> On Aug 9, 2013, at 7:20 PM, Justin Uberti <juberti@google.com> wrote:
>
> > Requiring a second offer to add the bundle-only m-lines defeats the
> purpose of bundle-only.
> > a) The delay to do an O/A will often exceed the time to allocate
> candidates, for small numbers of streams; bundle-only was meant to avoid
> this candidate latency.
>
> You lost me - I thought the "bundle-only" lines do not use their own
> transport, and instead require a regular "bundle" line to use its transport
> info.  Afaict, that means the bundle-only have no candidates.  No?  (or I
> could be misunderstanding your point)
>
> We're really only talking about the bundle-only lines in this discussion,
> I think.
>

I agree with what you are saying - I am just saying the latency effect of
the second O/A may be worse than the problem that bundle-only was trying to
solve.

>
>
> > b) Once you have the initial answer, you know whether the other side
> supports BUNDLE, so you can take the appropriate action (and so there is no
> need for bundle-only).
> >
>
> Christer's proposal is to send the regular m-lines, including regular
> bundle m-lines, in the initial Offer.  The Answer comes back for those, and
> tells you if the far-end does bundle.  You then send another Offer, with
> more *bundle* m-lines.  There is no "bundle-only" needed anymore.
>

Agree - there is no need for bundle-only with 2 O/As.

>
>
> > If SIP can't deal with the bundle-only concept, then WebRTC apps that
> care about latency will have to decide whether they are running in
> WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will
> be bundled in the initial offer; in compatibility mode, 2 offer-answers
> will be needed.
>
> Sure, that's WebRTC's decision.  Certainly for a JS<->Browser API, doing 2
> offer-answers seems silly to me.
>
> I don't know what you mean by "compatibility" mode.  What is a
> JS<->Browser API that's *not* "WebRTC-only" mode?  Shouldn't a Browser just
> always be in "WebRTC" mode?  If some specific *application* needs to do
> something else, I would think that's what JavaScript is for.
>
> Wasn't the premise supposed to be WebRTC does the basic things by default
> in SDP to make simple things easy to write in JS; but lets the advanced
> applications be possible by munging SDP and doing other things in JS?
>  Surely an application developer who wishes to interface to SIP-land for
> more than just the basic audio+video calls can figure out how to write some
> Javascript to do a double offer/answer, no?


There has been resistance to the idea that the app needs to decide if it's
interfacing with SIP-land or not. Which is understandable, but perhaps this
goal of app agnosticism is unattainable.

--089e013a02701ae87404e38d2abc
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Aug 9, 2013 at 5:13 PM, Hadriel Kaplan <span dir=3D"ltr">&l=
t;<a href=3D"mailto:hadriel.kaplan@oracle.com" target=3D"_blank">hadriel.ka=
plan@oracle.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
On Aug 9, 2013, at 7:20 PM, Justin Uberti &lt;<a href=3D"mailto:juberti@goo=
gle.com">juberti@google.com</a>&gt; wrote:<br>
<br>
&gt; Requiring a second offer to add the bundle-only m-lines defeats the pu=
rpose of bundle-only.<br>
&gt; a) The delay to do an O/A will often exceed the time to allocate candi=
dates, for small numbers of streams; bundle-only was meant to avoid this ca=
ndidate latency.<br>
<br>
</div>You lost me - I thought the &quot;bundle-only&quot; lines do not use =
their own transport, and instead require a regular &quot;bundle&quot; line =
to use its transport info. =C2=A0Afaict, that means the bundle-only have no=
 candidates. =C2=A0No? =C2=A0(or I could be misunderstanding your point)<br=
>


<br>
We&#39;re really only talking about the bundle-only lines in this discussio=
n, I think.<br></blockquote><div><br></div><div>I agree with what you are s=
aying - I am just saying the latency effect of the second O/A may be worse =
than the problem that bundle-only was trying to solve.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
<br>
&gt; b) Once you have the initial answer, you know whether the other side s=
upports BUNDLE, so you can take the appropriate action (and so there is no =
need for bundle-only).<br>
&gt;<br>
<br>
</div>Christer&#39;s proposal is to send the regular m-lines, including reg=
ular bundle m-lines, in the initial Offer. =C2=A0The Answer comes back for =
those, and tells you if the far-end does bundle. =C2=A0You then send anothe=
r Offer, with more *bundle* m-lines. =C2=A0There is no &quot;bundle-only&qu=
ot; needed anymore.<br>

</blockquote><div><br></div><div>Agree - there is no need for bundle-only w=
ith 2 O/As.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
<br>
&gt; If SIP can&#39;t deal with the bundle-only concept, then WebRTC apps t=
hat care about latency will have to decide whether they are running in WebR=
TC-only mode or compatibility mode. In WebRTC-only mode, streams will be bu=
ndled in the initial offer; in compatibility mode, 2 offer-answers will be =
needed.<br>


<br>
</div>Sure, that&#39;s WebRTC&#39;s decision. =C2=A0Certainly for a JS&lt;-=
&gt;Browser API, doing 2 offer-answers seems silly to me.<br>
<br>
I don&#39;t know what you mean by &quot;compatibility&quot; mode. =C2=A0Wha=
t is a JS&lt;-&gt;Browser API that&#39;s *not* &quot;WebRTC-only&quot; mode=
? =C2=A0Shouldn&#39;t a Browser just always be in &quot;WebRTC&quot; mode? =
=C2=A0If some specific *application* needs to do something else, I would th=
ink that&#39;s what JavaScript is for.<br>


<br>
Wasn&#39;t the premise supposed to be WebRTC does the basic things by defau=
lt in SDP to make simple things easy to write in JS; but lets the advanced =
applications be possible by munging SDP and doing other things in JS? =C2=
=A0Surely an application developer who wishes to interface to SIP-land for =
more than just the basic audio+video calls can figure out how to write some=
 Javascript to do a double offer/answer, no?</blockquote>

<div><br></div><div>There has been resistance to the idea that the app need=
s to decide if it&#39;s interfacing with SIP-land or not. Which is understa=
ndable, but perhaps this goal of app agnosticism is unattainable.</div>

</div></div></div>

--089e013a02701ae87404e38d2abc--

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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Matthew Kaufman'" <matthew@matthew.at>, <mmusic@ietf.org>
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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Matthew,

I like your argument of "shouldn't be reusing *this* standard for *that*
application". I have already mentioned in draft-partha-rtcweb-jsep-sip-00
that SDP usage in WebRTC is making the complication in interworking with
SIP. The current Unified plan (UP) is adding fuel to the fire. In case UP
plan is not going to work with SIP/RFC 3264 based mechanism, I agree with
your statement of "we're doing this "standards" thing wrong". 

I started thinking the standards are going wrong after the current UP
discussion as 

1) Some of the web developers are already not interested in current WebRTC
SDP (last month Public-webrtc discussion)
2) WebRTC SDP will not be suitable for future SIP based devices as well and
not just legacy PSTN/PLMN/Enterprise SIP devices.
3) My guess from your words is that WebRTC SDP is not so useful/flexible for
proprietary VoIP apps like Skype. 
4) I'm getting the feeling that WebRTC SDP mechanism will work only for
Jingle (XMPP based) sort of mechanism and not for anything else.
(Disclaimer: I don't know much about Jingle).

Having said that, I guess that UP will be updated to work at least with RFC
3264 (offer/answer) requirements and WebRTC SDP shall be reused for SIP
application. 

Thanks
Partha

PS: CU-WebRTC kind of WebRTC API will be more useful to interop with SIP
than current UP based WebRTC API as SDP required for SIP shall be built
using JavaScript/another W3C API in case CU-WebRTC kind of API exists. 

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Matthew Kaufman
> Sent: Saturday, August 10, 2013 4:41 AM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
> 
> On 8/9/2013 3:35 PM, Justin Uberti wrote:
> > That is not my understanding. If we don't care about intermediaries,
> > we could have made BUNDLE and UP a lot simpler.
> >
> > I think the expectation should be that Matthew's proposed case will
> > work, modulo any implementation bugs on the remote device.
> >
> 
> 
> I agree. Otherwise we're doing this "standards" thing wrong OR
> shouldn't
> be reusing *this* standard for *that* application.
> 
> Matthew Kaufman
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Sat Aug 10 15:15:26 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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From christer.holmberg@ericsson.com  Sat Aug 10 15:24:45 2013
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Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 10 Aug 2013 18:52:04 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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--047d7b5da905a447b004e3a243fb
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On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>> Requiring a second offer to add the bundle-only m-lines defeats the
> purpose of bundle-only.
> >> We already require a second offer. I know that some people, including
> yourself, want to change that, but that is nothing we have agreed on.
> >>
> >>> a) The delay to do an O/A will often exceed the time to allocate
> candidates, for small numbers of streams; bundle-only was meant to avoid
> this candidate latency.
> >> You do not have to allocate ports for the bundle-only m- lines.
> >>
> >>> b) Once you have the initial answer, you know whether the other side
> supports BUNDLE, so you can take the appropriate action (and so there is no
> need for bundle-only).
> >>>
> >>> If SIP can't deal with the bundle-only concept, then WebRTC apps that
> care about latency will have to decide whether they are running in
> WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will
> be bundled in the initial offer; in compatibility mode, 2 offer-answers
> will be needed.
> >> There are no "modes" in BUNDLE today :)
> >>
> > Of course there are. The handling in a reinvite case is distinct from
> the initial offer, so that is modal behavior.
>
> Sure, but you were talking about "WebRTC-only mode" and "compatibility
> mode", and my point is that currently the procedures for initial offers,
> subsequent offers etc are the same.
>
> >>> This would be a somewhat unfortunate outcome, so I'd like to see
> bundle-only (or something like it) work.
> >> I don't think the second offer is going to cause any latency. Also,
> keep in mind that you only need the 2nd Offer/Answer during the session
> establishment. As agreed in Berlin, in mid-session changes you do NOT need
> 2 offer/answers.
> >
> > Please provide support for this statement. It seems clear that a 2nd
> offer adds _at least_ 1 RTT latency.
>
> Sure, but I don't think that is going to be a practical problem.
>

Well, I'd like to see a more convincing argument than "some middleboxes are
sad" when you
do that.


First, there are deployed legacy SIP systems that use even more O/A
> transactions during session establishment. And, in your case I assume you
> are even going to use something more "lightweight" than SIP to carry your
> Offer/Answers.
>
> Second, as WebRTC mandates usage of ICE, you are often going to need a
> subsequent O/A anyway.
>

It's important to distinguish between round-trips that are in the critical
path for media
flowing and those which are not.

-Ekr

--047d7b5da905a447b004e3a243fb
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div class=3D"im"><br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines defeat=
s the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, includ=
ing yourself, want to change that, but that is nothing we have agreed on.<b=
r>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to alloca=
te candidates, for small numbers of streams; bundle-only was meant to avoid=
 this candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- lines.=A0=
<br>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the othe=
r side supports BUNDLE, so you can take the appropriate action (and so ther=
e is no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can&#39;t deal with the bundle-only concept, then WebRT=
C apps that care about latency will have to decide whether they are running=
 in WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams wi=
ll be bundled in the initial offer; in compatibility mode, 2 offer-answers =
will be needed.<br>


&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct from =
the initial offer, so that is modal behavior.=A0<br>
<br>
</div>Sure, but you were talking about &quot;WebRTC-only mode&quot; and &qu=
ot;compatibility mode&quot;, and my point is that currently the procedures =
for initial offers, subsequent offers etc are the same.<br>
<div class=3D"im"><br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I&#39;d like =
to see bundle-only (or something like it) work.<br>
&gt;&gt; I don&#39;t think the second offer is going to cause any latency. =
Also, keep in mind that you only need the 2nd Offer/Answer during the sessi=
on establishment. As agreed in Berlin, in mid-session changes you do NOT ne=
ed 2 offer/answers.<br>


&gt;<br>
&gt; Please provide support for this statement. It seems clear that a 2nd o=
ffer adds _at least_ 1 RTT latency.=A0<br>
<br>
</div>Sure, but I don&#39;t think that is going to be a practical problem.<=
br></blockquote><div><br></div><div>Well, I&#39;d like to see a more convin=
cing argument than &quot;some middleboxes are sad&quot; when you</div>

<div>do that.</div><div><br></div><div><br></div><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more &quot;lightweight&quot; than SIP to carry your=
 Offer/Answers.<br>


<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.<br></blockquote><div><br></div><div>It&#39;s important t=
o distinguish between round-trips that are in the critical path for media</=
div>

<div>flowing and those which are not.</div><div><br></div><div>-Ekr</div><d=
iv><br></div></div></div></div>

--047d7b5da905a447b004e3a243fb--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: "mmusic_ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

I think making sure all entities have the correct port numbers is critical =
in making sure the media will flow.

Regards,

Christer




Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Eric Rescorla [ekr@rtfm.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org]; =
Paul Kyzivat [pkyzivat@alum.mit.edu]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?



On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

>>> Requiring a second offer to add the bundle-only m-lines defeats the pur=
pose of bundle-only.
>> We already require a second offer. I know that some people, including yo=
urself, want to change that, but that is nothing we have agreed on.
>>
>>> a) The delay to do an O/A will often exceed the time to allocate candid=
ates, for small numbers of streams; bundle-only was meant to avoid this can=
didate latency.
>> You do not have to allocate ports for the bundle-only m- lines.
>>
>>> b) Once you have the initial answer, you know whether the other side su=
pports BUNDLE, so you can take the appropriate action (and so there is no n=
eed for bundle-only).
>>>
>>> If SIP can't deal with the bundle-only concept, then WebRTC apps that c=
are about latency will have to decide whether they are running in WebRTC-on=
ly mode or compatibility mode. In WebRTC-only mode, streams will be bundled=
 in the initial offer; in compatibility mode, 2 offer-answers will be neede=
d.
>> There are no "modes" in BUNDLE today :)
>>
> Of course there are. The handling in a reinvite case is distinct from the=
 initial offer, so that is modal behavior.

Sure, but you were talking about "WebRTC-only mode" and "compatibility mode=
", and my point is that currently the procedures for initial offers, subseq=
uent offers etc are the same.

>>> This would be a somewhat unfortunate outcome, so I'd like to see bundle=
-only (or something like it) work.
>> I don't think the second offer is going to cause any latency. Also, keep=
 in mind that you only need the 2nd Offer/Answer during the session establi=
shment. As agreed in Berlin, in mid-session changes you do NOT need 2 offer=
/answers.
>
> Please provide support for this statement. It seems clear that a 2nd offe=
r adds _at least_ 1 RTT latency.

Sure, but I don't think that is going to be a practical problem.

Well, I'd like to see a more convincing argument than "some middleboxes are=
 sad" when you
do that.


First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more "lightweight" than SIP to carry your Offer/Ans=
wers.

Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.

It's important to distinguish between round-trips that are in the critical =
path for media
flowing and those which are not.

-Ekr


--_000_7594FB04B1934943A5C02806D1A2204B1C43DBA9ESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body>
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div>I think making sure all entities have the correct port numbers is crit=
ical in making sure the media will flow.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Eric Rescorla [ekr@rtfm.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf=
.org]; Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</span>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
Hi,<br>
<div class=3D"im"><br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines defeat=
s the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, includ=
ing yourself, want to change that, but that is nothing we have agreed on.<b=
r>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to alloca=
te candidates, for small numbers of streams; bundle-only was meant to avoid=
 this candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- lines.&nb=
sp;<br>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the othe=
r side supports BUNDLE, so you can take the appropriate action (and so ther=
e is no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can't deal with the bundle-only concept, then WebRTC ap=
ps that care about latency will have to decide whether they are running in =
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will b=
e bundled in the initial offer; in compatibility
 mode, 2 offer-answers will be needed.<br>
&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct from =
the initial offer, so that is modal behavior.&nbsp;<br>
<br>
</div>
Sure, but you were talking about &quot;WebRTC-only mode&quot; and &quot;com=
patibility mode&quot;, and my point is that currently the procedures for in=
itial offers, subsequent offers etc are the same.<br>
<div class=3D"im"><br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I'd like to s=
ee bundle-only (or something like it) work.<br>
&gt;&gt; I don't think the second offer is going to cause any latency. Also=
, keep in mind that you only need the 2nd Offer/Answer during the session e=
stablishment. As agreed in Berlin, in mid-session changes you do NOT need 2=
 offer/answers.<br>
&gt;<br>
&gt; Please provide support for this statement. It seems clear that a 2nd o=
ffer adds _at least_ 1 RTT latency.&nbsp;<br>
<br>
</div>
Sure, but I don't think that is going to be a practical problem.<br>
</blockquote>
<div><br>
</div>
<div>Well, I'd like to see a more convincing argument than &quot;some middl=
eboxes are sad&quot; when you</div>
<div>do that.</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more &quot;lightweight&quot; than SIP to carry your=
 Offer/Answers.<br>
<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.<br>
</blockquote>
<div><br>
</div>
<div>It's important to distinguish between round-trips that are in the crit=
ical path for media</div>
<div>flowing and those which are not.</div>
<div><br>
</div>
<div>-Ekr</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Sun, 11 Aug 2013 04:15:49 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Sun, Aug 11, 2013 at 12:39 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
> I think making sure all entities have the correct port numbers is critical
> in making sure the media will flow.
>

There is a difference between "making sure the media will flow" and "before
media can
flow at all". For instance, Neither Firefox nor Chrome current does a
second O/A
transaction and yet media flows just fine in most cases, since they are
operating
in an unmediated environment. Obviously, this is a best-case scenario, but
similarly
there are going to be mediated environments which are perfectly happy with
a second
O/A exchange with non-zero ports.

This argues against designing a system in which none of the secondary media
flows
without a second O/A exchange.

-Ekr


-----Original Message-----
> *From:* Eric Rescorla [ekr@rtfm.com]
> *To:* Christer Holmberg [christer.holmberg@ericsson.com]
> *CC:* Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org];
> Paul Kyzivat [pkyzivat@alum.mit.edu]
> *Subject:* Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
> bundle-only m- lines?
>
>
>
> On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> >>> Requiring a second offer to add the bundle-only m-lines defeats the
>> purpose of bundle-only.
>> >> We already require a second offer. I know that some people, including
>> yourself, want to change that, but that is nothing we have agreed on.
>> >>
>> >>> a) The delay to do an O/A will often exceed the time to allocate
>> candidates, for small numbers of streams; bundle-only was meant to avoid
>> this candidate latency.
>> >> You do not have to allocate ports for the bundle-only m- lines.
>> >>
>> >>> b) Once you have the initial answer, you know whether the other side
>> supports BUNDLE, so you can take the appropriate action (and so there is no
>> need for bundle-only).
>> >>>
>> >>> If SIP can't deal with the bundle-only concept, then WebRTC apps that
>> care about latency will have to decide whether they are running in
>> WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will
>> be bundled in the initial offer; in compatibility mode, 2 offer-answers
>> will be needed.
>> >> There are no "modes" in BUNDLE today :)
>> >>
>> > Of course there are. The handling in a reinvite case is distinct from
>> the initial offer, so that is modal behavior.
>>
>>  Sure, but you were talking about "WebRTC-only mode" and "compatibility
>> mode", and my point is that currently the procedures for initial offers,
>> subsequent offers etc are the same.
>>
>> >>> This would be a somewhat unfortunate outcome, so I'd like to see
>> bundle-only (or something like it) work.
>> >> I don't think the second offer is going to cause any latency. Also,
>> keep in mind that you only need the 2nd Offer/Answer during the session
>> establishment. As agreed in Berlin, in mid-session changes you do NOT need
>> 2 offer/answers.
>> >
>> > Please provide support for this statement. It seems clear that a 2nd
>> offer adds _at least_ 1 RTT latency.
>>
>>  Sure, but I don't think that is going to be a practical problem.
>>
>
>  Well, I'd like to see a more convincing argument than "some middleboxes
> are sad" when you
> do that.
>
>
>  First, there are deployed legacy SIP systems that use even more O/A
>> transactions during session establishment. And, in your case I assume you
>> are even going to use something more "lightweight" than SIP to carry your
>> Offer/Answers.
>>
>> Second, as WebRTC mandates usage of ICE, you are often going to need a
>> subsequent O/A anyway.
>>
>
>  It's important to distinguish between round-trips that are in the
> critical path for media
> flowing and those which are not.
>
>  -Ekr
>
>

--001a11c2318eabcde404e3aa23c7
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">On Sun, Aug 11, 2013 at 12:39 AM, Christer Holmberg <span =
dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D=
"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<br><div class=
=3D"gmail_extra">

<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">



<div>
<div><span style=3D"font-size:11pt;font-family:Calibri,Arial,Helvetica,sans=
-serif">Hi,</span></div>
<div>=A0</div>
<div>I think making sure all entities have the correct port numbers is crit=
ical in making sure the media will flow.</div></div></blockquote><div><br><=
/div><div>There is a difference between &quot;making sure the media will fl=
ow&quot; and &quot;before media can</div>

<div>flow at all&quot;. For instance, Neither Firefox nor Chrome current do=
es a second O/A</div><div>transaction and yet media flows just fine in most=
 cases, since they are operating</div><div>in an unmediated environment. Ob=
viously, this is a best-case scenario, but similarly</div>

<div>there are going to be mediated environments which are perfectly happy =
with a second</div><div>O/A exchange with non-zero ports.</div><div><br></d=
iv><div>This argues against designing a system in which none of the seconda=
ry media flows</div>

<div>without a second O/A exchange.</div><div><br></div><div>-Ekr<br></div>=
<div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><span sty=
le><div class=3D"im">

-----Original Message-----<br>
<b>From:</b> Eric Rescorla [<a href=3D"mailto:ekr@rtfm.com" target=3D"_blan=
k">ekr@rtfm.com</a>]<br>
<b>To:</b> Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.=
com" target=3D"_blank">christer.holmberg@ericsson.com</a>]<br>
</div><div><div class=3D"h5"><b>CC:</b> Justin Uberti [<a href=3D"mailto:ju=
berti@google.com" target=3D"_blank">juberti@google.com</a>]; <a href=3D"mai=
lto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a> [<a href=3D"mail=
to:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a>]; Paul Kyzivat [<=
a href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit=
.edu</a>]<br>


<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundl=
e-only m- lines?</div></div></span><div><div class=3D"h5">
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmbe=
rg <span dir=3D"ltr">
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chr=
ister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi,<br>
<div><br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines defeat=
s the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, includ=
ing yourself, want to change that, but that is nothing we have agreed on.<b=
r>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to alloca=
te candidates, for small numbers of streams; bundle-only was meant to avoid=
 this candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- lines.=A0=
<br>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the othe=
r side supports BUNDLE, so you can take the appropriate action (and so ther=
e is no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can&#39;t deal with the bundle-only concept, then WebRT=
C apps that care about latency will have to decide whether they are running=
 in WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams wi=
ll be bundled in the initial offer; in compatibility
 mode, 2 offer-answers will be needed.<br>
&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct from =
the initial offer, so that is modal behavior.=A0<br>
<br>
</div>
Sure, but you were talking about &quot;WebRTC-only mode&quot; and &quot;com=
patibility mode&quot;, and my point is that currently the procedures for in=
itial offers, subsequent offers etc are the same.<br>
<div><br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I&#39;d like =
to see bundle-only (or something like it) work.<br>
&gt;&gt; I don&#39;t think the second offer is going to cause any latency. =
Also, keep in mind that you only need the 2nd Offer/Answer during the sessi=
on establishment. As agreed in Berlin, in mid-session changes you do NOT ne=
ed 2 offer/answers.<br>


&gt;<br>
&gt; Please provide support for this statement. It seems clear that a 2nd o=
ffer adds _at least_ 1 RTT latency.=A0<br>
<br>
</div>
Sure, but I don&#39;t think that is going to be a practical problem.<br>
</blockquote>
<div><br>
</div>
<div>Well, I&#39;d like to see a more convincing argument than &quot;some m=
iddleboxes are sad&quot; when you</div>
<div>do that.</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more &quot;lightweight&quot; than SIP to carry your=
 Offer/Answers.<br>


<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.<br>
</blockquote>
<div><br>
</div>
<div>It&#39;s important to distinguish between round-trips that are in the =
critical path for media</div>
<div>flowing and those which are not.</div>
<div><br>
</div>
<div>-Ekr</div>
<div><br>
</div>
</div>
</div>
</div>
</div>
</div></div></div>

</blockquote></div><br></div></div>

--001a11c2318eabcde404e3aa23c7--

From christer.holmberg@ericsson.com  Sun Aug 11 06:16:42 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: "mmusic_ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

>> I think making sure all entities have the correct port numbers is critic=
al in making sure the media will flow.
>
> There is a difference between "making sure the media will flow" and "befo=
re media can
> flow at all".
> For instance, Neither Firefox nor Chrome current does a second O/A
> transaction and yet media flows just fine in most cases, since they are o=
perating
> in an unmediated environment. Obviously, this is a best-case scenario, bu=
t similarly
> there are going to be mediated environments which are perfectly happy wit=
h a second
> O/A exchange with non-zero ports.

There are also environements where a single offer, with identical port numb=
ers, work just fine, because the receiver will be able to handle it even if=
 it doesn't support BUNDLE.

However, we are defining a generic mechanism, supposed to work in different=
 environments.=20

But, sure, if the community thinks that we shall define different options, =
depending on environments etc, we can do that...=20

Regards,

Christer




-----Original Message-----
From: Eric Rescorla [ekr@rtfm.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]

CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org]; =
Paul Kyzivat [pkyzivat@alum.mit.edu]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?





On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <christer.holmberg@erics=
son.com> wrote:

Hi,


>>> Requiring a second offer to add the bundle-only m-lines defeats the pur=
pose of bundle-only.
>> We already require a second offer. I know that some people, including yo=
urself, want to change that, but that is nothing we have agreed on.
>>
>>> a) The delay to do an O/A will often exceed the time to allocate candid=
ates, for small numbers of streams; bundle-only was meant to avoid this can=
didate latency.
>> You do not have to allocate ports for the bundle-only m- lines.=20
>>
>>> b) Once you have the initial answer, you know whether the other side su=
pports BUNDLE, so you can take the appropriate action (and so there is no n=
eed for bundle-only).
>>>
>>> If SIP can't deal with the bundle-only concept, then WebRTC apps that c=
are about latency will have to decide whether they are running in WebRTC-on=
ly mode or compatibility mode. In WebRTC-only mode, streams will be bundled=
 in the initial offer; in compatibility mode, 2 offer-answers will be neede=
d.
>> There are no "modes" in BUNDLE today :)
>>
> Of course there are. The handling in a reinvite case is distinct from the=
 initial offer, so that is modal behavior.=20


Sure, but you were talking about "WebRTC-only mode" and "compatibility mode=
", and my point is that currently the procedures for initial offers, subseq=
uent offers etc are the same.


>>> This would be a somewhat unfortunate outcome, so I'd like to see bundle=
-only (or something like it) work.
>> I don't think the second offer is going to cause any latency. Also, keep=
 in mind that you only need the 2nd Offer/Answer during the session establi=
shment. As agreed in Berlin, in mid-session changes you do NOT need 2 offer=
/answers.
>
> Please provide support for this statement. It seems clear that a 2nd offe=
r adds _at least_ 1 RTT latency.=20


Sure, but I don't think that is going to be a practical problem.



Well, I'd like to see a more convincing argument than "some middleboxes are=
 sad" when you
do that.




First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more "lightweight" than SIP to carry your Offer/Ans=
wers.

Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.



It's important to distinguish between round-trips that are in the critical =
path for media
flowing and those which are not.


-Ekr=

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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 11, 2013, at 7:15 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> There is a difference between "making sure the media will flow" and =
"before media can
> flow at all". For instance, Neither Firefox nor Chrome current does a =
second O/A
> transaction and yet media flows just fine in most cases, since they =
are operating
> in an unmediated environment. Obviously, this is a best-case scenario, =
but similarly
> there are going to be mediated environments which are perfectly happy =
with a second
> O/A exchange with non-zero ports.
>=20
> This argues against designing a system in which none of the secondary =
media flows
> without a second O/A exchange.

Actually I think it argues against designing a system which treats an =
Application Programming Interface the same as an on-the-wire =
Internet-wide negotiation mechanism, with a deployed infrastructure that =
has been growing larger for the past 10 years.  It also argues against =
tying the fate of WebRTC to SIP, with regards to SDP O/A.

One thing we could do, for example, is define BUNDLE as we've been =
describing (which is essentially no change, afaik) and define its usage =
for SDP O/A as we've been describing.  But we put a "unless the sender =
has specific knowledge otherwise, as it could in [jsep] for example".  =
Or something like that.  Then the JSEP draft could say what to do for =
WebRTC usage.

Take a step back and ignore this particular port-numbering issue for a =
second: in what context would it make sense for an API to need two =
separate rounds of "Offer/Answer" between the Javascript and the =
Browser?  I.e., if we weren't using SDP at all, and had never heard of =
RFC 3264, and just used a native Javascript API... what use-cases would =
require the JS script and Browser to communicate multiple times between =
each other for the same action?  I think error cases might, for example. =
 And I think if the JS hooked into a callback for ICE transport =
selection changes, then the Browser would need to tell it when those =
things changed, for example.  Can you think of any others?

-hadriel


From ekr@rtfm.com  Sun Aug 11 07:57:42 2013
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sun, 11 Aug 2013 07:50:41 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic_ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> I think making sure all entities have the correct port numbers is
> critical in making sure the media will flow.
> >
> > There is a difference between "making sure the media will flow" and
> "before media can
> > flow at all".
> > For instance, Neither Firefox nor Chrome current does a second O/A
> > transaction and yet media flows just fine in most cases, since they are
> operating
> > in an unmediated environment. Obviously, this is a best-case scenario,
> but similarly
> > there are going to be mediated environments which are perfectly happy
> with a second
> > O/A exchange with non-zero ports.
>
> There are also environements where a single offer, with identical port
> numbers, work just fine, because the receiver will be able to handle it
> even if it doesn't support BUNDLE.
>
> However, we are defining a generic mechanism, supposed to work in
> different environments.
>

"different environments" != "arbitrary lossage by any middlebox in the
universe"

Again, you need to distinguish between *endpoints* and filtering
middleboxes.



> But, sure, if the community thinks that we shall define different options,
> depending on environments etc, we can do that...
>

Which is not what I said.

-Ekr

Regards,
>
> Christer
>
>
>
>
> -----Original Message-----
> From: Eric Rescorla [ekr@rtfm.com]
> To: Christer Holmberg [christer.holmberg@ericsson.com]
>
> CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org];
> Paul Kyzivat [pkyzivat@alum.mit.edu]
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
> m- lines?
>
>
>
>
>
> On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
>
> >>> Requiring a second offer to add the bundle-only m-lines defeats the
> purpose of bundle-only.
> >> We already require a second offer. I know that some people, including
> yourself, want to change that, but that is nothing we have agreed on.
> >>
> >>> a) The delay to do an O/A will often exceed the time to allocate
> candidates, for small numbers of streams; bundle-only was meant to avoid
> this candidate latency.
> >> You do not have to allocate ports for the bundle-only m- lines.
> >>
> >>> b) Once you have the initial answer, you know whether the other side
> supports BUNDLE, so you can take the appropriate action (and so there is no
> need for bundle-only).
> >>>
> >>> If SIP can't deal with the bundle-only concept, then WebRTC apps that
> care about latency will have to decide whether they are running in
> WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will
> be bundled in the initial offer; in compatibility mode, 2 offer-answers
> will be needed.
> >> There are no "modes" in BUNDLE today :)
> >>
> > Of course there are. The handling in a reinvite case is distinct from
> the initial offer, so that is modal behavior.
>
>
> Sure, but you were talking about "WebRTC-only mode" and "compatibility
> mode", and my point is that currently the procedures for initial offers,
> subsequent offers etc are the same.
>
>
> >>> This would be a somewhat unfortunate outcome, so I'd like to see
> bundle-only (or something like it) work.
> >> I don't think the second offer is going to cause any latency. Also,
> keep in mind that you only need the 2nd Offer/Answer during the session
> establishment. As agreed in Berlin, in mid-session changes you do NOT need
> 2 offer/answers.
> >
> > Please provide support for this statement. It seems clear that a 2nd
> offer adds _at least_ 1 RTT latency.
>
>
> Sure, but I don't think that is going to be a practical problem.
>
>
>
> Well, I'd like to see a more convincing argument than "some middleboxes
> are sad" when you
> do that.
>
>
>
>
> First, there are deployed legacy SIP systems that use even more O/A
> transactions during session establishment. And, in your case I assume you
> are even going to use something more "lightweight" than SIP to carry your
> Offer/Answers.
>
> Second, as WebRTC mandates usage of ICE, you are often going to need a
> subsequent O/A anyway.
>
>
>
> It's important to distinguish between round-trips that are in the critical
> path for media
> flowing and those which are not.
>
>
> -Ekr
>

--047d7b6dc1201ef7a004e3ad24fa
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg <span dir=3D"ltr=
">&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">c=
hrister.holmberg@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">Hi,<br>
<br>
&gt;&gt; I think making sure all entities have the correct port numbers is =
critical in making sure the media will flow.<br>
&gt;<br>
&gt; There is a difference between &quot;making sure the media will flow&qu=
ot; and &quot;before media can<br>
&gt; flow at all&quot;.<br>
&gt; For instance, Neither Firefox nor Chrome current does a second O/A<br>
&gt; transaction and yet media flows just fine in most cases, since they ar=
e operating<br>
&gt; in an unmediated environment. Obviously, this is a best-case scenario,=
 but similarly<br>
&gt; there are going to be mediated environments which are perfectly happy =
with a second<br>
&gt; O/A exchange with non-zero ports.<br>
<br>
</div>There are also environements where a single offer, with identical por=
t numbers, work just fine, because the receiver will be able to handle it e=
ven if it doesn&#39;t support BUNDLE.<br>
<br>
However, we are defining a generic mechanism, supposed to work in different=
 environments.<br></blockquote><div><br></div><div>&quot;different environm=
ents&quot; !=3D &quot;arbitrary lossage by any middlebox in the universe&qu=
ot;</div>

<div><br></div><div>Again, you need to distinguish between *endpoints* and =
filtering middleboxes.</div><div><br></div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">


But, sure, if the community thinks that we shall define different options, =
depending on environments etc, we can do that...<br></blockquote><div><br><=
/div><div>Which is not what I said.</div><div><br></div><div>-Ekr</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
<br>
-----Original Message-----<br>
From: Eric Rescorla [<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>]<br>
To: Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.com">ch=
rister.holmberg@ericsson.com</a>]<br>
<br>
CC: Justin Uberti [<a href=3D"mailto:juberti@google.com">juberti@google.com=
</a>]; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a> [<a href=3D"m=
ailto:mmusic@ietf.org">mmusic@ietf.org</a>]; Paul Kyzivat [<a href=3D"mailt=
o:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>]<br>


Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?<br>
<br>
<br>
<br>
<br>
<br>
On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg &lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:=
<br>
<br>
Hi,<br>
<br>
<br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines defeat=
s the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, includ=
ing yourself, want to change that, but that is nothing we have agreed on.<b=
r>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to alloca=
te candidates, for small numbers of streams; bundle-only was meant to avoid=
 this candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- lines.<br=
>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the othe=
r side supports BUNDLE, so you can take the appropriate action (and so ther=
e is no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can&#39;t deal with the bundle-only concept, then WebRT=
C apps that care about latency will have to decide whether they are running=
 in WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams wi=
ll be bundled in the initial offer; in compatibility mode, 2 offer-answers =
will be needed.<br>


&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct from =
the initial offer, so that is modal behavior.<br>
<br>
<br>
Sure, but you were talking about &quot;WebRTC-only mode&quot; and &quot;com=
patibility mode&quot;, and my point is that currently the procedures for in=
itial offers, subsequent offers etc are the same.<br>
<br>
<br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I&#39;d like =
to see bundle-only (or something like it) work.<br>
&gt;&gt; I don&#39;t think the second offer is going to cause any latency. =
Also, keep in mind that you only need the 2nd Offer/Answer during the sessi=
on establishment. As agreed in Berlin, in mid-session changes you do NOT ne=
ed 2 offer/answers.<br>


&gt;<br>
&gt; Please provide support for this statement. It seems clear that a 2nd o=
ffer adds _at least_ 1 RTT latency.<br>
<br>
<br>
Sure, but I don&#39;t think that is going to be a practical problem.<br>
<br>
<br>
<br>
Well, I&#39;d like to see a more convincing argument than &quot;some middle=
boxes are sad&quot; when you<br>
do that.<br>
<br>
<br>
<br>
<br>
First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more &quot;lightweight&quot; than SIP to carry your=
 Offer/Answers.<br>


<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.<br>
<br>
<br>
<br>
It&#39;s important to distinguish between round-trips that are in the criti=
cal path for media<br>
flowing and those which are not.<br>
<br>
<br>
-Ekr</div></div></blockquote></div><br></div></div>

--047d7b6dc1201ef7a004e3ad24fa--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: "mmusic_ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

>>>> I think making sure all entities have the correct port numbers is crit=
ical in making sure the media will flow.
>>>
>>> There is a difference between "making sure the media will flow" and "be=
fore media can
>>> flow at all".
>>> For instance, Neither Firefox nor Chrome current does a second O/A
>>> transaction and yet media flows just fine in most cases, since they are=
 operating
>>> in an unmediated environment. Obviously, this is a best-case scenario, =
but similarly
>>> there are going to be mediated environments which are perfectly happy w=
ith a second
>>> O/A exchange with non-zero ports.
>> There are also environements where a single offer, with identical port n=
umbers, work just fine, because the receiver will be able to handle it even=
 if it doesn't support BUNDLE.
>>
>> However, we are defining a generic mechanism, supposed to work in differ=
ent environments.
>
> "different environments" !=3D "arbitrary lossage by any middlebox in the =
universe"
>
> Again, you need to distinguish between *endpoints* and filtering middlebo=
xes.

It depends on what you mean by "filtering". If a middlebox removes stuff fr=
om the signaling, there is nothing we can do about that.

The most simple case I am thinking about is a middlebox forwarding media ba=
sed on information in the SDP.

> But, sure, if the community thinks that we shall define different options=
, depending on environments etc, we can do that...
>
> Which is not what I said.

You did talk about different environments. Wouldn't we need to define the p=
rocedures for each environment?

Regards,

Christer



-----Original Message-----
From: Eric Rescorla [ekr@rtfm.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]

CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org]; =
Paul Kyzivat [pkyzivat@alum.mit.edu]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only =
m- lines?





On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg <christer.holmberg@erics=
son.com> wrote:

Hi,


>>> Requiring a second offer to add the bundle-only m-lines defeats the pur=
pose of bundle-only.
>> We already require a second offer. I know that some people, including yo=
urself, want to change that, but that is nothing we have agreed on.
>>
>>> a) The delay to do an O/A will often exceed the time to allocate candid=
ates, for small numbers of streams; bundle-only was meant to avoid this can=
didate latency.
>> You do not have to allocate ports for the bundle-only m- lines.
>>
>>> b) Once you have the initial answer, you know whether the other side su=
pports BUNDLE, so you can take the appropriate action (and so there is no n=
eed for bundle-only).
>>>
>>> If SIP can't deal with the bundle-only concept, then WebRTC apps that c=
are about latency will have to decide whether they are running in WebRTC-on=
ly mode or compatibility mode. In WebRTC-only mode, streams will be bundled=
 in the initial offer; in compatibility mode, 2 offer-answers will be neede=
d.
>> There are no "modes" in BUNDLE today :)
>>
> Of course there are. The handling in a reinvite case is distinct from the=
 initial offer, so that is modal behavior.


Sure, but you were talking about "WebRTC-only mode" and "compatibility mode=
", and my point is that currently the procedures for initial offers, subseq=
uent offers etc are the same.


>>> This would be a somewhat unfortunate outcome, so I'd like to see bundle=
-only (or something like it) work.
>> I don't think the second offer is going to cause any latency. Also, keep=
 in mind that you only need the 2nd Offer/Answer during the session establi=
shment. As agreed in Berlin, in mid-session changes you do NOT need 2 offer=
/answers.
>
> Please provide support for this statement. It seems clear that a 2nd offe=
r adds _at least_ 1 RTT latency.


Sure, but I don't think that is going to be a practical problem.



Well, I'd like to see a more convincing argument than "some middleboxes are=
 sad" when you
do that.




First, there are deployed legacy SIP systems that use even more O/A transac=
tions during session establishment. And, in your case I assume you are even=
 going to use something more "lightweight" than SIP to carry your Offer/Ans=
wers.

Second, as WebRTC mandates usage of ICE, you are often going to need a subs=
equent O/A anyway.



It's important to distinguish between round-trips that are in the critical =
path for media
flowing and those which are not.


-Ekr


From christer.holmberg@ericsson.com  Sun Aug 11 09:43:21 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>, Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Cc: "mmusic_ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Hi,

>> There is a difference between "making sure the media will flow" and=20
>> "before media can flow at all". For instance, Neither Firefox nor=20
>> Chrome current does a second O/A transaction and yet media flows just=20
>> fine in most cases, since they are operating in an unmediated=20
>> environment. Obviously, this is a best-case scenario, but similarly=20
>> there are going to be mediated environments which are perfectly happy wi=
th a second O/A exchange with non-zero ports.
>>=20
>> This argues against designing a system in which none of the secondary=20
>> media flows without a second O/A exchange.
>
> Actually I think it argues against designing a system which treats an App=
lication Programming Interface the same as an on-the-wire Internet-wide=20
> negotiation mechanism, with a deployed infrastructure that has been growi=
ng larger for the past 10 years.  It also argues against tying the fate of=
=20
> WebRTC to SIP, with regards to SDP O/A.
>
> One thing we could do, for example, is define BUNDLE as we've been descri=
bing (which is essentially no change, afaik) and define its usage for SDP O=
/A as we've been describing. =20
> But we put a "unless the sender has specific knowledge otherwise, as it c=
ould in [jsep] for example".  Or something like that.  Then the JSEP draft =
could say what to do for WebRTC usage.

You need to convince the people who have been against unless-the-sender-has=
-specific-knowledge procedures :)

> Take a step back and ignore this particular port-numbering issue for a se=
cond: in what context would it make sense for an API to need two separate=20
> rounds of "Offer/Answer" between the Javascript and the Browser?  I.e., i=
f we weren't using SDP at all, and had never heard of RFC 3264, and just us=
ed a=20
> native Javascript API... what use-cases would require the JS script and B=
rowser to communicate multiple times between each other for the same action=
?=20
> I think error cases might, for example.  And I think if the JS hooked int=
o a callback for ICE transport selection changes, then the Browser would ne=
ed to=20
> tell it when those things changed, for example.  Can you think of any oth=
ers?

We need to remember that there is other information than the port in the SD=
P.

For example, there are attributes which will apply to the whole bundle, and=
 the value set for those attribute may depend on how many m- lines there wi=
ll be in the bundle (which, obviously, you don't know until you get the fir=
st answer).

Regards,

Christer


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Eric Rescorla'" <ekr@rtfm.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
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Cc: "'mmusic_ietf.org'" <mmusic@ietf.org>, 'Paul Kyzivat' <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Eric,

 

<snip>

Again, you need to distinguish between *endpoints* and filtering
middleboxes.

</snip>

 

IMO, it is not possible to identify "endpoint" and "filtering middleboxes"
based on the protocol exchanges. Both webrtc browser and webrtc gateway
handles ICE, SRTP-DTLS, same codecs, SDP in the same way. There is no way to
identify WebRTC browser to distinguish whether the remote side is a browser
or media gateway during WebRTC session. Please explain in case it is
possible to distinguish the remote side is WebRTC browser or WebRTC GW from
the originating browser.

 

Thanks

Partha

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Eric Rescorla
Sent: Sunday, August 11, 2013 8:21 PM
To: Christer Holmberg
Cc: mmusic_ietf.org; Paul Kyzivat
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

 

 

 

On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:

Hi,

>> I think making sure all entities have the correct port numbers is
critical in making sure the media will flow.
>
> There is a difference between "making sure the media will flow" and
"before media can
> flow at all".
> For instance, Neither Firefox nor Chrome current does a second O/A
> transaction and yet media flows just fine in most cases, since they are
operating
> in an unmediated environment. Obviously, this is a best-case scenario, but
similarly
> there are going to be mediated environments which are perfectly happy with
a second
> O/A exchange with non-zero ports.

There are also environements where a single offer, with identical port
numbers, work just fine, because the receiver will be able to handle it even
if it doesn't support BUNDLE.

However, we are defining a generic mechanism, supposed to work in different
environments.

 

"different environments" != "arbitrary lossage by any middlebox in the
universe"

 

Again, you need to distinguish between *endpoints* and filtering
middleboxes.

 

 

But, sure, if the community thinks that we shall define different options,
depending on environments etc, we can do that...

 

Which is not what I said.

 

-Ekr

 

Regards,

Christer





-----Original Message-----
From: Eric Rescorla [ekr@rtfm.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]

CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org];
Paul Kyzivat [pkyzivat@alum.mit.edu]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?





On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:

Hi,


>>> Requiring a second offer to add the bundle-only m-lines defeats the
purpose of bundle-only.
>> We already require a second offer. I know that some people, including
yourself, want to change that, but that is nothing we have agreed on.
>>
>>> a) The delay to do an O/A will often exceed the time to allocate
candidates, for small numbers of streams; bundle-only was meant to avoid
this candidate latency.
>> You do not have to allocate ports for the bundle-only m- lines.
>>
>>> b) Once you have the initial answer, you know whether the other side
supports BUNDLE, so you can take the appropriate action (and so there is no
need for bundle-only).
>>>
>>> If SIP can't deal with the bundle-only concept, then WebRTC apps that
care about latency will have to decide whether they are running in
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will be
bundled in the initial offer; in compatibility mode, 2 offer-answers will be
needed.
>> There are no "modes" in BUNDLE today :)
>>
> Of course there are. The handling in a reinvite case is distinct from the
initial offer, so that is modal behavior.


Sure, but you were talking about "WebRTC-only mode" and "compatibility
mode", and my point is that currently the procedures for initial offers,
subsequent offers etc are the same.


>>> This would be a somewhat unfortunate outcome, so I'd like to see
bundle-only (or something like it) work.
>> I don't think the second offer is going to cause any latency. Also, keep
in mind that you only need the 2nd Offer/Answer during the session
establishment. As agreed in Berlin, in mid-session changes you do NOT need 2
offer/answers.
>
> Please provide support for this statement. It seems clear that a 2nd offer
adds _at least_ 1 RTT latency.


Sure, but I don't think that is going to be a practical problem.



Well, I'd like to see a more convincing argument than "some middleboxes are
sad" when you
do that.




First, there are deployed legacy SIP systems that use even more O/A
transactions during session establishment. And, in your case I assume you
are even going to use something more "lightweight" than SIP to carry your
Offer/Answers.

Second, as WebRTC mandates usage of ICE, you are often going to need a
subsequent O/A anyway.



It's important to distinguish between round-trips that are in the critical
path for media
flowing and those which are not.


-Ekr

 


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<div class=3DSection1>

<p class=3DMsoNormal>Eric,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&lt;snip&gt;<o:p></o:p></p>

<p class=3DMsoNormal>Again, you need to distinguish between *endpoints* =
and
filtering middleboxes.<o:p></o:p></p>

<p class=3DMsoNormal>&lt;/snip&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>IMO, it is not possible to identify =
&#8220;endpoint&#8221;
and &#8220;filtering middleboxes&#8221; based on the protocol exchanges. =
Both
webrtc browser and webrtc gateway handles ICE, SRTP-DTLS, same codecs, =
SDP in
the same way. There is no way to identify WebRTC browser to distinguish =
whether
the remote side is a browser or media gateway during WebRTC session. =
Please
explain in case it is possible to distinguish the remote side is WebRTC =
browser
or WebRTC GW from the originating browser.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Partha<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf Of =
</b>Eric
Rescorla<br>
<b>Sent:</b> Sunday, August 11, 2013 8:21 PM<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> mmusic_ietf.org; Paul Kyzivat<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for
bundle-only m- lines?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg =
&lt;<a
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;
wrote:<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>Hi,<br>
<br>
&gt;&gt; I think making sure all entities have the correct port numbers =
is
critical in making sure the media will flow.<br>
&gt;<br>
&gt; There is a difference between &quot;making sure the media will =
flow&quot;
and &quot;before media can<br>
&gt; flow at all&quot;.<br>
&gt; For instance, Neither Firefox nor Chrome current does a second =
O/A<br>
&gt; transaction and yet media flows just fine in most cases, since they =
are
operating<br>
&gt; in an unmediated environment. Obviously, this is a best-case =
scenario, but
similarly<br>
&gt; there are going to be mediated environments which are perfectly =
happy with
a second<br>
&gt; O/A exchange with non-zero ports.<o:p></o:p></p>

</div>

<p class=3DMsoNormal>There are also environements where a single offer, =
with
identical port numbers, work just fine, because the receiver will be =
able to
handle it even if it doesn't support BUNDLE.<br>
<br>
However, we are defining a generic mechanism, supposed to work in =
different
environments.<o:p></o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>&quot;different environments&quot; !=3D =
&quot;arbitrary
lossage by any middlebox in the universe&quot;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Again, you need to distinguish between *endpoints* =
and
filtering middleboxes.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal>But, sure, if the community thinks that we shall =
define
different options, depending on environments etc, we can do =
that...<o:p></o:p></p>

</blockquote>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Which is not what I said.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>-Ekr<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal>Regards,<br>
<br>
Christer<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><br>
<br>
<br>
<br>
-----Original Message-----<br>
From: Eric Rescorla [<a =
href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>]<br>
To: Christer Holmberg [<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>]<br>
<br>
CC: Justin Uberti [<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>];
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a> [<a
href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>]; Paul Kyzivat [<a
href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>]<br>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-only m-
lines?<br>
<br>
<br>
<br>
<br>
<br>
On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg &lt;<a
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt;
wrote:<br>
<br>
Hi,<br>
<br>
<br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines =
defeats
the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, =
including
yourself, want to change that, but that is nothing we have agreed =
on.<br>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to =
allocate
candidates, for small numbers of streams; bundle-only was meant to avoid =
this
candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- =
lines.<br>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the =
other
side supports BUNDLE, so you can take the appropriate action (and so =
there is
no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can't deal with the bundle-only concept, then WebRTC =
apps
that care about latency will have to decide whether they are running in
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams =
will be
bundled in the initial offer; in compatibility mode, 2 offer-answers =
will be
needed.<br>
&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct =
from the
initial offer, so that is modal behavior.<br>
<br>
<br>
Sure, but you were talking about &quot;WebRTC-only mode&quot; and
&quot;compatibility mode&quot;, and my point is that currently the =
procedures
for initial offers, subsequent offers etc are the same.<br>
<br>
<br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I'd like =
to see
bundle-only (or something like it) work.<br>
&gt;&gt; I don't think the second offer is going to cause any latency. =
Also,
keep in mind that you only need the 2nd Offer/Answer during the session
establishment. As agreed in Berlin, in mid-session changes you do NOT =
need 2
offer/answers.<br>
&gt;<br>
&gt; Please provide support for this statement. It seems clear that a =
2nd offer
adds _at least_ 1 RTT latency.<br>
<br>
<br>
Sure, but I don't think that is going to be a practical problem.<br>
<br>
<br>
<br>
Well, I'd like to see a more convincing argument than &quot;some =
middleboxes
are sad&quot; when you<br>
do that.<br>
<br>
<br>
<br>
<br>
First, there are deployed legacy SIP systems that use even more O/A
transactions during session establishment. And, in your case I assume =
you are
even going to use something more &quot;lightweight&quot; than SIP to =
carry your
Offer/Answers.<br>
<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a
subsequent O/A anyway.<br>
<br>
<br>
<br>
It's important to distinguish between round-trips that are in the =
critical path
for media<br>
flowing and those which are not.<br>
<br>
<br>
-Ekr<o:p></o:p></p>

</div>

</div>

</blockquote>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</div>

</div>

</body>

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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)]
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    <font size="-1">Hi Hadriel:<br>
      <br>
      <font size="-1">As <font size="-1">far as I rem<font size="-1">ember,
            the history of the three mechanisms <font size="-1">is ki<font
                size="-1">nd of fuzzy. At some point in time in th<font
                  size="-1">e past, 3GPP hadn't decided on the
                  architecture to <font size="-1">enable CS calls over
                    SIP/SDP, and they were looking into it. The
                    mechanisms were envisioned <font size="-1">by the
                      authors and comments on the mailing list, and put
                      at 3GPP's disposal for a choice.<br>
                      <br>
                      <font size="-1">I conc<font size="-1">ur with&nbsp;
                          your concern that just because we mand<font
                            size="-1">ate to implement <font size="-1">the
                              3 of them<font size="-1">, these are not
                                going to be implemented. I don't have
                                any problem in removing from the draft <font
                                  size="-1">those mechanisms <font
                                    size="-1">that we envision <font
                                      size="-1">they won't be used.<br>
                                      <br>
                                      <font size="-1">/Miguel</font><br>
                                    </font></font></font></font></font></font></font></font></font></font></font></font></font></font></font></font><br>
    </font>
    <div class="moz-cite-prefix">On 10/08/2013 1:28, Hadriel Kaplan
      wrote:<br>
    </div>
    <blockquote
      cite="mid:D4AF2203-B760-42C3-AB49-6CEA757634CE@oracle.com"
      type="cite">
      <pre wrap="">
Wow.  I can kinda understand why they'd say that (interop reasons, presumably), but that's a big deal.  The 3 mechanisms are very different, with very different architectural and deployment implications.  I have a very hard time believing any of us would truly implement all three.  In fact, I don't truly believe anyone would.

So... I hate to do this, but can the authors summarize the history of how we got 3 mechanisms to begin with, and who would really implement them all?

Or to put it another way, instead of specifying all 3, would it be impossible to take a step back and reconsider the real motivating use-case (which as far as I know is 3GPP), and only specify the one they need/use?

-hadriel


On Aug 9, 2013, at 6:10 PM, Flemming Andreasen <a class="moz-txt-link-rfc2396E" href="mailto:fandreas@cisco.com">&lt;fandreas@cisco.com&gt;</a> wrote:

</pre>
      <blockquote type="cite">
        <pre wrap="">Greetings MMUSIC

As you can see per the below, the IESG approved the Circuit-Switched Bearers draft for publication, however before doing so, they requested a couple of changes. The resulting approved document can be found at <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21">http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21</a>

One of the changes introduced resulted in all 3 correlation mechanisms now being mandatory-to-implement (previously they were not):
&lt;quote&gt;
   Implementations MUST support the three correlation mechanisms
   specified in 
Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
,
   and SHOULD include all supported mechanisms the initial Offer.

&lt;/quote&gt;
Since this is a non-trivial change that has not been discussed by the WG (and we know of at least one person having expressed concerns with it off-line), we are hereby solicting the WG participants for input on changes introduced as a result of the IESG review with a focus on the above. In doing so, please keep in mind that we are developing standards that work in the general Internet. 


The following URL provides a diff-file showing the changes resulting from the IESG review:
    <a class="moz-txt-link-freetext" href="http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&amp;url2=draft-ietf-mmusic-sdp-cs-21.txt">http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&amp;url2=draft-ietf-mmusic-sdp-cs-21.txt</a>

Comments are due by August 23 (two weeks from today)

Thanks 

-- Flemming (MMUSIC co-chair)



-------- Original Message --------
Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
Date:	Mon, 1 Jul 2013 11:56:23 -0700
From:	The IESG <a class="moz-txt-link-rfc2396E" href="mailto:iesg-secretary@ietf.org">&lt;iesg-secretary@ietf.org&gt;</a>
To:	IETF-Announce <a class="moz-txt-link-rfc2396E" href="mailto:ietf-announce@ietf.org">&lt;ietf-announce@ietf.org&gt;</a>
CC:	mmusic chair <a class="moz-txt-link-rfc2396E" href="mailto:mmusic-chairs@tools.ietf.org">&lt;mmusic-chairs@tools.ietf.org&gt;</a>, mmusic mailing list <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a>, RFC Editor <a class="moz-txt-link-rfc2396E" href="mailto:rfc-editor@rfc-editor.org">&lt;rfc-editor@rfc-editor.org&gt;</a>

The IESG has approved the following document:
- 'Session Description Protocol (SDP) Extension For Setting Audio and
   Video Media Streams Over Circuit-Switched Bearers In The Public
   Switched Telephone Network (PSTN)'
  (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard

This document is the product of the Multiparty Multimedia Session Control
Working Group.

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

A URL of this Internet Draft is:

<a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/">http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/</a>





*Technical Summary
*

The document describes use cases, requirements, and protocol extensions
for using the Session Description Protocol (SDP) Offer/Answer model for
establishing audio and video media streams over circuit-switched bearers
in the Publich Switched Telephone Network (PSTN)

*Working Group Summary
*

The WG had some discussion around the format to use for E.164 numbers
and whether to align this with the existing definition in RFC 3108. The
RFC 3108 definition was seen as deficient and the WG agreed it was
better to align with relevant parts of the tel URI format defined in RFC
3966, not least since SDP address types are defined in the context of a
particular network type, and hence RFC 3108 compatibility is not a
concern (the implication is that the "E164" address type may differ
between network types in SDP).

*Document Quality
*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

The document has received good overall review in the WG, some of which
resulted in changes to the detailed specification. The document has been
reviewed in detail several times, including of the last few versions.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.


*Personnel
*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo 
_______________________________________________
mmusic mailing list

<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>

.




_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
      </blockquote>
      <pre wrap="">
_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>

</pre>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="73">-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain</pre>
  </body>
</html>

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Date: Mon, 12 Aug 2013 11:18:32 -0400
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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 From a chair point of view, the intent with "scope is WebRTC" was 
largely what Paul described as option 1) below although I'd change 
"solely" to "primarily" (intent being that WebRTC is the primary driver, 
but if we can trivially and without controversy provide solutions that 
apply beyond that, then that would be desirable).

As Keith points out, this limited scope deliverable creates a new 
question in terms of scope for all the derived deliverables that have to 
be developed in the various WGs. It's not clear that we will have a 
single answer for all of these and taking BUNDLE as an example, it's 
currently an MMUSIC WG item, but its scope so far has not been limited 
to WebRTC (and I believe Colin argued in favor of keeping it that way at 
the Berlin meeting as well).

Thanks

-- Flemming


On 8/7/13 1:26 PM, DRAGE, Keith (Keith) wrote:
> I said that all the involved groups need to make a second step with all their deliverables, which is to decide whether what they are doing is used generally, or solely in the context of WEBRTC.
>
> Deciding whether it is 1) or 2) has not been defined in my understanding, and would really come under the scope of the second decision.
>
> I for one am hoping as much of this as possible comes out as being generally applicable, and obviously designed as such as a result, but the MMUSIC WG did not appear to be ready to say that last week. So some steps to get there.
>
> Keith
>
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: 07 August 2013 14:41
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>>
>> On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
>>> As one of those that used the term, my intention was that the initial
>>> decision was to agree to define what RTCWEB needed.
>>>
>>> As each dependent working group looks at the extensions they need to do,
>>> then they need to make the decision as to whether a wider scope of
>>> possible, desirable, or dangerous.
>>>
>>> So the BUNDLE status at the moment is that any changes for the unified
>>> plan should work for RTCWEB usage, but not for general. It would
>>> probably make sense to make it more generally applicable, but that is
>>> the next step in the decision process, and may need some other technical
>>> decisions before it happens.
>> I don't know how you intend the above to be parsed. Specifically, when
>> you say:
>>     "any changes for the unified plan should work for RTCWEB usage"
>>
>> Do you mean:
>>
>> 1) that the changes should be designed solely with RTCWEB usage in mind?
>> (But are still applicable in other environments that find them useful
>> and choose to adopt them.)
>>
>> Or do you mean:
>>
>> 2) that those changes should not be used in any environment other than
>> RTCWEB?
>>
>> I might be able to accept (1), but not (2).
>>
>> 	Thanks,
>> 	Paul
>>
>>
>>> Keith
>>>
>>> ------------------------------------------------------------------------
>>>
>>> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On
>>> Behalf Of *Bernard Aboba
>>> *Sent:* 07 August 2013 00:50
>>> *To:* Christer Holmberg; mmusic
>>> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>>>
>>> Christer said:
>>>
>>> "While cleaning up my MMUSIC notes today, I was listening to the audio
>>> recording, in order to get a clarification regarding what "scope is
>>> WEBRTC" really means.  However, unless I missed something, I didn't get
>>> a clear answer.
>>>
>>> Does it only cover the browser API (JSEP) or does it also cover
>>> something that can be sent out on the wire (as long as it is not SIP) by
>>> a browser?
>>>
>>> Feel free to tell me that it is a stupid question, and that the answer
>>> is very clear :)"
>>>
>>> [BA]  I actually think it is a good question :)
>>>
>>> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
>>> Plan" applies to use within  the JSEP API.    As a result, one
>>> might expect the SDP used in createOffer, setLocalDescription and
>>> setRemoteDescription to resemble that in the "Unified Plan" document.
>>> Since signaling is out of scope for WebRTC, applications that use the
>>> JSEP API might or might not use the "Unified Plan" SDP in signaling on
>>> the wire.
>>>
>>> However, "Unified Plan" includes normative references to published SDP
>>> specifications that were developed for general use, and the adoption of
>>> "Unified Plan" (or even an applicability statement within the "Unified
>>> Plan" document) doesn't change the applicability of those documents.
>>>
>>> Also, there may be some specifications that are part of the "Unified
>>> Plan" workload that may have applicability beyond WebRTC (e.g. a=group:
>>> SIMULCAST)
>>>
>>> Ultimately, I am expecting the work involved in "Unified Plan" to span
>>> multiple documents, and to clarify the applicability,  those individual
>>> documents will need to explain what they apply to (just WebRTC use or
>>> more general use).
>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> .
>


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Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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On 8/7/13 11:27 AM, Christer Holmberg wrote:
>
> If everyone has the same understanding, I would ask the chairs to make 
> it *very clear* in the final minutes ;)
>
I think the minutes should only reflect what was actually discussed in 
the meeting so I'm not sure we can record it there. However if we can 
conclude this thread with general agreement on the interpretation, then 
we can add a pointer/note to that effect.

Thanks

-- Flemming


> Regards,
>
> Christer
>
> *Lähettäjä:*Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
> *Lähetetty:* 7. elokuuta 2013 18:25
> *Vastaanottaja:* Christer Holmberg
> *Kopio:* Paul Kyzivat; mmusic@ietf.org
> *Aihe:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>
> On Wed, Aug 7, 2013 at 10:21 AM, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
> Hi,
>
> Ok, so what is "RTCWEB usage" then? :)
>
> For example, if I implement a JavaScript SIP app, and want to use the 
> UP with SIP on the wire, would that be a "RTCWEB usage"? My 
> understanding is that it would NOT be.
>
> My personal understanding of the scope is:
>
> [MB] That is also my understanding. [/MB]
>
>     1)JSEP; and
>
>     2)Browser-to-browser cases where we KNOW there are no
>     intermediaries etc in between.
>
>     Regards,
>
>     Christer
>
>     *Lähettäjä:*mmusic-bounces@ietf.org
>     <mailto:mmusic-bounces@ietf.org> [mailto:mmusic-bounces@ietf.org
>     <mailto:mmusic-bounces@ietf.org>] *Puolesta *Mary Barnes
>     *Lähetetty:* 7. elokuuta 2013 17:56
>     *Vastaanottaja:* Paul Kyzivat
>     *Kopio:* mmusic@ietf.org <mailto:mmusic@ietf.org>
>     *Aihe:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>
>     On Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat
>     <pkyzivat@alum.mit.edu <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>         On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
>
>         As one of those that used the term, my intention was that the
>         initial
>         decision was to agree to define what RTCWEB needed.
>
>         As each dependent working group looks at the extensions they
>         need to do,
>         then they need to make the decision as to whether a wider scope of
>         possible, desirable, or dangerous.
>
>         So the BUNDLE status at the moment is that any changes for the
>         unified
>         plan should work for RTCWEB usage, but not for general. It would
>         probably make sense to make it more generally applicable, but
>         that is
>         the next step in the decision process, and may need some other
>         technical
>         decisions before it happens.
>
>         I don't know how you intend the above to be parsed.
>         Specifically, when you say:
>
>
>           "any changes for the unified plan should work for RTCWEB usage"
>
>         Do you mean:
>
>         1) that the changes should be designed solely with RTCWEB
>         usage in mind? (But are still applicable in other environments
>         that find them useful and choose to adopt them.)
>
>         Or do you mean:
>
>         2) that those changes should not be used in any environment
>         other than RTCWEB?
>
>         I might be able to accept (1), but not (2).
>
>     [MB] I would agree with your (1), however, not with the
>     parenthetical part.  I would suggest that be "But MAY still be
>     applicable in other environments... "   My concern is that at this
>     point in time, you can't say that the changes ARE applicable to
>     other environments ....  [/MB]
>
>
>                 Thanks,
>                 Paul
>
>         Keith
>
>         ------------------------------------------------------------------------
>
>         *From:*mmusic-bounces@ietf.org
>         <mailto:mmusic-bounces@ietf.org>
>         [mailto:mmusic-bounces@ietf.org
>         <mailto:mmusic-bounces@ietf.org>] *On
>         Behalf Of *Bernard Aboba
>         *Sent:* 07 August 2013 00:50
>         *To:* Christer Holmberg; mmusic
>         *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
>
>
>
>         Christer said:
>
>         "While cleaning up my MMUSIC notes today, I was listening to
>         the audio
>         recording, in order to get a clarification regarding what
>         "scope is
>         WEBRTC" really means.  However, unless I missed something, I
>         didn't get
>         a clear answer.
>
>         Does it only cover the browser API (JSEP) or does it also cover
>         something that can be sent out on the wire (as long as it is
>         not SIP) by
>         a browser?
>
>         Feel free to tell me that it is a stupid question, and that
>         the answer
>         is very clear :)"
>
>         [BA]  I actually think it is a good question :)
>
>         One interpretation of "scope is WebRTC"  is that the  SDP in
>         "Unified
>         Plan" applies to use within  the JSEP API.    As a result, one
>         might expect the SDP used in createOffer, setLocalDescription and
>         setRemoteDescription to resemble that in the "Unified Plan"
>         document.
>         Since signaling is out of scope for WebRTC, applications that
>         use the
>         JSEP API might or might not use the "Unified Plan" SDP in
>         signaling on
>         the wire.
>
>         However, "Unified Plan" includes normative references to
>         published SDP
>         specifications that were developed for general use, and the
>         adoption of
>         "Unified Plan" (or even an applicability statement within the
>         "Unified
>         Plan" document) doesn't change the applicability of those
>         documents.
>
>         Also, there may be some specifications that are part of the
>         "Unified
>         Plan" workload that may have applicability beyond WebRTC (e.g.
>         a=group:
>         SIMULCAST)
>
>         Ultimately, I am expecting the work involved in "Unified Plan"
>         to span
>         multiple documents, and to clarify the applicability,  those
>         individual
>         documents will need to explain what they apply to (just WebRTC
>         use or
>         more general use).
>
>         _______________________________________________
>         mmusic mailing list
>         mmusic@ietf.org <mailto:mmusic@ietf.org>
>         https://www.ietf.org/mailman/listinfo/mmusic
>
>
>         _______________________________________________
>         mmusic mailing list
>         mmusic@ietf.org <mailto:mmusic@ietf.org>
>         https://www.ietf.org/mailman/listinfo/mmusic
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <br>
    <div class="moz-cite-prefix">On 8/7/13 11:27 AM, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C42157D@ESESSMB209.ericsson.se"
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">If everyone has the same understanding, I would
            ask the chairs to make it
            <b>very clear</b> in the final minutes ;)<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
      </div>
    </blockquote>
    I think the minutes should only reflect what was actually discussed
    in the meeting so I'm not sure we can record it there. However if we
    can conclude this thread with general agreement on the
    interpretation, then we can add a pointer/note to that effect. <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C42157D@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">L&auml;hett&auml;j&auml;:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
            Mary Barnes [<a class="moz-txt-link-freetext" href="mailto:mary.ietf.barnes@gmail.com">mailto:mary.ietf.barnes@gmail.com</a>]
            <br>
            <b>L&auml;hetetty:</b> 7. elokuuta 2013 18:25<br>
            <b>Vastaanottaja:</b> Christer Holmberg<br>
            <b>Kopio:</b> Paul Kyzivat; <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
            <b>Aihe:</b> Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"<o:p></o:p></span></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <div>
            <p class="MsoNormal" style="margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
            <div>
              <p class="MsoNormal">On Wed, Aug 7, 2013 at 10:21 AM,
                Christer Holmberg &lt;<a moz-do-not-send="true"
                  href="mailto:christer.holmberg@ericsson.com"
                  target="_blank">christer.holmberg@ericsson.com</a>&gt;
                wrote:<o:p></o:p></p>
              <div>
                <div>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                      lang="EN-US">Ok, so what is &#8221;RTCWEB usage&#8221; then?
                      :)</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                      lang="EN-US">&nbsp;</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                      lang="EN-US">For example, if I implement a
                      JavaScript SIP app, and want to use the UP with
                      SIP on the wire, would that be a &#8220;RTCWEB usage&#8221;?
                      My understanding is that it would NOT be.</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                      lang="EN-US">&nbsp;</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                      lang="EN-US">My personal understanding of the
                      scope is:</span><o:p></o:p></p>
                </div>
              </div>
              <div>
                <p class="MsoNormal">[MB] That is also my understanding.
                  [/MB]&nbsp;<o:p></o:p></p>
              </div>
              <blockquote style="border:none;border-left:solid #CCCCCC
                1.0pt;padding:0cm 0cm 0cm
                6.0pt;margin-left:4.8pt;margin-right:0cm">
                <div>
                  <div>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                    <p><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">1)</span><span
                        style="font-size:7.0pt;color:#1F497D"
                        lang="EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                      </span><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">JSEP; and</span><o:p></o:p></p>
                    <p><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">2)</span><span
                        style="font-size:7.0pt;color:#1F497D"
                        lang="EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                      </span><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">Browser-to-browser cases where we
                        KNOW there are no intermediaries etc in between.</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">Regards,</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">Christer</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">L&auml;hett&auml;j&auml;:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                        <a moz-do-not-send="true"
                          href="mailto:mmusic-bounces@ietf.org"
                          target="_blank">mmusic-bounces@ietf.org</a>
                        [mailto:<a moz-do-not-send="true"
                          href="mailto:mmusic-bounces@ietf.org"
                          target="_blank">mmusic-bounces@ietf.org</a>]
                        <b>Puolesta </b>Mary Barnes<br>
                        <b>L&auml;hetetty:</b> 7. elokuuta 2013 17:56<br>
                        <b>Vastaanottaja:</b> Paul Kyzivat<br>
                        <b>Kopio:</b> <a moz-do-not-send="true"
                          href="mailto:mmusic@ietf.org" target="_blank">mmusic@ietf.org</a><br>
                        <b>Aihe:</b> Re: [MMUSIC] UP: Meaning of "scope
                        is WEBRTC"</span><o:p></o:p></p>
                    <div>
                      <div>
                        <p class="MsoNormal"
                          style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
                        <div>
                          <div>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On
                              Wed, Aug 7, 2013 at 8:40 AM, Paul Kyzivat
                              &lt;<a moz-do-not-send="true"
                                href="mailto:pkyzivat@alum.mit.edu"
                                target="_blank">pkyzivat@alum.mit.edu</a>&gt;
                              wrote:<o:p></o:p></p>
                            <div>
                              <blockquote
                                style="border:none;border-left:solid
                                #CCCCCC 1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt">
                                <div>
                                  <p class="MsoNormal"
                                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On
                                    8/7/13 2:04 PM, DRAGE, Keith (Keith)
                                    wrote:<o:p></o:p></p>
                                  <p class="MsoNormal"
                                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">As
                                    one of those that used the term, my
                                    intention was that the initial<br>
                                    decision was to agree to define what
                                    RTCWEB needed.<br>
                                    <br>
                                    As each dependent working group
                                    looks at the extensions they need to
                                    do,<br>
                                    then they need to make the decision
                                    as to whether a wider scope of<br>
                                    possible, desirable, or dangerous.<br>
                                    <br>
                                    So the BUNDLE status at the moment
                                    is that any changes for the unified<br>
                                    plan should work for RTCWEB usage,
                                    but not for general. It would<br>
                                    probably make sense to make it more
                                    generally applicable, but that is<br>
                                    the next step in the decision
                                    process, and may need some other
                                    technical<br>
                                    decisions before it happens.<o:p></o:p></p>
                                  <p class="MsoNormal"
                                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
                                </div>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I
                                  don't know how you intend the above to
                                  be parsed. Specifically, when you say:<o:p></o:p></p>
                                <div>
                                  <p class="MsoNormal"
                                    style="mso-margin-top-alt:auto;margin-bottom:12.0pt"><br>
                                    &nbsp; "any changes for the unified plan
                                    should work for RTCWEB usage"<o:p></o:p></p>
                                </div>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Do
                                  you mean:<br>
                                  <br>
                                  1) that the changes should be designed
                                  solely with RTCWEB usage in mind? (But
                                  are still applicable in other
                                  environments that find them useful and
                                  choose to adopt them.)<br>
                                  <br>
                                  Or do you mean:<br>
                                  <br>
                                  2) that those changes should not be
                                  used in any environment other than
                                  RTCWEB?<br>
                                  <br>
                                  I might be able to accept (1), but not
                                  (2).<o:p></o:p></p>
                              </blockquote>
                              <div>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">[MB]
                                  I would agree with your (1), however,
                                  not with the parenthetical part. &nbsp;I
                                  would suggest that be "But MAY still
                                  be applicable in other environments...
                                  " &nbsp; My concern is that at this point
                                  in time, you can't say that the
                                  changes ARE applicable to other
                                  environments .... &nbsp;[/MB]<o:p></o:p></p>
                              </div>
                              <div>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
                              </div>
                              <blockquote
                                style="border:none;border-left:solid
                                #CCCCCC 1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt">
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;margin-bottom:12.0pt"><br>
                                  &nbsp; &nbsp; &nbsp; &nbsp; Thanks,<br>
                                  &nbsp; &nbsp; &nbsp; &nbsp; Paul<o:p></o:p></p>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Keith<br>
                                  <br>
------------------------------------------------------------------------<br>
                                  <br>
                                  *From:*<a moz-do-not-send="true"
                                    href="mailto:mmusic-bounces@ietf.org"
                                    target="_blank">mmusic-bounces@ietf.org</a>
                                  [mailto:<a moz-do-not-send="true"
                                    href="mailto:mmusic-bounces@ietf.org"
                                    target="_blank">mmusic-bounces@ietf.org</a>]
                                  *On<br>
                                  Behalf Of *Bernard Aboba<br>
                                  *Sent:* 07 August 2013 00:50<br>
                                  *To:* Christer Holmberg; mmusic<br>
                                  *Subject:* Re: [MMUSIC] UP: Meaning of
                                  "scope is WEBRTC"<o:p></o:p></p>
                                <div>
                                  <div>
                                    <p class="MsoNormal"
                                      style="mso-margin-top-alt:auto;margin-bottom:12.0pt"><br>
                                      <br>
                                      Christer said:<br>
                                      <br>
                                      "While cleaning up my MMUSIC notes
                                      today, I was listening to the
                                      audio<br>
                                      recording, in order to get a
                                      clarification regarding what
                                      "scope is<br>
                                      WEBRTC" really means. &nbsp;However,
                                      unless I missed something, I
                                      didn't get<br>
                                      a clear answer.<br>
                                      <br>
                                      Does it only cover the browser API
                                      (JSEP) or does it also cover<br>
                                      something that can be sent out on
                                      the wire (as long as it is not
                                      SIP) by<br>
                                      a browser?<br>
                                      <br>
                                      Feel free to tell me that it is a
                                      stupid question, and that the
                                      answer<br>
                                      is very clear :)"<br>
                                      <br>
                                      [BA] &nbsp;I actually think it is a
                                      good question :)<br>
                                      <br>
                                      One interpretation of "scope is
                                      WebRTC" &nbsp;is that the &nbsp;SDP in
                                      "Unified<br>
                                      Plan" applies to use within &nbsp;the
                                      JSEP API. &nbsp; &nbsp;As a result, one<br>
                                      might expect the SDP used in
                                      createOffer, setLocalDescription
                                      and<br>
                                      setRemoteDescription to resemble
                                      that in the "Unified Plan"
                                      document.<br>
                                      Since signaling is out of scope
                                      for WebRTC, applications that use
                                      the<br>
                                      JSEP API might or might not use
                                      the "Unified Plan" SDP in
                                      signaling on<br>
                                      the wire.<br>
                                      <br>
                                      However, "Unified Plan" includes
                                      normative references to published
                                      SDP<br>
                                      specifications that were developed
                                      for general use, and the adoption
                                      of<br>
                                      "Unified Plan" (or even an
                                      applicability statement within the
                                      "Unified<br>
                                      Plan" document) doesn't change the
                                      applicability of those documents.<br>
                                      <br>
                                      Also, there may be some
                                      specifications that are part of
                                      the "Unified<br>
                                      Plan" workload that may have
                                      applicability beyond WebRTC (e.g.
                                      a=group:<br>
                                      SIMULCAST)<br>
                                      <br>
                                      Ultimately, I am expecting the
                                      work involved in "Unified Plan" to
                                      span<br>
                                      multiple documents, and to clarify
                                      the applicability, &nbsp;those
                                      individual<br>
                                      documents will need to explain
                                      what they apply to (just WebRTC
                                      use or<br>
                                      more general use).<br>
                                      <br>
                                      <o:p></o:p></p>
                                  </div>
                                </div>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;margin-bottom:12.0pt">_______________________________________________<br>
                                  mmusic mailing list<br>
                                  <a moz-do-not-send="true"
                                    href="mailto:mmusic@ietf.org"
                                    target="_blank">mmusic@ietf.org</a><br>
                                  <a moz-do-not-send="true"
                                    href="https://www.ietf.org/mailman/listinfo/mmusic"
                                    target="_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
                                <p class="MsoNormal"
                                  style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><br>
_______________________________________________<br>
                                  mmusic mailing list<br>
                                  <a moz-do-not-send="true"
                                    href="mailto:mmusic@ietf.org"
                                    target="_blank">mmusic@ietf.org</a><br>
                                  <a moz-do-not-send="true"
                                    href="https://www.ietf.org/mailman/listinfo/mmusic"
                                    target="_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><o:p></o:p></p>
                              </blockquote>
                            </div>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
                          </div>
                        </div>
                      </div>
                    </div>
                  </div>
                </div>
              </blockquote>
            </div>
            <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From ron.even.tlv@gmail.com  Mon Aug 12 23:59:06 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Flemming Andreasen'" <fandreas@cisco.com>, "'DRAGE, Keith \(Keith\)'" <keith.drage@alcatel-lucent.com>
References: <7594FB04B1934943A5C02806D1A2204B1C41DB40@ESESSMB209.ericsson.se>	<BLU169-W132B4B2091B7DF8FC0BC926935D0@phx.gbl>	<949EF20990823C4C85C18D59AA11AD8B07D89D@FR712WXCHMBA11.zeu.alcatel-lucent.com>	<52024E57.9070602@alum.mit.edu>	<949EF20990823C4C85C18D59AA11AD8B07DDFC@FR712WXCHMBA11.zeu.alcatel-lucent.com> <5208FCC8.6000200@cisco.com>
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Date: Tue, 13 Aug 2013 09:56:50 +0300
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Cc: mmusic@ietf.org, 'Paul Kyzivat' <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
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Hi, 
I agree with Paul's first option.
I think that section 2 of the UP draft provides the solution overview and
some of the points are solely RTCweb while some  of the points may be used
by other solution.

One important point made in the presentation was that this does not change
RFC3264 so others can choose different solution, for example, instead of the
first two bullet items in section 2 while not changing RFC3264 . 

Since this solution does not change RFC3264 I see no problem on the scope of
work in other WG. It just need to be clear about the applicability of the
work and having it address both RFC3264 and UP. 

Roni Even

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Flemming Andreasen
> Sent: 12 August, 2013 6:19 PM
> To: DRAGE, Keith (Keith)
> Cc: mmusic@ietf.org; Paul Kyzivat
> Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
> 
>  From a chair point of view, the intent with "scope is WebRTC" was largely
> what Paul described as option 1) below although I'd change "solely" to
> "primarily" (intent being that WebRTC is the primary driver, but if we can
> trivially and without controversy provide solutions that apply beyond
that,
> then that would be desirable).
> 
> As Keith points out, this limited scope deliverable creates a new question
in
> terms of scope for all the derived deliverables that have to be developed
in
> the various WGs. It's not clear that we will have a single answer for all
of
> these and taking BUNDLE as an example, it's currently an MMUSIC WG item,
> but its scope so far has not been limited to WebRTC (and I believe Colin
> argued in favor of keeping it that way at the Berlin meeting as well).
> 
> Thanks
> 
> -- Flemming
> 
> 
> On 8/7/13 1:26 PM, DRAGE, Keith (Keith) wrote:
> > I said that all the involved groups need to make a second step with all
their
> deliverables, which is to decide whether what they are doing is used
> generally, or solely in the context of WEBRTC.
> >
> > Deciding whether it is 1) or 2) has not been defined in my
understanding,
> and would really come under the scope of the second decision.
> >
> > I for one am hoping as much of this as possible comes out as being
> generally applicable, and obviously designed as such as a result, but the
> MMUSIC WG did not appear to be ready to say that last week. So some steps
> to get there.
> >
> > Keith
> >
> >
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Paul Kyzivat
> >> Sent: 07 August 2013 14:41
> >> To: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
> >>
> >> On 8/7/13 2:04 PM, DRAGE, Keith (Keith) wrote:
> >>> As one of those that used the term, my intention was that the
> >>> initial decision was to agree to define what RTCWEB needed.
> >>>
> >>> As each dependent working group looks at the extensions they need to
> >>> do, then they need to make the decision as to whether a wider scope
> >>> of possible, desirable, or dangerous.
> >>>
> >>> So the BUNDLE status at the moment is that any changes for the
> >>> unified plan should work for RTCWEB usage, but not for general. It
> >>> would probably make sense to make it more generally applicable, but
> >>> that is the next step in the decision process, and may need some
> >>> other technical decisions before it happens.
> >> I don't know how you intend the above to be parsed. Specifically,
> >> when you say:
> >>     "any changes for the unified plan should work for RTCWEB usage"
> >>
> >> Do you mean:
> >>
> >> 1) that the changes should be designed solely with RTCWEB usage in
> mind?
> >> (But are still applicable in other environments that find them useful
> >> and choose to adopt them.)
> >>
> >> Or do you mean:
> >>
> >> 2) that those changes should not be used in any environment other
> >> than RTCWEB?
> >>
> >> I might be able to accept (1), but not (2).
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >>
> >>> Keith
> >>>
> >>> --------------------------------------------------------------------
> >>> ----
> >>>
> >>> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]
> *On
> >>> Behalf Of *Bernard Aboba
> >>> *Sent:* 07 August 2013 00:50
> >>> *To:* Christer Holmberg; mmusic
> >>> *Subject:* Re: [MMUSIC] UP: Meaning of "scope is WEBRTC"
> >>>
> >>> Christer said:
> >>>
> >>> "While cleaning up my MMUSIC notes today, I was listening to the
> >>> audio recording, in order to get a clarification regarding what
> >>> "scope is WEBRTC" really means.  However, unless I missed something,
> >>> I didn't get a clear answer.
> >>>
> >>> Does it only cover the browser API (JSEP) or does it also cover
> >>> something that can be sent out on the wire (as long as it is not
> >>> SIP) by a browser?
> >>>
> >>> Feel free to tell me that it is a stupid question, and that the
> >>> answer is very clear :)"
> >>>
> >>> [BA]  I actually think it is a good question :)
> >>>
> >>> One interpretation of "scope is WebRTC"  is that the  SDP in "Unified
> >>> Plan" applies to use within  the JSEP API.    As a result, one
> >>> might expect the SDP used in createOffer, setLocalDescription and
> >>> setRemoteDescription to resemble that in the "Unified Plan" document.
> >>> Since signaling is out of scope for WebRTC, applications that use
> >>> the JSEP API might or might not use the "Unified Plan" SDP in
> >>> signaling on the wire.
> >>>
> >>> However, "Unified Plan" includes normative references to published
> >>> SDP specifications that were developed for general use, and the
> >>> adoption of "Unified Plan" (or even an applicability statement
> >>> within the "Unified Plan" document) doesn't change the applicability
of
> those documents.
> >>>
> >>> Also, there may be some specifications that are part of the "Unified
> >>> Plan" workload that may have applicability beyond WebRTC (e.g.
> a=group:
> >>> SIMULCAST)
> >>>
> >>> Ultimately, I am expecting the work involved in "Unified Plan" to
> >>> span multiple documents, and to clarify the applicability,  those
> >>> individual documents will need to explain what they apply to (just
> >>> WebRTC use or more general use).
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> mmusic mailing list
> >>> mmusic@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/mmusic
> >>>
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> > .
> >
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From internet-drafts@ietf.org  Tue Aug 13 05:13:46 2013
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-msid-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

	Title           : Cross Session Stream Identification in the Session Descr=
iption Protocol
	Author(s)       : Harald Alvestrand
	Filename        : draft-ietf-mmusic-msid-01.txt
	Pages           : 14
	Date            : 2013-08-13

Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations between media streams.

   This mechanism is used to signal the association between the SDP
   concept of "m-line" and the WebRTC concept of "MediaStream" /
   "MediaStreamTrack" using SDP signaling.

   This document is a work item of the MMUSIC WG, whose discussion list
   is mmusic@ietf.org.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-msid

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-msid-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-msid-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From harald@alvestrand.no  Tue Aug 13 05:24:26 2013
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The auto-poster doesn't seem to want to tell mmusic about this....

this is updated according to Unified-plan. Since it was only 1 day 
before the old version had expired, the proofreading might be a little 
light - please point out places where I had missed changing "SSRC" to 
"m-line"!

I retained some description of the old mechanism (using a=ssrc) in an 
appendix. Please state opinions on whether this should be retained as 
informative, promoted to normative, or removed entirely in the next version.

Hope this informs!

        Harald


-------- Original Message --------
Subject: 	New Version Notification for draft-ietf-mmusic-msid-01.txt
Date: 	Tue, 13 Aug 2013 05:13:42 -0700
From: 	internet-drafts@ietf.org
To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand 
<harald@alvestrand.no>



A new version of I-D, draft-ietf-mmusic-msid-01.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-ietf-mmusic-msid
Revision:	 01
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2013-08-13
Group:		 mmusic
Number of pages: 14
URL:             http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
Status:          http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
Htmlized:        http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
Diff:            http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01

Abstract:
    This document specifies a grouping mechanism for RTP media streams
    that can be used to specify relations between media streams.

    This mechanism is used to signal the association between the SDP
    concept of "m-line" and the WebRTC concept of "MediaStream" /
    "MediaStreamTrack" using SDP signaling.

    This document is a work item of the MMUSIC WG, whose discussion list
    is mmusic@ietf.org.


                                                                                   


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--------------000203050105000906080708
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<html>
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    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    The auto-poster doesn't seem to want to tell mmusic about this....<br>
    <br>
    this is updated according to Unified-plan. Since it was only 1 day
    before the old version had expired, the proofreading might be a
    little light - please point out places where I had missed changing
    "SSRC" to "m-line"!<br>
    <br>
    I retained some description of the old mechanism (using a=ssrc) in
    an appendix. Please state opinions on whether this should be
    retained as informative, promoted to normative, or removed entirely
    in the next version.<br>
    <br>
    Hope this informs!<br>
    <br>
    Â Â Â Â Â Â  Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-ietf-mmusic-msid-01.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 13 Aug 2013 05:13:42 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Harald T. Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
              Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-ietf-mmusic-msid-01.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-ietf-mmusic-msid
Revision:	 01
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2013-08-13
Group:		 mmusic
Number of pages: 14
URL:             <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt">http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt</a>
Status:          <a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid">http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-mmusic-msid-01">http://tools.ietf.org/html/draft-ietf-mmusic-msid-01</a>
Diff:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01">http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01</a>

Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations between media streams.

   This mechanism is used to signal the association between the SDP
   concept of "m-line" and the WebRTC concept of "MediaStream" /
   "MediaStreamTrack" using SDP signaling.

   This document is a work item of the MMUSIC WG, whose discussion list
   is <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>.


                                                                                  


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</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------000203050105000906080708--

From magnus.westerlund@ericsson.com  Tue Aug 13 06:09:17 2013
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Date: Tue, 13 Aug 2013 15:10:06 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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References: <20130718161212.19084.19711.idtracker@ietfa.amsl.com> <51F61F88.1000503@ericsson.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] New Liaison Statement, "LS on Missing IANA Registration Information on SDP Attributes in RFC 2326"
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On 2013-07-29 09:53, Ari KerÃ¤nen wrote:
> MMUSIC,
> 
> We have received a liaison statement from 3GPP regarding missing SDP
> attributes at the IANA registry (see the e-mail below for details).
> 
> As the statement says:
>> SA4 has identified that IANA registrations of three SDP attributes
>> defined in RFC 2326, namely â€˜controlâ€™, â€˜rangeâ€™ and â€˜etagâ€™, are missing
> 
> As a solution, the RTSP 2.0 draft already registers 'control' and
> 'range' for RFC 2326 and RFC 2326bis
> (http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-34#section-22.15).
> However, in RFC2326bis 'etag' is renamed into 'mtag'.
> 
> Hence, it seems that we should also register 'etag' but for that
> reference RTSP 1.0 (RFC 2326) only.
> 
> Please review the liaison statement and our proposed solution and send
> your comments to us and to the list.

WG,

I volunteer to send an email registering these SDP attributes to IANA so
that a respond can be sent to 3GPP SA4 that they attribute has been
registered. I propose to include all attributes in that registration. I
can clarify in the RFC 2326bis draft that the SDP registrations are an
update to what we put in at this stage. I have also chosen to put "etag"
in the both category due to that is what has been done in the RFC2326bis
draft as the original text is far from clear if it is a session only or not.

Thus I propose to send the below to IANA on Monday (19th of Aug) unless
someone disagree.


> Dear IANA,
> 
> It has come to our attention that the SDP attributes defined in RFC 2326
> are not registered in the SDP attributes registry, i.e. related to RFC
> 4566 found at:
> www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml
> 
> In registry: att-field (both session and media level)
> 
> Please add the following SDP Attributes:
> 
> SDP Name: control
> Reference: RFC 2326
> 
> SDP Name: range
> Reference: RFC 2326
> 
> SDP Name: etag
> Reference: RFC 2326
> 
> Contact: Authors of RFC 2326
> 
> Thanks
> 
> Magnus Westerlund

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From fluffy@cisco.com  Tue Aug 13 15:54:30 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] UP: Can a partial offer be a full offer?
Thread-Index: AQHOmHgF5DCVxsS/xE6h7VSvSxBlBw==
Date: Tue, 13 Aug 2013 22:54:10 +0000
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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You only want to send the m blocks that have changed because if you send ot=
her ones, you up the chances of both sides sending the same m block at the =
same time and being back into the glare problem.=20

It seems to me the options are either 1 or 2 would make sense=20

1)  the sender SHOULD only send m m blocks with a change (which might be al=
l of them) and receiver MUST be prepared to receive any number.=20

2) you can only send one m block in  a partial Off or Ans - but you can hav=
e several outstanding at a time



On Aug 5, 2013, at 10:35 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
> =20
> I see no issue with multiple m- lines - just wanted to verify that it wou=
ld be allowed :)
> =20
> Regards,
> =20
> Christer
> =20
>=20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)
>=20
> -----Original Message-----
> From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> To: Christer Holmberg [christer.holmberg@ericsson.com]
> CC: mmusic [mmusic@ietf.org]
> Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
>=20
> On Aug 5, 2013, at 11:17 AM, Christer Holmberg <christer.holmberg@ericsso=
n.com> wrote:
>=20
> > Note that my re-INVITE/UPDATE example is based on the assumption that o=
ne can use partial offers/answers with SIP, on the wire. Not sure if that w=
as part of the agreed scope for the UP.
>=20
> Definitely NOT agreed on.  Doesn't mean it won't happen in the future, bu=
t the scope of UP was limited to WebRTC for now.  We can debate which parts=
 of UP are generically applicable, and which not.  For SIP, this isn't the =
right WG for that debate.
>=20
>=20
> > But, even without that, I think the question on including all m- lines =
on a partial offer still applies :)
>=20
> I don't see why WebRTC would restrict itself in any way regarding the num=
ber of m-lines in a partial offer/answer.  Nothing "breaks" if all m-lines =
are in it, does it?  The real question is if it can be greater than *one* m=
-line.  If it can, and I think it can, then I can't see how the exact quant=
ity of them matters.  The only downside is you increase the chance of glare=
, I assume.
>=20
> Afaik, the 'mid' attribute is there to identity which m-lines in the part=
ial offer correspond to which m-lines in the full ordered set. (the 'mid' c=
ould have even been an index number, except it's also used to avoid glare w=
hen adding m-lines, which a numeric index wouldn't have done)  So what's th=
e issue with one or multiple m-lines?
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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On Aug 6, 2013, at 10:14 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

>=20
> Hi,
> =20
> It has been indicated that POFs (partial offers) and PANs (partial answer=
s) form a stand-alone mechanism, instead of being an extension of RFC3264.

uh, I don't think so. THey seem sot be an optional extension to 3264

>=20
> Now, assuming that is true, we still need to describe the interaction bet=
ween POFs, PANs and 3264 Offer/Answers.
> =20
> Because, if I send a POF, and then receive a 3264 Offer (including the sa=
me m- line as the POF),  there is a glare and I need to reject the 3264 Off=
er.
> =20
> In the same way, if I send a 3264 Offer, and receive a POF (including the=
 same m- line as the 3264 Offer), there is a glare and I need to reject the=
 POF.
> =20
> I ASSUME the above is clear to everyone, including the authors of the UP,=
 but IF someone has a different opinion I'd like to hear about it :)
> =20
> Regards,
> =20
> Christer
> =20
> =20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] unified plan glareless atomicity improvement
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On Jul 31, 2013, at 6:16 AM, Martin Thomson <martin.thomson@gmail.com> wrot=
e:

> I think that the only guidance needed is something like:
> "Implementations MAY choose to exclusively use m-lines with a=3Dsendonly
> or a=3Drecvonly in order to further reduce the probably of encountering
> glare."

that can't be the default for non bundle lines or it won't work with existi=
ng stuff which was the whole point of this. It's not clear it helps reduce =
glare in general.








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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in	bundled m- lines (SDP Offers only)
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On Aug 7, 2013, at 3:50 PM, Parthasarathi R <partha@parthasarathi.co.in> wr=
ote:

> Hi all,
>=20
> It is a concern area for UP plan in case it works only for WebRTC and not
> for SIP/general. The changing current BUNDLE approach breaks interop with
> SIP/legacy endpoints.=20
>=20
> Thanks
> Partha

I disagree. The whole point was to make it work and I believe it does with =
the stuff I have tested.=20




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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] New Liaison Statement, "LS on Missing IANA Registration Information on SDP Attributes in RFC 2326"
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Subject: Re: [MMUSIC] New Liaison Statement, "LS on Missing IANA Registration Information on SDP Attributes in RFC 2326"
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Sounds reasonable to me.

On Aug 13, 2013, at 7:10 AM, Magnus Westerlund <magnus.westerlund@ericsson.=
com> wrote:

>=20
>=20
> On 2013-07-29 09:53, Ari Ker=E4nen wrote:
>> MMUSIC,
>>=20
>> We have received a liaison statement from 3GPP regarding missing SDP
>> attributes at the IANA registry (see the e-mail below for details).
>>=20
>> As the statement says:
>>> SA4 has identified that IANA registrations of three SDP attributes
>>> defined in RFC 2326, namely =91control=92, =91range=92 and =91etag=92, =
are missing
>>=20
>> As a solution, the RTSP 2.0 draft already registers 'control' and
>> 'range' for RFC 2326 and RFC 2326bis
>> (http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-34#section-22.1=
5).
>> However, in RFC2326bis 'etag' is renamed into 'mtag'.
>>=20
>> Hence, it seems that we should also register 'etag' but for that
>> reference RTSP 1.0 (RFC 2326) only.
>>=20
>> Please review the liaison statement and our proposed solution and send
>> your comments to us and to the list.
>=20
> WG,
>=20
> I volunteer to send an email registering these SDP attributes to IANA so
> that a respond can be sent to 3GPP SA4 that they attribute has been
> registered. I propose to include all attributes in that registration. I
> can clarify in the RFC 2326bis draft that the SDP registrations are an
> update to what we put in at this stage. I have also chosen to put "etag"
> in the both category due to that is what has been done in the RFC2326bis
> draft as the original text is far from clear if it is a session only or n=
ot.
>=20
> Thus I propose to send the below to IANA on Monday (19th of Aug) unless
> someone disagree.
>=20
>=20
>> Dear IANA,
>>=20
>> It has come to our attention that the SDP attributes defined in RFC 2326
>> are not registered in the SDP attributes registry, i.e. related to RFC
>> 4566 found at:
>> www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml
>>=20
>> In registry: att-field (both session and media level)
>>=20
>> Please add the following SDP Attributes:
>>=20
>> SDP Name: control
>> Reference: RFC 2326
>>=20
>> SDP Name: range
>> Reference: RFC 2326
>>=20
>> SDP Name: etag
>> Reference: RFC 2326
>>=20
>> Contact: Authors of RFC 2326
>>=20
>> Thanks
>>=20
>> Magnus Westerlund
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
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From martin.thomson@gmail.com  Tue Aug 13 16:27:29 2013
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From: Martin Thomson <martin.thomson@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 13 August 2013 23:58, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>
> On Jul 31, 2013, at 6:16 AM, Martin Thomson <martin.thomson@gmail.com> wrote:
>
>> I think that the only guidance needed is something like:
>> "Implementations MAY choose to exclusively use m-lines with a=sendonly
>> or a=recvonly in order to further reduce the probably of encountering
>> glare."
>
> that can't be the default for non bundle lines or it won't work with existing stuff which was the whole point of this. It's not clear it helps reduce glare in general.

You are somewhat correct.  I'm sure that it will "work" for some value
of "work", but that might not correspond with anything useful.

I was operating under the assumption that partial offers/answers
wouldn't reach old stuff.

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On Aug 8, 2013, at 12:18 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> =20
> As has been indicated, bundle-only can only be used when one knows that t=
he receiver, and possible intermediaries etc, support it.
> =20
>=20

I don't think that is the case. It can be used in an offer regardless of if=
 you know the answer can support it or not.=20

Related to other messages on the thread. Yes, I understand that ICE will ha=
ve a "clean up O/A" to fix all the ports to be the ones actually in use but=
 media flows before that "clean up". I have zero interest in exploding all =
this again to discuss a solution that would require more round trips before=
 media could flow and be less likely to work with existing SIP stuff. I thi=
nk the current design is better than one that would require multiple O/A to=
 get media to flow.=20



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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On Aug 9, 2013, at 10:16 AM, Parthasarathi R <partha@parthasarathi.co.in> w=
rote:

> This approach also
> Violates Sec 8.2 of RFC 3264 requirement
>=20
> " A stream that is offered with a port of zero MUST be marked with port
>   zero in the answer."

No this would not be violated. When one side was 3264 only, this would stil=
l happen. But when both sides support bundle-only, then this restriction no=
 longer holds. The bundle draft as an extension to 3264 needs to clearly de=
fine all that.=20


From pkyzivat@alum.mit.edu  Tue Aug 13 16:32:41 2013
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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A related thing that affects this:

session level attributes that affect each media level. E.g., session 
level c-line.

Do we assume that you must send a full offer to change such things?
(Of course you might choose to avoid this.)

Another thing: ISTM that every POF and PAN whould carry an o-line. That 
will indicate what it updates. And each one then needs to increment the 
sequence number.

	Thanks,
	Paul

On 8/14/13 12:54 AM, Cullen Jennings (fluffy) wrote:
>
> You only want to send the m blocks that have changed because if you send other ones, you up the chances of both sides sending the same m block at the same time and being back into the glare problem.
>
> It seems to me the options are either 1 or 2 would make sense
>
> 1)  the sender SHOULD only send m m blocks with a change (which might be all of them) and receiver MUST be prepared to receive any number.
>
> 2) you can only send one m block in  a partial Off or Ans - but you can have several outstanding at a time
>
>
>
> On Aug 5, 2013, at 10:35 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> I see no issue with multiple m- lines - just wanted to verify that it would be allowed :)
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> Sent from Windows using TouchDown (www.nitrodesk.com)
>>
>> -----Original Message-----
>> From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
>> To: Christer Holmberg [christer.holmberg@ericsson.com]
>> CC: mmusic [mmusic@ietf.org]
>> Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
>>
>> On Aug 5, 2013, at 11:17 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>
>>> Note that my re-INVITE/UPDATE example is based on the assumption that one can use partial offers/answers with SIP, on the wire. Not sure if that was part of the agreed scope for the UP.
>>
>> Definitely NOT agreed on.  Doesn't mean it won't happen in the future, but the scope of UP was limited to WebRTC for now.  We can debate which parts of UP are generically applicable, and which not.  For SIP, this isn't the right WG for that debate.
>>
>>
>>> But, even without that, I think the question on including all m- lines on a partial offer still applies :)
>>
>> I don't see why WebRTC would restrict itself in any way regarding the number of m-lines in a partial offer/answer.  Nothing "breaks" if all m-lines are in it, does it?  The real question is if it can be greater than *one* m-line.  If it can, and I think it can, then I can't see how the exact quantity of them matters.  The only downside is you increase the chance of glare, I assume.
>>
>> Afaik, the 'mid' attribute is there to identity which m-lines in the partial offer correspond to which m-lines in the full ordered set. (the 'mid' could have even been an index number, except it's also used to avoid glare when adding m-lines, which a numeric index wouldn't have done)  So what's the issue with one or multiple m-lines?
>>
>> -hadriel
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
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From pkyzivat@alum.mit.edu  Tue Aug 13 16:39:48 2013
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE QUESTIONS Q10/Q11: Scope of port zero usage in bundled m- lines (SDP Offers only)
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On 8/7/13 7:52 PM, Hadriel Kaplan wrote:
>
> On Aug 6, 2013, at 11:49 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> I don't really understand the "only for WebRTC" agreement.
>> I do see how defining extensions to O/A might be done only for WebRTC and not for 3264. (I don't *agree*, but I do *understand* that.)
>>
>> But BUNDLE is a change to SDP, and SDP extensions are required to specify how they interact with O/A.
>
> Is that actually true?  I don't see that requirement in RFC 4566 IANA registration procedures, nor in IANA's SDP parameters registry.

I don't know if it is written anywhere or not, but as a member of the 
SDP directorate *I* assume that SDP extensions need to discuss the O/A 
impact. Even before the SDP directorate we have been bugging authors 
about that.

Sometimes that impact may be *none*. I suppose it is conceivable that 
there would be an SDP extension that isn't intended to be used in any 
O/A context. But if that is the case, then I would expect *that* to be 
specified.

	Thanks,
	Paul

>> I don't think it makes any sense to say that such changes only apply to WebRTC. They should apply to anything that chooses to claim support to the BUNDLE draft.
>
> OK, but in a SIP-based usage, I think it's dangerous to offer port=0 media lines and negotiate to use those media lines with the far end due to an answer containing a new bundle attribute, without sending another offer indicating you're using those media lines.
>
> So I think if we want to use BUNDLE in SIP as well, we should really require it to do another offer/answer round.  In a WebRTC usage (ie, in Javascript API and JSEP), it can avoid the second round.
>
> One way to describe this in a BUNDLE RFC, I think, is to say it MUST do another offer/answer exchange, except when used in [draft-jsep] or as an application programming interface. (or something like that)
>
> -hadriel
>
>


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Subject: [MMUSIC] Will partial offers/answers leak to legacy stuff?
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On 8/14/13 1:27 AM, Martin Thomson wrote:

> I was operating under the assumption that partial offers/answers
> wouldn't reach old stuff.

Well, when we hear others assert such things in other contexts we give 
them a very hard time, and tell them that in spite of all their best 
intentions things will leak.

E.g., private network addresses, asserted identity.

	Thanks,
	Paul



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 8:34 AM, Martin Thomson <martin.thomson@gmail.com> wrot=
e:

> This formula is just a recommendation (a SHOULD, to be fair)

I think you need to read 2119 and understand the analysis was done when des=
igning ICE and came to the conclusion there were order of checks needs to b=
e coordinated between the two sides.=20

(I'm not saying that we can't find a way to improve on 5245, I'm just sayin=
g pretending you can ignore it because it has a SHOULD won't work)



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Cc: Jonathan Lennox <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On some NATs it was clear that from the source code that the rate limit was=
 caused by how fast the NAT could actually create new connections and it wa=
s the same code path for all address families. So at least in some cases, t=
he fact that the address family changed does not speed up the number of map=
ping per second the NAT can create. Hard do know how general this is.=20

Jonathan's Teredo point is very insightful.=20



On Aug 1, 2013, at 2:19 AM, Pal Martinsen (palmarti) <palmarti@cisco.com> w=
rote:

> Hi,
>=20
> I want to revisit a discussion from an earlier version of the draft.=20
>=20
> An earlier version of the draft suggested to do IPv4 and IPv6 checks in p=
arallel. I like that idea because:
>=20
> 1. It really speeds up the connectivity check phase
> 2. Should have no impact on the NAT port allocation problem. (Or is there=
 a way for an IPv6 conn check to have a negative impact on the IPv4 NAT?)
> 3. Easy to implement. I would just change the way I traverse the the orde=
red checklist. Send first available IPv6 test, then search the list for the=
 first available IPv4 check. To avoid timing issues a ice-option might be u=
sed to say that the checklist is done in parallel if it contains different =
address types. (Do we want to support IPv7.. :-) )
>=20
> .-.
> P=E5l-Erik
>=20
> On Jul 31, 2013, at 18:16 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>=20
>> On 31 July 2013 17:56, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote:
>>> overly complex
>>=20
>> Yes.  Given that this is a very small part of a spec that already fits
>> that description, we need to be careful.
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Tue Aug 13 16:52:55 2013
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Date: Wed, 14 Aug 2013 01:52:37 +0200
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Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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On 8/14/13 1:28 AM, Cullen Jennings (fluffy) wrote:
>
> On Aug 8, 2013, at 12:18 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>>
>> As has been indicated, bundle-only can only be used when one knows that the receiver, and possible intermediaries etc, support it.
>>
>>
>
> I don't think that is the case. It can be used in an offer regardless of if you know the answer can support it or not.

Agreed.

> Related to other messages on the thread. Yes, I understand that ICE will have a "clean up O/A" to fix all the ports to be the ones actually in use but media flows before that "clean up".

It may in some cases, not in others. (When intermediaries gate flows 
based on the answer then the media may not flow immediately.
Or maybe it will if the intermediary is sufficiently bundle-aware.

> I have zero interest in exploding all this again to discuss a solution that would require more round trips before media could flow and be less likely to work with existing SIP stuff. I think the current design is better than one that would require multiple O/A to get media to flow.

Exploring *what* again?
Its not like bundle-only has had such deep discussion.

What's clear is that bundle-only is a somewhat complex mechanism.
It isn't required if you are willing to do a 2nd o/a for the m-lines 
that would otherwise be bundle-only. It would indeed require another 
round trip before media could flow for *those* m-lines. Presumably they 
are "lower priority", so is that really a big deal? Is the extra 
machinery of bundle-only justified for this case?

	Thanks,
	Paul


From fluffy@cisco.com  Tue Aug 13 17:05:52 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Pr...
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Pr...
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The DTMF approach is spectacularly irritating. The user often gets an earfu=
l of DTMF and is really confused about why they got DTMF.=20

I don't care about the MUST implement all three but I think it is a bad ide=
a to say that the offer SHOULD use DTMF. I'd like to see DTMF be a "MAY" us=
e in the offer.=20


On uuie, it's might be worth noting that a valid reason not to use uuie in =
the offer is that something else has already used theses bits.=20




On Aug 9, 2013, at 4:10 PM, Flemming Andreasen <fandreas@cisco.com> wrote:

> Greetings MMUSIC
>=20
> As you can see per the below, the IESG approved the Circuit-Switched Bear=
ers draft for publication, however before doing so, they requested a couple=
 of changes. The resulting approved document can be found at http://tools.i=
etf.org/html/draft-ietf-mmusic-sdp-cs-21
>=20
> One of the changes introduced resulted in all 3 correlation mechanisms no=
w being mandatory-to-implement (previously they were not):
> <quote>
>    Implementations MUST support the three correlation mechanisms
>    specified in=20
> Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
> ,
>    and SHOULD include all supported mechanisms the initial Offer.
>=20
> </quote>
> Since this is a non-trivial change that has not been discussed by the WG =
(and we know of at least one person having expressed concerns with it off-l=
ine), we are hereby solicting the WG participants for input on changes intr=
oduced as a result of the IESG review with a focus on the above. In doing s=
o, please keep in mind that we are developing standards that work in the ge=
neral Internet.=20
>=20
>=20
> The following URL provides a diff-file showing the changes resulting from=
 the IESG review:
>     http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&=
url2=3Ddraft-ietf-mmusic-sdp-cs-21.txt
>=20
> Comments are due by August 23 (two weeks from today)
>=20
> Thanks=20
>=20
> -- Flemming (MMUSIC co-chair)
>=20
>=20
>=20
> -------- Original Message --------
> Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Ex=
tension For Setting Audio and Video Media Streams Over Circuit-Switched Bea=
rers In The Public Switched Telephone Network (PSTN)' to Proposed Standard =
(draft-ietf-mmusic-sdp-cs-21.txt)
> Date:	Mon, 1 Jul 2013 11:56:23 -0700
> From:	The IESG <iesg-secretary@ietf.org>
> To:	IETF-Announce <ietf-announce@ietf.org>
> CC:	mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list <mmu=
sic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
>=20
> The IESG has approved the following document:
> - 'Session Description Protocol (SDP) Extension For Setting Audio and
>    Video Media Streams Over Circuit-Switched Bearers In The Public
>    Switched Telephone Network (PSTN)'
>   (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard
>=20
> This document is the product of the Multiparty Multimedia Session Control
> Working Group.
>=20
> The IESG contact persons are Gonzalo Camarillo and Richard Barnes.
>=20
> A URL of this Internet Draft is:
>=20
> http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>=20
>=20
>=20
>=20
>=20
> *Technical Summary
> *
>=20
> The document describes use cases, requirements, and protocol extensions
> for using the Session Description Protocol (SDP) Offer/Answer model for
> establishing audio and video media streams over circuit-switched bearers
> in the Publich Switched Telephone Network (PSTN)
>=20
> *Working Group Summary
> *
>=20
> The WG had some discussion around the format to use for E.164 numbers
> and whether to align this with the existing definition in RFC 3108. The
> RFC 3108 definition was seen as deficient and the WG agreed it was
> better to align with relevant parts of the tel URI format defined in RFC
> 3966, not least since SDP address types are defined in the context of a
> particular network type, and hence RFC 3108 compatibility is not a
> concern (the implication is that the "E164" address type may differ
> between network types in SDP).
>=20
> *Document Quality
> *
>=20
> There are currently no known implementations of the draft, however the
> draft is a dependency for 3GPP, so future implementations are expected.
>=20
> The document has received good overall review in the WG, some of which
> resulted in changes to the detailed specification. The document has been
> reviewed in detail several times, including of the last few versions.
> The major contributors to these as well as earlier discussions are
> listed in the Acknowledgements section of the document.
>=20
>=20
> *Personnel
> *
>=20
> Document Shepherd: Flemming Andreasen
> Responsible AD: Gonzalo Camarillo=20
> _______________________________________________
> mmusic mailing list
>=20
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=20
> .
>=20
>=20
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From martin.thomson@gmail.com  Tue Aug 13 17:15:57 2013
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References: <CABkgnnVEzTABa4-YZWgBwiEBt6DXCr5ugHY0KOfkw-x5Y91Lyw@mail.gmail.com> <BLU169-W25092106DE55365B8C883893570@phx.gbl> <51F8D6B6.6080109@ericsson.com> <913383AAA69FF945B8F946018B75898A19008F0E@xmb-rcd-x10.cisco.com> <CABkgnnVSkzymi6faeQDpfk+pWVBtY1PWmZ456=Mm+ViCsoBS6w@mail.gmail.com> <C5E08FE080ACFD4DAE31E4BDBF944EB1136790C0@xmb-aln-x02.cisco.com>
Date: Wed, 14 Aug 2013 01:15:54 +0100
Message-ID: <CABkgnnVp-t5+Gg6nZ=gX8_RMwgNRgtGsjLjRqKTp=NBfC_qeyQ@mail.gmail.com>
From: Martin Thomson <martin.thomson@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 14 August 2013 00:46, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>> This formula is just a recommendation (a SHOULD, to be fair)
>
> I think you need to read 2119 and understand the analysis was done when designing ICE and came to the conclusion there were order of checks needs to be coordinated between the two sides.

I think that you misunderstand.  That particular requirement is
addressed by a MUST in Section 5.7.2
(http://tools.ietf.org/html/rfc5245#section-5.7.2), which provides the
ordering guarantee.  The text in 4.1.2.1 uses SHOULD, and I'm aware of
implementations that choose not to follow this guidance for the exact
same reasons that this draft (the eyeballs draft, that is) was
created.

From martin.thomson@gmail.com  Tue Aug 13 17:23:55 2013
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Date: Wed, 14 Aug 2013 01:23:43 +0100
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From: Martin Thomson <martin.thomson@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Will partial offers/answers leak to legacy stuff?
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On 14 August 2013 00:43, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> On 8/14/13 1:27 AM, Martin Thomson wrote:
>
>> I was operating under the assumption that partial offers/answers
>> wouldn't reach old stuff.
>
>
> Well, when we hear others assert such things in other contexts we give them
> a very hard time, and tell them that in spite of all their best intentions
> things will leak.

This is an interesting point, but I'm not sure if it's relevant
without knowing more about how people intend to standardize the use of
these things.  If you send them in SIP INVITE messages, then I'm
willing to entertain the discussion about leakage - we have to worry
at that point.

Until that happens, I will regard these partial thingies as square
pegs and SIP as a round hole.  And, while I expect that someone will
come along with a hammer shortly, allow me at least that brief time to
avoid worrying about these sorts of crazy things.

--Martin

From fluffy@cisco.com  Tue Aug 13 17:32:33 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Sorry, thought you were talking about the other thing.=20

On Aug 13, 2013, at 6:15 PM, Martin Thomson <martin.thomson@gmail.com> wrot=
e:

> On 14 August 2013 00:46, Cullen Jennings (fluffy) <fluffy@cisco.com> wrot=
e:
>>> This formula is just a recommendation (a SHOULD, to be fair)
>>=20
>> I think you need to read 2119 and understand the analysis was done when =
designing ICE and came to the conclusion there were order of checks needs t=
o be coordinated between the two sides.
>=20
> I think that you misunderstand.  That particular requirement is
> addressed by a MUST in Section 5.7.2
> (http://tools.ietf.org/html/rfc5245#section-5.7.2), which provides the
> ordering guarantee.  The text in 4.1.2.1 uses SHOULD, and I'm aware of
> implementations that choose not to follow this guidance for the exact
> same reasons that this draft (the eyeballs draft, that is) was
> created.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Tue Aug 13 19:24:20 2013
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Pr...
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I have no idea how these choices ended up in the draft.
IIRC they are the same choices used in VIPR. At least there I thought 
the point was that none of them are guaranteed to work, so having 
multiple choices increases the chances of success. But I think that 
means offering them all and then trying them successively until one 
works, or until one fails in a way that convinces you not to try any more.

	Thanks,
	Paul

On 8/14/13 2:05 AM, Cullen Jennings (fluffy) wrote:
>
>
> The DTMF approach is spectacularly irritating. The user often gets an earful of DTMF and is really confused about why they got DTMF.
>
> I don't care about the MUST implement all three but I think it is a bad idea to say that the offer SHOULD use DTMF. I'd like to see DTMF be a "MAY" use in the offer.
>
>
> On uuie, it's might be worth noting that a valid reason not to use uuie in the offer is that something else has already used theses bits.
>
>
>
>
> On Aug 9, 2013, at 4:10 PM, Flemming Andreasen <fandreas@cisco.com> wrote:
>
>> Greetings MMUSIC
>>
>> As you can see per the below, the IESG approved the Circuit-Switched Bearers draft for publication, however before doing so, they requested a couple of changes. The resulting approved document can be found at http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21
>>
>> One of the changes introduced resulted in all 3 correlation mechanisms now being mandatory-to-implement (previously they were not):
>> <quote>
>>     Implementations MUST support the three correlation mechanisms
>>     specified in
>> Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
>> ,
>>     and SHOULD include all supported mechanisms the initial Offer.
>>
>> </quote>
>> Since this is a non-trivial change that has not been discussed by the WG (and we know of at least one person having expressed concerns with it off-line), we are hereby solicting the WG participants for input on changes introduced as a result of the IESG review with a focus on the above. In doing so, please keep in mind that we are developing standards that work in the general Internet.
>>
>>
>> The following URL provides a diff-file showing the changes resulting from the IESG review:
>>      http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&url2=draft-ietf-mmusic-sdp-cs-21.txt
>>
>> Comments are due by August 23 (two weeks from today)
>>
>> Thanks
>>
>> -- Flemming (MMUSIC co-chair)
>>
>>
>>
>> -------- Original Message --------
>> Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
>> Date:	Mon, 1 Jul 2013 11:56:23 -0700
>> From:	The IESG <iesg-secretary@ietf.org>
>> To:	IETF-Announce <ietf-announce@ietf.org>
>> CC:	mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list <mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
>>
>> The IESG has approved the following document:
>> - 'Session Description Protocol (SDP) Extension For Setting Audio and
>>     Video Media Streams Over Circuit-Switched Bearers In The Public
>>     Switched Telephone Network (PSTN)'
>>    (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard
>>
>> This document is the product of the Multiparty Multimedia Session Control
>> Working Group.
>>
>> The IESG contact persons are Gonzalo Camarillo and Richard Barnes.
>>
>> A URL of this Internet Draft is:
>>
>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>>
>>
>>
>>
>>
>> *Technical Summary
>> *
>>
>> The document describes use cases, requirements, and protocol extensions
>> for using the Session Description Protocol (SDP) Offer/Answer model for
>> establishing audio and video media streams over circuit-switched bearers
>> in the Publich Switched Telephone Network (PSTN)
>>
>> *Working Group Summary
>> *
>>
>> The WG had some discussion around the format to use for E.164 numbers
>> and whether to align this with the existing definition in RFC 3108. The
>> RFC 3108 definition was seen as deficient and the WG agreed it was
>> better to align with relevant parts of the tel URI format defined in RFC
>> 3966, not least since SDP address types are defined in the context of a
>> particular network type, and hence RFC 3108 compatibility is not a
>> concern (the implication is that the "E164" address type may differ
>> between network types in SDP).
>>
>> *Document Quality
>> *
>>
>> There are currently no known implementations of the draft, however the
>> draft is a dependency for 3GPP, so future implementations are expected.
>>
>> The document has received good overall review in the WG, some of which
>> resulted in changes to the detailed specification. The document has been
>> reviewed in detail several times, including of the last few versions.
>> The major contributors to these as well as earlier discussions are
>> listed in the Acknowledgements section of the document.
>>
>>
>> *Personnel
>> *
>>
>> Document Shepherd: Flemming Andreasen
>> Responsible AD: Gonzalo Camarillo
>> _______________________________________________
>> mmusic mailing list
>>
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>> .
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From fippo@goodadvice.pages.de  Tue Aug 13 23:54:47 2013
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Date: Wed, 14 Aug 2013 08:54:23 +0200
From: Philipp Hancke <fippo@goodadvice.pages.de>
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Subject: [MMUSIC] RFC 5245: ice-ufrags of length 256 vs stun maximum username length of 512 bytes
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It seems that there is a minor inconsistency between the maximum 
username length in STUN and the way ice-ufrag is defined...

5245 allows ufrags of up to 256 characters:
	-- http://tools.ietf.org/html/rfc5245#section-15.4

The STUN username is formed as RFRAG:LFRAG
	http://tools.ietf.org/html/rfc5245#section-7.1.2.3

If both sides use 256 characters for their ufrags this would result in a 
513 byte STUN username. However, 5389 says the username has to be "less 
than 513 bytes".
	-- http://tools.ietf.org/html/rfc5389#section-15.3

Off by one?

For ice-pwd this isn't a problem, right?

From hadriel.kaplan@oracle.com  Wed Aug 14 07:31:50 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)]
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I propose we remove the UUI and DTMF correlation options, and just leave =
it as the callerid one only for now.

We can leave the ABNF with the current 'ext-mech' so that a future spec =
could define something else if need be, and the IANA registration =
procedures for doing so are already documented in this =
draft-ietf-mmusic-sdp-cs-21.

-hadriel


On Aug 12, 2013, at 8:51 AM, Miguel A. Garcia =
<Miguel.A.Garcia@ericsson.com> wrote:

> Hi Hadriel:
>=20
> As far as I remember, the history of the three mechanisms is kind of =
fuzzy. At some point in time in the past, 3GPP hadn't decided on the =
architecture to enable CS calls over SIP/SDP, and they were looking into =
it. The mechanisms were envisioned by the authors and comments on the =
mailing list, and put at 3GPP's disposal for a choice.
>=20
> I concur with  your concern that just because we mandate to implement =
the 3 of them, these are not going to be implemented. I don't have any =
problem in removing from the draft those mechanisms that we envision =
they won't be used.
>=20
> /Miguel
>=20
> On 10/08/2013 1:28, Hadriel Kaplan wrote:
>> Wow.  I can kinda understand why they'd say that (interop reasons, =
presumably), but that's a big deal.  The 3 mechanisms are very =
different, with very different architectural and deployment =
implications.  I have a very hard time believing any of us would truly =
implement all three.  In fact, I don't truly believe anyone would.
>>=20
>> So... I hate to do this, but can the authors summarize the history of =
how we got 3 mechanisms to begin with, and who would really implement =
them all?
>>=20
>> Or to put it another way, instead of specifying all 3, would it be =
impossible to take a step back and reconsider the real motivating =
use-case (which as far as I know is 3GPP), and only specify the one they =
need/use?
>>=20
>> -hadriel
>>=20
>>=20
>> On Aug 9, 2013, at 6:10 PM, Flemming Andreasen=20
>> <fandreas@cisco.com>
>>  wrote:
>>=20
>>=20
>>> Greetings MMUSIC
>>>=20
>>> As you can see per the below, the IESG approved the Circuit-Switched =
Bearers draft for publication, however before doing so, they requested a =
couple of changes. The resulting approved document can be found at=20
>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21
>>>=20
>>>=20
>>> One of the changes introduced resulted in all 3 correlation =
mechanisms now being mandatory-to-implement (previously they were not):
>>> <quote>
>>>    Implementations MUST support the three correlation mechanisms
>>>    specified in=20
>>> Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
>>> ,
>>>    and SHOULD include all supported mechanisms the initial Offer.
>>>=20
>>> </quote>
>>> Since this is a non-trivial change that has not been discussed by =
the WG (and we know of at least one person having expressed concerns =
with it off-line), we are hereby solicting the WG participants for input =
on changes introduced as a result of the IESG review with a focus on the =
above. In doing so, please keep in mind that we are developing standards =
that work in the general Internet.=20
>>>=20
>>>=20
>>> The following URL provides a diff-file showing the changes resulting =
from the IESG review:
>>>    =20
>>> =
http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&url2=3D=
draft-ietf-mmusic-sdp-cs-21.txt
>>>=20
>>>=20
>>> Comments are due by August 23 (two weeks from today)
>>>=20
>>> Thanks=20
>>>=20
>>> -- Flemming (MMUSIC co-chair)
>>>=20
>>>=20
>>>=20
>>> -------- Original Message --------
>>> Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol =
(SDP) Extension For Setting Audio and Video Media Streams Over =
Circuit-Switched Bearers In The Public Switched Telephone Network =
(PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
>>> Date:	Mon, 1 Jul 2013 11:56:23 -0700
>>> From:	The IESG=20
>>> <iesg-secretary@ietf.org>
>>>=20
>>> To:	IETF-Announce=20
>>> <ietf-announce@ietf.org>
>>>=20
>>> CC:	mmusic chair=20
>>> <mmusic-chairs@tools.ietf.org>, mmusic mailing list =
<mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
>>>=20
>>>=20
>>> The IESG has approved the following document:
>>> - 'Session Description Protocol (SDP) Extension For Setting Audio =
and
>>>    Video Media Streams Over Circuit-Switched Bearers In The Public
>>>    Switched Telephone Network (PSTN)'
>>>   (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard
>>>=20
>>> This document is the product of the Multiparty Multimedia Session =
Control
>>> Working Group.
>>>=20
>>> The IESG contact persons are Gonzalo Camarillo and Richard Barnes.
>>>=20
>>> A URL of this Internet Draft is:
>>>=20
>>>=20
>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>>=20
>>> *Technical Summary
>>> *
>>>=20
>>> The document describes use cases, requirements, and protocol =
extensions
>>> for using the Session Description Protocol (SDP) Offer/Answer model =
for
>>> establishing audio and video media streams over circuit-switched =
bearers
>>> in the Publich Switched Telephone Network (PSTN)
>>>=20
>>> *Working Group Summary
>>> *
>>>=20
>>> The WG had some discussion around the format to use for E.164 =
numbers
>>> and whether to align this with the existing definition in RFC 3108. =
The
>>> RFC 3108 definition was seen as deficient and the WG agreed it was
>>> better to align with relevant parts of the tel URI format defined in =
RFC
>>> 3966, not least since SDP address types are defined in the context =
of a
>>> particular network type, and hence RFC 3108 compatibility is not a
>>> concern (the implication is that the "E164" address type may differ
>>> between network types in SDP).
>>>=20
>>> *Document Quality
>>> *
>>>=20
>>> There are currently no known implementations of the draft, however =
the
>>> draft is a dependency for 3GPP, so future implementations are =
expected.
>>>=20
>>> The document has received good overall review in the WG, some of =
which
>>> resulted in changes to the detailed specification. The document has =
been
>>> reviewed in detail several times, including of the last few =
versions.
>>> The major contributors to these as well as earlier discussions are
>>> listed in the Acknowledgements section of the document.
>>>=20
>>>=20
>>> *Personnel
>>> *
>>>=20
>>> Document Shepherd: Flemming Andreasen
>>> Responsible AD: Gonzalo Camarillo=20
>>> _______________________________________________
>>> mmusic mailing list
>>>=20
>>>=20
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>=20
>>>=20
>>> .
>>>=20
>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> mmusic mailing list
>>>=20
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
>> mmusic mailing list
>>=20
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>>=20
>>=20
>=20
> --=20
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain
>=20


From pkyzivat@alum.mit.edu  Wed Aug 14 07:39:24 2013
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)]
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On 8/14/13 4:31 PM, Hadriel Kaplan wrote:
>
> I propose we remove the UUI and DTMF correlation options, and just leave it as the callerid one only for now.
>
> We can leave the ABNF with the current 'ext-mech' so that a future spec could define something else if need be, and the IANA registration procedures for doing so are already documented in this draft-ietf-mmusic-sdp-cs-21.

I don't think it makes sense to make such a change without some 
discussion whether the result will meet the needs that people have for 
the intended uses of this mechanism.

	Thanks,
	Paul

> -hadriel
>
>
> On Aug 12, 2013, at 8:51 AM, Miguel A. Garcia <Miguel.A.Garcia@ericsson.com> wrote:
>
>> Hi Hadriel:
>>
>> As far as I remember, the history of the three mechanisms is kind of fuzzy. At some point in time in the past, 3GPP hadn't decided on the architecture to enable CS calls over SIP/SDP, and they were looking into it. The mechanisms were envisioned by the authors and comments on the mailing list, and put at 3GPP's disposal for a choice.
>>
>> I concur with  your concern that just because we mandate to implement the 3 of them, these are not going to be implemented. I don't have any problem in removing from the draft those mechanisms that we envision they won't be used.
>>
>> /Miguel
>>
>> On 10/08/2013 1:28, Hadriel Kaplan wrote:
>>> Wow.  I can kinda understand why they'd say that (interop reasons, presumably), but that's a big deal.  The 3 mechanisms are very different, with very different architectural and deployment implications.  I have a very hard time believing any of us would truly implement all three.  In fact, I don't truly believe anyone would.
>>>
>>> So... I hate to do this, but can the authors summarize the history of how we got 3 mechanisms to begin with, and who would really implement them all?
>>>
>>> Or to put it another way, instead of specifying all 3, would it be impossible to take a step back and reconsider the real motivating use-case (which as far as I know is 3GPP), and only specify the one they need/use?
>>>
>>> -hadriel
>>>
>>>
>>> On Aug 9, 2013, at 6:10 PM, Flemming Andreasen
>>> <fandreas@cisco.com>
>>>   wrote:
>>>
>>>
>>>> Greetings MMUSIC
>>>>
>>>> As you can see per the below, the IESG approved the Circuit-Switched Bearers draft for publication, however before doing so, they requested a couple of changes. The resulting approved document can be found at
>>>> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21
>>>>
>>>>
>>>> One of the changes introduced resulted in all 3 correlation mechanisms now being mandatory-to-implement (previously they were not):
>>>> <quote>
>>>>     Implementations MUST support the three correlation mechanisms
>>>>     specified in
>>>> Section 5.2.3.2, Section 5.2.3.3 and Section 5.2.3.4
>>>> ,
>>>>     and SHOULD include all supported mechanisms the initial Offer.
>>>>
>>>> </quote>
>>>> Since this is a non-trivial change that has not been discussed by the WG (and we know of at least one person having expressed concerns with it off-line), we are hereby solicting the WG participants for input on changes introduced as a result of the IESG review with a focus on the above. In doing so, please keep in mind that we are developing standards that work in the general Internet.
>>>>
>>>>
>>>> The following URL provides a diff-file showing the changes resulting from the IESG review:
>>>>
>>>> http://tools.ietf.org/rfcdiff?url1=draft-ietf-mmusic-sdp-cs-18.txt&url2=draft-ietf-mmusic-sdp-cs-21.txt
>>>>
>>>>
>>>> Comments are due by August 23 (two weeks from today)
>>>>
>>>> Thanks
>>>>
>>>> -- Flemming (MMUSIC co-chair)
>>>>
>>>>
>>>>
>>>> -------- Original Message --------
>>>> Subject:	[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
>>>> Date:	Mon, 1 Jul 2013 11:56:23 -0700
>>>> From:	The IESG
>>>> <iesg-secretary@ietf.org>
>>>>
>>>> To:	IETF-Announce
>>>> <ietf-announce@ietf.org>
>>>>
>>>> CC:	mmusic chair
>>>> <mmusic-chairs@tools.ietf.org>, mmusic mailing list <mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
>>>>
>>>>
>>>> The IESG has approved the following document:
>>>> - 'Session Description Protocol (SDP) Extension For Setting Audio and
>>>>     Video Media Streams Over Circuit-Switched Bearers In The Public
>>>>     Switched Telephone Network (PSTN)'
>>>>    (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard
>>>>
>>>> This document is the product of the Multiparty Multimedia Session Control
>>>> Working Group.
>>>>
>>>> The IESG contact persons are Gonzalo Camarillo and Richard Barnes.
>>>>
>>>> A URL of this Internet Draft is:
>>>>
>>>>
>>>> http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *Technical Summary
>>>> *
>>>>
>>>> The document describes use cases, requirements, and protocol extensions
>>>> for using the Session Description Protocol (SDP) Offer/Answer model for
>>>> establishing audio and video media streams over circuit-switched bearers
>>>> in the Publich Switched Telephone Network (PSTN)
>>>>
>>>> *Working Group Summary
>>>> *
>>>>
>>>> The WG had some discussion around the format to use for E.164 numbers
>>>> and whether to align this with the existing definition in RFC 3108. The
>>>> RFC 3108 definition was seen as deficient and the WG agreed it was
>>>> better to align with relevant parts of the tel URI format defined in RFC
>>>> 3966, not least since SDP address types are defined in the context of a
>>>> particular network type, and hence RFC 3108 compatibility is not a
>>>> concern (the implication is that the "E164" address type may differ
>>>> between network types in SDP).
>>>>
>>>> *Document Quality
>>>> *
>>>>
>>>> There are currently no known implementations of the draft, however the
>>>> draft is a dependency for 3GPP, so future implementations are expected.
>>>>
>>>> The document has received good overall review in the WG, some of which
>>>> resulted in changes to the detailed specification. The document has been
>>>> reviewed in detail several times, including of the last few versions.
>>>> The major contributors to these as well as earlier discussions are
>>>> listed in the Acknowledgements section of the document.
>>>>
>>>>
>>>> *Personnel
>>>> *
>>>>
>>>> Document Shepherd: Flemming Andreasen
>>>> Responsible AD: Gonzalo Camarillo
>>>> _______________________________________________
>>>> mmusic mailing list
>>>>
>>>>
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>>
>>>> .
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>>
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> _______________________________________________
>>> mmusic mailing list
>>>
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>>
>>
>> --
>> Miguel A. Garcia
>> +34-91-339-3608
>> Ericsson Spain
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From martin.thomson@gmail.com  Wed Aug 14 09:10:52 2013
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Date: Wed, 14 Aug 2013 17:10:48 +0100
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Subject: Re: [MMUSIC] RFC 5245: ice-ufrags of length 256 vs stun maximum username length of 512 bytes
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On 14 August 2013 07:54, Philipp Hancke <fippo@goodadvice.pages.de> wrote:
>
> Off by one?

Maybe.  It certainly looks like an off-by-one error in the specs,
though you could technically argue that both sets of length
restrictions apply, since they are independent.  After all, not all
users of the STUN username attribute are also doing ICE.

> For ice-pwd this isn't a problem, right?

This doesn't transit the network in STUN packets.  In SDP, the maximum
size is 256 characters, but that's not a limit that other signaling
protocols need to respect.

From pkyzivat@alum.mit.edu  Wed Aug 14 10:28:19 2013
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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I have a fundamental question about this draft:

With Unified Plan we have a unique m-line for every MediaStreamTrack.
Given that, why do we need the msid attribute? Isn't it sufficient to 
define a new semantic ("mediastream"?) for the grouping framework? 
(RFC3388).

There would still be need for a way to convey the application data, but 
that can be done via another attribute ("mediastream-data"?) associated 
with the m-line.

	Thanks,
	Paul

On 8/13/13 2:24 PM, Harald Alvestrand wrote:
> The auto-poster doesn't seem to want to tell mmusic about this....
>
> this is updated according to Unified-plan. Since it was only 1 day
> before the old version had expired, the proofreading might be a little
> light - please point out places where I had missed changing "SSRC" to
> "m-line"!
>
> I retained some description of the old mechanism (using a=ssrc) in an
> appendix. Please state opinions on whether this should be retained as
> informative, promoted to normative, or removed entirely in the next version.
>
> Hope this informs!
>
>         Harald
>
>
> -------- Original Message --------
> Subject: 	New Version Notification for draft-ietf-mmusic-msid-01.txt
> Date: 	Tue, 13 Aug 2013 05:13:42 -0700
> From: 	internet-drafts@ietf.org
> To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand
> <harald@alvestrand.no>
>
>
>
> A new version of I-D, draft-ietf-mmusic-msid-01.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
>
> Filename:	 draft-ietf-mmusic-msid
> Revision:	 01
> Title:		 Cross Session Stream Identification in the Session Description Protocol
> Creation date:	 2013-08-13
> Group:		 mmusic
> Number of pages: 14
> URL:http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
> Status:http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
> Htmlized:http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
> Diff:http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01
>
> Abstract:
>     This document specifies a grouping mechanism for RTP media streams
>     that can be used to specify relations between media streams.
>
>     This mechanism is used to signal the association between the SDP
>     concept of "m-line" and the WebRTC concept of "MediaStream" /
>     "MediaStreamTrack" using SDP signaling.
>
>     This document is a work item of the MMUSIC WG, whose discussion list
>     ismmusic@ietf.org.
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> The IETF Secretariat
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From ari.keranen@ericsson.com  Thu Aug 15 05:11:12 2013
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Subject: Re: [MMUSIC] New Liaison Statement, "LS on Missing IANA Registration Information on SDP Attributes in RFC 2326"
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This looks good to me. Thanks Magnus!


Cheers,
Ari

On 8/13/13 4:10 PM, Magnus Westerlund wrote:
>
>
> On 2013-07-29 09:53, Ari KerÃ¤nen wrote:
>> MMUSIC,
>>
>> We have received a liaison statement from 3GPP regarding missing SDP
>> attributes at the IANA registry (see the e-mail below for details).
>>
>> As the statement says:
>>> SA4 has identified that IANA registrations of three SDP attributes
>>> defined in RFC 2326, namely â€˜controlâ€™, â€˜rangeâ€™ and â€˜etagâ€™, are missing
>>
>> As a solution, the RTSP 2.0 draft already registers 'control' and
>> 'range' for RFC 2326 and RFC 2326bis
>> (http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-34#section-22.15).
>> However, in RFC2326bis 'etag' is renamed into 'mtag'.
>>
>> Hence, it seems that we should also register 'etag' but for that
>> reference RTSP 1.0 (RFC 2326) only.
>>
>> Please review the liaison statement and our proposed solution and send
>> your comments to us and to the list.
>
> WG,
>
> I volunteer to send an email registering these SDP attributes to IANA so
> that a respond can be sent to 3GPP SA4 that they attribute has been
> registered. I propose to include all attributes in that registration. I
> can clarify in the RFC 2326bis draft that the SDP registrations are an
> update to what we put in at this stage. I have also chosen to put "etag"
> in the both category due to that is what has been done in the RFC2326bis
> draft as the original text is far from clear if it is a session only or not.
>
> Thus I propose to send the below to IANA on Monday (19th of Aug) unless
> someone disagree.
>
>
>> Dear IANA,
>>
>> It has come to our attention that the SDP attributes defined in RFC 2326
>> are not registered in the SDP attributes registry, i.e. related to RFC
>> 4566 found at:
>> www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml
>>
>> In registry: att-field (both session and media level)
>>
>> Please add the following SDP Attributes:
>>
>> SDP Name: control
>> Reference: RFC 2326
>>
>> SDP Name: range
>> Reference: RFC 2326
>>
>> SDP Name: etag
>> Reference: RFC 2326
>>
>> Contact: Authors of RFC 2326
>>
>> Thanks
>>
>> Magnus Westerlund
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> FÃ¤rÃ¶gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>


From harald@alvestrand.no  Thu Aug 15 07:59:12 2013
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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On 08/14/2013 07:28 PM, Paul Kyzivat wrote:
> I have a fundamental question about this draft:
>
> With Unified Plan we have a unique m-line for every MediaStreamTrack.
> Given that, why do we need the msid attribute? Isn't it sufficient to
> define a new semantic ("mediastream"?) for the grouping framework?
> (RFC3388).
>
> There would still be need for a way to convey the application data,
> but that can be done via another attribute ("mediastream-data"?)
> associated with the m-line.

I wondered about that too.
We have two IDs to convey:

- The ID of the MediaStream. The grouping can be a new semantic for the
grouping framework, but the actual ID value has to be carried somewhere
- it has to be associated with the group, so it can't be a session-level
attribute.

- The ID of the MediaStreamTrack. Now that MediaStreamTracks correspond
to m-lines, it makes sense to carry this in a media-level attribute.

I'm sure we can design another scheme for carrying those IDs around.
But I do wonder what the positive value of restructuring this is.

>
>     Thanks,
>     Paul
>
> On 8/13/13 2:24 PM, Harald Alvestrand wrote:
>> The auto-poster doesn't seem to want to tell mmusic about this....
>>
>> this is updated according to Unified-plan. Since it was only 1 day
>> before the old version had expired, the proofreading might be a little
>> light - please point out places where I had missed changing "SSRC" to
>> "m-line"!
>>
>> I retained some description of the old mechanism (using a=ssrc) in an
>> appendix. Please state opinions on whether this should be retained as
>> informative, promoted to normative, or removed entirely in the next
>> version.
>>
>> Hope this informs!
>>
>>         Harald
>>
>>
>> -------- Original Message --------
>> Subject:     New Version Notification for draft-ietf-mmusic-msid-01.txt
>> Date:     Tue, 13 Aug 2013 05:13:42 -0700
>> From:     internet-drafts@ietf.org
>> To:     Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand
>> <harald@alvestrand.no>
>>
>>
>>
>> A new version of I-D, draft-ietf-mmusic-msid-01.txt
>> has been successfully submitted by Harald Alvestrand and posted to the
>> IETF repository.
>>
>> Filename:     draft-ietf-mmusic-msid
>> Revision:     01
>> Title:         Cross Session Stream Identification in the Session
>> Description Protocol
>> Creation date:     2013-08-13
>> Group:         mmusic
>> Number of pages: 14
>> URL:http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
>> Status:http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
>> Htmlized:http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
>> Diff:http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01
>>
>> Abstract:
>>     This document specifies a grouping mechanism for RTP media streams
>>     that can be used to specify relations between media streams.
>>
>>     This mechanism is used to signal the association between the SDP
>>     concept of "m-line" and the WebRTC concept of "MediaStream" /
>>     "MediaStreamTrack" using SDP signaling.
>>
>>     This document is a work item of the MMUSIC WG, whose discussion list
>>     ismmusic@ietf.org.
>>
>>
>>
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> The IETF Secretariat
>>
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From martin.thomson@gmail.com  Thu Aug 15 08:32:40 2013
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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On 15 August 2013 15:58, Harald Alvestrand <harald@alvestrand.no> wrote:
> I wondered about that too.
> We have two IDs to convey:
>
> - The ID of the MediaStream. The grouping can be a new semantic for the
> grouping framework, but the actual ID value has to be carried somewhere
> - it has to be associated with the group, so it can't be a session-level
> attribute.
>
> - The ID of the MediaStreamTrack. Now that MediaStreamTracks correspond
> to m-lines, it makes sense to carry this in a media-level attribute.
>
> I'm sure we can design another scheme for carrying those IDs around.
> But I do wonder what the positive value of restructuring this is.

The MediaStream identifier gets lost if the grouping framework is
used.  Unless you define some sort of new mapping attribute, which
could end up looking like a=msid if you aren't careful.

The MediaStreamTrack identifier could be doubled onto the a=mid value.

I've always been lukewarm on transmitting identifiers in this fashion,
but I find it hard to find a better (i.e., cleaner, more obvious, less
redundant) solution than msid, except perhaps this one:

a=appgroup:MediaStreamId Track1 Track2
a=group:LS Track1 Track2
m=...
a=mid:Track1
m=...
a=mid:Track2

--Martin

From partha@parthasarathi.co.in  Thu Aug 15 09:48:37 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>
References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se>	<9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>, <5203D7E7.4070207@nostrum.com>	<7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se>	<1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com>	<5203EB64.70605@nostrum.com>	<65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com>	<5203F37A.8030204@nostrum.com>	<33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com>	<5203F4D8.9040904@nostrum.com>	<9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com>	<5204075B.1080108@nostrum.com> <A2E26DB5-CF47-44DF-8779-D470280977B6@oracle.com> <009d01ce951b$d52aa340$7f7fe9c0$@co.in> <C5E08FE080ACFD4DAE31E4BDBF944EB113678F9D@xmb-aln-x02.cisco.com>
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Date: Thu, 15 Aug 2013 22:18:19 +0530
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Cc: 'mmusic WG' <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Cullen,

The violation of RFC 3264 happens as follows:

1) In case m line has port "0" and bundle-only attribute exists in the
Answer, it is possible to interpret by the offerer in two ways as
	a) Answer is rejected by the answerer wherein bundle-only attribute
has to be ignored. This in inline with RFC 3264. Here, there is a need for
extra offer/answer to confirm bundle is supported or not.
	b) Answer is rejected by the answerer wherein bundle-only attribute
is supported. Here, there is no need of one more offer/answer but the
interop issues with SIP entities will happen.

2) There are middle-boxes which provides QoS based on the current
offer/answer. In case of WebRTC, the simplified topology shall be 

 "Browser----SIP server-----Browser/SIP UA"

In these scenario, SIP server initiates Qos based on offer/answer exchange
between browsers. These SIP servers will fail in case an optional attribute
like "bundle-only" updates RFC 3264 behavior like offer with port 0 and
answer with proper port.

Thanks
Partha

> -----Original Message-----
> From: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]
> Sent: Wednesday, August 14, 2013 4:59 AM
> To: Parthasarathi R
> Cc: mmusic WG
> Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-
> only m- lines?
> 
> 
> On Aug 9, 2013, at 10:16 AM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
> 
> > This approach also
> > Violates Sec 8.2 of RFC 3264 requirement
> >
> > " A stream that is offered with a port of zero MUST be marked with
> port
> >   zero in the answer."
> 
> No this would not be violated. When one side was 3264 only, this would
> still happen. But when both sides support bundle-only, then this
> restriction no longer holds. The bundle draft as an extension to 3264
> needs to clearly define all that.


From partha@parthasarathi.co.in  Thu Aug 15 10:13:48 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Eric Rescorla'" <ekr@rtfm.com>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>
References: <7594FB04B1934943A5C02806D1A2204B1C421C80@ESESSMB209.ericsson.se>	<9AA75667-A597-4EB1-87B4-FA79A2F35C3C@oracle.com>	<5203D7E7.4070207@nostrum.com>	<7594FB04B1934943A5C02806D1A2204B1C422522@ESESSMB209.ericsson.se>	<1687A002-7B01-4468-85DD-FF19BDFDAE14@oracle.com>	<5203EB64.70605@nostrum.com>	<65BCA443-98E1-4843-9522-248DD5B4ED68@oracle.com>	<5203F37A.8030204@nostrum.com>	<33EF0E51-D7E2-42BC-98AA-1A96E2E3CB4D@oracle.com>	<5203F4D8.9040904@nostrum.com>	<9BB1DB2F-32DB-453E-BAEF-3E600EAD3413@oracle.com>	<5204075B.1080108@nostrum.com>	<A2E26DB5-CF47-44DF-8779-D470280977B6@oracle.com>	<7594FB04B1934943A5C02806D1A2204B1C422D7D@ESESSMB209.ericsson.se>	<52051D0C.2070106@alum.mit.edu>	<7594FB04B1934943A5C02806D1A2204B1C42339E@ESESSMB209.ericsson.se>	<CAOJ7v-2LpF6HGApc5V-AA-D=D=EOtFmErmPSrRPKVdM8kQYHtw@mail.gmail.com>	<7594FB04B1934943A5C02806D1A2204B1C427679@ESESSMB208.ericsson.se>	<CAOJ7v-3Az7skFCy7KbNDWoMRUP+MPrVMK9Fj8uVAT_Z-dExJ2w@mail.gmail.com>	<7594FB04B1934943A5C02806 D1A2204B1C43BA02@ESESS MB209.ericsson.se>	<CABcZeBP0VZM2MLvQevQs8q0PcutD3jwQ+zWA73qBZ=6RRZeLhg@mail.gmail.com>	<7594FB04B1934943A5C02806D1A2204B1C43DBA9@ESESSMB209.ericsson.se>	<CABcZeBNvVp9hPw5KxN_HCckyMD1xOB0qh-Nbsenz1hek9qyqgg@mail.gmail.com>	<7594FB04B1934943A5C02806D1A2204B1C440F3B@ESESSMB209.ericsson.se> <CABcZeBMFRtVcJktoaH+9VFbE6XqDUow-Jar-cYhewSo8Zz6dpw@mail.gmail.com>
In-Reply-To: <CABcZeBMFRtVcJktoaH+9VFbE6XqDUow-Jar-cYhewSo8Zz6dpw@mail.gmail.com>
Date: Thu, 15 Aug 2013 22:43:26 +0530
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Cc: "'mmusic_ietf.org'" <mmusic@ietf.org>, 'Paul Kyzivat' <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only	m- lines?
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Eric,

 

<snip>

> For instance, Neither Firefox nor Chrome current does a second O/A
> transaction and yet media flows just fine in most cases, since they are
operating
> in an unmediated environment.

<snip>

 

Till now, my understanding is that Firefox & Chrome provides the running
code which is not fully compliant with all WebRTC specification. In fact, I
had come across the scenario wherein it is not clear in the signaling which
candidate pair is selected by the browser for media. Please correct me in
case the specification is not mandated to perform second O/A to confirm the
port selected for the media.

 

Thanks

Partha

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Eric Rescorla
Sent: Sunday, August 11, 2013 8:21 PM
To: Christer Holmberg
Cc: mmusic_ietf.org; Paul Kyzivat
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?

 

 

 

On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:

Hi,

>> I think making sure all entities have the correct port numbers is
critical in making sure the media will flow.
>
> There is a difference between "making sure the media will flow" and
"before media can
> flow at all".
> For instance, Neither Firefox nor Chrome current does a second O/A
> transaction and yet media flows just fine in most cases, since they are
operating
> in an unmediated environment. Obviously, this is a best-case scenario, but
similarly
> there are going to be mediated environments which are perfectly happy with
a second
> O/A exchange with non-zero ports.

There are also environements where a single offer, with identical port
numbers, work just fine, because the receiver will be able to handle it even
if it doesn't support BUNDLE.

However, we are defining a generic mechanism, supposed to work in different
environments.

 

"different environments" != "arbitrary lossage by any middlebox in the
universe"

 

Again, you need to distinguish between *endpoints* and filtering
middleboxes.

 

 

But, sure, if the community thinks that we shall define different options,
depending on environments etc, we can do that...

 

Which is not what I said.

 

-Ekr

 

Regards,

Christer





-----Original Message-----
From: Eric Rescorla [ekr@rtfm.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]

CC: Justin Uberti [juberti@google.com]; mmusic@ietf.org [mmusic@ietf.org];
Paul Kyzivat [pkyzivat@alum.mit.edu]
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only
m- lines?





On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:

Hi,


>>> Requiring a second offer to add the bundle-only m-lines defeats the
purpose of bundle-only.
>> We already require a second offer. I know that some people, including
yourself, want to change that, but that is nothing we have agreed on.
>>
>>> a) The delay to do an O/A will often exceed the time to allocate
candidates, for small numbers of streams; bundle-only was meant to avoid
this candidate latency.
>> You do not have to allocate ports for the bundle-only m- lines.
>>
>>> b) Once you have the initial answer, you know whether the other side
supports BUNDLE, so you can take the appropriate action (and so there is no
need for bundle-only).
>>>
>>> If SIP can't deal with the bundle-only concept, then WebRTC apps that
care about latency will have to decide whether they are running in
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams will be
bundled in the initial offer; in compatibility mode, 2 offer-answers will be
needed.
>> There are no "modes" in BUNDLE today :)
>>
> Of course there are. The handling in a reinvite case is distinct from the
initial offer, so that is modal behavior.


Sure, but you were talking about "WebRTC-only mode" and "compatibility
mode", and my point is that currently the procedures for initial offers,
subsequent offers etc are the same.


>>> This would be a somewhat unfortunate outcome, so I'd like to see
bundle-only (or something like it) work.
>> I don't think the second offer is going to cause any latency. Also, keep
in mind that you only need the 2nd Offer/Answer during the session
establishment. As agreed in Berlin, in mid-session changes you do NOT need 2
offer/answers.
>
> Please provide support for this statement. It seems clear that a 2nd offer
adds _at least_ 1 RTT latency.


Sure, but I don't think that is going to be a practical problem.



Well, I'd like to see a more convincing argument than "some middleboxes are
sad" when you
do that.




First, there are deployed legacy SIP systems that use even more O/A
transactions during session establishment. And, in your case I assume you
are even going to use something more "lightweight" than SIP to carry your
Offer/Answers.

Second, as WebRTC mandates usage of ICE, you are often going to need a
subsequent O/A anyway.



It's important to distinguish between round-trips that are in the critical
path for media
flowing and those which are not.


-Ekr

 


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<div class=3DSection1>

<p class=3DMsoNormal>Eric,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&lt;snip&gt;<o:p></o:p></p>

<p class=3DMsoNormal>&gt; For instance, Neither Firefox nor Chrome =
current does a
second O/A<br>
&gt; transaction and yet media flows just fine in most cases, since they =
are
operating<br>
&gt; in an unmediated environment.<o:p></o:p></p>

<p class=3DMsoNormal>&lt;snip&gt;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Till now, my understanding is that Firefox &amp; =
Chrome
provides the running code which is not fully compliant with all WebRTC =
specification.
In fact, I had come across the scenario wherein it is not clear in the
signaling which candidate pair is selected by the browser for media. =
Please
correct me in case the specification is not mandated to perform second =
O/A to
confirm the port selected for the media.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Partha<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
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<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf Of =
</b>Eric
Rescorla<br>
<b>Sent:</b> Sunday, August 11, 2013 8:21 PM<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> mmusic_ietf.org; Paul Kyzivat<br>
<b>Subject:</b> Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-only
m- lines?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>On Sun, Aug 11, 2013 at 6:11 AM, Christer Holmberg =
&lt;<a
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;
wrote:<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>Hi,<br>
<br>
&gt;&gt; I think making sure all entities have the correct port numbers =
is
critical in making sure the media will flow.<br>
&gt;<br>
&gt; There is a difference between &quot;making sure the media will =
flow&quot;
and &quot;before media can<br>
&gt; flow at all&quot;.<br>
&gt; For instance, Neither Firefox nor Chrome current does a second =
O/A<br>
&gt; transaction and yet media flows just fine in most cases, since they =
are
operating<br>
&gt; in an unmediated environment. Obviously, this is a best-case =
scenario, but
similarly<br>
&gt; there are going to be mediated environments which are perfectly =
happy with
a second<br>
&gt; O/A exchange with non-zero ports.<o:p></o:p></p>

</div>

<p class=3DMsoNormal>There are also environements where a single offer, =
with
identical port numbers, work just fine, because the receiver will be =
able to
handle it even if it doesn't support BUNDLE.<br>
<br>
However, we are defining a generic mechanism, supposed to work in =
different
environments.<o:p></o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>&quot;different environments&quot; !=3D =
&quot;arbitrary
lossage by any middlebox in the universe&quot;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Again, you need to distinguish between *endpoints* =
and
filtering middleboxes.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal>But, sure, if the community thinks that we shall =
define
different options, depending on environments etc, we can do =
that...<o:p></o:p></p>

</blockquote>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Which is not what I said.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>-Ekr<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<p class=3DMsoNormal>Regards,<br>
<br>
Christer<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><br>
<br>
<br>
<br>
-----Original Message-----<br>
From: Eric Rescorla [<a =
href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>]<br>
To: Christer Holmberg [<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>]<br>
<br>
CC: Justin Uberti [<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>];
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a> [<a
href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>]; Paul Kyzivat [<a
href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>]<br>
Subject: Re: [MMUSIC] BUNDLE goes UP: Do we need port zero for =
bundle-only m-
lines?<br>
<br>
<br>
<br>
<br>
<br>
On Sat, Aug 10, 2013 at 3:08 PM, Christer Holmberg &lt;<a
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt;
wrote:<br>
<br>
Hi,<br>
<br>
<br>
&gt;&gt;&gt; Requiring a second offer to add the bundle-only m-lines =
defeats
the purpose of bundle-only.<br>
&gt;&gt; We already require a second offer. I know that some people, =
including
yourself, want to change that, but that is nothing we have agreed =
on.<br>
&gt;&gt;<br>
&gt;&gt;&gt; a) The delay to do an O/A will often exceed the time to =
allocate
candidates, for small numbers of streams; bundle-only was meant to avoid =
this
candidate latency.<br>
&gt;&gt; You do not have to allocate ports for the bundle-only m- =
lines.<br>
&gt;&gt;<br>
&gt;&gt;&gt; b) Once you have the initial answer, you know whether the =
other
side supports BUNDLE, so you can take the appropriate action (and so =
there is
no need for bundle-only).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If SIP can't deal with the bundle-only concept, then WebRTC =
apps
that care about latency will have to decide whether they are running in
WebRTC-only mode or compatibility mode. In WebRTC-only mode, streams =
will be
bundled in the initial offer; in compatibility mode, 2 offer-answers =
will be
needed.<br>
&gt;&gt; There are no &quot;modes&quot; in BUNDLE today :)<br>
&gt;&gt;<br>
&gt; Of course there are. The handling in a reinvite case is distinct =
from the
initial offer, so that is modal behavior.<br>
<br>
<br>
Sure, but you were talking about &quot;WebRTC-only mode&quot; and
&quot;compatibility mode&quot;, and my point is that currently the =
procedures
for initial offers, subsequent offers etc are the same.<br>
<br>
<br>
&gt;&gt;&gt; This would be a somewhat unfortunate outcome, so I'd like =
to see
bundle-only (or something like it) work.<br>
&gt;&gt; I don't think the second offer is going to cause any latency. =
Also,
keep in mind that you only need the 2nd Offer/Answer during the session
establishment. As agreed in Berlin, in mid-session changes you do NOT =
need 2
offer/answers.<br>
&gt;<br>
&gt; Please provide support for this statement. It seems clear that a =
2nd offer
adds _at least_ 1 RTT latency.<br>
<br>
<br>
Sure, but I don't think that is going to be a practical problem.<br>
<br>
<br>
<br>
Well, I'd like to see a more convincing argument than &quot;some =
middleboxes
are sad&quot; when you<br>
do that.<br>
<br>
<br>
<br>
<br>
First, there are deployed legacy SIP systems that use even more O/A
transactions during session establishment. And, in your case I assume =
you are
even going to use something more &quot;lightweight&quot; than SIP to =
carry your
Offer/Answers.<br>
<br>
Second, as WebRTC mandates usage of ICE, you are often going to need a
subsequent O/A anyway.<br>
<br>
<br>
<br>
It's important to distinguish between round-trips that are in the =
critical path
for media<br>
flowing and those which are not.<br>
<br>
<br>
-Ekr<o:p></o:p></p>

</div>

</div>

</blockquote>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</div>

</div>

</body>

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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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I went through this version, with an eye to alignment with the Unified 
Plan. I have a number of comments, some nits, some not:

General:

The Unified Plan says a single m-line describes one media source. It 
allows this source to consist of multiple RTP Streams (identified by 
SSRC). And it allows associating RTP Streams with m-lines multiple ways, 
not just by SSRC.

This msid draft hasn't fully incorporated those concepts. It still seems 
to assume that a media source corresponds to a single SSRC. It would be 
helpful to have a section discussion the process of associating SSRCs to 
MedisStreamTracks. (And I think this needs to include the possibility 
that an SSRC might be moved from one MediaStreamTrack to another.)

Section 2:

    The identifier is a string of ASCII characters chosen from 0-9, a-z,
    A-Z and - (hyphen), consisting of between 1 and 64 characters.  It
    MUST be unique among the identifier values used in the same SDP
    session.  It is RECOMMENDED that is generated using a random-number
    generator.
...
    There may be multiple msid attributes on a single m-line.  There may
    also be multiple m-lines that have the same value for identifier and
    application data.

What does uniqueness mean in this context, since it can be used multiple 
times, with the same or different m-lines? I *think* you mean that a 
given identifier value is associated with a single MediaStream for the 
duration of the SDP session.

Also, msid attributes aren't "on" an m-line. I think it would be clearer 
to say "Multiple msid attributes may be associated with a single 
m-line." But with the Unified Plan this isn't true.

Also, I find the use of "identifier" and "token" very confusing. It is 
hard to keep track of the semantics intended for those things. It would 
be easier to follow if some more specific terms were used. For example, 
"mediastream-id" instead of "identifier" in the msid and msid-semantics 
attributes, and "semantics" instead of "token" in msid-semantics 
attribute. ("semantics" is what is used in 3388 and 5576.)

Section 4:

    o  When a description is updated to have more SSRCs with the same
       msid value, but different appdata values, the recipient can signal
       to its application that new MediaStreamTracks have been added to
       the media stream.

Is SSRC relevant any longer? s/SSRCs/m-lines/ ???

Also in Section 4:

    In addition to signaling that the track is closed when it disappears
    from the SDP, the track will also be signaled as being closed when
    all associated SSRCs have disappeared by the rules of [RFC3550]
    section 6.3.4 (BYE packet received) and 6.3.5 (timeout).

Here we can't avoid talking about SSRCs. But maybe more is needed about 
what SSRCs are associated with a particular m-line & msid value?

Section 4.1:

    Pre-WebRTC entities will not send msid.  This means that there will
    be some incoming RTP packets with SSRCs where the recipient does not
    know about a corresponding MediaStream id.

In the context of the Unified Plan I think this could be stated better:

    There may be some incoming RTP packets that the recipient cannot
    associate with a MediaStreamTrack.

Also:

    o  An msid-semantic:WMS attribute is present.  In this case, the
       session is WebRTC compatible, and the newly arrived SSRCs are
       either caused by a bug or by timing skew between the arrival of
       ...

I suggest:

    o  An msid-semantic:WMS attribute is present.  In this case, the
       session is WebRTC compatible, and the unassociated packets are
       either caused by a bug or by timing skew between the arrival of
       ...

	Thanks,
	Paul


On 8/13/13 2:24 PM, Harald Alvestrand wrote:
> The auto-poster doesn't seem to want to tell mmusic about this....
>
> this is updated according to Unified-plan. Since it was only 1 day
> before the old version had expired, the proofreading might be a little
> light - please point out places where I had missed changing "SSRC" to
> "m-line"!
>
> I retained some description of the old mechanism (using a=ssrc) in an
> appendix. Please state opinions on whether this should be retained as
> informative, promoted to normative, or removed entirely in the next version.
>
> Hope this informs!
>
>         Harald
>
>
> -------- Original Message --------
> Subject: 	New Version Notification for draft-ietf-mmusic-msid-01.txt
> Date: 	Tue, 13 Aug 2013 05:13:42 -0700
> From: 	internet-drafts@ietf.org
> To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand
> <harald@alvestrand.no>
>
>
>
> A new version of I-D, draft-ietf-mmusic-msid-01.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
>
> Filename:	 draft-ietf-mmusic-msid
> Revision:	 01
> Title:		 Cross Session Stream Identification in the Session Description Protocol
> Creation date:	 2013-08-13
> Group:		 mmusic
> Number of pages: 14
> URL:http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
> Status:http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
> Htmlized:http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
> Diff:http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01
>
> Abstract:
>     This document specifies a grouping mechanism for RTP media streams
>     that can be used to specify relations between media streams.
>
>     This mechanism is used to signal the association between the SDP
>     concept of "m-line" and the WebRTC concept of "MediaStream" /
>     "MediaStreamTrack" using SDP signaling.
>
>     This document is a work item of the MMUSIC WG, whose discussion list
>     ismmusic@ietf.org.
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> The IETF Secretariat
>
>
>
>
>
> _______________________________________________
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>


From christer.holmberg@ericsson.com  Fri Aug 16 00:17:13 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Hi,
=20
>> It has been indicated that POFs (partial offers) and PANs (partial answe=
rs) form a stand-alone mechanism, instead of being an extension of RFC3264.
>
> uh, I don't think so. THey seem sot be an optional extension to 3264

Well, that is not what was indicated during the meeting (if we don't count =
Hadriel :)...

Anyway, I guess that is part of the solution discussions. But, whatever it'=
s an extension or not, I don't think it changes the fact that there will in=
teractions between POF/PANs and 3264 O/As.

Regards,

Christer


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [MMUSIC] BUNDLE goes UP: Do we need port zero for bundle-only m- lines?
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Hi,

>> As has been indicated, bundle-only can only be used when one knows that =
the receiver, and possible intermediaries etc, support it.
> =20
>=20
>
> I don't think that is the case. It can be used in an offer regardless of =
if you know the answer can support it or not.=20
>
>  Related to other messages on the thread. Yes, I understand that ICE will=
 have a "clean up O/A" to fix all the ports to be the ones actually in use =
but media flows before=20
> that "clean up". I have zero interest in exploding all this again to disc=
uss a solution that would require more round trips before media could flow =
and be less likely to work=20
> with existing SIP stuff. I think the current design is better than one th=
at would require multiple O/A to get media to flow.=20

Discuss what???

In general, the current version of BUNDLE is based on the usage of 2 Offer/=
Answers - which is something you have agreed to.

If you only refer to bundle-only lines, that is a discussion that has just =
started...

Regards,

Christer






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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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Hi,

It is ok to only send m- line(s) that are changed.

But, again, when you in a POF/PAN change on m- line explicitly, other m- li=
nes may be changed implicitly. So, if you at the same time explicitly (usin=
g a separate POF/PAN) also try to change those other m- line, there will be=
 a collision.

Or, as Paul says, if multiple simultaneous POF/PANs change session level in=
formation, there will be collision.

MAYBE we could restrict usage of POF/PANs, and say that they MUST NOT be us=
ed to change session level information...

Regards,

Christer

-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Cullen Jennings (fluffy) [mailto:fluffy@cisco.com]=20
L=E4hetetty: 14. elokuuta 2013 1:54
Vastaanottaja: Christer Holmberg
Kopio: Hadriel Kaplan; mmusic
Aihe: Re: [MMUSIC] UP: Can a partial offer be a full offer?


You only want to send the m blocks that have changed because if you send ot=
her ones, you up the chances of both sides sending the same m block at the =
same time and being back into the glare problem.=20

It seems to me the options are either 1 or 2 would make sense=20

1)  the sender SHOULD only send m m blocks with a change (which might be al=
l of them) and receiver MUST be prepared to receive any number.=20

2) you can only send one m block in  a partial Off or Ans - but you can hav=
e several outstanding at a time



On Aug 5, 2013, at 10:35 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
> =20
> I see no issue with multiple m- lines - just wanted to verify that it wou=
ld be allowed :)
> =20
> Regards,
> =20
> Christer
> =20
>=20
>=20
> Sent from Windows using TouchDown (www.nitrodesk.com)
>=20
> -----Original Message-----
> From: Hadriel Kaplan [hadriel.kaplan@oracle.com]
> To: Christer Holmberg [christer.holmberg@ericsson.com]
> CC: mmusic [mmusic@ietf.org]
> Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
>=20
> On Aug 5, 2013, at 11:17 AM, Christer Holmberg <christer.holmberg@ericsso=
n.com> wrote:
>=20
> > Note that my re-INVITE/UPDATE example is based on the assumption that o=
ne can use partial offers/answers with SIP, on the wire. Not sure if that w=
as part of the agreed scope for the UP.
>=20
> Definitely NOT agreed on.  Doesn't mean it won't happen in the future, bu=
t the scope of UP was limited to WebRTC for now.  We can debate which parts=
 of UP are generically applicable, and which not.  For SIP, this isn't the =
right WG for that debate.
>=20
>=20
> > But, even without that, I think the question on including all m- lines =
on a partial offer still applies :)
>=20
> I don't see why WebRTC would restrict itself in any way regarding the num=
ber of m-lines in a partial offer/answer.  Nothing "breaks" if all m-lines =
are in it, does it?  The real question is if it can be greater than *one* m=
-line.  If it can, and I think it can, then I can't see how the exact quant=
ity of them matters.  The only downside is you increase the chance of glare=
, I assume.
>=20
> Afaik, the 'mid' attribute is there to identity which m-lines in the part=
ial offer correspond to which m-lines in the full ordered set. (the 'mid' c=
ould have even been an index number, except it's also used to avoid glare w=
hen adding m-lines, which a numeric index wouldn't have done)  So what's th=
e issue with one or multiple m-lines?
>=20
> -hadriel
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From fippo@goodadvice.pages.de  Fri Aug 16 00:48:36 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] RFC 5245: ice-ufrags of length 256 vs stun maximum username length of 512 bytes
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On Wed, 14 Aug 2013, Martin Thomson wrote:
> On 14 August 2013 07:54, Philipp Hancke <fippo@goodadvice.pages.de> wrote:
>>
>> Off by one?
>
> Maybe.  It certainly looks like an off-by-one error in the specs,
> though you could technically argue that both sets of length
> restrictions apply, since they are independent.  After all, not all
> users of the STUN username attribute are also doing ICE.

Right. I'd actually hope that implementations don't send anything longer 
than 512 bytes. I am not so sure about the comparison on the receiving 
end.
It's probably a good idea to mention this explicitly in 5245bis or
draft-ietf-mmusic-ice-sip-sdp though.

>> For ice-pwd this isn't a problem, right?
>
> This doesn't transit the network in STUN packets.  In SDP, the maximum
> size is 256 characters, but that's not a limit that other signaling
> protocols need to respect.

phew :-)

From martin.thomson@gmail.com  Fri Aug 16 09:13:05 2013
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Subject: Re: [MMUSIC] RFC 5245: ice-ufrags of length 256 vs stun maximum username length of 512 bytes
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On 16 August 2013 00:48, Philipp Hancke <fippo@goodadvice.pages.de> wrote:
> It's probably a good idea to mention this explicitly in 5245bis or
> draft-ietf-mmusic-ice-sip-sdp though.

Mention, or just fix.

I'm guessing that this has never really been tested in practice, so
fixing is probably quite feasible.

From christer.holmberg@ericsson.com  Mon Aug 19 06:03:42 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Hi,

We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, which d=
efines usage of UDPTL over DTLS, in order to provide secure fax.

The draft was previously submitted to DISPATCH. Based on discussions with t=
he ADs and chairs, it was decided that it shall be submitted to MMUSIC (not=
e that no DTLS extensions are needed).

As is indicated in the draft, 3GPP has performed a study on how to provide =
secure fax in the IMS, and the outcome was that secure fax shall be transpo=
rted using UDPTL over
DTLS. However, there is nothing "3GPP/IMS specific" about the mechanism, as=
 UDPTL is commonly used for fax also elsewhere.

Regards,

Christer

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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">We have submitted a new draft, draft-holmberg-mmusic=
-udptl-dtls-00, which defines usage of UDPTL over DTLS, in order to provide=
 secure fax.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft was previously submitted to DISPATCH. Base=
d on discussions with the ADs and chairs, it was decided that it shall be s=
ubmitted to MMUSIC (note that no DTLS extensions are needed).<o:p></o:p></p=
>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">As is indicated in the draft, 3GPP has performed a s=
tudy on how to provide secure fax in the IMS, and the outcome was that secu=
re fax shall be transported using UDPTL over<o:p></o:p></p>
<p class=3D"MsoNormal">DTLS. However, there is nothing &#8220;3GPP/IMS spec=
ific&#8221; about the mechanism, as UDPTL is commonly used for fax also els=
ewhere.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C46787FESESSMB209erics_--

From harald@alvestrand.no  Mon Aug 19 07:57:57 2013
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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On 08/15/2013 07:15 PM, Paul Kyzivat wrote:
> I went through this version, with an eye to alignment with the Unified 
> Plan. I have a number of comments, some nits, some not:

Thanks for doing this quickly!

I've incorporated these changes (with some rewording, as always). I'll 
give this a few more days to stew, and then emit a new version.

>
> General:
>
> The Unified Plan says a single m-line describes one media source. It 
> allows this source to consist of multiple RTP Streams (identified by 
> SSRC). And it allows associating RTP Streams with m-lines multiple 
> ways, not just by SSRC.
>
> This msid draft hasn't fully incorporated those concepts. It still 
> seems to assume that a media source corresponds to a single SSRC. It 
> would be helpful to have a section discussion the process of 
> associating SSRCs to MedisStreamTracks. (And I think this needs to 
> include the possibility that an SSRC might be moved from one 
> MediaStreamTrack to another.)

I tried replacing every occurence of SSRC with m-line, except where that 
didn't make sense (and in Appendix B, of course). Please point at the 
text where I missed, if it's not covered below!

>
> Section 2:
>
>    The identifier is a string of ASCII characters chosen from 0-9, a-z,
>    A-Z and - (hyphen), consisting of between 1 and 64 characters. It
>    MUST be unique among the identifier values used in the same SDP
>    session.  It is RECOMMENDED that is generated using a random-number
>    generator.
> ...
>    There may be multiple msid attributes on a single m-line. There may
>    also be multiple m-lines that have the same value for identifier and
>    application data.
>
> What does uniqueness mean in this context, since it can be used 
> multiple times, with the same or different m-lines? I *think* you mean 
> that a given identifier value is associated with a single MediaStream 
> for the duration of the SDP session.

Yes, but I can't write that in section 2, since MediaStreams aren't 
introduced until section 4. The meaning of "uniqueness" has to be 
defined by the msid-semantic, I think - should I add a sentence saying 
that this is so?

(It's tempting to declare that msid is useless for anything except 
WebRTC. But the Paris (?) IETF meeting told me to write a generic 
mechanism with a WebRTC specialization, so that's how it's written.)


>
> Also, msid attributes aren't "on" an m-line. I think it would be 
> clearer to say "Multiple msid attributes may be associated with a 
> single m-line." But with the Unified Plan this isn't true.

Sorry?

I think it's legal to say

m=video
a=msid: 1234 567
a=msid: ABCD EFG

This would mean (for the WMS context) that the same media content is 
present in both MediaStream 1234 and MediaStream ABCD, with 
MediaStreamTrack ids 567 and EFG, respectively.
>
> Also, I find the use of "identifier" and "token" very confusing. It is 
> hard to keep track of the semantics intended for those things. It 
> would be easier to follow if some more specific terms were used. For 
> example, "mediastream-id" instead of "identifier" in the msid and 
> msid-semantics attributes, and "semantics" instead of "token" in 
> msid-semantics attribute. ("semantics" is what is used in 3388 and 5576.)

You're talking about the ABNF in sections 2 and 3 now, right?
We could use more meaningful names for the tokens, but it's just another 
level of indirection - generally, I don't find that specs become much 
more readable by repeated productions of the form

    name1 = name2

unless we have to refer to the tokens by name from surrounding text.
But your mileage may vary.
>
> Section 4:
>
>    o  When a description is updated to have more SSRCs with the same
>       msid value, but different appdata values, the recipient can signal
>       to its application that new MediaStreamTracks have been added to
>       the media stream.
>
> Is SSRC relevant any longer? s/SSRCs/m-lines/ ???

Yep, I missed that (multiple misses in that section). Will update.

>
> Also in Section 4:
>
>    In addition to signaling that the track is closed when it disappears
>    from the SDP, the track will also be signaled as being closed when
>    all associated SSRCs have disappeared by the rules of [RFC3550]
>    section 6.3.4 (BYE packet received) and 6.3.5 (timeout).
>
> Here we can't avoid talking about SSRCs. But maybe more is needed 
> about what SSRCs are associated with a particular m-line & msid value?

Since this is in section 4 (WebRTC), I can happily point that problem 
off to unified-plan or its successors.
>
> Section 4.1:
>
>    Pre-WebRTC entities will not send msid.  This means that there will
>    be some incoming RTP packets with SSRCs where the recipient does not
>    know about a corresponding MediaStream id.
>
> In the context of the Unified Plan I think this could be stated better:
>
>    There may be some incoming RTP packets that the recipient cannot
>    associate with a MediaStreamTrack.

Rephrased, not quite like this.

>
> Also:
>
>    o  An msid-semantic:WMS attribute is present.  In this case, the
>       session is WebRTC compatible, and the newly arrived SSRCs are
>       either caused by a bug or by timing skew between the arrival of
>       ...
>
> I suggest:
>
>    o  An msid-semantic:WMS attribute is present.  In this case, the
>       session is WebRTC compatible, and the unassociated packets are
>       either caused by a bug or by timing skew between the arrival of

Done.

> ...
>
>     Thanks,
>     Paul
>
>
> On 8/13/13 2:24 PM, Harald Alvestrand wrote:
>> The auto-poster doesn't seem to want to tell mmusic about this....
>>
>> this is updated according to Unified-plan. Since it was only 1 day
>> before the old version had expired, the proofreading might be a little
>> light - please point out places where I had missed changing "SSRC" to
>> "m-line"!
>>
>> I retained some description of the old mechanism (using a=ssrc) in an
>> appendix. Please state opinions on whether this should be retained as
>> informative, promoted to normative, or removed entirely in the next 
>> version.
>>
>> Hope this informs!
>>
>>         Harald
>>
>>
>> -------- Original Message --------
>> Subject:     New Version Notification for draft-ietf-mmusic-msid-01.txt
>> Date:     Tue, 13 Aug 2013 05:13:42 -0700
>> From:     internet-drafts@ietf.org
>> To:     Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand
>> <harald@alvestrand.no>
>>
>>
>>
>> A new version of I-D, draft-ietf-mmusic-msid-01.txt
>> has been successfully submitted by Harald Alvestrand and posted to the
>> IETF repository.
>>
>> Filename:     draft-ietf-mmusic-msid
>> Revision:     01
>> Title:         Cross Session Stream Identification in the Session 
>> Description Protocol
>> Creation date:     2013-08-13
>> Group:         mmusic
>> Number of pages: 14
>> URL:http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
>> Status:http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
>> Htmlized:http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
>> Diff:http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01
>>
>> Abstract:
>>     This document specifies a grouping mechanism for RTP media streams
>>     that can be used to specify relations between media streams.
>>
>>     This mechanism is used to signal the association between the SDP
>>     concept of "m-line" and the WebRTC concept of "MediaStream" /
>>     "MediaStreamTrack" using SDP signaling.
>>
>>     This document is a work item of the MMUSIC WG, whose discussion list
>>     ismmusic@ietf.org.
>>
>>
>>
>>
>>
>> Please note that it may take a couple of minutes from the time of 
>> submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> The IETF Secretariat
>>
>>
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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On 08/16/2013 09:17 AM, Christer Holmberg wrote:
> Hi,
>   
>>> It has been indicated that POFs (partial offers) and PANs (partial answers) form a stand-alone mechanism, instead of being an extension of RFC3264.
>> uh, I don't think so. THey seem sot be an optional extension to 3264
> Well, that is not what was indicated during the meeting (if we don't count Hadriel :)...
>
> Anyway, I guess that is part of the solution discussions. But, whatever it's an extension or not, I don't think it changes the fact that there will interactions between POF/PANs and 3264 O/As.
>
My impression is that we're defining a new mechanism that contains 
3264-style O/A *and* POF/PAN.

Applications will then either use that mechanism or use 3264 only 
(analogous, kind of, to applications either using RTP/AVP or RTP/AVPF).

So I see no "interaction" as such, but the semantics of the new 
mechanism have to be defined, and, if we intend it for use in SIP, how 
to negotiate its use has to be defined.



From christer.holmberg@ericsson.com  Mon Aug 19 09:37:47 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Subject: Re: [MMUSIC] Interactions between POFs, PANs and 3264 Offer/Answers
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Hi Harald,

>>>> It has been indicated that POFs (partial offers) and PANs (partial ans=
wers) form a stand-alone mechanism, instead of being an extension of RFC326=
4.
>>> uh, I don't think so. THey seem sot be an optional extension to 3264
>> Well, that is not what was indicated during the meeting (if we don't cou=
nt Hadriel :)...
>>
>> Anyway, I guess that is part of the solution discussions. But, whatever =
it's an extension or not, I don't think it changes the fact that there will=
 interactions between POF/PANs and 3264 O/As.
>>
>> My impression is that we're defining a new mechanism that contains 3264-=
style O/A *and* POF/PAN.
>
> Applications will then either use that mechanism or use 3264 only (analog=
ous, kind of, to applications either using RTP/AVP or RTP/AVPF).
>
> So I see no "interaction" as such, but the semantics of the new mechanism=
 have to be defined, and, if we intend it for use in SIP, how to negotiate =
its use has to be defined.

The problem with SIP is that, in certain situations (e.g. in a (re-)INVITE =
request or response), you are REQUIRED to send an Offer. Now, whether that =
Offer must be a 3264 Offer, or if some "type" of Offer is also allowed, I d=
on't know :)

Regards,

Christer


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On Aug 19, 2013, at 6:03 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, =
which defines usage of UDPTL over DTLS, in order to provide secure fax.
> =20
> The draft was previously submitted to DISPATCH. Based on discussions =
with the ADs and chairs, it was decided that it shall be submitted to =
MMUSIC (note that no DTLS extensions are needed).
> =20
> As is indicated in the draft, 3GPP has performed a study on how to =
provide secure fax in the IMS, and the outcome was that secure fax shall =
be transported using UDPTL over
> DTLS.

Got a pointer to that study?  Seems easier to carry UDPTL over RTP, =
which would allow the RTP to be secured using SRTP (and thus the UDPTL =
would be secured using SRTP).  There is a spec floating around to do =
exactly that (carry fax over RTP so that SRTP can secure it).  Advantage =
of using SRTP to secure fax is it separates the keying mechanism from =
security, so that Security Descriptions / MIKEY / DTLS-SRTP / =
whatever-is-invented-in-2020 will work just as effectively for voice as =
for fax.  And also that upgrading from a voice call to a "fax" call has =
no additional complexities due to security ("please press START to begin =
the fax transmission").

-d


> However, there is nothing =933GPP/IMS specific=94 about the mechanism, =
as UDPTL is commonly used for fax also elsewhere.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Mon Aug 19 12:28:38 2013
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-ietf-mmusic-msid-01.txt
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inline

On 8/19/13 10:57 AM, Harald Alvestrand wrote:
> On 08/15/2013 07:15 PM, Paul Kyzivat wrote:
>> I went through this version, with an eye to alignment with the Unified
>> Plan. I have a number of comments, some nits, some not:
>
> Thanks for doing this quickly!
>
> I've incorporated these changes (with some rewording, as always). I'll
> give this a few more days to stew, and then emit a new version.
>
>>
>> General:
>>
>> The Unified Plan says a single m-line describes one media source. It
>> allows this source to consist of multiple RTP Streams (identified by
>> SSRC). And it allows associating RTP Streams with m-lines multiple
>> ways, not just by SSRC.
>>
>> This msid draft hasn't fully incorporated those concepts. It still
>> seems to assume that a media source corresponds to a single SSRC. It
>> would be helpful to have a section discussion the process of
>> associating SSRCs to MedisStreamTracks. (And I think this needs to
>> include the possibility that an SSRC might be moved from one
>> MediaStreamTrack to another.)
>
> I tried replacing every occurence of SSRC with m-line, except where that
> didn't make sense (and in Appendix B, of course). Please point at the
> text where I missed, if it's not covered below!

I didn't have anything specific in mind other than what is below.
But I just got the feeling when reading it that it was still oriented 
towards the old model.

>> Section 2:
>>
>>    The identifier is a string of ASCII characters chosen from 0-9, a-z,
>>    A-Z and - (hyphen), consisting of between 1 and 64 characters. It
>>    MUST be unique among the identifier values used in the same SDP
>>    session.  It is RECOMMENDED that is generated using a random-number
>>    generator.
>> ...
>>    There may be multiple msid attributes on a single m-line. There may
>>    also be multiple m-lines that have the same value for identifier and
>>    application data.
>>
>> What does uniqueness mean in this context, since it can be used
>> multiple times, with the same or different m-lines? I *think* you mean
>> that a given identifier value is associated with a single MediaStream
>> for the duration of the SDP session.
>
> Yes, but I can't write that in section 2, since MediaStreams aren't
> introduced until section 4. The meaning of "uniqueness" has to be
> defined by the msid-semantic, I think - should I add a sentence saying
> that this is so?

In part it is just nit picking by me.
It might be that you can just not say this at all, at least here.
In the context in which the statement is made it is hard to see what 
sort of uniqueness is being required.

> (It's tempting to declare that msid is useless for anything except
> WebRTC. But the Paris (?) IETF meeting told me to write a generic
> mechanism with a WebRTC specialization, so that's how it's written.)

IIUC, MediaStreamTrack ought to be synonymous with a CLUE 
capture-encoding. But I don't think we have anything in CLUE that 
corresponds to a MediaStream. I don't understand MediaStream very well, 
but perhaps it is indeed unique to WebRTC.

The taxonomy draft is perhaps the right reference here.

>> Also, msid attributes aren't "on" an m-line. I think it would be
>> clearer to say "Multiple msid attributes may be associated with a
>> single m-line." But with the Unified Plan this isn't true.
>
> Sorry?

I'm picking at the meaning of words. To me "on" an m-line means that it 
is present within the characters that comprise the m-line itself. No 
"attribute" works that way.

I presume you are using "on" as synonymous with "attached to", or 
"associated with". So I was simply suggesting being more explicit.

> I think it's legal to say
>
> m=video
> a=msid: 1234 567
> a=msid: ABCD EFG
>
> This would mean (for the WMS context) that the same media content is
> present in both MediaStream 1234 and MediaStream ABCD, with
> MediaStreamTrack ids 567 and EFG, respectively.

I wasn't talking about any of that.

Its a WebRTC issue whether the same logical stream of packets can be 
part of multiple MediaStreams or MediaStreamTracks. So I'll take your 
word for it.

>> Also, I find the use of "identifier" and "token" very confusing. It is
>> hard to keep track of the semantics intended for those things. It
>> would be easier to follow if some more specific terms were used. For
>> example, "mediastream-id" instead of "identifier" in the msid and
>> msid-semantics attributes, and "semantics" instead of "token" in
>> msid-semantics attribute. ("semantics" is what is used in 3388 and 5576.)
>
> You're talking about the ABNF in sections 2 and 3 now, right?

Yes.

> We could use more meaningful names for the tokens, but it's just another
> level of indirection - generally, I don't find that specs become much
> more readable by repeated productions of the form
>
>     name1 = name2
>
> unless we have to refer to the tokens by name from surrounding text.

It words two ways:
- the text referring back to the elements from the ABNF
- when I look at the ABNF, understanding what concept in the text
   this element refers to.

When reading, I encountered both of those problems.
Eventually I can figure it out by going through an exercise of "what 
could this possibly mean". But it would be nice if it was clearer

	Thanks,
	Paul

> But your mileage may vary.

I realize this is to some extent a matter of taste.

Personally, I find that ABNF non-terminals are cheap, so I am happy to 
use more of them to increase clarity. In particular I'm inclined to 
ensure that there is a well-named non-terminal for every element that 
needs to be referenced from the text.

	Thanks,
	Paul

>> Section 4:
>>
>>    o  When a description is updated to have more SSRCs with the same
>>       msid value, but different appdata values, the recipient can signal
>>       to its application that new MediaStreamTracks have been added to
>>       the media stream.
>>
>> Is SSRC relevant any longer? s/SSRCs/m-lines/ ???
>
> Yep, I missed that (multiple misses in that section). Will update.
>
>>
>> Also in Section 4:
>>
>>    In addition to signaling that the track is closed when it disappears
>>    from the SDP, the track will also be signaled as being closed when
>>    all associated SSRCs have disappeared by the rules of [RFC3550]
>>    section 6.3.4 (BYE packet received) and 6.3.5 (timeout).
>>
>> Here we can't avoid talking about SSRCs. But maybe more is needed
>> about what SSRCs are associated with a particular m-line & msid value?
>
> Since this is in section 4 (WebRTC), I can happily point that problem
> off to unified-plan or its successors.
>>
>> Section 4.1:
>>
>>    Pre-WebRTC entities will not send msid.  This means that there will
>>    be some incoming RTP packets with SSRCs where the recipient does not
>>    know about a corresponding MediaStream id.
>>
>> In the context of the Unified Plan I think this could be stated better:
>>
>>    There may be some incoming RTP packets that the recipient cannot
>>    associate with a MediaStreamTrack.
>
> Rephrased, not quite like this.
>
>>
>> Also:
>>
>>    o  An msid-semantic:WMS attribute is present.  In this case, the
>>       session is WebRTC compatible, and the newly arrived SSRCs are
>>       either caused by a bug or by timing skew between the arrival of
>>       ...
>>
>> I suggest:
>>
>>    o  An msid-semantic:WMS attribute is present.  In this case, the
>>       session is WebRTC compatible, and the unassociated packets are
>>       either caused by a bug or by timing skew between the arrival of
>
> Done.
>
>> ...
>>
>>     Thanks,
>>     Paul
>>
>>
>> On 8/13/13 2:24 PM, Harald Alvestrand wrote:
>>> The auto-poster doesn't seem to want to tell mmusic about this....
>>>
>>> this is updated according to Unified-plan. Since it was only 1 day
>>> before the old version had expired, the proofreading might be a little
>>> light - please point out places where I had missed changing "SSRC" to
>>> "m-line"!
>>>
>>> I retained some description of the old mechanism (using a=ssrc) in an
>>> appendix. Please state opinions on whether this should be retained as
>>> informative, promoted to normative, or removed entirely in the next
>>> version.
>>>
>>> Hope this informs!
>>>
>>>         Harald
>>>
>>>
>>> -------- Original Message --------
>>> Subject:     New Version Notification for draft-ietf-mmusic-msid-01.txt
>>> Date:     Tue, 13 Aug 2013 05:13:42 -0700
>>> From:     internet-drafts@ietf.org
>>> To:     Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand
>>> <harald@alvestrand.no>
>>>
>>>
>>>
>>> A new version of I-D, draft-ietf-mmusic-msid-01.txt
>>> has been successfully submitted by Harald Alvestrand and posted to the
>>> IETF repository.
>>>
>>> Filename:     draft-ietf-mmusic-msid
>>> Revision:     01
>>> Title:         Cross Session Stream Identification in the Session
>>> Description Protocol
>>> Creation date:     2013-08-13
>>> Group:         mmusic
>>> Number of pages: 14
>>> URL:http://www.ietf.org/internet-drafts/draft-ietf-mmusic-msid-01.txt
>>> Status:http://datatracker.ietf.org/doc/draft-ietf-mmusic-msid
>>> Htmlized:http://tools.ietf.org/html/draft-ietf-mmusic-msid-01
>>> Diff:http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-msid-01
>>>
>>> Abstract:
>>>     This document specifies a grouping mechanism for RTP media streams
>>>     that can be used to specify relations between media streams.
>>>
>>>     This mechanism is used to signal the association between the SDP
>>>     concept of "m-line" and the WebRTC concept of "MediaStream" /
>>>     "MediaStreamTrack" using SDP signaling.
>>>
>>>     This document is a work item of the MMUSIC WG, whose discussion list
>>>     ismmusic@ietf.org.
>>>
>>>
>>>
>>>
>>>
>>> Please note that it may take a couple of minutes from the time of
>>> submission
>>> until the htmlized version and diff are available at tools.ietf.org.
>>>
>>> The IETF Secretariat
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From christer.holmberg@ericsson.com  Tue Aug 20 02:42:33 2013
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To: Dan Wing <dwing@cisco.com>
Thread-Topic: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Hi Dan,

I'll get back to your comments later, but just a question for clarification=
:

When you say "UDPTL over RTP", you are NOT referring to RFC 4612, right?=20

Or, did you mean to say "fax over RTP"? :)

Regards,

Christer

-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: 19. elokuuta 2013 20:43
To: Christer Holmberg
Cc: mmusic; mmusic-chairs@tools.ietf.org
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00


On Aug 19, 2013, at 6:03 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
> =20
> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, which=
 defines usage of UDPTL over DTLS, in order to provide secure fax.
> =20
> The draft was previously submitted to DISPATCH. Based on discussions with=
 the ADs and chairs, it was decided that it shall be submitted to MMUSIC (n=
ote that no DTLS extensions are needed).
> =20
> As is indicated in the draft, 3GPP has performed a study on how to=20
> provide secure fax in the IMS, and the outcome was that secure fax shall =
be transported using UDPTL over DTLS.

Got a pointer to that study?  Seems easier to carry UDPTL over RTP, which w=
ould allow the RTP to be secured using SRTP (and thus the UDPTL would be se=
cured using SRTP).  There is a spec floating around to do exactly that (car=
ry fax over RTP so that SRTP can secure it).  Advantage of using SRTP to se=
cure fax is it separates the keying mechanism from security, so that Securi=
ty Descriptions / MIKEY / DTLS-SRTP / whatever-is-invented-in-2020 will wor=
k just as effectively for voice as for fax.  And also that upgrading from a=
 voice call to a "fax" call has no additional complexities due to security =
("please press START to begin the fax transmission").

-d


> However, there is nothing "3GPP/IMS specific" about the mechanism, as UDP=
TL is commonly used for fax also elsewhere.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: Dan Wing <dwing@cisco.com>
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Date: Tue, 20 Aug 2013 08:33:39 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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On Aug 20, 2013, at 2:42 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi Dan,
>=20
> I'll get back to your comments later, but just a question for =
clarification:
>=20
> When you say "UDPTL over RTP", you are NOT referring to RFC 4612, =
right?=20

I believe I am.  However, I thought there was an ITU specification for =
T.38 over RTP -- perhaps there is just that IETF specification.  Are you =
not a fan of RFC4612?

>=20
> Or, did you mean to say "fax over RTP"? :)

Maybe you could educate me on the difference.  My fax knowledge has =
faded with time.

-d

>=20
> Regards,
>=20
> Christer
>=20
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]=20
> Sent: 19. elokuuta 2013 20:43
> To: Christer Holmberg
> Cc: mmusic; mmusic-chairs@tools.ietf.org
> Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
>=20
>=20
> On Aug 19, 2013, at 6:03 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>> Hi,
>>=20
>> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, =
which defines usage of UDPTL over DTLS, in order to provide secure fax.
>>=20
>> The draft was previously submitted to DISPATCH. Based on discussions =
with the ADs and chairs, it was decided that it shall be submitted to =
MMUSIC (note that no DTLS extensions are needed).
>>=20
>> As is indicated in the draft, 3GPP has performed a study on how to=20
>> provide secure fax in the IMS, and the outcome was that secure fax =
shall be transported using UDPTL over DTLS.
>=20
> Got a pointer to that study?  Seems easier to carry UDPTL over RTP, =
which would allow the RTP to be secured using SRTP (and thus the UDPTL =
would be secured using SRTP).  There is a spec floating around to do =
exactly that (carry fax over RTP so that SRTP can secure it).  Advantage =
of using SRTP to secure fax is it separates the keying mechanism from =
security, so that Security Descriptions / MIKEY / DTLS-SRTP / =
whatever-is-invented-in-2020 will work just as effectively for voice as =
for fax.  And also that upgrading from a voice call to a "fax" call has =
no additional complexities due to security ("please press START to begin =
the fax transmission").
>=20
> -d
>=20
>=20
>> However, there is nothing "3GPP/IMS specific" about the mechanism, as =
UDPTL is commonly used for fax also elsewhere.
>>=20
>> Regards,
>>=20
>> Christer
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>=20


From christer.holmberg@ericsson.com  Tue Aug 20 09:32:33 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dan Wing <dwing@cisco.com>
Thread-Topic: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Hi Dan,

>> I'll get back to your comments later, but just a question for clarificat=
ion:
>>
>> When you say "UDPTL over RTP", you are NOT referring to RFC 4612, right?
>
> I believe I am.  However, I thought there was an ITU specification for T.=
38 over RTP -- perhaps there is just that IETF specification. =20

The latest version of T.38 also specifies the usage of T.38 over RTP. =20

RFC 4612 defines the new MIME type and fmtp attribute values, also used in =
T.38, but refers to T.38 for the fax mechanism itself.

> Are you not a fan of RFC4612?

I'm a fan of Kelly Clarkson ;)

>> Or, did you mean to say "fax over RTP"? :)
>
> Maybe you could educate me on the difference.  My fax knowledge has faded=
 with time.

When you use RTP, you don't use UDPTL. The UDPTL layer is "replaced" by the=
 RTP layer, and the fax stuff is transported as an RTP payload type.

Regards,

Christer





> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: 19. elokuuta 2013 20:43
> To: Christer Holmberg
> Cc: mmusic; mmusic-chairs@tools.ietf.org
> Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
>
>
> On Aug 19, 2013, at 6:03 AM, Christer Holmberg <christer.holmberg@ericsso=
n.com> wrote:
>
>> Hi,
>>
>> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, whic=
h defines usage of UDPTL over DTLS, in order to provide secure fax.
>>
>> The draft was previously submitted to DISPATCH. Based on discussions wit=
h the ADs and chairs, it was decided that it shall be submitted to MMUSIC (=
note that no DTLS extensions are needed).
>>
>> As is indicated in the draft, 3GPP has performed a study on how to
>> provide secure fax in the IMS, and the outcome was that secure fax shall=
 be transported using UDPTL over DTLS.
>
> Got a pointer to that study?  Seems easier to carry UDPTL over RTP, which=
 would allow the RTP to be secured using SRTP (and thus the UDPTL would be =
secured using SRTP).  There is a spec floating around to do exactly that (c=
arry fax over RTP so that SRTP can secure it).  Advantage of using SRTP to =
secure fax is it separates the keying mechanism from security, so that Secu=
rity Descriptions / MIKEY / DTLS-SRTP / whatever-is-invented-in-2020 will w=
ork just as effectively for voice as for fax.  And also that upgrading from=
 a voice call to a "fax" call has no additional complexities due to securit=
y ("please press START to begin the fax transmission").
>
> -d
>
>
>> However, there is nothing "3GPP/IMS specific" about the mechanism, as UD=
PTL is commonly used for fax also elsewhere.
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>=

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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On 8/16/13 3:35 AM, Christer Holmberg wrote:

> MAYBE we could restrict usage of POF/PANs, and say that they MUST NOT be used to change session level information...

I was assuming it was obvious that this must not be the case.
The POFs and PANs (AFAIK) only contain the media sections they intend to 
change, so they don't contain the session level stuff.
If you change session level stuff, then you should need a real offer.

Or is somebody suggesting to analyze the changes, and discover that some 
session level things don't apply to all media sections (e.g., because 
the media section overrides them, or they just aren't pertinent for that 
kind of media) and so you can send a POF that includes just the session 
level stuff and the media sections that are affected. ???

I don't think much of that.

	Thanks,
	Paul

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] UP: Can a partial offer be a full offer?
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Date: Wed, 21 Aug 2013 06:19:32 +0000
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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Hi,

>> MAYBE we could restrict usage of POF/PANs, and say that they MUST NOT be=
 used to change session level information...
>
> I was assuming it was obvious that this must not be the case.
> The POFs and PANs (AFAIK) only contain the media sections they intend to =
change, so they don't contain the session level stuff.
> If you change session level stuff, then you should need a real offer.

Keep in mind that the o- line, which IS session level, IS modified each tim=
e a POF/PAN is sent.

Maybe we should use media level o- lines for POFs/PANs....

Regards,

Christer


From pkyzivat@alum.mit.edu  Wed Aug 21 06:34:22 2013
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On 8/21/13 2:19 AM, Christer Holmberg wrote:
> Hi,
>
>>> MAYBE we could restrict usage of POF/PANs, and say that they MUST NOT be used to change session level information...
>>
>> I was assuming it was obvious that this must not be the case.
>> The POFs and PANs (AFAIK) only contain the media sections they intend to change, so they don't contain the session level stuff.
>> If you change session level stuff, then you should need a real offer.
>
> Keep in mind that the o- line, which IS session level, IS modified each time a POF/PAN is sent.

Good point.

> Maybe we should use media level o- lines for POFs/PANs....

I don't know how you could do that in a backward compatible way.

	Thanks,
	Paul


From hadriel.kaplan@oracle.com  Wed Aug 21 07:33:25 2013
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Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On Aug 21, 2013, at 9:34 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

>> Maybe we should use media level o- lines for POFs/PANs....
>=20
> I don't know how you could do that in a backward compatible way.

Backward compatible to what?  There's no POFs/PANs thing today.  You =
don't even need to use SDP syntax if you don't want to, or you could use =
some diff-style format of SDP lines, or use what look and act exactly =
like SDP lines but aren't technically "SDP" to avoid any legacy RFC =
rules.

-hadriel


From pkyzivat@alum.mit.edu  Wed Aug 21 10:24:44 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] UP: Can a partial offer be a full offer?
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On 8/21/13 10:33 AM, Hadriel Kaplan wrote:
>
> On Aug 21, 2013, at 9:34 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>>> Maybe we should use media level o- lines for POFs/PANs....
>>
>> I don't know how you could do that in a backward compatible way.
>
> Backward compatible to what?  There's no POFs/PANs thing today.  You don't even need to use SDP syntax if you don't want to, or you could use some diff-style format of SDP lines, or use what look and act exactly like SDP lines but aren't technically "SDP" to avoid any legacy RFC rules.

Maybe we are thinking of this differently, but my thinking is that 
consistency with a full SDP must be maintained. If you start off with an 
O/A, then some POFs/PANs, eventually you may need to do another O/A. And 
at that point there needs to be consistency.

I think that means that POFs/PANs always contain a subset of a full SDP, 
and act as deltas, implicitly updating a full SDP. That means everything 
in the POFs/PANs needs to be valid SDP. And when that stuff is used in a 
full O/A, there need to be rules for how those are reconciled.

So, to introduce a media level o-line, it would need to be defined for 
SDP too.

It might be possible to arrange so that it won't be seen by an endpoint 
that doesn't support POFs/PANs, but that will take some care, and is 
likely to be fragile.

	Thanks,
	Paul


From aallen@blackberry.com  Wed Aug 21 10:31:31 2013
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From: Andrew Allen <aallen@blackberry.com>
To: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Propo
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Propo
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In my view since including all the correlation attributes in the SDP is a SH=
OULD strength implementing all the correlation attributes should  also be of=
 SHOULD strength or at most MUST implement except when it is known that a pa=
rticular correlation mechanism will work for the deployment scenario that  c=
ircuit-switched bearer is used. It seems to me kind of silly to have a MUST=
 implement (but might not ever try and use requirement).



To my knowledge 3GPP are currently the only party interested in using this d=
raft. To publish this draft with mandatory aspects which are not needed and=
 not acceptable to 3GPP application will in my view be a mistake and result=
 in an RFC that will either not be implemented or will not be implemented as=
 written because the overhead of implementing the DTMF and UUI correlation m=
echanisms is not justified and since these are not needed in the 3GPP use ca=
se and will not work with the 3GPP architecture either.



In the 3GPP application of this draft the negotiation of the circuit-switche=
d bearer does not take place end to end between the terminals but instead be=
tween the terminal and an IMS network infrastructure component (the SCC AS)=
 that acts as B2BUA and the CS call is then interworked to SIP and patched i=
n with the session with the remote party. In addition the E.164 number provi=
ded by the IMS network infrastructure component is unique for that circuit-s=
witched bearer setup so no additional correlation mechanism is really needed=
. The Mobile terminal always acts as the active party in establishing the ci=
rcuit switched bearer in the 3GPP scenario.



Mandating the callerid mechanism for correlation is acceptable to me since i=
t basically has no impact on the mobile terminal or the IMS network infrastr=
ucture component except the inclusion of the cs-correlation attribute and th=
e inclusion of that by the interworking MSC in the P-Asserted-Identity (whic=
h is already done). It may have the advantage that now the IMS network infra=
structure component does not have to allocate a unique E.164 number for each=
 session but could just use the callerid for the correlation. I think this a=
lso does mean that the SCC AS can then be sure that the CS session setup is=
 coming from the terminal  that was negotiated it in the O/A exchange and no=
t some attacker trying to hijack sessions using as the dialed number an E.16=
4 number that was previously assigned to him. Also since in the 3GPP scenari=
o the circuit-switched network is local to and under the control of the oper=
ator the callerid mechanism can be guaranteed to work.



The DTMF and UUI mechanisms are quite intensive to implement and both have i=
ssues for the 3GPP application. DTMF will result in a delay in the setup end=
 to end of the media and also would require media to be routed to the IMS ne=
twork infrastructure component (which is only a signaling B2BUA) and force i=
t to have to support  DTMF tone detection. UUI will almost certainly not wor=
k as most operators switchers do not pass this information and also would re=
quire support for the CUSS UUI mechanisms to deliver the UUI in SIP from the=
 interworking switch to the SCC AS.


I am not against the end to end use of CS SDP but I think it does have some=
 issues such as the cost of long distance or even international circuit swit=
ched bearers when the only leg that is a problem is the access for one mobil=
e user. Also a device that needs to use circuit switched bearers cannot do s=
o with those device that don't also support CS SDP or that also have access=
 to a circuit-switched connection. So I don't think the correlation issues w=
ith end to end use are the primary obstacles to successful interoperability.=
 One way to address the interoperability issues is an architecture like 3GPP=
's where the circuit-switched bearer is negotiated between the device and an=
 interworking function at the edge of the network which is where the quality=
 problem exists in the first place not to set up a CS call end to end.

In my view it is not the end of the world if in some scenarios the user has=
 to manually accept the circuit switched bearer and patch it into the multim=
edia session. In fact there may be some really valid use cases where the dev=
ice that receives the CS setup is not the same device that negotiated the CS=
 session using SDP. Think of a webex type conference where you connect into=
 a conference using SIP and then get the conference bridge to call you back=
 with a PSTN call to your POTS phone - Manual acceptance seems fine to me in=
 this case and automatic correlation will not work. Similarly an internet  g=
aming session established using SIP might call the user back at his POTS lin=
e for audio conversations associated with the game.


One possibility is to state in the draft that if you cannot confirm that the=
 CS call is part of the session negotiated using SDP using a correlation mec=
hanism or some other means you MUST NOT accept the CS bearer as part of the=
 session without manual user acceptance.

I am Ok for the callerid correlation mechanism to be made mandatory to imple=
ment. I think this is the mechanism with the least burden for implementation=
s and also the one most likely to work. I do not agree that the DTMF and the=
 UUI should be mandatory to implement.

I also think it should be realized that implementations may only support one=
 direction for the CS call establishment and that the endpoints only need to=
 support the aspects of the correlation mechanism for the direction(s) they=
 support. Also some users might also not wish to be a passive party (such as=
 when calling party pays for example when cellular roaming) even if they hav=
e the capability to do so. Thus some implementations may only support the as=
pects of the correlation mechanism for the direction (active or passive) tha=
t they support.

Andrew

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Flemming Andreasen
Sent: Friday, August 09, 2013 6:10 PM
To: mmusic
Subject: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol=
 Action: 'Session Description Protocol (SDP) Extension For Setting Audio and=
 Video Media Streams Over Circuit-Switched Bearers In The Public Switched Te=
lephone Network (PSTN)' to Propos...

Greetings MMUSIC

As you can see per the below, the IESG approved the Circuit-Switched Bearers=
 draft for publication, however before doing so, they requested a couple of=
 changes. The resulting approved document can be found at http://tools.ietf.=
org/html/draft-ietf-mmusic-sdp-cs-21

One of the changes introduced resulted in all 3 correlation mechanisms now b=
eing mandatory-to-implement (previously they were not):
<quote>


   Implementations MUST support the three correlation mechanisms

   specified in Section 5.2.3.2<http://tools.ietf.org/html/draft-ietf-mmusic=
-sdp-cs-21#section-5.2.3.2>, Section 5.2.3.3<http://tools.ietf.org/html/draf=
t-ietf-mmusic-sdp-cs-21#section-5.2.3.3> and Section 5.2.3.4<http://tools.ie=
tf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.4>,

   and SHOULD include all supported mechanisms the initial Offer.
</quote>
Since this is a non-trivial change that has not been discussed by the WG (an=
d we know of at least one person having expressed concerns with it off-line)=
, we are hereby solicting the WG participants for input on changes introduce=
d as a result of the IESG review with a focus on the above. In doing so, ple=
ase keep in mind that we are developing standards that work in the general I=
nternet.


The following URL provides a diff-file showing the changes resulting from th=
e IESG review:
    http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&url=
2=3Ddraft-ietf-mmusic-sdp-cs-21.txt

Comments are due by August 23 (two weeks from today)

Thanks

-- Flemming (MMUSIC co-chair)


-------- Original Message --------
Subject:

[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For=
 Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The=
 Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf-=
mmusic-sdp-cs-21.txt)

Date:

Mon, 1 Jul 2013 11:56:23 -0700

From:

The IESG <iesg-secretary@ietf.org><mailto:iesg-secretary@ietf.org>

To:

IETF-Announce <ietf-announce@ietf.org><mailto:ietf-announce@ietf.org>

CC:

mmusic chair <mmusic-chairs@tools.ietf.org><mailto:mmusic-chairs@tools.ietf.=
org>, mmusic mailing list <mmusic@ietf.org><mailto:mmusic@ietf.org>, RFC Edi=
tor <rfc-editor@rfc-editor.org><mailto:rfc-editor@rfc-editor.org>



The IESG has approved the following document:

- 'Session Description Protocol (SDP) Extension For Setting Audio and

   Video Media Streams Over Circuit-Switched Bearers In The Public

   Switched Telephone Network (PSTN)'

  (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard



This document is the product of the Multiparty Multimedia Session Control

Working Group.



The IESG contact persons are Gonzalo Camarillo and Richard Barnes.



A URL of this Internet Draft is:

http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/









*Technical Summary

*



The document describes use cases, requirements, and protocol extensions

for using the Session Description Protocol (SDP) Offer/Answer model for

establishing audio and video media streams over circuit-switched bearers

in the Publich Switched Telephone Network (PSTN)



*Working Group Summary

*



The WG had some discussion around the format to use for E.164 numbers

and whether to align this with the existing definition in RFC 3108. The

RFC 3108 definition was seen as deficient and the WG agreed it was

better to align with relevant parts of the tel URI format defined in RFC

3966, not least since SDP address types are defined in the context of a

particular network type, and hence RFC 3108 compatibility is not a

concern (the implication is that the "E164" address type may differ

between network types in SDP).



*Document Quality

*



There are currently no known implementations of the draft, however the

draft is a dependency for 3GPP, so future implementations are expected.



The document has received good overall review in the WG, some of which

resulted in changes to the detailed specification. The document has been

reviewed in detail several times, including of the last few versions.

The major contributors to these as well as earlier discussions are

listed in the Acknowledgements section of the document.





*Personnel

*



Document Shepherd: Flemming Andreasen

Responsible AD: Gonzalo Camarillo

_______________________________________________

mmusic mailing list

mmusic@ietf.org<mailto:mmusic@ietf.org>

https://www.ietf.org/mailman/listinfo/mmusic

.





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<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">In my view since=
 including all the correlation attributes in the SDP is a SHOULD strength im=
plementing all the correlation attributes should&nbsp; also be of SHOULD str=
ength or at most MUST implement except
 when it is known that a particular correlation mechanism will work for the=
 deployment scenario that&nbsp; circuit-switched bearer is used. It seems to=
 me kind of silly to have a MUST implement (but might not ever try and use r=
equirement).
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">To my knowledge 3=
GPP are currently the only party interested in using this draft. To publish=
 this draft with mandatory aspects which are not needed and not acceptable t=
o 3GPP application will in my view
 be a mistake and result in an RFC that will either not be implemented or wi=
ll not be implemented as written because the overhead of implementing the DT=
MF and UUI correlation mechanisms is not justified and since these are not n=
eeded in the 3GPP use case and
 will not work with the 3GPP architecture either.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">In the 3GPP appli=
cation of this draft the negotiation of the circuit-switched bearer does not=
 take place end to end between the terminals but instead between the termina=
l and an IMS network infrastructure
 component (the SCC AS) that acts as B2BUA and the CS call is then interwork=
ed to SIP and patched in with the session with the remote party. In addition=
 the E.164 number provided by the IMS network infrastructure component is un=
ique for that circuit-switched
 bearer setup so no additional correlation mechanism is really needed. The M=
obile terminal always acts as the active party in establishing the circuit s=
witched bearer in the 3GPP scenario.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">Mandating the cal=
lerid mechanism for correlation is acceptable to me since it basically has n=
o impact on the mobile terminal or the IMS network infrastructure component=
 except the inclusion of the cs-correlation
 attribute and the inclusion of that by the interworking MSC in the P-Assert=
ed-Identity (which is already done). It may have the advantage that now the=
 IMS network infrastructure component does not have to allocate a unique E.1=
64 number for each session but
 could just use the callerid for the correlation. I think this also does mea=
n that the SCC AS can then be sure that the CS session setup is coming from=
 the terminal &nbsp;that was negotiated it in the O/A exchange and not some=
 attacker trying to hijack sessions
 using as the dialed number an E.164 number that was previously assigned to=
 him. Also since in the 3GPP scenario the circuit-switched network is local=
 to and under the control of the operator the callerid mechanism can be guar=
anteed to work.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">The DTMF and UUI=
 mechanisms are quite intensive to implement and both have issues for the 3G=
PP application. DTMF will result in a delay in the setup end to end of the m=
edia and also would require media
 to be routed to the IMS network infrastructure component (which is only a s=
ignaling B2BUA) and force it to have to support&nbsp; DTMF tone detection. U=
UI will almost certainly not work as most operators switchers do not pass th=
is information and also would require
 support for the CUSS UUI mechanisms to deliver the UUI in SIP from the inte=
rworking switch to the SCC AS.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext">I am not against the end to end use of CS=
 SDP but I think it does have some issues such as the cost of long distance=
 or even international circuit switched bearers when the
 only leg that is a problem is the access for one mobile user. Also a device=
 that needs to use circuit switched bearers cannot do so with those device t=
hat don&#8217;t also support CS SDP or that also have access to a circuit-sw=
itched connection. So I don&#8217;t think
 the correlation issues with end to end use are the primary obstacles to suc=
cessful interoperability. One way to address the interoperability issues is=
 an architecture like 3GPP's where the circuit-switched bearer is negotiated=
 between the device and an interworking
 function at the edge of the network which is where the quality problem exis=
ts in the first place not to set up a CS call end to end.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext">In my view it is not the end of the world=
 if in some scenarios the user has to manually accept the circuit switched b=
earer and patch it into the multimedia session. In fact
 there may be some really valid use cases where the device that receives the=
 CS setup is not the same device that negotiated the CS session using SDP. T=
hink of a webex type conference where you connect into a conference using SI=
P and then get the conference
 bridge to call you back with a PSTN call to your POTS phone &#8211; Manual=
 acceptance seems fine to me in this case and automatic correlation will not=
 work. Similarly an internet&nbsp; gaming session established using SIP migh=
t call the user back at his POTS line for
 audio conversations associated with the game. <o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext">One possibility is to state in the draft=
 that if you cannot confirm that the CS call is part of the session negotiat=
ed using SDP using a correlation mechanism or some other
 means you MUST NOT accept the CS bearer as part of the session without manu=
al user acceptance.<br>
<br>
</span><span style=3D"font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
;color:windowtext">I am Ok for the callerid correlation mechanism to be made=
 mandatory to implement. I think this is the mechanism with the least burden=
 for implementations and also the one most likely to
 work. I do not agree that the DTMF and the UUI should be mandatory to imple=
ment.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:windowtext">I also think it should be realized that i=
mplementations may only support one direction for the CS call establishment=
 and that the endpoints only need to support the aspects
 of the correlation mechanism for the direction(s) they support. Also some u=
sers might also not wish to be a passive party (such as when calling party p=
ays for example when cellular roaming) even if they have the capability to d=
o so. Thus some implementations
 may only support the aspects of the correlation mechanism for the direction=
 (active or passive) that they support.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:windowtext">Andrew<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span>=
</p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0=
in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot;=
Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span=
 style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&q=
uot;;color:windowtext"> mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.=
org]
<b>On Behalf Of </b>Flemming Andreasen<br>
<b>Sent:</b> Friday, August 09, 2013 6:10 PM<br>
<b>To:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: P=
rotocol Action: 'Session Description Protocol (SDP) Extension For Setting Au=
dio and Video Media Streams Over Circuit-Switched Bearers In The Public Swit=
ched Telephone Network (PSTN)'
 to Propos...<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Greetings MMUSIC<br>
<br>
As you can see per the below, the IESG approved the Circuit-Switched Bearers=
 draft for publication, however before doing so, they requested a couple of=
 changes. The resulting approved document can be found at
<a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21">http://to=
ols.ietf.org/html/draft-ietf-mmusic-sdp-cs-21</a><br>
<br>
One of the changes introduced resulted in all 3 correlation mechanisms now b=
eing mandatory-to-implement (previously they were not):<br>
&lt;quote&gt;<br>
<br>
<o:p></o:p></p>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&nb=
sp;&nbsp; Implementations MUST support the three correlation mechanisms<o:p>=
</o:p></span></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&nb=
sp;&nbsp; specified in <a href=3D"http://tools.ietf.org/html/draft-ietf-mmus=
ic-sdp-cs-21#section-5.2.3.2">Section 5.2.3.2</a>, <a href=3D"http://tools.i=
etf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.3">Section 5.2.3.3</a=
> and <a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#sect=
ion-5.2.3.4">Section 5.2.3.4</a>,<o:p></o:p></span></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&nb=
sp;&nbsp; and SHOULD include all supported mechanisms the initial Offer.<o:p=
></o:p></span></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&lt;/quote&gt;<br>
Since this is a non-trivial change that has not been discussed by the WG (an=
d we know of at least one person having expressed concerns with it off-line)=
, we are hereby solicting the WG participants for input on changes introduce=
d as a result of the IESG review
 with a focus on the above. In doing so, please keep in mind that we are dev=
eloping standards that work in the general Internet.
<br>
<br>
<br>
The following URL provides a diff-file showing the changes resulting from th=
e IESG review:<br>
&nbsp;&nbsp;&nbsp; <a href=3D"http://tools.ietf.org/rfcdiff?url1=3Ddraft-iet=
f-mmusic-sdp-cs-18.txt&amp;url2=3Ddraft-ietf-mmusic-sdp-cs-21.txt" target=3D=
"_blank">
http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&amp;url=
2=3Ddraft-ietf-mmusic-sdp-cs-21.txt</a><br>
<br>
Comments are due by August 23 (two weeks from today)<br>
<br>
Thanks <br>
<br>
-- Flemming (MMUSIC co-chair)<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><br>
<br>
-------- Original Message -------- <o:p></o:p></p>
<table class=3D"MsoNormalTable" border=3D"0" cellspacing=3D"0" cellpadding=
=3D"0">
<tbody>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Subject=
: <o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">[MMUSIC] Protocol Action: 'Session Description Protoc=
ol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Sw=
itched Bearers In The Public Switched Telephone Network (PSTN)' to Proposed=
 Standard (draft-ietf-mmusic-sdp-cs-21.txt)<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Date: <=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">Mon, 1 Jul 2013 11:56:23 -0700<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>From: <=
o:p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">The IESG <a href=3D"mailto:iesg-secretary@ietf.org">&=
lt;iesg-secretary@ietf.org&gt;</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>To: <o:=
p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">IETF-Announce <a href=3D"mailto:ietf-announce@ietf.or=
g">&lt;ietf-announce@ietf.org&gt;</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>CC: <o:=
p></o:p></b></p>
</td>
<td style=3D"padding:0in 0in 0in 0in">
<p class=3D"MsoNormal">mmusic chair <a href=3D"mailto:mmusic-chairs@tools.ie=
tf.org">&lt;mmusic-chairs@tools.ietf.org&gt;</a>, mmusic mailing list
<a href=3D"mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a>, RFC Editor <=
a href=3D"mailto:rfc-editor@rfc-editor.org">
&lt;rfc-editor@rfc-editor.org&gt;</a><o:p></o:p></p>
</td>
</tr>
</tbody>
</table>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<pre>The IESG has approved the following document:<o:p></o:p></pre>
<pre>- 'Session Description Protocol (SDP) Extension For Setting Audio and<o=
:p></o:p></pre>
<pre>&nbsp;&nbsp; Video Media Streams Over Circuit-Switched Bearers In The P=
ublic<o:p></o:p></pre>
<pre>&nbsp;&nbsp; Switched Telephone Network (PSTN)'<o:p></o:p></pre>
<pre>&nbsp; (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard<o:p></o:p=
></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>This document is the product of the Multiparty Multimedia Session Contr=
ol<o:p></o:p></pre>
<pre>Working Group.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The IESG contact persons are Gonzalo Camarillo and Richard Barnes.<o:p>=
</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>A URL of this Internet Draft is:<o:p></o:p></pre>
<pre><a href=3D"http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/">h=
ttp://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/</a><o:p></o:p></pre=
>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Technical Summary<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The document describes use cases, requirements, and protocol extensions=
<o:p></o:p></pre>
<pre>for using the Session Description Protocol (SDP) Offer/Answer model for=
<o:p></o:p></pre>
<pre>establishing audio and video media streams over circuit-switched bearer=
s<o:p></o:p></pre>
<pre>in the Publich Switched Telephone Network (PSTN)<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Working Group Summary<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The WG had some discussion around the format to use for E.164 numbers<o=
:p></o:p></pre>
<pre>and whether to align this with the existing definition in RFC 3108. The=
<o:p></o:p></pre>
<pre>RFC 3108 definition was seen as deficient and the WG agreed it was<o:p>=
</o:p></pre>
<pre>better to align with relevant parts of the tel URI format defined in RF=
C<o:p></o:p></pre>
<pre>3966, not least since SDP address types are defined in the context of a=
<o:p></o:p></pre>
<pre>particular network type, and hence RFC 3108 compatibility is not a<o:p>=
</o:p></pre>
<pre>concern (the implication is that the &quot;E164&quot; address type may=
 differ<o:p></o:p></pre>
<pre>between network types in SDP).<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Document Quality<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>There are currently no known implementations of the draft, however the<=
o:p></o:p></pre>
<pre>draft is a dependency for 3GPP, so future implementations are expected.=
<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The document has received good overall review in the WG, some of which<=
o:p></o:p></pre>
<pre>resulted in changes to the detailed specification. The document has bee=
n<o:p></o:p></pre>
<pre>reviewed in detail several times, including of the last few versions.<o=
:p></o:p></pre>
<pre>The major contributors to these as well as earlier discussions are<o:p>=
</o:p></pre>
<pre>listed in the Acknowledgements section of the document.<o:p></o:p></pre=
>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Personnel<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Document Shepherd: Flemming Andreasen<o:p></o:p></pre>
<pre>Responsible AD: Gonzalo Camarillo <o:p></o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>mmusic mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><o:p></o:p></pre>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ie=
tf.org/mailman/listinfo/mmusic</a><o:p></o:p></pre>
<pre>.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
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On Aug 21, 2013, at 1:24 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 8/21/13 10:33 AM, Hadriel Kaplan wrote:
>> Backward compatible to what?  There's no POFs/PANs thing today.  You =
don't even need to use SDP syntax if you don't want to, or you could use =
some diff-style format of SDP lines, or use what look and act exactly =
like SDP lines but aren't technically "SDP" to avoid any legacy RFC =
rules.
>=20
> Maybe we are thinking of this differently, but my thinking is that =
consistency with a full SDP must be maintained. If you start off with an =
O/A, then some POFs/PANs, eventually you may need to do another O/A. And =
at that point there needs to be consistency.

Sure, the state changes and the rules that do that must be consistent.  =
The encoding syntax and rules for mandatory lines in a POF/PAN need not =
be. (not that I'm seriously proposing a different encoding syntax be =
created, but just saying "we aren't constrained to SDP offer/answer MUST =
rules for this POF/PAN piece")


> I think that means that POFs/PANs always contain a subset of a full =
SDP, and act as deltas, implicitly updating a full SDP. That means =
everything in the POFs/PANs needs to be valid SDP.

So are you saying it needs to contain an "v=3D", "s=3D", and "t=3D" =
lines?  Because those are mandatory for "SDP", afaik. =20
[In fact afaict, technically "e=3D" and "p=3D" were mandatory =
originally, but RFC 3264 downgraded them to optional. (and then later =
RFC 4566 did the same for "SDP")]

This POF/PAN thing can pick+choose what pieces of SDP-ish stuff it needs =
to convey, and how.


> And when that stuff is used in a full O/A, there need to be rules for =
how those are reconciled.

Sure.

> So, to introduce a media level o-line, it would need to be defined for =
SDP too.

Or we could keep an o=3D line above the m=3D section of the POF/PAN, or =
define some other attribute altogether to perform the same function for =
the POFs/PANs and just define that it updates the sess-version value, or =
whatever.  My only point is we don't need to worry about =
backwards-compatibility.  We do need to worry about name collision with =
SDP, since we're using SDP syntax encoding.  But we could, for example, =
define attributes that are only usable for POFs/PANs and not full SDP =
O/A.

-hadriel


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Jonathan Lennox'" <jonathan@vidyo.com>, "'Bernard Aboba'" <bernard_aboba@hotmail.com>
References: <BLU169-W630D4FBAA70899F6C54A0593830@phx.gbl>, <3879D71E758A7E4AA99A35DD8D41D3D91D487E63@xmb-rcd-x14.cisco.com>, <EE556E46-54C1-4AAB-B03A-56FB8971D8A2@vidyo.com>	<BLU169-W1208949649FACAC3137B05C93630@phx.gbl> <3AD93742-DBA4-4EDD-80AD-A667994ECE8D@vidyo.com>
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Date: Thu, 22 Aug 2013 12:05:13 +0300
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RFC 6190 Single Session Transport
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Hi Jonathan,

Maybe if you want to use FEC only on the base layer and not on all layers.

Roni

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Jonathan Lennox
Sent: 19 July, 2013 5:54 PM
To: Bernard Aboba
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RFC 6190 Single Session Transport

 

 

On Jul 18, 2013, at 10:02 PM, Bernard Aboba <bernard_aboba@hotmail.com>
wrote:





Jonathan said: 

"I don't understand how single-session multi-source transmission will work
without either a) signaling "a=ssrc-group DDP" decoding dependencies, or b)
making various implementation-specific assumptions about the structure of
the SVC streams."
 
[BA] In terms of SDP choices for expressing layering within a single m line,
you can either go the a=ssrc-group:DDP route, or use distinct payload types
for each layer and RFC 5583 a=group:DDP (this is what "Unified Plan" Section
4.7 appears to advocate).  

I'm not sure how "MST within a single session" could be expressed in SDP
without BUNDLE.  Was the idea to use multiple m lines and group them
together but use the same port? 

 

I think the authors of RFC 6190 didn't see a use case for MST within a
single RTP session -- all of the interesting MST use cases they saw involved
separate transport flows for the separate layers, and if you're sending all
the layers on a single transport flow, SST is significantly simpler.

 

What requirement are you trying to achieve that's met by MST in a single
session, but not by SST?

 

--

Jonathan Lennox
jonathan@vidyo.com

 


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi Jonathan,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Maybe if you want to use FEC only on the base layer and not on all =
layers.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf Of =
</b>Jonathan Lennox<br><b>Sent:</b> 19 July, 2013 5:54 PM<br><b>To:</b> =
Bernard Aboba<br><b>Cc:</b> mmusic@ietf.org<br><b>Subject:</b> Re: =
[MMUSIC] RFC 6190 Single Session =
Transport<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal>On =
Jul 18, 2013, at 10:02 PM, Bernard Aboba &lt;<a =
href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&g=
t; wrote:<o:p></o:p></p></div><p =
class=3DMsoNormal><br><br><o:p></o:p></p><div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-family:"Calibri","sans-serif"'>Jonathan said:<span =
class=3Dapple-converted-space>&nbsp;</span><o:p></o:p></span></p><div><p =
class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif"'>&quot;I don't understand =
how single-session multi-source transmission will work without either a) =
signaling &quot;a=3Dssrc-group DDP&quot; decoding dependencies, or b) =
making various implementation-specific assumptions about the structure =
of the SVC streams.&quot;<br>&nbsp;<br>[BA] In terms of SDP choices for =
expressing layering within a single m line, you can either go the =
a=3Dssrc-group:DDP route, or use distinct payload types for each layer =
and RFC 5583 a=3Dgroup:DDP (this is what &quot;Unified Plan&quot; =
Section 4.7 appears to advocate).&nbsp;<span =
class=3Dapple-converted-space>&nbsp;</span><br><br>I'm not sure how =
&quot;MST within a single session&quot; could be expressed in SDP =
without BUNDLE.&nbsp; Was the idea to use multiple m lines and group =
them together but use the same port?<span =
class=3Dapple-converted-space>&nbsp;</span><o:p></o:p></span></p></div></=
div></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>I think the authors of RFC 6190 didn't see a use case =
for MST within a single RTP session -- all of the interesting MST use =
cases they saw involved separate transport flows for the separate =
layers, and if you're sending all the layers on a single transport flow, =
SST is significantly simpler.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>What requirement are you trying to achieve that's met =
by MST in a single session, but not by SST?<o:p></o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:13.5pt;font-family:"Helvetica","sans-serif";color:blac=
k'>--<o:p></o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:13.5pt'><span =
style=3D'font-size:13.5pt;font-family:"Helvetica","sans-serif";color:blac=
k'>Jonathan Lennox<br><a =
href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</a><o:p></o:p></spa=
n></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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From christer.holmberg@ericsson.com  Thu Aug 22 02:37:22 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dan Wing <dwing@cisco.com>
Thread-Topic: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Cc: "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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Hi Dan,

Regarding the 3GPP SA3 study, the information I gave was a little misleadin=
g. From a security perspective SRTP would be fine.

The focus of the SA3 study was on how to provide security for the existing =
fax transmission mechanism, which uses UDPTL/UDP.

3GPP already mandates IMS terminals to support UDPTL/UDP for unsecure fax. =
And, new terminals (supporting secure fax) are still required to also suppo=
rt unsecure fax, in order to communicate with legacy terminals when unsecur=
e fax is sufficient.=20

So, using UDPTL/DTLS/UDP for secure fax makes more sense, as it avoids impl=
ementing different mechanisms for secure and unsecure fax - UDPTL/DTLS/UDP =
only requires a new layer between UDPTL and UDP, it does not require changi=
ng the upper layers (UDPTL and above).

Hopefully this clarifies :)

Regards,

Christer





-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: 19. elokuuta 2013 20:43
To: Christer Holmberg
Cc: mmusic; mmusic-chairs@tools.ietf.org
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00


On Aug 19, 2013, at 6:03 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
> =20
> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, which=
 defines usage of UDPTL over DTLS, in order to provide secure fax.
> =20
> The draft was previously submitted to DISPATCH. Based on discussions with=
 the ADs and chairs, it was decided that it shall be submitted to MMUSIC (n=
ote that no DTLS extensions are needed).
> =20
> As is indicated in the draft, 3GPP has performed a study on how to=20
> provide secure fax in the IMS, and the outcome was that secure fax shall =
be transported using UDPTL over DTLS.

Got a pointer to that study?  Seems easier to carry UDPTL over RTP, which w=
ould allow the RTP to be secured using SRTP (and thus the UDPTL would be se=
cured using SRTP).  There is a spec floating around to do exactly that (car=
ry fax over RTP so that SRTP can secure it).  Advantage of using SRTP to se=
cure fax is it separates the keying mechanism from security, so that Securi=
ty Descriptions / MIKEY / DTLS-SRTP / whatever-is-invented-in-2020 will wor=
k just as effectively for voice as for fax.  And also that upgrading from a=
 voice call to a "fax" call has no additional complexities due to security =
("please press START to begin the fax transmission").

-d


> However, there is nothing "3GPP/IMS specific" about the mechanism, as UDP=
TL is commonly used for fax also elsewhere.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: Atle Monrad <atle.monrad@ericsson.com>
To: Andrew Allen <aallen@blackberry.com>, Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Propo
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Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Propo
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Folks

I'd like to support Andrew on this topic.

I don't know if bundling these attributes together in one draft was done ju=
st for convenience or of there are other reasons. What is clear is that the=
se three attributes were not related initially.

3GPP has no use of all attributes as explained by Andrew. I hope that this =
late change can be reverted (or modified in a less radical way) and the dra=
ft completed soon.

3GPP has been waiting for completion of this draft for long now. I think we=
 can assume that there are implementations in the field, thus these last mi=
nute changes may cause problems.

Please also consider that the 3GPP-feature using this draft is "IMS Central=
ized Service Control". This feature is part of 3GPP Rel-8, a release that w=
as completed from 3GPP point if view by June 2008.

Thanks
/atle

________________________________


Atle Monrad
3GPP CT Chairman

Group Function Technology - Standardization and Technical Regulation
Ericsson


From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Andrew Allen
Sent: 21. august 2013 19:31
To: Flemming Andreasen; mmusic
Subject: Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Pro=
tocol Action: 'Session Description Protocol (SDP) Extension For Setting Aud=
io and Video Media Streams Over Circuit-Switched Bearers In The Public Swit=
ched Telephone Network (PSTN)' to Propo


In my view since including all the correlation attributes in the SDP is a S=
HOULD strength implementing all the correlation attributes should  also be =
of SHOULD strength or at most MUST implement except when it is known that a=
 particular correlation mechanism will work for the deployment scenario tha=
t  circuit-switched bearer is used. It seems to me kind of silly to have a =
MUST implement (but might not ever try and use requirement).



To my knowledge 3GPP are currently the only party interested in using this =
draft. To publish this draft with mandatory aspects which are not needed an=
d not acceptable to 3GPP application will in my view be a mistake and resul=
t in an RFC that will either not be implemented or will not be implemented =
as written because the overhead of implementing the DTMF and UUI correlatio=
n mechanisms is not justified and since these are not needed in the 3GPP us=
e case and will not work with the 3GPP architecture either.



In the 3GPP application of this draft the negotiation of the circuit-switch=
ed bearer does not take place end to end between the terminals but instead =
between the terminal and an IMS network infrastructure component (the SCC A=
S) that acts as B2BUA and the CS call is then interworked to SIP and patche=
d in with the session with the remote party. In addition the E.164 number p=
rovided by the IMS network infrastructure component is unique for that circ=
uit-switched bearer setup so no additional correlation mechanism is really =
needed. The Mobile terminal always acts as the active party in establishing=
 the circuit switched bearer in the 3GPP scenario.



Mandating the callerid mechanism for correlation is acceptable to me since =
it basically has no impact on the mobile terminal or the IMS network infras=
tructure component except the inclusion of the cs-correlation attribute and=
 the inclusion of that by the interworking MSC in the P-Asserted-Identity (=
which is already done). It may have the advantage that now the IMS network =
infrastructure component does not have to allocate a unique E.164 number fo=
r each session but could just use the callerid for the correlation. I think=
 this also does mean that the SCC AS can then be sure that the CS session s=
etup is coming from the terminal  that was negotiated it in the O/A exchang=
e and not some attacker trying to hijack sessions using as the dialed numbe=
r an E.164 number that was previously assigned to him. Also since in the 3G=
PP scenario the circuit-switched network is local to and under the control =
of the operator the callerid mechanism can be guaranteed to work.



The DTMF and UUI mechanisms are quite intensive to implement and both have =
issues for the 3GPP application. DTMF will result in a delay in the setup e=
nd to end of the media and also would require media to be routed to the IMS=
 network infrastructure component (which is only a signaling B2BUA) and for=
ce it to have to support  DTMF tone detection. UUI will almost certainly no=
t work as most operators switchers do not pass this information and also wo=
uld require support for the CUSS UUI mechanisms to deliver the UUI in SIP f=
rom the interworking switch to the SCC AS.


I am not against the end to end use of CS SDP but I think it does have some=
 issues such as the cost of long distance or even international circuit swi=
tched bearers when the only leg that is a problem is the access for one mob=
ile user. Also a device that needs to use circuit switched bearers cannot d=
o so with those device that don't also support CS SDP or that also have acc=
ess to a circuit-switched connection. So I don't think the correlation issu=
es with end to end use are the primary obstacles to successful interoperabi=
lity. One way to address the interoperability issues is an architecture lik=
e 3GPP's where the circuit-switched bearer is negotiated between the device=
 and an interworking function at the edge of the network which is where the=
 quality problem exists in the first place not to set up a CS call end to e=
nd.

In my view it is not the end of the world if in some scenarios the user has=
 to manually accept the circuit switched bearer and patch it into the multi=
media session. In fact there may be some really valid use cases where the d=
evice that receives the CS setup is not the same device that negotiated the=
 CS session using SDP. Think of a webex type conference where you connect i=
nto a conference using SIP and then get the conference bridge to call you b=
ack with a PSTN call to your POTS phone - Manual acceptance seems fine to m=
e in this case and automatic correlation will not work. Similarly an intern=
et  gaming session established using SIP might call the user back at his PO=
TS line for audio conversations associated with the game.


One possibility is to state in the draft that if you cannot confirm that th=
e CS call is part of the session negotiated using SDP using a correlation m=
echanism or some other means you MUST NOT accept the CS bearer as part of t=
he session without manual user acceptance.

I am Ok for the callerid correlation mechanism to be made mandatory to impl=
ement. I think this is the mechanism with the least burden for implementati=
ons and also the one most likely to work. I do not agree that the DTMF and =
the UUI should be mandatory to implement.

I also think it should be realized that implementations may only support on=
e direction for the CS call establishment and that the endpoints only need =
to support the aspects of the correlation mechanism for the direction(s) th=
ey support. Also some users might also not wish to be a passive party (such=
 as when calling party pays for example when cellular roaming) even if they=
 have the capability to do so. Thus some implementations may only support t=
he aspects of the correlation mechanism for the direction (active or passiv=
e) that they support.

Andrew

From: mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [mailto:mmusi=
c-bounces@ietf.org] On Behalf Of Flemming Andreasen
Sent: Friday, August 09, 2013 6:10 PM
To: mmusic
Subject: [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: Protoco=
l Action: 'Session Description Protocol (SDP) Extension For Setting Audio a=
nd Video Media Streams Over Circuit-Switched Bearers In The Public Switched=
 Telephone Network (PSTN)' to Propos...

Greetings MMUSIC

As you can see per the below, the IESG approved the Circuit-Switched Bearer=
s draft for publication, however before doing so, they requested a couple o=
f changes. The resulting approved document can be found at http://tools.iet=
f.org/html/draft-ietf-mmusic-sdp-cs-21

One of the changes introduced resulted in all 3 correlation mechanisms now =
being mandatory-to-implement (previously they were not):
<quote>

   Implementations MUST support the three correlation mechanisms

   specified in Section 5.2.3.2<http://tools.ietf.org/html/draft-ietf-mmusi=
c-sdp-cs-21#section-5.2.3.2>, Section 5.2.3.3<http://tools.ietf.org/html/dr=
aft-ietf-mmusic-sdp-cs-21#section-5.2.3.3> and Section 5.2.3.4<http://tools=
.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.4>,

   and SHOULD include all supported mechanisms the initial Offer.
</quote>
Since this is a non-trivial change that has not been discussed by the WG (a=
nd we know of at least one person having expressed concerns with it off-lin=
e), we are hereby solicting the WG participants for input on changes introd=
uced as a result of the IESG review with a focus on the above. In doing so,=
 please keep in mind that we are developing standards that work in the gene=
ral Internet.


The following URL provides a diff-file showing the changes resulting from t=
he IESG review:
    http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&ur=
l2=3Ddraft-ietf-mmusic-sdp-cs-21.txt

Comments are due by August 23 (two weeks from today)

Thanks

-- Flemming (MMUSIC co-chair)


-------- Original Message --------
Subject:

[MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For=
 Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The=
 Public Switched Telephone Network (PSTN)' to Proposed Standard (draft-ietf=
-mmusic-sdp-cs-21.txt)

Date:

Mon, 1 Jul 2013 11:56:23 -0700

From:

The IESG <iesg-secretary@ietf.org><mailto:iesg-secretary@ietf.org>

To:

IETF-Announce <ietf-announce@ietf.org><mailto:ietf-announce@ietf.org>

CC:

mmusic chair <mmusic-chairs@tools.ietf.org><mailto:mmusic-chairs@tools.ietf=
.org>, mmusic mailing list <mmusic@ietf.org><mailto:mmusic@ietf.org>, RFC E=
ditor <rfc-editor@rfc-editor.org><mailto:rfc-editor@rfc-editor.org>



The IESG has approved the following document:

- 'Session Description Protocol (SDP) Extension For Setting Audio and

   Video Media Streams Over Circuit-Switched Bearers In The Public

   Switched Telephone Network (PSTN)'

  (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard



This document is the product of the Multiparty Multimedia Session Control

Working Group.



The IESG contact persons are Gonzalo Camarillo and Richard Barnes.



A URL of this Internet Draft is:

http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/









*Technical Summary

*



The document describes use cases, requirements, and protocol extensions

for using the Session Description Protocol (SDP) Offer/Answer model for

establishing audio and video media streams over circuit-switched bearers

in the Publich Switched Telephone Network (PSTN)



*Working Group Summary

*



The WG had some discussion around the format to use for E.164 numbers

and whether to align this with the existing definition in RFC 3108. The

RFC 3108 definition was seen as deficient and the WG agreed it was

better to align with relevant parts of the tel URI format defined in RFC

3966, not least since SDP address types are defined in the context of a

particular network type, and hence RFC 3108 compatibility is not a

concern (the implication is that the "E164" address type may differ

between network types in SDP).



*Document Quality

*



There are currently no known implementations of the draft, however the

draft is a dependency for 3GPP, so future implementations are expected.



The document has received good overall review in the WG, some of which

resulted in changes to the detailed specification. The document has been

reviewed in detail several times, including of the last few versions.

The major contributors to these as well as earlier discussions are

listed in the Acknowledgements section of the document.





*Personnel

*



Document Shepherd: Flemming Andreasen

Responsible AD: Gonzalo Camarillo

_______________________________________________

mmusic mailing list

mmusic@ietf.org<mailto:mmusic@ietf.org>

https://www.ietf.org/mailman/listinfo/mmusic

.




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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
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p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I&#8217;d like to support=
 Andrew on this topic.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I don&#8217;t know if bun=
dling these attributes together in one draft was done just for convenience =
or of there are other reasons. What is clear is that these three
 attributes were not related initially.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">3GPP has no use of all at=
tributes as explained by Andrew. I hope that this late change can be revert=
ed (or modified in a less radical way) and the draft completed
 soon.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">3GPP has been waiting for=
 completion of this draft for long now. I think we can assume that there ar=
e implementations in the field, thus these last minute changes
 may cause problems.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Please also consider that=
 the 3GPP-feature using this draft is &#8220;IMS Centralized Service Contro=
l&#8221;. This feature is part of 3GPP Rel-8, a release that was completed
 from 3GPP point if view by June 2008.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">/atle<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><b><span style=3D"font=
-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F=
497D">________________________________</span></b><span style=3D"font-size:1=
0.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D">
<br>
<br>
<br>
<b>Atle Monrad</b><br>
3GPP CT Chairman<br>
<br>
Group Function Technology &#8211; Standardization and Technical Regulation<=
/span><span style=3D"font-size:10.0pt;color:#1F497D">
</span><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;=
sans-serif&quot;;color:#1F497D"><br>
Ericsson</span><span style=3D"font-family:&quot;Arial&quot;,&quot;sans-seri=
f&quot;;color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-fa=
mily:&quot;Arial&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o=
:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> mmusic-bounces@ietf.org [mailto:mmusic-bounces@ie=
tf.org]
<b>On Behalf Of </b>Andrew Allen<br>
<b>Sent:</b> 21. august 2013 19:31<br>
<b>To:</b> Flemming Andreasen; mmusic<br>
<b>Subject:</b> Re: [MMUSIC] Comments on IESG Changes in the -cs draft ? [F=
wd: Protocol Action: 'Session Description Protocol (SDP) Extension For Sett=
ing Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)'
 to Propo<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">In my view since=
 including all the correlation attributes in the SDP is a SHOULD strength i=
mplementing all the correlation attributes should&nbsp; also be of SHOULD s=
trength or at most MUST implement except
 when it is known that a particular correlation mechanism will work for the=
 deployment scenario that&nbsp; circuit-switched bearer is used. It seems t=
o me kind of silly to have a MUST implement (but might not ever try and use=
 requirement).
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">To my knowledge =
3GPP are currently the only party interested in using this draft. To publis=
h this draft with mandatory aspects which are not needed and not acceptable=
 to 3GPP application will in my view
 be a mistake and result in an RFC that will either not be implemented or w=
ill not be implemented as written because the overhead of implementing the =
DTMF and UUI correlation mechanisms is not justified and since these are no=
t needed in the 3GPP use case and
 will not work with the 3GPP architecture either.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">In the 3GPP appl=
ication of this draft the negotiation of the circuit-switched bearer does n=
ot take place end to end between the terminals but instead between the term=
inal and an IMS network infrastructure
 component (the SCC AS) that acts as B2BUA and the CS call is then interwor=
ked to SIP and patched in with the session with the remote party. In additi=
on the E.164 number provided by the IMS network infrastructure component is=
 unique for that circuit-switched
 bearer setup so no additional correlation mechanism is really needed. The =
Mobile terminal always acts as the active party in establishing the circuit=
 switched bearer in the 3GPP scenario.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">Mandating the ca=
llerid mechanism for correlation is acceptable to me since it basically has=
 no impact on the mobile terminal or the IMS network infrastructure compone=
nt except the inclusion of the cs-correlation
 attribute and the inclusion of that by the interworking MSC in the P-Asser=
ted-Identity (which is already done). It may have the advantage that now th=
e IMS network infrastructure component does not have to allocate a unique E=
.164 number for each session but
 could just use the callerid for the correlation. I think this also does me=
an that the SCC AS can then be sure that the CS session setup is coming fro=
m the terminal &nbsp;that was negotiated it in the O/A exchange and not som=
e attacker trying to hijack sessions
 using as the dialed number an E.164 number that was previously assigned to=
 him. Also since in the 3GPP scenario the circuit-switched network is local=
 to and under the control of the operator the callerid mechanism can be gua=
ranteed to work.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt">The DTMF and UUI=
 mechanisms are quite intensive to implement and both have issues for the 3=
GPP application. DTMF will result in a delay in the setup end to end of the=
 media and also would require media
 to be routed to the IMS network infrastructure component (which is only a =
signaling B2BUA) and force it to have to support&nbsp; DTMF tone detection.=
 UUI will almost certainly not work as most operators switchers do not pass=
 this information and also would require
 support for the CUSS UUI mechanisms to deliver the UUI in SIP from the int=
erworking switch to the SCC AS.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext">I am not against the end to end use of =
CS SDP but I think it does have some issues such as the cost of long distan=
ce or even international circuit switched bearers when the
 only leg that is a problem is the access for one mobile user. Also a devic=
e that needs to use circuit switched bearers cannot do so with those device=
 that don&#8217;t also support CS SDP or that also have access to a circuit=
-switched connection. So I don&#8217;t think
 the correlation issues with end to end use are the primary obstacles to su=
ccessful interoperability. One way to address the interoperability issues i=
s an architecture like 3GPP's where the circuit-switched bearer is negotiat=
ed between the device and an interworking
 function at the edge of the network which is where the quality problem exi=
sts in the first place not to set up a CS call end to end.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext">In my view it is not the end of the wor=
ld if in some scenarios the user has to manually accept the circuit switche=
d bearer and patch it into the multimedia session. In fact
 there may be some really valid use cases where the device that receives th=
e CS setup is not the same device that negotiated the CS session using SDP.=
 Think of a webex type conference where you connect into a conference using=
 SIP and then get the conference
 bridge to call you back with a PSTN call to your POTS phone &#8211; Manual=
 acceptance seems fine to me in this case and automatic correlation will no=
t work. Similarly an internet&nbsp; gaming session established using SIP mi=
ght call the user back at his POTS line for
 audio conversations associated with the game. <o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext">One possibility is to state in the draf=
t that if you cannot confirm that the CS call is part of the session negoti=
ated using SDP using a correlation mechanism or some other
 means you MUST NOT accept the CS bearer as part of the session without man=
ual user acceptance.<br>
<br>
I am Ok for the callerid correlation mechanism to be made mandatory to impl=
ement. I think this is the mechanism with the least burden for implementati=
ons and also the one most likely to work. I do not agree that the DTMF and =
the UUI should be mandatory to implement.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:windowtext">I also think it should be realized that=
 implementations may only support one direction for the CS call establishme=
nt and that the endpoints only need to support the aspects
 of the correlation mechanism for the direction(s) they support. Also some =
users might also not wish to be a passive party (such as when calling party=
 pays for example when cellular roaming) even if they have the capability t=
o do so. Thus some implementations
 may only support the aspects of the correlation mechanism for the directio=
n (active or passive) that they support.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:windowtext">Andrew<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:windowtext"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext">
<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [<a =
href=3D"mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
<b>On Behalf Of </b>Flemming Andreasen<br>
<b>Sent:</b> Friday, August 09, 2013 6:10 PM<br>
<b>To:</b> mmusic<br>
<b>Subject:</b> [MMUSIC] Comments on IESG Changes in the -cs draft ? [Fwd: =
Protocol Action: 'Session Description Protocol (SDP) Extension For Setting =
Audio and Video Media Streams Over Circuit-Switched Bearers In The Public S=
witched Telephone Network (PSTN)'
 to Propos...<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Greetings MMUSIC<br>
<br>
As you can see per the below, the IESG approved the Circuit-Switched Bearer=
s draft for publication, however before doing so, they requested a couple o=
f changes. The resulting approved document can be found at
<a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21">http://t=
ools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21</a><br>
<br>
One of the changes introduced resulted in all 3 correlation mechanisms now =
being mandatory-to-implement (previously they were not):<br>
&lt;quote&gt;<o:p></o:p></p>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&n=
bsp;&nbsp; Implementations MUST support the three correlation mechanisms<o:=
p></o:p></span></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&n=
bsp;&nbsp; specified in <a href=3D"http://tools.ietf.org/html/draft-ietf-mm=
usic-sdp-cs-21#section-5.2.3.2">Section 5.2.3.2</a>, <a href=3D"http://tool=
s.ietf.org/html/draft-ietf-mmusic-sdp-cs-21#section-5.2.3.3">Section 5.2.3.=
3</a> and <a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-21=
#section-5.2.3.4">Section 5.2.3.4</a>,<o:p></o:p></span></pre>
<pre style=3D"page-break-before:always"><span style=3D"font-size:12.0pt">&n=
bsp;&nbsp; and SHOULD include all supported mechanisms the initial Offer.<o=
:p></o:p></span></pre>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&lt;/quote&gt;<br>
Since this is a non-trivial change that has not been discussed by the WG (a=
nd we know of at least one person having expressed concerns with it off-lin=
e), we are hereby solicting the WG participants for input on changes introd=
uced as a result of the IESG review
 with a focus on the above. In doing so, please keep in mind that we are de=
veloping standards that work in the general Internet.
<br>
<br>
<br>
The following URL provides a diff-file showing the changes resulting from t=
he IESG review:<br>
&nbsp;&nbsp;&nbsp; <a href=3D"http://tools.ietf.org/rfcdiff?url1=3Ddraft-ie=
tf-mmusic-sdp-cs-18.txt&amp;url2=3Ddraft-ietf-mmusic-sdp-cs-21.txt" target=
=3D"_blank">
http://tools.ietf.org/rfcdiff?url1=3Ddraft-ietf-mmusic-sdp-cs-18.txt&amp;ur=
l2=3Ddraft-ietf-mmusic-sdp-cs-21.txt</a><br>
<br>
Comments are due by August 23 (two weeks from today)<br>
<br>
Thanks <br>
<br>
-- Flemming (MMUSIC co-chair)<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><br>
<br>
-------- Original Message -------- <o:p></o:p></p>
<table class=3D"MsoNormalTable" border=3D"0" cellspacing=3D"0" cellpadding=
=3D"0">
<tbody>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Subjec=
t: <o:p></o:p></b></p>
</td>
<td style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal">[MMUSIC] Protocol Action: 'Session Description Proto=
col (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-=
Switched Bearers In The Public Switched Telephone Network (PSTN)' to Propos=
ed Standard (draft-ietf-mmusic-sdp-cs-21.txt)<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>Date: =
<o:p></o:p></b></p>
</td>
<td style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal">Mon, 1 Jul 2013 11:56:23 -0700<o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>From: =
<o:p></o:p></b></p>
</td>
<td style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal">The IESG <a href=3D"mailto:iesg-secretary@ietf.org">=
&lt;iesg-secretary@ietf.org&gt;</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>To: <o=
:p></o:p></b></p>
</td>
<td style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal">IETF-Announce <a href=3D"mailto:ietf-announce@ietf.o=
rg">&lt;ietf-announce@ietf.org&gt;</a><o:p></o:p></p>
</td>
</tr>
<tr>
<td nowrap=3D"" valign=3D"top" style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal" align=3D"right" style=3D"text-align:right"><b>CC: <o=
:p></o:p></b></p>
</td>
<td style=3D"padding:0cm 0cm 0cm 0cm">
<p class=3D"MsoNormal">mmusic chair <a href=3D"mailto:mmusic-chairs@tools.i=
etf.org">&lt;mmusic-chairs@tools.ietf.org&gt;</a>, mmusic mailing list
<a href=3D"mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a>, RFC Editor =
<a href=3D"mailto:rfc-editor@rfc-editor.org">
&lt;rfc-editor@rfc-editor.org&gt;</a><o:p></o:p></p>
</td>
</tr>
</tbody>
</table>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<pre>The IESG has approved the following document:<o:p></o:p></pre>
<pre>- 'Session Description Protocol (SDP) Extension For Setting Audio and<=
o:p></o:p></pre>
<pre>&nbsp;&nbsp; Video Media Streams Over Circuit-Switched Bearers In The =
Public<o:p></o:p></pre>
<pre>&nbsp;&nbsp; Switched Telephone Network (PSTN)'<o:p></o:p></pre>
<pre>&nbsp; (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard<o:p></o:=
p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>This document is the product of the Multiparty Multimedia Session Cont=
rol<o:p></o:p></pre>
<pre>Working Group.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The IESG contact persons are Gonzalo Camarillo and Richard Barnes.<o:p=
></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>A URL of this Internet Draft is:<o:p></o:p></pre>
<pre><a href=3D"http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/">=
http://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/</a><o:p></o:p></p=
re>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Technical Summary<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The document describes use cases, requirements, and protocol extension=
s<o:p></o:p></pre>
<pre>for using the Session Description Protocol (SDP) Offer/Answer model fo=
r<o:p></o:p></pre>
<pre>establishing audio and video media streams over circuit-switched beare=
rs<o:p></o:p></pre>
<pre>in the Publich Switched Telephone Network (PSTN)<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Working Group Summary<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The WG had some discussion around the format to use for E.164 numbers<=
o:p></o:p></pre>
<pre>and whether to align this with the existing definition in RFC 3108. Th=
e<o:p></o:p></pre>
<pre>RFC 3108 definition was seen as deficient and the WG agreed it was<o:p=
></o:p></pre>
<pre>better to align with relevant parts of the tel URI format defined in R=
FC<o:p></o:p></pre>
<pre>3966, not least since SDP address types are defined in the context of =
a<o:p></o:p></pre>
<pre>particular network type, and hence RFC 3108 compatibility is not a<o:p=
></o:p></pre>
<pre>concern (the implication is that the &quot;E164&quot; address type may=
 differ<o:p></o:p></pre>
<pre>between network types in SDP).<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Document Quality<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>There are currently no known implementations of the draft, however the=
<o:p></o:p></pre>
<pre>draft is a dependency for 3GPP, so future implementations are expected=
.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The document has received good overall review in the WG, some of which=
<o:p></o:p></pre>
<pre>resulted in changes to the detailed specification. The document has be=
en<o:p></o:p></pre>
<pre>reviewed in detail several times, including of the last few versions.<=
o:p></o:p></pre>
<pre>The major contributors to these as well as earlier discussions are<o:p=
></o:p></pre>
<pre>listed in the Acknowledgements section of the document.<o:p></o:p></pr=
e>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>*Personnel<o:p></o:p></pre>
<pre>*<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>Document Shepherd: Flemming Andreasen<o:p></o:p></pre>
<pre>Responsible AD: Gonzalo Camarillo <o:p></o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>mmusic mailing list<o:p></o:p></pre>
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From ron.even.tlv@gmail.com  Thu Aug 22 07:02:43 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, <mmusic@ietf.org>
References: <201304251725.r3PHPqeV3429515@shell01.TheWorld.com>	<3879D71E758A7E4AA99A35DD8D41D3D90F6DC561@xmb-rcd-x14.cisco.com>	<51798419.7070103@nostrum.com> <517A23B4.3060801@ericsson.com>	<201304261820.r3QIKq913501941@shell01.TheWorld.com>	<51909E36.9050407@ericsson.com> <C5E08FE080ACFD4DAE31E4BDBF944EB113508AF8@xmb-aln-x02.cisco.com>
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Subject: Re: [MMUSIC] Should we update the IANA registry to reflect RFC 5761?
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Hi,
Based on this thread and the proposal in Adam's  MMSUIC unified proposal we
will write a document that will update RFC3551 and outline the available
ranges.
We will check if we need any comment to the closed RTP Payload types (PT)
for standard audio and video encodings .
This will be done in AVTcore based on the conclusion from the MMUSIC session
in Berlin
Roni Even 
AVTcore co-chair

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Cullen Jennings (fluffy)
> Sent: 21 May, 2013 5:51 PM
> To: mmusic@ietf.org WG
> Cc: payload@ietf.org
> Subject: Re: [MMUSIC] Should we update the IANA registry to reflect RFC
> 5761?
> 
> 
> On May 13, 2013, at 2:03 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
> 
> > I also think you should check with IANA if one can touch a closed
> > registry at all, or if we are restricted to clarifying notes for the
> > registry.
> 
> If we have consensus for the change and IESG approval, I think we can do
> whatever we want to the IANA registries. If we want a totally different
> format for the registry, we can do that - we just need a draft that
updates
> the previous RFC that defined the registries.
> 
> However, I agree with Keith the registry is for code point allocation, not
> explaining life to implementors. If we need an information RFC to
summarize
> and explain all the info to implementors, that would be easy to publish if
> someone wrote it.
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Roni Even <ron.even.tlv@gmail.com>, "'Jonathan Lennox'" <jonathan@vidyo.com>, "'Bernard Aboba'" <bernard_aboba@hotmail.com>
Thread-Topic: [MMUSIC] RFC 6190 Single Session Transport
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Subject: Re: [MMUSIC] RFC 6190 Single Session Transport
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Selective forwarding of layers at a switch is simpler with MST versus SST, =
since SST requires rewriting timestamps and sequence numbers to drop layers=
. RTCP reporting per layer is another benefit of MST. These advantages stem=
 from using multiple SSRCs independent of whether they use the same or sepa=
rate transport flows.

I think the entire nomenclature of SST/MST should evolve to Single/Multiple=
 SSRC (not Session) Transport, since that is really what matters. BUNDLE an=
d UP make this evolution inevitable. Note that MST is already widely deploy=
ed this way, which is non-standard. So  updating the nomenclature would ali=
gn with both current deployments and future standards like BUNDLE/UP.

Mo


From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Roni Even
Sent: Thursday, August 22, 2013 5:05 AM
To: 'Jonathan Lennox'; 'Bernard Aboba'
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RFC 6190 Single Session Transport

Hi Jonathan,
Maybe if you want to use FEC only on the base layer and not on all layers.
Roni

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Jonathan Lennox
Sent: 19 July, 2013 5:54 PM
To: Bernard Aboba
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RFC 6190 Single Session Transport


On Jul 18, 2013, at 10:02 PM, Bernard Aboba <bernard_aboba@hotmail.com<mail=
to:bernard_aboba@hotmail.com>> wrote:

Jonathan said:
"I don't understand how single-session multi-source transmission will work =
without either a) signaling "a=3Dssrc-group DDP" decoding dependencies, or =
b) making various implementation-specific assumptions about the structure o=
f the SVC streams."

[BA] In terms of SDP choices for expressing layering within a single m line=
, you can either go the a=3Dssrc-group:DDP route, or use distinct payload t=
ypes for each layer and RFC 5583 a=3Dgroup:DDP (this is what "Unified Plan"=
 Section 4.7 appears to advocate).

I'm not sure how "MST within a single session" could be expressed in SDP wi=
thout BUNDLE.  Was the idea to use multiple m lines and group them together=
 but use the same port?

I think the authors of RFC 6190 didn't see a use case for MST within a sing=
le RTP session -- all of the interesting MST use cases they saw involved se=
parate transport flows for the separate layers, and if you're sending all t=
he layers on a single transport flow, SST is significantly simpler.

What requirement are you trying to achieve that's met by MST in a single se=
ssion, but not by SST?

--
Jonathan Lennox
jonathan@vidyo.com<mailto:jonathan@vidyo.com>


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Selective forwarding of layers at a swi=
tch is simpler with MST versus SST, since SST requires rewriting timestamps=
 and sequence numbers to drop layers. RTCP reporting per
 layer is another benefit of MST. These advantages stem from using multiple=
 SSRCs independent of whether they use the same or separate transport flows=
.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">I think the entire nomenclature of SST/=
MST should evolve to Single/Multiple SSRC (not Session) Transport, since th=
at is really what matters. BUNDLE and UP make this evolution
 inevitable. Note that MST is already widely deployed this way, which is no=
n-standard. So&nbsp; updating the nomenclature would align with both curren=
t deployments and future standards like BUNDLE/UP.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Mo<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> mmusic-b=
ounces@ietf.org [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Roni Even<br>
<b>Sent:</b> Thursday, August 22, 2013 5:05 AM<br>
<b>To:</b> 'Jonathan Lennox'; 'Bernard Aboba'<br>
<b>Cc:</b> mmusic@ietf.org<br>
<b>Subject:</b> Re: [MMUSIC] RFC 6190 Single Session Transport<o:p></o:p></=
span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Jonathan,<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Maybe if you want to use =
FEC only on the base layer and not on all layers.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Roni<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> mmusic-b=
ounces@ietf.org [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>Jonathan Lennox<br>
<b>Sent:</b> 19 July, 2013 5:54 PM<br>
<b>To:</b> Bernard Aboba<br>
<b>Cc:</b> mmusic@ietf.org<br>
<b>Subject:</b> Re: [MMUSIC] RFC 6190 Single Session Transport<o:p></o:p></=
span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Jul 18, 2013, at 10:02 PM, Bernard Aboba &lt;<a h=
ref=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; =
wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-f=
amily:&quot;Calibri&quot;,&quot;sans-serif&quot;">Jonathan said:<span class=
=3D"apple-converted-space">&nbsp;</span><o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;">&quot;I don't understand how single-session multi-source=
 transmission will work without either a) signaling &quot;a=3Dssrc-group DD=
P&quot; decoding dependencies, or b) making various implementation-specific
 assumptions about the structure of the SVC streams.&quot;<br>
&nbsp;<br>
[BA] In terms of SDP choices for expressing layering within a single m line=
, you can either go the a=3Dssrc-group:DDP route, or use distinct payload t=
ypes for each layer and RFC 5583 a=3Dgroup:DDP (this is what &quot;Unified =
Plan&quot; Section 4.7 appears to advocate).&nbsp;<span class=3D"apple-conv=
erted-space">&nbsp;</span><br>
<br>
I'm not sure how &quot;MST within a single session&quot; could be expressed=
 in SDP without BUNDLE.&nbsp; Was the idea to use multiple m lines and grou=
p them together but use the same port?<span class=3D"apple-converted-space"=
>&nbsp;</span><o:p></o:p></span></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I think the authors of RFC 6190 didn't see a use cas=
e for MST within a single RTP session -- all of the interesting MST use cas=
es they saw involved separate transport flows for the separate layers, and =
if you're sending all the layers on
 a single transport flow, SST is significantly simpler.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">What requirement are you trying to achieve that's me=
t by MST in a single session, but not by SST?<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;;color:black">--<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:13.5pt"><span style=3D"font-s=
ize:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;;color:b=
lack">Jonathan Lennox<br>
<a href=3D"mailto:jonathan@vidyo.com">jonathan@vidyo.com</a><o:p></o:p></sp=
an></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_3879D71E758A7E4AA99A35DD8D41D3D91D508265xmbrcdx14ciscoc_--

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Subject: [MMUSIC] RTSP 2.0: Inconsistent identifiers between Range and Accept-Ranges
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WG,

One of Elwyn's comments are in regards to the lack of IANA registry for
new Accept-Ranges formats. My immediate reaction is that should not be
needed as range formats are common between the Range and Accept-Ranges
header. But, then I checked our definitions are realized that we have
made a blunder to not reuse the range identifiers from Range in
Accept-Ranges.

To resolve this issue we either can do two things.

1) Change the Accept-Ranges header to use the Range header identifiers.
This will affect any implementations of Accept-Ranges header and the
values it uses. Because Accept-Ranges has the following identifiers:

Accept-Ranges: NPT SMPTE UTC

While ranges has the following:
Range: npt, smpte, smpte-30-drop, smpte-25, clock

So in addition to being different there are actually more resolution in
formats for Range header than in Accept ranges.

2) Define a registry and clarify the mapping between the accept-ranges
identifiers and the Range ones.

I personally think 1) is long term better. But, the question is how it
affects any existing implementations of 2.0?

Feedback much desired

Cheers

Magnus Westerlund

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Hi,

Continuing with the issues after Section 11.

Once more a big thanks for the work you have put into this. It is really
helpful, both from clarity point of view and finding some actual
protocol issues.



On 2013-06-06 02:11, Elwyn Davies wrote:

 >
> s13.4.1, para 2:
>> Upon receipt of the PLAY request, the server MUST position the normal
>>    play time to the beginning of the range specified in the received
>>    Range header 
> 
> Should this mention the caveat that (if I understood earlier statements
> correctly) the server may not be able to deliver exactly on this MUST
> and then sets it the nearest it can manage? This is borne out by the
> example later in this section where the server sets the start point back
> 10ms (and says so), etc.  Seek-Style affects the interpreation of MUST!

Yes, I have added that this needs to happen ".. within the limits of the
media resource and in accordance with the Seek-Style header (Section
X.Y) ..".



> 
> s13.4.4/.5/.7/.8: In these sections the Media-Property random access
> property is stated as being required to be Random-Access.  Is
> Conditional Random Access a possible value for this property? The
> confusion in s4.9.1 does not allow me to conclude that this shouldn't be
> a value for the property (sorry for the double negative but this looks a
> bit confused).

These section lists the Media-Properties for the media resource one
request delivery for. Thus from a PLAY request perspective using a
Seek-Style: CoRAP header is fine for the ones that has Random Access
property as Random-Access in these section.


> 
> s13.5.1:  Apologies if this question is stupid - I am not a media stream
> wxpert.  Suppose we have an aggregated stream with (say) a number of
> recorded media items making up the aggregate.  The client sets it on to
> play to the end.  What happens if not all the streams end at the same
> time?  Do we get PLAY_NOTIFY End-of-Stream notifications for each
> individual stream as it ends or just one when the complete aggregate
> ends?

Not at all a stupid question. This should have been clearly specified
for all usages of PLAY_NOTIFY in regards to aggregated sessions. Thus I
have added both a general requirement on PLAY_NOTIFY level:

PLAY_NOTIFY requests MAY use both aggregate control URI and individual
media resource URIs depending on scope of the notification. This scope
may have important distinctions for aggregated sessions, and each reason
for a PLAY_NOTIFY request needs to specify the interpretation and if
aggregated control URIs or individual URIs may be used in requests.

And specified the following for end-of-stream:

For RTSP Session where media resources under aggregated control the
media resources will normally end at approximately the same time,
although some small differences may exist, on the scale of  a few
hundred of milliseconds. In those cases a RTSP session under aggregated
control SHOULD send only a single PLAY_NOTIFY request. By using the
aggregate control URI in the PLAY_NOTIFY request the RTSP server
indicates that this applies to all media resources within the session.
In cases RTP is used for media delivery corresponding RTP-Info needs to
be included for all media resources. In cases where one or more media
resource has significantly shorter duration than some other resources in
the aggregated session the server MAY send end-of-stream notifications
using individual media resource URIs to indicate to agents that there
will be no more media for this particular media resource related to the
current active PLAY request. In such cases when the remaining media
resources comes to end-of-stream they MUST send a PLAY_NOTIFY request
using the aggregate control URI to indicate that no more resources remains.

And the following for Media-Properties-Update

The Media-Properties header has session scope, thus for aggregated
sessions the PLAY_NOTIFY request MUST be using the aggregated control URI.

And for Scale-Change

PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated
control URI in the request. The scale change for any aggregated session
do apply to all media streams part of the aggregate.



> 
> s13.5.2, para 2:
>> that changes the basis for making
>>    estimates on how the media range progress. 
> I can't parse the last part of this and I am not sure what it should be.

I hope this is better

This notification MUST be sent for media that are Time-Progressing every
time an event happens that changes the basis for making estimates on how
the available for play-back media range will progress with wall clock time.


> 
> s14: What happens if the upper layer PDU is longer than (2^16 - 1)?

I have added this note:

Note that this mechanism does not support larger PDUs than 65535 bytes,
which is the same that regular IPv4 and IPv6 is capable. If the media
delivery protocol intended to be used has larger PDUs than that, the
definition of such mechanisms usage of this mechanism will require the
definition of a PDU fragmentation mechanism.



> 
> s18.10, private para:  Is there any way in RTSP for the cache to know
> whether this is a private cache for the user that should cache this
> content?

To my understanding this is something the cache itself knows through its
behavior and deployment.


> s18.10, last para:  I am not absolutely certain, but I think this para
> is specific to the max-age cache type rather than being generic as
> implied by the indentation. [If so use the <vspace blankLines=1/> trick
> to provide a pseudo-paragraph break]

Yes, you are correct. I did the equally easy choice for hanging lists to
simply make a new paragraph without any paragraph text.

> s18.14: Consider adding the 'identity' value explicitly to the ABNF.

If I understand you correctly what you are requesting is that "identity"
is listed as an construct in the content-coding ABNF construction that
are used by Accept-Encoding and Content-Encoding headers.

Accept-Encoding   =  "Accept-Encoding" HCOLON
                     [ encoding *(COMMA encoding) ]
encoding          =  codings [SEMI accept-params]
codings           =  content-coding / "*"
content-coding    =  "identity" / token
Content-Encoding   =  "Content-Encoding" HCOLON
                      content-coding *(COMMA content-coding)


> 
> s18.29, para 2: Are (or should there be) constraints on the ordering of
> media ranges for a non-continuous media stream?  It would probably make
> the client's job easier if they were in incresing order. 

This is a interesting question. If one simply expresses what pieces of
media that are currently available then using a requirement that the
first ranges end-time must be prior any of the subsequent ranges start
times would work. However, my head gets stuck on considering when
desired playback is non linear. But that is probably a higher function
question.

So a possible addition after this existing sentence:

The server MAY include more than one range specification of any given
time format to indicate media that has non-continuous range.

would be:
The range specifications shall be ordered with the range with the lowest
value or earliest start time first, followed by ranges with increasingly
higher values or later start time.

Comments on this?

> s18.41: It appears from a literal interpretation (the third sentence is
> an unqualified MUST) that the responder has to respond with error code
> 551 and an (empty?) list of Unsupported features if it actually does
> support all the features.  I suspect this is not what is wanted.

Correct, reformulated proposal is:

In case any of the feature-tags listed by the Require header are not
supported by the server or client receiving the request, it MUST respond
to the request using the error code 551 (Option Not Supported) and
include  the Unsupported header listing those feature-tags which are NOT
supported.

> s18.45, para 4: Harummph! Political Correctness strikes again.  Every
> protocol should (and probably does) have a CRAP policy. ;-)

Yes, the lower case o is silent here ;-).


> 
> s18.52, 'setup' section:  There seems to be some inconsistency in the
> spcification of the 'active/passive/actpass' values.  a) Is there some
> reason for specifying them in square brackets?

Not, that I can remember. Removing both brackets and quotes.

 b) should they say
> ["active:"] or ["active":] - both formats are used?  The ABNF has the
> plain words with no brackets or quotes.

The later would be correct, but now it will only say
active: <description>

The ABNF is clear that it will be
RTP/SAVPF/TCP;unicast;setup=active;connection=new


> 
> 18.52, 'connection' section, 1st line:
> s/Clients use the setup parameter/Clients use the connection parameter/
> (I assume).

Had to tweak that sentence a bit more:

Clients use the connection parameter in a transport specification part
of the Transport header in a SETUP request, to indicate the client's
preference for either reusing an existing connection between client and
server (in which case the client sets the "connection" parameter to
"existing"), or requesting the creation of a new connection between
client and server (in which cast the client sets the "connection"
parameter to "new").



> 
> s21.1, transfer of sensitive info: Is there an equivalent of the Referer
> header in RTSP?  Not sure there is but it would be good to explain one
> way or the other (either there isn't one so it's less of a problem or
> this is the equivalent...)

Yes, the Referer header is retained also in RTSP.


> 
> s21.1, DNS Spoofing:  Presumably RTSP will interact significantly with
> content delivery networks.  Some of these work by DNS 'spoofing' that
> was in its infancy when RFC 2616 was published.  Are there any other
> issues that come out of the greater prevalence of CDNs since 1999?

I have no experience with large scale deployment of RTSP. To my
knowledge there hasn't been the same development with CDNs for RTSP as
HTTP. This I think is dependent on several factors. One has been the
issues with deploying RTSP on global scale due to NAT traversal etc,
instead progressive download and HTTP adaptive streaming has mostly
taken this market. Thus most large scale deployments has been in
contexts within single operators providing Video On Demand services etc.

Thus, I don't really know of any additional issues around this.


> 
> s21.1, Session Hijacking:  The last part of this is written so that it
> appears to mandate authentication in all cases.  As far as I understand
> it is up to the client to decide if it needs authentication etc and some
> content will be offered on unsecured URIs.  I think this needs to be
> clearer that authentication will mitigate the problem but whether it is
> done is up to the client if the server doesn't care.

Is this better?

Another choice for preventing session hijacking is to use client
authentication and only allow the authenticated client creating the
session to access that session.

> 
> s21.1, Persistently suspicious behaviour:  There seems to be a mismatch
> between the title (persistently..) and the first sentence (a single
> instance).  Do we have examples of the sort of security risk which is
> supposed to be met with a 403 response the first time?

Yes, it is a bit inconsistent. I propose:

Suspicious behavior: RTSP servers upon detecting intances of behavior
which is deemed a security risk SHOULD return error code 403
(Forbidden). RTSP servers SHOULD also be aware of attempts to probe the
server for weaknesses and entry points and MAY arbitrarily disconnect
and ignore further requests from clients which are deemed to be in
violation of local security policy.



> 
> s21.1, last sentence:
>>    Some of the above threats are such that
>>    the implementation of the RTSP functionality itself needs to consider
>>    which policy and strategy it uses to mitigate them.
> It strikes me that this is not very helpful to a prospective
> implementer.  The point of this section is at least in part to provide
> that policy and strategy.  I am not quite sure how to rewrite this (or
> if indeed other people think it does need rewriting). 

No, it is not helpful in other ways then, you need to think about this
and consider what mitigations you think should be implemented and how
that affects your implementation design.

For now I am leaving this as it is.



> s22.3:  The rules about allocating in the sections above x50 should be
> explained here (see s8.1.1).  The  numbers below x50 in each section
> should be marked as a special category of reserved so they are only
> allocated to HTTP analogues.  Is an experimental/private segment in each
> category desirable?   

Yes I think adding something like the below to at least make clear the
intention that RTSP specific status codes should first be registered in
the x50-x99 range.

New RTSP functionality requiring Status Codes should first be registered
in the range x50-x99. Only when the range is full should registrations
be done in the x00-x49 range, unless it is to adopt an HTTP extension
also to RTSP. The reason is to enable any HTTP extension to be adopted
to RTSP without needing to renumber any related status codes.


Regarding experimental and private I am very torn about such range.
Unless someone else in the WG thinks this is worth the work I will leave
it out.

> 
> s22.9: Being allowed to register new Range header formats will currently
> be restricted to times specified by NPT, SMPTE or UTC because no
> registry is defined for Accept-Ranges.  Is this what is intended?

No, the intention is that Range header format registry is shared between
Range and Accept-Ranges. However, I see the description is not clear on
this and needs to be amended.

I have sent a separate email on this.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Subject: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Hi,

One of the open issues related to BUNDLE presented in Berlin (Q5 in my slid=
es) was whether, once a BUNDLE address is re-negotiated (i.e. the usage of =
BUNDLE etc has already been negotiated), it is allowed to assign the new su=
ggested BUNDLE address to all "m=3D" lines immediately, instead of doing th=
e 2-step Offer/Answer procedure. The outcome was that it shall be allowed.

So, based on that, I am suggesting the change below to section 6.1.4 of BUN=
DLE.

NOTE that this only applies to the case where a BUNDLE address is RE-negoti=
ated (e.g. in a re-INVITE). There is still a discussion about the initial O=
ffer, and that outcome of that (and the outcome of the UP discussions) may =
obviously also impact the text. But, let's keep that separate for now, and =
only focus on the case when a BUNDLE address is RE-negotiated :)

Regards,

Christer

-------------------------------------------

OLD TEXT (BUNDLE-04):
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

6.4.1.  SDP Offerer Bundle Address Request and Usage

   An Offerer can assign a BUNDLE address to multiple "m=3D" lines in a
   BUNDLE group, once the Answerer has selected the BUNDLE address for
   the Offerer.  An Offerer MUST NOT assign a BUNDLE address to multiple
   "m=3D" lines until the Answerer has selected the BUNDLE address.

   OPEN ISSUE: Should it be allowed to assign a new BUNDLE address to
   multiple "m=3D" lines in a BUNDLE group, before the Answerer has
   selected the BUNDLE address?

   In order to negotiate (or, to re-negotiate) the BUNDLE address
   associated with a BUNDLE group, the Offerer, in the SDP Offer,
   assigns a unique address to each "m=3D" line in the BUNDLE group.  In
   addition, the Offerer indicates which unique address it wishes the
   Answerer to select as the Offerer's BUNDLE address.  The Offerer
   places the mid, associated with the unique address requested to be
   selected as the BUNDLE address, first in the SDP group:BUNDLE
   attribute mid list.  The Answerer will then select the BUNDLE address
   for the Offerer ([ref-to-be-added]).

   If the Offerer, in a subsequent SDP Offer, wants to re-negotiate the
   BUNDLE address associated with a BUNDLE group, it MAY assign the
   previously negotiated BUNDLE address as a unique address to one of
   the "m=3D" lines in the BUNDLE group.

   If the Offerer assigns the previously selected BUNDLE address to more
   than one "m=3D" line in a BUNDLE group, the first mid in the SDP
   group:BUNDLE attribute mid list MUST represent an "m=3D" line to which
   the BUNDLE address is assigned.  Hence, in order to re-negotiate the
   BUNDLE address, the Offerer needs to assign a unique address to each
   "m=3D" line in the BUNDLE group, as described above.

   An Offerer MUST NOT assign a BUNDLE address to an "m=3D" line that is
   not associated with a BUNDLE group.  An Offerer MUST NOT assign a
   BUNDLE address, that has been negotiated for a BUNDLE group, to an
   "m=3D" line that is associated with another BUNDLE group.

   Section 10.1 shows an example of a Bundle Address Request.



SUGGESTED NEW TEXT:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D


6.4.1.  SDP Offerer Bundle Address Request and Usage



   An Offerer can assign a BUNDLE address to multiple "m=3D" lines in a

   BUNDLE group, once the Answerer has selected the BUNDLE address for

   the Offerer.  An Offerer MUST NOT assign a BUNDLE address to multiple

   "m=3D" lines until the Answerer has selected the BUNDLE address.



   Before the Answerer has selected a BUNDLE address for the Offerer,

   in order to negotiate the BUNDLE address associated with a BUNDLE

   group, the Offerer, in the SDP Offer, assigns a unique address to

   each "m=3D" line in the BUNDLE group. In addition, the Offerer indicates

   which unique address it wishes the Answerer to select as the

   Offerer's BUNDLE address. The Offerer places the mid, associated

   with the unique address requested to be selected as the BUNDLE address,

   first in the SDP group:BUNDLE attribute mid list.  The Answerer will

   then select the BUNDLE address for the Offerer ([ref-to-be-added]).



   If an Offerer, in a subsequent SDP Offer, wants to re-negotiate the

   BUNDLE address associated with a BUNDLE group, it can either:



-    Assign the requested new BUNDLE address to all "m=3D" lines in the

BUNDLE group; or

-    In the same way as described in the procedures above, assign a

unique address to each "m=3D" line in the BUNDLE group (the previously

negotiated BUNDLE address can also be assigned as a unique address

to one of the "m=3D" lines in the BUNDLE group).



   In both cases, the Offerer places the mid, associated with the unique

   address requested to be selected as the BUNDLE address, first in the

   SDP group:BUNDLE attribute mid list (if the requested new BUNDLE address

   is assigned to all "m=3D" lines in the BUNDLE group, all mids in the SDP

   group:BUNDLE attribute mid list will represent that same address).



   An Offerer MUST NOT assign a BUNDLE address to an "m=3D" line that is

   not associated with a BUNDLE group.  An Offerer MUST NOT assign a

   BUNDLE address, that has been negotiated for a BUNDLE group, to an

   "m=3D" line that is associated with another BUNDLE group.



   Section 10.1 shows an example of a Bundle Address Request.


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"FI">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal">One of the open issues related to BUNDLE presented i=
n Berlin (Q5 in my slides) was whether, once a BUNDLE address is re-negotia=
ted (i.e. the usage of BUNDLE etc has already been negotiated), it is allow=
ed to assign the new suggested BUNDLE
 address to all &#8220;m=3D&#8221; lines immediately, instead of doing the =
2-step Offer/Answer procedure. The outcome was that it shall be allowed.<o:=
p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So, based on that, I am suggesting the change below =
to section 6.1.4 of BUNDLE.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">NOTE that this only applies to the case where a BUND=
LE address is RE-negotiated (e.g. in a re-INVITE). There is still a discuss=
ion about the initial Offer, and that outcome of that (and the outcome of t=
he UP discussions) may obviously also
 impact the text. But, let&#8217;s keep that separate for now, and only foc=
us on the case when a BUNDLE address is RE-negotiated :)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">-------------------------------------------<o:p></o:=
p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">OLD TEXT (BUNDLE-04):<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">6.4.1.&nbsp; SDP Offerer Bundle Address Request and Usage<=
o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; An Offerer can assign a BUNDLE address to mul=
tiple &quot;m=3D&quot; lines in a<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; BUNDLE group, once the Answerer has selected =
the BUNDLE address for<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; the Offerer.&nbsp; An Offerer MUST NOT assign=
 a BUNDLE address to multiple<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; &quot;m=3D&quot; lines until the Answerer has=
 selected the BUNDLE address.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; OPEN ISSUE: Should it be allowed to assign a =
new BUNDLE address to<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; multiple &quot;m=3D&quot; lines in a BUNDLE g=
roup, before the Answerer has<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; selected the BUNDLE address?<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; In order to negotiate (or, to re-negotiate) t=
he BUNDLE address<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; associated with a BUNDLE group, the Offerer, =
in the SDP Offer,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; assigns a unique address to each &quot;m=3D&q=
uot; line in the BUNDLE group.&nbsp; In<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; addition, the Offerer indicates which unique =
address it wishes the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; Answerer to select as the Offerer's BUNDLE ad=
dress.&nbsp; The Offerer<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; places the mid, associated with the unique ad=
dress requested to be<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; selected as the BUNDLE address, first in the =
SDP group:BUNDLE<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; attribute mid list.&nbsp; The Answerer will t=
hen select the BUNDLE address<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; for the Offerer ([ref-to-be-added]).<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; If the Offerer, in a subsequent SDP Offer, wa=
nts to re-negotiate the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; BUNDLE address associated with a BUNDLE group=
, it MAY assign the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; previously negotiated BUNDLE address as a uni=
que address to one of<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; the &quot;m=3D&quot; lines in the BUNDLE grou=
p.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; If the Offerer assigns the previously selecte=
d BUNDLE address to more<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; than one &quot;m=3D&quot; line in a BUNDLE gr=
oup, the first mid in the SDP<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; group:BUNDLE attribute mid list MUST represen=
t an &quot;m=3D&quot; line to which<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; the BUNDLE address is assigned.&nbsp; Hence, =
in order to re-negotiate the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; BUNDLE address, the Offerer needs to assign a=
 unique address to each<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; &quot;m=3D&quot; line in the BUNDLE group, as=
 described above.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; An Offerer MUST NOT assign a BUNDLE address t=
o an &quot;m=3D&quot; line that is<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; not associated with a BUNDLE group.&nbsp; An =
Offerer MUST NOT assign a<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; BUNDLE address, that has been negotiated for =
a BUNDLE group, to an<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; &quot;m=3D&quot; line that is associated with=
 another BUNDLE group.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; Section 10.1 shows an example of a Bundle Add=
ress Request.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">SUGGESTED NEW TEXT:<o:p></o:p></p>
<p class=3D"MsoNormal">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<pre>6.4.1.&nbsp; SDP Offerer Bundle Address Request and Usage<o:p></o:p></=
pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; An Offerer can assign a BUNDLE address to multiple &quot;=
m=3D&quot; lines in a<o:p></o:p></pre>
<pre>&nbsp;&nbsp; BUNDLE group, once the Answerer has selected the BUNDLE a=
ddress for<o:p></o:p></pre>
<pre>&nbsp;&nbsp; the Offerer.&nbsp; An Offerer MUST NOT assign a BUNDLE ad=
dress to multiple<o:p></o:p></pre>
<pre>&nbsp;&nbsp; &quot;m=3D&quot; lines until the Answerer has selected th=
e BUNDLE address.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; Before the Answerer has selected a BUNDLE address for the=
 Offerer,<o:p></o:p></pre>
<pre>&nbsp;&nbsp; in order to negotiate the BUNDLE address associated with =
a BUNDLE <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;group, the Offerer, in the SDP Offer, assigns a uniq=
ue address to <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;each &quot;m=3D&quot; line in the BUNDLE group. In a=
ddition, the Offerer indicates <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;which unique address it wishes the Answerer to selec=
t as the <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;Offerer's BUNDLE address. The Offerer places the mid=
, associated <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;with the unique address requested to be selected as =
the BUNDLE address, <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;first in the SDP group:BUNDLE attribute mid list.&nb=
sp; The Answerer will <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;then select the BUNDLE address for the Offerer ([ref=
-to-be-added]).<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; If an Offerer, in a subsequent SDP Offer, wants to re-neg=
otiate the<o:p></o:p></pre>
<pre>&nbsp;&nbsp; BUNDLE address associated with a BUNDLE group, it can eit=
her:<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre style=3D"margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo=
1"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=3D"fon=
t:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp; </span></span><![en=
dif]>Assign the requested new BUNDLE address to all &#8220;m=3D&#8221; line=
s in the <o:p></o:p></pre>
<pre style=3D"margin-left:36.0pt">BUNDLE group; or<o:p></o:p></pre>
<pre style=3D"margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo=
1"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=3D"fon=
t:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp; </span></span><![en=
dif]>In the same way as described in the procedures above, assign a <o:p></=
o:p></pre>
<pre style=3D"margin-left:36.0pt">unique address to each &#8220;m=3D&#8221;=
 line in the BUNDLE group (the previously <o:p></o:p></pre>
<pre style=3D"margin-left:36.0pt">negotiated BUNDLE address can also be ass=
igned as a unique address <o:p></o:p></pre>
<pre style=3D"margin-left:36.0pt">to one of the &#8220;m=3D&#8221; lines in=
 the BUNDLE group).<o:p></o:p></pre>
<pre style=3D"margin-left:36.0pt"><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; In both cases, the Offerer places the mid, associated wit=
h the unique <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;address requested to be selected as the BUNDLE addre=
ss, first in the <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;SDP group:BUNDLE attribute mid list (if the requeste=
d new BUNDLE address <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;is assigned to all &#8220;m=3D&#8221; lines in the B=
UNDLE group, all mids in the SDP <o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;group:BUNDLE attribute mid list will represent that =
same address).<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; An Offerer MUST NOT assign a BUNDLE address to an &quot;m=
=3D&quot; line that is<o:p></o:p></pre>
<pre>&nbsp;&nbsp; not associated with a BUNDLE group.&nbsp; An Offerer MUST=
 NOT assign a<o:p></o:p></pre>
<pre>&nbsp;&nbsp; BUNDLE address, that has been negotiated for a BUNDLE gro=
up, to an<o:p></o:p></pre>
<pre>&nbsp;&nbsp; &quot;m=3D&quot; line that is associated with another BUN=
DLE group.<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>&nbsp;&nbsp; Section 10.1 shows an example of a Bundle Address Request=
.<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From christer.holmberg@ericsson.com  Fri Aug 23 03:53:06 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: DECISION (BERLIN-Q1/Q2/Q7): Usage of specific PT value for single codec configuration
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Date: Fri, 23 Aug 2013 10:52:55 +0000
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Subject: [MMUSIC] DECISION (BERLIN-Q1/Q2/Q7): Usage of specific PT value for single codec configuration
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Hi,

One of the open issues related to BUNDLE presented in Berlin (Q1, Q2 and Q7=
 in my slides) was regarding the usage of any given PT value.

My take of the outcome was that:


-          We agreed that any given PT value can only be used for a single =
codec configuration (Q1)

-          We agreed that we need some definition text for "codec configura=
tion" (Q2)

-          We agreed that, within a BUNDLE group, any given PT value can be=
 used in multiple m- lines, as long as it is used for a single codec config=
uration (Q7).

Jonathan Lennox promised to suggest some definition text for "codec configu=
ration", so let's wait for that until we go into those discussions.

But, if you have any issues with the agreements in general, please bring th=
em up :)

Regards,

Christer




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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">One of the open issues related to BUNDLE presented i=
n Berlin (Q1, Q2 and Q7 in my slides) was regarding the usage of any given =
PT value.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">My take of the outcome was that:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>We <b>agreed</b> that any given PT value can only b=
e used for a single codec configuration (Q1)<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>We <b>agreed</b> that we need some definition text =
for &#8220;codec configuration&#8221; (Q2)<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo1"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>We <b>agreed</b> that, within a BUNDLE group, any g=
iven PT value can be used in multiple m- lines, as long as it is used for a=
 single codec configuration (Q7).<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Jonathan Lennox promised to suggest some definition =
text for &#8220;codec configuration&#8221;, so let&#8217;s wait for that un=
til we go into those discussions.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">But, if you have any issues with the agreements in g=
eneral, please bring them up :)<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"FI"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C47766DESESSMB209erics_--

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Hi,

Now getting to implement your response in our source.

Some few additional comments or notes below.

On 2013-06-25 14:47, Elwyn Davies wrote:
> 
> On Mon, 2013-06-24 at 16:51 +0200, Magnus Westerlund wrote:
>> Elwyn,
>>
>> Follow up on some of the nits that is not simply to implement or where I
>> disagree or want to provide feedback.
>>
>> This email covers all issues up to and including Section 11. That is as
>> far as I have managed to process them yet. I will now handle over the
>> rest of this list to my co-authors.
>>
>> On 2013-06-06 02:11, Elwyn Davies wrote:




>>>
>>>
>>> ss4.4/4.5/4.6: If I read the ABNF in s20.2.1 correctly, the unqualified 
>>> values "smpte", "npt" and "clock" are valid  in the Range parameter.  
>>> What is the meaning of such an unqualified value?
>>
>> You are correct about that the basic definition in ABNF allows for
>> empty. The reason for this is the following stated in Section 18.38:
>>
>>     It MAY be
>>    included in GET_PARAMETER requests from the client to the server with
>>    only a Range format and no value to request the current media
>>    position, whether the session is in Play or Ready state in the
>>    included format.
>>
> Two points:
> - A suggestion: Make ss4.4/4.5/4.6 into subsections of a new s4.4.  Add
> a bit of text in the new s4.4 saying that the range specifiers are used
> in the RANGE header and pointing to s18.38 for the unqualified use.
> (consistent with s2.7 which gives usage).
> - An issue I hadn't spotted: I haven't checked but are the unqualified
> values allowed in the SDP extension case (s20.3)?
>>
>>

I think your suggestion is a good one and I have written such a section,
initial proposal below:

4.4.  Media Time Formats

   RTSP currently supports three different media time formats defined
   below.  Additional time formats may be specified in the future.
   These time formats can be used with the Range header (Section 18.38)
   to request playback and specify at which media position protocol
   requests actually will or has taken place.  They are also used in
   description of the media's properties using the Media-Range header
   (Section 18.29).  The format identifier only are used in Accept-
   Ranges header (Section 18.5) to declare supported time formats and
   also in the Range header (Section 18.38) to request the time format
   used in the response.

4.4.1.  SMPTE Relative Timestamps


Regarding the SDP a=range attribute. The ABNF allows the unqualified
usage. The textual specification of a=range attribute in D.1.6 is silent
on unqualified usage.

The only usage of unqualified usage in a=range would be to indicate that
although this media has unknown or time-progressing properties it does
support the time formats for the a=range with unqualified values that
has been included.



>>
>>>
>>> s10.2, para 6 and following note:  I can't see that the note following 
>>> para 6 can be justified: since the restriction on multiple connections 
>>> from a client to a server is only a SHOULD, an interoperable server MUST 
>>> be capable of handling multiple connections from a client unless the 
>>> server can explicitly send back a TOO MANY CONNECTIONS status.
>>
>> See separate email
> 
> Should I have had this already?

It was sent to the MMUSIC WG mailing list on the 24th of June. I have
forwarded if to you personally.

> 
>>
>>>
>>> s10.6: RFC 3986 has capability for dealing with IP versions beyond 6... 
>>> should these be carried over into RTSP?  The IANA registry says they are.
>>
>> I am uncertain what you want here? Yes, the URI format allows for future
>> definitions of literal IP address versions. However, RTSP does contain
>> explicit IPv4 and IPv6 address fields that aren't URIs. Thus, future IP
>> versions will require some extension definition to function.
> 
> Perhaps an excess of zeal on my behalf.  Perhaps a (non-normative) note
> saying what you just wrote: 'Although the general URI format envisages 
> potential future new versions of the literal IP address, usage of any
> such new version would require other modifications to the RTSP
> specofication (e.g., ...[any relevant section(s)]).' 

Yes, I added such text.


>>
>>>
>>> s10.7, last para: Should the doubling of the wait timer have an upper limit?
>>
>> Yes, I added an upper limit of 30 minutes. This is way beyond what any
>> human will accept, only robots will keep on going at this timeout value.
>> And I think that is large enough to avoid any issues.
> 
> Agreed
> 
>>
>> I also propose to clarify the range for the variations and added the
>> classical 0.5 to 1.5 times the deterministic value.
>>
>> Text proposal.
>>
>> That timer MUST be doubled for each additional failure to connect or
>> receive response until the value exceeds 30 minutes when the timers
>> deterministic value may be set to 30 minutes. It is RECOMMENDED to not
>   ^^^^^^^^^^^^^
>   base?
>> set the same value in the timer for each scheduling, but instead to add
>> a variation to the mean value, resulting in picking a random value
>> within the range of 0.5 to 1.5 of the mean value.

This has gotten changed a bit more and currently reads:

A more complex case may arise when a load balancing RTSP proxy is in
use, i.e., where an RTSP proxy is used to select amongst a set of RTSP
servers to handle the requests, or when multiple server addresses are
available for a given server name. The proxy or client may receive a 503
(Service Unavailable) or 553 (Proxy Unavailable) from one of its RTSP
servers or proxies, or a TCP timeout (if the server is even unable to
handle the request message). The proxy or client simply retries the
other addresses or configured proxies, but may also receive a 503
(Service Unavailable) or 553 (Proxy Unavailable) response or TCP
timeouts from those addresses. In such a situation, where none of the
RTSP servers/proxies/addresses can handle the request, the RTSP agent
has to wait before it can send any new requests to the RTSP server. Any
additional request to a specific address MUST be delayed according to
the Retry-After headers received. For addresses where no response was
received or TCP timeout occurred, an initial wait timer SHOULD be set to
5 seconds. That timer MUST be doubled for each additional failure to
connect or receive response until the value exceeds 30 minutes when the
timers mean value may be set to 30 minutes. It is REQUIRED to not set
the same value in the timer for each scheduling, but instead to add a
variation to the mean value, resulting in picking a random value within
the range of 0.5 to 1.5 of the mean value.


>>
>>>
>>> s11, para 4: According to s22.1.3, play.basic, play.scale and play.speed
>>> are 'defined' in this draft.  The nearest thing to a definition of
>>> play.basic that I can find is the last sentence of para 4 in this
>>> section.  I am not convinced that I could produce an interoperable
>>> implementation of a server with this feature-tag with the one sentence
>>> 'definition'.  References to where play.xxx are defined would be useful
>>> (play.scale in s18.44, play.speed in s18.48).
>>>
>>
>> Hmmm, the other feature-tags than play.basic are pretty straightforward
>> and well contained. play-basic is really a minimal RTSP server
>> implementation according to the whole spec for media playback.
>>
>> This can probably be more clearly expressed somewhere, but it can't
>> really be enumerated. Then we are back to the issue we had with RFC 2326
>> minimal implementation list that was incomplete and lacked a lot of
>> things really necessary.
>>
>> This issue still haven't resulted in any changes to our source.
> 
> Ah!  I see I have (re-)opened a can of worms!  One thought for providing
> the proverbial larger can into which the worms can be stuffed back:
> 
> Define 'play.basic' as the default that you get back if none of the
> others apply.  How's that for weasel words!

After my leave I have written a bit of definition for play.basic:

11.1.  Feature Tag: play.basic

   The play.basic feature tag represents an RTSP implementation
   according to all normative RTSP protocol features specified in this
   specification.  This specification is both a RTSP core specification
   as well intended to enable setup and control of playback of media.
   Thus following all normative parts in the main sections (the ones
   with numbers), not the appendices (starting with letters), unless
   explicitly specified in a main section for being a required appendix.

      Note: This feature tag does not mandate any media delivery
      protocol, such as RTP.

      In RTSP 1.0 there was a minimal implementation section.  However,
      that was not consistent with the rest of the specification.  So
      rather than making an attempt to explicitly enumerate the features
      for play.basic this specification have to be read in completeness
      and the necessary features normatively defined as being required
      are included.

>>
>>
>>> s11, Supported (and Proxy-Supported) bullets: 
>>> OLD:
>>>          This results in the
>>>          receiver is learning the requester's feature support.
>>> NEW:
>>>          This results in the
>>>          receiver learning the requester's feature support 
>>>          relevant to the specified resource.
>>> [Note: I added 'specified resource' here and then thought about whether
>>> this made any sense.  It probably needs to be clearer what set of
>>> features are included in the Supported list for both requester and
>>> responder.  Are they restricted to those relevant to the specified
>>> resource that might of course be the whole server, proxy - or client?? -
>>> in which case there isn't an issue, but it might be a specific
>>> presentation - what happens then?  Affects s13.1, para 3 also.]
>>
>> Actually, I don't think it is specific to the media resource. The issue
>> with making it specific to a media resource is that a RTSP client needs
>> to be able to look into the future and supply an answer for a media
>> resource it may not yet know sufficient to provide a tailored response
>> for. I think it is simpler and better to provide general capabilities in
>> the supported header and then later determine how well they apply for
>> the media resource itself.
> 
> So maybe make it clear that feature support returned is just what the
> responder can do generally and is not specific to the URI provided.

Yes, I have tried to make it clear that it is not resource specific.

   Supported:  This header is used to determine the complete set of
         functionality that both client and server have in general and
         is not dependent on specific resource.  The intended usage is
         to determine before one needs to use a functionality that it is
         supported.  It can be used in any method, but OPTIONS is the
         most suitable one as it at the same time determines all methods
         that are implemented.  When sending a request the requester
         declares all its capabilities by including all supported
         feature-tags.  This results in the receiver is learning the
         requester's feature support.  The receiver then includes its
         set of features in the response.

   Proxy-Supported:  This header is used similarly to the Supported
         header, but instead of giving the supported functionality of
         the client or server it provides both the requester and the
         responder a view of the common functionality supported in
         general by all members of the proxy chain between the two
         supports and not dependent on the resource.  Proxies are
         required to add this header whenever the Supported header is
         present, but proxies may also add it independently of the
         requester.

I also added variations of this sentence to the Supported and
Proxy-Supported headers description:

These feature tags are the ones the server or client support in general,
and is not specific to the request resource.

> 
> Either that or make the request valid only for non-specific URIs (i.e.
> at the server/client level) if that is possible (not sure how that would
> work for server asking client if that is allowed).

Doesn't really work as supported is intended to be possible to add to
any request to determine the functionality supported by the replying agent.

cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From pkyzivat@alum.mit.edu  Tue Aug 27 06:42:51 2013
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Date: Tue, 27 Aug 2013 09:42:44 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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This seems reasonable to me.

	Thanks,
	Paul

On 8/23/13 6:39 AM, Christer Holmberg wrote:
> Hi,
>
> One of the open issues related to BUNDLE presented in Berlin (Q5 in my
> slides) was whether, once a BUNDLE address is re-negotiated (i.e. the
> usage of BUNDLE etc has already been negotiated), it is allowed to
> assign the new suggested BUNDLE address to all “m=” lines immediately,
> instead of doing the 2-step Offer/Answer procedure. The outcome was that
> it shall be allowed.
>
> So, based on that, I am suggesting the change below to section 6.1.4 of
> BUNDLE.
>
> NOTE that this only applies to the case where a BUNDLE address is
> RE-negotiated (e.g. in a re-INVITE). There is still a discussion about
> the initial Offer, and that outcome of that (and the outcome of the UP
> discussions) may obviously also impact the text. But, let’s keep that
> separate for now, and only focus on the case when a BUNDLE address is
> RE-negotiated :)
>
> Regards,
>
> Christer
>
> -------------------------------------------
>
> OLD TEXT (BUNDLE-04):
>
> ====================
>
> 6.4.1.  SDP Offerer Bundle Address Request and Usage
>
>     An Offerer can assign a BUNDLE address to multiple "m=" lines in a
>
>     BUNDLE group, once the Answerer has selected the BUNDLE address for
>
>     the Offerer.  An Offerer MUST NOT assign a BUNDLE address to multiple
>
>     "m=" lines until the Answerer has selected the BUNDLE address.
>
>     OPEN ISSUE: Should it be allowed to assign a new BUNDLE address to
>
>     multiple "m=" lines in a BUNDLE group, before the Answerer has
>
>     selected the BUNDLE address?
>
>     In order to negotiate (or, to re-negotiate) the BUNDLE address
>
>     associated with a BUNDLE group, the Offerer, in the SDP Offer,
>
>     assigns a unique address to each "m=" line in the BUNDLE group.  In
>
>     addition, the Offerer indicates which unique address it wishes the
>
>     Answerer to select as the Offerer's BUNDLE address.  The Offerer
>
>     places the mid, associated with the unique address requested to be
>
>     selected as the BUNDLE address, first in the SDP group:BUNDLE
>
>     attribute mid list.  The Answerer will then select the BUNDLE address
>
>     for the Offerer ([ref-to-be-added]).
>
>     If the Offerer, in a subsequent SDP Offer, wants to re-negotiate the
>
>     BUNDLE address associated with a BUNDLE group, it MAY assign the
>
>     previously negotiated BUNDLE address as a unique address to one of
>
>     the "m=" lines in the BUNDLE group.
>
>     If the Offerer assigns the previously selected BUNDLE address to more
>
>     than one "m=" line in a BUNDLE group, the first mid in the SDP
>
>     group:BUNDLE attribute mid list MUST represent an "m=" line to which
>
>     the BUNDLE address is assigned.  Hence, in order to re-negotiate the
>
>     BUNDLE address, the Offerer needs to assign a unique address to each
>
>     "m=" line in the BUNDLE group, as described above.
>
>     An Offerer MUST NOT assign a BUNDLE address to an "m=" line that is
>
>     not associated with a BUNDLE group.  An Offerer MUST NOT assign a
>
>     BUNDLE address, that has been negotiated for a BUNDLE group, to an
>
>     "m=" line that is associated with another BUNDLE group.
>
>     Section 10.1 shows an example of a Bundle Address Request.
>
> SUGGESTED NEW TEXT:
>
> ===================
>
> 6.4.1.  SDP Offerer Bundle Address Request and Usage
>
>
>
>     An Offerer can assign a BUNDLE address to multiple "m=" lines in a
>
>     BUNDLE group, once the Answerer has selected the BUNDLE address for
>
>     the Offerer.  An Offerer MUST NOT assign a BUNDLE address to multiple
>
>     "m=" lines until the Answerer has selected the BUNDLE address.
>
>
>
>     Before the Answerer has selected a BUNDLE address for the Offerer,
>
>     in order to negotiate the BUNDLE address associated with a BUNDLE
>
>     group, the Offerer, in the SDP Offer, assigns a unique address to
>
>     each "m=" line in the BUNDLE group. In addition, the Offerer indicates
>
>     which unique address it wishes the Answerer to select as the
>
>     Offerer's BUNDLE address. The Offerer places the mid, associated
>
>     with the unique address requested to be selected as the BUNDLE address,
>
>     first in the SDP group:BUNDLE attribute mid list.  The Answerer will
>
>     then select the BUNDLE address for the Offerer ([ref-to-be-added]).
>
>
>
>     If an Offerer, in a subsequent SDP Offer, wants to re-negotiate the
>
>     BUNDLE address associated with a BUNDLE group, it can either:
>
>
>
> -     Assign the requested new BUNDLE address to all “m=” lines in the
>
> BUNDLE group; or
>
> -     In the same way as described in the procedures above, assign a
>
> unique address to each “m=” line in the BUNDLE group (the previously
>
> negotiated BUNDLE address can also be assigned as a unique address
>
> to one of the “m=” lines in the BUNDLE group).
>
>
>
>     In both cases, the Offerer places the mid, associated with the unique
>
>     address requested to be selected as the BUNDLE address, first in the
>
>     SDP group:BUNDLE attribute mid list (if the requested new BUNDLE address
>
>     is assigned to all “m=” lines in the BUNDLE group, all mids in the SDP
>
>     group:BUNDLE attribute mid list will represent that same address).
>
>
>
>     An Offerer MUST NOT assign a BUNDLE address to an "m=" line that is
>
>     not associated with a BUNDLE group.  An Offerer MUST NOT assign a
>
>     BUNDLE address, that has been negotiated for a BUNDLE group, to an
>
>     "m=" line that is associated with another BUNDLE group.
>
>
>
>     Section 10.1 shows an example of a Bundle Address Request.
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From magnus.westerlund@ericsson.com  Wed Aug 28 02:46:40 2013
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Subject: [MMUSIC] RTSP 2.0: Body format negotiation
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WG,

As a result of the below quoted discussion I have created a proposed
text change to make the format negotiation more explicit and clear and
linked to the RTSP methods that have need for it.

The changes I have done in an attempt to address this can be summarized as:
- Create a general negotiation description in a sub-section to the
Message Body section
- Updated GET_PARAMETER and SET_PARAMETER methods to reference this and
be more explicit about handling request formats and returning the same
format.
- Mandate that anyone capable of responding to a GET_PARAMETER and
SET_PARAMETER request MUST support text/parameters
- Updating the Accept header to actually contain description of how you
use it. Missed import from HTTP spec when removing reference.

Please review these changes and provide comments.

Actual text proposal below including whole sections:

9.3.  Message Body Format Negotiation

   The content format of the message body is provided using the Content-
   Type header (Section 18.18).  To enable the responder of a request to
   determine which media type it should use, the requestor may include
   the Accept header (Section 18.1) in a request to identify supported
   media types or media type ranges suitable to the response.  In cases
   the responder is not supporting any of the specified formats, then
   the request response will be a 406 (Not Acceptable) error code.

   The media types that may be used on requests with message bodies
   needs to be determined through the use of feature-tags, specification
   requirement or trial and error.  Trial and error works in the regards
   that in case the responder is not supporting the media type of the
   message body it will respond with a 415 (Unsupported Media Type).

   The formats supported and their negotiation is done individually on a
   per method and direction (request or response body) direction.
   Requirements on supporting particular media types for use as message
   bodies in requests and response SHALL also be specified on per method
   and direction basis.


13.8.  GET_PARAMETER

   The GET_PARAMETER request retrieves the value of any specified
   parameter or parameters for a presentation or stream specified in the
   URI.  If the Session header is present in a request, the value of a
   parameter MUST be retrieved in the specified session context.  There
   are two ways of specifying the parameters to be retrieved.

   The first is by including headers which have been defined such that
   you can use them for this purpose.  Headers for this purpose should
   allow empty, or stripped value parts to avoid having to specify bogus
   data when indicating the desire to retrieve a value.  The successful
   completion of the request should also be evident from any filled out
   values in the response.  The headers in this specification that MAY
   be used for retrieving their current value using GET_PARAMETER below.
   Additional headers MAY be specified in the future:

   o  Accept-Ranges

   o  Media-Range

   o  Media-Properties

   o  Range

   o  RTP-Info

   The other way is to specify a message body that lists the
   parameter(s) that are desired to be retrieved.  The Content-Type
   header (Section 18.18) is used to specify which format the message
   body has.  If the receiver of the request is not supporting the media
   type used for the message body, it SHALL respond using the error code
   415 (Unsupported Media Type).  The responder to a GET_PARAMETER
   request MUST use the media type of the request for the response.  For
   additional considerations regarding message body negotiation see
   Section 9.3.

   RTSP Agents implementing support for responding to GET_PARAMETER
   requests SHALL implement the text/parameters format (Appendix F).
   This to ensure that at least one known format for parameter are
   implemented and thus prevent parameter format negotiation failure.

   Parameters specified within the body of the message must all be
   understood by the request receiving agent.  If one or more parameters
   are not understood a 451 (Parameter Not Understood) MUST be sent
   including a body listing these parameters that weren't understood.
   If all parameters are understood their values are filled in and
   returned in the response message body.

   The method MAY also be used without a message body or any header that
   requests parameters for keep-alive purpose.  The keep-alive timer has
   been updated for any request that is successful, i.e., a 200 OK
   response is received.  Any non-required header present in such a
   request may or may not have been processed.  Normally the presence of
   filled out values in the header will be indication that the header
   has been processed.  However, for cases when this is difficult to
   determine, it is recommended to use a feature-tag and the Require
   header.  For this reason it is usually easier if any parameters to be
   retrieved are sent in the body, rather than using any header.

   Example:

     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 431
           User-Agent: PhonyClient/1.2
           Session: 12345678
           Content-Length: 26
           Content-Type: text/parameters

           packets_received
           jitter

     C->S: RTSP/2.0 200 OK
           CSeq: 431
           Session: 12345678
           Server: PhonyServer/1.1
           Date: Mon, 08 Mar 2010 13:43:23 GMT
           Content-Length: 38
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838


13.9.  SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URI.  The
   method MAY also be used without a message body.  It is the
   RECOMMENDED method to be used in a request sent for the sole purpose
   of updating the keep-alive timer.  If this request is successful,
   i.e., a 200 OK response is received, then the keep-alive timer has
   been updated.  Any non-required header present in such a request may
   or may not have been processed.  To allow a client to determine if
   any such header has been processed, it is necessary to use a feature
   tag and the Require header.  Due to this reason it is RECOMMENDED
   that any parameters are sent in the body, rather than using any
   header.

   When using a message body to list the parameter(s) that are desired
   to be set the Content-Type header (Section 18.18) is used to specify
   which format the message body has.  If the receiver of the request is
   not supporting the media type used for the message body, it SHALL
   respond using the error code 415 (Unsupported Media Type).  For
   additional considerations regarding message body negotiation see
   Section 9.3.

   RTSP Agents implementing support for responding to SET_PARAMETER
   requests SHALL implement the text/parameters format (Appendix F).
   This to ensure that at least one known format for parameters are
   implemented and thus prevent parameter format negotiation failure.

   A request is RECOMMENDED to only contain a single parameter to allow
   the client to determine why a particular request failed.  If the
   request contains several parameters, the server MUST only act on the
   request if all of the parameters can be set successfully.  A server
   MUST allow a parameter to be set repeatedly to the same value, but it
   MAY disallow changing parameter values.  If the receiver of the
   request does not understand or cannot locate a parameter, error 451
   (Parameter Not Understood) MUST be used.  When a parameter is not
   allowed to change, the error code is 458 (Parameter Is Read-Only).
   The response body MUST contain only the parameters that have errors.
   Otherwise no body MUST be returned.  The response body MUST use the
   media type of the request for the response.

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

      Restricting setting transport parameters to SETUP is for the
      benefit of firewalls.

      The parameters are split in a fine-grained fashion so that there
      can be more meaningful error indications.  However, it may make
      sense to allow the setting of several parameters if an atomic
      setting is desirable.  Imagine device control where the client
      does not want the camera to pan unless it can also tilt to the
      right angle at the same time.

   Example:

     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 421
           User-Agent: PhonyClient/1.2
           Session: iixT43KLc
           Date: Mon, 08 Mar 2010 14:45:04 GMT
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/2.0 451 Parameter Not Understood
           CSeq: 421
           Session: iixT43KLc
           Server: PhonyServer/1.0
           Date: Mon, 08 Mar 2010 14:44:56 GMT
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff



18.1.  Accept

   The Accept request-header field can be used to specify certain
   presentation description and parameter media types [RFC6838] which
   are acceptable for the response to DESCRIBE and GET_PARAMETER
   requests.

   See Section 20.2.3 for the syntax.

   The asterisk "*" character is used to group media types into ranges,
   with "*/*" indicating all media types and "type/*" indicating all
   subtypes of that type.  The media-range MAY include media type
   parameters that are applicable to that range.

   Each media-range MAY be followed by one or more accept-params,
   beginning with the "q" parameter for indicating a relative quality
   factor.  The first "q" parameter (if any) separates the media-range
   parameter(s) from the accept-params.  Quality factors allow the user
   or user agent to indicate the relative degree of preference for that
   media-range, using the qvalue scale from 0 to 1 (section 3.9).  The
   default value is q=1.

   Example of use:

     Accept: application/example ;q=0.7, application/sdp

   Indicates that the requesting agent prefers the media type
   application/sdp through the default 1.0 rating but also accepts
   the application/example media type with 0.7 quality rating.

   If no Accept header field is present, then it is assumed that the
   client accepts all media types.  If an Accept header field is
   present, and if the server cannot send a response which is acceptable
   according to the combined Accept field value, then the server SHOULD
   send a 406 (not acceptable) response.


On 2013-06-10 09:52, Magnus Westerlund wrote:
> On 2013-06-07 17:40, Elwyn Davies wrote:
>> On Fri, 2013-06-07 at 16:05 +0200, Magnus Westerlund wrote:
>>
>>>> Appendix F: I missed that the text/parameter format appeared in the
>>>> examples for GET_PARAMETER and SET_PARAMETER. It isn't stated in the
>>>> definitions of these methods what encodings are acceptable for the
>>>> message bodies that may go with these methods and their responses.
>>>
>>> Exactly, that is intentional. These methods can use anything that has a
>>> MIME type. Also it has HTTP's mechanisms for determining what formats
>>> one supports.
>>>
>>>> Clearly the new text/parameter MIME format is one.  Is it the only one?
>>>
>>> It is the only one defined by IETF for this purpose. That is why it is
>>> in the appendix so that RTSP users shall have something to define the
>>> parameters they want in. However, they have to accept the limitations of
>>> a basic text format.
>>>
>>>> I guess you could use a application/json or an XML format but I guess
>>>> these would also either have to be called out explicitly in the method
>>>> descriptions or put in as feature extensions.  This needs to be
>>>> clarified in the method descriptions (s13).  The upshot of all this is
>>>> that I think Appendix F needs to be pulled back into Section 20 as
>>>> currently it is the only way to encode the message bodies.
>>>
>>> I can agree that the format negotiation for the bodies of
>>> GET/SET_PARAMETER is not clear. I will look at proposing some
>>> improvements of the text related to the handling of method bodies and
>>> their format negotiation.
>> Good. I don't see where the server tells what formats it accepts for
>> either GET_PARAMETER or SET_PARAMETER.  Also the Accept spec doesn't say
>> anything about SET_PARAMETER.  AFAICS the client could tell the server
>> what formats it accepts as a side-effect of DESCRIBE but that's a bit
>> arcane - and it is not clear how you would separate the formats for
>> DESCRIBE from those for GET_PARAM and SET_PARAM.
> 
> Yes, I think this negotiation should happen on per methods basis.
>>>
>>> However, I am not pulling in Appendix F. It is an optional to use format
>>> for parameter value pairs that can be used in these methods, it is not
>>> required. Nor, does the specification define any parameters that are
>>> transferred using this interface. The things that are in the appendix
>>> are not core protocol, they represent either informational things, like
>>> the examples and the state machine. The other appendices are normative
>>> definitions of particular choices for things that can be combined or
>>> used with RTSP, like RTP as media transport.
>>>
>> OK, it is possible to use GET_PARAM for basic requirements without using
>> message bodies, so you can get away without defining a lowest common
>> denominator format. However, the use of message bodies for SET_PARAM is
>> RECOMMENDED so it seems a little odd not to ensure that server and
>> client will have at least one format in common. (And its not as if we
>> are dealing with a hugely complicated bit of spec for
>> text/parameters! :-) ).  Then, given the example in GET_PARAMETER it
>> seems sensible to advise implementing text/parameters as the default for
>> GET_PARAM also.
> 
> Sure, I personally don't have any issue with making it at least SHOULD
> implement text/parameters. But from my horizon a forward pointer with
> the normative requirements is sufficient to accomplish that.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From dwing@cisco.com  Wed Aug 28 10:41:20 2013
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
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On Aug 22, 2013, at 2:37 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi Dan,
>=20
> Regarding the 3GPP SA3 study, the information I gave was a little =
misleading. =46rom a security perspective SRTP would be fine.
>=20
> The focus of the SA3 study was on how to provide security for the =
existing fax transmission mechanism, which uses UDPTL/UDP.
>=20
> 3GPP already mandates IMS terminals to support UDPTL/UDP for unsecure =
fax. And, new terminals (supporting secure fax) are still required to =
also support unsecure fax, in order to communicate with legacy terminals =
when unsecure fax is sufficient.=20
>=20
> So, using UDPTL/DTLS/UDP for secure fax makes more sense, as it avoids =
implementing different mechanisms for secure and unsecure fax - =
UDPTL/DTLS/UDP only requires a new layer between UDPTL and UDP, it does =
not require changing the upper layers (UDPTL and above).
>=20
> Hopefully this clarifies :)

(Sorry for my delay.  I was on vacation.)

Thanks for the clarification.


Back to your document, it says:

   Since the DTLS record layer "application_data" packet does not
   indicate whether it carries UDPTL, or some other protocol, the usage
   of a dedicated DTLS association for transport of UDPTL needs to be
   negotiated, e.g. using the Session Description Protocol (SDP)
   [RFC4566] and the SDP offer/answer mechanism [RFC3264].

   Therefore, this document specifies a new <proto> value [RFC4566] for
   the SDP "m=3D" line [RFC3264], in order to indicate UDPTL over DTLS =
in
   SDP messages [RFC4566].

Have you considered doing UDPTL negotiation in the DTLS handshake =
itself, using =
http://tools.ietf.org/html/draft-ietf-tls-applayerprotoneg?  That seems =
ideally suited for indicating the application layer protocol, perhaps in =
addition to the SDP signaling described in your I-D.

-d


>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]=20
> Sent: 19. elokuuta 2013 20:43
> To: Christer Holmberg
> Cc: mmusic; mmusic-chairs@tools.ietf.org
> Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
>=20
>=20
> On Aug 19, 2013, at 6:03 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>> Hi,
>>=20
>> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, =
which defines usage of UDPTL over DTLS, in order to provide secure fax.
>>=20
>> The draft was previously submitted to DISPATCH. Based on discussions =
with the ADs and chairs, it was decided that it shall be submitted to =
MMUSIC (note that no DTLS extensions are needed).
>>=20
>> As is indicated in the draft, 3GPP has performed a study on how to=20
>> provide secure fax in the IMS, and the outcome was that secure fax =
shall be transported using UDPTL over DTLS.
>=20
> Got a pointer to that study?  Seems easier to carry UDPTL over RTP, =
which would allow the RTP to be secured using SRTP (and thus the UDPTL =
would be secured using SRTP).  There is a spec floating around to do =
exactly that (carry fax over RTP so that SRTP can secure it).  Advantage =
of using SRTP to secure fax is it separates the keying mechanism from =
security, so that Security Descriptions / MIKEY / DTLS-SRTP / =
whatever-is-invented-in-2020 will work just as effectively for voice as =
for fax.  And also that upgrading from a voice call to a "fax" call has =
no additional complexities due to security ("please press START to begin =
the fax transmission").
>=20
> -d
>=20
>=20
>> However, there is nothing "3GPP/IMS specific" about the mechanism, as =
UDPTL is commonly used for fax also elsewhere.
>>=20
>> Regards,
>>=20
>> Christer
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>=20


From christer.holmberg@ericsson.com  Thu Aug 29 01:17:38 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dan Wing <dwing@cisco.com>
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Hi Dan,

The protocols listed in draft-ietf-tls-applayerprotoneg-01 are used without=
 SIP/SDP, so using such mechanism makes sense for those protocols.

In SIP, SDP is used to negotiate the media (including nonsecure fax), and I=
 see no reason why we should introduce a new mechanism for secure fax.=20

Regards,

Christer


-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: 28. elokuuta 2013 20:41
To: Christer Holmberg
Cc: mmusic WG
Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00


On Aug 22, 2013, at 2:37 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi Dan,
>=20
> Regarding the 3GPP SA3 study, the information I gave was a little mislead=
ing. From a security perspective SRTP would be fine.
>=20
> The focus of the SA3 study was on how to provide security for the existin=
g fax transmission mechanism, which uses UDPTL/UDP.
>=20
> 3GPP already mandates IMS terminals to support UDPTL/UDP for unsecure fax=
. And, new terminals (supporting secure fax) are still required to also sup=
port unsecure fax, in order to communicate with legacy terminals when unsec=
ure fax is sufficient.=20
>=20
> So, using UDPTL/DTLS/UDP for secure fax makes more sense, as it avoids im=
plementing different mechanisms for secure and unsecure fax - UDPTL/DTLS/UD=
P only requires a new layer between UDPTL and UDP, it does not require chan=
ging the upper layers (UDPTL and above).
>=20
> Hopefully this clarifies :)

(Sorry for my delay.  I was on vacation.)

Thanks for the clarification.


Back to your document, it says:

   Since the DTLS record layer "application_data" packet does not
   indicate whether it carries UDPTL, or some other protocol, the usage
   of a dedicated DTLS association for transport of UDPTL needs to be
   negotiated, e.g. using the Session Description Protocol (SDP)
   [RFC4566] and the SDP offer/answer mechanism [RFC3264].

   Therefore, this document specifies a new <proto> value [RFC4566] for
   the SDP "m=3D" line [RFC3264], in order to indicate UDPTL over DTLS in
   SDP messages [RFC4566].

Have you considered doing UDPTL negotiation in the DTLS handshake itself, u=
sing http://tools.ietf.org/html/draft-ietf-tls-applayerprotoneg?  That seem=
s ideally suited for indicating the application layer protocol, perhaps in =
addition to the SDP signaling described in your I-D.

-d


>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: 19. elokuuta 2013 20:43
> To: Christer Holmberg
> Cc: mmusic; mmusic-chairs@tools.ietf.org
> Subject: Re: [MMUSIC] Draft new: draft-holmberg-mmusic-udptl-dtls-00
>=20
>=20
> On Aug 19, 2013, at 6:03 AM, Christer Holmberg <christer.holmberg@ericsso=
n.com> wrote:
>=20
>> Hi,
>>=20
>> We have submitted a new draft, draft-holmberg-mmusic-udptl-dtls-00, whic=
h defines usage of UDPTL over DTLS, in order to provide secure fax.
>>=20
>> The draft was previously submitted to DISPATCH. Based on discussions wit=
h the ADs and chairs, it was decided that it shall be submitted to MMUSIC (=
note that no DTLS extensions are needed).
>>=20
>> As is indicated in the draft, 3GPP has performed a study on how to=20
>> provide secure fax in the IMS, and the outcome was that secure fax shall=
 be transported using UDPTL over DTLS.
>=20
> Got a pointer to that study?  Seems easier to carry UDPTL over RTP, which=
 would allow the RTP to be secured using SRTP (and thus the UDPTL would be =
secured using SRTP).  There is a spec floating around to do exactly that (c=
arry fax over RTP so that SRTP can secure it).  Advantage of using SRTP to =
secure fax is it separates the keying mechanism from security, so that Secu=
rity Descriptions / MIKEY / DTLS-SRTP / whatever-is-invented-in-2020 will w=
ork just as effectively for voice as for fax.  And also that upgrading from=
 a voice call to a "fax" call has no additional complexities due to securit=
y ("please press START to begin the fax transmission").
>=20
> -d
>=20
>=20
>> However, there is nothing "3GPP/IMS specific" about the mechanism, as UD=
PTL is commonly used for fax also elsewhere.
>>=20
>> Regards,
>>=20
>> Christer
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>=20


From christer.holmberg@ericsson.com  Fri Aug 30 05:44:21 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Hi,

We also need to decide whether that change also applies to when new media d=
escriptions (m- line) are added to a previously negotiated BUNDLE group.

Currently the text says that, when an m- line is added, it must contain eit=
her a unique address, or a previously negotiated BUNDLE address.

But, since we now allow to assign a new, yet not confirmed, BUNDLE address =
to multiple m- lines, I guess that applies also to a new m- line.

If so, we can make the text in Section 6.4.3, because it more or less uses =
the procedures defined in Section 6.4.1.

Regards,

Christer

------------------------------------------


OLD TEXT (-04):
---------------------


6.4.3.  Adding a media description to a BUNDLE group

   When an Offerer adds an "m=3D" line to a BUNDLE group, the Offerer MUST
   assign either a unique address, or the BUNDLE address associated with
   the BUNDLE group, to the added "m=3D" line.  In addition, the Offerer
   MUST assign a mid value to the "m=3D" line, and place the mid in the
   SDP group:BUNDLE attribute mid list associated with the BUNDLE group,
   in order to group the "m=3D" line to the BUNDLE group.

   NOTE: If the Offerer assigns a unique address to the added "m=3D" line,
   it allows the Answerer to move the "m=3D" line out of the BUNDLE group
   without having to reject the "m=3D" line ([ref-to-be-added]).

   To the previously added "m=3D" lines in the BUNDLE group, the Offerer
   assigns either unique addresses or the BUNDLE address associated with
   the BUNDLE group, according to the procedures in Section 6.4.1.

   Section 10.3 shows an example of an SDP Offer used to add an "m=3D"
   line to a BUNDLE group.


NEW SUGGESTED TEXT (-04):
--------------------------------------


6.4.3.  Adding a media description to a BUNDLE group

   When an Offerer adds a new "m=3D" line to a previously negotiated BUNDLE=
 group, the
   Offerer assigns an address to the new "m=3D" line according to the proce=
dures in Section=20
   6.4.1. If the Offerer assigns a BUNDLE address (previously negotiated, o=
r a suggested=20
   new Address) to the previously negotiated "m=3D" lines in the BUNDLE gro=
up, the Offerer=20
   can assign the same address to the new "m=3D" line. Alternatively, the O=
fferer can assign=20
   a unique address to the new "m=3D" line.

   NOTE: If the Offerer assigns a unique address to the added "m=3D" line,
   it allows the Answerer to move the "m=3D" line out of the BUNDLE group
   without having to reject the "m=3D" line ([ref-to-be-added]).

   Section 10.3 shows an example of an SDP Offer used to add an "m=3D"
   line to a BUNDLE group.

















-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Paul Kyzivat
Sent: 27. elokuuta 2013 16:43
To: mmusic@ietf.org
Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-n=
egotiated, the Offerer can assign the requsted new address to all m- lines =
immediately, without first offering unique addresses.

This seems reasonable to me.

	Thanks,
	Paul

On 8/23/13 6:39 AM, Christer Holmberg wrote:
> Hi,
>
> One of the open issues related to BUNDLE presented in Berlin (Q5 in my
> slides) was whether, once a BUNDLE address is re-negotiated (i.e. the=20
> usage of BUNDLE etc has already been negotiated), it is allowed to=20
> assign the new suggested BUNDLE address to all "m=3D" lines immediately,=
=20
> instead of doing the 2-step Offer/Answer procedure. The outcome was=20
> that it shall be allowed.
>
> So, based on that, I am suggesting the change below to section 6.1.4=20
> of BUNDLE.
>
> NOTE that this only applies to the case where a BUNDLE address is=20
> RE-negotiated (e.g. in a re-INVITE). There is still a discussion about=20
> the initial Offer, and that outcome of that (and the outcome of the UP
> discussions) may obviously also impact the text. But, let's keep that=20
> separate for now, and only focus on the case when a BUNDLE address is=20
> RE-negotiated :)
>
> Regards,
>
> Christer
>
> -------------------------------------------
>
> OLD TEXT (BUNDLE-04):
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
> 6.4.1.  SDP Offerer Bundle Address Request and Usage
>
>     An Offerer can assign a BUNDLE address to multiple "m=3D" lines in a
>
>     BUNDLE group, once the Answerer has selected the BUNDLE address=20
> for
>
>     the Offerer.  An Offerer MUST NOT assign a BUNDLE address to=20
> multiple
>
>     "m=3D" lines until the Answerer has selected the BUNDLE address.
>
>     OPEN ISSUE: Should it be allowed to assign a new BUNDLE address to
>
>     multiple "m=3D" lines in a BUNDLE group, before the Answerer has
>
>     selected the BUNDLE address?
>
>     In order to negotiate (or, to re-negotiate) the BUNDLE address
>
>     associated with a BUNDLE group, the Offerer, in the SDP Offer,
>
>     assigns a unique address to each "m=3D" line in the BUNDLE group. =20
> In
>
>     addition, the Offerer indicates which unique address it wishes the
>
>     Answerer to select as the Offerer's BUNDLE address.  The Offerer
>
>     places the mid, associated with the unique address requested to be
>
>     selected as the BUNDLE address, first in the SDP group:BUNDLE
>
>     attribute mid list.  The Answerer will then select the BUNDLE=20
> address
>
>     for the Offerer ([ref-to-be-added]).
>
>     If the Offerer, in a subsequent SDP Offer, wants to re-negotiate=20
> the
>
>     BUNDLE address associated with a BUNDLE group, it MAY assign the
>
>     previously negotiated BUNDLE address as a unique address to one of
>
>     the "m=3D" lines in the BUNDLE group.
>
>     If the Offerer assigns the previously selected BUNDLE address to=20
> more
>
>     than one "m=3D" line in a BUNDLE group, the first mid in the SDP
>
>     group:BUNDLE attribute mid list MUST represent an "m=3D" line to=20
> which
>
>     the BUNDLE address is assigned.  Hence, in order to re-negotiate=20
> the
>
>     BUNDLE address, the Offerer needs to assign a unique address to=20
> each
>
>     "m=3D" line in the BUNDLE group, as described above.
>
>     An Offerer MUST NOT assign a BUNDLE address to an "m=3D" line that=20
> is
>
>     not associated with a BUNDLE group.  An Offerer MUST NOT assign a
>
>     BUNDLE address, that has been negotiated for a BUNDLE group, to an
>
>     "m=3D" line that is associated with another BUNDLE group.
>
>     Section 10.1 shows an example of a Bundle Address Request.
>
> SUGGESTED NEW TEXT:
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
> 6.4.1.  SDP Offerer Bundle Address Request and Usage
>
>
>
>     An Offerer can assign a BUNDLE address to multiple "m=3D" lines in a
>
>     BUNDLE group, once the Answerer has selected the BUNDLE address=20
> for
>
>     the Offerer.  An Offerer MUST NOT assign a BUNDLE address to=20
> multiple
>
>     "m=3D" lines until the Answerer has selected the BUNDLE address.
>
>
>
>     Before the Answerer has selected a BUNDLE address for the Offerer,
>
>     in order to negotiate the BUNDLE address associated with a BUNDLE
>
>     group, the Offerer, in the SDP Offer, assigns a unique address to
>
>     each "m=3D" line in the BUNDLE group. In addition, the Offerer=20
> indicates
>
>     which unique address it wishes the Answerer to select as the
>
>     Offerer's BUNDLE address. The Offerer places the mid, associated
>
>     with the unique address requested to be selected as the BUNDLE=20
> address,
>
>     first in the SDP group:BUNDLE attribute mid list.  The Answerer=20
> will
>
>     then select the BUNDLE address for the Offerer ([ref-to-be-added]).
>
>
>
>     If an Offerer, in a subsequent SDP Offer, wants to re-negotiate=20
> the
>
>     BUNDLE address associated with a BUNDLE group, it can either:
>
>
>
> -     Assign the requested new BUNDLE address to all "m=3D" lines in the
>
> BUNDLE group; or
>
> -     In the same way as described in the procedures above, assign a
>
> unique address to each "m=3D" line in the BUNDLE group (the previously
>
> negotiated BUNDLE address can also be assigned as a unique address
>
> to one of the "m=3D" lines in the BUNDLE group).
>
>
>
>     In both cases, the Offerer places the mid, associated with the=20
> unique
>
>     address requested to be selected as the BUNDLE address, first in=20
> the
>
>     SDP group:BUNDLE attribute mid list (if the requested new BUNDLE=20
> address
>
>     is assigned to all "m=3D" lines in the BUNDLE group, all mids in the=
=20
> SDP
>
>     group:BUNDLE attribute mid list will represent that same address).
>
>
>
>     An Offerer MUST NOT assign a BUNDLE address to an "m=3D" line that=20
> is
>
>     not associated with a BUNDLE group.  An Offerer MUST NOT assign a
>
>     BUNDLE address, that has been negotiated for a BUNDLE group, to an
>
>     "m=3D" line that is associated with another BUNDLE group.
>
>
>
>     Section 10.1 shows an example of a Bundle Address Request.
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

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Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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On Aug 30, 2013, at 8:43 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> We also need to decide whether that change also applies to when new =
media descriptions (m- line) are added to a previously negotiated BUNDLE =
group.
> Currently the text says that, when an m- line is added, it must =
contain either a unique address, or a previously negotiated BUNDLE =
address.
> But, since we now allow to assign a new, yet not confirmed, BUNDLE =
address to multiple m- lines, I guess that applies also to a new m- =
line.

But then if the far-end answered with accepting it for the same bundle, =
then you'd have to send another offer with the m-line using the correct =
port of the bundle group, right?

-hadriel



From christer.holmberg@ericsson.com  Fri Aug 30 06:39:19 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Date: Fri, 30 Aug 2013 13:39:02 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C47762C@ESESSMB209.ericsson.se> <521CACD4.6060904@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B1C4809D7@ESESSMB209.ericsson.se> <968A751E-C145-4799-9479-417D850F50A6@oracle.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Hi,

>> We also need to decide whether that change also applies to when new medi=
a descriptions (m- line) are added to a previously negotiated BUNDLE group.
>> Currently the text says that, when an m- line is added, it must contain =
either a unique address, or a previously negotiated BUNDLE address.
>> But, since we now allow to assign a new, yet not confirmed, BUNDLE addre=
ss to multiple m- lines, I guess that applies also to a new m- line.
>
> But then if the far-end answered with accepting it for the same bundle, t=
hen you'd have to send another offer with the m-line using the correct port=
 of the bundle group, right?

The change is that an Offerer is now allowed to assign the same address (th=
e requested BUNDLE address) to all m- lines in the BUNDLE group BEFORE the =
Answerer has "confirmed" the address=20
to become the Offerer's BUNDLE address.

So, if the requested BUNDLE address was assigned to all m- lines in the BUN=
DLE group, and the Answerer accepts it, there is obviously no need to send =
a second offer.

But, if any of the m- lines in the offer did NOT contain the selected BUNDL=
E address, a second Offer is still required :)

Regards,

Christer



From christer.holmberg@ericsson.com  Fri Aug 30 06:48:23 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Date: Fri, 30 Aug 2013 13:48:13 +0000
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Subject: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Hi,

I started to draft some text for the definition of the SDP 'bundle-only' at=
tribute (my assumption is that it will be defined in draft-bundle).

Note that, at this point, I don't suggest any normative text regarding the =
usage of port zero in the SDP Offer, as we are still discussing that (the o=
utcome of that discussion may affect at least the last paragraph in section=
 x.2.1.).

Also note that I don't say anything about in which "scope/context" (webrtc =
etc) the usage of the attribute is allowed.

So, leaving the usage of port zero, and the scope question, aside for the m=
oment, please take a look and indicate whether you have any issues (at this=
 point, I ask you to focus on the technical stuff - and stuff which is uncl=
ear) with the text :)

Regards,

Christer


--------------------------


X.  SDP 'bundle-only' attribute

X.1.  General

This section defines a new SDP media-level attribute, 'bundle-only'. The at=
tribute
is used, in an SDP Offer, to indicate that an "m=3D" line, to which the att=
ribute
has been assigned, is only to be accepted by the Answerer if the Answerer k=
eeps
the "m=3D" line within the BUNDLE group to which the "m=3D" line is associa=
ted in the
SDP Offer.


X.2.  SDP Offer/Answer Procedures

X.2.1.  SDP Offer Procedures

When an Offerer sends an SDP Offer, which contains a BUNDLE group, it can a=
ssign the
SDP 'bundle-only' attribute to an "m=3D" line associated with the BUNDLE gr=
oup,
in order to request that the Answerer only accepts the "m=3D" line if is ke=
pt within
the BUNDLE group to which the "m=3D" line is associated in the SDP Offer.

An Offerer MUST NOT assign the SDP 'bundle-only' attribute to an "m=3D" lin=
e that is not
associated with a BUNDLE group.

If an Offerer receives an SDP Answer where an "m=3D" line, to which it assi=
gned an
'bundle-only' attribute in the associated Offer, has been removed from a BU=
NDLE group, but
has not been rejected ([ref-to-section- 6.5.2.]), the Offerer MUST send a n=
ew Offer, in which
it assigns a zero port value to the "m=3D" line.


X.2.2.  SDP Answerer Procedures

When an Answerer receives an SDP Offer ([ref-to-section- 6.5.]), the Answer=
er MAY choose
to remove an "m=3D" line out of a BUNDLE group ([ref-to-section- 6.5.2.]). =
If an SDP
'bundle-only' attribute has been assigned to such "m=3D" line, the Answerer=
 MUST reject
the "m=3D" line, according to the procedures in Section 6.5.3.

An Answerer MUST NOT insert an SDP 'bundle-only' attribute in an SDP Answer=
.





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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">I started to draft some text for the definition of the SDP=
 'bundle-only' attribute (my assumption is that it will be defined in draft=
-bundle).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Note that, at this point, I don&#8217;t suggest any normat=
ive text regarding the usage of port zero in the SDP Offer, as we are still=
 discussing that (the outcome of that discussion may
 affect at least the last paragraph in section x.2.1.).<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Also note that I don&#8217;t say anything about in which &=
#8220;scope/context&#8221; (webrtc etc) the usage of the attribute is allow=
ed.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">So, leaving the usage of port zero, and the scope question=
, aside for the moment, please take a look and indicate whether you have an=
y issues (at this point, I ask you to focus on
 the technical stuff &#8211; and stuff which is unclear) with the text :)<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Christer<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">--------------------------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">X.&nbsp; SDP 'bundle-only' attribute<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">X.1.&nbsp; General<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">This section defines a new SDP media-level attribute, 'bun=
dle-only'. The attribute<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">is used, in an SDP Offer, to indicate that an &quot;m=3D&q=
uot; line, to which the attribute<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">has been assigned, is only to be accepted by the Answerer =
if the Answerer keeps
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">the &quot;m=3D&quot; line within the BUNDLE group to which=
 the &quot;m=3D&quot; line is associated in the
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">SDP Offer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">X.2.&nbsp; SDP Offer/Answer Procedures<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">X.2.1.&nbsp; SDP Offer Procedures<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">When an Offerer sends an SDP Offer, which contains a BUNDL=
E group, it can assign the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">SDP 'bundle-only' attribute to an &quot;m=3D&quot; line as=
sociated with the BUNDLE group,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">in order to request that the Answerer only accepts the &qu=
ot;m=3D&quot; line if is kept within
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">the BUNDLE group to which the &quot;m=3D&quot; line is ass=
ociated in the SDP Offer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">An Offerer MUST NOT assign the SDP 'bundle-only' attribute=
 to an &quot;m=3D&quot; line that is not<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">associated with a BUNDLE group.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">If an Offerer receives an SDP Answer where an &quot;m=3D&q=
uot; line, to which it assigned an<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">'bundle-only' attribute in the associated Offer, has been =
removed from a BUNDLE group, but<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">has not been rejected ([ref-to-section-</span>
<span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;">6.5.2.=
]), the Offerer MUST send a new Offer, in which<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">it assigns a zero port value to the &quot;m=3D&quot; line.=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">X.2.2.&nbsp; SDP Answerer Procedures<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">When an Answerer receives an SDP Offer ([ref-to-section-</=
span>
<span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;">6.5.])=
, the Answerer MAY choose
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">to remove an &quot;m=3D&quot; line out of a BUNDLE group (=
[ref-to-section-</span>
<span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;">6.5.2.=
]). If an SDP <o:p>
</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">'bundle-only' attribute has been assigned to such &quot;m=
=3D&quot; line, the Answerer MUST reject
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">the &quot;m=3D&quot; line, according to the procedures in =
Section 6.5.3.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">An Answerer MUST NOT insert an SDP 'bundle-only' attribute=
 in an SDP Answer.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C480AB8ESESSMB209erics_--

From magnus.westerlund@ericsson.com  Fri Aug 30 08:37:06 2013
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Date: Fri, 30 Aug 2013 17:37:30 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP 2.0: Client Authentication
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WG,

As part of Robert's comments is this minor comment:

>
>> As written, a client and server can use HTTP Basic authentication over
>> TCP that is not protected with TLS. Consider restricting it's use to TLS
>> only. Are you sure you don't need to say anything RTSP specific about
>> DIGEST computations (I don't think you need to go as far as SIP did
>> talking about DIGEST, but I'm surprised you haven't run into issues
>> relying only on 2617 literally.
> Are you referring to Section 19.1 here?
Yes
>
> Yes, I would not have any issue with requiring Basic to be combined with
> TLS. I think that is appropriate.
>
> When it comes to DIGEST, are you referring to section 22.4 in RFC 3261?
Yes
>
> That section indicates that some clarifications are likely needed.

This actually resulted in finding a bit more shortcomings around the
client authentication. The high level summary of what my proposal for
fixing this is:

- Mandate TLS for basic authentication.
- Added the missing Authentication-Info and Proxy-Authentication-Info
- Corrected ABNF that was missing basic
- Written an equivalent of the RFC 2617 clarification section of
  RFC 3261.
- Tried to press on the hop-by-hop nature of proxy-authorization
- Put up a warning flag for server to client requests lack authentication.

This massive amount obviously needs review.

The resulting relevant text sections are:

18 Header tables:

   +------------------+-------+-----+----+-----+-----+-----+-----+-----+
   | Header           | Where | Pro | DE | OPT | STP | PLY | PSE | TRD |
   |                  |       | xy  | S  |     |     |     |     |     |
   +------------------+-------+-----+----+-----+-----+-----+-----+-----+

   | Authentication-I | r     |     | o  | o   | o   | o   | o   | o/- |
   | nfo              |       |     |    |     |     |     |     |     |

   | Authorization    | R     |     | o  | o   | o   | o   | o   | o   |


   | Proxy-           | 407     | amr | m  | m  | m  | m   | m   | m   |
   | Authenticate     |         |     |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy-Authentica | r       | amd | o  | o  | o  | o   | o   | o/- |
   | tion-Info        |         | r   |    |    |    |     |     |     |
   |                  |         |     |    |    |    |     |     |     |
   | Proxy-           | R       | rd  | o  | o  | o  | o   | o   | o   |
   | Authorization    |         |     |    |    |    |     |     |     |

   |                  |         |     |    |    |    |     |     |     |
   | WWW-             | 401     |     | m  | m  | m  | m   | m   | m   |
   | Authenticate     |         |     |    |    |    |     |     |     |
   +------------------+---------+-----+----+----+----+-----+-----+-----+

   +-------------------------+---------+-------+-----+-----+-----+-----+
   | Header                  | Where   | Proxy | GPR | SPR | RDR | PNY |
   +-------------------------+---------+-------+-----+-----+-----+-----+
   | Authentication-Info     | r       |       | o/- | o/- | -   | -   |
   |                         |         |       |     |     |     |     |
   | Authorization           | R       |       | o   | o   | o   | -   |
   |                         |         |       |     |     |     |     |
   | Proxy-Authenticate      | 407     | amdr  | m   | m   | m   | -   |
   |                         |         |       |     |     |     |     |
   | Proxy-Authentication-In | r       | amdr  | o/- | o/- | -   | -   |
   | fo                      |         |       |     |     |     |     |
   |                         |         |       |     |     |     |     |
   | Proxy-Authorization     | R       | amdr  | o   | o   | o   | -   |


      +------------------+---------+-------+-----+-----+-----+-----+
      | Header           | Where   | Proxy | GPR | SPR | RDR | PNY |
      +------------------+---------+-------+-----+-----+-----+-----+

      |                  |         |       |     |     |     |     |
      | WWW-Authenticate | 401     |       | m   | m   | m   | -   |
      +------------------+---------+-------+-----+-----+-----+-----+


18.7.  Authentication-Info

   The Authentication-Info header is used by the server to communicate
   some information regarding the successful authentication in the
   response message.  This usage of this header is specified in
   [RFC2617] with some RTSP clarification in Section 19.1.  This header
   MUST only be used in response messages related to client to server
   requests.

18.8.  Authorization

   An RTSP client that wishes to authenticate itself with a server using
   authentication mechanism from HTTP [RFC2617] , usually, but not
   necessarily, after receiving a 401 response, does so by including an
   Authorization request-header field with the request.  The
   Authorization field value consists of credentials containing the
   authentication information of the user agent for the realm of the
   resource being requested.  This header MUST only be used in client to
   server requests.

   If a request is authenticated and a realm specified, the same
   credentials SHOULD be valid for all other requests within this realm
   (assuming that the authentication scheme itself does not require
   otherwise, such as credentials that vary according to a challenge
   value or using synchronized clocks).

   When a shared cache (see Section 16) receives a request containing an
   Authorization field, it MUST NOT return the corresponding response as
   a reply to any other request, unless one of the following specific
   exceptions holds:

   1.  If the response includes the "max-age" cache-control directive,
       the cache MAY use that response in replying to a subsequent
       request.  But (if the specified maximum age has passed) a proxy
       cache MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.  (This is the defined behavior
       for max-age.)  If the response includes "max-age=0", the proxy
       MUST always revalidate it before re-using it.

   2.  If the response includes the "must-revalidate" cache-control
       directive, the cache MAY use that response in replying to a
       subsequent request.  But if the response is stale, all caches
       MUST first revalidate it with the origin server, using the
       request-headers from the new request to allow the origin server
       to authenticate the new request.

   3.  If the response includes the "public" cache-control directive, it
       MAY be returned in reply to any subsequent request.


18.34.  Proxy-Authenticate

   The Proxy-Authenticate response-header field MUST be included as part
   of a 407 (Proxy Authentication Required) response.  The field value
   consists of a challenge that indicates the authentication scheme and
   parameters applicable to the proxy for this Request-URI.

   The HTTP access authentication process is described in [RFC2617].
   Unlike WWW-Authenticate, the Proxy-Authenticate header field applies
   only to the current connection and SHOULD NOT be passed on to
   downstream agents.  This header MUST only be used in response
   messages related to client to server requests.

18.35.  Proxy-Authentication-Info

   The Proxy-Authentication-Info header is used by the proxy to
   communicate some information regarding the successful authentication
   to the proxy in the message response.  The content and usage of this
   header is described in the HTTP access authentication [RFC2617] that
   is also used by RTSP and clarified in Section 19.1.  This header MUST
   only be used in response messages related to client to server
   requests.  This header has hop by hop scope.

18.36.  Proxy-Authorization

   The Proxy-Authorization request-header field allows the client to
   identify itself (or its user) to a proxy which requires
   authentication.  The Proxy-Authorization field value consists of
   credentials containing the authentication information of the user
   agent for the proxy and/or realm of the resource being requested.

   The HTTP access authentication process is described in [RFC2617].
   Unlike Authorization, the Proxy-Authorization header field applies
   only to the next hop proxy.  This header MUST only be used in client
   to server requests.

18.58.  WWW-Authenticate

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages.  The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the Request-URI.  This header MUST only be
   used in response messages related to client to server requests.

   The HTTP access authentication process is described in [RFC2617] with
   some clarification in Section 19.1.  User agents are advised to take
   special care in parsing the WWW- Authenticate field value as it might
   contain more than one challenge, or if more than one WWW-Authenticate
   header field is provided, the contents of a challenge itself can
   contain a comma-separated list of authentication parameters.



19.1.  RTSP and HTTP Authentication

   RTSP and HTTP share common authentication schemes, and thus follow
   the same usage guidelines as specified in [RFC2617] with the
   additions for digest specified below in Section 19.1.1.  Servers
   SHOULD implement both basic and digest [RFC2617] authentication.
   Clients MUST implement both basic and digest authentication [RFC2617]
   so that a server that requires the client to authenticate can trust
   that the capability is present.

   It should be stressed that using the HTTP authentication alone does
   not provide full control message security.  Therefore, in
   environments requiring tighter security for the control messages, TLS
   SHOULD be used, see Section 19.2.  Any RTSP message containing an
   Authorization header containg a basic authorization MUST be using a
   TLS connection with confidentiality protection enabled, i.e. no NULL
   encryption.

   In cases where there is a chain of proxies between the client and the
   server, each proxy may individually request the client or previous
   proxy to authenticate itself.  This is done using the Proxy-
   Authenticate (Section 18.34), the Proxy-Authorization (Section 18.36)
   and the Proxy-Authentication-Info (Section 18.35) headers.  These
   headers are hop-by-hop headers and have only scope for the current
   connection.  Thus if a chain exist, the proxy connecting to another
   proxy will have to act as a client to authorize itself towards the
   next proxy.  The WWW-Authenticate (Section 18.58), Authorization
   (Section 18.8) and Authentication-Info (Section 18.7) headers are
   end-to-end and must not be modified by proxies.

   This authentication mechanism works only for client to server
   requests as currently defined.  This leaves server to client request
   outside of the context of TLS based communication more vulnerable to
   message injection attacks on the client.  Based on the server to
   client methods that exist, the potential risks are various; hijacking
   (REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY) or attacks
   with uncertain results (SET_PARAMETER).


19.1.1.  Digest Authentication

   This section describes the modifications and clarifications required
   to apply the HTTP Digest authentication scheme to RTSP.  The RTSP
   scheme usage is almost completely identical to that for HTTP
   [RFC2617].  These are based on the procedures defined for SIP 2.0
   [RFC3261].

   The rules for Digest authentication follow those defined in
   [RFC2617], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the
   following differences:

   1.  Use the ABNF specified in this document, rather than the one in
       [RFC2617].  That way the following is assured:

       *  Using the right RTSP URIs allowed in the challenge as well as
          in the digest.

       *  Resolved the error in the "uri" parameter of the Authorization
          header in [RFC2617].

   2.  If MTags are used then the example procedure for choosing a nonce
       based on Etag can work based on replacing ETag with the MTag.

   3.  As a clarification to the calculation of the A2 value for message
       integrity assurance in the Digest authentication scheme,
       implementers should assume, when the entity-body is empty (that
       is, when the RTSP messages have no message body) that the hash of
       the message-body resolves to the MD5 hash of an empty string, or:
       H(entity-body) = MD5("") = "d41d8cd98f00b204e9800998ecf8427e".

   4.  RFC 2617 notes that a cnonce value MUST NOT be sent in an
       Authorization (and by extension Proxy-Authorization) header field
       if no qop directive has been sent.  Therefore, any algorithms
       that have a dependency on the cnonce (including "MD5-Sess")
       require that the qop directive be sent.  Use of the "qop"
       parameter is optional in RFC 2617 for the purposes of backwards
       compatibility with RFC 2069; since this specification defines
       RTSP 2.0 there is no backwards compatibilit issue with mandating.
       Thus, all RTSP agents MUST implement qop-options.



   Authentication-Info = "Authentication-Info" HCOLON auth-info
   auth-info         = auth-info-entry *(COMMA auth-info-entry)
   auth-info-entry   = nextnonce
                     / message-qop
                     / response-auth
                     / cnonce
                     / nonce-count
   nextnonce         = "nextnonce" "=" nonce-value
   response-auth     = "rspauth" "=" response-digest
   response-digest   = DQ *LHEX DQ
   Authorization     =  "Authorization" HCOLON credentials
   credentials       =  basic-credential
                     /  digest-credential
                     /  other-response
   basic-credential  = "Basic" LWS basic-credentials
   basic-credentials = base64 ; Base64 encoding of user-password
   user-password     = basic-username ":" password
   basic-username    = *CF-TEXT
   CF-TEXT           = %x20-39 / %x3B-7E / %x80-FF ; TEXT without :
   password          = *TEXT
   digest-credential = ("Digest" LWS digest-response)
   digest-response   =  dig-resp *(COMMA dig-resp)
   dig-resp          =  username / realm / nonce / digest-uri
                     /  dresponse / algorithm / cnonce
                     /  opaque / message-qop
                     /  nonce-count / auth-param
   username          =  "username" EQUAL username-value
   username-value    =  quoted-string
   digest-uri        =  "uri" EQUAL LDQUOT digest-uri-value RDQUOT
   digest-uri-value  =  RTSP-REQ-URI
   message-qop       =  "qop" EQUAL qop-value
   cnonce            =  "cnonce" EQUAL cnonce-value
   cnonce-value      =  nonce-value
   nonce-count       =  "nc" EQUAL nc-value
   nc-value          =  8LHEX
   dresponse         =  "response" EQUAL request-digest
   request-digest    =  LDQUOT 32LHEX RDQUOT
   auth-param        =  auth-param-name EQUAL
                        ( token / quoted-string )
   auth-param-name   =  token
   other-response    =  auth-scheme LWS auth-param
                        *(COMMA auth-param)
   auth-scheme       =  token

   Proxy-Authenticate =  "Proxy-Authenticate" HCOLON challenge-list
   challenge-list     =  challenge *(COMMA challenge)
   challenge          =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                      /  ("Basic" LWS realm)
                      /  other-challenge
   other-challenge    =  auth-scheme LWS auth-param
                         *(COMMA auth-param)
   digest-cln         =  realm / domain / nonce
                      /  opaque / stale / algorithm
                      /  qop-options / auth-param
   realm              =  "realm" EQUAL realm-value
   realm-value        =  quoted-string
   domain             =  "domain" EQUAL LDQUOT RTSP-REQ-Ref
                         *(1*SP RTSP-REQ-Ref ) RDQUOT
   nonce              =  "nonce" EQUAL nonce-value
   nonce-value        =  quoted-string
   opaque             =  "opaque" EQUAL quoted-string
   stale              =  "stale" EQUAL ( "true" / "false" )
   algorithm          =  "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
   qop-options        =  "qop" EQUAL LDQUOT qop-value
                         *("," qop-value) RDQUOT
   qop-value          =  "auth" / "auth-int" / token

   Proxy-Auth-Info    = "Proxy-Authentication-Info" HCOLON auth-info
   Proxy-Authorization = "Proxy-Authorization" HCOLON credentials

   WWW-Authenticate  = "WWW-Authenticate" HCOLON challenge-list
-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From martin.thomson@gmail.com  Fri Aug 30 09:29:59 2013
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From: Martin Thomson <martin.thomson@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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I think that you need to include in X.2.2 what the answerer puts in
port for the "m=3D" line if it intends to accept a bundle-only line.

On 30 August 2013 06:48, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>
>
> Hi,
>
>
>
> I started to draft some text for the definition of the SDP 'bundle-only'
> attribute (my assumption is that it will be defined in draft-bundle).
>
>
>
> Note that, at this point, I don=E2=80=99t suggest any normative text rega=
rding the
> usage of port zero in the SDP Offer, as we are still discussing that (the
> outcome of that discussion may affect at least the last paragraph in sect=
ion
> x.2.1.).
>
>
>
> Also note that I don=E2=80=99t say anything about in which =E2=80=9Cscope=
/context=E2=80=9D (webrtc
> etc) the usage of the attribute is allowed.
>
>
>
> So, leaving the usage of port zero, and the scope question, aside for the
> moment, please take a look and indicate whether you have any issues (at t=
his
> point, I ask you to focus on the technical stuff =E2=80=93 and stuff whic=
h is
> unclear) with the text :)
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> --------------------------
>
>
>
>
>
> X.  SDP 'bundle-only' attribute
>
>
>
> X.1.  General
>
>
>
> This section defines a new SDP media-level attribute, 'bundle-only'. The
> attribute
>
> is used, in an SDP Offer, to indicate that an "m=3D" line, to which the
> attribute
>
> has been assigned, is only to be accepted by the Answerer if the Answerer
> keeps
>
> the "m=3D" line within the BUNDLE group to which the "m=3D" line is assoc=
iated
> in the
>
> SDP Offer.
>
>
>
>
>
> X.2.  SDP Offer/Answer Procedures
>
>
>
> X.2.1.  SDP Offer Procedures
>
>
>
> When an Offerer sends an SDP Offer, which contains a BUNDLE group, it can
> assign the
>
> SDP 'bundle-only' attribute to an "m=3D" line associated with the BUNDLE
> group,
>
> in order to request that the Answerer only accepts the "m=3D" line if is =
kept
> within
>
> the BUNDLE group to which the "m=3D" line is associated in the SDP Offer.
>
>
>
> An Offerer MUST NOT assign the SDP 'bundle-only' attribute to an "m=3D" l=
ine
> that is not
>
> associated with a BUNDLE group.
>
>
>
> If an Offerer receives an SDP Answer where an "m=3D" line, to which it
> assigned an
>
> 'bundle-only' attribute in the associated Offer, has been removed from a
> BUNDLE group, but
>
> has not been rejected ([ref-to-section- 6.5.2.]), the Offerer MUST send a
> new Offer, in which
>
> it assigns a zero port value to the "m=3D" line.
>
>
>
>
>
> X.2.2.  SDP Answerer Procedures
>
>
>
> When an Answerer receives an SDP Offer ([ref-to-section- 6.5.]), the
> Answerer MAY choose
>
> to remove an "m=3D" line out of a BUNDLE group ([ref-to-section- 6.5.2.])=
. If
> an SDP
>
> 'bundle-only' attribute has been assigned to such "m=3D" line, the Answer=
er
> MUST reject
>
> the "m=3D" line, according to the procedures in Section 6.5.3.
>
>
>
> An Answerer MUST NOT insert an SDP 'bundle-only' attribute in an SDP Answ=
er.
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

From ron.even.tlv@gmail.com  Fri Aug 30 09:36:01 2013
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Subject: [MMUSIC] FW: New Version Notification for draft-wu-avtcore-dynamic-pt-usage-00.txt
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Hi,
Apologize for the cross posting. Please comment only in avtcore mailing =
list

We have submitted a draft that clarifies the usage of dynamic payload =
type numbers.
This is based on email discussion in MMUSIC=20
Thanks
Roni Even

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: 30 August, 2013 7:29 PM
To: Qin Wu; Wenson Wu; Rachel Huang; Roni Even; Rachel Huang
Subject: New Version Notification for =
draft-wu-avtcore-dynamic-pt-usage-00.txt


A new version of I-D, draft-wu-avtcore-dynamic-pt-usage-00.txt
has been successfully submitted by Qin Wu and posted to the IETF =
repository.

Filename:	 draft-wu-avtcore-dynamic-pt-usage
Revision:	 00
Title:		 Guideline for dynamic payload type number usage policy
Creation date:	 2013-08-30
Group:		 Individual Submission
Number of pages: 5
URL:             =
http://www.ietf.org/internet-drafts/draft-wu-avtcore-dynamic-pt-usage-00.=
txt
Status:          =
http://datatracker.ietf.org/doc/draft-wu-avtcore-dynamic-pt-usage
Htmlized:        =
http://tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage-00


Abstract:
   The RTP Profile for Audio and Video Conferences with Minimal Control
   (RTP/AVP) is the basis for many other profiles, such as the Secure
   Real-time Transport Protocol (RTP/SAVP), the Extended RTP Profile for
   Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
   AVPF), and the Extended Secure RTP Profile for RTCP-Based Feedback
   (RTP/SAVPF).  This document updates RFC 3551 and provide guidelines
   for payload type number usage policy.

                                                                         =
        =20


Please note that it may take a couple of minutes from the time of =
submission until the htmlized version and diff are available at =
tools.ietf.org.

The IETF Secretariat


From christer.holmberg@ericsson.com  Fri Aug 30 09:36:54 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Subject: Re: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Hi,

>I think that you need to include in X.2.2 what the answerer puts in
>port for the "m=3D" line if it intends to accept a bundle-only line.

Good point.=20

I'll write that it adds the same BUNDLE address that is assigned to any oth=
er m- line within the same BUNDLE group.

Regards,

Christer


On 30 August 2013 06:48, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>
>
> Hi,
>
>
>
> I started to draft some text for the definition of the SDP 'bundle-only'
> attribute (my assumption is that it will be defined in draft-bundle).
>
>
>
> Note that, at this point, I don=92t suggest any normative text regarding =
the
> usage of port zero in the SDP Offer, as we are still discussing that (the
> outcome of that discussion may affect at least the last paragraph in sect=
ion
> x.2.1.).
>
>
>
> Also note that I don=92t say anything about in which =93scope/context=94 =
(webrtc
> etc) the usage of the attribute is allowed.
>
>
>
> So, leaving the usage of port zero, and the scope question, aside for the
> moment, please take a look and indicate whether you have any issues (at t=
his
> point, I ask you to focus on the technical stuff =96 and stuff which is
> unclear) with the text :)
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> --------------------------
>
>
>
>
>
> X.  SDP 'bundle-only' attribute
>
>
>
> X.1.  General
>
>
>
> This section defines a new SDP media-level attribute, 'bundle-only'. The
> attribute
>
> is used, in an SDP Offer, to indicate that an "m=3D" line, to which the
> attribute
>
> has been assigned, is only to be accepted by the Answerer if the Answerer
> keeps
>
> the "m=3D" line within the BUNDLE group to which the "m=3D" line is assoc=
iated
> in the
>
> SDP Offer.
>
>
>
>
>
> X.2.  SDP Offer/Answer Procedures
>
>
>
> X.2.1.  SDP Offer Procedures
>
>
>
> When an Offerer sends an SDP Offer, which contains a BUNDLE group, it can
> assign the
>
> SDP 'bundle-only' attribute to an "m=3D" line associated with the BUNDLE
> group,
>
> in order to request that the Answerer only accepts the "m=3D" line if is =
kept
> within
>
> the BUNDLE group to which the "m=3D" line is associated in the SDP Offer.
>
>
>
> An Offerer MUST NOT assign the SDP 'bundle-only' attribute to an "m=3D" l=
ine
> that is not
>
> associated with a BUNDLE group.
>
>
>
> If an Offerer receives an SDP Answer where an "m=3D" line, to which it
> assigned an
>
> 'bundle-only' attribute in the associated Offer, has been removed from a
> BUNDLE group, but
>
> has not been rejected ([ref-to-section- 6.5.2.]), the Offerer MUST send a
> new Offer, in which
>
> it assigns a zero port value to the "m=3D" line.
>
>
>
>
>
> X.2.2.  SDP Answerer Procedures
>
>
>
> When an Answerer receives an SDP Offer ([ref-to-section- 6.5.]), the
> Answerer MAY choose
>
> to remove an "m=3D" line out of a BUNDLE group ([ref-to-section- 6.5.2.])=
. If
> an SDP
>
> 'bundle-only' attribute has been assigned to such "m=3D" line, the Answer=
er
> MUST reject
>
> the "m=3D" line, according to the procedures in Section 6.5.3.
>
>
>
> An Answerer MUST NOT insert an SDP 'bundle-only' attribute in an SDP Answ=
er.
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>=

From hadriel.kaplan@oracle.com  Fri Aug 30 10:07:41 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Subject: Re: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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This presumes there is such a thing as "bundle-only" m-lines.

I don't believe we've determined consensus or not on your previous =
proposal to get rid of them, which was a long thread starting here I =
think:
https://www.ietf.org/mail-archive/web/mmusic/current/msg12114.html

-hadriel


On Aug 30, 2013, at 9:48 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> =20
> Hi,
> =20
> I started to draft some text for the definition of the SDP =
'bundle-only' attribute (my assumption is that it will be defined in =
draft-bundle).
> =20
> Note that, at this point, I don=92t suggest any normative text =
regarding the usage of port zero in the SDP Offer, as we are still =
discussing that (the outcome of that discussion may affect at least the =
last paragraph in section x.2.1.).
> =20
> Also note that I don=92t say anything about in which =93scope/context=94=
 (webrtc etc) the usage of the attribute is allowed.
> =20
> So, leaving the usage of port zero, and the scope question, aside for =
the moment, please take a look and indicate whether you have any issues =
(at this point, I ask you to focus on the technical stuff =96 and stuff =
which is unclear) with the text :)
> =20
> Regards,
> =20
> Christer
> =20

From christer.holmberg@ericsson.com  Fri Aug 30 11:25:09 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Subject: Re: [MMUSIC] BUNDLE DISCUSION Q8 (NEW): SDP 'bundle-only' attribute
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Hi,

>This presumes there is such a thing as "bundle-only" m-lines.
>
>I don't believe we've determined consensus or not on your previous proposa=
l to get rid of them, which was a long thread starting here I think:
>https://www.ietf.org/mail-archive/web/mmusic/current/msg12114.html

My issue has not been with the 'bundle-attribute', but with the usage of po=
rt zero, and that e-mail was a suggestion on how to get rid of that.

However, when re-reading that e-mail, if we would go ahead with my suggesti=
on it might also remove the need for the attribute...

Regards,

Christer



On Aug 30, 2013, at 9:48 AM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> =20
> Hi,
> =20
> I started to draft some text for the definition of the SDP 'bundle-only' =
attribute (my assumption is that it will be defined in draft-bundle).
> =20
> Note that, at this point, I don't suggest any normative text regarding th=
e usage of port zero in the SDP Offer, as we are still discussing that (the=
 outcome of that discussion may affect at least the last paragraph in secti=
on x.2.1.).
> =20
> Also note that I don't say anything about in which "scope/context" (webrt=
c etc) the usage of the attribute is allowed.
> =20
> So, leaving the usage of port zero, and the scope question, aside for the=
 moment, please take a look and indicate whether you have any issues (at th=
is point, I ask you to focus on the technical stuff - and stuff which is un=
clear) with the text :)
> =20
> Regards,
> =20
> Christer
> =20

From hadriel.kaplan@oracle.com  Fri Aug 30 13:28:02 2013
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Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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On Aug 30, 2013, at 9:39 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> We also need to decide whether that change also applies to when new =
media descriptions (m- line) are added to a previously negotiated BUNDLE =
group.
>>> Currently the text says that, when an m- line is added, it must =
contain either a unique address, or a previously negotiated BUNDLE =
address.
>>> But, since we now allow to assign a new, yet not confirmed, BUNDLE =
address to multiple m- lines, I guess that applies also to a new m- =
line.
>>=20
>> But then if the far-end answered with accepting it for the same =
bundle, then you'd have to send another offer with the m-line using the =
correct port of the bundle group, right?
>=20
> The change is that an Offerer is now allowed to assign the same =
address (the requested BUNDLE address) to all m- lines in the BUNDLE =
group BEFORE the Answerer has "confirmed" the address=20
> to become the Offerer's BUNDLE address.
>=20
> So, if the requested BUNDLE address was assigned to all m- lines in =
the BUNDLE group, and the Answerer accepts it, there is obviously no =
need to send a second offer.
>=20
> But, if any of the m- lines in the offer did NOT contain the selected =
BUNDLE address, a second Offer is still required :)

Actually, on reflection it doesn't need to send a new offer even if the =
far-end didn't want bundle. =20

As the offerer, if you want to add an m-line with bundle, simply use the =
the same port numbers as the existing bundle.  If the answerer responds =
with its same port number for its side of the bundle, you're done.  If =
the answerer decides it didn't want that new m-line to be bundled, it =
will simply pick a new port number in its answer.  You're still done.  =
Because it's not the same 5-tuple. (it happens to be the same local =
source port number on the offerer, but not the same dest port)=20

If the offerer now wants a new local port number for some reason (like =
it can't handle distinguishing full 5-tuples) it can send another offer =
to use a new port number for that m-line.

-hadriel


From christer.holmberg@ericsson.com  Fri Aug 30 23:55:42 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Hi,

>>>> We also need to decide whether that change also applies to when new me=
dia descriptions (m- line) are added to a previously negotiated BUNDLE grou=
p.
>>>> Currently the text says that, when an m- line is added, it must contai=
n either a unique address, or a previously negotiated BUNDLE address.
>>>> But, since we now allow to assign a new, yet not confirmed, BUNDLE add=
ress to multiple m- lines, I guess that applies also to a new m- line.
>>>
>>> But then if the far-end answered with accepting it for the same bundle,=
 then you'd have to send another offer with the m-line using the correct po=
rt of the bundle group, right?
>>
>> The change is that an Offerer is now allowed to assign the same address =
(the requested BUNDLE address) to all m- lines in the BUNDLE group BEFORE t=
he Answerer has "confirmed" the address
>> to become the Offerer's BUNDLE address.
>>
>> So, if the requested BUNDLE address was assigned to all m- lines in the =
BUNDLE group, and the Answerer accepts it, there is obviously no need to se=
nd a second offer.
>>
>> But, if any of the m- lines in the offer did NOT contain the selected BU=
NDLE address, a second Offer is still required :)
>
> Actually, on reflection it doesn't need to send a new offer even if the f=
ar-end didn't want bundle.
>
> As the offerer, if you want to add an m-line with bundle, simply use the =
the same port numbers as the existing bundle.  If the answerer responds=20
> with its same port number for its side of the bundle, you're done.  If th=
e answerer decides it didn't want that new m-line to be bundled, it will si=
mply=20
> pick a new port number in its answer.  You're still done.  Because it's n=
ot the same 5-tuple. (it happens to be the same local source port number on=
 the offerer, but not the same dest port)
>
> If the offerer now wants a new local port number for some reason (like it=
 can't handle distinguishing full 5-tuples) it can send another offer to us=
e a new port number for that m-line.

That is a separate topic, more related to the Answerer procedures, and is n=
ot alligned with the current BUNDLE text. Currently, if an m- line (no matt=
er whether it is a new m- line, or a previously added one) in an Offer cont=
ains a BUNDLE address, and the Answerer doesn't want it to be bundled, it M=
UST be rejected with port zero.

We have been discussing, and it is what my original bundle proposal describ=
ed, the usage of a single address on the Offerer side even if the Answerer =
uses separate ports, and I thought we decided not to go that way...

Regards,

Christer=

From hadriel.kaplan@oracle.com  Sat Aug 31 04:52:51 2013
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From: Hadriel Kaplan <hadriel.kaplan@oracle.com>
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Date: Sat, 31 Aug 2013 07:52:15 -0400
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References: <7594FB04B1934943A5C02806D1A2204B1C47762C@ESESSMB209.ericsson.se> <521CACD4.6060904@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B1C4809D7@ESESSMB209.ericsson.se> <968A751E-C145-4799-9479-417D850F50A6@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C480A75@ESESSMB209.ericsson.se>, <5DE22A0F-257F-4F86-B2A8-7E8627D96D6A@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C4814A1@ESESSMB209.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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On Aug 31, 2013, at 2:55 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> That is a separate topic, more related to the Answerer procedures, and =
is not alligned with the current BUNDLE text. Currently, if an m- line =
(no matter whether it is a new m- line, or a previously added one) in an =
Offer contains a BUNDLE address, and the Answerer doesn't want it to be =
bundled, it MUST be rejected with port zero.
>=20
> We have been discussing, and it is what my original bundle proposal =
described, the usage of a single address on the Offerer side even if the =
Answerer uses separate ports, and I thought we decided not to go that =
way=85

No the issue with the Offer using the same ports is an issue for the =
*initial* offer, before the offerer knows the far-end supports bundle.  =
In the above I was talking about the subsequent offers/answers, once the =
offerer knows the far-end supports bundle.

Regardless, I suppose we really do need to have the 'bundle-only' =
attribute anyway, just to enable the offerer to say "I am only offering =
these m-lines using the same bundle transport, and if you don't want to =
do that then I want them to not be used at all".

-hadriel


From christer.holmberg@ericsson.com  Sat Aug 31 08:36:24 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <hadriel.kaplan@oracle.com>
Thread-Topic: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
Thread-Index: Ac6f7OG1P0YUwhF6Remol1dqLcLlOQDLal4AAJiyiVD//+hjAP//16UQgACiigCAAM7gU4AAM26AgABeIwo=
Date: Sat, 31 Aug 2013 15:36:06 +0000
Message-ID: <7594FB04B1934943A5C02806D1A2204B1C481723@ESESSMB209.ericsson.se>
References: <7594FB04B1934943A5C02806D1A2204B1C47762C@ESESSMB209.ericsson.se> <521CACD4.6060904@alum.mit.edu> <7594FB04B1934943A5C02806D1A2204B1C4809D7@ESESSMB209.ericsson.se> <968A751E-C145-4799-9479-417D850F50A6@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C480A75@ESESSMB209.ericsson.se>, <5DE22A0F-257F-4F86-B2A8-7E8627D96D6A@oracle.com> <7594FB04B1934943A5C02806D1A2204B1C4814A1@ESESSMB209.ericsson.se>, <792914D0-6BF9-4D46-80C5-25D4678D707C@oracle.com>
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Subject: Re: [MMUSIC] DECISION (BERLIN-Q5): When the BUNDLE address is RE-negotiated, the Offerer can assign the requsted new address to all m- lines immediately, without first offering unique addresses.
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Hi,

>> That is a separate topic, more related to the Answerer procedures, and i=
s not alligned with the current BUNDLE text. Currently, if an=20
>> m- line (no matter whether it is a new m- line, or a previously added on=
e) in an Offer contains a BUNDLE address, and the Answerer=20
>> doesn't want it to be bundled, it MUST be rejected with port zero.
>>
>> We have been discussing, and it is what my original bundle proposal desc=
ribed, the usage of a single address on the Offerer side even if the Answer=
er uses separate ports, and I thought we decided not to go that way=85
>
> No the issue with the Offer using the same ports is an issue for the *ini=
tial* offer, before the offerer knows the far-end supports bundle.  In the =
above I was talking about the subsequent offers/answers, once the offerer k=
nows the far-end supports bundle.

Fair enough. But, it's still not specific to 'bundle-only' m- lines.

> Regardless, I suppose we really do need to have the 'bundle-only' attribu=
te anyway, just to enable the offerer to say "I am only offering these m-li=
nes using the same bundle transport, and if you don't want to do that then =
I want them to not be used at all".

With the CURRENT rules you do that by assigning the BUNDLE address to those=
 m- lines, because the Answerer must reject them in case they are removed f=
rom the BUNDLE group.

But, IF we want to change that (for mid-call O/As), there might be a need f=
or an explicit attribute.

In any case, we can work on the text for now, and then later throw it away =
if we find out it's not needed :)

Regards,

Christer=
