
From harald@alvestrand.no  Mon Jan  6 10:30:58 2014
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Subject: Re: [MMUSIC] Some comments on draft-ietf-mmusic-msid-02
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Thanks for the review! I'm now trying to get the -03 version out (at
long last).

The comments have been incorporated. Hope they work out to your
satisfaction!

On 12/17/2013 11:01 PM, Paul Kyzivat wrote:
> In Vancouver I volunteered to look at this version of the msid draft,
> especially wrt its relationship to the app-token draft.
>
> After looking, IMO some things can be done to this draft that will not
> effect its substance, but will more clearly separate its concerns from
> those of app-token and unified-plan.
>
> The unified-plan has gotten us to a point where individual source
> streams each have their own m-line.
>
> The app-token draft is defining a mechanism that will provide allow a
> level of indirection to SSRCs. That hasn't matured to the point of
> being picked up in unified-plan, but I hope it eventually will be.
>
> The problem I see now is that this msid draft is still filled with
> lots of reference to SSRCs, even though the mechanism it specifies no
> longer is at all dependent on SSRCs. The mechanism is solely for the
> purpose of attaching related identifiers to m-lines and groups of
> m-lines. How those m-lines are associated with SSRCs and/or appids is
> better left to other documents.
>
> If it is so decoupled, then it will be easy to progress this one. Then
> we can leave it to unified-plan and app-token to do their thing. And
> in the future, if new mechanisms are defined to bind streams to
> m-lines, then this document will remain valid.
>
> Some specifics:
>
> * Section 1.2:
>
>    The SDP grouping framework [RFC5888] can be used to group m-lines.
>    However, there is sometimes the need for an application to specify
>    some application-level information about the association between the
>    SSRC and the group.  This is not possible using the SDP grouping
>    framework.
>
> s/SSRC/m-line/

Done.
>
> * Section 1.3:
>
>    In the RTP specification, media streams are identified using the SSRC
>    field.  Streams are grouped into RTP Sessions, and also carry a
>    CNAME.  Neither CNAME nor RTP session correspond to a MediaStream.
>    Therefore, the association of an RTP media stream to MediaStreams
>    need to be explicitly signaled.
>
> I think this entire paragraph can be dropped. The following paragraph
> is sufficient.

This is background indicating why CNAME is not an appropriate mechanism.
While this debate seems to be long over, I'd like to keep the paragraph.

>
> * Section 2:
>
>    Endpoints can update the associations between SSRCs as expressed by
>    msid attributes at any time; the semantics and restrictions of such
>    grouping and ungrouping are application dependent.
>
> s/SSRCs/m-lines/

In this context, I prefer to use "RTP media streams".

>
> * Section 4:
>
>    This section creates a new semantic for use with the framework
>    defined in Section 2, to be used for associating SSRCs representing
>    MediaStreamTracks within MediaStreams as defined in
>    [W3C.WD-webrtc-20120209].
>
> s/SSRCs/m-lines/

Done.

>
>    In a WebRTC-compatible SDP description, all SSRCs intending to be
>    sent from one peer will be identified in the SDP generated by that
>    entity.
>
> I'm not sure about the above. It's true now based on unified-plan. It
> might be altered by the use of app-token.

If unified-plan is changed, this will have to be changed too. But it
doesn't matter for this specification, actually, so I'm deleting this
paragraph.

>
>    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).
>
> Another one I'm not sure about. If app-token is used to define
> m-lines, then it ceases to be so clear when the track should close.
> Maybe it should close if a BYE or timeout is received on the last SSRC
> bound to the app-token. Or maybe not. This probably needs more work.
> But is this the right draft to specify that?

This is definitely not the right draft to describe the use of app-token.

This text resulted from previous discussions, and was a question in an
earlier version - I believe Colin Perkins suggested that the
disappearance according to RTP rules should cause the ending of the
track too.

If one track is associated with multiple SSRCs, then it is logical that
all the SSRCs have to disappear before it is ended; this is already in
the text.
No change.

>
> * Section 5:
>
>    o  The attribute gives an association over a set of m-lines.  It can
>       be used to signal the relationship between a WebRTC MediaStream
>       and a set of SSRCs.
>
> s/SSRCs/m-lines/

Done.

>
> * Appendices: I'm ignoring them.
>
>     Thanks,
>     Paul
>
> On 12/17/13 7:24 AM, Ari Keränen wrote:
>> On 12/16/13 4:42 PM, Harald Alvestrand wrote:
>>> On 12/16/2013 03:37 PM, Ari Keränen wrote:
>>>> Hi Roni and Paul,
>>>>
>>>> At the Vancouver meeting we discussed that the MSID draft would need
>>>> more review and input, also regarding it's relationship with the
>>>> app-token draft. You guys volunteered to have a look at the new
>>>> version Harald submitted during the meeting:
>>>> http://tools.ietf.org/html/draft-ietf-mmusic-msid-02
>>>>
>>>> When you have a chance, would be great if you can check out the draft
>>>> and post your comments on the list.
>>>>
>>>>
>>>> Thanks,
>>>> Ari
>>> There was also significant input on the document in the WEBRTC WG
>>> meeting in Shenzhen - especially around the subject of non-signalled
>>> SSRCs.
>>>
>>> I'll issue an updated version based on that; if you can get feedback in
>>> soon I can incorporate that too.
>>>
>>
>> Thanks for the update Harald! Could you summarize that input to the
>> mailing list so that the WG is up to speed with the discussion?
>>
>>
>> Cheers,
>> Ari
>>
>


-- 
Surveillance is pervasive. Go Dark.


From internet-drafts@ietf.org  Mon Jan  6 10:33:53 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-msid-03.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 Sessio=
n Description Protocol
        Author          : Harald Alvestrand
	Filename        : draft-ietf-mmusic-msid-03.txt
	Pages           : 15
	Date            : 2014-01-06

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-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-msid-03


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 pkyzivat@alum.mit.edu  Mon Jan  6 11:53:20 2014
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Subject: Re: [MMUSIC] Some comments on draft-ietf-mmusic-msid-02
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More comments inline. I'm not commenting on things were we are in full 
agreement.

*** Note to Roni and Jonathan: need input from you on this ***

On 1/6/14 1:30 PM, Harald Alvestrand wrote:
> Thanks for the review! I'm now trying to get the -03 version out (at
> long last).
>
> The comments have been incorporated. Hope they work out to your
> satisfaction!
>
> On 12/17/2013 11:01 PM, Paul Kyzivat wrote:
>> In Vancouver I volunteered to look at this version of the msid draft,
>> especially wrt its relationship to the app-token draft.
>>
>> After looking, IMO some things can be done to this draft that will not
>> effect its substance, but will more clearly separate its concerns from
>> those of app-token and unified-plan.
>>
>> The unified-plan has gotten us to a point where individual source
>> streams each have their own m-line.
>>
>> The app-token draft is defining a mechanism that will provide allow a
>> level of indirection to SSRCs. That hasn't matured to the point of
>> being picked up in unified-plan, but I hope it eventually will be.
>>
>> The problem I see now is that this msid draft is still filled with
>> lots of reference to SSRCs, even though the mechanism it specifies no
>> longer is at all dependent on SSRCs. The mechanism is solely for the
>> purpose of attaching related identifiers to m-lines and groups of
>> m-lines. How those m-lines are associated with SSRCs and/or appids is
>> better left to other documents.
>>
>> If it is so decoupled, then it will be easy to progress this one. Then
>> we can leave it to unified-plan and app-token to do their thing. And
>> in the future, if new mechanisms are defined to bind streams to
>> m-lines, then this document will remain valid.
>>
>> Some specifics:
>>
>> * Section 1.2:
>>
>>     The SDP grouping framework [RFC5888] can be used to group m-lines.
>>     However, there is sometimes the need for an application to specify
>>     some application-level information about the association between the
>>     SSRC and the group.  This is not possible using the SDP grouping
>>     framework.
>>
>> s/SSRC/m-line/
>
> Done.
>>
>> * Section 1.3:
>>
>>     In the RTP specification, media streams are identified using the SSRC
>>     field.  Streams are grouped into RTP Sessions, and also carry a
>>     CNAME.  Neither CNAME nor RTP session correspond to a MediaStream.
>>     Therefore, the association of an RTP media stream to MediaStreams
>>     need to be explicitly signaled.
>>
>> I think this entire paragraph can be dropped. The following paragraph
>> is sufficient.
>
> This is background indicating why CNAME is not an appropriate mechanism.
> While this debate seems to be long over, I'd like to keep the paragraph.

I can live with that.

>> * Section 2:
>>
>>     Endpoints can update the associations between SSRCs as expressed by
>>     msid attributes at any time; the semantics and restrictions of such
>>     grouping and ungrouping are application dependent.
>>
>> s/SSRCs/m-lines/
>
> In this context, I prefer to use "RTP media streams".

I think that still leaves things a bit confusing. There are two kinds of 
interesting changes possible:

1) the association between msid attributes and m-lines can be updated
2) the association between m-lines and RTP streams can be updated

Of these, (1) is directly relevant to this draft, and should be made clear.

Meanwhile, (2) is also important, and may also be worth pointing out in 
this draft. But the mechanics of doing so are in the scope of other drafts.

>> * Section 4:
>>
>>     This section creates a new semantic for use with the framework
>>     defined in Section 2, to be used for associating SSRCs representing
>>     MediaStreamTracks within MediaStreams as defined in
>>     [W3C.WD-webrtc-20120209].
>>
>> s/SSRCs/m-lines/
>
> Done.
>
>>
>>     In a WebRTC-compatible SDP description, all SSRCs intending to be
>>     sent from one peer will be identified in the SDP generated by that
>>     entity.
>>
>> I'm not sure about the above. It's true now based on unified-plan. It
>> might be altered by the use of app-token.
>
> If unified-plan is changed, this will have to be changed too. But it
> doesn't matter for this specification, actually, so I'm deleting this
> paragraph.

OK.

>>
>>     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).
>>
>> Another one I'm not sure about. If app-token is used to define
>> m-lines, then it ceases to be so clear when the track should close.
>> Maybe it should close if a BYE or timeout is received on the last SSRC
>> bound to the app-token. Or maybe not. This probably needs more work.
>> But is this the right draft to specify that?
>
> This is definitely not the right draft to describe the use of app-token.

Or the use of Unified!

> This text resulted from previous discussions, and was a question in an
> earlier version - I believe Colin Perkins suggested that the
> disappearance according to RTP rules should cause the ending of the
> track too.

I think this may now require further discussion.

> If one track is associated with multiple SSRCs, then it is logical that
> all the SSRCs have to disappear before it is ended; this is already in
> the text.
> No change.

The problem, if there is one, is what it means for SSRCs to be 
associated with an m-line. IIUC, with appid there is only one SSRC 
associated with the m-line at any one time, but which SSRC it is may change.

The problem would come if a BYE was received on the SSRC currently 
associated with the m-line based on appid, and then another packet came 
on a different SSRC and an appid token that ties it to this m-line. I 
don't know what should happen in this case. I'd like to hear from Roni 
and Jonathan about it.

Clearly what should happen in that case is not something for this draft 
to address. I just don't want this draft to constrain what should happen 
in that case.

>> * Section 5:
>>
>>     o  The attribute gives an association over a set of m-lines.  It can
>>        be used to signal the relationship between a WebRTC MediaStream
>>        and a set of SSRCs.
>>
>> s/SSRCs/m-lines/
>
> Done.

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Thu Jan  9 04:25:21 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: James Rafferty <jrafferty@humancomm.com>, "mmusic@ietf.org" <mmusic@ietf.org>, Gonzalo Salgueiro <gsalguei@cisco.com>
Thread-Topic: [MMUSIC] Review (Rafferty) of draft-ietf-mmusic-udptl-dtls-02
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Subject: Re: [MMUSIC] Review (Rafferty) of draft-ietf-mmusic-udptl-dtls-02
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Hi James,

Thanks for your review! Comments inline.

> This is a review ofdraft-ietf-mmusic-udptl-dtls-02.   In general, the=20
> draft is in good shape.  My comments and suggestions are included below.
>
> Section 1 - first bullet:  add "the" before application layer

Ok.

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

> Section 1, Page 2  - On the paragraph before Table 2, the current last se=
ntence reads:  "The protocol stack for integrity and confidentiality protec=
ted fax transport using UDPTL over DTLS is shown in Table 2."
>
> A couple of word changes would make it read more clearly:
> "The protocol stack which enhances fax transport to offer integrity and c=
onfidentiality using UDPTL over DTLS is shown in Table 2."

Looks ok.

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

> Section 1, Page 3:  Under the "primary motivations" , I'd suggest broaden=
ing the applicability of the last bullet point to read:
>
> "3GPP and the IP fax community need a mechanism to transport UDPTL over D=
TLS  in order to provide secure fax in IMS and other SIP-based networks."

Looks ok.

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

> Section 2
>
> for the 2nd sentence of the second paragraph, clarify by replacing "it"  =
by "session"

Ok.

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

> Section 5 - Security Considerations
>
> I think it would help to pull in material about the threat model before j=
umping directly into the solution (my latest draft got dinged in the IESG s=
ecurity review on this point).
>
> From a fax community perspective, the relevant threats seem to be:
>
> 1.  Confidentiality - Address potential for 3rd parties to intercept and =
decode messages
>
> 2.  Integrity - Ensure that the parties exchanging the messages are prope=
rly authenticated and that the message has not been altered during transpor=
t.
>
> These points are addressed somewhat in the body of this section, but it w=
ould be useful to start with the threats model and then identify how the so=
lution meets the requirements.

We'll look into it.

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

> Section 6 - IANA Considerations
>
> For consistency with the IANA content for "proto"  in RFC 4566, it looks =
like the entry for the SDP name in Table 3 should be in quotes or "UDP/TLS/=
UDPTL"

Ok.

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

> Example A.3
>
> This is an important example, since most (if not all) current UDPTL sessi=
ons will start as audio sessions.
>
> To highlight this point, I'd suggest adding a second sentence in this sec=
tion such as:
>
> "By current conventions, most non-secured UDPTL sessions are initiated vi=
a a re-INVITE after the SIP session has begun as an audio session and this =
example shows to address this scenario for initiating UDPTL sessions secure=
d via DTLS."

In general I am ok with the suggestion, but perhaps we could replace "By cu=
rrent conventions" with "Traditionally". Because, we don't know how things =
will be done in the future.

I also suggest some other editorial changes to the sentence. Something like=
:

OLD:

	Figure 6 focuses on T.38 fax securely transported using UDPTL over
   	DTLS replacing audio media stream in an existing audio-only session.


NEW:

	Traditionally, most session with non-secure transport of T.38 fax,
	transported using UDPTL, are established by modifying an ongoing=20
	audio session into a fax session.
	Figure 6 shows an example message flow of modifying
	an existing audio session into a session with T.38 fax=20
	securely transported using UDPTL over DTLS.

...or something like that. I'll need to make it "fit" into the existing tex=
t :)

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

> That concludes my comments on this draft.

Thank You very much! :)

Regards,

Christer

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From: Tom Kristensen <2mkristensen@gmail.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>
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Cc: Tom Kristensen <tomkrist@cisco.com>, "bfcpbis-chairs@tools.ietf.org" <bfcpbis-chairs@tools.ietf.org>, "draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org" <draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org>, "<mmusic@ietf.org>" <mmusic@ietf.org>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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Great. Thanks for the review. Comments inline.

On 20 December 2013 14:26, Ali C. Begen (abegen) <abegen@cisco.com> wrote:

> I am the assigned SDP directorate reviewer for this draft. For background
> on the SDP directorate, please see the FAQ at
>
> http://www.ietf.org/iesg/directorate/sdp.html
>
> Please wait for direction from your document shepherd or AD before posting
> a new version of the draft.
>

Aye aye, Captain!

Summary:
> - There are a few minor issues as indicated below.
>
> Issues:
> 1) Section 3 uses m=<media> <port> <transport> <fmt> and refers to 4566
> but 4566 actually uses m=<media> <port> <proto> <fmt> .
>
> The newly defined 4 values (TCP/BFCP, TCP/TLS/BFCP and the UDP
> equivalents) are actually the proto values that are registered with IANA
> (not transport values). Thus, I recommend changing the text to reflect this.
>

 I agree. This is inherited from RFC 4583, but as long as RFC 4566 is
referenced we should use a corresponding definition. I will change.


> 2) Definition of "floorctrl" attribute. I think the answer has to include
> only one role if the offer had this attribute. If that is correct, I
> suggest the following change:
>
> OLD:
>    If an SDP media description in an offer contains a 'floorctrl'
>    attribute, the answerer accepting that media MUST include one in the
>    corresponding media description of the answer.
>
> NEW:
>    If an SDP media description in an offer contains a 'floorctrl'
>    attribute, the answerer accepting that media MUST include only one in
> the
>    corresponding media description of the answer.
>

Yes, the answer must have one - and only one - floorctrl attribute value,
i.e. role. However, in the last sentence of Section 4 it is stated that:

   The following is an example of a 'floorctrl' attribute in an offer.
   When this attribute appears in an answer, it only carries one role:


I think that is sufficient and we haven't seen any cases of
misinterpretation of this in the field!

3) There are a few "SHOULD"s. I think they are OK but it is not clear when
> it is OK to ignore that SHOULD. If there are no such use cases, maybe they
> can be replaced with a "MUST"?
>

This is inherited from the original RFC 4583, that's the main reason for me
to suggest not changing this in this draft. Hopefully, that's fine?

-- Tom

-- 
# Cisco                         |  http://www.cisco.com/telepresence/
## tomkrist@cisco.com  |  http://www.tandberg.com
###                               |  http://folk.uio.no/tomkri/

--047d7bd751320e5fae04ef9b7e23
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<div dir=3D"ltr">Great. Thanks for the review. Comments inline.<br><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On 20 December 2013 14:26=
, Ali C. Begen (abegen) <span dir=3D"ltr">&lt;<a href=3D"mailto:abegen@cisc=
o.com" target=3D"_blank">abegen@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">I am the assigned SDP directorate reviewer for this draft.=
 For background on the SDP directorate, please see the FAQ at<br>

<br>
<a href=3D"http://www.ietf.org/iesg/directorate/sdp.html" target=3D"_blank"=
>http://www.ietf.org/iesg/directorate/sdp.html</a><br>
<br>
Please wait for direction from your document shepherd or AD before posting =
a new version of the draft.<br></blockquote><div><br></div><div style>Aye a=
ye, Captain!</div><div style><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb=
(204,204,204);border-left-style:solid;padding-left:1ex">

Summary:<br>
- There are a few minor issues as indicated below.<br>
<br>
Issues:<br>
1) Section 3 uses m=3D&lt;media&gt; &lt;port&gt; &lt;transport&gt; &lt;fmt&=
gt; and refers to 4566 but 4566 actually uses m=3D&lt;media&gt; &lt;port&gt=
; &lt;proto&gt; &lt;fmt&gt; .<br>
<br>
The newly defined 4 values (TCP/BFCP, TCP/TLS/BFCP and the UDP equivalents)=
 are actually the proto values that are registered with IANA (not transport=
 values). Thus, I recommend changing the text to reflect this.<br></blockqu=
ote>
<div><br></div><div style>=A0I agree. This is inherited from RFC 4583, but =
as long as RFC 4566 is referenced we should use a corresponding definition.=
 I will change.</div><div>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">

2) Definition of &quot;floorctrl&quot; attribute. I think the answer has to=
 include only one role if the offer had this attribute. If that is correct,=
 I suggest the following change:<br>
<br>
OLD:<br>
=A0 =A0If an SDP media description in an offer contains a &#39;floorctrl&#3=
9;<br>
=A0 =A0attribute, the answerer accepting that media MUST include one in the=
<br>
=A0 =A0corresponding media description of the answer.<br>
<br>
NEW:<br>
=A0 =A0If an SDP media description in an offer contains a &#39;floorctrl&#3=
9;<br>
=A0 =A0attribute, the answerer accepting that media MUST include only one i=
n the<br>
=A0 =A0corresponding media description of the answer.<br></blockquote><div>=
=A0</div><div style>Yes, the answer must have one - and only one - floorctr=
l attribute value, i.e. role. However, in the last sentence of Section 4 it=
 is stated that:=A0</div>
<div style><br></div><div style><pre class=3D"" style=3D"font-size:1em;marg=
in-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   The following is an examp=
le of a &#39;floorctrl&#39; attribute in an offer.
   When this attribute appears in an answer, it only carries one role:</pre=
></div><div style><br></div><div style>I think that is sufficient and we ha=
ven&#39;t seen any cases of misinterpretation of this in the field!</div>
<div style><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bord=
er-left-style:solid;padding-left:1ex">
3) There are a few &quot;SHOULD&quot;s. I think they are OK but it is not c=
lear when it is OK to ignore that SHOULD. If there are no such use cases, m=
aybe they can be replaced with a &quot;MUST&quot;?<br></blockquote><div>
<br></div><div style>This is inherited from the original RFC 4583, that&#39=
;s the main reason for me to suggest not changing this in this draft. Hopef=
ully, that&#39;s fine?</div></div><div class=3D"gmail_extra"><br></div>-- T=
om<br clear=3D"all">
<div><br></div>-- <br># Cisco =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 | =A0<a href=3D"http://www.cisco.com/telepresence/" target=3D"_blank">h=
ttp://www.cisco.com/telepresence/</a><br>## <a href=3D"mailto:tomkrist@cisc=
o.com" target=3D"_blank">tomkrist@cisco.com</a> =A0| =A0<a href=3D"http://w=
ww.tandberg.com" target=3D"_blank">http://www.tandberg.com</a><br>
### =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0<a hre=
f=3D"http://folk.uio.no/tomkri/" target=3D"_blank">http://folk.uio.no/tomkr=
i/</a>
</div></div>

--047d7bd751320e5fae04ef9b7e23--

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#2: Rules for Answerer payload type selection

Changes (by fandreas@cisco.com):

 * status:  new => closed
 * resolution:   => fixed


-- 
-------------------------------------+------------------------------------
  Reporter:  hkaplan@acmepacket.com  |      Owner:  hkaplan@acmepacket.com
      Type:  defect                  |     Status:  closed
  Priority:  major                   |  Component:  media-loopback
   Version:  1.0                     |   Severity:  In WG Last Call
Resolution:  fixed                   |   Keywords:
-------------------------------------+------------------------------------

Ticket URL: <http://tools.ietf.org/wg/mmusic/trac/ticket/2#comment:2>
mmusic <http://tools.ietf.org/mmusic/>


From ari.keranen@ericsson.com  Fri Jan 10 08:00:29 2014
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Subject: Re: [MMUSIC] Fwd: New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Hi,

I read the latest draft and have a few questions. In Section 3 the draft 
says:

>    ICE [RFC5245] section 4.1.2 states that the formula in section
>    4.1.2.1 SHOULD be used.  Failing to do so may lead to ICE taking
>    longer to converge as the checklist no longer will be coordinated.
>    Therefore responsiveness of ICE candidate checks are improved when
>    both sides support Happy-Eyeballs, both sides have the same number of
>    candidate pairs, and both sides use the same Happy Eyeballs promotion
>    algorithm.

Do you mean the checklist is no longer coordinated (i.e., matching on 
both sides) if all those requirements aren't fulfilled? That sounds a 
bit strange but that's the impression I got from the text.

Also, if there is no coordination/signaling between the endpoints, what 
are the odds they'd end up using the same promotion algorithm?

>    If each ICE agent uses a different algorithm to promote IPv4
>    candidates, ICE connectivity checks will be as responsive as the
>    least aggressive algorithm.

What is "least aggressive algorithm"?


Cheers,
Ari

From internet-drafts@ietf.org  Fri Jan 10 08:40:25 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-ice-sip-sdp-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           : Using Interactive Connectivity Establishment (ICE=
) with Session Description Protocol (SDP) offer/answer and Session Initiati=
on Protocol (SIP)
        Authors         : Marc Petit-Huguenin
                          Ari Keranen
	Filename        : draft-ietf-mmusic-ice-sip-sdp-01.txt
	Pages           : 41
	Date            : 2014-01-10

Abstract:
   This document describes how Interactive Connectivity Establishment
   (ICE) is used with Session Description Protocol (SDP) offer/answer
   and Session Initiation Protocol (SIP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-ice-sip-sdp/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-ice-sip-sdp-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 abegen@cisco.com  Fri Jan 10 09:34:44 2014
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Tom Kristensen <2mkristensen@gmail.com>
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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Cc: "Tom Kristensen \(tomkrist\)" <tomkrist@cisco.com>, "bfcpbis-chairs@tools.ietf.org" <bfcpbis-chairs@tools.ietf.org>, "draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org" <draft-ietf-bfcpbis-rfc4583bis@tools.ietf.org>, "<mmusic@ietf.org>" <mmusic@ietf.org>
Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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On Jan 10, 2014, at 2:50 AM, Tom Kristensen <2mkristensen@gmail.com> wrote:

> 3) There are a few "SHOULD"s. I think they are OK but it is not clear whe=
n it is OK to ignore that SHOULD. If there are no such use cases, maybe the=
y can be replaced with a "MUST"?
>=20
> This is inherited from the original RFC 4583, that's the main reason for =
me to suggest not changing this in this draft. Hopefully, that's fine?

Well I guess IESG will have issues with this, so take this as a warning :)

>=20
> -- Tom


From internet-drafts@ietf.org  Fri Jan 10 09:37:58 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-ice-sip-sdp-02.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           : Using Interactive Connectivity Establishment (ICE=
) with Session Description Protocol (SDP) offer/answer and Session Initiati=
on Protocol (SIP)
        Authors         : Marc Petit-Huguenin
                          Ari Keranen
	Filename        : draft-ietf-mmusic-ice-sip-sdp-02.txt
	Pages           : 41
	Date            : 2014-01-10

Abstract:
   This document describes how Interactive Connectivity Establishment
   (ICE) is used with Session Description Protocol (SDP) offer/answer
   and Session Initiation Protocol (SIP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-ice-sip-sdp/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-ice-sip-sdp-02


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 palmarti@cisco.com  Mon Jan 13 03:46:34 2014
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: =?Windows-1252?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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On 10 Jan 2014, at 17:00 pm, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote=
:

> Hi,
>=20
> I read the latest draft and have a few questions. In Section 3 the draft =
says:
>=20
>>   ICE [RFC5245] section 4.1.2 states that the formula in section
>>   4.1.2.1 SHOULD be used.  Failing to do so may lead to ICE taking
>>   longer to converge as the checklist no longer will be coordinated.
>>   Therefore responsiveness of ICE candidate checks are improved when
>>   both sides support Happy-Eyeballs, both sides have the same number of
>>   candidate pairs, and both sides use the same Happy Eyeballs promotion
>>   algorithm.
>=20
> Do you mean the checklist is no longer coordinated (i.e., matching on bot=
h sides) if all those requirements aren't fulfilled? That sounds a bit stra=
nge but that's the impression I got from the text.

What we tried to say was that if the two agents have different Happy Eyebal=
l Algorithms ICE might take longer to complete.
Both Alice and Bob used the following formula to sort the checklist: After =
N numbers of IPv6 addresses, add M numbers of IPv4 addresses.=20

   Alice(N=3D4, M=3D1)                   Bob(N=3D2,M=3D1)
1     IPv6                                     IPv6
2     IPv6                                     IPv6
3     IPv6                                     IPv4(*)
4     IPv6                                     IPv6
5     IPv4(*)                                 IPv6
6     IPv6                                     IPv4(*)
7     IPv6                                     IPv6
8     IPv6                                     IPv6
9     IPv6                                     IPv4(*)
10   IPv4(*)                                 IPv6
11   IPv6                                     IPv6
12   IPv6                                     IPv4(*)
13   IPv6                                     IPv6
14   IPv6                                     IPv6
15   IPv4(*)
16   IPv4(*)

 The initial thought was that having for example different M and N values w=
ould cause ICE to take longer to complete. But due to triggered checks diff=
erent M and N values might actually speed up the ICE negotiation.=20

I will update the text using this simple example algorithm, and rewrite Sec=
tion 3.

>=20
> Also, if there is no coordination/signaling between the endpoints, what a=
re the odds they'd end up using the same promotion algorithm?
>=20
Exactly, and that might be an advantage..=20

>>   If each ICE agent uses a different algorithm to promote IPv4
>>   candidates, ICE connectivity checks will be as responsive as the
>>   least aggressive algorithm.
>=20
> What is "least aggressive algorithm=94?

Good question. Regarding what the meaning was I think it was wron.


Rereading the entire draft myself. I realise it is in a pretty bad state. I=
 will do a full rewrite focusing on:

1. Explaining the need for the draft. (Get IPv6 path a head start so we cho=
ose that, but try to minimize the delay if the path is broken)
2. Get rid of generic IP address language. We are dealing with IPv6 and IPv=
4. That is not going to change for a while. (Haing language as if IPv8 was =
just around the corner just makes it harder to read)
3. Add example algorithm as described in this mail

Thanks for the review, will submit a 05 version before next IETF.=20

.-.
P=E5l-Erik

>=20

>=20
> Cheers,
> Ari


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To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, "Pal Martinsen (palmarti)" <palmarti@cisco.com>
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Hi Ari,

On 1/10/14 9:30 PM, "Ari Ker=E4nen" <ari.keranen@ericsson.com> wrote:

>Hi,
>
>I read the latest draft and have a few questions. In Section 3 the draft
>says:
>
>>    ICE [RFC5245] section 4.1.2 states that the formula in section
>>    4.1.2.1 SHOULD be used.  Failing to do so may lead to ICE taking
>>    longer to converge as the checklist no longer will be coordinated.
>>    Therefore responsiveness of ICE candidate checks are improved when
>>    both sides support Happy-Eyeballs, both sides have the same number of
>>    candidate pairs, and both sides use the same Happy Eyeballs promotion
>>    algorithm.
>
>Do you mean the checklist is no longer coordinated (i.e., matching on
>both sides) if all those requirements aren't fulfilled? That sounds a
>bit strange but that's the impression I got from the text.
>
>Also, if there is no coordination/signaling between the endpoints, what
>are the odds they'd end up using the same promotion algorithm?
>
>>    If each ICE agent uses a different algorithm to promote IPv4
>>    candidates, ICE connectivity checks will be as responsive as the
>>    least aggressive algorithm.
>
>What is "least aggressive algorithm"?

Since there is no signaling between the end points, each end point could
promote IPv4 to different positions in the checklist. Based on candidate
pair priority =3D 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0), the end point tha=
t
has promoted IPv4 in the least aggressive fashion i.e. IPv4 lower in its
list when compared to the other end point, will influence the candidate
priority pairing due to (2^32*MIN(G,D)) in the formula.


Thanks,
Prashanth

>
>
>Cheers,
>Ari
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, Tom Kristensen <2mkristensen@gmail.com>
Thread-Topic: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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Subject: Re: [MMUSIC] SDP Directorate: Review of draft-ietf-bfcpbis-rfc4583bis-08
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I am afraid Ali is right. Let's take the SHOULDs 1 by 1 and see how to
deal with them.

1) From section 3:
"The fmt (format) list is ignored for BFCP.  The fmt list of BFCP 'm'
The fmt (format) list is ignored for BFCP.  The fmt list of BFCP 'm'
         lines SHOULD contain a single "*" character."


This is inherited from RFC 4583. I see no reason to not use '*'. Unless
someone can identify one, I suggest changing to "MUST".

2) From section 4:
"Endpoints that use the offer/answer model to establish BFCP
   connections MUST support the 'floorctrl' attribute.  A floor control
   server acting as an offerer or as an answerer SHOULD include this
   attribute in its session descriptions.
   If the 'floorctrl' attribute is not used in an offer/answer exchange,
   by default the offerer and the answerer will act as a floor control
   client and as a floor control server, respectively."


I think this is okay as is.

3) From section 5:
"Endpoints that use the offer/answer model to establish BFCP
   connections MUST support the 'confid' and the 'userid' attributes.  A
   floor control server acting as an offerer or as an answerer SHOULD
   include these attributes in its session descriptions."

This is inherited form RFC 4583. I have seen clients not including these
in an offer, but I cannot think of a case in which a server would not
include them these. I suggest changing to "MUST".

4) From section 6:
"Endpoints that use the offer/answer model to establish BFCP
   connections MUST support the 'floorid' and the 'label' attributes.  A
   floor control server acting as an offerer or as an answerer SHOULD
   include these attributes in its session descriptions."


This is inherited form RFC 4583. I cannot think of a case in which a
server would not include these. I suggest changing to "MUST".


5) There are several SHOULDs in section 7, which defines the new bfcpver
attribute. As this attribute did not exist in RFC 4583, the SHOULDs are
warranted for backward compatibility. However, they can be made mandatory
for implementations of BFCP/UDP as well as those compliant with this
update to RFC 4583.

The only should for which this is not the case is the following:

"Tokens representing versions SHOULD be
   integers matching the "Version" field that would be presented in the
   BFCP COMMON-HEADER [8
<http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8>]."


I think this can be changed to "MUST" for all cases.

6) From section 8.1:
"When the existing TCP connection is reset following the rules in [8
<http://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4583bis-08#ref-8>],
   the client SHOULD generate an offer towards the floor control server
   in order to reestablish the connection.  If a TCP connection cannot
   deliver a BFCP message and times out, the entity that attempted to
   send the message (i.e., the one that detected the TCP timeout) SHOULD
   generate an offer in order to reestablish the TCP connection."

I am not sure about these. I think they could be changed to "MUST".
Hopefully others can chime in.

7) Section 9 reads as follows:
"Endpoints that use the offer/
   answer model to establish BFCP connections MUST support the
   'fingerprint' attribute and SHOULD include it in their session
   descriptions."

I think this can be changed to "MUST"; however, I would appreciate
additional opinions on this one as well.

Cheers,
Charles


On 1/10/14 9:34 AM, "Ali C. Begen (abegen)" <abegen@cisco.com> wrote:

>
>On Jan 10, 2014, at 2:50 AM, Tom Kristensen <2mkristensen@gmail.com>
>wrote:
>
>> 3) There are a few "SHOULD"s. I think they are OK but it is not clear
>>when it is OK to ignore that SHOULD. If there are no such use cases,
>>maybe they can be replaced with a "MUST"?
>>=20
>> This is inherited from the original RFC 4583, that's the main reason
>>for me to suggest not changing this in this draft. Hopefully, that's
>>fine?
>
>Well I guess IESG will have issues with this, so take this as a warning :)
>
>>=20
>> -- Tom
>
>_______________________________________________
>mmusic mailing list
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Review (by Albrecht) of draft-ietf-mmusic-udptl-dtls-02
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Date: Tue, 14 Jan 2014 12:46:48 +0000
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Subject: Re: [MMUSIC] Review (by Albrecht) of draft-ietf-mmusic-udptl-dtls-02
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Hi Albrecht,

I am sorry that it took a while to address your comments, but here we go :)

Comment inline. I try to shorten your suggested text, by keeping the essent=
ial parts, and prevent adding long notes.

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

to 3.1<http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-02#section-3=
.1>.  Secure Channel Establishment
      Paragraph:

   In addition to the usual contents of an SDP media description ("m=3D"

   line) specified for UDPTL over the UDP, each SDP media description

   for UDPTL over DTLS over the UDP will also contain several SDP

   attributes, as specified in [RFC4145<http://tools.ietf.org/html/rfc4145>=
] and [RFC4572<http://tools.ietf.org/html/rfc4572>].

proposed to be detailed by:


   In addition to the usual contents of an SDP media description ("m=3D"

   line) specified for UDPTL over the UDP, each SDP media description

   for UDPTL over DTLS over the UDP will also contain several SDP

   attributes, which were introduced in the context of TCP and TLS (but reu=
sed here for DTLS-over-UDP), as specified in [RFC4145<http://tools.ietf.org=
/html/rfc4145>] (TCP) and [RFC4572<http://tools.ietf.org/html/rfc4572>] (TL=
S).

I suggest the following:

In addition to the usual contents of an SDP media description ("m=3D"

      line) specified for UDPTL over UDP, each SDP media description

      for UDPTL over DTLS over UDP will also contain several SDP

      attributes (which were introduced in the context of TCP [RFC4145]

      and TLS [4572], and re-used in this document).



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


List item:

   o  The endpoint MUST use the SDP setup attribute [RFC4145<http://tools.i=
etf.org/html/rfc4145>].  The

      offerer SHOULD assign the SDP setup attribute with a setup:actpass

      value, and MAY assign the SDP setup attribute with a setup:active

      value or setup:passive value.  The offerer MUST NOT assign the SDP

      setup attribute with a setup:holdconn value.  If the offerer ...

proposed to add further background information:

   o  The endpoint MUST use the SDP setup attribute [RFC4145<http://tools.i=
etf.org/html/rfc4145>].  The

      offerer SHOULD assign the SDP setup attribute with a setup:actpass

      value, and MAY assign the SDP setup attribute with a setup:active

      value or setup:passive value (NOTE: these rules allow to decouple the=
 establishment direction of the DTLS session from the establishment directi=
on of the SIP session (which may be demanded due to security considerations=
, NAT traversal aspects, etc.)).
The offerer MUST NOT assign the SDP

      setup attribute with a setup:holdconn value.  If the offerer ....

I suggest to simply say following:

                "In order to negotiate the TLS roles, the endpoint MUST use=
 the SDP setup attribute [RFC4145<http://tools.ietf.org/html/rfc4145>]."

I don't think we need the other text. It's really not the main scope of the=
 document. Other DTLS protocols also use the setup
attribute to determine the TLS roles, and we are simply doing the same thin=
g.

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



to 3.2<http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-02#section-3=
.2>.  Secure Channel Usage

      Paragraph:

   DTLS is used as specified in [RFC6347<http://tools.ietf.org/html/rfc6347=
>].  Once the DTLS handshake is

   completed, the UDPTL packets SHALL be transported in DTLS record

   layer "application_data" packets.

proposed to be detailed by:

   DTLS is used as specified in [RFC6347<http://tools.ietf.org/html/rfc6347=
>].  Once the DTLS handshake is

   successfully completed, the UDPTL packets SHALL be transported in DTLS r=
ecord

   layer "application_data" packets.
NOTE: such a proceeding ensures that the application (i.e., T.38) does not =
start to transmit facsimile data already during an ongoing underlying DTLS =
session establishment. This aspect might be important for end-to-end scenar=
ios with gateway equipment and a possible "UDPTL/DTLS/UDP-to-UDPTL/UDP" int=
erworking function).

I suggest the following:


        "DTLS is used as specified in [RFC6347<http://tools.ietf.org/html/r=
fc6347>].  Once the DTLS handshake is

        successfully completed (in order to prevent facsimile data from bei=
ng

        transmitted insecurely), the UDPTL packets SHALL be transported in =
DTLS

        record layer "application_data" packets."



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

Regards,

Christer


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Albrecht,<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I am sorry that it too=
k a while to address your comments, but here we go :)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Comment inline. I try =
to shorten your suggested text, by keeping the essential parts, and prevent=
 adding long notes.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">-------=
----------------------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
<o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><a name=3D"section-3.1"><b><span lang=3D"EN-GB" styl=
e=3D"font-family:Consolas">to
</span></b></a><span lang=3D"EN-GB"><a href=3D"http://tools.ietf.org/html/d=
raft-ietf-mmusic-udptl-dtls-02#section-3.1"><b><span style=3D"font-family:C=
onsolas">3.1</span></b></a></span><b><span lang=3D"EN-GB" style=3D"font-fam=
ily:Consolas">.&nbsp; Secure Channel Establishment<o:p></o:p></span></b></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paragraph:<o:p></o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp; &nbsp;In addition to the usual contents of=
 an SDP media description (&quot;m=3D&quot;<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; line) specified for UDPTL over the U=
DP, each SDP media description<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; for UDPTL over DTLS over the UDP wil=
l also contain several SDP<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; attributes, as specified in [<a href=
=3D"http://tools.ietf.org/html/rfc4145" title=3D"&quot;TCP-Based Media Tran=
sport in the Session Description Protocol (SDP)&quot;">RFC4145</a>] and [<a=
 href=3D"http://tools.ietf.org/html/rfc4572" title=3D"&quot;Connection-Orie=
nted Media Transport over the Transport Layer Security (TLS) Protocol in th=
e Session Description Protocol (SDP)&quot;">RFC4572</a>].<o:p></o:p></span>=
</pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
<o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:36.0pt"><span lang=3D"EN-GB" st=
yle=3D"font-family:Consolas">proposed to be detailed by:<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
<o:p>&nbsp;</o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; In addition to the usual contents of=
 an SDP media description (&quot;m=3D&quot;<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; line) specified for UDPTL over the U=
DP, each SDP media description<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; for UDPTL over DTLS over the UDP wil=
l also contain several SDP<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; attributes, <span style=3D"color:red=
">which were introduced in the context of TCP and TLS (but reused here for =
DTLS-over-UDP),</span> as specified in [<a href=3D"http://tools.ietf.org/ht=
ml/rfc4145" title=3D"&quot;TCP-Based Media Transport in the Session Descrip=
tion Protocol (SDP)&quot;">RFC4145</a>] <span style=3D"color:red">(TCP)</sp=
an> and [<a href=3D"http://tools.ietf.org/html/rfc4572" title=3D"&quot;Conn=
ection-Oriented Media Transport over the Transport Layer Security (TLS) Pro=
tocol in the Session Description Protocol (SDP)&quot;">RFC4572</a>] <span s=
tyle=3D"color:red">(TLS)</span>.<o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">I sugge=
st the following:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:36.0pt"><span lang=3D"EN-GB">In=
 addition to the usual contents of an SDP media description (&quot;m=3D&quo=
t;</span><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p></o:p></span></p=
>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;line) specified fo=
r UDPTL over UDP, each SDP media description<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;for UDPTL over DTL=
S over UDP will also contain several SDP<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;attributes <span s=
tyle=3D"color:#00B050">(which were introduced in the context of TCP [RFC414=
5] <o:p></o:p></span></span></pre>
<pre><span lang=3D"EN-GB" style=3D"color:#00B050">&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;and TLS [4572], and re-used in this document).<o:p></o:p></span>=
</pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">-------=
----------------------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
List item:<o:p></o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; o&nbsp; The endpoint MUST use the SD=
P setup attribute [<a href=3D"http://tools.ietf.org/html/rfc4145" title=3D"=
&quot;TCP-Based Media Transport in the Session Description Protocol (SDP)&q=
uot;">RFC4145</a>].&nbsp; The<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; offerer SHOULD ass=
ign the SDP setup attribute with a setup:actpass<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; value, and MAY ass=
ign the SDP setup attribute with a setup:active<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; value or setup:pas=
sive value.&nbsp; The offerer MUST NOT assign the SDP<o:p></o:p></span></pr=
e>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; setup attribute wi=
th a setup:holdconn value.&nbsp; If the offerer &#8230;<o:p></o:p></span></=
pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
<o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
proposed to add further background information:<o:p></o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; o&nbsp; The endpoint MUST use the SD=
P setup attribute [<a href=3D"http://tools.ietf.org/html/rfc4145" title=3D"=
&quot;TCP-Based Media Transport in the Session Description Protocol (SDP)&q=
uot;">RFC4145</a>].&nbsp; The<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; offerer SHOULD ass=
ign the SDP setup attribute with a setup:actpass<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; value, and MAY ass=
ign the SDP setup attribute with a setup:active<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; value or setup:pas=
sive value <span style=3D"color:red">(NOTE: these rules allow to decouple t=
he establishment direction of the DTLS session from the establishment direc=
tion of the SIP session (which may be demanded due to security consideratio=
ns, NAT traversal aspects, etc.)).</span>&nbsp; <br>The offerer MUST NOT as=
sign the SDP<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; setup attribute wi=
th a setup:holdconn value.&nbsp; If the offerer &#8230;.<o:p></o:p></span><=
/pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">I sugge=
st to simply say following:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; &#8220;</span><span lang=3D"EN-GB" style=3D"color:#00B050">In orde=
r to negotiate the TLS roles,</span><span lang=3D"EN-GB" style=3D"color:#1F=
497D">
</span><span lang=3D"EN-GB">the endpoint MUST use the SDP setup attribute [=
<a href=3D"http://tools.ietf.org/html/rfc4145" title=3D"&quot;TCP-Based Med=
ia Transport in the Session Description Protocol (SDP)&quot;">RFC4145</a>].=
&#8221;</span><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I don&#8217;t think we=
 need the other text. It's really not the main scope of the document. Other=
 DTLS protocols also use the setup
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">attribute to determine=
 the TLS roles, and we are simply doing the same thing.<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-family:Consolas;color:#1F497D"><=
o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">-------=
----------------------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-GB" style=3D"font-family:Consola=
s">to <a name=3D"section-3.2">
</a><a href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-02#s=
ection-3.2">3.2</a>.&nbsp; Secure Channel Usage<o:p></o:p></span></b></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-GB" style=3D"font-family:Consola=
s"><o:p>&nbsp;</o:p></span></b></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paragraph:<o:p></o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; DTLS is used as specified in [<a hre=
f=3D"http://tools.ietf.org/html/rfc6347" title=3D"&quot;Datagram Transport =
Layer Security Version 1.2&quot;">RFC6347</a>].&nbsp; Once the DTLS handsha=
ke is<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; completed, the UDPTL packets SHALL b=
e transported in DTLS record<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; layer &quot;application_data&quot; p=
ackets.<o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas">=
<o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:36.0pt"><span lang=3D"EN-GB" st=
yle=3D"font-family:Consolas">proposed to be detailed by:<o:p></o:p></span><=
/p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; DTLS is used as specified in [<a hre=
f=3D"http://tools.ietf.org/html/rfc6347" title=3D"&quot;Datagram Transport =
Layer Security Version 1.2&quot;">RFC6347</a>].&nbsp; Once the DTLS handsha=
ke is<o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; <span style=3D"color:red">successful=
ly</span> completed, the UDPTL packets SHALL be transported in DTLS record<=
o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp; layer &quot;application_data&quot; p=
ackets.<br><span style=3D"color:red">NOTE: such a proceeding ensures that t=
he application (i.e., T.38) does not start to transmit facsimile data alrea=
dy during an ongoing underlying DTLS session establishment. This aspect mig=
ht be important for end-to-end scenarios with gateway equipment and a possi=
ble &#8220;UDPTL/DTLS/UDP-to-UDPTL/UDP&#8221; interworking function).</span=
><o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">I sugge=
st the following:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<pre><span lang=3D"EN-GB">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <span =
style=3D"color:#00B050">&#8220;DTLS is used as specified in [<a href=3D"htt=
p://tools.ietf.org/html/rfc6347" title=3D"&quot;Datagram Transport Layer Se=
curity Version 1.2&quot;"><span style=3D"color:#00B050">RFC6347</span></a>]=
.&nbsp; Once the DTLS handshake is<o:p></o:p></span></span></pre>
<pre><span lang=3D"EN-GB" style=3D"color:#00B050">&nbsp;&nbsp; &nbsp;&nbsp;=
&nbsp;&nbsp; successfully completed (in order to prevent facsimile data fro=
m being <o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB" style=3D"color:#00B050">&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; transmitted insecurely), the UDPTL packets SHALL be trans=
ported in DTLS <o:p></o:p></span></pre>
<pre><span lang=3D"EN-GB" style=3D"color:#00B050">&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; record layer &quot;application_data&quot; packets.&#8221;=
<o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"font-family:Consolas;c=
olor:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">-------=
----------------------<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">Regards=
,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D">Christe=
r<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-GB" style=3D"color:#1F497D"><o:p>&n=
bsp;</o:p></span></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C5F9C35ESESSMB209erics_--

From abegen@cisco.com  Tue Jan 14 11:49:44 2014
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "<mmusic@ietf.org>" <mmusic@ietf.org>
Thread-Topic: Issue tracker for 4566bis
Thread-Index: AQHPEWG8F/foggAQEEyGL+H7582LcA==
Date: Tue, 14 Jan 2014 19:49:28 +0000
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Subject: [MMUSIC] Issue tracker for 4566bis
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Hi everyone

Revising 4566 is underway as you know and there have been some issues with =
4566 and some folks want them to be fixed in 4566bis. However, due to convo=
luted discussions, some of these issues go unfixed. To make the process eas=
ier, it was suggested to establish an issue tracker.

I have not done this before, but found the link below:
http://tools.ietf.org/wg/mmusic/trac/report/1

Is this a good idea to use this or do you have better suggestions?

Thanks,
-acbegen


From pkyzivat@alum.mit.edu  Wed Jan 15 10:24:36 2014
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] Issue tracker for 4566bis
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On 1/14/14 2:49 PM, Ali C. Begen (abegen) wrote:
> Hi everyone
>
> Revising 4566 is underway as you know and there have been some issues with 4566 and some folks want them to be fixed in 4566bis. However, due to convoluted discussions, some of these issues go unfixed. To make the process easier, it was suggested to establish an issue tracker.
>
> I have not done this before, but found the link below:
> http://tools.ietf.org/wg/mmusic/trac/report/1
>
> Is this a good idea to use this or do you have better suggestions?

This is just a link to a report of active tickets.
Are you just asking if doing something analogous for 4566bis would be a 
good thing?

I think the issue tracker is fine if you, as editor, use it to track the 
issues with your document.

But I strongly suggest that you be the only one to open, update, and 
close tickets for 4566bis. Don't leave it for anybody who thinks they 
have an issue to open their own ticket, and don't use updates to the 
tickets as a way to debate an issue. I saw that done once or twice and 
it created a mess - confusing established mechanisms for discussing 
issues via email.

So IMO we should continue to *discuss* proposed changes on the mmusic 
list. Based on those discussions you can make tickets as you see fit to 
track the status of identified issues and their incorporation into the 
draft.

	Thanks,
	Paul


From internet-drafts@ietf.org  Wed Jan 15 14:34:20 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-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           : Interactive Connectivity Establishment (ICE): A P=
rotocol for Network Address Translator (NAT) Traversal for Offer/Answer Pro=
tocols
        Authors         : Ari Keranen
                          Jonathan Rosenberg
	Filename        : draft-ietf-mmusic-rfc5245bis-01.txt
	Pages           : 87
	Date            : 2014-01-15

Abstract:
   This document describes a protocol for Network Address Translator
   (NAT) traversal for UDP-based multimedia sessions established with
   the offer/answer model.  This protocol is called Interactive
   Connectivity Establishment (ICE).  ICE makes use of the Session
   Traversal Utilities for NAT (STUN) protocol and its extension,
   Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
   utilizing the offer/answer model, such as the Session Initiation
   Protocol (SIP).

   This document obsoletes RFC 5245.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc5245bis/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc5245bis-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rfc5245bis-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 abegen@cisco.com  Thu Jan 16 09:37:43 2014
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] Issue tracker for 4566bis
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Subject: Re: [MMUSIC] Issue tracker for 4566bis
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On Jan 15, 2014, at 10:24 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 1/14/14 2:49 PM, Ali C. Begen (abegen) wrote:
>> Hi everyone
>>=20
>> Revising 4566 is underway as you know and there have been some issues wi=
th 4566 and some folks want them to be fixed in 4566bis. However, due to co=
nvoluted discussions, some of these issues go unfixed. To make the process =
easier, it was suggested to establish an issue tracker.
>>=20
>> I have not done this before, but found the link below:
>> http://tools.ietf.org/wg/mmusic/trac/report/1
>>=20
>> Is this a good idea to use this or do you have better suggestions?
>=20
> This is just a link to a report of active tickets.
> Are you just asking if doing something analogous for 4566bis would be a g=
ood thing?
>=20
> I think the issue tracker is fine if you, as editor, use it to track the =
issues with your document.
>=20
> But I strongly suggest that you be the only one to open, update, and clos=
e tickets for 4566bis. Don't leave it for anybody who thinks they have an i=
ssue to open their own ticket, and don't use updates to the tickets as a wa=
y to debate an issue. I saw that done once or twice and it created a mess -=
 confusing established mechanisms for discussing issues via email.
>=20
> So IMO we should continue to *discuss* proposed changes on the mmusic lis=
t. Based on those discussions you can make tickets as you see fit to track =
the status of identified issues and their incorporation into the draft.

Fine with me. I will try to create and track the issues in that list (so ev=
erybody can know where we are) and people should post their comments about =
the 4566bis in the list. But, please be clear on what the problem is and pr=
ovide a solution if you can so that we can move faster.

thanks.
-acbegen



>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Thu Jan 16 14:28:36 2014
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Subject: Re: [MMUSIC] Issue tracker for 4566bis
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Ali,

I'll be interested to see the set of issues you post initially.
I've raised things from time to time in the past, and you may already 
have them on your private list.

	Thanks,
	Paul

On 1/16/14 12:37 PM, Ali C. Begen (abegen) wrote:
>
> On Jan 15, 2014, at 10:24 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 1/14/14 2:49 PM, Ali C. Begen (abegen) wrote:
>>> Hi everyone
>>>
>>> Revising 4566 is underway as you know and there have been some issues with 4566 and some folks want them to be fixed in 4566bis. However, due to convoluted discussions, some of these issues go unfixed. To make the process easier, it was suggested to establish an issue tracker.
>>>
>>> I have not done this before, but found the link below:
>>> http://tools.ietf.org/wg/mmusic/trac/report/1
>>>
>>> Is this a good idea to use this or do you have better suggestions?
>>
>> This is just a link to a report of active tickets.
>> Are you just asking if doing something analogous for 4566bis would be a good thing?
>>
>> I think the issue tracker is fine if you, as editor, use it to track the issues with your document.
>>
>> But I strongly suggest that you be the only one to open, update, and close tickets for 4566bis. Don't leave it for anybody who thinks they have an issue to open their own ticket, and don't use updates to the tickets as a way to debate an issue. I saw that done once or twice and it created a mess - confusing established mechanisms for discussing issues via email.
>>
>> So IMO we should continue to *discuss* proposed changes on the mmusic list. Based on those discussions you can make tickets as you see fit to track the status of identified issues and their incorporation into the draft.
>
> Fine with me. I will try to create and track the issues in that list (so everybody can know where we are) and people should post their comments about the 4566bis in the list. But, please be clear on what the problem is and provide a solution if you can so that we can move faster.
>
> thanks.
> -acbegen
>
>
>
>>
>> 	Thanks,
>> 	Paul
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>


From internet-drafts@ietf.org  Fri Jan 17 02:29:20 2014
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Date: Fri, 17 Jan 2014 02:29:19 -0800
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-udptl-dtls-03.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           : UDP Transport Layer (UDPTL) over Datagram Transpo=
rt Layer Security (DTLS)
        Authors         : Christer Holmberg
                          Ivo Sedlacek
                          Gonzalo Salgueiro
	Filename        : draft-ietf-mmusic-udptl-dtls-03.txt
	Pages           : 21
	Date            : 2014-01-17

Abstract:
   This document specifies how the UDP Transport Layer (UDPTL) protocol,
   the predominant transport protocol for T.38 fax, can be transported
   over the Datagram Transport Layer Security (DTLS) protocol, how the
   usage of UDPTL over DTLS is indicated in the Session Description
   Protocol (SDP), and how UDPTL over DTLS is negotiated in a session
   established using the Session Initiation Protocol (SIP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-udptl-dtls/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-udptl-dtls-03


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 christer.holmberg@ericsson.com  Fri Jan 17 02:33:48 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: Draft new version:  draft-ietf-mmusic-udptl-dtls-03
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Subject: [MMUSIC] Draft new version:  draft-ietf-mmusic-udptl-dtls-03
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Hi fax lovers,

Based on the excellent and detailed document reviews performed by James R, =
David H, Albrecht S and Oscar O, we've submitted a new version (-03) of dra=
ft-ietf-mmusic-udptl-dtls.

Regards,

Christer

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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
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<p class=3D"MsoNormal"><span lang=3D"FI">Hi fax lovers,<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">Based on the excellent and detailed document reviews=
 performed by James R, David H, Albrecht S and Oscar O, we&#8217;ve submitt=
ed a new version (-03) of draft-ietf-mmusic-udptl-dtls.<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_7594FB04B1934943A5C02806D1A2204B1D0FDD10ESESSMB209erics_--

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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc2326bis-39.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           : Real Time Streaming Protocol 2.0 (RTSP)
        Authors         : Henning Schulzrinne
                          Anup Rao
                          Rob Lanphier
                          Magnus Westerlund
                          Martin Stiemerling
	Filename        : draft-ietf-mmusic-rfc2326bis-39.txt
	Pages           : 309
	Date            : 2014-01-17

Abstract:
   This memorandum defines RTSP version 2.0 which obsoletes RTSP version
   1.0 defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-layer
   protocol for setup and control of the delivery of data with real-time
   properties.  RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video.  Sources of data can include both live data feeds and stored
   clips.  This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc2326bis/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-39

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rfc2326bis-39


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 magnus.westerlund@ericsson.com  Fri Jan 17 03:04:22 2014
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WG,

Martin and I have now worked through the discusses and comments received
in IESG review. Thus this is our suggestion on how to resolve them. I
encourage you to review the changes while the ADs reviews the discuss
positions. If you find issues we need to hear comments ASAP.

The main changes are:

- rewritten structure for a large number of IANA entries

- Additional text regarding security and sensitivity of control vs media.

- ABNF chagnes regarding UTF-8

- Rewritten NPT text and ABNF.

- Updated examples for Date to follow the time format actually specified.

Cheers

Magnus

On 2014-01-17 11:58, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
>         Title           : Real Time Streaming Protocol 2.0 (RTSP)
>         Authors         : Henning Schulzrinne
>                           Anup Rao
>                           Rob Lanphier
>                           Magnus Westerlund
>                           Martin Stiemerling
> 	Filename        : draft-ietf-mmusic-rfc2326bis-39.txt
> 	Pages           : 309
> 	Date            : 2014-01-17
> 
> Abstract:
>    This memorandum defines RTSP version 2.0 which obsoletes RTSP version
>    1.0 defined in RFC 2326.
> 
>    The Real Time Streaming Protocol, or RTSP, is an application-layer
>    protocol for setup and control of the delivery of data with real-time
>    properties.  RTSP provides an extensible framework to enable
>    controlled, on-demand delivery of real-time data, such as audio and
>    video.  Sources of data can include both live data feeds and stored
>    clips.  This protocol is intended to control multiple data delivery
>    sessions, provide a means for choosing delivery channels such as UDP,
>    multicast UDP and TCP, and provide a means for choosing delivery
>    mechanisms based upon RTP (RFC 3550).
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc2326bis/
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-39
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rfc2326bis-39
> 
> 
> 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/
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From dhanes@cisco.com  Fri Jan 17 06:33:02 2014
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From: "David Hanes (dhanes)" <dhanes@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Draft new version:  draft-ietf-mmusic-udptl-dtls-03
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--_000_CEFEA89ED391dhanesciscocom_
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Hi Christer,

This new draft looks good and addresses the concerns that I raised.

Thanks,
David



From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Friday, January 17, 2014 5:33 AM
To: "mmusic@ietf.org<mailto:mmusic@ietf.org>" <mmusic@ietf.org<mailto:mmusi=
c@ietf.org>>
Cc: "mmusic-chairs@tools.ietf.org<mailto:mmusic-chairs@tools.ietf.org>" <mm=
usic-chairs@tools.ietf.org<mailto:mmusic-chairs@tools.ietf.org>>
Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-03

Hi fax lovers,

Based on the excellent and detailed document reviews performed by James R, =
David H, Albrecht S and Oscar O, we=92ve submitted a new version (-03) of d=
raft-ietf-mmusic-udptl-dtls.

Regards,

Christer

--_000_CEFEA89ED391dhanesciscocom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>Hi Christer,</div>
<div><br>
</div>
<div>This new draft looks good and addresses the concerns that I raised.&nb=
sp;</div>
<div><br>
</div>
<div>Thanks,</div>
<div>David</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
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 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, January 17, 2014 5:33=
 AM<br>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:mmusic@=
ietf.org">mmusic@ietf.org</a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org">=
mmusic@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:mmusic-=
chairs@tools.ietf.org">mmusic-chairs@tools.ietf.org</a>&quot; &lt;<a href=
=3D"mailto:mmusic-chairs@tools.ietf.org">mmusic-chairs@tools.ietf.org</a>&g=
t;<br>
<span style=3D"font-weight:bold">Subject: </span>[MMUSIC] Draft new version=
: draft-ietf-mmusic-udptl-dtls-03<br>
</div>
<div><br>
</div>
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<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"FI">Hi fax lovers,<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">Based on the excellent and detailed document reviews=
 performed by James R, David H, Albrecht S and Oscar O, we=92ve submitted a=
 new version (-03) of draft-ietf-mmusic-udptl-dtls.<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>
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From gsalguei@cisco.com  Fri Jan 17 11:43:32 2014
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: John Shaheen <John.Shaheen@dialogic.com>
Thread-Topic: Working Group Last Call Review of UDPTL over DTLS Draft
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Date: Fri, 17 Jan 2014 19:43:17 +0000
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Cc: "foip@sipforum.org" <FoIP@sipforum.org>, mmusic <mmusic@ietf.org>, Sebastien BOIRE-LAVIGNE <Sebastien.Boire-Lavigne@sagemcom.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Working Group Last Call Review of UDPTL over DTLS Draft
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[Copying the MMUSIC list for awareness...]

John -=20

I appreciate you taking the time to review the draft.

Cheers,

Gonzalo



On Jan 17, 2014, at 2:39 PM, John Shaheen <John.Shaheen@dialogic.com> wrote=
:

> Hi Gonzalo,
> I have reviewed the UDPTL over DTLS draft with my team and we do not see =
any issues.=20
> =20
> Regards,
> John Shaheen
> =20
> John G. Shaheen III
> Engineering Manager, Brooktrout
> Dialogic, Inc.
>=20
> Tel:       781.433.9514
> Fax:      781.433.9255
> Email:    john.shaheen@dialogic.com
> =20
> <image001.png>
> =20
> <image002.png> <image003.png> <image004.png> <image005.png> <image006.png=
> <image007.png> <image008.png>=20
> =20
> This e-mail is intended only for the named recipient(s) and may contain i=
nformation that is privileged, confidential and/or exempt from disclosure u=
nder applicable law. No waiver of privilege, confidence or otherwise is int=
ended by virtue of communication via the internet. Any unauthorized use,dis=
semination or copying is strictly prohibited. If you have received this e-m=
ail in error, or are not named as a recipient, please immediately notify th=
e sender and destroy all copies of this e-mail.
> =20
> From: foip-bounces@sipforum.org [mailto:foip-bounces@sipforum.org] On Beh=
alf Of Gonzalo Salgueiro (gsalguei)
> Sent: Tuesday, December 10, 2013 12:57 PM
> To: foip@sipforum.org; Sebastien BOIRE-LAVIGNE
> Cc: Ivo Sedlacek; Christer Holmberg
> Subject: [Foip] Working Group Last Call Review of UDPTL over DTLS Draft
> =20
> FoIP TG -=20
> =20
> The authors believe that the UDPTL over DTLS draft is ready to be moved f=
orward toward publication. For that to happen, we would need folks to perfo=
rm a detailed review of the draft, and indicate on the list (mmusic@ietf.or=
g) about any possible issues.  Feedback indicating that you have read the d=
raft and that everything looks good is even helpful.  The draft can be foun=
d here:
> =20
> http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls
>=20
> The authors would appreciate it if folks from this group would be willing=
 to perform such a review.
>=20
> Thanks!
>=20
> Gonzalo


From christer.holmberg@ericsson.com  Fri Jan 17 21:35:01 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, John Shaheen <John.Shaheen@dialogic.com>
Thread-Topic: Working Group Last Call Review of UDPTL over DTLS Draft
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Subject: Re: [MMUSIC] Working Group Last Call Review of UDPTL over DTLS Draft
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Thanks, John! :)

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

"Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com> wrote:


[Copying the MMUSIC list for awareness...]

John -

I appreciate you taking the time to review the draft.

Cheers,

Gonzalo



On Jan 17, 2014, at 2:39 PM, John Shaheen <John.Shaheen@dialogic.com> wrote=
:

> Hi Gonzalo,
> I have reviewed the UDPTL over DTLS draft with my team and we do not see =
any issues.
>
> Regards,
> John Shaheen
>
> John G. Shaheen III
> Engineering Manager, Brooktrout
> Dialogic, Inc.
>
> Tel:       781.433.9514
> Fax:      781.433.9255
> Email:    john.shaheen@dialogic.com
>
> <image001.png>
>
> <image002.png> <image003.png> <image004.png> <image005.png> <image006.png=
> <image007.png> <image008.png>
>
> This e-mail is intended only for the named recipient(s) and may contain i=
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ended by virtue of communication via the internet. Any unauthorized use,dis=
semination or copying is strictly prohibited. If you have received this e-m=
ail in error, or are not named as a recipient, please immediately notify th=
e sender and destroy all copies of this e-mail.
>
> From: foip-bounces@sipforum.org [mailto:foip-bounces@sipforum.org] On Beh=
alf Of Gonzalo Salgueiro (gsalguei)
> Sent: Tuesday, December 10, 2013 12:57 PM
> To: foip@sipforum.org; Sebastien BOIRE-LAVIGNE
> Cc: Ivo Sedlacek; Christer Holmberg
> Subject: [Foip] Working Group Last Call Review of UDPTL over DTLS Draft
>
> FoIP TG -
>
> The authors believe that the UDPTL over DTLS draft is ready to be moved f=
orward toward publication. For that to happen, we would need folks to perfo=
rm a detailed review of the draft, and indicate on the list (mmusic@ietf.or=
g) about any possible issues.  Feedback indicating that you have read the d=
raft and that everything looks good is even helpful.  The draft can be foun=
d here:
>
> http://tools.ietf.org/html/draft-ietf-mmusic-udptl-dtls
>
> The authors would appreciate it if folks from this group would be willing=
 to perform such a review.
>
> Thanks!
>
> Gonzalo


From christer.holmberg@ericsson.com  Sun Jan 19 23:05:18 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Shida Schubert <shida@agnada.com>, "David Hanes (dhanes)" <dhanes@cisco.com>
Thread-Topic: [MMUSIC] Draft new version:  draft-ietf-mmusic-udptl-dtls-03
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Thanks Shida! :)

Regards,

Christer

L=E4hett=E4j=E4: Shida Schubert [mailto:shida@agnada.com]
L=E4hetetty: 20. tammikuuta 2014 8:59
Vastaanottaja: David Hanes (dhanes)
Kopio: Christer Holmberg; mmusic@ietf.org; mmusic-chairs@tools.ietf.org
Aihe: Re: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-03


 I read the previous draft and didn't see any issues beside
raised by others.

 I glanced over the new draft and the changes look good.

 Regards
  Shida

On Jan 17, 2014, at 6:32 AM, David Hanes (dhanes) <dhanes@cisco.com<mailto:=
dhanes@cisco.com>> wrote:


Hi Christer,

This new draft looks good and addresses the concerns that I raised.

Thanks,
David



From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Friday, January 17, 2014 5:33 AM
To: "mmusic@ietf.org<mailto:mmusic@ietf.org>" <mmusic@ietf.org<mailto:mmusi=
c@ietf.org>>
Cc: "mmusic-chairs@tools.ietf.org<mailto:mmusic-chairs@tools.ietf.org>" <mm=
usic-chairs@tools.ietf.org<mailto:mmusic-chairs@tools.ietf.org>>
Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-03

Hi fax lovers,

Based on the excellent and detailed document reviews performed by James R, =
David H, Albrecht S and Oscar O, we've submitted a new version (-03) of dra=
ft-ietf-mmusic-udptl-dtls.

Regards,

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


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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">Thanks Shida!
</span><span style=3D"font-size:11.0pt;font-family:Wingdings;color:#1F497D"=
>J</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D"><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">Regards,<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">Christer<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>
<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;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> Shida Schubert [mailto:shida@agnada.com]
<br>
<b>L=E4hetetty:</b> 20. tammikuuta 2014 8:59<br>
<b>Vastaanottaja:</b> David Hanes (dhanes)<br>
<b>Kopio:</b> Christer Holmberg; mmusic@ietf.org; mmusic-chairs@tools.ietf.=
org<br>
<b>Aihe:</b> Re: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-0=
3<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;I read the previous draft and didn't see any i=
ssues beside&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">raised by others.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;I glanced over the new draft and the changes l=
ook good.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;Regards<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; Shida<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Jan 17, 2014, at 6:32 AM, David Hanes (dhanes) &l=
t;<a href=3D"mailto:dhanes@cisco.com">dhanes@cisco.com</a>&gt; wrote:<o:p><=
/o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi Christer,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">This new draft looks good and addresses=
 the concerns that I raised.&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Thanks,<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">David<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;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 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">From:<span class=3D"apple-converted-=
space">&nbsp;</span></span></b><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;">Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com"><span style=3D"color:purple">chr=
ister.holmberg@ericsson.com</span></a>&gt;<br>
<b>Date:<span class=3D"apple-converted-space">&nbsp;</span></b>Friday, Janu=
ary 17, 2014 5:33 AM<br>
<b>To:<span class=3D"apple-converted-space">&nbsp;</span></b>&quot;<a href=
=3D"mailto:mmusic@ietf.org"><span style=3D"color:purple">mmusic@ietf.org</s=
pan></a>&quot; &lt;<a href=3D"mailto:mmusic@ietf.org"><span style=3D"color:=
purple">mmusic@ietf.org</span></a>&gt;<br>
<b>Cc:<span class=3D"apple-converted-space">&nbsp;</span></b>&quot;<a href=
=3D"mailto:mmusic-chairs@tools.ietf.org"><span style=3D"color:purple">mmusi=
c-chairs@tools.ietf.org</span></a>&quot; &lt;<a href=3D"mailto:mmusic-chair=
s@tools.ietf.org"><span style=3D"color:purple">mmusic-chairs@tools.ietf.org=
</span></a>&gt;<br>
<b>Subject:<span class=3D"apple-converted-space">&nbsp;</span></b>[MMUSIC] =
Draft new version: draft-ietf-mmusic-udptl-dtls-03<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">Hi fax lovers,</span><span lang=3D"EN-U=
S" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-ser=
if&quot;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><span lang=3D"EN-US" style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
"><o:p></o:p></span></p>
</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;">Based on the excellent a=
nd detailed document reviews performed by James R, David H, Albrecht S and =
Oscar O, we&#8217;ve submitted a new version (-03) of draft-ietf-mmusic-udp=
tl-dtls.<o:p></o:p></span></p>
</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;">&nbsp;<o:p></o:p></span>=
</p>
</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;">Regards,<o:p></o:p></spa=
n></p>
</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;">&nbsp;<o:p></o:p></span>=
</p>
</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;">Christer<o:p></o:p></spa=
n></p>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">_______________________________________=
________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org"><span style=3D"color:purple">mmusic@ietf=
.org</span></a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic"><span style=3D"col=
or:purple">https://www.ietf.org/mailman/listinfo/mmusic</span></a><o:p></o:=
p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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 I read the previous draft and didn't see any issues beside=20
raised by others.=20

 I glanced over the new draft and the changes look good.=20

 Regards
  Shida

On Jan 17, 2014, at 6:32 AM, David Hanes (dhanes) <dhanes@cisco.com> =
wrote:

> Hi Christer,
>=20
> This new draft looks good and addresses the concerns that I raised.=20
>=20
> Thanks,
> David
>=20
>=20
>=20
> From: Christer Holmberg <christer.holmberg@ericsson.com>
> Date: Friday, January 17, 2014 5:33 AM
> To: "mmusic@ietf.org" <mmusic@ietf.org>
> Cc: "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>
> Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-03
>=20
> Hi fax lovers,
> =20
> Based on the excellent and detailed document reviews performed by =
James R, David H, Albrecht S and Oscar O, we=92ve submitted a new =
version (-03) of draft-ietf-mmusic-udptl-dtls.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


--Apple-Mail=_830E5732-9BF4-4A72-957F-DC8862A385CD
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	charset=windows-1252

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div><br></div><div>&nbsp;I read the previous draft =
and didn't see any issues beside&nbsp;</div><div>raised by =
others.&nbsp;</div><div><br></div><div>&nbsp;I glanced over the new =
draft and the changes look =
good.&nbsp;</div><div><br></div><div>&nbsp;Regards</div><div>&nbsp; =
Shida</div><div><br></div><div><div>On Jan 17, 2014, at 6:32 AM, David =
Hanes (dhanes) &lt;<a =
href=3D"mailto:dhanes@cisco.com">dhanes@cisco.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"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; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
font-size: 14px; font-family: Calibri, sans-serif;"><div>Hi =
Christer,</div><div><br></div><div>This new draft looks good and =
addresses the concerns that I =
raised.&nbsp;</div><div><br></div><div>Thanks,</div><div>David</div><div><=
br></div><div><br></div><div><br></div><span =
id=3D"OLK_SRC_BODY_SECTION"><div style=3D"font-family: Calibri; =
font-size: 11pt; text-align: left; border-width: 1pt medium medium; =
border-style: solid none none; padding: 3pt 0in 0in; border-top-color: =
rgb(181, 196, 223);"><span style=3D"font-weight: bold;">From:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>Christer Holmberg =
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" style=3D"color: =
purple; text-decoration: =
underline;">christer.holmberg@ericsson.com</a>&gt;<br><span =
style=3D"font-weight: bold;">Date:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>Friday, January 17, =
2014 5:33 AM<br><span style=3D"font-weight: bold;">To:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>"<a =
href=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a>" &lt;<a href=3D"mailto:mmusic@ietf.org" =
style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a>&gt;<br><span style=3D"font-weight: =
bold;">Cc:<span class=3D"Apple-converted-space">&nbsp;</span></span>"<a =
href=3D"mailto:mmusic-chairs@tools.ietf.org" style=3D"color: purple; =
text-decoration: underline;">mmusic-chairs@tools.ietf.org</a>" &lt;<a =
href=3D"mailto:mmusic-chairs@tools.ietf.org" style=3D"color: purple; =
text-decoration: =
underline;">mmusic-chairs@tools.ietf.org</a>&gt;<br><span =
style=3D"font-weight: bold;">Subject:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>[MMUSIC] Draft new =
version: draft-ietf-mmusic-udptl-dtls-03<br></div><div><br></div><div =
xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" =
xmlns=3D"http://www.w3.org/TR/REC-html40"><div lang=3D"EN-US" =
link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" style=3D"page: =
WordSection1;"><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif;"><span lang=3D"FI">Hi fax =
lovers,<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif;"><span =
lang=3D"FI"><o:p>&nbsp;</o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;">Based on =
the excellent and detailed document reviews performed by James R, David =
H, Albrecht S and Oscar O, we=92ve submitted a new version (-03) of =
draft-ietf-mmusic-udptl-dtls.<o:p></o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;"><o:p>&nbsp;</o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;">Regards,<o:p></o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;"><o:p>&nbsp;</o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;">Christer<o:p></o:p></div></div></div></div></span>___________=
____________________________________<br>mmusic mailing list<br><a =
href=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a><br><a =
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purple; text-decoration: =
underline;">https://www.ietf.org/mailman/listinfo/mmusic</a></div></blockq=
uote></div><br></body></html>=

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From: Shida Schubert <shida@agnada.com>
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Cc: "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Draft new version:  draft-ietf-mmusic-udptl-dtls-03
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 I read the previous draft and didn't see any issues beside=20
raised by others.=20

 I glanced over the new draft and the changes look good.=20

 Regards
  Shida

On Jan 17, 2014, at 6:32 AM, David Hanes (dhanes) <dhanes@cisco.com> =
wrote:

> Hi Christer,
>=20
> This new draft looks good and addresses the concerns that I raised.=20
>=20
> Thanks,
> David
>=20
>=20
>=20
> From: Christer Holmberg <christer.holmberg@ericsson.com>
> Date: Friday, January 17, 2014 5:33 AM
> To: "mmusic@ietf.org" <mmusic@ietf.org>
> Cc: "mmusic-chairs@tools.ietf.org" <mmusic-chairs@tools.ietf.org>
> Subject: [MMUSIC] Draft new version: draft-ietf-mmusic-udptl-dtls-03
>=20
> Hi fax lovers,
> =20
> Based on the excellent and detailed document reviews performed by =
James R, David H, Albrecht S and Oscar O, we=92ve submitted a new =
version (-03) of draft-ietf-mmusic-udptl-dtls.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div><br></div><div>&nbsp;I read the previous draft =
and didn't see any issues beside&nbsp;</div><div>raised by =
others.&nbsp;</div><div><br></div><div>&nbsp;I glanced over the new =
draft and the changes look =
good.&nbsp;</div><div><br></div><div>&nbsp;Regards</div><div>&nbsp; =
Shida</div><br><div><div>On Jan 17, 2014, at 6:32 AM, David Hanes =
(dhanes) &lt;<a href=3D"mailto:dhanes@cisco.com">dhanes@cisco.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"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; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
font-size: 14px; font-family: Calibri, sans-serif;"><div>Hi =
Christer,</div><div><br></div><div>This new draft looks good and =
addresses the concerns that I =
raised.&nbsp;</div><div><br></div><div>Thanks,</div><div>David</div><div><=
br></div><div><br></div><div><br></div><span =
id=3D"OLK_SRC_BODY_SECTION"><div style=3D"font-family: Calibri; =
font-size: 11pt; text-align: left; border-width: 1pt medium medium; =
border-style: solid none none; padding: 3pt 0in 0in; border-top-color: =
rgb(181, 196, 223);"><span style=3D"font-weight: bold;">From:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>Christer Holmberg =
&lt;<a href=3D"mailto:christer.holmberg@ericsson.com" style=3D"color: =
purple; text-decoration: =
underline;">christer.holmberg@ericsson.com</a>&gt;<br><span =
style=3D"font-weight: bold;">Date:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>Friday, January 17, =
2014 5:33 AM<br><span style=3D"font-weight: bold;">To:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>"<a =
href=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a>" &lt;<a href=3D"mailto:mmusic@ietf.org" =
style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a>&gt;<br><span style=3D"font-weight: =
bold;">Cc:<span class=3D"Apple-converted-space">&nbsp;</span></span>"<a =
href=3D"mailto:mmusic-chairs@tools.ietf.org" style=3D"color: purple; =
text-decoration: underline;">mmusic-chairs@tools.ietf.org</a>" &lt;<a =
href=3D"mailto:mmusic-chairs@tools.ietf.org" style=3D"color: purple; =
text-decoration: =
underline;">mmusic-chairs@tools.ietf.org</a>&gt;<br><span =
style=3D"font-weight: bold;">Subject:<span =
class=3D"Apple-converted-space">&nbsp;</span></span>[MMUSIC] Draft new =
version: draft-ietf-mmusic-udptl-dtls-03<br></div><div><br></div><div =
xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" =
xmlns=3D"http://www.w3.org/TR/REC-html40"><div lang=3D"EN-US" =
link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" style=3D"page: =
WordSection1;"><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif;"><span lang=3D"FI">Hi fax =
lovers,<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif;"><span =
lang=3D"FI"><o:p>&nbsp;</o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;">Based on =
the excellent and detailed document reviews performed by James R, David =
H, Albrecht S and Oscar O, we=92ve submitted a new version (-03) of =
draft-ietf-mmusic-udptl-dtls.<o:p></o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;"><o:p>&nbsp;</o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;">Regards,<o:p></o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;"><o:p>&nbsp;</o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif;">Christer<o:p></o:p></div></div></div></div></span>___________=
____________________________________<br>mmusic mailing list<br><a =
href=3D"mailto:mmusic@ietf.org" style=3D"color: purple; text-decoration: =
underline;">mmusic@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic" style=3D"color: =
purple; text-decoration: =
underline;">https://www.ietf.org/mailman/listinfo/mmusic</a></div></blockq=
uote></div><br></body></html>=

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Subject: [MMUSIC] RFC 7104 on Duplication Grouping Semantics in the Session Description Protocol
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7104

        Title:      Duplication Grouping Semantics in the 
                    Session Description Protocol 
        Author:     A. Begen, Y. Cai,
                    H. Ou
        Status:     Standards Track
        Stream:     IETF
        Date:       January 2014
        Mailbox:    abegen@cisco.com, 
                    yiqunc@microsoft.com, 
                    hou@cisco.com
        Pages:      10
        Characters: 17719
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-mmusic-duplication-grouping-04.txt

        URL:        http://www.rfc-editor.org/rfc/rfc7104.txt

Packet loss is undesirable for real-time multimedia sessions, but it
can occur due to congestion or other unplanned network outages.  This
is especially true for IP multicast networks, where packet loss
patterns can vary greatly between receivers.  One technique that can
be used to recover from packet loss without incurring unbounded delay
for all the receivers is to duplicate the packets and send them in
separate redundant streams.  This document defines the semantics for
grouping redundant streams in the Session Description Protocol (SDP).
The semantics defined in this document are to be used with the SDP
Grouping Framework.  Grouping semantics at the Synchronization Source
(SSRC) level are also defined in this document for RTP streams using
SSRC multiplexing.

This document is a product of the Multiparty Multimedia Session Control Working Group of the IETF.

This is now a Proposed Standard.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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The RFC Editor Team
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Subject: [MMUSIC] Comments on ABNF in draft-petithuguenin-mmusic-ice-sip-sdp-01
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Hi,

As part of the RTSP for NAT work I actually noticed an issue with the
ABNF that is defined for ICE in SDP.

>From Section 8.1:

   candidate-attribute   = "candidate" ":" foundation SP component-id SP
                           transport SP
                           priority SP
                           connection-address SP     ;from RFC 4566
                           port         ;port from RFC 4566
                           SP cand-type
                           [SP rel-addr]
                           [SP rel-port]
                           *(SP extension-att-name SP
                                extension-att-value)

   foundation            = 1*32ice-char
   component-id          = 1*5DIGIT
   transport             = "UDP" / transport-extension
   transport-extension   = token              ; from RFC 3261
   priority              = 1*10DIGIT
   cand-type             = "typ" SP candidate-types



Petit-Huguenin & KeranenExpires August 29, 2013                 [Page 9]

Internet-Draft             ICE SIP/SDP Usage               February 2013


   candidate-types       = "host" / "srflx" / "prflx" / "relay" / token
   rel-addr              = "raddr" SP connection-address
   rel-port              = "rport" SP port
   extension-att-name    = byte-string    ;from RFC 4566
   extension-att-value   = byte-string
   ice-char              = ALPHA / DIGIT / "+" / "/"

The issue is what is allowed in extension-att-name and
extension-att-value. They are listed here as byte-string. That is an
extremely wide allowance in what values to be included.

First of all it includes SP the character used for delimiting the
-att-name against the -att-value and the next set of name and value. I
can't see that one can allow that, and if one have values that includes
space, then they must be escaped.

Further I wonder if it wise to include the non-visible values from the
below 128 range of the byte. It might be okay, but some thought needs to
be put into this.

Cheers

Magnus Westerlund

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From iesg-secretary@ietf.org  Wed Jan 22 04:25:18 2014
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Cc: mmusic chair <mmusic-chairs@tools.ietf.org>, mmusic mailing list <mmusic@ietf.org>, RFC Editor <rfc-editor@rfc-editor.org>
Subject: [MMUSIC] Protocol Action: 'Delayed Duplication Attribute in the Session Description Protocol' to Proposed Standard (draft-ietf-mmusic-delayed-duplication-03.txt)
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The IESG has approved the following document:
- 'Delayed Duplication Attribute in the Session Description Protocol'
  (draft-ietf-mmusic-delayed-duplication-03.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-delayed-duplication/




Technical Summary

   A straightforward approach to provide protection against packet
   losses due to network outages with a longest duration of T time units
   is to simply duplicate the original packets and send each copy
   separated in time by at least T time units.  This approach is
   commonly referred to as Time-shifted Redundancy, Temporal Redundancy
   or simply Delayed Duplication.  This document defines an attribute to
   indicate the presence of temporally redundant media streams and the
   duplication delay in the Session Description Protocol.


Working Group Summary

  There is WG consensus behind this draft. All issues raised during
  and before WGLC have been addressed.


Document Quality

  The document has received reviews from working group participants
  both at MMUSIC and AVTEXT and the document shepherd.  There is
  currently at least one vendor implementing this mechanism and
  various other vendors have expressed interest for this technique.


Personnel

Who is the Document Shepherd? Who is the Responsible Area
Director?

  Ari Keranen is the Document Shepherd, and the Responsible Area
  Director is Gonzalo Camarillo.


RFC Editor's Note

  Please add the following site as a reference to Lucky 13 mentioned in Section 5.
  http://www.isg.rhul.ac.uk/tls/Lucky13.html

From magnus.westerlund@ericsson.com  Wed Jan 22 06:01:31 2014
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Date: Wed, 22 Jan 2014 15:00:56 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org
Subject: [MMUSIC] draft-ietf-mmusic-rtsp-nat: ICE restart fails, then what?
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WG,

I received a comment in the doc shepherd preparation from Flemming. I
need some feedback on this.

This is in regards to ICE restart.

One initiate an ICE restart when having an existing promoted and
used candidate pair. So RFC 5245 is clear that you continue to use the
candidate pair while restarting, but it is not clear that you will
continue to use it if an ICE restart fails.

My intuition would be to keep on using the candidate pair if it still
works. But, if one looks at 8.1.2 and applies that to a restart the
result is different. I mean the following text:

   o  If the state of the check list is Failed, ICE has not been able to
      complete for this media stream.  The correct behavior depends on
      the state of the check lists for other media streams:

      *  If all check lists are Failed, ICE processing overall is
         considered to be in the Failed state, and the agent SHOULD
         consider the session a failure, SHOULD NOT restart ICE, and the
         controlling agent SHOULD terminate the entire session.

Thus, one should terminate the session. I think there is possibility
that there should be a bit more room for attempting rectifying actions.

Thus my current proposal for Section 6.12 of draft-ietf-mmmusic-rtsp-nat
is the following:

   If the RTSP session is in playing state at the time of sending the
   SETUP request requiring ICE restart, then the ICE connectivity checks
   SHALL use Regular nomination.  Any ongoing media delivery continues
   on the previously nominated candidate pairs until the new pairs have
   been nominated for the individual candidate.  Once the nomination of
   the new candidate pair has completed, all unused candidates may be
   released.  If the ICE processing fails and no new candidate pairs are
   promoted for use, then one MAY continue to use the previously
   nominated candidate pairs while they still function.  If they have
   appear to fail to transport media packets anymore then the client can
   select between two actions.  First, if it has any actions available
   that might make ICE work, like trying another STUN/TURN server or
   change the transport parameters it modifies the session, and if ICE
   is still to be used restart ICE.  If the client is unable to modify
   the ICE parameters, it MUST NOT restart the ICE processing, and
   SHOULD terminate the RTSP session.


Have I missed anything about this interaction with ICE?

Any issues with what I propose?

Thanks

Magnus Westerlund

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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: "Adam Gensler (agensler)" <agensler@cisco.com>
Thread-Topic: Review of draft-ietf-mmusic-udptl-dtls-03
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Cc: mmusic <mmusic@ietf.org>, "christer.holmberg@ericsson.com" <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Review of draft-ietf-mmusic-udptl-dtls-03
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Thanks very much for your detailed review, Adam.  I'm copying the MMUSIC@ l=
ist for awareness.

Cheers,

Gonzalo


On Jan 22, 2014, at 2:16 PM, Adam Gensler (agensler) <agensler@cisco.com> w=
rote:

> Gonzalo, Christer,
>=20
> After speaking with Gonzalo I've reviewed draft-ietf-mmusic-udptl-dtls-03=
.
> Overall this draft is excellent and appears to cover all the corner cases
> I could think of. I do have a few comments, see below:
>=20
> 1. In section 3.1, secure channel establishment, there is the following
> text:
>=20
> "If the endpoint supports, and is willing to use, a cipher suite with an
> associated certificate, it MUST include an SDP fingerprint attribute
> [RFC4572 <http://tools.ietf.org/html/rfc4572>] in the SDP."
>=20
> "If a cipher suite with an associated certificate is selected during the
> DTLS handshake, the certificate received during the DTLS handshake MUST
> match the fingerprint received in the SDP fingerprint attribute."
>=20
> To me this seems to indicate the particular cipher suites are associated
> with a particular certificate. While it's true that the certificate plays
> a role in overall TLS negotiation, it does not provide a list of ciphers
> to use. I think what this is saying is that if the client wishes to use a
> certificate for DTLS it should provide a certificate fingerprint. Is that
> right? Given the overall complexity of PKI in general I think being extra
> clear here would help avoid any confusion.
>=20
>=20
> 2. Section 4.1 describes handling anonymous faxes with freshly generated
> self-signed certificates. It goes on to say:
>=20
> "...and the content of the subjectAltName attribute inside the certificat=
e
> MUST NOT contain information that either allows correlation or
> identification of the user making anonymous calls."
>=20
> While the subjectAltName could potentially have identifiable data in it,
> it is not always present as a field within the certificate. Some
> certificates may have a subjectAltName, some may not. However, all
> certificates should have a commonName. I think that is the item that
> should be specifically called out as a place to avoid identifiable
> information. Or, perhaps, simply stipulating that no fields within the
> certificate should contain identifiable information.
>=20
>=20
> 3. Section 4.3 describes rekeying a connection that is established. The
> DTLS RFC describes how to handle this, but gives two options to an
> endpoint where out-of-order packets are received that have been encrypted
> with different keys; discard the packet, or, retain two sets of keying
> material for some recommended interval. DTLS is not specific as to what t=
o
> do here. Section 4.3 says the receiver SHOULD maintain the keying
> material. Given the nature of T.38, I think it may be prudent to say the
> receiver MUST maintain the keying material. Dropping and/or discarding
> T.38 packets would never be my preference.
>=20
> If there's any questions or concerns about the above please let me know.
>=20
> Best regards,
> Adam Gensler
>=20
>=20


From fandreas@cisco.com  Wed Jan 22 16:28:35 2014
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To: SM <sm@resistor.net>, "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Subject: Re: [MMUSIC] Last Call: <draft-ietf-mmusic-sdp-g723-g729-04.txt> (Offer/Answer Considerations for G723 Annex A and G729 Annex	B) to Proposed Standard
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Greetings

The chairs have reviewed this thread carefully and also discussed it 
with our AD. To conclude, we believe we have concensus to proceed with 
the document as follows:
- The document does not formally update RFC 4856 (or any other RFC) and 
hence the front page and abstract will be updated accordingly.
- The document will remain Standards Track.
- Since the "annexa" and "annexb" parameters are defined in RFC 4856, it 
makes sense to keep the reference to RFC 4856 in the introductory text, 
along the lines of Muthu's suggested "New Introduction" text in the 
previous message in this thread.

If anybody disagrees with this conclusion, please let the chairs know.

Thanks

     Ari & Flemming (MMUSIC co-chairs)




On 12/4/13, 11:54 PM, SM wrote:
> Hi Muthu,
> At 20:13 04-12-2013, Muthu Arul Mozhi Perumal (mperumal) wrote:
>> I see your point.
>>
>> Our opinion was that an RFC that specifies the mapping of media type 
>> parameters with SDP and registers the media types with IANA should 
>> also describe its usage, especially with the offer/answer model (e.g, 
>> RFC435). In that sense we thought draft-ietf-mmusic-sdp-g723-g729 is 
>> an update to RFC4856.
>>
>> Your opinion is that they should be delinked -- one RFC can register 
>> the media types and another RFC can describe the usage for the 
>> registered media types independently. I am fine with that. The 
>> abstract and introduction in draft-ietf-mmusic-sdp-g723-g729 can 
>> easily be changed to reflect it.
>
> Yes, but you can have both in one RFC.  The registration can be 
> requested in the IANA Considerations section.
>
>> ---------------------------------------------------------------------
>> Current Abstract:
>>    RFC4856 describes the annexa parameter for G723 and the annexb
>>    parameter for G729, G729D and G729E. However, the specification does
>>    not describe the offerer and answerer behavior when the value of the
>>    annexa or annexb parameter does not match in the Session Description
>>    protocol(SDP) offer and answer.  This document provides the offer/
>>    answer considerations for these parameters and updates RFC4856.
>>
>> New Abstract:
>>    There are no specifications that describe the offerer and answerer
>>    behavior when the value of the annexa parameter of G729 or the annexb
>>    parameter of G729, G729D and G729E does not match in the Session
>>    Description protocol(SDP) offer and answer. This document provides 
>> the
>>    offer/answer considerations for these parameters.
>
> I suggest removing "there are no specifications ..." and explain what 
> the document specifies.
>
>> Current Introduction:
>>    [RFC4856] describes the annexa parameter for G723 as follows:
>>
>>       annexa: indicates that Annex A, voice activity detection, is used
>>       or preferred.  Permissible values are "yes" and "no" (without the
>>       quotes); "yes" is implied if this parameter is omitted.
>>
>>    Also, [RFC4856] describes the annexb parameter for G729, G729D and
>>    G729E as follows:
>>
>>       annexb: indicates that Annex B, voice activity detection, is used
>>       or preferred.  Permissible values are "yes" and "no" (without the
>>       quotes); "yes" is implied if this parameter is omitted.
>
> The above starts with RFC 4856.  I suggest starting with the RFC 4566 
> followed by the RFC 3264 text (I did not verify the reference).  You 
> can then specify the fmtp parameters for Annex A, etc.  Once you have 
> done that, verify whether there are some registry requirements and add 
> text for that in a IANA Considerations section.
>
> Regards,
> -sm
> .
>


From internet-drafts@ietf.org  Thu Jan 23 01:58:32 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-11.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           : The Evaluation of Different Network Address Trans=
lator (NAT) Traversal Techniques for Media Controlled by Real-time Streamin=
g Protocol (RTSP)
        Authors         : Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-11.txt
	Pages           : 39
	Date            : 2014-01-23

Abstract:
   This document describes several Network Address Translator (NAT)
   traversal techniques that were considered to be used for establishing
   the RTP media flows controlled by the Real-time Streaming Protocol
   (RTSP).  Each technique includes a description of how it would be
   used, the security implications of using it and any other deployment
   considerations it has.  There are also discussions on how NAT
   traversal techniques relate to firewalls and how each technique can
   be applied in different use cases.  These findings were used when
   selecting the NAT traversal for RTSP 2.0, which is specified in a
   separate document.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat-evaluation/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-evaluation-11

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rtsp-nat-evaluation-11


Please note that it may take a couple of minutes from the time of submission
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From internet-drafts@ietf.org  Thu Jan 23 02:03:30 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-18.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           : A Network Address Translator (NAT) Traversal Mech=
anism for Media Controlled by Real-Time Streaming Protocol (RTSP)
        Authors         : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-18.txt
	Pages           : 32
	Date            : 2014-01-23

Abstract:
   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams set up and
   controlled with Real-time Streaming Protocol version 2 (RTSP 2.0).
   It uses Interactive Connectivity Establishment (ICE) adapted to use
   RTSP as a signalling channel, defining the necessary RTSP extensions
   and procedures.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-18

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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           : The Evaluation of Different Network Address Trans=
lator (NAT) Traversal Techniques for Media Controlled by Real-time Streamin=
g Protocol (RTSP)
        Authors         : Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-12.txt
	Pages           : 39
	Date            : 2014-01-23

Abstract:
   This document describes several Network Address Translator (NAT)
   traversal techniques that were considered to be used for establishing
   the RTP media flows controlled by the Real-time Streaming Protocol
   (RTSP).  Each technique includes a description of how it would be
   used, the security implications of using it and any other deployment
   considerations it has.  There are also discussions on how NAT
   traversal techniques relate to firewalls and how each technique can
   be applied in different use cases.  These findings were used when
   selecting the NAT traversal for RTSP 2.0, which is specified in a
   separate document.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat-evaluation/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-evaluation-12

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rtsp-nat-evaluation-12


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.

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ftp://ftp.ietf.org/internet-drafts/


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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-12.txt
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WG,

This revision is the result of feedback from Flemming during his doc
shepherd write-up process and review.

What has been changed is:
- Made the copied text into a quote to avoid having to put pre RFC 5378
disclaimer on the document. This was what caused the rapid 11-12 version
as I had realized that I had edited the text from RFC 4787.
- Lots of small editorial fixes.
- More references in appropriate places, including to the latching
document.
- Reformulated text regarding security considerations for latching.

Cheers

Magnus

On 2014-01-23 11:29, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
>         Title           : The Evaluation of Different Network Address Translator (NAT) Traversal Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
>         Authors         : Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-12.txt
> 	Pages           : 39
> 	Date            : 2014-01-23
> 
> Abstract:
>    This document describes several Network Address Translator (NAT)
>    traversal techniques that were considered to be used for establishing
>    the RTP media flows controlled by the Real-time Streaming Protocol
>    (RTSP).  Each technique includes a description of how it would be
>    used, the security implications of using it and any other deployment
>    considerations it has.  There are also discussions on how NAT
>    traversal techniques relate to firewalls and how each technique can
>    be applied in different use cases.  These findings were used when
>    selecting the NAT traversal for RTSP 2.0, which is specified in a
>    separate document.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat-evaluation/
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-evaluation-12
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-evaluation-12
> 
> 
> 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/
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From magnus.westerlund@ericsson.com  Thu Jan 23 02:41:26 2014
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Date: Thu, 23 Jan 2014 11:41:21 +0100
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-18.txt
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WG,

This revision is the result of feedback from Flemming during his doc
shepherd write-up process and review.

The changes made are:
- Lots of editorial corrections
- Section 4.2: Added escaping for ICE extensions resulting in new
parameters and values.
- Changed some lower case should, must etc to their upper case versions
where appropriate.
- Section 6.10 has been expanded to clarify when 480 response may result
in freeing of candidates.
- Section 6.12 has text on what to do when a ICE restart fails.

Cheers

Magnus



On 2014-01-23 11:03, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
>         Title           : A Network Address Translator (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming Protocol (RTSP)
>         Authors         : Jeff Goldberg
>                           Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-18.txt
> 	Pages           : 32
> 	Date            : 2014-01-23
> 
> Abstract:
>    This document defines a solution for Network Address Translation
>    (NAT) traversal for datagram based media streams set up and
>    controlled with Real-time Streaming Protocol version 2 (RTSP 2.0).
>    It uses Interactive Connectivity Establishment (ICE) adapted to use
>    RTSP as a signalling channel, defining the necessary RTSP extensions
>    and procedures.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat/
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-18
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-18
> 
> 
> 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/
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From harald@alvestrand.no  Thu Jan 23 03:22:25 2014
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Subject: [MMUSIC] MSID draft -03 is out
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This draft was published on January 6.

Copy of the changelog:

C.9.  Changes from mmusic-msid-02 to -03

    Corrected even more cases where the word "ssrc" was not changed to
    "M-line".

    Added the functionality of using an asterisk (*) in the msid-semantic
    line, in order to remove the need for listing all msids in the msid-
    semantic line whne only one msid-semantic is in use.

    Removed some now-unnecessary text.

Comments are welcome!

         Harald


From adam@nostrum.com  Thu Jan 23 12:18:07 2014
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Subject: [MMUSIC] Telephone-event and multiple clock rates
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This is a multi-part message in MIME format.
--------------010400040301040307060201
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I've had an interesting issue brought to my attention regarding RFC4473 
"telephone-event" handling.

Section 2.1 requires that "Named telephone events are carried as part of 
the audio stream and MUST use the same sequence number and timestamp 
base as the regular audio channel".

On its face, this would imply that offering voice codecs with varying 
clock rates would result in the need to offer several PTs with 
telephone-event, varying only in frequency.

So, for example, if we were setting up a media session that looked like this

    m=audio 12346 RTP/AVP 0 97 98 99
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 opus/4800 0/2
    a=rtpmap:98 AMR-WB/16000
    a=rtpmap:99 speex/32000

But want to add telephone-event to it, we would need to do this:

    m=audio 12346 RTP/AVP 0 97 98 99 100 101 102 103
    a=rtpmap:0 PCMU/8000
    a=rtpmap:97 opus/48000/2
    a=rtpmap:98 AMR-WB/16000
    a=rtpmap:99 speex/32000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15
    a=rtpmap:101 telephone-event/48000
    a=fmtp:101 0-15
    a=rtpmap:102 telephone-event/16000
    a=fmtp:102 0-15
    a=rtpmap:103 telephone-event/32000
    a=fmtp:103 0-15


Rather than this:

    m=audio 12346 RTP/AVP 0 97 98 99 100
    a=rtpmap:0 PCMU/8000
    a=rtpmap:0 PCMA/8000
    a=rtpmap:97 opus/48000/2
    a=rtpmap:98 AMR-WB/16000
    a=rtpmap:99 speex/32000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15


And then it would be incumbent on the remote end to ensure that they use 
the *correct* DTMF PT to match the codec that they're using for speech.

This all seems like patent nonsense, and it burns an additional PT for 
each clock rate. Is that really what was intended by this language in 
RFC 4733?

/a

--------------010400040301040307060201
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <tt>I've had an interesting issue brought to my attention regarding
      RFC4473 "telephone-event" handling.</tt><tt><br>
    </tt><tt><br>
    </tt><tt>Section 2.1 requires that "Named telephone events are
      carried as part of the audio stream and MUST use the same sequence
      number and timestamp base as the regular audio channel".</tt><tt><br>
    </tt><tt><br>
    </tt><tt>On its face, this would imply that offering voice codecs
      with varying clock rates would result in the need to offer several
      PTs with telephone-event, varying only in frequency.</tt><tt><br>
    </tt><tt><br>
    </tt><tt>So, for example, if we were setting up a media session that
      looked like this</tt><tt><br>
    </tt><tt><br>
    </tt>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <tt> Â Â  m</tt><tt>=audio 12346 RTP/AVP 0 </tt><tt>97 98 99<br>
    </tt><tt>Â Â  a=rtpmap:0 PCMU/8000</tt><tt><br>
    </tt><tt>Â Â  a=rtpmap:97 opus/4800</tt>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <tt>0/2<br>
      Â Â  a=rtpmap:98 AMR-WB/16000<br>
    </tt><tt>Â Â  a=rtpmap:99 speex/3200</tt><tt>0<br>
      <br>
    </tt><tt>But want to add telephone-event to it, we would need to do
      this:</tt><tt><br>
    </tt><br>
    <tt>Â Â  m</tt><tt>=audio 12346 RTP/AVP 0 </tt><tt>97 98 99 100 101
      102 103<br>
    </tt><tt><tt>Â Â  a=rtpmap:0 PCMU/8000</tt><tt><br>
      </tt><tt>Â Â  a=rtpmap:97 opus/4800</tt><tt>0/2<br>
        Â Â  a=rtpmap:98 AMR-WB/16000<br>
      </tt><tt>Â Â  a=rtpmap:99 speex/3200</tt><tt>0<br>
        Â Â  a=rtpmap:100 telephone-event/8000<br>
        Â Â  a=fmtp:100 0-15<br>
      </tt></tt><tt><tt>Â Â  a=rtpmap:101 telephone-event/48000<br>
        Â Â  a=fmtp:101 0-15<br>
      </tt></tt><tt><tt>Â Â  a=rtpmap:102 telephone-event/16000<br>
        Â Â  a=fmtp:102 0-15<br>
      </tt></tt><tt><tt>Â Â  a=rtpmap:103 telephone-event/32000<br>
        Â Â  a=fmtp:103 0-15<br>
        <br>
      </tt></tt><tt><br>
      Rather than this:</tt><br>
    <br>
    <tt>Â Â  m</tt><tt>=audio 12346 RTP/AVP 0 </tt><tt>97 98 99 100<br>
    </tt><tt><tt>Â Â  a=rtpmap:0 PCMU/8000</tt><tt><br>
        Â Â  a=rtpmap:0 PCMA/8000</tt><tt><br>
      </tt><tt>Â Â  a=rtpmap:97 opus/4800</tt><tt>0/2<br>
        Â Â  a=rtpmap:98 AMR-WB/16000<br>
      </tt><tt>Â Â  a=rtpmap:99 speex/3200</tt><tt>0<br>
        Â Â  a=rtpmap:100 telephone-event/8000<br>
        Â Â  a=fmtp:100 0-15<br>
      </tt></tt><tt><br>
    </tt><tt><br>
      And then it would be incumbent on the remote end to ensure that
      they use the *correct* DTMF PT to match the codec that they're
      using for speech.<br>
      <br>
      This all seems like patent nonsense, and it burns an additional PT
      for each clock rate. Is that really what was intended by this
      language in RFC 4733?</tt><tt><br>
    </tt><tt><br>
    </tt><tt>/a</tt><tt><br>
    </tt>
  </body>
</html>

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From christer.holmberg@ericsson.com  Thu Jan 23 17:43:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, "Adam Gensler (agensler)" <agensler@cisco.com>
Thread-Topic: Review of draft-ietf-mmusic-udptl-dtls-03
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Review of draft-ietf-mmusic-udptl-dtls-03
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Hi Adam,

Thank You very much for the review! I am currently on a business trip, but =
I'll address your comments asap.

Regards,

Christer


-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Gonzalo Salgueiro (gsalguei) [mailto:gsalguei@cisco.com]=20
L=E4hetetty: 22. tammikuuta 2014 21:20
Vastaanottaja: Adam Gensler (agensler)
Kopio: Gonzalo Salgueiro (gsalguei); Christer Holmberg; mmusic
Aihe: Re: Review of draft-ietf-mmusic-udptl-dtls-03
T=E4rkeys: Suuri

Thanks very much for your detailed review, Adam.  I'm copying the MMUSIC@ l=
ist for awareness.

Cheers,

Gonzalo


On Jan 22, 2014, at 2:16 PM, Adam Gensler (agensler) <agensler@cisco.com> w=
rote:

> Gonzalo, Christer,
>=20
> After speaking with Gonzalo I've reviewed draft-ietf-mmusic-udptl-dtls-03=
.
> Overall this draft is excellent and appears to cover all the corner=20
> cases I could think of. I do have a few comments, see below:
>=20
> 1. In section 3.1, secure channel establishment, there is the=20
> following
> text:
>=20
> "If the endpoint supports, and is willing to use, a cipher suite with=20
> an associated certificate, it MUST include an SDP fingerprint=20
> attribute
> [RFC4572 <http://tools.ietf.org/html/rfc4572>] in the SDP."
>=20
> "If a cipher suite with an associated certificate is selected during=20
> the DTLS handshake, the certificate received during the DTLS handshake=20
> MUST match the fingerprint received in the SDP fingerprint attribute."
>=20
> To me this seems to indicate the particular cipher suites are=20
> associated with a particular certificate. While it's true that the=20
> certificate plays a role in overall TLS negotiation, it does not=20
> provide a list of ciphers to use. I think what this is saying is that=20
> if the client wishes to use a certificate for DTLS it should provide a=20
> certificate fingerprint. Is that right? Given the overall complexity=20
> of PKI in general I think being extra clear here would help avoid any con=
fusion.
>=20
>=20
> 2. Section 4.1 describes handling anonymous faxes with freshly=20
> generated self-signed certificates. It goes on to say:
>=20
> "...and the content of the subjectAltName attribute inside the=20
> certificate MUST NOT contain information that either allows=20
> correlation or identification of the user making anonymous calls."
>=20
> While the subjectAltName could potentially have identifiable data in=20
> it, it is not always present as a field within the certificate. Some=20
> certificates may have a subjectAltName, some may not. However, all=20
> certificates should have a commonName. I think that is the item that=20
> should be specifically called out as a place to avoid identifiable=20
> information. Or, perhaps, simply stipulating that no fields within the=20
> certificate should contain identifiable information.
>=20
>=20
> 3. Section 4.3 describes rekeying a connection that is established.=20
> The DTLS RFC describes how to handle this, but gives two options to an=20
> endpoint where out-of-order packets are received that have been=20
> encrypted with different keys; discard the packet, or, retain two sets=20
> of keying material for some recommended interval. DTLS is not specific=20
> as to what to do here. Section 4.3 says the receiver SHOULD maintain=20
> the keying material. Given the nature of T.38, I think it may be=20
> prudent to say the receiver MUST maintain the keying material.=20
> Dropping and/or discarding
> T.38 packets would never be my preference.
>=20
> If there's any questions or concerns about the above please let me know.
>=20
> Best regards,
> Adam Gensler
>=20
>=20


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From: "Stach, Thomas" <thomas.stach@unify.com>
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Thread-Topic: [MMUSIC] Telephone-event and multiple clock rates
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: "Stach, Thomas" <thomas.stach@unify.com>, Adam Roach <adam@nostrum.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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Hi,

My understanding for the reasoning behind this definition in the RFC is
due to that the events is supposed to be connected to the same time-line
as the audio. Thus it ended up using the same SSRC. To avoid timestamp
rate switching which we know has issues
(https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-rates/)
it was mandated to provide one PT for each rate negotiated for the audio
payload types.

On 2014-01-24 08:19, Stach, Thomas wrote:
> 
> It still remains interesting what his means if two codecs with different
> clock rates were negotiated and codec-switching-on-the-fly happens.
> 

So there is a difference in negotiating possible to use, another to
actively switch between them within a RTP session for a particular media
source. If you actually switch then you have to take the considerations
in draft-ietf-avtext-multiple-clock-rates into account.

If you are not switching, then these are only ensuring that you can use
the events for what ever audio payload format you select to use out of
the set offered and accepted.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
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From: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: Comments on ABNF in draft-petithuguenin-mmusic-ice-sip-sdp-01
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Subject: Re: [MMUSIC] Comments on ABNF in draft-petithuguenin-mmusic-ice-sip-sdp-01
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Hi Magnus,

Good point. That ABNF indeed seems to be overly liberal on the extension at=
tribute names and values. Perhaps e.g., token would be more appropriate at =
least for the extension-att-name (like att-field in RFC4566).=20

For extension-att-value, are there any reasonable use cases where e.g., VCH=
AR would not be sufficient? Are RFC4566 att-values with non-printable chara=
cters used somewhere?


Cheers,
Ari

On 22 Jan 2014, at 11:35, Magnus Westerlund <magnus.westerlund@ericsson.com=
> wrote:

> Hi,
>=20
> As part of the RTSP for NAT work I actually noticed an issue with the
> ABNF that is defined for ICE in SDP.
>=20
> From Section 8.1:
>=20
>   candidate-attribute   =3D "candidate" ":" foundation SP component-id SP
>                           transport SP
>                           priority SP
>                           connection-address SP     ;from RFC 4566
>                           port         ;port from RFC 4566
>                           SP cand-type
>                           [SP rel-addr]
>                           [SP rel-port]
>                           *(SP extension-att-name SP
>                                extension-att-value)
>=20
>   foundation            =3D 1*32ice-char
>   component-id          =3D 1*5DIGIT
>   transport             =3D "UDP" / transport-extension
>   transport-extension   =3D token              ; from RFC 3261
>   priority              =3D 1*10DIGIT
>   cand-type             =3D "typ" SP candidate-types
>=20
>=20
>=20
> Petit-Huguenin & KeranenExpires August 29, 2013                 [Page 9]
>=20
> Internet-Draft             ICE SIP/SDP Usage               February 2013
>=20
>=20
>   candidate-types       =3D "host" / "srflx" / "prflx" / "relay" / token
>   rel-addr              =3D "raddr" SP connection-address
>   rel-port              =3D "rport" SP port
>   extension-att-name    =3D byte-string    ;from RFC 4566
>   extension-att-value   =3D byte-string
>   ice-char              =3D ALPHA / DIGIT / "+" / "/"
>=20
> The issue is what is allowed in extension-att-name and
> extension-att-value. They are listed here as byte-string. That is an
> extremely wide allowance in what values to be included.
>=20
> First of all it includes SP the character used for delimiting the
> -att-name against the -att-value and the next set of name and value. I
> can't see that one can allow that, and if one have values that includes
> space, then they must be escaped.
>=20
> Further I wonder if it wise to include the non-visible values from the
> below 128 range of the byte. It might be okay, but some thought needs to
> be put into this.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20


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Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Hi,

The way forward you proposed sounds good. Looking forward to the updated dr=
aft. However, I didn=92t really understand how is the lack of coordination =
and different promo algorithms an advantage?


Cheers,
Ari

On 13 Jan 2014, at 13:46, Pal Martinsen (palmarti) <palmarti@cisco.com> wro=
te:

>>=20
>>=20
>> Also, if there is no coordination/signaling between the endpoints, what =
are the odds they'd end up using the same promotion algorithm?
>>=20
> Exactly, and that might be an advantage..=20


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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 1/24/14 04:29, Magnus Westerlund wrote:
> My understanding for the reasoning behind this definition in the RFC is
> due to that the events is supposed to be connected to the same time-line
> as the audio. Thus it ended up using the same SSRC. To avoid timestamp
> rate switching which we know has issues
> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-rates/)
> it was mandated to provide one PT for each rate negotiated for the audio
> payload types.

To make sure I understand what you're saying correctly: you are 
asserting that having four different clock rates for voice codecs 
requires us to use four different PTs for telephone-event if we're going 
to have them on the same SSRC. Is that correct?

/a

From magnus.westerlund@ericsson.com  Mon Jan 27 00:47:34 2014
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 2014-01-24 16:59, Adam Roach wrote:
> On 1/24/14 04:29, Magnus Westerlund wrote:
>> My understanding for the reasoning behind this definition in the RFC is
>> due to that the events is supposed to be connected to the same time-line
>> as the audio. Thus it ended up using the same SSRC. To avoid timestamp
>> rate switching which we know has issues
>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-rates/)
>>
>> it was mandated to provide one PT for each rate negotiated for the audio
>> payload types.
> 
> To make sure I understand what you're saying correctly: you are
> asserting that having four different clock rates for voice codecs
> requires us to use four different PTs for telephone-event if we're going
> to have them on the same SSRC. Is that correct?

Yes!

All to avoid the issue that with rate switching within an SSRC. This is
both potential loss of time synchronization as well as degrading RTP
timestamp based RTCP measurements. And the above draft (soon RFC) does
require new implementations to change their SSRC when doing
rate-switching. That clearly have signaling implications in the context
of WebRTC.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
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 randell-ietf@jesup.org  Mon Jan 27 00:48:27 2014
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 1/24/2014 10:59 AM, Adam Roach wrote:
> On 1/24/14 04:29, Magnus Westerlund wrote:
>> My understanding for the reasoning behind this definition in the RFC is
>> due to that the events is supposed to be connected to the same time-line
>> as the audio. Thus it ended up using the same SSRC. To avoid timestamp
>> rate switching which we know has issues
>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-rates/) 
>>
>> it was mandated to provide one PT for each rate negotiated for the audio
>> payload types.
>
> To make sure I understand what you're saying correctly: you are 
> asserting that having four different clock rates for voice codecs 
> requires us to use four different PTs for telephone-event if we're 
> going to have them on the same SSRC. Is that correct?

If those four PT's (not codecs) are accepted (and thus can be switched 
between on-the-fly), yes.  Multiple clock rates plays hell with RTP 
timestamps.

You could use a separate SSRC (or m=audio line) for each clock rate 
instead.  This would likely mean glitching at transitions however, so 
not a great solution, but works.  You might be able to avoid glitches by 
making sure you send RTCP for the alternate SSRCs so synchronization 
info is available, though this may require the receiver to know somehow 
that the SSRC's need to be "lip-synced" (in this case audio to audio 
though).  Overall - here be (baby) dragons in the glitch-avoidance 
variation...

In theory you could allow one telephone-event PT and do a renegotiate 
after an effective codec/PT change.  You might need to be careful about 
asynchronicity due to the renegotiation and not re-use the same PT for a 
different rate of telephone-event (ping-pong between two).  There would 
exist a window when there was a switch where it would be problematic to 
send telephone-event (and you'd need to require that both sides switch 
when one does). Overall - here be (big) dragons.

-- 
Randell Jesup -- rjesup a t mozilla d o t com


From christer.holmberg@ericsson.com  Mon Jan 27 03:36:21 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: Questions on draft-ietf-mmusic-sctp-sdp-05 
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Subject: [MMUSIC] Questions on draft-ietf-mmusic-sctp-sdp-05
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Hi,

Section 5.1 of draft-ietf-mmusic-sctp-sdp-05 describes how an offered SCTP =
association can be rejected, by in the Answer setting the port value in the=
 associated SDP sctpmap attribute to zero.

Q1:         When an SCTP association is rejected, shall the associated port=
 value in the m- line fmt list also be set to zero? The text only talks abo=
ut the sctpmap attribute.

The draft does not talk about subsequent offers.

Q2:         Do I need to include all sctpmap attributes in subsequent offer=
s, even if some are associated with previously rejected STCP associations, =
in order to maintain the order?

Q3:         Can I, in a subsequent offer, remove an SCTP association by set=
ting the associated port value to zero? If so, do I set the port value to z=
ero in the associated sctpmap attribute, and/or the m- line fmt list?

The draft only talks about having a single media-subtype per SCTP associati=
on.

Q4:         Is it possible to define multiple media-subtypes per SCTP assoc=
iation?

                Example:

                m=3Dapplication 54111 DTLS/SCTP 5000
              c=3DIN IP4 79.97.215.79
              a=3Dsctpmap:5000 webrtc-datachannel 16
              a=3Dsctpmap:5000 bfcp 2
              a=3Dsctpmap:5000 t38 1

                (Each media-subtype could use different SCTP streams within=
 the SCTP association)


Thanks!

Regards,

Christer

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<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">Section 5.1 of draft-ietf-mmusic-sctp-sdp-05 describ=
es how an offered SCTP association can be rejected, by in the Answer settin=
g the port value in the associated SDP sctpmap attribute to zero.<o:p></o:p=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q1:</b>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; When an SCTP association is rejected, shall the associated port valu=
e in the m- line fmt list also be set to zero? The text only talks about th=
e sctpmap attribute.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft does not talk about subsequent offers.<o:p=
></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q2:</b>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; Do I need to include all sctpmap attributes in subsequent offers, ev=
en if some are associated with previously rejected STCP associations, in or=
der to maintain the order?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q3:</b>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; Can I, in a subsequent offer, remove an SCTP association by setting =
the associated port value to zero? If so, do I set the port value to zero i=
n the associated sctpmap attribute, and/or the m- line fmt list?<o:p></o:p>=
</p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft only talks about having a single media-sub=
type per SCTP association.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><b>Q4</b>:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; Is it possible to define multiple media-subtypes per SCTP associatio=
n?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Example:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Dapplication 54111 DTLS/SCTP <b>5=
000</b><o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4 79.97.215.79<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:<b>5000</b> webrtc-datachannel 16<o:=
p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:<b>5000</b> bfcp 2<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; a=3Dsctpmap:<b>5000</b> t38 1<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; (Each media-subtype could use differ=
ent SCTP streams within the SCTP association)<o:p></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">Thanks!<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_7594FB04B1934943A5C02806D1A2204B1D13BB05ESESSMB209erics_--

From christer.holmberg@ericsson.com  Tue Jan 28 05:02:57 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Telephone-event and multiple clock rates
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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Hi,

I guess it would be good to add some text to BUNDLE, to indicate that you m=
ust include the telephone-event PTs for every m- line which may use them. I=
.e. you cannot indicate telephone-event for one m- line, and then use them =
on every m- line within the bundle group.

But, I guess there will be no need for people to support telephone-events o=
n more than one m- line to begin with, so...

Regards,

Christer


-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
Sent: 27. tammikuuta 2014 10:48
To: mmusic@ietf.org
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates

On 1/24/2014 10:59 AM, Adam Roach wrote:
> On 1/24/14 04:29, Magnus Westerlund wrote:
>> My understanding for the reasoning behind this definition in the RFC=20
>> is due to that the events is supposed to be connected to the same=20
>> time-line as the audio. Thus it ended up using the same SSRC. To=20
>> avoid timestamp rate switching which we know has issues
>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
>> tes/)
>>
>> it was mandated to provide one PT for each rate negotiated for the=20
>> audio payload types.
>
> To make sure I understand what you're saying correctly: you are=20
> asserting that having four different clock rates for voice codecs=20
> requires us to use four different PTs for telephone-event if we're=20
> going to have them on the same SSRC. Is that correct?

If those four PT's (not codecs) are accepted (and thus can be switched betw=
een on-the-fly), yes.  Multiple clock rates plays hell with RTP timestamps.

You could use a separate SSRC (or m=3Daudio line) for each clock rate inste=
ad.  This would likely mean glitching at transitions however, so not a grea=
t solution, but works.  You might be able to avoid glitches by making sure =
you send RTCP for the alternate SSRCs so synchronization info is available,=
 though this may require the receiver to know somehow that the SSRC's need =
to be "lip-synced" (in this case audio to audio though).  Overall - here be=
 (baby) dragons in the glitch-avoidance variation...

In theory you could allow one telephone-event PT and do a renegotiate after=
 an effective codec/PT change.  You might need to be careful about asynchro=
nicity due to the renegotiation and not re-use the same PT for a different =
rate of telephone-event (ping-pong between two).  There would exist a windo=
w when there was a switch where it would be problematic to send telephone-e=
vent (and you'd need to require that both sides switch when one does). Over=
all - here be (big) dragons.

--
Randell Jesup -- rjesup a t mozilla d o t com

_______________________________________________
mmusic mailing list
mmusic@ietf.org
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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: =?Windows-1252?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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On 24 Jan 2014, at 14:59 pm, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote=
:

> Hi,
>=20
> The way forward you proposed sounds good. Looking forward to the updated =
draft. However, I didn=92t really understand how is the lack of coordinatio=
n and different promo algorithms an advantage?

My thought at the time was due to triggered checks it could in some cases b=
e good to have uncoordinated checklists for IPv6 due to less NATs and other=
 restrictive =93things=94. But that is clearly a bad idea. Sorry for the de=
railing.=20

Just posted -05 version of the draft.=20

Name:		draft-reddy-mmusic-ice-happy-eyeballs
Revision:	05
Title:		Happy Eyeballs Extension for ICE
Document date:	2014-01-28
Group:		Individual Submission
Pages:		5
URL:            http://www.ietf.org/internet-drafts/draft-reddy-mmusic-ice-=
happy-eyeballs-05.txt
Status:         https://datatracker.ietf.org/doc/draft-reddy-mmusic-ice-hap=
py-eyeballs/
Htmlized:       http://tools.ietf.org/html/draft-reddy-mmusic-ice-happy-eye=
balls-05
Diff:           http://www.ietf.org/rfcdiff?url2=3Ddraft-reddy-mmusic-ice-h=
appy-eyeballs-05


Not sure if we hit the right balance between keeping it short, but still re=
adable without having all the ICE details fresh in memory.=20

Do we ask for WG adoption, or do we need another MMUSIC session discussing =
this?

.-.
P=E5l-Erik

>=20
>=20
> Cheers,
> Ari
>=20
> On 13 Jan 2014, at 13:46, Pal Martinsen (palmarti) <palmarti@cisco.com> w=
rote:
>=20
>>>=20
>>>=20
>>> Also, if there is no coordination/signaling between the endpoints, what=
 are the odds they'd end up using the same promotion algorithm?
>>>=20
>> Exactly, and that might be an advantage..=20
>=20


From pkyzivat@alum.mit.edu  Tue Jan 28 08:32:07 2014
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 1/28/14 8:02 AM, Christer Holmberg wrote:
> Hi,
>
> I guess it would be good to add some text to BUNDLE, to indicate that you must include the telephone-event PTs for every m- line which may use them. I.e. you cannot indicate telephone-event for one m- line, and then use them on every m- line within the bundle group.
>
> But, I guess there will be no need for people to support telephone-events on more than one m- line to begin with, so...

Why would you say that?

	Thanks,
	Paul

> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>
> On 1/24/2014 10:59 AM, Adam Roach wrote:
>> On 1/24/14 04:29, Magnus Westerlund wrote:
>>> My understanding for the reasoning behind this definition in the RFC
>>> is due to that the events is supposed to be connected to the same
>>> time-line as the audio. Thus it ended up using the same SSRC. To
>>> avoid timestamp rate switching which we know has issues
>>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
>>> tes/)
>>>
>>> it was mandated to provide one PT for each rate negotiated for the
>>> audio payload types.
>>
>> To make sure I understand what you're saying correctly: you are
>> asserting that having four different clock rates for voice codecs
>> requires us to use four different PTs for telephone-event if we're
>> going to have them on the same SSRC. Is that correct?
>
> If those four PT's (not codecs) are accepted (and thus can be switched between on-the-fly), yes.  Multiple clock rates plays hell with RTP timestamps.
>
> You could use a separate SSRC (or m=audio line) for each clock rate instead.  This would likely mean glitching at transitions however, so not a great solution, but works.  You might be able to avoid glitches by making sure you send RTCP for the alternate SSRCs so synchronization info is available, though this may require the receiver to know somehow that the SSRC's need to be "lip-synced" (in this case audio to audio though).  Overall - here be (baby) dragons in the glitch-avoidance variation...
>
> In theory you could allow one telephone-event PT and do a renegotiate after an effective codec/PT change.  You might need to be careful about asynchronicity due to the renegotiation and not re-use the same PT for a different rate of telephone-event (ping-pong between two).  There would exist a window when there was a switch where it would be problematic to send telephone-event (and you'd need to require that both sides switch when one does). Overall - here be (big) dragons.
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
> _______________________________________________
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From christer.holmberg@ericsson.com  Tue Jan 28 10:25:23 2014
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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] Telephone-event and multiple clock rates
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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Hi Paul,

I can't think of a use-case with multiple audio m- lines, where each would =
need to support telephone-events.

Perhaps such use-cases do exist, but I don't they are very common.

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:


On 1/28/14 8:02 AM, Christer Holmberg wrote:
> Hi,
>
> I guess it would be good to add some text to BUNDLE, to indicate that you=
 must include the telephone-event PTs for every m- line which may use them.=
 I.e. you cannot indicate telephone-event for one m- line, and then use the=
m on every m- line within the bundle group.
>
> But, I guess there will be no need for people to support telephone-events=
 on more than one m- line to begin with, so...

Why would you say that?

        Thanks,
        Paul

> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>
> On 1/24/2014 10:59 AM, Adam Roach wrote:
>> On 1/24/14 04:29, Magnus Westerlund wrote:
>>> My understanding for the reasoning behind this definition in the RFC
>>> is due to that the events is supposed to be connected to the same
>>> time-line as the audio. Thus it ended up using the same SSRC. To
>>> avoid timestamp rate switching which we know has issues
>>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
>>> tes/)
>>>
>>> it was mandated to provide one PT for each rate negotiated for the
>>> audio payload types.
>>
>> To make sure I understand what you're saying correctly: you are
>> asserting that having four different clock rates for voice codecs
>> requires us to use four different PTs for telephone-event if we're
>> going to have them on the same SSRC. Is that correct?
>
> If those four PT's (not codecs) are accepted (and thus can be switched be=
tween on-the-fly), yes.  Multiple clock rates plays hell with RTP timestamp=
s.
>
> You could use a separate SSRC (or m=3Daudio line) for each clock rate ins=
tead.  This would likely mean glitching at transitions however, so not a gr=
eat solution, but works.  You might be able to avoid glitches by making sur=
e you send RTCP for the alternate SSRCs so synchronization info is availabl=
e, though this may require the receiver to know somehow that the SSRC's nee=
d to be "lip-synced" (in this case audio to audio though).  Overall - here =
be (baby) dragons in the glitch-avoidance variation...
>
> In theory you could allow one telephone-event PT and do a renegotiate aft=
er an effective codec/PT change.  You might need to be careful about asynch=
ronicity due to the renegotiation and not re-use the same PT for a differen=
t rate of telephone-event (ping-pong between two).  There would exist a win=
dow when there was a switch where it would be problematic to send telephone=
-event (and you'd need to require that both sides switch when one does). Ov=
erall - here be (big) dragons.
>
> --
> Randell Jesup -- rjesup a t mozilla d o t 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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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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On 01/28/2014 02:02 PM, Christer Holmberg wrote:
> Hi,
>
> I guess it would be good to add some text to BUNDLE, to indicate that you must include the telephone-event PTs for every m- line which may use them. I.e. you cannot indicate telephone-event for one m- line, and then use them on every m- line within the bundle group.
>
> But, I guess there will be no need for people to support telephone-events on more than one m- line to begin with, so...

Telephone-events are a hack for gateways into contexts where DTMF
signalling makes sense - which is primarily the telephone network and
telephone-related systems.

In practice, I don't expect gateways into those contexts to support very
many clock rates - they should support G.711's 8000 and Opus' 48K, and
hopefully that's the end of it.

I don't want to worry very much about two extra lines of SDP; let's just
declare (probably in -rtcweb-audio-) that if multiple audio codec
bitrates are offered, and one wishes to use DTMF no matter what codec is
selected, one must offer DTMF at each clock rate offered - each offer
consuming one PT.

>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>
> On 1/24/2014 10:59 AM, Adam Roach wrote:
>> On 1/24/14 04:29, Magnus Westerlund wrote:
>>> My understanding for the reasoning behind this definition in the RFC 
>>> is due to that the events is supposed to be connected to the same 
>>> time-line as the audio. Thus it ended up using the same SSRC. To 
>>> avoid timestamp rate switching which we know has issues
>>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
>>> tes/)
>>>
>>> it was mandated to provide one PT for each rate negotiated for the 
>>> audio payload types.
>> To make sure I understand what you're saying correctly: you are 
>> asserting that having four different clock rates for voice codecs 
>> requires us to use four different PTs for telephone-event if we're 
>> going to have them on the same SSRC. Is that correct?
> If those four PT's (not codecs) are accepted (and thus can be switched between on-the-fly), yes.  Multiple clock rates plays hell with RTP timestamps.
>
> You could use a separate SSRC (or m=audio line) for each clock rate instead.  This would likely mean glitching at transitions however, so not a great solution, but works.  You might be able to avoid glitches by making sure you send RTCP for the alternate SSRCs so synchronization info is available, though this may require the receiver to know somehow that the SSRC's need to be "lip-synced" (in this case audio to audio though).  Overall - here be (baby) dragons in the glitch-avoidance variation...
>
> In theory you could allow one telephone-event PT and do a renegotiate after an effective codec/PT change.  You might need to be careful about asynchronicity due to the renegotiation and not re-use the same PT for a different rate of telephone-event (ping-pong between two).  There would exist a window when there was a switch where it would be problematic to send telephone-event (and you'd need to require that both sides switch when one does). Overall - here be (big) dragons.
>
> --
> Randell Jesup -- rjesup a t mozilla d o t 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


-- 
Surveillance is pervasive. Go Dark.


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Randell Jesup <randell-ietf@jesup.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Telephone-event and multiple clock rates
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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Well I could postulate a webrtc endpoint forming the collection endpoint fo=
r a telephone voting system.

Keith=20

> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Christer Holmberg
> Sent: 28 January 2014 13:03
> To: Randell Jesup; mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>=20
> Hi,
>=20
> I guess it would be good to add some text to BUNDLE, to=20
> indicate that you must include the telephone-event PTs for=20
> every m- line which may use them. I.e. you cannot indicate=20
> telephone-event for one m- line, and then use them on every=20
> m- line within the bundle group.
>=20
> But, I guess there will be no need for people to support=20
> telephone-events on more than one m- line to begin with, so...
>=20
> Regards,
>=20
> Christer
>=20
>=20
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of=20
> Randell Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>=20
> On 1/24/2014 10:59 AM, Adam Roach wrote:
> > On 1/24/14 04:29, Magnus Westerlund wrote:
> >> My understanding for the reasoning behind this definition=20
> in the RFC=20
> >> is due to that the events is supposed to be connected to the same=20
> >> time-line as the audio. Thus it ended up using the same SSRC. To=20
> >> avoid timestamp rate switching which we know has issues=20
> >>=20
> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >> tes/)
> >>
> >> it was mandated to provide one PT for each rate negotiated for the=20
> >> audio payload types.
> >
> > To make sure I understand what you're saying correctly: you are=20
> > asserting that having four different clock rates for voice codecs=20
> > requires us to use four different PTs for telephone-event if we're=20
> > going to have them on the same SSRC. Is that correct?
>=20
> If those four PT's (not codecs) are accepted (and thus can be=20
> switched between on-the-fly), yes.  Multiple clock rates=20
> plays hell with RTP timestamps.
>=20
> You could use a separate SSRC (or m=3Daudio line) for each=20
> clock rate instead.  This would likely mean glitching at=20
> transitions however, so not a great solution, but works.  You=20
> might be able to avoid glitches by making sure you send RTCP=20
> for the alternate SSRCs so synchronization info is available,=20
> though this may require the receiver to know somehow that the=20
> SSRC's need to be "lip-synced" (in this case audio to audio=20
> though).  Overall - here be (baby) dragons in the=20
> glitch-avoidance variation...
>=20
> In theory you could allow one telephone-event PT and do a=20
> renegotiate after an effective codec/PT change.  You might=20
> need to be careful about asynchronicity due to the=20
> renegotiation and not re-use the same PT for a different rate=20
> of telephone-event (ping-pong between two).  There would=20
> exist a window when there was a switch where it would be=20
> problematic to send telephone-event (and you'd need to=20
> require that both sides switch when one does). Overall - here=20
> be (big) dragons.
>=20
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
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> mmusic@ietf.org
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> =

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Randell Jesup <randell-ietf@jesup.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Telephone-event and multiple clock rates
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Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
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> Well I could postulate a webrtc endpoint forming the collection endpoint =
for a telephone voting system.

Sure, if you are going to represent every new incoming "vote call" by addin=
g an audio m- line to a single ongoing session.

Regards,

Christer


> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Christer=20
> Holmberg
> Sent: 28 January 2014 13:03
> To: Randell Jesup; mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>=20
> Hi,
>=20
> I guess it would be good to add some text to BUNDLE, to indicate that=20
> you must include the telephone-event PTs for every m- line which may=20
> use them. I.e. you cannot indicate telephone-event for one m- line,=20
> and then use them on every
> m- line within the bundle group.
>=20
> But, I guess there will be no need for people to support=20
> telephone-events on more than one m- line to begin with, so...
>=20
> Regards,
>=20
> Christer
>=20
>=20
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell=20
> Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>=20
> On 1/24/2014 10:59 AM, Adam Roach wrote:
> > On 1/24/14 04:29, Magnus Westerlund wrote:
> >> My understanding for the reasoning behind this definition
> in the RFC
> >> is due to that the events is supposed to be connected to the same=20
> >> time-line as the audio. Thus it ended up using the same SSRC. To=20
> >> avoid timestamp rate switching which we know has issues
> >>=20
> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >> tes/)
> >>
> >> it was mandated to provide one PT for each rate negotiated for the=20
> >> audio payload types.
> >
> > To make sure I understand what you're saying correctly: you are=20
> > asserting that having four different clock rates for voice codecs=20
> > requires us to use four different PTs for telephone-event if we're=20
> > going to have them on the same SSRC. Is that correct?
>=20
> If those four PT's (not codecs) are accepted (and thus can be switched=20
> between on-the-fly), yes.  Multiple clock rates plays hell with RTP=20
> timestamps.
>=20
> You could use a separate SSRC (or m=3Daudio line) for each clock rate=20
> instead.  This would likely mean glitching at transitions however, so=20
> not a great solution, but works.  You might be able to avoid glitches=20
> by making sure you send RTCP for the alternate SSRCs so=20
> synchronization info is available, though this may require the=20
> receiver to know somehow that the SSRC's need to be "lip-synced" (in=20
> this case audio to audio though).  Overall - here be (baby) dragons in=20
> the glitch-avoidance variation...
>=20
> In theory you could allow one telephone-event PT and do a renegotiate=20
> after an effective codec/PT change.  You might need to be careful=20
> about asynchronicity due to the renegotiation and not re-use the same=20
> PT for a different rate of telephone-event (ping-pong between two). =20
> There would exist a window when there was a switch where it would be=20
> problematic to send telephone-event (and you'd need to require that=20
> both sides switch when one does). Overall - here be (big) dragons.
>=20
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>=20
> _______________________________________________
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>=20

From christer.holmberg@ericsson.com  Wed Jan 29 01:58:16 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, "Adam Gensler (agensler)" <agensler@cisco.com>
Thread-Topic: Review of draft-ietf-mmusic-udptl-dtls-03
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Subject: Re: [MMUSIC] Review of draft-ietf-mmusic-udptl-dtls-03
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Hi Adam,

Again, thanks for your review! See my comments inline.

> After speaking with Gonzalo I've reviewed draft-ietf-mmusic-udptl-dtls-03=
.
> Overall this draft is excellent and appears to cover all the corner=20
> cases I could think of. I do have a few comments, see below:
>=20
> 1. In section 3.1, secure channel establishment, there is the=20
> following text:
>=20
> "If the endpoint supports, and is willing to use, a cipher suite with=20
> an associated certificate, it MUST include an SDP fingerprint=20
> attribute [RFC4572 <http://tools.ietf.org/html/rfc4572>] in the SDP."
>=20
> "If a cipher suite with an associated certificate is selected during=20
> the DTLS handshake, the certificate received during the DTLS handshake=20
> MUST match the fingerprint received in the SDP fingerprint attribute."
>=20
> To me this seems to indicate the particular cipher suites are=20
> associated with a particular certificate. While it's true that the=20
> certificate plays a role in overall TLS negotiation, it does not=20
> provide a list of ciphers to use.

The intention of the text is NOT to indicate that a particular cipher suite=
s are associated with a particular certificate.=20

> I think what this is saying is that
> if the client wishes to use a certificate for DTLS it should provide a=20
> certificate fingerprint. Is that right?

Only if the cipher suite has an associated certificate.

But, e.g. a PSK based cipher suite  (RFC4279) has no associated certificate=
 and thus there is no point of indicating certificate fingerprint for PSK b=
ased cipher suite.

> Given the overall complexity of PKI in general I think being extra clear =
here would help avoid any confusion.

We did spend quite a bit of time on this part of the text, and I can't see =
what modifications I should do in order to make it more clear.

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

> 2. Section 4.1 describes handling anonymous faxes with freshly=20
> generated self-signed certificates. It goes on to say:
>=20
> "...and the content of the subjectAltName attribute inside the=20
> certificate MUST NOT contain information that either allows=20
> correlation or identification of the user making anonymous calls."
>=20
> While the subjectAltName could potentially have identifiable data in=20
> it, it is not always present as a field within the certificate. Some=20
> certificates may have a subjectAltName, some may not. However, all=20
> certificates should have a commonName. I think that is the item that=20
> should be specifically called out as a place to avoid identifiable=20
> information. Or, perhaps, simply stipulating that no fields within the=20
> certificate should contain identifiable information.

That's a good point.

What about something like:

                "...and attributes inside the certificate SHALL NOT contain=
 information that either allows correlation or
                identification of the user making anonymous calls. This is =
particularly important for the subjectAltName=20
                and commonName attributes"

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

> 3. Section 4.3 describes rekeying a connection that is established.=20
> The DTLS RFC describes how to handle this, but gives two options to an=20
> endpoint where out-of-order packets are received that have been=20
> encrypted with different keys; discard the packet, or, retain two sets=20
> of keying material for some recommended interval. DTLS is not specific=20
> as to what to do here. Section 4.3 says the receiver SHOULD maintain=20
> the keying material. Given the nature of T.38, I think it may be=20
> prudent to say the receiver MUST maintain the keying material.
> Dropping and/or discarding
> T.38 packets would never be my preference.

We can change it to MUST.

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

Regards,

Christer



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Cool dude. BTW, how much sqft does wardrobe come to?
On Jan 29, 2014 12:51 AM, <mmusic-request@ietf.org> wrote:

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> Today's Topics:
>
>    1. Re: Telephone-event and multiple clock rates (Christer Holmberg)
>    2. Re: New Version Notification for
>       draft-reddy-mmusic-ice-happy-eyeballs-04.txt
>       (Pal Martinsen (palmarti))
>    3. Re: Telephone-event and multiple clock rates (Paul Kyzivat)
>    4. Re: Telephone-event and multiple clock rates (Christer Holmberg)
>    5. Re: Telephone-event and multiple clock rates (Harald Alvestrand)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 28 Jan 2014 13:02:50 +0000
> From: Christer Holmberg <christer.holmberg@ericsson.com>
> To: Randell Jesup <randell-ietf@jesup.org>, "mmusic@ietf.org"
>         <mmusic@ietf.org>
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> Message-ID:
>         <7594FB04B1934943A5C02806D1A2204B1D14595C@ESESSMB209.ericsson.se>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi,
>
> I guess it would be good to add some text to BUNDLE, to indicate that you
> must include the telephone-event PTs for every m- line which may use them.
> I.e. you cannot indicate telephone-event for one m- line, and then use them
> on every m- line within the bundle group.
>
> But, I guess there will be no need for people to support telephone-events
> on more than one m- line to begin with, so...
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 27. tammikuuta 2014 10:48
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
>
> On 1/24/2014 10:59 AM, Adam Roach wrote:
> > On 1/24/14 04:29, Magnus Westerlund wrote:
> >> My understanding for the reasoning behind this definition in the RFC
> >> is due to that the events is supposed to be connected to the same
> >> time-line as the audio. Thus it ended up using the same SSRC. To
> >> avoid timestamp rate switching which we know has issues
> >> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >> tes/)
> >>
> >> it was mandated to provide one PT for each rate negotiated for the
> >> audio payload types.
> >
> > To make sure I understand what you're saying correctly: you are
> > asserting that having four different clock rates for voice codecs
> > requires us to use four different PTs for telephone-event if we're
> > going to have them on the same SSRC. Is that correct?
>
> If those four PT's (not codecs) are accepted (and thus can be switched
> between on-the-fly), yes.  Multiple clock rates plays hell with RTP
> timestamps.
>
> You could use a separate SSRC (or m=audio line) for each clock rate
> instead.  This would likely mean glitching at transitions however, so not a
> great solution, but works.  You might be able to avoid glitches by making
> sure you send RTCP for the alternate SSRCs so synchronization info is
> available, though this may require the receiver to know somehow that the
> SSRC's need to be "lip-synced" (in this case audio to audio though).
>  Overall - here be (baby) dragons in the glitch-avoidance variation...
>
> In theory you could allow one telephone-event PT and do a renegotiate
> after an effective codec/PT change.  You might need to be careful about
> asynchronicity due to the renegotiation and not re-use the same PT for a
> different rate of telephone-event (ping-pong between two).  There would
> exist a window when there was a switch where it would be problematic to
> send telephone-event (and you'd need to require that both sides switch when
> one does). Overall - here be (big) dragons.
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 28 Jan 2014 13:49:19 +0000
> From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
> To: Ari Ker?nen <ari.keranen@ericsson.com>
> Cc: mmusic <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)"
>         <tireddy@cisco.com>
> Subject: Re: [MMUSIC] New Version Notification for
>         draft-reddy-mmusic-ice-happy-eyeballs-04.txt
> Message-ID: <7C0E6D63-D816-4415-B621-73A0309C6AB2@cisco.com>
> Content-Type: text/plain; charset="Windows-1252"
>
>
> On 24 Jan 2014, at 14:59 pm, Ari Ker?nen <ari.keranen@ericsson.com> wrote:
>
> > Hi,
> >
> > The way forward you proposed sounds good. Looking forward to the updated
> draft. However, I didn?t really understand how is the lack of coordination
> and different promo algorithms an advantage?
>
> My thought at the time was due to triggered checks it could in some cases
> be good to have uncoordinated checklists for IPv6 due to less NATs and
> other restrictive ?things?. But that is clearly a bad idea. Sorry for the
> derailing.
>
> Just posted -05 version of the draft.
>
> Name:           draft-reddy-mmusic-ice-happy-eyeballs
> Revision:       05
> Title:          Happy Eyeballs Extension for ICE
> Document date:  2014-01-28
> Group:          Individual Submission
> Pages:          5
> URL:
> http://www.ietf.org/internet-drafts/draft-reddy-mmusic-ice-happy-eyeballs-05.txt
> Status:
> https://datatracker.ietf.org/doc/draft-reddy-mmusic-ice-happy-eyeballs/
> Htmlized:
> http://tools.ietf.org/html/draft-reddy-mmusic-ice-happy-eyeballs-05
> Diff:
> http://www.ietf.org/rfcdiff?url2=draft-reddy-mmusic-ice-happy-eyeballs-05
>
>
> Not sure if we hit the right balance between keeping it short, but still
> readable without having all the ICE details fresh in memory.
>
> Do we ask for WG adoption, or do we need another MMUSIC session discussing
> this?
>
> .-.
> P?l-Erik
>
> >
> >
> > Cheers,
> > Ari
> >
> > On 13 Jan 2014, at 13:46, Pal Martinsen (palmarti) <palmarti@cisco.com>
> wrote:
> >
> >>>
> >>>
> >>> Also, if there is no coordination/signaling between the endpoints,
> what are the odds they'd end up using the same promotion algorithm?
> >>>
> >> Exactly, and that might be an advantage..
> >
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 28 Jan 2014 11:32:02 -0500
> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> Message-ID: <52E7DB82.2080907@alum.mit.edu>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 1/28/14 8:02 AM, Christer Holmberg wrote:
> > Hi,
> >
> > I guess it would be good to add some text to BUNDLE, to indicate that
> you must include the telephone-event PTs for every m- line which may use
> them. I.e. you cannot indicate telephone-event for one m- line, and then
> use them on every m- line within the bundle group.
> >
> > But, I guess there will be no need for people to support
> telephone-events on more than one m- line to begin with, so...
>
> Why would you say that?
>
>         Thanks,
>         Paul
>
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> > Sent: 27. tammikuuta 2014 10:48
> > To: mmusic@ietf.org
> > Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> >
> > On 1/24/2014 10:59 AM, Adam Roach wrote:
> >> On 1/24/14 04:29, Magnus Westerlund wrote:
> >>> My understanding for the reasoning behind this definition in the RFC
> >>> is due to that the events is supposed to be connected to the same
> >>> time-line as the audio. Thus it ended up using the same SSRC. To
> >>> avoid timestamp rate switching which we know has issues
> >>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >>> tes/)
> >>>
> >>> it was mandated to provide one PT for each rate negotiated for the
> >>> audio payload types.
> >>
> >> To make sure I understand what you're saying correctly: you are
> >> asserting that having four different clock rates for voice codecs
> >> requires us to use four different PTs for telephone-event if we're
> >> going to have them on the same SSRC. Is that correct?
> >
> > If those four PT's (not codecs) are accepted (and thus can be switched
> between on-the-fly), yes.  Multiple clock rates plays hell with RTP
> timestamps.
> >
> > You could use a separate SSRC (or m=audio line) for each clock rate
> instead.  This would likely mean glitching at transitions however, so not a
> great solution, but works.  You might be able to avoid glitches by making
> sure you send RTCP for the alternate SSRCs so synchronization info is
> available, though this may require the receiver to know somehow that the
> SSRC's need to be "lip-synced" (in this case audio to audio though).
>  Overall - here be (baby) dragons in the glitch-avoidance variation...
> >
> > In theory you could allow one telephone-event PT and do a renegotiate
> after an effective codec/PT change.  You might need to be careful about
> asynchronicity due to the renegotiation and not re-use the same PT for a
> different rate of telephone-event (ping-pong between two).  There would
> exist a window when there was a switch where it would be problematic to
> send telephone-event (and you'd need to require that both sides switch when
> one does). Overall - here be (big) dragons.
> >
> > --
> > Randell Jesup -- rjesup a t mozilla d o t 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
> >
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 28 Jan 2014 18:25:17 +0000
> From: Christer Holmberg <christer.holmberg@ericsson.com>
> To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org"
>         <mmusic@ietf.org>
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> Message-ID: <9dt0s2vbp3f9xidieks341ev.1390933514153@email.android.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Paul,
>
> I can't think of a use-case with multiple audio m- lines, where each would
> need to support telephone-events.
>
> Perhaps such use-cases do exist, but I don't they are very common.
>
> Regards,
>
> Christer
>
> Sent from my Sony Ericsson Xperia arc S
>
> Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>
> On 1/28/14 8:02 AM, Christer Holmberg wrote:
> > Hi,
> >
> > I guess it would be good to add some text to BUNDLE, to indicate that
> you must include the telephone-event PTs for every m- line which may use
> them. I.e. you cannot indicate telephone-event for one m- line, and then
> use them on every m- line within the bundle group.
> >
> > But, I guess there will be no need for people to support
> telephone-events on more than one m- line to begin with, so...
>
> Why would you say that?
>
>         Thanks,
>         Paul
>
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> > Sent: 27. tammikuuta 2014 10:48
> > To: mmusic@ietf.org
> > Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> >
> > On 1/24/2014 10:59 AM, Adam Roach wrote:
> >> On 1/24/14 04:29, Magnus Westerlund wrote:
> >>> My understanding for the reasoning behind this definition in the RFC
> >>> is due to that the events is supposed to be connected to the same
> >>> time-line as the audio. Thus it ended up using the same SSRC. To
> >>> avoid timestamp rate switching which we know has issues
> >>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >>> tes/)
> >>>
> >>> it was mandated to provide one PT for each rate negotiated for the
> >>> audio payload types.
> >>
> >> To make sure I understand what you're saying correctly: you are
> >> asserting that having four different clock rates for voice codecs
> >> requires us to use four different PTs for telephone-event if we're
> >> going to have them on the same SSRC. Is that correct?
> >
> > If those four PT's (not codecs) are accepted (and thus can be switched
> between on-the-fly), yes.  Multiple clock rates plays hell with RTP
> timestamps.
> >
> > You could use a separate SSRC (or m=audio line) for each clock rate
> instead.  This would likely mean glitching at transitions however, so not a
> great solution, but works.  You might be able to avoid glitches by making
> sure you send RTCP for the alternate SSRCs so synchronization info is
> available, though this may require the receiver to know somehow that the
> SSRC's need to be "lip-synced" (in this case audio to audio though).
>  Overall - here be (baby) dragons in the glitch-avoidance variation...
> >
> > In theory you could allow one telephone-event PT and do a renegotiate
> after an effective codec/PT change.  You might need to be careful about
> asynchronicity due to the renegotiation and not re-use the same PT for a
> different rate of telephone-event (ping-pong between two).  There would
> exist a window when there was a switch where it would be problematic to
> send telephone-event (and you'd need to require that both sides switch when
> one does). Overall - here be (big) dragons.
> >
> > --
> > Randell Jesup -- rjesup a t mozilla d o t 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
> >
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 28 Jan 2014 19:01:46 +0100
> From: Harald Alvestrand <harald@alvestrand.no>
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> Message-ID: <52E7F08A.6070200@alvestrand.no>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 01/28/2014 02:02 PM, Christer Holmberg wrote:
> > Hi,
> >
> > I guess it would be good to add some text to BUNDLE, to indicate that
> you must include the telephone-event PTs for every m- line which may use
> them. I.e. you cannot indicate telephone-event for one m- line, and then
> use them on every m- line within the bundle group.
> >
> > But, I guess there will be no need for people to support
> telephone-events on more than one m- line to begin with, so...
>
> Telephone-events are a hack for gateways into contexts where DTMF
> signalling makes sense - which is primarily the telephone network and
> telephone-related systems.
>
> In practice, I don't expect gateways into those contexts to support very
> many clock rates - they should support G.711's 8000 and Opus' 48K, and
> hopefully that's the end of it.
>
> I don't want to worry very much about two extra lines of SDP; let's just
> declare (probably in -rtcweb-audio-) that if multiple audio codec
> bitrates are offered, and one wishes to use DTMF no matter what codec is
> selected, one must offer DTMF at each clock rate offered - each offer
> consuming one PT.
>
> >
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of Randell Jesup
> > Sent: 27. tammikuuta 2014 10:48
> > To: mmusic@ietf.org
> > Subject: Re: [MMUSIC] Telephone-event and multiple clock rates
> >
> > On 1/24/2014 10:59 AM, Adam Roach wrote:
> >> On 1/24/14 04:29, Magnus Westerlund wrote:
> >>> My understanding for the reasoning behind this definition in the RFC
> >>> is due to that the events is supposed to be connected to the same
> >>> time-line as the audio. Thus it ended up using the same SSRC. To
> >>> avoid timestamp rate switching which we know has issues
> >>> (https://datatracker.ietf.org/doc/draft-ietf-avtext-multiple-clock-ra
> >>> tes/)
> >>>
> >>> it was mandated to provide one PT for each rate negotiated for the
> >>> audio payload types.
> >> To make sure I understand what you're saying correctly: you are
> >> asserting that having four different clock rates for voice codecs
> >> requires us to use four different PTs for telephone-event if we're
> >> going to have them on the same SSRC. Is that correct?
> > If those four PT's (not codecs) are accepted (and thus can be switched
> between on-the-fly), yes.  Multiple clock rates plays hell with RTP
> timestamps.
> >
> > You could use a separate SSRC (or m=audio line) for each clock rate
> instead.  This would likely mean glitching at transitions however, so not a
> great solution, but works.  You might be able to avoid glitches by making
> sure you send RTCP for the alternate SSRCs so synchronization info is
> available, though this may require the receiver to know somehow that the
> SSRC's need to be "lip-synced" (in this case audio to audio though).
>  Overall - here be (baby) dragons in the glitch-avoidance variation...
> >
> > In theory you could allow one telephone-event PT and do a renegotiate
> after an effective codec/PT change.  You might need to be careful about
> asynchronicity due to the renegotiation and not re-use the same PT for a
> different rate of telephone-event (ping-pong between two).  There would
> exist a window when there was a switch where it would be problematic to
> send telephone-event (and you'd need to require that both sides switch when
> one does). Overall - here be (big) dragons.
> >
> > --
> > Randell Jesup -- rjesup a t mozilla d o t 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
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
>
> ------------------------------
>
> Subject: Digest Footer
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> ------------------------------
>
> End of mmusic Digest, Vol 117, Issue 19
> ***************************************
>

--089e0160d2b85f48d004f11b6bfe
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">Cool dude. BTW, how much sqft does wardrobe come to?</p>
<div class=3D"gmail_quote">On Jan 29, 2014 12:51 AM,  &lt;<a href=3D"mailto=
:mmusic-request@ietf.org">mmusic-request@ietf.org</a>&gt; wrote:<br type=3D=
"attribution"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
Send mmusic mailing list submissions to<br>
=A0 =A0 =A0 =A0 <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<br>
To subscribe or unsubscribe via the World Wide Web, visit<br>
=A0 =A0 =A0 =A0 <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
or, via email, send a message with subject or body &#39;help&#39; to<br>
=A0 =A0 =A0 =A0 <a href=3D"mailto:mmusic-request@ietf.org">mmusic-request@i=
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<br>
You can reach the person managing the list at<br>
=A0 =A0 =A0 =A0 <a href=3D"mailto:mmusic-owner@ietf.org">mmusic-owner@ietf.=
org</a><br>
<br>
When replying, please edit your Subject line so it is more specific<br>
than &quot;Re: Contents of mmusic digest...&quot;<br>
<br>
<br>
Today&#39;s Topics:<br>
<br>
=A0 =A01. Re: Telephone-event and multiple clock rates (Christer Holmberg)<=
br>
=A0 =A02. Re: New Version Notification for<br>
=A0 =A0 =A0 draft-reddy-mmusic-ice-happy-eyeballs-04.txt<br>
=A0 =A0 =A0 (Pal Martinsen (palmarti))<br>
=A0 =A03. Re: Telephone-event and multiple clock rates (Paul Kyzivat)<br>
=A0 =A04. Re: Telephone-event and multiple clock rates (Christer Holmberg)<=
br>
=A0 =A05. Re: Telephone-event and multiple clock rates (Harald Alvestrand)<=
br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Tue, 28 Jan 2014 13:02:50 +0000<br>
From: Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m">christer.holmberg@ericsson.com</a>&gt;<br>
To: Randell Jesup &lt;<a href=3D"mailto:randell-ietf@jesup.org">randell-iet=
f@jesup.org</a>&gt;, &quot;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.o=
rg</a>&quot;<br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&=
gt;<br>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
Message-ID:<br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"mailto:7594FB04B1934943A5C02806D1A2204B1D145=
95C@ESESSMB209.ericsson.se">7594FB04B1934943A5C02806D1A2204B1D14595C@ESESSM=
B209.ericsson.se</a>&gt;<br>
Content-Type: text/plain; charset=3D&quot;us-ascii&quot;<br>
<br>
Hi,<br>
<br>
I guess it would be good to add some text to BUNDLE, to indicate that you m=
ust include the telephone-event PTs for every m- line which may use them. I=
.e. you cannot indicate telephone-event for one m- line, and then use them =
on every m- line within the bundle group.<br>

<br>
But, I guess there will be no need for people to support telephone-events o=
n more than one m- line to begin with, so...<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
-----Original Message-----<br>
From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-boun=
ces@ietf.org</a>] On Behalf Of Randell Jesup<br>
Sent: 27. tammikuuta 2014 10:48<br>
To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
<br>
On 1/24/2014 10:59 AM, Adam Roach wrote:<br>
&gt; On 1/24/14 04:29, Magnus Westerlund wrote:<br>
&gt;&gt; My understanding for the reasoning behind this definition in the R=
FC<br>
&gt;&gt; is due to that the events is supposed to be connected to the same<=
br>
&gt;&gt; time-line as the audio. Thus it ended up using the same SSRC. To<b=
r>
&gt;&gt; avoid timestamp rate switching which we know has issues<br>
&gt;&gt; (<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtext-mul=
tiple-clock-ra" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ie=
tf-avtext-multiple-clock-ra</a><br>
&gt;&gt; tes/)<br>
&gt;&gt;<br>
&gt;&gt; it was mandated to provide one PT for each rate negotiated for the=
<br>
&gt;&gt; audio payload types.<br>
&gt;<br>
&gt; To make sure I understand what you&#39;re saying correctly: you are<br=
>
&gt; asserting that having four different clock rates for voice codecs<br>
&gt; requires us to use four different PTs for telephone-event if we&#39;re=
<br>
&gt; going to have them on the same SSRC. Is that correct?<br>
<br>
If those four PT&#39;s (not codecs) are accepted (and thus can be switched =
between on-the-fly), yes. =A0Multiple clock rates plays hell with RTP times=
tamps.<br>
<br>
You could use a separate SSRC (or m=3Daudio line) for each clock rate inste=
ad. =A0This would likely mean glitching at transitions however, so not a gr=
eat solution, but works. =A0You might be able to avoid glitches by making s=
ure you send RTCP for the alternate SSRCs so synchronization info is availa=
ble, though this may require the receiver to know somehow that the SSRC&#39=
;s need to be &quot;lip-synced&quot; (in this case audio to audio though). =
=A0Overall - here be (baby) dragons in the glitch-avoidance variation...<br=
>

<br>
In theory you could allow one telephone-event PT and do a renegotiate after=
 an effective codec/PT change. =A0You might need to be careful about asynch=
ronicity due to the renegotiation and not re-use the same PT for a differen=
t rate of telephone-event (ping-pong between two). =A0There would exist a w=
indow when there was a switch where it would be problematic to send telepho=
ne-event (and you&#39;d need to require that both sides switch when one doe=
s). Overall - here be (big) dragons.<br>

<br>
--<br>
Randell Jesup -- rjesup a t mozilla d o t com<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>
<br>
------------------------------<br>
<br>
Message: 2<br>
Date: Tue, 28 Jan 2014 13:49:19 +0000<br>
From: &quot;Pal Martinsen (palmarti)&quot; &lt;<a href=3D"mailto:palmarti@c=
isco.com">palmarti@cisco.com</a>&gt;<br>
To: Ari Ker?nen &lt;<a href=3D"mailto:ari.keranen@ericsson.com">ari.keranen=
@ericsson.com</a>&gt;<br>
Cc: mmusic &lt;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&gt;, =
&quot;Tirumaleswar Reddy \(tireddy\)&quot;<br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"mailto:tireddy@cisco.com">tireddy@cisco.com<=
/a>&gt;<br>
Subject: Re: [MMUSIC] New Version Notification for<br>
=A0 =A0 =A0 =A0 draft-reddy-mmusic-ice-happy-eyeballs-04.txt<br>
Message-ID: &lt;<a href=3D"mailto:7C0E6D63-D816-4415-B621-73A0309C6AB2@cisc=
o.com">7C0E6D63-D816-4415-B621-73A0309C6AB2@cisco.com</a>&gt;<br>
Content-Type: text/plain; charset=3D&quot;Windows-1252&quot;<br>
<br>
<br>
On 24 Jan 2014, at 14:59 pm, Ari Ker?nen &lt;<a href=3D"mailto:ari.keranen@=
ericsson.com">ari.keranen@ericsson.com</a>&gt; wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; The way forward you proposed sounds good. Looking forward to the updat=
ed draft. However, I didn?t really understand how is the lack of coordinati=
on and different promo algorithms an advantage?<br>
<br>
My thought at the time was due to triggered checks it could in some cases b=
e good to have uncoordinated checklists for IPv6 due to less NATs and other=
 restrictive ?things?. But that is clearly a bad idea. Sorry for the derail=
ing.<br>

<br>
Just posted -05 version of the draft.<br>
<br>
Name: =A0 =A0 =A0 =A0 =A0 draft-reddy-mmusic-ice-happy-eyeballs<br>
Revision: =A0 =A0 =A0 05<br>
Title: =A0 =A0 =A0 =A0 =A0Happy Eyeballs Extension for ICE<br>
Document date: =A02014-01-28<br>
Group: =A0 =A0 =A0 =A0 =A0Individual Submission<br>
Pages: =A0 =A0 =A0 =A0 =A05<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/internet-drafts/=
draft-reddy-mmusic-ice-happy-eyeballs-05.txt" target=3D"_blank">http://www.=
ietf.org/internet-drafts/draft-reddy-mmusic-ice-happy-eyeballs-05.txt</a><b=
r>
Status: =A0 =A0 =A0 =A0 <a href=3D"https://datatracker.ietf.org/doc/draft-r=
eddy-mmusic-ice-happy-eyeballs/" target=3D"_blank">https://datatracker.ietf=
.org/doc/draft-reddy-mmusic-ice-happy-eyeballs/</a><br>
Htmlized: =A0 =A0 =A0 <a href=3D"http://tools.ietf.org/html/draft-reddy-mmu=
sic-ice-happy-eyeballs-05" target=3D"_blank">http://tools.ietf.org/html/dra=
ft-reddy-mmusic-ice-happy-eyeballs-05</a><br>
Diff: =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddra=
ft-reddy-mmusic-ice-happy-eyeballs-05" target=3D"_blank">http://www.ietf.or=
g/rfcdiff?url2=3Ddraft-reddy-mmusic-ice-happy-eyeballs-05</a><br>
<br>
<br>
Not sure if we hit the right balance between keeping it short, but still re=
adable without having all the ICE details fresh in memory.<br>
<br>
Do we ask for WG adoption, or do we need another MMUSIC session discussing =
this?<br>
<br>
.-.<br>
P?l-Erik<br>
<br>
&gt;<br>
&gt;<br>
&gt; Cheers,<br>
&gt; Ari<br>
&gt;<br>
&gt; On 13 Jan 2014, at 13:46, Pal Martinsen (palmarti) &lt;<a href=3D"mail=
to:palmarti@cisco.com">palmarti@cisco.com</a>&gt; wrote:<br>
&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Also, if there is no coordination/signaling between the endpoi=
nts, what are the odds they&#39;d end up using the same promotion algorithm=
?<br>
&gt;&gt;&gt;<br>
&gt;&gt; Exactly, and that might be an advantage..<br>
&gt;<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 3<br>
Date: Tue, 28 Jan 2014 11:32:02 -0500<br>
From: Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@al=
um.mit.edu</a>&gt;<br>
To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
Message-ID: &lt;<a href=3D"mailto:52E7DB82.2080907@alum.mit.edu">52E7DB82.2=
080907@alum.mit.edu</a>&gt;<br>
Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed<br>
<br>
On 1/28/14 8:02 AM, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt; I guess it would be good to add some text to BUNDLE, to indicate that =
you must include the telephone-event PTs for every m- line which may use th=
em. I.e. you cannot indicate telephone-event for one m- line, and then use =
them on every m- line within the bundle group.<br>

&gt;<br>
&gt; But, I guess there will be no need for people to support telephone-eve=
nts on more than one m- line to begin with, so...<br>
<br>
Why would you say that?<br>
<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic=
-bounces@ietf.org</a>] On Behalf Of Randell Jesup<br>
&gt; Sent: 27. tammikuuta 2014 10:48<br>
&gt; To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
&gt;<br>
&gt; On 1/24/2014 10:59 AM, Adam Roach wrote:<br>
&gt;&gt; On 1/24/14 04:29, Magnus Westerlund wrote:<br>
&gt;&gt;&gt; My understanding for the reasoning behind this definition in t=
he RFC<br>
&gt;&gt;&gt; is due to that the events is supposed to be connected to the s=
ame<br>
&gt;&gt;&gt; time-line as the audio. Thus it ended up using the same SSRC. =
To<br>
&gt;&gt;&gt; avoid timestamp rate switching which we know has issues<br>
&gt;&gt;&gt; (<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtext=
-multiple-clock-ra" target=3D"_blank">https://datatracker.ietf.org/doc/draf=
t-ietf-avtext-multiple-clock-ra</a><br>
&gt;&gt;&gt; tes/)<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; it was mandated to provide one PT for each rate negotiated for=
 the<br>
&gt;&gt;&gt; audio payload types.<br>
&gt;&gt;<br>
&gt;&gt; To make sure I understand what you&#39;re saying correctly: you ar=
e<br>
&gt;&gt; asserting that having four different clock rates for voice codecs<=
br>
&gt;&gt; requires us to use four different PTs for telephone-event if we&#3=
9;re<br>
&gt;&gt; going to have them on the same SSRC. Is that correct?<br>
&gt;<br>
&gt; If those four PT&#39;s (not codecs) are accepted (and thus can be swit=
ched between on-the-fly), yes. =A0Multiple clock rates plays hell with RTP =
timestamps.<br>
&gt;<br>
&gt; You could use a separate SSRC (or m=3Daudio line) for each clock rate =
instead. =A0This would likely mean glitching at transitions however, so not=
 a great solution, but works. =A0You might be able to avoid glitches by mak=
ing sure you send RTCP for the alternate SSRCs so synchronization info is a=
vailable, though this may require the receiver to know somehow that the SSR=
C&#39;s need to be &quot;lip-synced&quot; (in this case audio to audio thou=
gh). =A0Overall - here be (baby) dragons in the glitch-avoidance variation.=
..<br>

&gt;<br>
&gt; In theory you could allow one telephone-event PT and do a renegotiate =
after an effective codec/PT change. =A0You might need to be careful about a=
synchronicity due to the renegotiation and not re-use the same PT for a dif=
ferent rate of telephone-event (ping-pong between two). =A0There would exis=
t a window when there was a switch where it would be problematic to send te=
lephone-event (and you&#39;d need to require that both sides switch when on=
e does). Overall - here be (big) dragons.<br>

&gt;<br>
&gt; --<br>
&gt; Randell Jesup -- rjesup a t mozilla d o t com<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>
&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>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Tue, 28 Jan 2014 18:25:17 +0000<br>
From: Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m">christer.holmberg@ericsson.com</a>&gt;<br>
To: Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum=
.mit.edu</a>&gt;, &quot;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org<=
/a>&quot;<br>
=A0 =A0 =A0 =A0 &lt;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&=
gt;<br>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
Message-ID: &lt;<a href=3D"mailto:9dt0s2vbp3f9xidieks341ev.1390933514153@em=
ail.android.com">9dt0s2vbp3f9xidieks341ev.1390933514153@email.android.com</=
a>&gt;<br>
Content-Type: text/plain; charset=3D&quot;us-ascii&quot;<br>
<br>
Hi Paul,<br>
<br>
I can&#39;t think of a use-case with multiple audio m- lines, where each wo=
uld need to support telephone-events.<br>
<br>
Perhaps such use-cases do exist, but I don&#39;t they are very common.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Sony Ericsson Xperia arc S<br>
<br>
Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit=
.edu</a>&gt; wrote:<br>
<br>
<br>
On 1/28/14 8:02 AM, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt; I guess it would be good to add some text to BUNDLE, to indicate that =
you must include the telephone-event PTs for every m- line which may use th=
em. I.e. you cannot indicate telephone-event for one m- line, and then use =
them on every m- line within the bundle group.<br>

&gt;<br>
&gt; But, I guess there will be no need for people to support telephone-eve=
nts on more than one m- line to begin with, so...<br>
<br>
Why would you say that?<br>
<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<br>
<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic=
-bounces@ietf.org</a>] On Behalf Of Randell Jesup<br>
&gt; Sent: 27. tammikuuta 2014 10:48<br>
&gt; To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
&gt;<br>
&gt; On 1/24/2014 10:59 AM, Adam Roach wrote:<br>
&gt;&gt; On 1/24/14 04:29, Magnus Westerlund wrote:<br>
&gt;&gt;&gt; My understanding for the reasoning behind this definition in t=
he RFC<br>
&gt;&gt;&gt; is due to that the events is supposed to be connected to the s=
ame<br>
&gt;&gt;&gt; time-line as the audio. Thus it ended up using the same SSRC. =
To<br>
&gt;&gt;&gt; avoid timestamp rate switching which we know has issues<br>
&gt;&gt;&gt; (<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtext=
-multiple-clock-ra" target=3D"_blank">https://datatracker.ietf.org/doc/draf=
t-ietf-avtext-multiple-clock-ra</a><br>
&gt;&gt;&gt; tes/)<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; it was mandated to provide one PT for each rate negotiated for=
 the<br>
&gt;&gt;&gt; audio payload types.<br>
&gt;&gt;<br>
&gt;&gt; To make sure I understand what you&#39;re saying correctly: you ar=
e<br>
&gt;&gt; asserting that having four different clock rates for voice codecs<=
br>
&gt;&gt; requires us to use four different PTs for telephone-event if we&#3=
9;re<br>
&gt;&gt; going to have them on the same SSRC. Is that correct?<br>
&gt;<br>
&gt; If those four PT&#39;s (not codecs) are accepted (and thus can be swit=
ched between on-the-fly), yes. =A0Multiple clock rates plays hell with RTP =
timestamps.<br>
&gt;<br>
&gt; You could use a separate SSRC (or m=3Daudio line) for each clock rate =
instead. =A0This would likely mean glitching at transitions however, so not=
 a great solution, but works. =A0You might be able to avoid glitches by mak=
ing sure you send RTCP for the alternate SSRCs so synchronization info is a=
vailable, though this may require the receiver to know somehow that the SSR=
C&#39;s need to be &quot;lip-synced&quot; (in this case audio to audio thou=
gh). =A0Overall - here be (baby) dragons in the glitch-avoidance variation.=
..<br>

&gt;<br>
&gt; In theory you could allow one telephone-event PT and do a renegotiate =
after an effective codec/PT change. =A0You might need to be careful about a=
synchronicity due to the renegotiation and not re-use the same PT for a dif=
ferent rate of telephone-event (ping-pong between two). =A0There would exis=
t a window when there was a switch where it would be problematic to send te=
lephone-event (and you&#39;d need to require that both sides switch when on=
e does). Overall - here be (big) dragons.<br>

&gt;<br>
&gt; --<br>
&gt; Randell Jesup -- rjesup a t mozilla d o t com<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>
&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>
<br>
------------------------------<br>
<br>
Message: 5<br>
Date: Tue, 28 Jan 2014 19:01:46 +0100<br>
From: Harald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no">harald@=
alvestrand.no</a>&gt;<br>
To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
Message-ID: &lt;<a href=3D"mailto:52E7F08A.6070200@alvestrand.no">52E7F08A.=
6070200@alvestrand.no</a>&gt;<br>
Content-Type: text/plain; charset=3DISO-8859-1<br>
<br>
On 01/28/2014 02:02 PM, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt; I guess it would be good to add some text to BUNDLE, to indicate that =
you must include the telephone-event PTs for every m- line which may use th=
em. I.e. you cannot indicate telephone-event for one m- line, and then use =
them on every m- line within the bundle group.<br>

&gt;<br>
&gt; But, I guess there will be no need for people to support telephone-eve=
nts on more than one m- line to begin with, so...<br>
<br>
Telephone-events are a hack for gateways into contexts where DTMF<br>
signalling makes sense - which is primarily the telephone network and<br>
telephone-related systems.<br>
<br>
In practice, I don&#39;t expect gateways into those contexts to support ver=
y<br>
many clock rates - they should support G.711&#39;s 8000 and Opus&#39; 48K, =
and<br>
hopefully that&#39;s the end of it.<br>
<br>
I don&#39;t want to worry very much about two extra lines of SDP; let&#39;s=
 just<br>
declare (probably in -rtcweb-audio-) that if multiple audio codec<br>
bitrates are offered, and one wishes to use DTMF no matter what codec is<br=
>
selected, one must offer DTMF at each clock rate offered - each offer<br>
consuming one PT.<br>
<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic [mailto:<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic=
-bounces@ietf.org</a>] On Behalf Of Randell Jesup<br>
&gt; Sent: 27. tammikuuta 2014 10:48<br>
&gt; To: <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; Subject: Re: [MMUSIC] Telephone-event and multiple clock rates<br>
&gt;<br>
&gt; On 1/24/2014 10:59 AM, Adam Roach wrote:<br>
&gt;&gt; On 1/24/14 04:29, Magnus Westerlund wrote:<br>
&gt;&gt;&gt; My understanding for the reasoning behind this definition in t=
he RFC<br>
&gt;&gt;&gt; is due to that the events is supposed to be connected to the s=
ame<br>
&gt;&gt;&gt; time-line as the audio. Thus it ended up using the same SSRC. =
To<br>
&gt;&gt;&gt; avoid timestamp rate switching which we know has issues<br>
&gt;&gt;&gt; (<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avtext=
-multiple-clock-ra" target=3D"_blank">https://datatracker.ietf.org/doc/draf=
t-ietf-avtext-multiple-clock-ra</a><br>
&gt;&gt;&gt; tes/)<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; it was mandated to provide one PT for each rate negotiated for=
 the<br>
&gt;&gt;&gt; audio payload types.<br>
&gt;&gt; To make sure I understand what you&#39;re saying correctly: you ar=
e<br>
&gt;&gt; asserting that having four different clock rates for voice codecs<=
br>
&gt;&gt; requires us to use four different PTs for telephone-event if we&#3=
9;re<br>
&gt;&gt; going to have them on the same SSRC. Is that correct?<br>
&gt; If those four PT&#39;s (not codecs) are accepted (and thus can be swit=
ched between on-the-fly), yes. =A0Multiple clock rates plays hell with RTP =
timestamps.<br>
&gt;<br>
&gt; You could use a separate SSRC (or m=3Daudio line) for each clock rate =
instead. =A0This would likely mean glitching at transitions however, so not=
 a great solution, but works. =A0You might be able to avoid glitches by mak=
ing sure you send RTCP for the alternate SSRCs so synchronization info is a=
vailable, though this may require the receiver to know somehow that the SSR=
C&#39;s need to be &quot;lip-synced&quot; (in this case audio to audio thou=
gh). =A0Overall - here be (baby) dragons in the glitch-avoidance variation.=
..<br>

&gt;<br>
&gt; In theory you could allow one telephone-event PT and do a renegotiate =
after an effective codec/PT change. =A0You might need to be careful about a=
synchronicity due to the renegotiation and not re-use the same PT for a dif=
ferent rate of telephone-event (ping-pong between two). =A0There would exis=
t a window when there was a switch where it would be problematic to send te=
lephone-event (and you&#39;d need to require that both sides switch when on=
e does). Overall - here be (big) dragons.<br>

&gt;<br>
&gt; --<br>
&gt; Randell Jesup -- rjesup a t mozilla d o t com<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>
&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>
<br>
<br>
--<br>
Surveillance is pervasive. Go Dark.<br>
<br>
<br>
<br>
------------------------------<br>
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URL: http://datatracker.ietf.org/wg/mmusic/charter/

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Changed milestone "Submit a framework for SDP attributes when
multiplexing as Proposed Standard", set state to active from review,
accepting new milestone.

URL: http://datatracker.ietf.org/wg/mmusic/charter/

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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           : Offer/Answer Considerations for G723 Annex A and =
G729 Annex B
        Authors         : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-05.txt
	Pages           : 8
	Date            : 2014-01-29

Abstract:
   This document provides the offer/answer considerations for the annexa
   parameter of G723 and the annexb parameter of G729, G729D and G729E
   when the value of the annexa or annexb parameter does not match in
   the Session Description protocol (SDP) offer and answer.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-g723-g729-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-05


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:
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: I-D Action: draft-ietf-mmusic-sdp-g723-g729-05.txt
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WG,

This addresses the IETF Last Call and Gen-ART review comments.

Muthu

-----Original Message-----
From: I-D-Announce [mailto:i-d-announce-bounces@ietf.org] On Behalf Of inte=
rnet-drafts@ietf.org
Sent: Wednesday, January 29, 2014 11:34 PM
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-05.txt

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           : Offer/Answer Considerations for G723 Annex A and =
G729 Annex B
        Authors         : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-05.txt
	Pages           : 8
	Date            : 2014-01-29

Abstract:
   This document provides the offer/answer considerations for the annexa
   parameter of G723 and the annexb parameter of G729, G729D and G729E
   when the value of the annexa or annexb parameter does not match in
   the Session Description protocol (SDP) offer and answer.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-g723-g729/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-g723-g729-05

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-05

Please note that it may take a couple of minutes from the time of submissio=
n
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/

_______________________________________________
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Flemming Andreasen (fandreas)" <fandreas@cisco.com>, SM <sm@resistor.net>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] Last Call: <draft-ietf-mmusic-sdp-g723-g729-04.txt> (Offer/Answer Considerations for G723 Annex A and G729 Annex	B) to Proposed Standard
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References: <20131030131748.6987.86198.idtracker@ietfa.amsl.com> <6.2.5.6.2.20131030185231.0ddfb7a8@resistor.net> <00a101ced981$77414d60$65c3e820$@co.in> <00ab01cee555$05f53560$11dfa020$@co.in> <6.2.5.6.2.20131119110213.0cb0dd40@resistor.net> <003301cee952$852cd120$8f867360$@co.in> <6.2.5.6.2.20131129000812.0bdaa088@resistor.net> <016d01ceed31$aa7a1130$ff6e3390$@co.in> <6.2.5.6.2.20131130003721.0d65a980@resistor.net> <E721D8C6A2E1544DB2DEBC313AF54DE22436AD4D@xmb-rcd-x02.cisco.com> <6.2.5.6.2.20131203011102.0db5d8c0@resistor.net> <E721D8C6A2E1544DB2DEBC313AF54DE2243737E1@xmb-rcd-x02.cisco.com> <6.2.5.6.2.20131204072722.0e064590@resistor.net> <E721D8C6A2E1544DB2DEBC313AF54DE22437429E@xmb-rcd-x02.cisco.com> <949EF20990823C4C85C18D59AA11AD8B0F23CB@FR712WXCHMBA11.zeu.alcatel-lucent.com> <E721D8C6A2E1544DB2DEBC313AF54DE2243749E2@xmb-rcd-x02.cisco.com> <6.2.5.6.2.20131204204318.0b8c2430@resistor.net> <52E0622E.5070806@cisco.com>
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We just published the -05 version addressing these comments..

Muthu

|-----Original Message-----
|From: Flemming Andreasen (fandreas)
|Sent: Thursday, January 23, 2014 5:59 AM
|To: SM; Muthu Arul Mozhi Perumal (mperumal); DRAGE, Keith (Keith); Ari Ker=
=E4nen
|Cc: mmusic@ietf.org
|Subject: Re: [MMUSIC] Last Call: <draft-ietf-mmusic-sdp-g723-g729-04.txt> =
(Offer/Answer Considerations
|for G723 Annex A and G729 Annex B) to Proposed Standard
|
|Greetings
|
|The chairs have reviewed this thread carefully and also discussed it
|with our AD. To conclude, we believe we have concensus to proceed with
|the document as follows:
|- The document does not formally update RFC 4856 (or any other RFC) and
|hence the front page and abstract will be updated accordingly.
|- The document will remain Standards Track.
|- Since the "annexa" and "annexb" parameters are defined in RFC 4856, it
|makes sense to keep the reference to RFC 4856 in the introductory text,
|along the lines of Muthu's suggested "New Introduction" text in the
|previous message in this thread.
|
|If anybody disagrees with this conclusion, please let the chairs know.
|
|Thanks
|
|     Ari & Flemming (MMUSIC co-chairs)
|
|
|
|
|On 12/4/13, 11:54 PM, SM wrote:
|> Hi Muthu,
|> At 20:13 04-12-2013, Muthu Arul Mozhi Perumal (mperumal) wrote:
|>> I see your point.
|>>
|>> Our opinion was that an RFC that specifies the mapping of media type
|>> parameters with SDP and registers the media types with IANA should
|>> also describe its usage, especially with the offer/answer model (e.g,
|>> RFC435). In that sense we thought draft-ietf-mmusic-sdp-g723-g729 is
|>> an update to RFC4856.
|>>
|>> Your opinion is that they should be delinked -- one RFC can register
|>> the media types and another RFC can describe the usage for the
|>> registered media types independently. I am fine with that. The
|>> abstract and introduction in draft-ietf-mmusic-sdp-g723-g729 can
|>> easily be changed to reflect it.
|>
|> Yes, but you can have both in one RFC.  The registration can be
|> requested in the IANA Considerations section.
|>
|>> ---------------------------------------------------------------------
|>> Current Abstract:
|>>    RFC4856 describes the annexa parameter for G723 and the annexb
|>>    parameter for G729, G729D and G729E. However, the specification does
|>>    not describe the offerer and answerer behavior when the value of the
|>>    annexa or annexb parameter does not match in the Session Description
|>>    protocol(SDP) offer and answer.  This document provides the offer/
|>>    answer considerations for these parameters and updates RFC4856.
|>>
|>> New Abstract:
|>>    There are no specifications that describe the offerer and answerer
|>>    behavior when the value of the annexa parameter of G729 or the annex=
b
|>>    parameter of G729, G729D and G729E does not match in the Session
|>>    Description protocol(SDP) offer and answer. This document provides
|>> the
|>>    offer/answer considerations for these parameters.
|>
|> I suggest removing "there are no specifications ..." and explain what
|> the document specifies.
|>
|>> Current Introduction:
|>>    [RFC4856] describes the annexa parameter for G723 as follows:
|>>
|>>       annexa: indicates that Annex A, voice activity detection, is used
|>>       or preferred.  Permissible values are "yes" and "no" (without the
|>>       quotes); "yes" is implied if this parameter is omitted.
|>>
|>>    Also, [RFC4856] describes the annexb parameter for G729, G729D and
|>>    G729E as follows:
|>>
|>>       annexb: indicates that Annex B, voice activity detection, is used
|>>       or preferred.  Permissible values are "yes" and "no" (without the
|>>       quotes); "yes" is implied if this parameter is omitted.
|>
|> The above starts with RFC 4856.  I suggest starting with the RFC 4566
|> followed by the RFC 3264 text (I did not verify the reference).  You
|> can then specify the fmtp parameters for Annex A, etc.  Once you have
|> done that, verify whether there are some registry requirements and add
|> text for that in a IANA Considerations section.
|>
|> Regards,
|> -sm
|> .
|>


From keith.drage@alcatel-lucent.com  Wed Jan 29 16:39:33 2014
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "avtcore@ietf.org" <avtcore@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "clue@ietf.org" <clue@ietf.org>
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Date: Thu, 30 Jan 2014 00:39:25 +0000
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Subject: [MMUSIC] draft-ietf-avtext-rtp-grouping-taxonomy-00
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To all potential users of the taxonomy document in other working groups.

We could do with a few people reading the above draft and identifying omiss=
ions, open issues, or corrections, so the editors can get a new version out=
 prior to the AVTEXT meeting in London.

https://datatracker.ietf.org/doc/draft-ietf-avtext-rtp-grouping-taxonomy/

In particular, those of you who have drafts that might use some of this ter=
minology.

Please respond to the avtext list.

Regards

Keith
AVTEXT WG cochair.=

From Christian.Groves@nteczone.com  Wed Jan 29 20:24:45 2014
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Subject: Re: [MMUSIC] Questions on draft-ietf-mmusic-sctp-sdp-05
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Hello,

Not an answer but a further question related to Q4.
Is it possible to have several of the same media-subtypes per SCTP 
association?
  i.e.         a=sctpmap:5000 webrtc-datachannel 16
               a=sctpmap:5001 bfcp 2
               a=sctpmap:5002 bfcp 1

I think it has been pointed about before but the media-subtypes ABNF is 
incomplete.

The draft also indicates that the use of existing MIME subtypes for the 
media-subtype format is encouraged. I assume this means the use of one 
of the values from the IANA media type registry 
<http://www.iana.org/assignments/media-types/media-types.xhtml>? If so 
then "bfcp" is not a registered type there. However T38 (audio or image) is.

draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00 also uses BFCP and 
MSRP. It mentions these in terms of "sub-protocol" which I assume more 
relates to the proto field at 
http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-2. 
However BFCP and MSRP are not individual values there either and it 
doesn't define t38.

I couldn't find the rtcweb draft that proposes to register 
webrtc-Datachannel in either of the above registries.

I saw draft-ejzak-dispatch-webrtc-data-channel-sdpneg proposing a 
a=webrtc-Datachannel attribute. I thought SDP attributes needed a static 
name (i.e. so it can be registered by IANA). In cl.4.3 of the draft it 
appears that the attribute name is based on the "protocol" in the 
sctpmap. However this would imply a dynamic allocation of attribute names.

I'm confused. Can any one clarify what the media-subtype is meant to be?

Regards,
Christian


On 27/01/2014 10:36 PM, Christer Holmberg wrote:
>
> Hi,
>
> Section 5.1 of draft-ietf-mmusic-sctp-sdp-05 describes how an offered 
> SCTP association can be rejected, by in the Answer setting the port 
> value in the associated SDP sctpmap attribute to zero.
>
> *Q1:*         When an SCTP association is rejected, shall the 
> associated port value in the m- line fmt list also be set to zero? The 
> text only talks about the sctpmap attribute.
>
> The draft does not talk about subsequent offers.
>
> *Q2:*         Do I need to include all sctpmap attributes in 
> subsequent offers, even if some are associated with previously 
> rejected STCP associations, in order to maintain the order?
>
> *Q3:*         Can I, in a subsequent offer, remove an SCTP association 
> by setting the associated port value to zero? If so, do I set the port 
> value to zero in the associated sctpmap attribute, and/or the m- line 
> fmt list?
>
> The draft only talks about having a single media-subtype per SCTP 
> association.
>
> *Q4*:         Is it possible to define multiple media-subtypes per 
> SCTP association?
>
>                 Example:
>
>                 m=application 54111 DTLS/SCTP *5000*
>
>               c=IN IP4 79.97.215.79
>
>               a=sctpmap:*5000* webrtc-datachannel 16
>
>               a=sctpmap:*5000* bfcp 2
>
>               a=sctpmap:*5000* t38 1
>
>                 (Each media-subtype could use different SCTP streams 
> within the SCTP association)
>
> Thanks!
>
> Regards,
>
> Christer
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Wed Jan 29 21:12:03 2014
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Subject: Re: [MMUSIC] Questions on draft-ietf-mmusic-sctp-sdp-05
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On 1/29/14 11:24 PM, Christian Groves wrote:
> Hello,
>
> Not an answer but a further question related to Q4.
> Is it possible to have several of the same media-subtypes per SCTP
> association?
>   i.e.         a=sctpmap:5000 webrtc-datachannel 16
>                a=sctpmap:5001 bfcp 2
>                a=sctpmap:5002 bfcp 1

AFAICT this should be ok.

BUT, I find associating a media subtype with an sctp association 
nonsensical. Each association represents a set of independent channels. 
What sense does it make to associate a media sub-type with set of 
channels? The existing media subtypes, such as bfcp, are are intended to 
be mapped onto a packet stream, such as you get with UDP or RTP.

A "type" that is associated with an sctp association ought to describe a 
particular usage of a *set* of packet streams, or channels. That is a 
very different sort of thing.

> I think it has been pointed about before but the media-subtypes ABNF is
> incomplete.
>
> The draft also indicates that the use of existing MIME subtypes for the
> media-subtype format is encouraged. I assume this means the use of one
> of the values from the IANA media type registry
> <http://www.iana.org/assignments/media-types/media-types.xhtml>? If so
> then "bfcp" is not a registered type there. However T38 (audio or image)
> is.
>
> draft-ejzak-dispatch-webrtc-data-channel-sdpneg-00 also uses BFCP and
> MSRP. It mentions these in terms of "sub-protocol" which I assume more
> relates to the proto field at
> http://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-2.
> However BFCP and MSRP are not individual values there either and it
> doesn't define t38.

There are a lot of details to iron out, but at least the ejzak draft is 
independently declaring the data type per channel within an association.

	Thanks,
	Paul

> I couldn't find the rtcweb draft that proposes to register
> webrtc-Datachannel in either of the above registries.
>
> I saw draft-ejzak-dispatch-webrtc-data-channel-sdpneg proposing a
> a=webrtc-Datachannel attribute. I thought SDP attributes needed a static
> name (i.e. so it can be registered by IANA). In cl.4.3 of the draft it
> appears that the attribute name is based on the "protocol" in the
> sctpmap. However this would imply a dynamic allocation of attribute names.
>
> I'm confused. Can any one clarify what the media-subtype is meant to be?
>
> Regards,
> Christian
>
>
> On 27/01/2014 10:36 PM, Christer Holmberg wrote:
>>
>> Hi,
>>
>> Section 5.1 of draft-ietf-mmusic-sctp-sdp-05 describes how an offered
>> SCTP association can be rejected, by in the Answer setting the port
>> value in the associated SDP sctpmap attribute to zero.
>>
>> *Q1:*         When an SCTP association is rejected, shall the
>> associated port value in the m- line fmt list also be set to zero? The
>> text only talks about the sctpmap attribute.
>>
>> The draft does not talk about subsequent offers.
>>
>> *Q2:*         Do I need to include all sctpmap attributes in
>> subsequent offers, even if some are associated with previously
>> rejected STCP associations, in order to maintain the order?
>>
>> *Q3:*         Can I, in a subsequent offer, remove an SCTP association
>> by setting the associated port value to zero? If so, do I set the port
>> value to zero in the associated sctpmap attribute, and/or the m- line
>> fmt list?
>>
>> The draft only talks about having a single media-subtype per SCTP
>> association.
>>
>> *Q4*:         Is it possible to define multiple media-subtypes per
>> SCTP association?
>>
>>                 Example:
>>
>>                 m=application 54111 DTLS/SCTP *5000*
>>
>>               c=IN IP4 79.97.215.79
>>
>>               a=sctpmap:*5000* webrtc-datachannel 16
>>
>>               a=sctpmap:*5000* bfcp 2
>>
>>               a=sctpmap:*5000* t38 1
>>
>>                 (Each media-subtype could use different SCTP streams
>> within the SCTP association)
>>
>> Thanks!
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
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From christer.holmberg@ericsson.com  Wed Jan 29 23:23:51 2014
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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] Questions on draft-ietf-mmusic-sctp-sdp-05
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Subject: Re: [MMUSIC] Questions on draft-ietf-mmusic-sctp-sdp-05
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Hi,

> Not an answer but a further question related to Q4.
> Is it possible to have several of the same media-subtypes per SCTP=20
> association?
>   i.e.         a=3Dsctpmap:5000 webrtc-datachannel 16
>                a=3Dsctpmap:5001 bfcp 2
>                a=3Dsctpmap:5002 bfcp 1

As the port values are different, aren't you using two DIFFERENT SCTP assoc=
iations in the example?

Section 4.1 of the draft says:

	"Running SCTP over DTLS make possible to have multiple SCTP
   	associations on top of the same DTLS connection; each SCTP
   	association make use of a distinct port number that is mainly used to
   	demultiplex the associations."

Regards,

Christer

From Christian.Groves@nteczone.com  Thu Jan 30 01:46:39 2014
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Subject: Re: [MMUSIC] Questions on draft-ietf-mmusic-sctp-sdp-05
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Hello Christer,

I guess then in my example there is different SCTP associations. I 
should have probably said the same media-subtype in several a=sctpmap 
attributes under the same m-line.

Christian

On 30/01/2014 6:23 PM, Christer Holmberg wrote:
> Hi,
>
>> Not an answer but a further question related to Q4.
>> Is it possible to have several of the same media-subtypes per SCTP
>> association?
>>    i.e.         a=sctpmap:5000 webrtc-datachannel 16
>>                 a=sctpmap:5001 bfcp 2
>>                 a=sctpmap:5002 bfcp 1
> As the port values are different, aren't you using two DIFFERENT SCTP associations in the example?
>
> Section 4.1 of the draft says:
>
> 	"Running SCTP over DTLS make possible to have multiple SCTP
>     	associations on top of the same DTLS connection; each SCTP
>     	association make use of a distinct port number that is mainly used to
>     	demultiplex the associations."
>
> Regards,
>
> Christer
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From ari.keranen@ericsson.com  Thu Jan 30 03:34:40 2014
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From: =?windows-1252?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Hi,

On 28/01/14 15:49, Pal Martinsen (palmarti) wrote:
> Just posted -05 version of the draft.

Thanks! Here's a few comments on the latest version.

The abstract says:

>    This document provides guidelines on how to make ICE [RFC5245]
>    conclude faster in IPv4/IPv6 dual-stack scenarios where broken paths
>    exist.  This will lead to more sustained IPv6 deployment as users
>    will no longer have an incentive to disable IPv6.

Out of (much of) context, especially the second sentence seems like a 
quite strong statement. I guess there could be other reasons than this 
for disabling v6, for example.

And nit: usually abstract should not contain references and open the ICE 
acronym.

>    An ICE agent can choose an algorithm or a technique of its choice to
>    ensure that the resulting check lists have a fair intermingled mix of
>    IPv4 and IPv6 address families.  Modifying the check list directly
>    can lead to uncoordinated local and remote check lists that result in
>    ICE taking longer to complete.

Not only can the tests take longer, but they can potentially also fail 
if the lists get badly out of sync since once you have made a NAT 
binding with a connectivity check packet, the check to other direction 
must come before the binding expires. But I suppose this will not be a 
problem with the proposed solution since the lists will be in sync?

>    The local and remote agent can have different algorithms for choosing
>    the type preference value without any impact on coordination between
>    the local and remote check list.

What do you mean by "without any impact on coordination"?

>    Where G is the candidate provided by the controlling agent and D the

s/candidate/priority of the candidate/

> Do we ask for WG adoption, or do we need another MMUSIC session discussing this?

Given this new approach, I think it's too early to ask for WG adoption. 
Rather we need more discussion especially here on the list and possibly 
at the meeting.

What I would find helpful for discussion is that if you could provide a 
step-by-step example of how the candidate lists are formed and 
connectivity checks done in cases where a) both support this with same 
promo algo b) both support this but with different promo algo c) only 
one endpoint supports this.

Also maybe a recommended promo algo would make sense so that there is at 
least a fair chance that both will use the same algo/values and get most 
out of this.


Cheers,
Ari


From ari.keranen@nomadiclab.com  Thu Jan 30 05:01:08 2014
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Cc: draft-ietf-mmusic-rtsp-nat@tools.ietf.org
Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat: ICE restart fails, then what?
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Hi Magnus,

I agree that one should not terminate existing session that works just 
because an ICE restart fails. I think the text you quoted is rather for 
a situation where this is the first session and there are no existing 
(working) sessions.

Hence, your text proposal looks good to me, with a few nits (see inline).

On 22/01/14 16:00, Magnus Westerlund wrote:
[...]
> Thus my current proposal for Section 6.12 of draft-ietf-mmmusic-rtsp-nat
> is the following:
>
>     If the RTSP session is in playing state at the time of sending the
>     SETUP request requiring ICE restart, then the ICE connectivity checks
>     SHALL use Regular nomination.  Any ongoing media delivery continues
>     on the previously nominated candidate pairs until the new pairs have
>     been nominated for the individual candidate.  Once the nomination of

"nominating pairs for candidate" sounds a bit strange. Should that be 
e.g., "nominated for the individual stream"?

>     the new candidate pair has completed, all unused candidates may be
>     released.  If the ICE processing fails and no new candidate pairs are
>     promoted for use, then one MAY continue to use the previously

s/promoted/nominated/

>     nominated candidate pairs while they still function.  If they have
s/have//

>     appear to fail to transport media packets anymore then the client can
>     select between two actions.  First, if it has any actions available
>     that might make ICE work, like trying another STUN/TURN server or
>     change the transport parameters it modifies the session, and if ICE
>     is still to be used restart ICE.

The sentence above probably needs some more commas in right places (or 
maybe break it in pieces).


Cheers,
Ari

From balmenyh@cisco.com  Thu Jan 30 05:30:15 2014
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Subject: [MMUSIC] Candidate priority in draft-reddy-mmusic-ice-happy-eyeballs-05
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Hi,

This is a comment related to 
http://www.ietf.org/id/draft-reddy-mmusic-ice-happy-eyeballs-05.txt. 
Section 4 states:

    ICE [RFC5245] section 4.1.2.2 has guidelines for how the type
    preference value should be chosen.  Instead of having a static value
    for IPv4 and a static value for IPv6 type of addresses, it is
    possible to choose this value dynamically in such a way that IPv4 and
    IPv6 address candidate priorities ends up intermingled.

    The local and remote agent can have different algorithms for choosing
    the type preference value without any impact on coordination between
    the local and remote check list.

I believe the intention here was to reference the local preference, 
instead of the type preference. The former is meant to distinguish 
between network interfaces from which the candidates have been gathered, 
while the later was intended to distinguish the type of the candidate 
obtained. It sounds to me, that IPv4 / IPv6 difference falls therefore 
in the local preference category, and not the type preference.

 From a different perspective, applying the formula given above:

         priority = (2^24)*(type preference) +
                    (2^8)*(local preference) +
                    (2^0)*(256 - component ID)

it also makes sense to "play" with the less significant bits, which are 
the local preference, as opposed to the more significant bits, which are 
the type preference. This will indeed lead to a finer grained 
intermingling of the IPv4 and IPv6 candidates.

Regards,

Balint

From magnus.westerlund@ericsson.com  Thu Jan 30 05:45:40 2014
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Subject: Re: [MMUSIC] draft-ietf-mmusic-rtsp-nat: ICE restart fails, then what?
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On 2014-01-30 14:01, Ari Keranen wrote:
> Hi Magnus,
> 
> I agree that one should not terminate existing session that works just
> because an ICE restart fails. I think the text you quoted is rather for
> a situation where this is the first session and there are no existing
> (working) sessions.
> 
> Hence, your text proposal looks good to me, with a few nits (see inline).

Thanks for the review, I will fix the nits and push a new version.

Cheers

Magnus

> 
> On 22/01/14 16:00, Magnus Westerlund wrote:
> [...]
>> Thus my current proposal for Section 6.12 of draft-ietf-mmmusic-rtsp-nat
>> is the following:
>>
>>     If the RTSP session is in playing state at the time of sending the
>>     SETUP request requiring ICE restart, then the ICE connectivity checks
>>     SHALL use Regular nomination.  Any ongoing media delivery continues
>>     on the previously nominated candidate pairs until the new pairs have
>>     been nominated for the individual candidate.  Once the nomination of
> 
> "nominating pairs for candidate" sounds a bit strange. Should that be
> e.g., "nominated for the individual stream"?
> 
>>     the new candidate pair has completed, all unused candidates may be
>>     released.  If the ICE processing fails and no new candidate pairs are
>>     promoted for use, then one MAY continue to use the previously
> 
> s/promoted/nominated/
> 
>>     nominated candidate pairs while they still function.  If they have
> s/have//
> 
>>     appear to fail to transport media packets anymore then the client can
>>     select between two actions.  First, if it has any actions available
>>     that might make ICE work, like trying another STUN/TURN server or
>>     change the transport parameters it modifies the session, and if ICE
>>     is still to be used restart ICE.
> 
> The sentence above probably needs some more commas in right places (or
> maybe break it in pieces).
> 
> 
> Cheers,
> Ari
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
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 ari.keranen@ericsson.com  Thu Jan 30 06:25:45 2014
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Hi,

I reviewed the UDPTL DTLS draft and overall it seems to be in good 
shape. Please find below a few (mostly editorial) comments to consider.


1.  Introduction

    This was mainly because of the challenges involved with
    physical access to telephony equipment.

s/physical access/malevolent physical access/
(or something else saying that there's an unauthorized accessing for 
making an attack in question)


    This document specifies the transport of UDPTL over DTLS using the
    DTLS record layer "application_data" packets [RFC6347].

"application_data" is actually defined in the TLS RFC (5246) and 6347 
doesn't mention that name (as such) at all. Maybe having both RFCs as 
reference makes sense.


4.2.2.  Latching Control without ICE

    and the passive MUST NOT wait for a STUN answer before sending its

s/passive/passive side/

Moreover, the term "Latching" may not be obvious for everyone. Do you 
mean the same as in draft-ietf-mmusic-latching? A sentence explaining 
this could help or alternatively changing the title to more descriptive one.


4.2.3.  STUN Interaction
    The UA MUST demultiplex packets arriving on the IP address and port
    associated with the DTLS association as follows:

Why is this a MUST? Wouldn't other ways be OK too (say, magic cookie)? 
Maybe s/MUST/can/?


A.3.  Message Flow Of T.38 Fax Replacing Audio Media Stream in An
       Existing Audio-Only Session

    Traditionally, most session with non-secure transport of T.38 fax,
    transported using UDPTL, are established by modifying an ongoing

s/session/sessions/


Cheers,
Ari

From internet-drafts@ietf.org  Thu Jan 30 06:28:53 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-19.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           : A Network Address Translator (NAT) Traversal Mech=
anism for Media Controlled by Real-Time Streaming Protocol (RTSP)
        Authors         : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-19.txt
	Pages           : 32
	Date            : 2014-01-30

Abstract:
   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams set up and
   controlled with Real-time Streaming Protocol version 2 (RTSP 2.0).
   It uses Interactive Connectivity Establishment (ICE) adapted to use
   RTSP as a signalling channel, defining the necessary RTSP extensions
   and procedures.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-19

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-rtsp-nat-19


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/


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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-19.txt
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Hi,

This version just addresses some nits from Ari.

Cheers

Magnus

On 2014-01-30 15:28, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
> 
>         Title           : A Network Address Translator (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming Protocol (RTSP)
>         Authors         : Jeff Goldberg
>                           Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-19.txt
> 	Pages           : 32
> 	Date            : 2014-01-30
> 
> Abstract:
>    This document defines a solution for Network Address Translation
>    (NAT) traversal for datagram based media streams set up and
>    controlled with Real-time Streaming Protocol version 2 (RTSP 2.0).
>    It uses Interactive Connectivity Establishment (ICE) adapted to use
>    RTSP as a signalling channel, defining the necessary RTSP extensions
>    and procedures.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rtsp-nat/
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rtsp-nat-19
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-rtsp-nat-19
> 
> 
> 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/
> 
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From internet-drafts@ietf.org  Fri Jan 31 02:25:02 2014
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.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           : Session Description Protocol (SDP) Extension For =
Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The =
Public Switched Telephone Network (PSTN)
        Authors         : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
	Pages           : 37
	Date            : 2014-01-31

Abstract:
   This memo 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 Public Switched Telephone Network (PSTN).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22


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 Simo.Veikkolainen@nokia.com  Fri Jan 31 03:02:13 2014
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From: <Simo.Veikkolainen@nokia.com>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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Hello WG,

We have submitted a new version of http://tools.ietf.org/html/draft-ietf-mm=
usic-sdp-cs-22=20

To recap, there was discussion on the  MMUSIC WG list on which correlation =
mechanisms should be mandatory to implement (see Gonzalo's email http://www=
.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by Gon=
zalo, 3GPP is going to use this draft, more specifically the plan now is to=
 use the caller id based correlation mechanism. This was the message I got =
from Andrew after a recent 3GPP CT1 meeting.

We have now made the following changes to the draft:
-	the endpoint may choose which correlation mechanisms to implement (before=
 all three were MTI).
-	we have introduced a fourth correlation type ("external") which indicates=
 explicit support for cases where also some external means, typically the h=
uman user, can do the correlation. This can be as a fallback in cases where=
 no other common correlation mechanism is found, or the negotiated correlat=
ion mechanisms fail due to being not supported in the PSTN network. This is=
 much like current systems like Webex work.

This was also the guidance from the IESG informal telechat in December wher=
e Gonzalo presented the draft.

We would like to hear from the WG whether there are any concerns with these=
 changes.

On behalf of the authors,
Simo

PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web/mm=
usic/current/msg12792.html to next version, slipped my mind this time.


-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext internet-dra=
fts@ietf.org
Sent: 31 January, 2014 12:25
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt


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           : Session Description Protocol (SDP) Extension For =
Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The =
Public Switched Telephone Network (PSTN)
        Authors         : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
	Pages           : 37
	Date            : 2014-01-31

Abstract:
   This memo 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 Public Switched Telephone Network (PSTN).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-cs-22


Please note that it may take a couple of minutes from the time of submissio=
n 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/

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

From christer.holmberg@ericsson.com  Fri Jan 31 03:56:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, "draft-ietf-mmusic-udptl-dtls@tools.ietf.org" <draft-ietf-mmusic-udptl-dtls@tools.ietf.org>
Thread-Topic: draft-ietf-mmusic-udptl-dtls-03 review
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Subject: Re: [MMUSIC] draft-ietf-mmusic-udptl-dtls-03 review
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Hi Ari,

Thanks for your review! Comments inline.

> 1.  Introduction
>
>    This was mainly because of the challenges involved with
>    physical access to telephony equipment.
>
> s/physical access/malevolent physical access/ (or something else=20
> saying that there's an unauthorized accessing for making an attack in
> question)

I will change as suggested.

>    This document specifies the transport of UDPTL over DTLS using the
>    DTLS record layer "application_data" packets [RFC6347].
>
> "application_data" is actually defined in the TLS RFC (5246) and 6347=20
> doesn't mention that name (as such) at all. Maybe having both RFCs as=20
> reference makes sense.

I suggest the following modification:

	"This document specifies the transport of UDPTL over DTLS using the
   	DTLS record layer "application_data" packets [RFC5246] [RFC6347]."


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

> 4.2.2.  Latching Control without ICE
>
>    and the passive MUST NOT wait for a STUN answer before sending its
>
> s/passive/passive side/

I will change as suggested.

> Moreover, the term "Latching" may not be obvious for everyone. Do you=20
> mean the same as in draft-ietf-mmusic-latching? A sentence explaining=20
> this could help or alternatively changing the title to more descriptive o=
ne.

I would rather suggest to change the section titles, as following:

4.2.1. NAT Traversal With ICE
4.2.2. NAT Traversal Without ICE

...because, the text doesn't really talk so much about latching per se, but=
 more about the opening of pin holes etc.

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

> 4.2.3.  STUN Interaction
>    The UA MUST demultiplex packets arriving on the IP address and port
>    associated with the DTLS association as follows:
>
> Why is this a MUST? Wouldn't other ways be OK too (say, magic cookie)?=20
> Maybe s/MUST/can/?

I suggest to add "e.g.", to show that the documented demux mechanism is onl=
y one way it can be done.

	"The UA MUST demultiplex packets arriving on the IP address and port
	associated with the DTLS association, e.g. as follows:"

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

> A.3.  Message Flow Of T.38 Fax Replacing Audio Media Stream in An
>       Existing Audio-Only Session
>
>    Traditionally, most session with non-secure transport of T.38 fax,
>    transported using UDPTL, are established by modifying an ongoing
>
> s/session/sessions/

I will change as suggested.

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

Thanks!

Regards,

Christer


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To: Christer Holmberg <christer.holmberg@ericsson.com>, "draft-ietf-mmusic-udptl-dtls@tools.ietf.org" <draft-ietf-mmusic-udptl-dtls@tools.ietf.org>
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Subject: Re: [MMUSIC] draft-ietf-mmusic-udptl-dtls-03 review
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Hi Christer,

Looks good, thanks!


Cheers,
Ari

On 31/01/14 13:56, Christer Holmberg wrote:
> Hi Ari,
>
> Thanks for your review! Comments inline.
>
>> 1.  Introduction
>>
>>     This was mainly because of the challenges involved with
>>     physical access to telephony equipment.
>>
>> s/physical access/malevolent physical access/ (or something else
>> saying that there's an unauthorized accessing for making an attack in
>> question)
>
> I will change as suggested.
>
>>     This document specifies the transport of UDPTL over DTLS using the
>>     DTLS record layer "application_data" packets [RFC6347].
>>
>> "application_data" is actually defined in the TLS RFC (5246) and 6347
>> doesn't mention that name (as such) at all. Maybe having both RFCs as
>> reference makes sense.
>
> I suggest the following modification:
>
> 	"This document specifies the transport of UDPTL over DTLS using the
>     	DTLS record layer "application_data" packets [RFC5246] [RFC6347]."
>
>
> ---------------------------
>
>> 4.2.2.  Latching Control without ICE
>>
>>     and the passive MUST NOT wait for a STUN answer before sending its
>>
>> s/passive/passive side/
>
> I will change as suggested.
>
>> Moreover, the term "Latching" may not be obvious for everyone. Do you
>> mean the same as in draft-ietf-mmusic-latching? A sentence explaining
>> this could help or alternatively changing the title to more descriptive one.
>
> I would rather suggest to change the section titles, as following:
>
> 4.2.1. NAT Traversal With ICE
> 4.2.2. NAT Traversal Without ICE
>
> ...because, the text doesn't really talk so much about latching per se, but more about the opening of pin holes etc.
>
> ---------------------------
>
>> 4.2.3.  STUN Interaction
>>     The UA MUST demultiplex packets arriving on the IP address and port
>>     associated with the DTLS association as follows:
>>
>> Why is this a MUST? Wouldn't other ways be OK too (say, magic cookie)?
>> Maybe s/MUST/can/?
>
> I suggest to add "e.g.", to show that the documented demux mechanism is only one way it can be done.
>
> 	"The UA MUST demultiplex packets arriving on the IP address and port
> 	associated with the DTLS association, e.g. as follows:"
>
> ---------------------------
>
>> A.3.  Message Flow Of T.38 Fax Replacing Audio Media Stream in An
>>        Existing Audio-Only Session
>>
>>     Traditionally, most session with non-secure transport of T.38 fax,
>>     transported using UDPTL, are established by modifying an ongoing
>>
>> s/session/sessions/
>
> I will change as suggested.
>
> ---------------------------
>
> Thanks!
>
> Regards,
>
> Christer
>


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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: =?Windows-1252?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
Thread-Topic: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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Subject: Re: [MMUSIC] New Version Notification for draft-reddy-mmusic-ice-happy-eyeballs-04.txt
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On 30 Jan 2014, at 12:34 pm, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote=
:

> Hi,
>=20
> On 28/01/14 15:49, Pal Martinsen (palmarti) wrote:
>> Just posted -05 version of the draft.
>=20
> Thanks! Here's a few comments on the latest version.
>=20
> The abstract says:
>=20
>>   This document provides guidelines on how to make ICE [RFC5245]
>>   conclude faster in IPv4/IPv6 dual-stack scenarios where broken paths
>>   exist.  This will lead to more sustained IPv6 deployment as users
>>   will no longer have an incentive to disable IPv6.
>=20
> Out of (much of) context, especially the second sentence seems like a qui=
te strong statement. I guess there could be other reasons than this for dis=
abling v6, for example.

Agree. Sentence removed.

>=20
> And nit: usually abstract should not contain references and open the ICE =
acronym.
>=20
Fixed.


>>   An ICE agent can choose an algorithm or a technique of its choice to
>>   ensure that the resulting check lists have a fair intermingled mix of
>>   IPv4 and IPv6 address families.  Modifying the check list directly
>>   can lead to uncoordinated local and remote check lists that result in
>>   ICE taking longer to complete.
>=20
> Not only can the tests take longer, but they can potentially also fail if=
 the lists get badly out of sync since once you have made a NAT binding wit=
h a connectivity check packet, the check to other direction must come befor=
e the binding expires. But I suppose this will not be a problem with the pr=
oposed solution since the lists will be in sync?
>=20
Added failure as well to further scare the reader. :-)

And yes, the list must be kept in sync.

>>   The local and remote agent can have different algorithms for choosing
>>   the type preference value without any impact on coordination between
>>   the local and remote check list.
>=20
> What do you mean by "without any impact on coordination=94?

kept in sync?

New text:

The local and remote agent can have different algorithms for choosing
the local preference value without impacting the synchronisation
between the local and remote check list.
>=20

>>   Where G is the candidate provided by the controlling agent and D the
>=20
> s/candidate/priority of the candidate/
>=20
tnx


>> Do we ask for WG adoption, or do we need another MMUSIC session discussi=
ng this?
>=20
> Given this new approach, I think it's too early to ask for WG adoption. R=
ather we need more discussion especially here on the list and possibly at t=
he meeting.
>=20
Ok.=20

Just a bit afraid of wasting cycles if this proves to not so useful.=20

> What I would find helpful for discussion is that if you could provide a s=
tep-by-step example of how the candidate lists are formed and connectivity =
checks done in cases where a) both support this with same promo algo b) bot=
h support this but with different promo algo c) only one endpoint supports =
this.
>=20

As I do believe in running code, I wrote a simple implementation of the alg=
orithm described below.

It can be downloaded from:
https://github.com/palerikm/HappyE-ICE-Test

>From README:

Compiling
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Run ./bootstrap to create the configure script.

then do ./configure
and
=20
make

Running
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

There are two txt files with candidates. One with local candidates where pr=
iorities needs to be calculated and a predefined remote candidate file. (Pl=
ease Note parsing is crude so keep the files clean..)

the program can be run as follows:

src/icehetest 4 candidates.txt remoteCandidates.txt=20

4 indicates the number of concecutive candidates if same address type allow=
ed.


> Also maybe a recommended promo algo would make sense so that there is at =
least a fair chance that both will use the same algo/values and get most ou=
t of this.
>=20
First shot an an algorithm:

5.  Example Algorithm for Choosing the Local Preference

   The value space for the local preference is from 0 to 65535
   inclusive.  This value space can be divided up in chunks for each IP
   address family.

   An IPv6 and IPv4 start priority must be given.  In this example IPv6
   starts at 60000 and IPv4 at 59000.  This leaves enough address space
   to further play with the values if pr interface priorities needs to
   be added.  The highest value should be given to the address family
   that should be prioritized.

        IPv6    IPv4
         Start   Start
   65535  60k     59k    58k    57k    56k    55k                    0
   +--------+------+------+------+------+------+---------------------+
   |        | IPv6 | IPv4 | IPv6 | IPv4 | IPv6 |                     |
   |        | (1)  |  (1) |  (2) |  (2) |  (3) |                     |
   +--------+------+------+------+------+------+---------------------+
             <- N->

        =20
   The local preference can be calculated by the given formula:


         local_preference =3D N*2*(Cn/Cmax)

   Where N is the absolute value of IPv6_start-IPv4_start.  This ensures
   a positive number even if IPv4 is the highest priority.  Cn is the
   number of current candidates of a specific IP address type and
   candidate type (HOST, SRFLX, RELAY).  Cmax is the number of allowed
   consecutive candidates of the same IP address type.



.-.
P=E5l-Erik

>=20
> Cheers,
> Ari
>=20


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From: "Adam Gensler (agensler)" <agensler@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
Thread-Topic: Review of draft-ietf-mmusic-udptl-dtls-03
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Subject: Re: [MMUSIC] Review of draft-ietf-mmusic-udptl-dtls-03
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Christer,

Sorry for the delay, I've been offline for a few days. See inline with
[adam]:


On 1/29/14, 4:58 AM, "Christer Holmberg" <christer.holmberg@ericsson.com>
wrote:

>Hi Adam,
>
>Again, thanks for your review! See my comments inline.
>
>> After speaking with Gonzalo I've reviewed
>>draft-ietf-mmusic-udptl-dtls-03.
>> Overall this draft is excellent and appears to cover all the corner
>> cases I could think of. I do have a few comments, see below:
>>=20
>> 1. In section 3.1, secure channel establishment, there is the
>> following text:
>>=20
>> "If the endpoint supports, and is willing to use, a cipher suite with
>> an associated certificate, it MUST include an SDP fingerprint
>> attribute [RFC4572 <http://tools.ietf.org/html/rfc4572>] in the SDP."
>>=20
>> "If a cipher suite with an associated certificate is selected during
>> the DTLS handshake, the certificate received during the DTLS handshake
>> MUST match the fingerprint received in the SDP fingerprint attribute."
>>=20
>> To me this seems to indicate the particular cipher suites are
>> associated with a particular certificate. While it's true that the
>> certificate plays a role in overall TLS negotiation, it does not
>> provide a list of ciphers to use.
>
>The intention of the text is NOT to indicate that a particular cipher
>suites are associated with a particular certificate.
>
>> I think what this is saying is that
>> if the client wishes to use a certificate for DTLS it should provide a
>> certificate fingerprint. Is that right?
>
>Only if the cipher suite has an associated certificate.
>
>But, e.g. a PSK based cipher suite  (RFC4279) has no associated
>certificate and thus there is no point of indicating certificate
>fingerprint for PSK based cipher suite.
>
>> Given the overall complexity of PKI in general I think being extra
>>clear here would help avoid any confusion.
>
>We did spend quite a bit of time on this part of the text, and I can't
>see what modifications I should do in order to make it more clear.
>
>------------------------


[adam] Thanks for the clarification here. Given this extra context I agree
and I'm not sure how to adjust this any further.


>
>> 2. Section 4.1 describes handling anonymous faxes with freshly
>> generated self-signed certificates. It goes on to say:
>>=20
>> "...and the content of the subjectAltName attribute inside the
>> certificate MUST NOT contain information that either allows
>> correlation or identification of the user making anonymous calls."
>>=20
>> While the subjectAltName could potentially have identifiable data in
>> it, it is not always present as a field within the certificate. Some
>> certificates may have a subjectAltName, some may not. However, all
>> certificates should have a commonName. I think that is the item that
>> should be specifically called out as a place to avoid identifiable
>> information. Or, perhaps, simply stipulating that no fields within the
>> certificate should contain identifiable information.
>
>That's a good point.
>
>What about something like:
>
>                "...and attributes inside the certificate SHALL NOT
>contain information that either allows correlation or
>                identification of the user making anonymous calls. This
>is particularly important for the subjectAltName
>                and commonName attributes"
>
>------------------------


[adam] This sounds good to me.


>
>> 3. Section 4.3 describes rekeying a connection that is established.
>> The DTLS RFC describes how to handle this, but gives two options to an
>> endpoint where out-of-order packets are received that have been
>> encrypted with different keys; discard the packet, or, retain two sets
>> of keying material for some recommended interval. DTLS is not specific
>> as to what to do here. Section 4.3 says the receiver SHOULD maintain
>> the keying material. Given the nature of T.38, I think it may be
>> prudent to say the receiver MUST maintain the keying material.
>> Dropping and/or discarding
>> T.38 packets would never be my preference.
>
>We can change it to MUST.
>
>------------------------


[adam] Perfect, I think that will help avoid any confusion around this,
should the situation occur.


>
>Regards,
>
>Christer
>
>


From christer.holmberg@ericsson.com  Fri Jan 31 09:47:03 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Adam Gensler (agensler)" <agensler@cisco.com>, "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
Thread-Topic: Review of draft-ietf-mmusic-udptl-dtls-03
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Subject: Re: [MMUSIC] Review of draft-ietf-mmusic-udptl-dtls-03
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Hi Adam,

Thanks for your reply!

We'll submit a new version of the draft, with the changes implemented, asap=
 :)

Regards,

Christer

Sent from my Sony Ericsson Xperia arc S

"Adam Gensler (agensler)" <agensler@cisco.com> wrote:


Christer,

Sorry for the delay, I've been offline for a few days. See inline with
[adam]:


On 1/29/14, 4:58 AM, "Christer Holmberg" <christer.holmberg@ericsson.com>
wrote:

>Hi Adam,
>
>Again, thanks for your review! See my comments inline.
>
>> After speaking with Gonzalo I've reviewed
>>draft-ietf-mmusic-udptl-dtls-03.
>> Overall this draft is excellent and appears to cover all the corner
>> cases I could think of. I do have a few comments, see below:
>>
>> 1. In section 3.1, secure channel establishment, there is the
>> following text:
>>
>> "If the endpoint supports, and is willing to use, a cipher suite with
>> an associated certificate, it MUST include an SDP fingerprint
>> attribute [RFC4572 <http://tools.ietf.org/html/rfc4572>] in the SDP."
>>
>> "If a cipher suite with an associated certificate is selected during
>> the DTLS handshake, the certificate received during the DTLS handshake
>> MUST match the fingerprint received in the SDP fingerprint attribute."
>>
>> To me this seems to indicate the particular cipher suites are
>> associated with a particular certificate. While it's true that the
>> certificate plays a role in overall TLS negotiation, it does not
>> provide a list of ciphers to use.
>
>The intention of the text is NOT to indicate that a particular cipher
>suites are associated with a particular certificate.
>
>> I think what this is saying is that
>> if the client wishes to use a certificate for DTLS it should provide a
>> certificate fingerprint. Is that right?
>
>Only if the cipher suite has an associated certificate.
>
>But, e.g. a PSK based cipher suite  (RFC4279) has no associated
>certificate and thus there is no point of indicating certificate
>fingerprint for PSK based cipher suite.
>
>> Given the overall complexity of PKI in general I think being extra
>>clear here would help avoid any confusion.
>
>We did spend quite a bit of time on this part of the text, and I can't
>see what modifications I should do in order to make it more clear.
>
>------------------------


[adam] Thanks for the clarification here. Given this extra context I agree
and I'm not sure how to adjust this any further.


>
>> 2. Section 4.1 describes handling anonymous faxes with freshly
>> generated self-signed certificates. It goes on to say:
>>
>> "...and the content of the subjectAltName attribute inside the
>> certificate MUST NOT contain information that either allows
>> correlation or identification of the user making anonymous calls."
>>
>> While the subjectAltName could potentially have identifiable data in
>> it, it is not always present as a field within the certificate. Some
>> certificates may have a subjectAltName, some may not. However, all
>> certificates should have a commonName. I think that is the item that
>> should be specifically called out as a place to avoid identifiable
>> information. Or, perhaps, simply stipulating that no fields within the
>> certificate should contain identifiable information.
>
>That's a good point.
>
>What about something like:
>
>                "...and attributes inside the certificate SHALL NOT
>contain information that either allows correlation or
>                identification of the user making anonymous calls. This
>is particularly important for the subjectAltName
>                and commonName attributes"
>
>------------------------


[adam] This sounds good to me.


>
>> 3. Section 4.3 describes rekeying a connection that is established.
>> The DTLS RFC describes how to handle this, but gives two options to an
>> endpoint where out-of-order packets are received that have been
>> encrypted with different keys; discard the packet, or, retain two sets
>> of keying material for some recommended interval. DTLS is not specific
>> as to what to do here. Section 4.3 says the receiver SHOULD maintain
>> the keying material. Given the nature of T.38, I think it may be
>> prudent to say the receiver MUST maintain the keying material.
>> Dropping and/or discarding
>> T.38 packets would never be my preference.
>
>We can change it to MUST.
>
>------------------------


[adam] Perfect, I think that will help avoid any confusion around this,
should the situation occur.


>
>Regards,
>
>Christer
>
>


From pkyzivat@alum.mit.edu  Fri Jan 31 10:52:59 2014
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
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I just reviewed the new version. First I looked at just the diffs. But 
those raised some more general questions in my mind, so I re-reviewed 
more of the draft. I'm left with some considerable confusion about how 
this is intended to work in practice, especially with extern 
correlation, but even with the other forms.

In particular, I am unclear of the sequencing of O/A processing, 
correlation processing, and signaling completion. (I realize this is an 
mmusic draft and theoretically independent of sip, but forgive me if I 
discuss this in the context of sip.)

Normally, when first establishing a session with sip, the first O/A may 
occur before the signaling session is established. But early media may 
flow after the first O/A. Often it is expected that the user will be be 
aware and involved in this early media.

Once a session is up, it can be changed with a reinvite containing a new 
O/A. If the O/A and the reinvite are successful, then the results of the 
new O/A replace the old one. If the reinvite fails, then the session is 
supposed to continue as it was before the reinvite. In the interim, from 
the sending of the reinvite until its success or failure, things are a 
little confusing - presumably both old and new media streams must be 
maintained, but which are connected to the user is unspecified.

I see nothing in this draft that explains how the CS media, and 
correlation verification, fit into the above process.

Consider a new call that offers CS media:

- The first O/A must complete before the CS call can be attempted.

- must the CS call complete and be correlated successfully before
   the INVITE gets a 200? If the invite is allowed to complete first,
   what happens if no CS call is established, or if it is but
   correlation fails?

- when does CS media start to flow to the user as part of the
   SIP call? Is it before or after correlation? (E.g., does the
   user hear the DTMF exchange?)

- if media flows to the user before correlation, then what if
   correlation fails? Will another call be accepted and correlation
   be attempted again?

- what if multiple incoming calls arrive before correlation
   has been completed? (If correlation occurs before exposing the
   media to the user, then it isn't an issue.)

- extern correlation complicates this. It isn't like the other
   correlation mechanisms - the user must be exposed to the media
   to decide if there is correlation. How would the user indicate
   "this is the wrong call" in a way that would allow correlation
   to be attempted on another call?

Now consider a call that is initially established with one RTP media 
stream. Later, a reinvite is sent with an offer proposing to to replace 
that with a CS media stream:

- again, there must be an answer before the CS call can be attempted.

- are you assuming that the reinvite will complete before the CS
   call does? If so, then you are in a break-before-make situation -
   the RTP session is gone before the CS session takes its place.

- all the issues above also apply here.

ISTM that there is a bunch of missing explanation of how this is 
expected to work.

I also have a nit in section 5.3.2:

    The fourth correlation mechanism declares support for cases where
    correlation is done by external means.  Typically this means that the
    decision is left for the human user.  This is the way how some
    current conferencing systems operate: ...

"This is the way how some current conferencing systems operate"

should be:

"This is how some current conferencing systems operate"

	Thanks,
	Paul

On 1/31/14 6:01 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello WG,
>
> We have submitted a new version of http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>
> To recap, there was discussion on the  MMUSIC WG list on which correlation mechanisms should be mandatory to implement (see Gonzalo's email http://www.ietf.org/mail-archive/web/mmusic/current/msg12274.html ). As stated by Gonzalo, 3GPP is going to use this draft, more specifically the plan now is to use the caller id based correlation mechanism. This was the message I got from Andrew after a recent 3GPP CT1 meeting.
>
> We have now made the following changes to the draft:
> -	the endpoint may choose which correlation mechanisms to implement (before all three were MTI).
> -	we have introduced a fourth correlation type ("external") which indicates explicit support for cases where also some external means, typically the human user, can do the correlation. This can be as a fallback in cases where no other common correlation mechanism is found, or the negotiated correlation mechanisms fail due to being not supported in the PSTN network. This is much like current systems like Webex work.
>
> This was also the guidance from the IESG informal telechat in December where Gonzalo presented the draft.
>
> We would like to hear from the WG whether there are any concerns with these changes.
>
> On behalf of the authors,
> Simo
>
> PS. I'll add the proposed change in http://www.ietf.org/mail-archive/web/mmusic/current/msg12792.html to next version, slipped my mind this time.
>
>
> -----Original Message-----
> From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of ext internet-drafts@ietf.org
> Sent: 31 January, 2014 12:25
> To: i-d-announce@ietf.org
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-22.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
>
>          Title           : Session Description Protocol (SDP) Extension For Setting Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
>          Authors         : Miguel A. Garcia-Martin
>                            Simo Veikkolainen
> 	Filename        : draft-ietf-mmusic-sdp-cs-22.txt
> 	Pages           : 37
> 	Date            : 2014-01-31
>
> Abstract:
>     This memo 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 Public Switched Telephone Network (PSTN).
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-cs/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-sdp-cs-22
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-mmusic-sdp-cs-22
>
>
> 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/
>
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