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Cc: Ben Campbell <ben@nostrum.com>
Subject: [MMUSIC] SDP comments to draft-ietf-simple-msrp-cema-02.txt
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Here are some comments related to the new "msrp-cema" attribute defined 
in draft-ietf-simple-msrp-cema-02.txt

- I am missing a section that has the syntax of the new extension. While 
it's true that the syntax is fairly simple, any extension should clearly 
specify the syntax (in this case, ABNF). This formal syntax could be:

        attribute          /= msrp-cema-attr
        ; attribute defined in RFC 4566
        msrp-cema-attr     = "msrp-cema"

I also suggest to add a section titled "The SDP 'msrp-cema' attribute" 
where you first give an overview of what the attribute does at a very 
high level (in this case, it indicates support for the CEMA mechanism), 
and then you include the formal syntax.

- General. When you mention an SDP attribute, the "a=" is nor part of the 
attribute itself. So, it is not correct to say:

   the a=path attribute

The correct one is:

   the 'path' attribute


- The title of sections 4.2 and 4.3 could be improved. There is not such 
things as "MSRP offer" or "MSRP answer". We have the notion of "SDP 
offer" and "SDP answer", as well as "SDP offerer" and "SDP answerer", in 
which case, the sections should be titled "SDP offerer procedures" and 
"SDP answerer procedures".

Furthermore, it is quite confusing to follow these sections because they 
refer to the "MSRP endpoint" and the "remote MSRP endpoint". Obviously, 
what is the "MSRP endpoint" in Section 4.2, becomes the "remote MSRP 
endpoint" in Section 4.3, and vice versa. I suggest to re-write these 
sections in terms of "the SDP offerer" and "the SDP answerer". This will 
make the text much clearer, and the terms will not reverse when we change 
sections.

- Section 4.1. I think those three bullet points are the real 
"Applicability Statement" rather than something general to the mechanism. 
To me, they should be moved to Section 3.

- Sections 4.1 to 4.3, the text says that an "MSRP endpoint becomes 
'active'" or 'passive'. I was confused with what the text was meaning. I 
guess it means from the point of view of the TCP connection used for 
MSRP. I think this should be clarified.

- Section 4.2, 2nd bullet point. I am missing that this bullet point says 
how to populate the c/m lines in SDP. Perhaps it is covered by the first 
paragraph of this Section 4.2, which basically refers to RFC 4975 and 
4976. But then, the third bullet point describes the m/c line, so, at 
this point in time, I had my doubts whether bullet point forgets to say 
something of the c/m line or it is populated as per RFC 4975/4976.

- In order to add a bit of clarity and separate the generation of the SDP 
offer from receiving the SDP answer, I would suggest to add the following 
text at the beginning of the paragraph that introduces the SDP answer in 
Section 4.2:

  "When receiving an SDP answer," and then continue with the current text 
"if the MSRP media description of the SDP answer ..."

- Section 4.2, 1st bullet point of the SDP answer (and in other parts of 
4.2 and 4.3), the text says: "... does not match the information ...". I 
have a problem with the word "match". Most likely, "match" implies 
"equality". So we are talking of the equality of the c/m lines with a 
URI. These two can never be equal. In particular, a URI can eve be a 
domain name or host name, in which case, will never be equal to an IP 
address and a port number. Most likely you want to say "resolve", but I 
am not sure.

- I was trying to analyze if all the different cases are described in 
Section 4.2. Here is one that I believe is not covered: "if the SDP 
answer does not contain an 'msrp-cema' attribute and none of the 3 
conditions are met".

- Section 4.3, first paragraph. The text says "MUST reject". I think it 
would be nice to have a motivation to have such a strong action.

- Section 5.4. The text says:  "Middleboxes
    cannot be deployed in environments that require end-to-end SDP
    protection using SIP identity [RFC4916]."

I am trying to see the connection between "end-to-end SDP protection" and 
"SIP identity". I would have removed one of them from the text, depending 
on what you want to say. Perhaps you want to say that the CEMA extension 
does not work if "end-to-end integrity protection of SDP is used".

- Section 7.1 does not mention the registry where IANA has to operate. 
Start the section saying:

  "This document instructs IANA to add a attribute to the 'att-field 
(media level only)' registry of the SDP parameters registry, according to 
the following data"

- Section 7.1, "Purpose". I will not speak of the "sending UA" in an SDP 
registry, but of the "SDP creator".



Nits:

- First paragraph in Section 6.3:  " ... manipulating message content is 
CEMA provides ..."
Most likely s/is/in

- I would add formal references whenever an RFC is mentioned, especially 
if they are part of a normative sentence. For example, "... according to 
the procedures of RFC 4975 [RFC4975]".

- I believe the text should say "a Message Session ..." rather than "an 
Message Session".

But "an MSRP message", rather than "a MSRP message"

- Section 4.1, first paragraph. Replace "... and what SDP information" 
with "... and which SDP information"

- Section 4.1, first paragraph, at the end. Replace "TLS connection for 
the MSRP messages" with "TLS connection for sending and receiving MSRP 
messages".

/Miguel

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From miguel.a.garcia@ericsson.com  Wed Oct  5 06:41:31 2011
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: Re: [MMUSIC] Poll for consensus on WG item: SDP extension to classify traffic
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I believe we have support to adopt the traffic-class draft as WG item.

Please, authors, submit the next version as a WG item.

/Miguel

On 30/09/2011 21:21, Ali C. Begen (abegen) wrote:
> +1
>
> -acbegen
>
> On Sep 29, 2011, at 11:35 PM, "Gonzalo Salgueiro" <gsalguei@cisco.com
> <mailto:gsalguei@cisco.com>> wrote:
>
>> Fully support it.
>>
>> Gonzalo
>>
>> On Sep 29, 2011, at 4:32 PM, Miguel A. Garcia wrote:
>>
>>> We have a milestone to submit an "SDP extension to classify traffic"
>>> in August 2012.
>>>
>>> We have a draft that can be used as a design base to meet this goal,
>>> this is the "trafficlass" attribute:
>>>
>>> <http://datatracker.ietf.org/doc/draft-polk-mmusic-traffic-class-for-sdp/>http://datatracker.ietf.org/doc/draft-polk-mmusic-traffic-class-for-sdp/
>>>
>>> We would like to poll the WG for consensus to adopt such draft as a WG
>>> item in order to meet the mentioned milestone.
>>>
>>> Please indicate whether you support it or if not.
>>>
>>> Miguel and Flemming
>>> --
>>> Miguel A. Garcia
>>> +34-91-339-3608
>>> Ericsson Spain
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org <mailto:mmusic@ietf.org>
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org <mailto:mmusic@ietf.org>
>> https://www.ietf.org/mailman/listinfo/mmusic

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From miguel.a.garcia@ericsson.com  Wed Oct  5 06:49:58 2011
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Cc: Flemming Andreasen <fandreas@cisco.com>, draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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This is to announce a 2 week Working Group Last Call for

     draft-ietf-mmusic-media-loopback-16

as Proposed Standard. Please review and provide any comments you may have 
on the document before Thursday, October 20th, 2011. Comments should be 
sent to the document authors and the MMUSIC WG list.

Thanks

     Miguel & Flemming
-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From HKaplan@acmepacket.com  Wed Oct  5 13:07:56 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>, "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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When looking at the diff between this version of the draft and the previous=
, I note the majority of the changes are around removing the "priming" mech=
anism in its entirety.  And there is only one small paragraph about making =
this work behind NATs, which basically says "SHOULD use" ICE/STUN/TURN.

What surprises me about this is I don't recall any WG consensus to make the=
 change to pull out the priming mechanism entirely this way.  I only recall=
 discussion around saying one SHOULD use ICE but MAY use the priming mechan=
ism if ICE isn't supported; and keeping the priming mechanism defined in th=
e loopback draft to explain how it works.

So... (1) why was this significant change made without apparent WG consensu=
s, and (2) how do we expect the priming mechanism to interoperate if we don=
't define how it works?  Or have we decided making loopback calls to SIP UA=
's behind NATs is no longer important/relevant?

-hadriel


On Oct 5, 2011, at 9:52 AM, Miguel A. Garcia wrote:

> This is to announce a 2 week Working Group Last Call for
>=20
>    draft-ietf-mmusic-media-loopback-16
>=20
> as Proposed Standard. Please review and provide any comments you may have=
 on the document before Thursday, October 20th, 2011. Comments should be se=
nt to the document authors and the MMUSIC WG list.
>=20
> Thanks
>=20
>    Miguel & Flemming
> --=20
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Cc: Flemming Andreasen <fandreas@cisco.com>, draft-ietf-simple-chat@tools.ietf.org
Subject: Re: [MMUSIC] SDP Reviewers Needed for draft-ietf-simple-chat-10
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Hi

I have reviewed draft-ietf-simple-chat-10 for SDP extension issues, and 
I have two comments:

1) Need to consider extensibility rules for the new SDP attribute and 
any IANA registry needed for that
2) Security considerations (privacy concerns) seem to be missing for the 
SDP attribute.

I also have a few editorial nit fixes that I have forwarded to the 
authors separately.

Thanks

-- Flemming


On 9/21/11 1:06 PM, Flemming Andreasen wrote:
> Folks
>
> The SIMPLE WG has completed WGLC on
>
>     "Multi-party Chat Using the Message Session Relay Protocol (MSRP)" 
> (http://tools.ietf.org/html/draft-ietf-simple-chat-10).
>
> The draft extends SDP and requires review by the experts of MMUSIC. 
> The extension is listed in Section 8 with some examples in Section 9.
>
> If you would like to volunteer as a reviewer, please let the chairs 
> now. Also, please note that we have been asked to complete the review 
> relatively quickly.
>
> Thanks
>
>      Flemming & Miguel
>
>
>

From pkyzivat@alum.mit.edu  Thu Oct  6 14:00:39 2011
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Subject: Re: [MMUSIC] Draft submission: draft-holmberg-mmusic-sdp-multiplex-negotiation
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On 9/16/11 4:35 AM, Miguel A. Garcia wrote:
> I think there are two major points of difference between the TOGETHER
> and the MULTIPLEX approach:
>
> 1) Usage of port numbers in the SDP offer
>
> 2) Combination of additional parameters from the SDP offer
>
> Let me analyze 1) in more detail. I believe a stream of media is
> identified by the transport addresses of both endpoints and the protocol
> (RTP/RTCP in this case). So, if we take the MULTIPLEX approach, the
> offerer will include the same port number in all its multiplexed
> streams. Now, if the answerer decides to multiplex, then we will have a
> single stream, because the transport addresses of all the RTP flows are
> equal.
>
> In this MULTIPLEX approach, if the answerer does not support the
> extension or does not want to use it, it will create an SDP answer that
> contains different port numbers. Therefore, each RTP flow will be send
> over a different 5-tuple, and each RTP flow will constitute a different
> media stream, even when all the transport addresses at one endpoint are
> equal. So I don't see this creating a problem with any existing endpoint
> that does not support the extension.

IIUC, o/a results in an exchange of addresses/ports at which media will 
be *received*. It does *not* define a 5-tuple. You learn the 5-tuple 
from the o/a only if you assume media is sent from the same addr/port at 
which it is received.

Of course its fairly common to assume this symmetric usage. In 
retrospect it should probably have been mandated. But it wasn't.

If symmetric RTP is a fundamental requirement of the multiplexing, then 
it should be called out as such.

	Thanks,
	Paul

From miguel.a.garcia@ericsson.com  Fri Oct  7 03:29:54 2011
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We have two related milestones to submit SDP extensions to signal 
parallax information and to signal stereoscopic 3D video. This milestones 
are currently due on September 2012.

We have two drafts that are believed to serve as a design base to meet 
these milestones:

http://datatracker.ietf.org/doc/draft-greevenbosch-mmusic-parallax-attribute/
http://datatracker.ietf.org/doc/draft-greevenbosch-mmusic-signal-3d-format/

We would like to poll the WG for consensus to adopt those drafts as a WG
items in order to meet these milestones.

Please indicate whether you support the adoption of one, the other, or both.

     Miguel and Flemming
-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From miguel.a.garcia@ericsson.com  Fri Oct  7 03:31:44 2011
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Subject: Re: [MMUSIC] Poll for consensus on parallax and 3D format drafts
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Now with the name of the draft in the title of this e-mail.

/Miguel

On 07/10/2011 12:33, Miguel A. Garcia wrote:
> We have two related milestones to submit SDP extensions to signal
> parallax information and to signal stereoscopic 3D video. This milestones
> are currently due on September 2012.
>
> We have two drafts that are believed to serve as a design base to meet
> these milestones:
>
> http://datatracker.ietf.org/doc/draft-greevenbosch-mmusic-parallax-attribute/
> http://datatracker.ietf.org/doc/draft-greevenbosch-mmusic-signal-3d-format/
>
> We would like to poll the WG for consensus to adopt those drafts as a WG
> items in order to meet these milestones.
>
> Please indicate whether you support the adoption of one, the other, or both.
>
>       Miguel and Flemming

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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Cc: mmusic <mmusic@ietf.org>, draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Hi Hadriel

The issue was discussed on the mailing list in August, where the authors 
suggested it as a path forward and the chairs asked the WG for feedback 
on this approach, see e.g.

http://www.ietf.org/mail-archive/web/mmusic/current/msg08857.html

The issue was also discussed in the last IETF MMUSIC meeting (see 
http://www.ietf.org/proceedings/81/minutes/mmusic.txt)

To date, we have only seen opposition to this proposal from you (on both 
the mailing list and in the meeting), and we are lacking an alternative 
path forward that can satisfy all involved parties, including the 
general AD and WG chair concerns around definining an alternative NAT 
traversal mechanism in this draft without proper discussion and general 
considerations around the need for an alternative to ICE (per the above 
thread). To recap, the proposal we made was:

<quote>
To summarize, the current latching mechanism and the associated payload 
type definitions for it will be removed, and ICE will be the only NAT 
traversal mechanism specified for loopback in this draft. The loopback 
payload types that are needed in general (i.e. without latching) will 
remain in the draft and the PAYLOAD WG will review them as part of this 
draft.

If the authors and/or the WG believe that we should continue working on 
the latching (or other) NAT traversal mechanisms different from ICE, 
then that is something that will be discussed and/or developed outside 
of the media loopback work.
</quote>

If you do not find this path forward acceptable, can you please suggest 
an alternative that satisfies the general concerns around defining an 
alternative NAT traversal mechanism in this draft ?

Thanks

-- Flemming & Miguel

On 10/5/11 4:10 PM, Hadriel Kaplan wrote:
> When looking at the diff between this version of the draft and the previous, I note the majority of the changes are around removing the "priming" mechanism in its entirety.  And there is only one small paragraph about making this work behind NATs, which basically says "SHOULD use" ICE/STUN/TURN.
>
> What surprises me about this is I don't recall any WG consensus to make the change to pull out the priming mechanism entirely this way.  I only recall discussion around saying one SHOULD use ICE but MAY use the priming mechanism if ICE isn't supported; and keeping the priming mechanism defined in the loopback draft to explain how it works.
>
> So... (1) why was this significant change made without apparent WG consensus, and (2) how do we expect the priming mechanism to interoperate if we don't define how it works?  Or have we decided making loopback calls to SIP UA's behind NATs is no longer important/relevant?
>
> -hadriel
>
>
> On Oct 5, 2011, at 9:52 AM, Miguel A. Garcia wrote:
>
>    
>> This is to announce a 2 week Working Group Last Call for
>>
>>     draft-ietf-mmusic-media-loopback-16
>>
>> as Proposed Standard. Please review and provide any comments you may have on the document before Thursday, October 20th, 2011. Comments should be sent to the document authors and the MMUSIC WG list.
>>
>> Thanks
>>
>>     Miguel&  Flemming
>> -- 
>> Miguel A. Garcia
>> +34-91-339-3608
>> Ericsson Spain
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>      
>
>    

From abegen@cisco.com  Mon Oct 10 11:17:40 2011
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I revised the two drafts per the feedback in Quebec.

If there are further comments, we would like to hear about them. As of =
now, there are no open issues from our standpoint except possibly =
defining a new RTCP XR report, which can be done later. Thus, we would =
like these two drafts to be considered and adopted by the MMUSIC wg.

https://datatracker.ietf.org/doc/draft-begen-mmusic-redundancy-grouping/
https://datatracker.ietf.org/doc/draft-begen-mmusic-temporal-interleaving=
/=20

CC'ing AVT(core) since the topic is relevant but please post your =
comments to mmusic where the drafts belong to.

-acbegen

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On 2011-10-10 20:17, Ali C. Begen (abegen) wrote:
> I revised the two drafts per the feedback in Quebec.
> 
> If there are further comments, we would like to hear about them. As
> of now, there are no open issues from our standpoint except possibly
> defining a new RTCP XR report, which can be done later. Thus, we
> would like these two drafts to be considered and adopted by the
> MMUSIC wg.
> 
> https://datatracker.ietf.org/doc/draft-begen-mmusic-redundancy-grouping/
>
> https://datatracker.ietf.org/doc/draft-begen-mmusic-temporal-interleaving/
> 
> CC'ing AVT(core) since the topic is relevant but please post your
> comments to mmusic where the drafts belong to.
> 

Hi,

(as individual)

I have looked at the redundancy-grouping draft and still think it fails
as RTP specifications for how duplication should be done in RTP, either
using SSRC multiplexing with two different SSRCs in the same RTP session
or for the alternative how having two different RTP sessions. This first
appears to be sufficiently specified, where my only comment is the
dependency on signalling to determine that this grouping is in place. I
do propose that an RTP/RTCP based complement is derived. I would
personally suggest that my own proposal for a new SDES item SRCNAME is
used but that should clearly be discussed.

For the case where multiple RTP sessions are used, it is still not clear
how a receiver determine which SSRC in each RTP session that are
related. And I would like to point out that CNAME is not sufficient.

I still think we should consider splitting this into one AVTCORE
document discussing the RTP aspects of duplication, and another MMUSIC
document defining the grouping semantics.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, October 11, 2011 5:42 AM
> To: Ali C. Begen (abegen)
> Cc: mmusic (E-mail); avt@ietf.org
> Subject: Re: [AVTCORE] Duplication semantics drafts
>=20
> On 2011-10-10 20:17, Ali C. Begen (abegen) wrote:
> > I revised the two drafts per the feedback in Quebec.
> >
> > If there are further comments, we would like to hear about them. As
> > of now, there are no open issues from our standpoint except possibly
> > defining a new RTCP XR report, which can be done later. Thus, we
> > would like these two drafts to be considered and adopted by the
> > MMUSIC wg.
> >
> > =
https://datatracker.ietf.org/doc/draft-begen-mmusic-redundancy-grouping/
> >
> > =
https://datatracker.ietf.org/doc/draft-begen-mmusic-temporal-interleaving=
/
> >
> > CC'ing AVT(core) since the topic is relevant but please post your
> > comments to mmusic where the drafts belong to.
> >
>=20
> Hi,
>=20
> (as individual)
>=20
> I have looked at the redundancy-grouping draft and still think it =
fails
> as RTP specifications for how duplication should be done in RTP, =
either
> using SSRC multiplexing with two different SSRCs in the same RTP =
session
> or for the alternative how having two different RTP sessions. This =
first
> appears to be sufficiently specified, where my only comment is the
> dependency on signalling to determine that this grouping is in place. =
I
> do propose that an RTP/RTCP based complement is derived. I would
> personally suggest that my own proposal for a new SDES item SRCNAME is
> used but that should clearly be discussed.

Right, the ssrc muxing case is straightforward and easy to implement. =
Everything works fine without the need for defining new stuff except the =
grouping semantics. I am not sure why you think there is a problem in =
the signaling.

If duplication will be done across sessions (your second case), then it =
gets complicated. And as I understand your claim is regular CNAME falls =
short to define this relation. Is that correct? And your proposal is =
SRCNAME, right?
=20
> For the case where multiple RTP sessions are used, it is still not =
clear
> how a receiver determine which SSRC in each RTP session that are
> related. And I would like to point out that CNAME is not sufficient.
>=20
> I still think we should consider splitting this into one AVTCORE
> document discussing the RTP aspects of duplication, and another MMUSIC
> document defining the grouping semantics.

This sounds like a good plan to me.
-acbegen
=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------


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On 2011-10-11 12:46, Ali C. Begen (abegen) wrote:
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> (as Individual)

>> I have looked at the redundancy-grouping draft and still think it fails
>> as RTP specifications for how duplication should be done in RTP, either
>> using SSRC multiplexing with two different SSRCs in the same RTP session
>> or for the alternative how having two different RTP sessions. This first
>> appears to be sufficiently specified, where my only comment is the
>> dependency on signalling to determine that this grouping is in place. I
>> do propose that an RTP/RTCP based complement is derived. I would
>> personally suggest that my own proposal for a new SDES item SRCNAME is
>> used but that should clearly be discussed.
> 
> Right, the ssrc muxing case is straightforward and easy to implement.
> Everything works fine without the need for defining new stuff except
> the grouping semantics. I am not sure why you think there is a problem
>  in the signaling.

Yes, it works as long as the application can accept to rely on the
signalling protocol. However, if one would like to to this in session
where one has a dynamic set of participants and it is large enough that
it isn't feasible to updated the signalling for each new participant
that joins the session and is a potential sender. Then it would be good
to have something that works on the RTP/RTCP level.


> 
> If duplication will be done across sessions (your second case), then
it gets complicated. And as I understand your claim is regular CNAME
falls short to define this relation. Is that correct? And your proposal
is SRCNAME, right?

Yes, the show case when CNAME falls short is for any case where an
end-point will have multiple media flows in the same session. So in this
case you have 2 SSRC in one session carrying the source (original)
stream and 2 SSRCs in the second carrying the duplicated data. As the
two source SSRC come from the same synchronization context they do carry
the same CNAME. Thus a receiver will have two sets of SSRCs that both
carry the same CNAME. Thus that can't be used to bind the correct set of
SSRCs together. Using the sequence numbers are prone to errors and can
easily fail. I don't like overriding the SSRC randomization and require
the duplicated streams to have the same SSRC as the original. It also
creates separate solutions for the two cases.

I therefore prefer to see a solution that can handle both cases and
targeted at this particular problem.

There will be a dedicated version of the SRCNAME draft submitted before
the deadline.

> 
>> For the case where multiple RTP sessions are used, it is still not clear
>> how a receiver determine which SSRC in each RTP session that are
>> related. And I would like to point out that CNAME is not sufficient.
>>
>> I still think we should consider splitting this into one AVTCORE
>> document discussing the RTP aspects of duplication, and another MMUSIC
>> document defining the grouping semantics.
> 
> This sounds like a good plan to me.

Good

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From HKaplan@acmepacket.com  Wed Oct 12 07:59:20 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Flemming Andreasen <fandreas@cisco.com>
Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On Oct 8, 2011, at 8:54 AM, Flemming Andreasen wrote:

> The issue was discussed on the mailing list in August, where the authors =
suggested it as a path forward and the chairs asked the WG for feedback on =
this approach, see e.g.
>=20
> http://www.ietf.org/mail-archive/web/mmusic/current/msg08857.html
>=20
> The issue was also discussed in the last IETF MMUSIC meeting (see http://=
www.ietf.org/proceedings/81/minutes/mmusic.txt)
>=20
> To date, we have only seen opposition to this proposal from you (on both =
the mailing list and in the meeting), and we are lacking an alternative pat=
h forward that can satisfy all involved parties, including the general AD a=
nd WG chair concerns around definining an alternative NAT traversal mechani=
sm in this draft without proper discussion and general considerations aroun=
d the need for an alternative to ICE (per the above thread).

In the email thread you referenced above there was in fact someone else who=
 was for keeping the priming mechanism, and it wasn't me.

In fact, in that whole thread there was only one other person than yourself=
 who was in favor of only having ICE in the draft.  Of course there was onl=
y one other person than myself who was in favor of keeping the primer mecha=
nism.

My point is that there has been no consensus either way.  The authors only =
changed the draft due to your objections, afaict, not due to some large con=
sensus to do so.


> If you do not find this path forward acceptable, can you please suggest a=
n alternative that satisfies the general concerns around defining an altern=
ative NAT traversal mechanism in this draft ?

There have been no "general concerns" as far as I can tell, other than "we =
have ICE defined so why not use it".   My concerns with mandating *only* IC=
E as opposed to having a SHOULD use ICE but MAY use primer are detailed her=
e:=20
http://www.ietf.org/mail-archive/web/mmusic/current/msg08611.html

There was no feedback/response to that, afaict.

-hadriel


From internet-drafts@ietf.org  Thu Oct 13 05:19:02 2011
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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=
 Working Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-08.txt
	Pages           : 30
	Date            : 2011-10-13

   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).


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt

From Simo.Veikkolainen@nokia.com  Thu Oct 13 05:21:12 2011
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Subject: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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Hello all,

We just submitted a new version of draft-ietf-mmusic-sdp-cs.

Changes in the -08 version include:

Added a new section "Offer/Answer extensions" detailing the process of hand=
ling the O/A exchange.=20

Clarified that multiple extensions for cs-correlation are allowed.

Added a request for IANA to create a sub-registry for "cs-correlation" attr=
ibute values, with "Specification Required" policy.=20

Clarified how dynamic payloads are handled. Basically we stick to normal SD=
P processing here, i.e. if dynamic payload types are used, a=3Drtpmap attri=
bute(s) are required. This may need more thought still.

Changed a lot of SHOULDs into MUSTs.

Extended the example in Section 6.

Flemming had a comment on mandating E.164 numbers in the international form=
at. After some thinking, we still say that the enpoints SHOULD include the =
number in international format. The network may anyway mangle the number pr=
esentation, so we thought mandating the intl format is not absolutely neces=
sary.

TODO: Security considerations needs more work based on Flemming's earlier c=
omments (including caller's E164 number in SDP).
TODO: Clarification on the mechanism works with 3PCC.
TODO:Change the UUIE syntax to allow for a discriminator byte.

We hope to address the remaning issues still before the IETF submission dea=
dline.

Regards,
Simo


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext internet-drafts@ietf.org
Sent: 13 October, 2011 15:19
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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=
 Working Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-08.txt
	Pages           : 30
	Date            : 2011-10-13

   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).


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic

From magnus.westerlund@ericsson.com  Fri Oct 14 06:19:36 2011
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MMUSIC,

I have submitted a draft that looks as multiplexing multiple RTP
sessions on the same lower layer transport. It is also extending
draft-holmberg-mmusic-sdp-multiplex-negotiation-00 to achieve
negotiation of the multiplexing taking place and which session IDs to
use for each media description.

Feedback appreciated

Magnus Westerlund

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : Multiple RTP Session on a Single Lower-Layer Transport
	Author(s)       : Magnus Westerlund
	Filename        : draft-westerlund-avtcore-transport-multiplexing-00.txt
	Pages           : 23
	Date            : 2011-10-14

   This document specifies how multiple RTP sessions are to be
   multiplexed on the same lower-layer transport, e.g. a UDP flow.  It
   discusses various requirements that have been raised and their
   feasibility, which results in a solution with a certain
   applicability.  A solution is recommended and that solution is
   provided in more detail, including signalling and examples.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-westerlund-avtcore-transport-mult=
iplexing-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-westerlund-avtcore-transport-multi=
plexing-00.txt
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From pkyzivat@alum.mit.edu  Fri Oct 14 11:46:37 2011
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Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of comments:

5.2.1

This says addresses for the E164 address type are represented according 
to ITU-T E.164. If I have followed the reference correctly, it only 
specifies the country codes, not the encoding of numbers including 
country codes. I think it would be better to define this by reference to 
RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or 
global-number-digits defined there. (I'm not sure which of those is most 
appropriate for your use.)

Also, you give the example

     c=PSTN - -

That requires that "-" be a defined address type. AFAIK it is not.
Did you mean it to be

     c=PSTN E164 -

5.2.3.1

The syntax given for a=cs-correlation is inconsistent with the syntax in 
5.6 and with usage throughout the document.

5.2.3.3

You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the 
actual max information content is actually either 32 or 128 octets plus 
a leading protocol discriminator byte.

Its unclear what you intend to be included in the specified uuie element 
in the sdp - does it include the protocol discriminator or not?

Also, the content and protocol discriminator contain octets, not just 
alphanumerics, but the syntax in 5.6 only permits alphanumerics. Is it 
your *intent* to limit the protocol discriminator and the value to that 
subset described by alphanumerics? And is that a reasonable thing to do? 
(For this use, it *might* be sufficient to prescribe a specific value 
for the protocol discriminator.)

5.5.1 / 5.5.4

The description of a=connection here is more appropriate to section 
5.5.4. Here, for the initial offer, a=connection:new is the only thing 
that makes sense.

In 5.5.4 it would be helpful to explain in more detail how a=connection 
is used to control whether an existing call is retained, dropped, or 
dropped and reestablished.

(In some cases it will be desired to do another o/a while leaving the 
pstn call undisturbed. In other cases it may the explicit intent to 
cause the call to be reestablished.)

5.6

Is there a reason why you don't use the definition of an E.164 number 
from RFC 3966, rather than using your own? What is the significance of 
making the "+" optional? And why is caller-id-value defined differently 
than e164-address?

Figure 4, and others:

The E.164 number used (+15551234) isn't a valid one for country code 1.

General Comment:

Everything is described, and examples show, only a single m= line, and 
all the cs-correlation and setup things used within that one media 
section. There is no explanation of what to do if there is more than one 
pstn m= line, as might arise if the intent is to have an audio/video 
call. Without any explanation, it would seem that everything should be 
specified separately for each m= line, which might result in *two* pstn 
calls. Presumably that isn't want is desired.

More definition is needed if the same pstn call is to be used for two 
media streams.

	Thanks,
	Paul

On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
> Hello all,
>
> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
>
> Changes in the -08 version include:
>
> Added a new section "Offer/Answer extensions" detailing the process of handling the O/A exchange.
>
> Clarified that multiple extensions for cs-correlation are allowed.
>
> Added a request for IANA to create a sub-registry for "cs-correlation" attribute values, with "Specification Required" policy.
>
> Clarified how dynamic payloads are handled. Basically we stick to normal SDP processing here, i.e. if dynamic payload types are used, a=rtpmap attribute(s) are required. This may need more thought still.
>
> Changed a lot of SHOULDs into MUSTs.
>
> Extended the example in Section 6.
>
> Flemming had a comment on mandating E.164 numbers in the international format. After some thinking, we still say that the enpoints SHOULD include the number in international format. The network may anyway mangle the number presentation, so we thought mandating the intl format is not absolutely necessary.
>
> TODO: Security considerations needs more work based on Flemming's earlier comments (including caller's E164 number in SDP).
> TODO: Clarification on the mechanism works with 3PCC.
> TODO:Change the UUIE syntax to allow for a discriminator byte.
>
> We hope to address the remaning issues still before the IETF submission deadline.
>
> Regards,
> Simo
>
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of ext internet-drafts@ietf.org
> Sent: 13 October, 2011 15:19
> To: i-d-announce@ietf.org
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
> 	Author(s)       : Miguel A. Garcia-Martin
>                            Simo Veikkolainen
> 	Filename        : draft-ietf-mmusic-sdp-cs-08.txt
> 	Pages           : 30
> 	Date            : 2011-10-13
>
>     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).
>
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
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> mmusic@ietf.org
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>


From keith.drage@alcatel-lucent.com  Sat Oct 15 05:52:57 2011
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
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Paul wrote

> 5.2.1
>=20
> This says addresses for the E164 address type are represented according
> to ITU-T E.164. If I have followed the reference correctly, it only
> specifies the country codes, not the encoding of numbers including
> country codes. I think it would be better to define this by reference to
> RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or
> global-number-digits defined there. (I'm not sure which of those is most
> appropriate for your use.)
>

E.164 defines the country codes in an annex. The main body of the recommend=
ation defines the structure of a number consisting of country code and nati=
onal significant number for international number. And national and subscrib=
er prefix and appropriate part of the number for national number and subscr=
iber number usage. The definition also includes various length limitations,=
 and identifies local service numbers.

However I suspect that the reference here needs to explicitly limit it to i=
nternational numbers defined by E.164.

I see no need to go via RFC 3966 which becomes only significant in terms of=
 referencing if you want to define local numbers.

Regards

Keith

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Paul Kyzivat
> Sent: 14 October 2011 19:47
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
>=20
> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of comments:
>=20
> 5.2.1
>=20
> This says addresses for the E164 address type are represented according
> to ITU-T E.164. If I have followed the reference correctly, it only
> specifies the country codes, not the encoding of numbers including
> country codes. I think it would be better to define this by reference to
> RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or
> global-number-digits defined there. (I'm not sure which of those is most
> appropriate for your use.)
>=20
> Also, you give the example
>=20
>      c=3DPSTN - -
>=20
> That requires that "-" be a defined address type. AFAIK it is not.
> Did you mean it to be
>=20
>      c=3DPSTN E164 -
>=20
> 5.2.3.1
>=20
> The syntax given for a=3Dcs-correlation is inconsistent with the syntax i=
n
> 5.6 and with usage throughout the document.
>=20
> 5.2.3.3
>=20
> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the
> actual max information content is actually either 32 or 128 octets plus
> a leading protocol discriminator byte.
>=20
> Its unclear what you intend to be included in the specified uuie element
> in the sdp - does it include the protocol discriminator or not?
>=20
> Also, the content and protocol discriminator contain octets, not just
> alphanumerics, but the syntax in 5.6 only permits alphanumerics. Is it
> your *intent* to limit the protocol discriminator and the value to that
> subset described by alphanumerics? And is that a reasonable thing to do?
> (For this use, it *might* be sufficient to prescribe a specific value
> for the protocol discriminator.)
>=20
> 5.5.1 / 5.5.4
>=20
> The description of a=3Dconnection here is more appropriate to section
> 5.5.4. Here, for the initial offer, a=3Dconnection:new is the only thing
> that makes sense.
>=20
> In 5.5.4 it would be helpful to explain in more detail how a=3Dconnection
> is used to control whether an existing call is retained, dropped, or
> dropped and reestablished.
>=20
> (In some cases it will be desired to do another o/a while leaving the
> pstn call undisturbed. In other cases it may the explicit intent to
> cause the call to be reestablished.)
>=20
> 5.6
>=20
> Is there a reason why you don't use the definition of an E.164 number
> from RFC 3966, rather than using your own? What is the significance of
> making the "+" optional? And why is caller-id-value defined differently
> than e164-address?
>=20
> Figure 4, and others:
>=20
> The E.164 number used (+15551234) isn't a valid one for country code 1.
>=20
> General Comment:
>=20
> Everything is described, and examples show, only a single m=3D line, and
> all the cs-correlation and setup things used within that one media
> section. There is no explanation of what to do if there is more than one
> pstn m=3D line, as might arise if the intent is to have an audio/video
> call. Without any explanation, it would seem that everything should be
> specified separately for each m=3D line, which might result in *two* pstn
> calls. Presumably that isn't want is desired.
>=20
> More definition is needed if the same pstn call is to be used for two
> media streams.
>=20
> 	Thanks,
> 	Paul
>=20
> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
> > Hello all,
> >
> > We just submitted a new version of draft-ietf-mmusic-sdp-cs.
> >
> > Changes in the -08 version include:
> >
> > Added a new section "Offer/Answer extensions" detailing the process of
> handling the O/A exchange.
> >
> > Clarified that multiple extensions for cs-correlation are allowed.
> >
> > Added a request for IANA to create a sub-registry for "cs-correlation"
> attribute values, with "Specification Required" policy.
> >
> > Clarified how dynamic payloads are handled. Basically we stick to norma=
l
> SDP processing here, i.e. if dynamic payload types are used, a=3Drtpmap
> attribute(s) are required. This may need more thought still.
> >
> > Changed a lot of SHOULDs into MUSTs.
> >
> > Extended the example in Section 6.
> >
> > Flemming had a comment on mandating E.164 numbers in the international
> format. After some thinking, we still say that the enpoints SHOULD includ=
e
> the number in international format. The network may anyway mangle the
> number presentation, so we thought mandating the intl format is not
> absolutely necessary.
> >
> > TODO: Security considerations needs more work based on Flemming's
> earlier comments (including caller's E164 number in SDP).
> > TODO: Clarification on the mechanism works with 3PCC.
> > TODO:Change the UUIE syntax to allow for a discriminator byte.
> >
> > We hope to address the remaning issues still before the IETF submission
> deadline.
> >
> > Regards,
> > Simo
> >
> >
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behal=
f
> Of ext internet-drafts@ietf.org
> > Sent: 13 October, 2011 15:19
> > To: i-d-announce@ietf.org
> > Cc: mmusic@ietf.org
> > Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched Bearers In
> The Public Switched Telephone Network (PSTN)
> > 	Author(s)       : Miguel A. Garcia-Martin
> >                            Simo Veikkolainen
> > 	Filename        : draft-ietf-mmusic-sdp-cs-08.txt
> > 	Pages           : 30
> > 	Date            : 2011-10-13
> >
> >     This memo describes use cases, requirements, and protocol extension=
s
> >     for using the Session Description Protocol (SDP) Offer/Answer model
> >     for establishing audio and video media streams over circuit-switche=
d
> >     bearers in the Public Switched Telephone Network (PSTN).
> >
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > This Internet-Draft can be retrieved at:
> > ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
> > _______________________________________________
> > 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
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From capelastegui@dit.upm.es  Sat Oct 15 06:49:08 2011
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From: "Pedro Capelastegui" <capelastegui@dit.upm.es>
To: "'IETF - MMUSIC'" <mmusic@ietf.org>, "'Bert Greevenbosch'" <Bert.Greevenbosch@huawei.com>
Date: Fri, 14 Oct 2011 18:18:10 +0200
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Subject: [MMUSIC] Comments on draft-greevenbosch-mmusic-signal-3d-format-01
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Hi,

=20

I have reviewed draft-greevenbosch-mmusic-signal-3d-format-01, and I =
have the following comments.

=20

                - Backwards compatibility

=20

When a 3D-enabled device using this specification calls a legacy device, =
it is very likely that the answerer will accept the session and end up =
receiving media that it can=E2=80=99t display properly, such as =
frame-packed video or a depth map. I think the default behaviour here =
should be that whenever the SDP answer omits the 3dFormat attribute, the =
offerer only sends 2D video (if available), avoiding further offers if =
possible. This speeds up session setup with legacy agents, and prevents =
transmitting useless media.

=20

To implement this change, I would modify a few paragraphs in section 8.

=20

In 8.1 (Frame Packing), replace:

=20

=E2=80=9Cthe offerer MAY update the offer with a 2D stream.  If the =
offerer is the streaming server, it  MAY choose to stream the frame =
packed video as it is.=E2=80=9C

=20

with

=20

=E2=80=9Cthe offerer SHOULD treat that media stream as a 2D =
stream.=E2=80=9D

=20

In 8.2 (2D and auxiliary as a single stream), replace:

               =20

=E2=80=9C the offerer MAY update its offer by offering only a 2D  video =
stream.=E2=80=9D

=20

with

               =20

=E2=80=9CSHOULD treat that media stream as a 2D stream.=E2=80=9D

=20

In 8.3 (2D and auxiliary as two separate streams), replace:

               =20

" If the answerer selects only the auxiliary video, the offerer  SHOULD =
update its offer without the auxiliary video."

=20

with

               =20

"If the answerer selects only the auxiliary video, the offerer  MAY =
treat that stream as a 2D video stream. If it does not, the offerer =
SHOULD update its offer without the auxiliary video."

=E2=80=83

=20

                - H.264/MVC

=20

I think that compatibility with the MVC format is a must for this =
specification. This will probably require defining new types of 3D =
formats and components. Unfortunately, the payload format for MVC =
(draft-ietf-payload-rtp-mvc-0) is still unfinished, with most of the SDP =
section missing=E2=80=A6

=20

=20

                - Depth Maps in auxiliary streams.

=20

Some transmission options use a single stream containing one view =
(centre or left) and its associated depth map (or parallax map). =
However, there is no mention as to how the view and map are packed in =
that stream. I suppose the specification assumes the use of the =
auxiliary data format defined in MPEG-C part 3, but this should probably =
be indicated explicitly.

=20

Also, perhaps the =E2=80=9C2DA=E2=80=9D format should leave the door =
open to alternate packing mechanisms other than MPEG-C part 3. I =
don=E2=80=99t know of any currently existing alternatives, but it is not =
unreasonable to expect them at some point in the future.

=20

               =20

                - Parallax Maps

=20

The draft provides no reference for the use of parallax maps, so =
it=E2=80=99s probably worth mentioning that they are defined, like depth =
maps, in [ISO/IEC 23002-3]. Also, I have not found any examples of 3D =
video applications based on parallax maps in the literature.  How likely =
is it that this scenario will see actual use?

=20

=20

                - Frame Sequential packing

=20

I think that the mechanism for identifying L- and R- streams in a frame =
sequential stream should be clearly explained.

=20

=20

                - Multiple 3D streams

=20

On page 14, it says that =E2=80=9CThe answerer SHOULD select only one 3D =
format=E2=80=9D. This means that, for any given session, only one 3D =
video stream can be used. I think this is too limiting, since it =
prevents describing sessions where one agent has multiple 3D displays, =
among others.=20

=20

=20

                - Multiple Depth Maps.

=20

The document assumes that depth maps will only be used in 2D+depth =
streams. However, I think that the combination of multiple views and =
depth maps (e.g. L/R views and 2 depth maps) is an interesting scenario =
that may deserve consideration.

=20

=20

Regards,

Pedro

=20

=20

-------------------------------------------------------------------------=
--

=20

Pedro Capelastegui

Researcher

Universidad Polit=C3=A9cnica de Madrid


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-GB link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoPlainText>Hi,<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I have =
reviewed draft-greevenbosch-mmusic-signal-3d-format-01, and I have the =
following comments.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Backwards =
compatibility</u><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>When a =
3D-enabled device using this specification calls a legacy device, it is =
very likely that the answerer will accept the session and end up =
receiving media that it can=E2=80=99t display properly, such as =
frame-packed video or a depth map. I think the default behaviour here =
should be that whenever the SDP answer omits the 3dFormat attribute, the =
offerer only sends 2D video (if available), avoiding further offers if =
possible. This speeds up session setup with legacy agents, and prevents =
transmitting useless media.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>To =
implement this change, I would modify a few paragraphs in section =
8.<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>In 8.1 (Frame Packing), replace:<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=E2=80=9C<i>the offerer MAY update the offer with a =
2D stream.=C2=A0 If the offerer is the streaming server, it=C2=A0 MAY =
choose to stream the frame packed video as it =
is</i>.=E2=80=9C<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>with<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=E2=80=9C<i>the offerer SHOULD treat that media =
stream as a 2D stream.</i>=E2=80=9D<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>In 8.2 =
(2D and auxiliary as a single stream), replace:<o:p></o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <o:p></o:p></p><p =
class=3DMsoPlainText>=E2=80=9C <i>the offerer MAY update its offer by =
offering only a 2D=C2=A0 video stream.</i>=E2=80=9D<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>with<o:p></o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <o:p></o:p></p><p =
class=3DMsoPlainText>=E2=80=9C<i>SHOULD treat that media stream as a 2D =
stream.</i>=E2=80=9D<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>In 8.3 =
(2D and auxiliary as two separate streams), replace:<o:p></o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <o:p></o:p></p><p =
class=3DMsoPlainText>&quot; <i>If the answerer selects only the =
auxiliary video, the offerer=C2=A0 SHOULD update its offer without the =
auxiliary video</i>.&quot;<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>with<o:p></o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <o:p></o:p></p><p =
class=3DMsoPlainText>&quot;<i>If the answerer selects only the auxiliary =
video, the offerer=C2=A0 MAY treat that stream as a 2D video stream. If =
it does not, the offerer SHOULD update its offer without the auxiliary =
video.</i>&quot;<o:p></o:p></p><p =
class=3DMsoPlainText>=E2=80=83<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>H.264/MVC</u><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I =
think that compatibility with the MVC format is a must for this =
specification. This will probably require defining new types of 3D =
formats and components. Unfortunately, the payload format for MVC =
(draft-ietf-payload-rtp-mvc-0) is still unfinished, with most of the SDP =
section missing=E2=80=A6<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Depth Maps in auxiliary =
streams.</u><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Some =
transmission options use a single stream containing one view (centre or =
left) and its associated depth map (or parallax map). However, there is =
no mention as to how the view and map are packed in that stream. I =
suppose the specification assumes the use of the auxiliary data format =
defined in MPEG-C part 3, but this should probably be indicated =
explicitly.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Also, =
perhaps the =E2=80=9C2DA=E2=80=9D format should leave the door open to =
alternate packing mechanisms other than MPEG-C part 3. I don=E2=80=99t =
know of any currently existing alternatives, but it is not unreasonable =
to expect them at some point in the future.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <o:p></o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Parallax =
Maps</u><o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>The draft provides no reference for the use of =
parallax maps, so it=E2=80=99s probably worth mentioning that they are =
defined, like depth maps, in [ISO/IEC 23002-3]. Also, I have not found =
any examples of 3D video applications based on parallax maps in the =
literature.=C2=A0 How likely is it that this scenario will see actual =
use?<o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Frame Sequential =
packing</u><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>I =
think that the mechanism for identifying L- and R- streams in a frame =
sequential stream should be clearly explained.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Multiple 3D =
streams</u><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>On =
page 14, it says that =E2=80=9CThe answerer SHOULD select only one 3D =
format=E2=80=9D. This means that, for any given session, only one 3D =
video stream can be used. I think this is too limiting, since it =
prevents describing sessions where one agent has multiple 3D displays, =
among others. <o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 - <u>Multiple Depth =
Maps.</u><o:p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>The document assumes that depth maps will only be =
used in 2D+depth streams. However, I think that the combination of =
multiple views and depth maps (e.g. L/R views and 2 depth maps) is an =
interesting scenario that may deserve consideration.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>Regards,<o:p></o:p></p><p =
class=3DMsoPlainText>Pedro<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>----------------------------------------------------=
-----------------------<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Pedro =
Capelastegui<o:p></o:p></p><p =
class=3DMsoPlainText>Researcher<o:p></o:p></p><p =
class=3DMsoPlainText>Universidad Polit=C3=A9cnica de =
Madrid<o:p></o:p></p></div></body></html>
------=_NextPart_000_0010_01CC8A9D.A33CC870--


From miguel.a.garcia@ericsson.com  Sun Oct 16 01:25:17 2011
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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Hi Keith.

I am a bit confused with your comment indicated that the reference should 
be limited to international numbers defined by E.164. I think E.164 is 
all about international numbers. As a matter of fact, the title of E.164 
is "The international public telecommunication numbering plan", and the 
scope clearly indicates that it is about international numbers.

However, I think we need to elaborate a bit more the '+' sign, which is 
not part of the E.164 specification, but a common trick to represent a 
full international E.164 number.

/Miguel

On 15/10/2011 14:52, DRAGE, (Keith) wrote:
> Paul wrote
>
>> 5.2.1
>>
>> This says addresses for the E164 address type are represented
>> according to ITU-T E.164. If I have followed the reference
>> correctly, it only specifies the country codes, not the encoding of
>> numbers including country codes. I think it would be better to
>> define this by reference to RFC 3966 (TEL) and to either
>> telephone-subscriber, or global-number, or global-number-digits
>> defined there. (I'm not sure which of those is most appropriate for
>> your use.)
>>
>
> E.164 defines the country codes in an annex. The main body of the
> recommendation defines the structure of a number consisting of country
> code and national significant number for international number. And
> national and subscriber prefix and appropriate part of the number for
> national number and subscriber number usage. The definition also
> includes various length limitations, and identifies local service
> numbers.
>
> However I suspect that the reference here needs to explicitly limit it
> to international numbers defined by E.164.
>
> I see no need to go via RFC 3966 which becomes only significant in
> terms of referencing if you want to define local numbers.
>
> Regards
>
> Keith
>
>> -----Original Message----- From: mmusic-bounces@ietf.org
>> [mailto:mmusic-bounces@ietf.org] On Behalf Of Paul Kyzivat Sent: 14
>> October 2011 19:47 To: mmusic@ietf.org Subject: Re: [MMUSIC] FW: I-D
>> Action: draft-ietf-mmusic-sdp-cs-08.txt
>>
>> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of
>> comments:
>>
>> 5.2.1
>>
>> This says addresses for the E164 address type are represented
>> according to ITU-T E.164. If I have followed the reference
>> correctly, it only specifies the country codes, not the encoding of
>> numbers including country codes. I think it would be better to
>> define this by reference to RFC 3966 (TEL) and to either
>> telephone-subscriber, or global-number, or global-number-digits
>> defined there. (I'm not sure which of those is most appropriate for
>> your use.)
>>
>> Also, you give the example
>>
>> c=PSTN - -
>>
>> That requires that "-" be a defined address type. AFAIK it is not.
>> Did you mean it to be
>>
>> c=PSTN E164 -
>>
>> 5.2.3.1
>>
>> The syntax given for a=cs-correlation is inconsistent with the
>> syntax in 5.6 and with usage throughout the document.
>>
>> 5.2.3.3
>>
>> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK
>> the actual max information content is actually either 32 or 128
>> octets plus a leading protocol discriminator byte.
>>
>> Its unclear what you intend to be included in the specified uuie
>> element in the sdp - does it include the protocol discriminator or
>> not?
>>
>> Also, the content and protocol discriminator contain octets, not
>> just alphanumerics, but the syntax in 5.6 only permits
>> alphanumerics. Is it your *intent* to limit the protocol
>> discriminator and the value to that subset described by
>> alphanumerics? And is that a reasonable thing to do? (For this use,
>> it *might* be sufficient to prescribe a specific value for the
>> protocol discriminator.)
>>
>> 5.5.1 / 5.5.4
>>
>> The description of a=connection here is more appropriate to section
>> 5.5.4. Here, for the initial offer, a=connection:new is the only
>> thing that makes sense.
>>
>> In 5.5.4 it would be helpful to explain in more detail how
>> a=connection is used to control whether an existing call is
>> retained, dropped, or dropped and reestablished.
>>
>> (In some cases it will be desired to do another o/a while leaving
>> the pstn call undisturbed. In other cases it may the explicit intent
>> to cause the call to be reestablished.)
>>
>> 5.6
>>
>> Is there a reason why you don't use the definition of an E.164
>> number from RFC 3966, rather than using your own? What is the
>> significance of making the "+" optional? And why is caller-id-value
>> defined differently than e164-address?
>>
>> Figure 4, and others:
>>
>> The E.164 number used (+15551234) isn't a valid one for country code
>> 1.
>>
>> General Comment:
>>
>> Everything is described, and examples show, only a single m= line,
>> and all the cs-correlation and setup things used within that one
>> media section. There is no explanation of what to do if there is
>> more than one pstn m= line, as might arise if the intent is to have
>> an audio/video call. Without any explanation, it would seem that
>> everything should be specified separately for each m= line, which
>> might result in *two* pstn calls. Presumably that isn't want is
>> desired.
>>
>> More definition is needed if the same pstn call is to be used for
>> two media streams.
>>
>> Thanks, Paul
>>
>> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
>>> Hello all,
>>>
>>> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
>>>
>>> Changes in the -08 version include:
>>>
>>> Added a new section "Offer/Answer extensions" detailing the
>>> process of
>> handling the O/A exchange.
>>>
>>> Clarified that multiple extensions for cs-correlation are
>>> allowed.
>>>
>>> Added a request for IANA to create a sub-registry for
>>> "cs-correlation"
>> attribute values, with "Specification Required" policy.
>>>
>>> Clarified how dynamic payloads are handled. Basically we stick to
>>> normal
>> SDP processing here, i.e. if dynamic payload types are used,
>> a=rtpmap attribute(s) are required. This may need more thought
>> still.
>>>
>>> Changed a lot of SHOULDs into MUSTs.
>>>
>>> Extended the example in Section 6.
>>>
>>> Flemming had a comment on mandating E.164 numbers in the
>>> international
>> format. After some thinking, we still say that the enpoints SHOULD
>> include the number in international format. The network may anyway
>> mangle the number presentation, so we thought mandating the intl
>> format is not absolutely necessary.
>>>
>>> TODO: Security considerations needs more work based on Flemming's
>> earlier comments (including caller's E164 number in SDP).
>>> TODO: Clarification on the mechanism works with 3PCC. TODO:Change
>>> the UUIE syntax to allow for a discriminator byte.
>>>
>>> We hope to address the remaning issues still before the IETF
>>> submission
>> deadline.
>>>
>>> Regards, Simo
>>>
>>>
>>> -----Original Message----- From: mmusic-bounces@ietf.org
>>> [mailto:mmusic-bounces@ietf.org] On Behalf
>> Of ext internet-drafts@ietf.org
>>> Sent: 13 October, 2011 15:19 To: i-d-announce@ietf.org Cc:
>>> mmusic@ietf.org Subject: [MMUSIC] I-D Action:
>>> draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched
>> Bearers In The Public Switched Telephone Network (PSTN)
>>> Author(s)       : Miguel A. Garcia-Martin Simo Veikkolainen
>>> Filename        : draft-ietf-mmusic-sdp-cs-08.txt Pages
>>> : 30 Date            : 2011-10-13
>>>
>>> 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).
>>>
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>>
>>>
>>>
Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> This Internet-Draft can be retrieved at:
>>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>>
>>>
_______________________________________________
>>> mmusic mailing list mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> _______________________________________________ mmusic mailing
>>> list mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________ mmusic mailing list
>> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________ mmusic mailing list
> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From miguel.a.garcia@ericsson.com  Sun Oct 16 03:16:34 2011
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Hi Paul.

You can find my comments inline, except for one that we are discussing 
with Keith in a separate e-mail.

On 14/10/2011 20:46, Paul Kyzivat wrote:
> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of comments:
>
> 5.2.1
>
> This says addresses for the E164 address type are represented according
> to ITU-T E.164. If I have followed the reference correctly, it only
> specifies the country codes, not the encoding of numbers including
> country codes. I think it would be better to define this by reference to
> RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or
> global-number-digits defined there. (I'm not sure which of those is most
> appropriate for your use.)
>
> Also, you give the example
>
>       c=PSTN - -
>
> That requires that "-" be a defined address type. AFAIK it is not.
> Did you mean it to be
>
>       c=PSTN E164 -


No, we meant:

     c=PSTN - -

The idea is frequently a device will know that is Circuit-Switched 
capable, but it wouldn't know its own E.164 number, or even it is 
allocated an E.164 number. It could be possible that the device is aware 
of a different numbering plan (for example, E.212).

So, in summary: we want to express that the device has a PSTN 
Circuit-Switched connection, but it is not aware of its address in any 
numbering plan.

In Section 8.3, we register with IANA both the E.164 and the dash "-" tokens.



>
> 5.2.3.1
>
> The syntax given for a=cs-correlation is inconsistent with the syntax in
> 5.6 and with usage throughout the document.

I am comparing Sections 5.2.3.1 and 5.6, and the usage throughout the 
document and I don't find this inconsistency. Can you elaborate a bit 
more what this inconsistency is?


>
> 5.2.3.3
>
> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the
> actual max information content is actually either 32 or 128 octets plus
> a leading protocol discriminator byte.

Right. The terminology in Q.931 helps to mislead. The User-User 
Information Element contains three parts:

- 1-octet User-user information element identifier
- 1-octet "Length of user-user contents"
- 1-octet "Protocol discriminator"
- "User Information".

The "User Information can be up to 32 or 128 octets, depending on 
networks. Together with the other 3 octets, it makes a maximum of 35 or 
131 octets.

I guess the draft should elaborate the composition of this information 
element and clarify that it includes the 3 initial octets.


>
> Its unclear what you intend to be included in the specified uuie element
> in the sdp - does it include the protocol discriminator or not?

Yes, as I described above.

>
> Also, the content and protocol discriminator contain octets, not just
> alphanumerics, but the syntax in 5.6 only permits alphanumerics. Is it
> your *intent* to limit the protocol discriminator and the value to that
> subset described by alphanumerics? And is that a reasonable thing to do?
> (For this use, it *might* be sufficient to prescribe a specific value
> for the protocol discriminator.)

This is an error that we should fix. It is not the intention of this 
draft to limit the protocol discriminator, so, we need to address this 
issue and allow any possible value. We will fix it in the next version of 
the draft.



>
> 5.5.1 / 5.5.4
>
> The description of a=connection here is more appropriate to section
> 5.5.4. Here, for the initial offer, a=connection:new is the only thing
> that makes sense.

Ok, we can move the a=connection:existing to 5.5.4.

>
> In 5.5.4 it would be helpful to explain in more detail how a=connection
> is used to control whether an existing call is retained, dropped, or
> dropped and reestablished.
 >
>
> (In some cases it will be desired to do another o/a while leaving the
> pstn call undisturbed. In other cases it may the explicit intent to
> cause the call to be reestablished.)

ok, will do.


>
> 5.6
>
> Is there a reason why you don't use the definition of an E.164 number
> from RFC 3966, rather than using your own?

RFC 3966 defines how to represent an E.164 number as a URL. Here, we do 
not intent to use URLs. Additionally, the tel: URL provide representation 
of either global and local numbers, ISDN subaddresses, extensions, and 
phone context, which are not intended to be used here.

 > What is the significance of
> making the "+" optional?

The only reason I can think, is that the draft has a "SHOULD" strength 
for using the international format of E.164. If someone really knows what 
he is doing and can use a national format, it wouldn't use the "+" sign. 
I recognize that this is not a very strong reason, but it is the only 
justification I can find.



 > And why is caller-id-value defined differently
> than e164-address?

Yes, I think they should be equal. Probably we should get rid of 
"caller-id-value" and use "e164-address" instead.



> Figure 4, and others:
>
> The E.164 number used (+15551234) isn't a valid one for country code 1.

Is it because there are not enough number of digits for the subscriber 
number?

>
> General Comment:
>
> Everything is described, and examples show, only a single m= line, and
> all the cs-correlation and setup things used within that one media
> section. There is no explanation of what to do if there is more than one
> pstn m= line, as might arise if the intent is to have an audio/video
> call. Without any explanation, it would seem that everything should be
> specified separately for each m= line, which might result in *two* pstn
> calls. Presumably that isn't want is desired.
>
> More definition is needed if the same pstn call is to be used for two
> media streams.

Yes, I agree with you. I think we should add a section describing how to 
setup an circuit-switched audio/video call.

>
> 	Thanks,
> 	Paul

Thanks for your comments,

        Miguel


>
> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
>> Hello all,
>>
>> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
>>
>> Changes in the -08 version include:
>>
>> Added a new section "Offer/Answer extensions" detailing the process of handling the O/A exchange.
>>
>> Clarified that multiple extensions for cs-correlation are allowed.
>>
>> Added a request for IANA to create a sub-registry for "cs-correlation" attribute values, with "Specification Required" policy.
>>
>> Clarified how dynamic payloads are handled. Basically we stick to normal SDP processing here, i.e. if dynamic payload types are used, a=rtpmap attribute(s) are required. This may need more thought still.
>>
>> Changed a lot of SHOULDs into MUSTs.
>>
>> Extended the example in Section 6.
>>
>> Flemming had a comment on mandating E.164 numbers in the international format. After some thinking, we still say that the enpoints SHOULD include the number in international format. The network may anyway mangle the number presentation, so we thought mandating the intl format is not absolutely necessary.
>>
>> TODO: Security considerations needs more work based on Flemming's earlier comments (including caller's E164 number in SDP).
>> TODO: Clarification on the mechanism works with 3PCC.
>> TODO:Change the UUIE syntax to allow for a discriminator byte.
>>
>> We hope to address the remaning issues still before the IETF submission deadline.
>>
>> Regards,
>> Simo
>>
>>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of ext internet-drafts@ietf.org
>> Sent: 13 October, 2011 15:19
>> To: i-d-announce@ietf.org
>> Cc: mmusic@ietf.org
>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
>> 	Author(s)       : Miguel A. Garcia-Martin
>>                             Simo Veikkolainen
>> 	Filename        : draft-ietf-mmusic-sdp-cs-08.txt
>> 	Pages           : 30
>> 	Date            : 2011-10-13
>>
>>      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).
>>
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> This Internet-Draft can be retrieved at:
>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>> _______________________________________________
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>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>> _______________________________________________
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>
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-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From miguel.a.garcia@ericsson.com  Sun Oct 16 13:42:35 2011
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Subject: [MMUSIC] Comments to draft-ietf-mmusic-media-loopback-16
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Hi,

Here are my comments to version 16 of the media loopback draft.

- Since this draft defines both an SDP extension and an RTP payload 
extension, I wonder whether this draft has been reviewed by the AVTEXT 
working group. Can the authors comment?

- Continuing with the comment above, I noticed that the title of the 
draft reflects the SDP extension, but is mute towards the RTP extension. 
I recommend to become "An extension to SDP and RTP for media loopback". 
This clearly reflects the actual contents of the draft.

- Terminology. I don't find any justification for this draft to depend on 
RFC 3261 and for making references to SIP INVITEs and alike. Instead, the 
draft should be based on the SDP offer/answer model (RFC 3264). As part 
of this, I will add a paragraph in Section 2 saying that ... "The 
following terms are borrowed from RFC 3264: offer, offerer, answer, 
answerer". I will rename section 3 to "SDP offerer behavior", Section 4 
to "SDP answerer behavior". An in general, I will not talk of the 
"offering entity" (section 3), but the "offerer", or the "answering 
entity" (section 4), but the "answerer".

- I will add a new paragraph at the end of Section 3 indicating where the 
"loopback" attribute is defined.

- Section 4, the last paragraph is not correct, because it puts a 
normative requirement (MAY) for answerers that are not compliant with 
this specification. So, if an answerer is not compliant with this 
specification, how can it comply with the normative MAY?

I suggest to replace the "MAY" with "is expected".

- I would rename Section 5 to "New SDP attributes"

- I would start Section 5.1 with a sentence that says: "This 
specification defines a new 'loopback' attribute which indicates the type 
of loopback that the entity is able to do".

- The syntax of the loopback attribute in Section 5.1 can be enhanced. I 
believe this one is more according to what we are used in SDP (only the 
first three lines differ with the draft):

         attribute             /= loopback-attr
         ; attribute defined in RFC 4566
         loopback-attr          = "loopback:" loopback-type
         loopback-type          = loopback-choice [1*SP loopback-choice]
         loopback-choice        = loopback-type-pkt / loopback-type-media
         loopback-type-pkt      = "rtp-pkt-loopback"
         loopback-type-media    = "rtp-media-loopback"


- The last two paragraphs in Section 5.1 describe the packet loopback and 
the media loopback. However, in Section 1.1, there are three modes 
defined. I would like to have a matching between the two values and the 
three modes. For example, you could add a sentence indicating in each of 
these two values "this value is used when the loopback is of type xxx 
(see Section 1.X)."

- Section 5.2 actually defines two attributes (loopback-source and 
loopback-mirror). First, I would like to ask if the authors have 
considered to have a single attribute, that you could cal 
"loopback-mode", with two values "source" and "mirror". If this is not 
desirable, then you need to split section 5.2 into two sections, each one 
devoted to a single attribute. Make sure that the name of the attribute 
is listed in the title of the section, so that it is also visible in the 
table of contents.

- If you continue with the two attributes approach, then the syntax for 
one of then would be:

         attribute             /= loopback-src-attr
         ; attribute defined in RFC 4566
         loopback-src-attr      = "loopback-source:" fmt *(SP fmt)
         ; fmt defined in RFC 4566

- Section 5.3, after the example. There is a note that contains a 
normative statement (MUST NOT). In principle, notes should be informative 
by nature, so just remove the word "Note:" from the beginning of the 
paragraph.

- Section 5.3, the paragraph after the Note refers to the "m= line" when 
the draft says "The port number and the address in the offer (m= 
line)..." However, the address in SDP goes in the c= line, therefore, 
whenever you refer to port number and address, you should refer to the 
"m/c= lines".

- Still talking of the same sentence as before. The sentence reads:

     The port number and the address in the offer (m= line) indicate
     where the offerer would like to send and receive the media stream.

I disagree with the "send" in the above sentence. According to RFC 3264, 
the port number and the address in either the offer or the answer 
represents where the offerer or answerer would like to receive the media 
stream, respectively.

Note that if you really want to keep the "send" in the above sentence, 
you need to mandate the usage of symmetric RTP and RTCP, RFC 4961. I 
haven't seen that dependency so far, so I believe that the "send" is 
incorrect in the sentence.

Also note that the resolution of this issue applies to Section 5.4, first 
paragraph as well.

- Examples at the end of Section 5.3 It would be nice to elaborate these 
examples, describing what they are showing. Also, at this point in the 
draft, the "encaprtp" and "rtploopback" tokens have not been introduced, 
so it would be nice to point to the sections where it will later be defined.

- Section 5.4, first paragraph, first sentence, replace "a loopback 
answer" with "an SDP answer". Later in this same paragraph replace 
"offerere" with "offerer".

- There is some missing words in the second sentence of the first 
paragraph in Section 5.4. The sentence is incomplete.

- Section 5.4, second paragraph, write "attributes" in plural in the 
first sentence.

- Section 5.5, first and second sentences, read:

     The answer to a loopback-source MUST be loopback-mirror.  The
     answer to a loopback-mirror MUST be loopback-source.

First, the sentences should be expressed in terms of "SDP offer" and "SDP 
answer" "containing a loopback-source" or "loopback-mirror". For example:

     An offer containing a "loopback-source" attribute must be answered 
with an SDP answer containing a "loopback-mirror attribute in the same 
media description. An offer containing...

But, having said that... I wonder whether the above sentence makes sense 
in this section. This section is titled "Offerer processing of the 
answer". Therefore, I expect to find there what the offerer should do 
when it receives an SDP answer, not what the answerer should do to 
process the offer and generate the answer. So, move these two sentences 
to the Section 5.4.

- The third and fourth sentences in Section 5.5 read:

     The
     loopback-mode line MUST contain at least one codec the answerer is
     willing to send or receive depending on whether it is the loopback-
     source or the loopback-mirror.  In addition, the "m=" line MUST
     contain at least one codec that the answerer is willing to send or
     receive depending on whether it is the loopback-mirror or the
     loopback-source.

Well, it seems that these sentencse also describe how the answerer 
generates the answer, so it should be moved to Section 5.4.

- Second paragraph in Section 5.5. Replace "target UA" with "answerer". 
But most important, this paragraph needs to say what happens if the offer 
receives an answer that does not include the 
loopback-mirror/loopback-source attribute. I believe it should say that a 
session is still established, but loopback packets are not expected. It 
is up to the offerer to close down the session by regular means (high 
level protocol such as a SIP BYE or sending a new offer with the port 
number set to zero).

- Section 5.6, I wonder way the "may" in the first sentence is not a MAY.

- Section 5.6, replace the reference to SIP. You just need the reference 
to Section 8 of RFC 3264.

- Section 7.1.2. The paragraph under "Receive Timestamp: 32 bits". The 
sentence reads:

     The
     Receive timestamp MUST be based on the same clock used by the
     loopback-source.

I wonder if there are mechanisms in place to implement the MUST, i.e., to 
make sure that exactly the same clock is used by both endpoints. Is there 
such mechanism?

The resolution of this issue also applies to Section 7.2.1.

- I don't understand the only normative sentence in Section 7.2 (Direct 
loopback). It reads:

     This payload format MUST NOT be used in cases where
     the MTU on the loopback path will cause fragmentation of looped
     back RTP packets.

So, if direct loopback does not increment the size of the payload, why is 
there a "MUST NOT".

I would understand if the sentence were written in Section 7.1, so that 
it would be applicable to the Encapsulated Payload Format". But I don't 
see how this sentence applies to Direct loopback.

- Section 10. All the examples, replace "client", "server" and "INVITE 
request", and "response" with:

An [offerer|answerer] sends an SDP [offer|answer] which looks like:

- Section 10. In all the examples, replace "s=Example" with "s=" (source, 
RFC 3264). And you can omit the i= and e= lines as well, they do not 
really add anything, nor they are mandated.

- Section 10.3. Re-title this section to "SDP answer rejecting loopback 
media".

- The note at the end of Section 10.3 reads:

     NOTE: Loopback request may be rejected by either not including the
     loopback mode attribute (for backward compatibility) or setting the
     media port number to zero, or both, in the response.

First, I don't like this note here, because it may induce to double 
specification, specially since you have Section 5.4 that describes how to 
reject a loopback request.

And actually, here you have double incongruent specification. Section 
5.4.1 only offers one mechanism for rejecting a session: setting the port 
to zero like in regular SDP. Nothing is written about rejecting a session 
by not including the loopback mode attribute.

Not including a loopback mode attribute in the answer is not a rejection, 
it is merely a session with no support for the loopback feature, but the 
session is established.

I recommend to delete this note.

- Section 11. I disagree with the dependency on RFC 3261. Please remove. 
Also, I am surprised of not seen any dependency of the security 
considerations of RTP/RTCP.


/Miguel






-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From keith.drage@alcatel-lucent.com  Sun Oct 16 14:43:00 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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I think the issue is that you are variously using all sorts of terminology.=
 I see in the document:

	E.164 number
	in E.164 format
	E.164 address

Your main definition is:

   For the address type, we initially consider the possibility of
   describing E.164 telephone numbers.  We define a new "E164" address
   type.  When used, the "E164" address type indicates that the
   connection address contains a telephone number represented according
   to the ITU-T E.164 [ITU.E164.1991] recommendation.

Even if the prime purpose of E.164 is to define international numbers there=
 are multiple types of number defined in E.164, e.g.=20

	4.10	global subscriber number (GSN)
	4.18	national (significant) number [N(S)N]
	4.24	subscriber number (SN)

The definition you are looking for is:

4.16	international public telecommunication number
	F:	num=E9ro de t=E9l=E9communications publiques internationales
	S:	n=FAmero de telecomunicaciones p=FAblicas internacionales
A string of decimal digits that, for a geographic country code, uniquely id=
entifies a subscriber or a point where a service is provided. For the case =
of a global service code, it identifies the subscriber of the service. For =
Networks, it identifies a subscriber of the Network.
An international public telecommunication number (hereafter referred to as =
"international E.164 numbers") can act in the "role" of both a name and an =
address. Portability is reducing a number's role as an address. Numbers are=
 increasingly acting in the role of a name only.
The number, which includes the country code and subsequent digits, but not =
the international prefix, contains the information necessary to route the c=
all to this termination point on a public network (it may also contain the =
supplementary information necessary to forward it on a private network). It=
 is sometimes referred to as an "E.164 number" or "international number".

So my view is that your main definition can be construed as referring to an=
y of the numbers defined in E.164 which I have listed. The term "telephone =
number" refers to both national and international numbers.=20

For the remainder of the document, while E.164 number is permissible as an =
indication of your required term, I think you would be better using one of =
the preferred terms either "international E.164 numbers" or "international =
public telecommunication number". Whichever you use, use it consistently th=
roughout the document.

If you are specifically indicating the address type, I suggest you consiste=
ntly put the E.164 in quotation marks, followed by the words "address type"=
 in full.

The "+" is actually part of ITU-T Recommendation E.123: "Notation for natio=
nal and international telephone numbers, e-mail addresses and Web addresses=
". However you cannot use E.123 as a blanket reference as it also includes =
space characters and parentheses, unless you wish these to also be valid.

Regards

Keith

> -----Original Message-----
> From: Miguel A. Garcia [mailto:Miguel.A.Garcia@ericsson.com]
> Sent: 16 October 2011 09:25
> To: DRAGE, Keith (Keith)
> Cc: Paul Kyzivat; mmusic@ietf.org
> Subject: Re: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
>=20
> Hi Keith.
>=20
> I am a bit confused with your comment indicated that the reference should
> be limited to international numbers defined by E.164. I think E.164 is
> all about international numbers. As a matter of fact, the title of E.164
> is "The international public telecommunication numbering plan", and the
> scope clearly indicates that it is about international numbers.
>=20
> However, I think we need to elaborate a bit more the '+' sign, which is
> not part of the E.164 specification, but a common trick to represent a
> full international E.164 number.
>=20
> /Miguel
>=20
> On 15/10/2011 14:52, DRAGE, (Keith) wrote:
> > Paul wrote
> >
> >> 5.2.1
> >>
> >> This says addresses for the E164 address type are represented
> >> according to ITU-T E.164. If I have followed the reference
> >> correctly, it only specifies the country codes, not the encoding of
> >> numbers including country codes. I think it would be better to
> >> define this by reference to RFC 3966 (TEL) and to either
> >> telephone-subscriber, or global-number, or global-number-digits
> >> defined there. (I'm not sure which of those is most appropriate for
> >> your use.)
> >>
> >
> > E.164 defines the country codes in an annex. The main body of the
> > recommendation defines the structure of a number consisting of country
> > code and national significant number for international number. And
> > national and subscriber prefix and appropriate part of the number for
> > national number and subscriber number usage. The definition also
> > includes various length limitations, and identifies local service
> > numbers.
> >
> > However I suspect that the reference here needs to explicitly limit it
> > to international numbers defined by E.164.
> >
> > I see no need to go via RFC 3966 which becomes only significant in
> > terms of referencing if you want to define local numbers.
> >
> > Regards
> >
> > Keith
> >
> >> -----Original Message----- From: mmusic-bounces@ietf.org
> >> [mailto:mmusic-bounces@ietf.org] On Behalf Of Paul Kyzivat Sent: 14
> >> October 2011 19:47 To: mmusic@ietf.org Subject: Re: [MMUSIC] FW: I-D
> >> Action: draft-ietf-mmusic-sdp-cs-08.txt
> >>
> >> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of
> >> comments:
> >>
> >> 5.2.1
> >>
> >> This says addresses for the E164 address type are represented
> >> according to ITU-T E.164. If I have followed the reference
> >> correctly, it only specifies the country codes, not the encoding of
> >> numbers including country codes. I think it would be better to
> >> define this by reference to RFC 3966 (TEL) and to either
> >> telephone-subscriber, or global-number, or global-number-digits
> >> defined there. (I'm not sure which of those is most appropriate for
> >> your use.)
> >>
> >> Also, you give the example
> >>
> >> c=3DPSTN - -
> >>
> >> That requires that "-" be a defined address type. AFAIK it is not.
> >> Did you mean it to be
> >>
> >> c=3DPSTN E164 -
> >>
> >> 5.2.3.1
> >>
> >> The syntax given for a=3Dcs-correlation is inconsistent with the
> >> syntax in 5.6 and with usage throughout the document.
> >>
> >> 5.2.3.3
> >>
> >> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK
> >> the actual max information content is actually either 32 or 128
> >> octets plus a leading protocol discriminator byte.
> >>
> >> Its unclear what you intend to be included in the specified uuie
> >> element in the sdp - does it include the protocol discriminator or
> >> not?
> >>
> >> Also, the content and protocol discriminator contain octets, not
> >> just alphanumerics, but the syntax in 5.6 only permits
> >> alphanumerics. Is it your *intent* to limit the protocol
> >> discriminator and the value to that subset described by
> >> alphanumerics? And is that a reasonable thing to do? (For this use,
> >> it *might* be sufficient to prescribe a specific value for the
> >> protocol discriminator.)
> >>
> >> 5.5.1 / 5.5.4
> >>
> >> The description of a=3Dconnection here is more appropriate to section
> >> 5.5.4. Here, for the initial offer, a=3Dconnection:new is the only
> >> thing that makes sense.
> >>
> >> In 5.5.4 it would be helpful to explain in more detail how
> >> a=3Dconnection is used to control whether an existing call is
> >> retained, dropped, or dropped and reestablished.
> >>
> >> (In some cases it will be desired to do another o/a while leaving
> >> the pstn call undisturbed. In other cases it may the explicit intent
> >> to cause the call to be reestablished.)
> >>
> >> 5.6
> >>
> >> Is there a reason why you don't use the definition of an E.164
> >> number from RFC 3966, rather than using your own? What is the
> >> significance of making the "+" optional? And why is caller-id-value
> >> defined differently than e164-address?
> >>
> >> Figure 4, and others:
> >>
> >> The E.164 number used (+15551234) isn't a valid one for country code
> >> 1.
> >>
> >> General Comment:
> >>
> >> Everything is described, and examples show, only a single m=3D line,
> >> and all the cs-correlation and setup things used within that one
> >> media section. There is no explanation of what to do if there is
> >> more than one pstn m=3D line, as might arise if the intent is to have
> >> an audio/video call. Without any explanation, it would seem that
> >> everything should be specified separately for each m=3D line, which
> >> might result in *two* pstn calls. Presumably that isn't want is
> >> desired.
> >>
> >> More definition is needed if the same pstn call is to be used for
> >> two media streams.
> >>
> >> Thanks, Paul
> >>
> >> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
> >>> Hello all,
> >>>
> >>> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
> >>>
> >>> Changes in the -08 version include:
> >>>
> >>> Added a new section "Offer/Answer extensions" detailing the
> >>> process of
> >> handling the O/A exchange.
> >>>
> >>> Clarified that multiple extensions for cs-correlation are
> >>> allowed.
> >>>
> >>> Added a request for IANA to create a sub-registry for
> >>> "cs-correlation"
> >> attribute values, with "Specification Required" policy.
> >>>
> >>> Clarified how dynamic payloads are handled. Basically we stick to
> >>> normal
> >> SDP processing here, i.e. if dynamic payload types are used,
> >> a=3Drtpmap attribute(s) are required. This may need more thought
> >> still.
> >>>
> >>> Changed a lot of SHOULDs into MUSTs.
> >>>
> >>> Extended the example in Section 6.
> >>>
> >>> Flemming had a comment on mandating E.164 numbers in the
> >>> international
> >> format. After some thinking, we still say that the enpoints SHOULD
> >> include the number in international format. The network may anyway
> >> mangle the number presentation, so we thought mandating the intl
> >> format is not absolutely necessary.
> >>>
> >>> TODO: Security considerations needs more work based on Flemming's
> >> earlier comments (including caller's E164 number in SDP).
> >>> TODO: Clarification on the mechanism works with 3PCC. TODO:Change
> >>> the UUIE syntax to allow for a discriminator byte.
> >>>
> >>> We hope to address the remaning issues still before the IETF
> >>> submission
> >> deadline.
> >>>
> >>> Regards, Simo
> >>>
> >>>
> >>> -----Original Message----- From: mmusic-bounces@ietf.org
> >>> [mailto:mmusic-bounces@ietf.org] On Behalf
> >> Of ext internet-drafts@ietf.org
> >>> Sent: 13 October, 2011 15:19 To: i-d-announce@ietf.org Cc:
> >>> mmusic@ietf.org Subject: [MMUSIC] I-D Action:
> >>> draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched
> >> Bearers In The Public Switched Telephone Network (PSTN)
> >>> Author(s)       : Miguel A. Garcia-Martin Simo Veikkolainen
> >>> Filename        : draft-ietf-mmusic-sdp-cs-08.txt Pages
> >>> : 30 Date            : 2011-10-13
> >>>
> >>> 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).
> >>>
> >>>
> >>> A URL for this Internet-Draft is:
> >>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
> >>>
> >>>
> >>>
> Internet-Drafts are also available by anonymous FTP at:
> >>> ftp://ftp.ietf.org/internet-drafts/
> >>>
> >>> This Internet-Draft can be retrieved at:
> >>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
> >>>
> >>>
> _______________________________________________
> >>> mmusic mailing list mmusic@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/mmusic
> >>> _______________________________________________ mmusic mailing
> >>> list mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
> >>>
> >>
> >> _______________________________________________ mmusic mailing list
> >> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
> > _______________________________________________ mmusic mailing list
> > mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> --
> Miguel A. Garcia
> +34-91-339-3608
> Ericsson Spain

From miguel.a.garcia@ericsson.com  Sun Oct 16 23:56:02 2011
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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Very good comments Keith, thanks.

We will have to do a pass throughout the document trying to make it 
consistent. I think we should be using the term "international E.164 
number". And I believe we shouldn't make a reference to E.123, but 
explain the usage of the plus sign.

BR,

    Miguel

On 16/10/2011 23:42, DRAGE, Keith (Keith) wrote:
> I think the issue is that you are variously using all sorts of terminology. I see in the document:
>
>          E.164 number
>          in E.164 format
>          E.164 address
>
> Your main definition is:
>
>     For the address type, we initially consider the possibility of
>     describing E.164 telephone numbers.  We define a new "E164" address
>     type.  When used, the "E164" address type indicates that the
>     connection address contains a telephone number represented according
>     to the ITU-T E.164 [ITU.E164.1991] recommendation.
>
> Even if the prime purpose of E.164 is to define international numbers there are multiple types of number defined in E.164, e.g.
>
>          4.10    global subscriber number (GSN)
>          4.18    national (significant) number [N(S)N]
>          4.24    subscriber number (SN)
>
> The definition you are looking for is:
>
> 4.16    international public telecommunication number
>          F:      numéro de télécommunications publiques internationales
>          S:      número de telecomunicaciones públicas internacionales
> A string of decimal digits that, for a geographic country code, uniquely identifies a subscriber or a point where a service is provided. For the case of a global service code, it identifies the subscriber of the service. For Networks, it identifies a subscriber of the Network.
> An international public telecommunication number (hereafter referred to as "international E.164 numbers") can act in the "role" of both a name and an address. Portability is reducing a number's role as an address. Numbers are increasingly acting in the role of a name only.
> The number, which includes the country code and subsequent digits, but not the international prefix, contains the information necessary to route the call to this termination point on a public network (it may also contain the supplementary information necessary to forward it on a private network). It is sometimes referred to as an "E.164 number" or "international number".
>
> So my view is that your main definition can be construed as referring to any of the numbers defined in E.164 which I have listed. The term "telephone number" refers to both national and international numbers.
>
> For the remainder of the document, while E.164 number is permissible as an indication of your required term, I think you would be better using one of the preferred terms either "international E.164 numbers" or "international public telecommunication number". Whichever you use, use it consistently throughout the document.
>
> If you are specifically indicating the address type, I suggest you consistently put the E.164 in quotation marks, followed by the words "address type" in full.
>
> The "+" is actually part of ITU-T Recommendation E.123: "Notation for national and international telephone numbers, e-mail addresses and Web addresses". However you cannot use E.123 as a blanket reference as it also includes space characters and parentheses, unless you wish these to also be valid.
>
> Regards
>
> Keith
>
>> -----Original Message-----
>> From: Miguel A. Garcia [mailto:Miguel.A.Garcia@ericsson.com]
>> Sent: 16 October 2011 09:25
>> To: DRAGE, Keith (Keith)
>> Cc: Paul Kyzivat; mmusic@ietf.org
>> Subject: Re: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
>>
>> Hi Keith.
>>
>> I am a bit confused with your comment indicated that the reference should
>> be limited to international numbers defined by E.164. I think E.164 is
>> all about international numbers. As a matter of fact, the title of E.164
>> is "The international public telecommunication numbering plan", and the
>> scope clearly indicates that it is about international numbers.
>>
>> However, I think we need to elaborate a bit more the '+' sign, which is
>> not part of the E.164 specification, but a common trick to represent a
>> full international E.164 number.
>>
>> /Miguel
>>
>> On 15/10/2011 14:52, DRAGE, (Keith) wrote:
>>> Paul wrote
>>>
>>>> 5.2.1
>>>>
>>>> This says addresses for the E164 address type are represented
>>>> according to ITU-T E.164. If I have followed the reference
>>>> correctly, it only specifies the country codes, not the encoding of
>>>> numbers including country codes. I think it would be better to
>>>> define this by reference to RFC 3966 (TEL) and to either
>>>> telephone-subscriber, or global-number, or global-number-digits
>>>> defined there. (I'm not sure which of those is most appropriate for
>>>> your use.)
>>>>
>>>
>>> E.164 defines the country codes in an annex. The main body of the
>>> recommendation defines the structure of a number consisting of country
>>> code and national significant number for international number. And
>>> national and subscriber prefix and appropriate part of the number for
>>> national number and subscriber number usage. The definition also
>>> includes various length limitations, and identifies local service
>>> numbers.
>>>
>>> However I suspect that the reference here needs to explicitly limit it
>>> to international numbers defined by E.164.
>>>
>>> I see no need to go via RFC 3966 which becomes only significant in
>>> terms of referencing if you want to define local numbers.
>>>
>>> Regards
>>>
>>> Keith
>>>
>>>> -----Original Message----- From: mmusic-bounces@ietf.org
>>>> [mailto:mmusic-bounces@ietf.org] On Behalf Of Paul Kyzivat Sent: 14
>>>> October 2011 19:47 To: mmusic@ietf.org Subject: Re: [MMUSIC] FW: I-D
>>>> Action: draft-ietf-mmusic-sdp-cs-08.txt
>>>>
>>>> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of
>>>> comments:
>>>>
>>>> 5.2.1
>>>>
>>>> This says addresses for the E164 address type are represented
>>>> according to ITU-T E.164. If I have followed the reference
>>>> correctly, it only specifies the country codes, not the encoding of
>>>> numbers including country codes. I think it would be better to
>>>> define this by reference to RFC 3966 (TEL) and to either
>>>> telephone-subscriber, or global-number, or global-number-digits
>>>> defined there. (I'm not sure which of those is most appropriate for
>>>> your use.)
>>>>
>>>> Also, you give the example
>>>>
>>>> c=PSTN - -
>>>>
>>>> That requires that "-" be a defined address type. AFAIK it is not.
>>>> Did you mean it to be
>>>>
>>>> c=PSTN E164 -
>>>>
>>>> 5.2.3.1
>>>>
>>>> The syntax given for a=cs-correlation is inconsistent with the
>>>> syntax in 5.6 and with usage throughout the document.
>>>>
>>>> 5.2.3.3
>>>>
>>>> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK
>>>> the actual max information content is actually either 32 or 128
>>>> octets plus a leading protocol discriminator byte.
>>>>
>>>> Its unclear what you intend to be included in the specified uuie
>>>> element in the sdp - does it include the protocol discriminator or
>>>> not?
>>>>
>>>> Also, the content and protocol discriminator contain octets, not
>>>> just alphanumerics, but the syntax in 5.6 only permits
>>>> alphanumerics. Is it your *intent* to limit the protocol
>>>> discriminator and the value to that subset described by
>>>> alphanumerics? And is that a reasonable thing to do? (For this use,
>>>> it *might* be sufficient to prescribe a specific value for the
>>>> protocol discriminator.)
>>>>
>>>> 5.5.1 / 5.5.4
>>>>
>>>> The description of a=connection here is more appropriate to section
>>>> 5.5.4. Here, for the initial offer, a=connection:new is the only
>>>> thing that makes sense.
>>>>
>>>> In 5.5.4 it would be helpful to explain in more detail how
>>>> a=connection is used to control whether an existing call is
>>>> retained, dropped, or dropped and reestablished.
>>>>
>>>> (In some cases it will be desired to do another o/a while leaving
>>>> the pstn call undisturbed. In other cases it may the explicit intent
>>>> to cause the call to be reestablished.)
>>>>
>>>> 5.6
>>>>
>>>> Is there a reason why you don't use the definition of an E.164
>>>> number from RFC 3966, rather than using your own? What is the
>>>> significance of making the "+" optional? And why is caller-id-value
>>>> defined differently than e164-address?
>>>>
>>>> Figure 4, and others:
>>>>
>>>> The E.164 number used (+15551234) isn't a valid one for country code
>>>> 1.
>>>>
>>>> General Comment:
>>>>
>>>> Everything is described, and examples show, only a single m= line,
>>>> and all the cs-correlation and setup things used within that one
>>>> media section. There is no explanation of what to do if there is
>>>> more than one pstn m= line, as might arise if the intent is to have
>>>> an audio/video call. Without any explanation, it would seem that
>>>> everything should be specified separately for each m= line, which
>>>> might result in *two* pstn calls. Presumably that isn't want is
>>>> desired.
>>>>
>>>> More definition is needed if the same pstn call is to be used for
>>>> two media streams.
>>>>
>>>> Thanks, Paul
>>>>
>>>> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
>>>>> Hello all,
>>>>>
>>>>> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
>>>>>
>>>>> Changes in the -08 version include:
>>>>>
>>>>> Added a new section "Offer/Answer extensions" detailing the
>>>>> process of
>>>> handling the O/A exchange.
>>>>>
>>>>> Clarified that multiple extensions for cs-correlation are
>>>>> allowed.
>>>>>
>>>>> Added a request for IANA to create a sub-registry for
>>>>> "cs-correlation"
>>>> attribute values, with "Specification Required" policy.
>>>>>
>>>>> Clarified how dynamic payloads are handled. Basically we stick to
>>>>> normal
>>>> SDP processing here, i.e. if dynamic payload types are used,
>>>> a=rtpmap attribute(s) are required. This may need more thought
>>>> still.
>>>>>
>>>>> Changed a lot of SHOULDs into MUSTs.
>>>>>
>>>>> Extended the example in Section 6.
>>>>>
>>>>> Flemming had a comment on mandating E.164 numbers in the
>>>>> international
>>>> format. After some thinking, we still say that the enpoints SHOULD
>>>> include the number in international format. The network may anyway
>>>> mangle the number presentation, so we thought mandating the intl
>>>> format is not absolutely necessary.
>>>>>
>>>>> TODO: Security considerations needs more work based on Flemming's
>>>> earlier comments (including caller's E164 number in SDP).
>>>>> TODO: Clarification on the mechanism works with 3PCC. TODO:Change
>>>>> the UUIE syntax to allow for a discriminator byte.
>>>>>
>>>>> We hope to address the remaning issues still before the IETF
>>>>> submission
>>>> deadline.
>>>>>
>>>>> Regards, Simo
>>>>>
>>>>>
>>>>> -----Original Message----- From: mmusic-bounces@ietf.org
>>>>> [mailto:mmusic-bounces@ietf.org] On Behalf
>>>> Of ext internet-drafts@ietf.org
>>>>> Sent: 13 October, 2011 15:19 To: i-d-announce@ietf.org Cc:
>>>>> mmusic@ietf.org Subject: [MMUSIC] I-D Action:
>>>>> draft-ietf-mmusic-sdp-cs-08.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 Up Audio and Video Media Streams Over Circuit-Switched
>>>> Bearers In The Public Switched Telephone Network (PSTN)
>>>>> Author(s)       : Miguel A. Garcia-Martin Simo Veikkolainen
>>>>> Filename        : draft-ietf-mmusic-sdp-cs-08.txt Pages
>>>>> : 30 Date            : 2011-10-13
>>>>>
>>>>> 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).
>>>>>
>>>>>
>>>>> A URL for this Internet-Draft is:
>>>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>>>>
>>>>>
>>>>>
>> Internet-Drafts are also available by anonymous FTP at:
>>>>> ftp://ftp.ietf.org/internet-drafts/
>>>>>
>>>>> This Internet-Draft can be retrieved at:
>>>>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>>>>
>>>>>
>> _______________________________________________
>>>>> mmusic mailing list mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>> _______________________________________________ mmusic mailing
>>>>> list mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>>>>
>>>>
>>>> _______________________________________________ mmusic mailing list
>>>> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>> _______________________________________________ mmusic mailing list
>>> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> --
>> Miguel A. Garcia
>> +34-91-339-3608
>> Ericsson Spain

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

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Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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Miguel - comments inline.

	Thanks,
	Paul

On 10/16/11 6:16 AM, Miguel A. Garcia wrote:
> Hi Paul.
>
> You can find my comments inline, except for one that we are discussing
> with Keith in a separate e-mail.
>
> On 14/10/2011 20:46, Paul Kyzivat wrote:
>> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of comments:
>>
>> 5.2.1
>>
>> This says addresses for the E164 address type are represented according
>> to ITU-T E.164. If I have followed the reference correctly, it only
>> specifies the country codes, not the encoding of numbers including
>> country codes. I think it would be better to define this by reference to
>> RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or
>> global-number-digits defined there. (I'm not sure which of those is most
>> appropriate for your use.)
>>
>> Also, you give the example
>>
>> c=PSTN - -
>>
>> That requires that "-" be a defined address type. AFAIK it is not.
>> Did you mean it to be
>>
>> c=PSTN E164 -
>
>
> No, we meant:
>
> c=PSTN - -
>
> The idea is frequently a device will know that is Circuit-Switched
> capable, but it wouldn't know its own E.164 number, or even it is
> allocated an E.164 number. It could be possible that the device is aware
> of a different numbering plan (for example, E.212).
>
> So, in summary: we want to express that the device has a PSTN
> Circuit-Switched connection, but it is not aware of its address in any
> numbering plan.
>
> In Section 8.3, we register with IANA both the E.164 and the dash "-"
> tokens.

OK, I see its registered. And I found it mentioned in passing.
But I don't find a specification for what the connection-address 
may/must be when the addrtype is "-".

In 4566 there is:

    When the <addrtype> is IP4 and IP6, the connection address is defined
    as follows:

    o  ...

It would be helpful if this doc had a similar thing:

    When the <addrtype> is PSTN, the connection address is defined
    as follows:

    o  an e164-address

    When the <addrtype> is "-", the connection address is defined
    as follows:

    o  the value "-", signifying that the address is unknown.
OR
    o  any syntactically valid value, which is to be ignored..

>> 5.2.3.1
>>
>> The syntax given for a=cs-correlation is inconsistent with the syntax in
>> 5.6 and with usage throughout the document.
>
> I am comparing Sections 5.2.3.1 and 5.6, and the usage throughout the
> document and I don't find this inconsistency. Can you elaborate a bit
> more what this inconsistency is?

5.2.3.1 has:

    a=cs-correlation: callerid:<callerid-value> |
                      uuie:<uuie-value>           |
                      dtmf:<dtmf-value>           |
                      *[extension-name:<extension-value>]

This requires callerid, uuid, and dtmf to be mutually exclusive, rather 
than permitting one of each as 5.6 allows. Also, the ":" and value are 
required rather than being optional.

>> 5.2.3.3
>>
>> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the
>> actual max information content is actually either 32 or 128 octets plus
>> a leading protocol discriminator byte.
>
> Right. The terminology in Q.931 helps to mislead. The User-User
> Information Element contains three parts:
>
> - 1-octet User-user information element identifier
> - 1-octet "Length of user-user contents"
> - 1-octet "Protocol discriminator"
> - "User Information".
>
> The "User Information can be up to 32 or 128 octets, depending on
> networks. Together with the other 3 octets, it makes a maximum of 35 or
> 131 octets.
>
> I guess the draft should elaborate the composition of this information
> element and clarify that it includes the 3 initial octets.

Based on the examples, and some common sense, I think you don't want to 
specify the first two of the above in the SDP. AFAIK the first has a 
well defined value, and the second is well defined by counting the bytes 
in the uuie-value.

>> Its unclear what you intend to be included in the specified uuie element
>> in the sdp - does it include the protocol discriminator or not?
>
> Yes, as I described above.
>
>>
>> Also, the content and protocol discriminator contain octets, not just
>> alphanumerics, but the syntax in 5.6 only permits alphanumerics. Is it
>> your *intent* to limit the protocol discriminator and the value to that
>> subset described by alphanumerics? And is that a reasonable thing to do?
>> (For this use, it *might* be sufficient to prescribe a specific value
>> for the protocol discriminator.)
>
> This is an error that we should fix. It is not the intention of this
> draft to limit the protocol discriminator, so, we need to address this
> issue and allow any possible value. We will fix it in the next version
> of the draft.
>
>
>
>>
>> 5.5.1 / 5.5.4
>>
>> The description of a=connection here is more appropriate to section
>> 5.5.4. Here, for the initial offer, a=connection:new is the only thing
>> that makes sense.
>
> Ok, we can move the a=connection:existing to 5.5.4.
>
>>
>> In 5.5.4 it would be helpful to explain in more detail how a=connection
>> is used to control whether an existing call is retained, dropped, or
>> dropped and reestablished.
>  >
>>
>> (In some cases it will be desired to do another o/a while leaving the
>> pstn call undisturbed. In other cases it may the explicit intent to
>> cause the call to be reestablished.)
>
> ok, will do.
>
>
>>
>> 5.6
>>
>> Is there a reason why you don't use the definition of an E.164 number
>> from RFC 3966, rather than using your own?
>
> RFC 3966 defines how to represent an E.164 number as a URL. Here, we do
> not intent to use URLs. Additionally, the tel: URL provide
> representation of either global and local numbers, ISDN subaddresses,
> extensions, and phone context, which are not intended to be used here.

I meant not the entire tel syntax, but by reference to one of the ABNF 
symbols defined in 3966. That has the benefit of being more accessible 
to those reading ietf drafts, and to give the syntax in ABNF that can be 
readily incorporated with the rest of the syntax.

Of course that would allowing visual-separator. I don't know if that is 
a big deal or not.

>  > What is the significance of
>> making the "+" optional?
>
> The only reason I can think, is that the draft has a "SHOULD" strength
> for using the international format of E.164. If someone really knows
> what he is doing and can use a national format, it wouldn't use the "+"
> sign. I recognize that this is not a very strong reason, but it is the
> only justification I can find.

Well, if the intent is to allow a national format, then it should be 
made clear that lack of the "+" means that. (And then how on how one is 
supposed to know *which* national format.)

I realize there is text that says the actual callerid received may not 
be in E.164 format and so the comparison should take that into account. 
But I didn't think that applied to the number specified in SDP.

>  > And why is caller-id-value defined differently
>> than e164-address?
>
> Yes, I think they should be equal. Probably we should get rid of
> "caller-id-value" and use "e164-address" instead.
>
>
>
>> Figure 4, and others:
>>
>> The E.164 number used (+15551234) isn't a valid one for country code 1.
>
> Is it because there are not enough number of digits for the subscriber
> number?

Right. Its just a nit. There is no area code in it.

>> General Comment:
>>
>> Everything is described, and examples show, only a single m= line, and
>> all the cs-correlation and setup things used within that one media
>> section. There is no explanation of what to do if there is more than one
>> pstn m= line, as might arise if the intent is to have an audio/video
>> call. Without any explanation, it would seem that everything should be
>> specified separately for each m= line, which might result in *two* pstn
>> calls. Presumably that isn't want is desired.
>>
>> More definition is needed if the same pstn call is to be used for two
>> media streams.
>
> Yes, I agree with you. I think we should add a section describing how to
> setup an circuit-switched audio/video call.
>
>>
>> Thanks,
>> Paul
>
> Thanks for your comments,
>
> Miguel
>
>
>>
>> On 10/13/11 8:21 AM, Simo.Veikkolainen@nokia.com wrote:
>>> Hello all,
>>>
>>> We just submitted a new version of draft-ietf-mmusic-sdp-cs.
>>>
>>> Changes in the -08 version include:
>>>
>>> Added a new section "Offer/Answer extensions" detailing the process
>>> of handling the O/A exchange.
>>>
>>> Clarified that multiple extensions for cs-correlation are allowed.
>>>
>>> Added a request for IANA to create a sub-registry for
>>> "cs-correlation" attribute values, with "Specification Required" policy.
>>>
>>> Clarified how dynamic payloads are handled. Basically we stick to
>>> normal SDP processing here, i.e. if dynamic payload types are used,
>>> a=rtpmap attribute(s) are required. This may need more thought still.
>>>
>>> Changed a lot of SHOULDs into MUSTs.
>>>
>>> Extended the example in Section 6.
>>>
>>> Flemming had a comment on mandating E.164 numbers in the
>>> international format. After some thinking, we still say that the
>>> enpoints SHOULD include the number in international format. The
>>> network may anyway mangle the number presentation, so we thought
>>> mandating the intl format is not absolutely necessary.
>>>
>>> TODO: Security considerations needs more work based on Flemming's
>>> earlier comments (including caller's E164 number in SDP).
>>> TODO: Clarification on the mechanism works with 3PCC.
>>> TODO:Change the UUIE syntax to allow for a discriminator byte.
>>>
>>> We hope to address the remaning issues still before the IETF
>>> submission deadline.
>>>
>>> Regards,
>>> Simo
>>>
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of ext internet-drafts@ietf.org
>>> Sent: 13 October, 2011 15:19
>>> To: i-d-announce@ietf.org
>>> Cc: mmusic@ietf.org
>>> Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-cs-08.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 Up
>>> Audio and Video Media Streams Over Circuit-Switched Bearers In The
>>> Public Switched Telephone Network (PSTN)
>>> Author(s) : Miguel A. Garcia-Martin
>>> Simo Veikkolainen
>>> Filename : draft-ietf-mmusic-sdp-cs-08.txt
>>> Pages : 30
>>> Date : 2011-10-13
>>>
>>> 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).
>>>
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>>
>>> Internet-Drafts are also available by anonymous FTP at:
>>> ftp://ftp.ietf.org/internet-drafts/
>>>
>>> This Internet-Draft can be retrieved at:
>>> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-08.txt
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>


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Date: Mon, 17 Oct 2011 21:58:13 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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Cc: mmusic <mmusic@ietf.org>, "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On 10/12/11 10:59 AM, Hadriel Kaplan wrote:
> On Oct 8, 2011, at 8:54 AM, Flemming Andreasen wrote:
>
>    
>> The issue was discussed on the mailing list in August, where the authors suggested it as a path forward and the chairs asked the WG for feedback on this approach, see e.g.
>>
>> http://www.ietf.org/mail-archive/web/mmusic/current/msg08857.html
>>
>> The issue was also discussed in the last IETF MMUSIC meeting (see http://www.ietf.org/proceedings/81/minutes/mmusic.txt)
>>
>> To date, we have only seen opposition to this proposal from you (on both the mailing list and in the meeting), and we are lacking an alternative path forward that can satisfy all involved parties, including the general AD and WG chair concerns around definining an alternative NAT traversal mechanism in this draft without proper discussion and general considerations around the need for an alternative to ICE (per the above thread).
>>      
> In the email thread you referenced above there was in fact someone else who was for keeping the priming mechanism, and it wasn't me.
>
> In fact, in that whole thread there was only one other person than yourself who was in favor of only having ICE in the draft.  Of course there was only one other person than myself who was in favor of keeping the primer mechanism.
>
> My point is that there has been no consensus either way.  The authors only changed the draft due to your objections, afaict, not due to some large consensus to do so.
>    

To be clear, the author(s), myself, and our responsible AD did send a 
note to the list suggesting this as a path forward. I asked the WG 
members to speak up if they disagreed with the suggested path forward, 
and you are the only person to argue against subsequently. While I'll 
agree there is not unanimous agreement in the group with the direction, 
I think we need to hear from more than 1 person to declare a lack of 
(rough) concensus. Having said that, it would be better if we could 
reach an agreement that is satisfactory to all.


>> If you do not find this path forward acceptable, can you please suggest an alternative that satisfies the general concerns around defining an alternative NAT traversal mechanism in this draft ?
>>      
> There have been no "general concerns" as far as I can tell, other than "we have ICE defined so why not use it".
Yes - we have a Standards Track and general solution available, so we 
should use it.

>    My concerns with mandating *only* ICE as opposed to having a SHOULD use ICE but MAY use primer are detailed here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg08611.html
>
> There was no feedback/response to that, afaict.
>    
You are referencing a ~6 month old e-mail, and there has since then been 
several more e-mail exchanges on this subject, an explanation of the 
issues with the old approach, and a thread plus WG meeting discussion on 
the need for an ICE alternative (like latching), which didn't garner any 
takers so far. If you believe those arguments still stand and offer a 
viable path forward, then please resurrect them (either in the context 
of this e-mail thread or as a new thread), and we (and hopefully other 
members of the WG) can discuss further. Again, to recap the concerns 
with the old approach:
- We have a general and Standards Track NAT traversal solution in the 
form of ICE so we should use it
- If there is a need for an ICE alternative, then we need to discuss 
that more generally and come up with a proper solution
- If latching is indeed needed as an alternative solution to ICE, then 
it needs to be properly documented and the limitations of it spelled out 
clearly. Furthermore, the loopback draft does not seem like the right 
place for this.

Thanks

-- Flemming


> -hadriel
>
>
>    

From miguel.a.garcia@ericsson.com  Tue Oct 18 13:03:35 2011
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Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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Hi Paul, see inline comments.

On 18/10/2011 0:27, Paul Kyzivat wrote:
> Miguel - comments inline.
>
> 	Thanks,
> 	Paul
>
> On 10/16/11 6:16 AM, Miguel A. Garcia wrote:
>> Hi Paul.
>>
>> You can find my comments inline, except for one that we are discussing
>> with Keith in a separate e-mail.
>>
>> On 14/10/2011 20:46, Paul Kyzivat wrote:
>>> I reviewed draft-ietf-mmusic-sdp-cs-08 and have a number of comments:
>>>
>>> 5.2.1
>>>
>>> This says addresses for the E164 address type are represented according
>>> to ITU-T E.164. If I have followed the reference correctly, it only
>>> specifies the country codes, not the encoding of numbers including
>>> country codes. I think it would be better to define this by reference to
>>> RFC 3966 (TEL) and to either telephone-subscriber, or global-number, or
>>> global-number-digits defined there. (I'm not sure which of those is most
>>> appropriate for your use.)
>>>
>>> Also, you give the example
>>>
>>> c=PSTN - -
>>>
>>> That requires that "-" be a defined address type. AFAIK it is not.
>>> Did you mean it to be
>>>
>>> c=PSTN E164 -
>>
>>
>> No, we meant:
>>
>> c=PSTN - -
>>
>> The idea is frequently a device will know that is Circuit-Switched
>> capable, but it wouldn't know its own E.164 number, or even it is
>> allocated an E.164 number. It could be possible that the device is aware
>> of a different numbering plan (for example, E.212).
>>
>> So, in summary: we want to express that the device has a PSTN
>> Circuit-Switched connection, but it is not aware of its address in any
>> numbering plan.
>>
>> In Section 8.3, we register with IANA both the E.164 and the dash "-"
>> tokens.
>
> OK, I see its registered. And I found it mentioned in passing.
> But I don't find a specification for what the connection-address
> may/must be when the addrtype is "-".
>
> In 4566 there is:
>
>      When the<addrtype>  is IP4 and IP6, the connection address is defined
>      as follows:
>
>      o  ...
>
> It would be helpful if this doc had a similar thing:
>
>      When the<addrtype>  is PSTN, the connection address is defined
>      as follows:
>
>      o  an e164-address
>
>      When the<addrtype>  is "-", the connection address is defined
>      as follows:
>
>      o  the value "-", signifying that the address is unknown.
> OR
>      o  any syntactically valid value, which is to be ignored..


Good point. We will add it to the draft.


>
>>> 5.2.3.1
>>>
>>> The syntax given for a=cs-correlation is inconsistent with the syntax in
>>> 5.6 and with usage throughout the document.
>>
>> I am comparing Sections 5.2.3.1 and 5.6, and the usage throughout the
>> document and I don't find this inconsistency. Can you elaborate a bit
>> more what this inconsistency is?
>
> 5.2.3.1 has:
>
>      a=cs-correlation: callerid:<callerid-value>  |
>                        uuie:<uuie-value>            |
>                        dtmf:<dtmf-value>            |
>                        *[extension-name:<extension-value>]
>
> This requires callerid, uuid, and dtmf to be mutually exclusive, rather
> than permitting one of each as 5.6 allows. Also, the ":" and value are
> required rather than being optional.


Aha, I see it know. The correct one is in 5.6, so we will change 5.2.3.1.

>
>>> 5.2.3.3
>>>
>>> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the
>>> actual max information content is actually either 32 or 128 octets plus
>>> a leading protocol discriminator byte.
>>
>> Right. The terminology in Q.931 helps to mislead. The User-User
>> Information Element contains three parts:
>>
>> - 1-octet User-user information element identifier
>> - 1-octet "Length of user-user contents"
>> - 1-octet "Protocol discriminator"
>> - "User Information".
>>
>> The "User Information can be up to 32 or 128 octets, depending on
>> networks. Together with the other 3 octets, it makes a maximum of 35 or
>> 131 octets.
>>
>> I guess the draft should elaborate the composition of this information
>> element and clarify that it includes the 3 initial octets.
>
> Based on the examples, and some common sense, I think you don't want to
> specify the first two of the above in the SDP. AFAIK the first has a
> well defined value, and the second is well defined by counting the bytes
> in the uuie-value.

It is true that the fist octet is fixed and the second is the length, so 
they do not really convey useful information. But we do really need the 
third octet (the protocol discriminator). Based on this, I would prefer 
to include the whole object (User-user information element) rather than 
the object minus two octets.

>
>>> Its unclear what you intend to be included in the specified uuie element
>>> in the sdp - does it include the protocol discriminator or not?
>>
>> Yes, as I described above.
>>
>>>
>>> Also, the content and protocol discriminator contain octets, not just
>>> alphanumerics, but the syntax in 5.6 only permits alphanumerics. Is it
>>> your *intent* to limit the protocol discriminator and the value to that
>>> subset described by alphanumerics? And is that a reasonable thing to do?
>>> (For this use, it *might* be sufficient to prescribe a specific value
>>> for the protocol discriminator.)
>>
>> This is an error that we should fix. It is not the intention of this
>> draft to limit the protocol discriminator, so, we need to address this
>> issue and allow any possible value. We will fix it in the next version
>> of the draft.
>>
>>
>>
>>>
>>> 5.5.1 / 5.5.4
>>>
>>> The description of a=connection here is more appropriate to section
>>> 5.5.4. Here, for the initial offer, a=connection:new is the only thing
>>> that makes sense.
>>
>> Ok, we can move the a=connection:existing to 5.5.4.
>>
>>>
>>> In 5.5.4 it would be helpful to explain in more detail how a=connection
>>> is used to control whether an existing call is retained, dropped, or
>>> dropped and reestablished.
>>   >
>>>
>>> (In some cases it will be desired to do another o/a while leaving the
>>> pstn call undisturbed. In other cases it may the explicit intent to
>>> cause the call to be reestablished.)
>>
>> ok, will do.
>>
>>
>>>
>>> 5.6
>>>
>>> Is there a reason why you don't use the definition of an E.164 number
>>> from RFC 3966, rather than using your own?
>>
>> RFC 3966 defines how to represent an E.164 number as a URL. Here, we do
>> not intent to use URLs. Additionally, the tel: URL provide
>> representation of either global and local numbers, ISDN subaddresses,
>> extensions, and phone context, which are not intended to be used here.
>
> I meant not the entire tel syntax, but by reference to one of the ABNF
> symbols defined in 3966. That has the benefit of being more accessible
> to those reading ietf drafts, and to give the syntax in ABNF that can be
> readily incorporated with the rest of the syntax.
>
> Of course that would allowing visual-separator. I don't know if that is
> a big deal or not.

Actually, this field should only contain digits.

>
>>   >  What is the significance of
>>> making the "+" optional?
>>
>> The only reason I can think, is that the draft has a "SHOULD" strength
>> for using the international format of E.164. If someone really knows
>> what he is doing and can use a national format, it wouldn't use the "+"
>> sign. I recognize that this is not a very strong reason, but it is the
>> only justification I can find.
>
> Well, if the intent is to allow a national format, then it should be
> made clear that lack of the "+" means that. (And then how on how one is
> supposed to know *which* national format.)
>
> I realize there is text that says the actual callerid received may not
> be in E.164 format and so the comparison should take that into account.
> But I didn't think that applied to the number specified in SDP.

That is correct. That does not apply to the number included in SDP, but 
to the received Caller-ID in Q.931. Some operators of some countries send 
this number in international format; others send it in national format.

Unless someone has a very good reason, I guess we can preclude national 
numbers from SDP, so we should remove the '+' sign from the syntax.

>
>>   >  And why is caller-id-value defined differently
>>> than e164-address?
>>
>> Yes, I think they should be equal. Probably we should get rid of
>> "caller-id-value" and use "e164-address" instead.
>>
>>
>>
>>> Figure 4, and others:
>>>
>>> The E.164 number used (+15551234) isn't a valid one for country code 1.
>>
>> Is it because there are not enough number of digits for the subscriber
>> number?
>
> Right. Its just a nit. There is no area code in it.
>
>>> General Comment:
>>>
>>> Everything is described, and examples show, only a single m= line, and
>>> all the cs-correlation and setup things used within that one media
>>> section. There is no explanation of what to do if there is more than one
>>> pstn m= line, as might arise if the intent is to have an audio/video
>>> call. Without any explanation, it would seem that everything should be
>>> specified separately for each m= line, which might result in *two* pstn
>>> calls. Presumably that isn't want is desired.
>>>
>>> More definition is needed if the same pstn call is to be used for two
>>> media streams.
>>
>> Yes, I agree with you. I think we should add a section describing how to
>> setup an circuit-switched audio/video call.
>>
>>>
>>> Thanks,
>>> Paul
>>
>> Thanks for your comments,
>>
>> Miguel
>>

/Miguel

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From christer.holmberg@ericsson.com  Wed Oct 19 02:38:25 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: IETF - MMUSIC <mmusic@ietf.org>
Date: Wed, 19 Oct 2011 11:38:22 +0200
Thread-Topic: Draft new name: draft-holmberg-mmusic-sdp-bundle-negotiation (previously known as draft-holmberg-mmusic-sdp-multiplex-negotiation)
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Hi,

I've submitted a new draft:

draft-holmberg-mmusic-sdp-bundle-negotiation-00

It replaces draft-holmberg-mmusic-sdp-multiplex-negotiation, and the name o=
f the group extension is now 'BUNDLE' (instead of 'MULTIPLEX'), based on of=
f-line comments that the 'multiplexing' terminology caused some confusion.

Regards,

Christer



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<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">I've submitted a new draft:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">draft-holmberg-mmusic-sdp-bundle-negotiation-00</font=
></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">It replaces draft-holmberg-mmusic-sdp-multiplex-negot=
iation, and the name of the group extension is now 'BUNDLE' (instead of 'MU=
LTIPLEX'), based on off-line comments that the 'multiplexing' terminology c=
aused some confusion.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
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From harald@alvestrand.no  Wed Oct 19 02:59:28 2011
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On 10/19/11 11:38, Christer Holmberg wrote:
> Hi,
> I've submitted a new draft:
> draft-holmberg-mmusic-sdp-bundle-negotiation-00
> It replaces draft-holmberg-mmusic-sdp-multiplex-negotiation, and the 
> name of the group extension is now 'BUNDLE' (instead of 'MULTIPLEX'), 
> based on off-line comments that the 'multiplexing' terminology caused 
> some confusion.

And since the discussion seemed to indicate that Christer's way of 
writing drafts harmonizes better with mmusic than my way of writing 
drafts ..... I'm happy to cede the pen to Christer, and support his 
efforts in getting this draft improved and (eventually) adopted and 
published.

Expect no more updates to draft-alvestrand-one-rtp.

                 Harald

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


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    On 10/19/11 11:38, Christer Holmberg wrote:
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A058522341F489A@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
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        <div>&nbsp;</div>
        <div><font size="2">Hi,</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">I've submitted a new draft:</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">draft-holmberg-mmusic-sdp-bundle-negotiation-00</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">It replaces
            draft-holmberg-mmusic-sdp-multiplex-negotiation, and the
            name of the group extension is now 'BUNDLE' (instead of
            'MULTIPLEX'), based on off-line comments that the
            'multiplexing' terminology caused some confusion.</font></div>
      </font></blockquote>
    <br>
    And since the discussion seemed to indicate that Christer's way of
    writing drafts harmonizes better with mmusic than my way of writing
    drafts ..... I'm happy to cede the pen to Christer, and support his
    efforts in getting this draft improved and (eventually) adopted and
    published.<br>
    <br>
    Expect no more updates to draft-alvestrand-one-rtp.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A058522341F489A@ESESSCMS0356.eemea.ericsson.se"
      type="cite"><font face="Arial, sans-serif" size="3">
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Regards,</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Christer</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
      </font>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
mmusic mailing list
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<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
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Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Flemming Andreasen" <fandreas@cisco.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>
Cc: mmusic <mmusic@ietf.org>, draft-ietf-mmusic-media-loopback@tools.ietf.org
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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I would like to echo concerns voiced by Hadriel (in the linked e-mail
below) and particularly that loopback call should look as much similar
to regular calls as possible. Forcing deployments, which do not use ICE
for regular calls to use ICE for loopback calls, does not sound like a
good idea to me.=20

Thanks,
Tolga

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
Behalf
> Of Flemming Andreasen
> Sent: Monday, October 17, 2011 9:58 PM
> To: Hadriel Kaplan
> Cc: mmusic; <draft-ietf-mmusic-media-loopback@tools.ietf.org>
> Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
>=20
>=20
>=20
> On 10/12/11 10:59 AM, Hadriel Kaplan wrote:
> > On Oct 8, 2011, at 8:54 AM, Flemming Andreasen wrote:
> >
> >
> >> The issue was discussed on the mailing list in August, where the
> authors suggested it as a path forward and the chairs asked the WG for
> feedback on this approach, see e.g.
> >>
> >> http://www.ietf.org/mail-archive/web/mmusic/current/msg08857.html
> >>
> >> The issue was also discussed in the last IETF MMUSIC meeting (see
> http://www.ietf.org/proceedings/81/minutes/mmusic.txt)
> >>
> >> To date, we have only seen opposition to this proposal from you (on
> both the mailing list and in the meeting), and we are lacking an
> alternative path forward that can satisfy all involved parties,
including
> the general AD and WG chair concerns around definining an alternative
NAT
> traversal mechanism in this draft without proper discussion and
general
> considerations around the need for an alternative to ICE (per the
above
> thread).
> >>
> > In the email thread you referenced above there was in fact someone
else
> who was for keeping the priming mechanism, and it wasn't me.
> >
> > In fact, in that whole thread there was only one other person than
> yourself who was in favor of only having ICE in the draft.  Of course
> there was only one other person than myself who was in favor of
keeping
> the primer mechanism.
> >
> > My point is that there has been no consensus either way.  The
authors
> only changed the draft due to your objections, afaict, not due to some
> large consensus to do so.
> >
>=20
> To be clear, the author(s), myself, and our responsible AD did send a
> note to the list suggesting this as a path forward. I asked the WG
> members to speak up if they disagreed with the suggested path forward,
> and you are the only person to argue against subsequently. While I'll
> agree there is not unanimous agreement in the group with the
direction,
> I think we need to hear from more than 1 person to declare a lack of
> (rough) concensus. Having said that, it would be better if we could
> reach an agreement that is satisfactory to all.
>=20
>=20
> >> If you do not find this path forward acceptable, can you please
suggest
> an alternative that satisfies the general concerns around defining an
> alternative NAT traversal mechanism in this draft ?
> >>
> > There have been no "general concerns" as far as I can tell, other
than
> "we have ICE defined so why not use it".
> Yes - we have a Standards Track and general solution available, so we
> should use it.
>=20
> >    My concerns with mandating *only* ICE as opposed to having a
SHOULD
> use ICE but MAY use primer are detailed here:
> > http://www.ietf.org/mail-archive/web/mmusic/current/msg08611.html
> >
> > There was no feedback/response to that, afaict.
> >
> You are referencing a ~6 month old e-mail, and there has since then
been
> several more e-mail exchanges on this subject, an explanation of the
> issues with the old approach, and a thread plus WG meeting discussion
on
> the need for an ICE alternative (like latching), which didn't garner
any
> takers so far. If you believe those arguments still stand and offer a
> viable path forward, then please resurrect them (either in the context
> of this e-mail thread or as a new thread), and we (and hopefully other
> members of the WG) can discuss further. Again, to recap the concerns
> with the old approach:
> - We have a general and Standards Track NAT traversal solution in the
> form of ICE so we should use it
> - If there is a need for an ICE alternative, then we need to discuss
> that more generally and come up with a proper solution
> - If latching is indeed needed as an alternative solution to ICE, then
> it needs to be properly documented and the limitations of it spelled
out
> clearly. Furthermore, the loopback draft does not seem like the right
> place for this.
>=20
> Thanks
>=20
> -- Flemming
>=20
>=20
> > -hadriel
> >
> >
> >
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From pkyzivat@alum.mit.edu  Thu Oct 20 09:39:36 2011
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] FW:  I-D Action: draft-ietf-mmusic-sdp-cs-08.txt
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On 10/18/11 4:03 PM, Miguel A. Garcia wrote:

trimming out the settled stuff

>>>> 5.2.3.3
>>>>
>>>> You specify that the UUIE has a max size of 35 or 131 octets. AFAIK the
>>>> actual max information content is actually either 32 or 128 octets plus
>>>> a leading protocol discriminator byte.
>>>
>>> Right. The terminology in Q.931 helps to mislead. The User-User
>>> Information Element contains three parts:
>>>
>>> - 1-octet User-user information element identifier
>>> - 1-octet "Length of user-user contents"
>>> - 1-octet "Protocol discriminator"
>>> - "User Information".
>>>
>>> The "User Information can be up to 32 or 128 octets, depending on
>>> networks. Together with the other 3 octets, it makes a maximum of 35 or
>>> 131 octets.
>>>
>>> I guess the draft should elaborate the composition of this information
>>> element and clarify that it includes the 3 initial octets.
>>
>> Based on the examples, and some common sense, I think you don't want to
>> specify the first two of the above in the SDP. AFAIK the first has a
>> well defined value, and the second is well defined by counting the bytes
>> in the uuie-value.
>
> It is true that the fist octet is fixed and the second is the length, so
> they do not really convey useful information. But we do really need the
> third octet (the protocol discriminator). Based on this, I would prefer
> to include the whole object (User-user information element) rather than
> the object minus two octets.

Really??? I find that just an invitation to confusion and error.

Also, you should take a look at what is being done in the CUSS wg - 
draft-ietf-cuss-sip-uui-02.

For this purpose you have some latitude in what you allow. It isn't 
really necessary to allow all data bytes, all lengths, or all protocol 
discriminators. You only need to allow enough to work in all networks 
and to provide sufficient randomness. The cuss work wants to provide 
access to the complete mechanism, and even beyond that when not going 
over a PSTN link. So you might choose different approach. But its at 
least worth a look.

(Personally, I haven't been happy with CUSS including the protocol 
discriminator in the string with the data, but I haven't convinced 
anyone else of that.)

>>>> 5.6
>>>>
>>>> Is there a reason why you don't use the definition of an E.164 number
>>>> from RFC 3966, rather than using your own?
>>>
>>> RFC 3966 defines how to represent an E.164 number as a URL. Here, we do
>>> not intent to use URLs. Additionally, the tel: URL provide
>>> representation of either global and local numbers, ISDN subaddresses,
>>> extensions, and phone context, which are not intended to be used here.
>>
>> I meant not the entire tel syntax, but by reference to one of the ABNF
>> symbols defined in 3966. That has the benefit of being more accessible
>> to those reading ietf drafts, and to give the syntax in ABNF that can be
>> readily incorporated with the rest of the syntax.
>>
>> Of course that would allowing visual-separator. I don't know if that is
>> a big deal or not.
>
> Actually, this field should only contain digits.

My point in calling out 3966 is that it is more familiar to the sip 
community. But this isn't a big deal as long as the reference is specific.

>>> > What is the significance of
>>>> making the "+" optional?
>>>
>>> The only reason I can think, is that the draft has a "SHOULD" strength
>>> for using the international format of E.164. If someone really knows
>>> what he is doing and can use a national format, it wouldn't use the "+"
>>> sign. I recognize that this is not a very strong reason, but it is the
>>> only justification I can find.
>>
>> Well, if the intent is to allow a national format, then it should be
>> made clear that lack of the "+" means that. (And then how on how one is
>> supposed to know *which* national format.)
>>
>> I realize there is text that says the actual callerid received may not
>> be in E.164 format and so the comparison should take that into account.
>> But I didn't think that applied to the number specified in SDP.
>
> That is correct. That does not apply to the number included in SDP, but
> to the received Caller-ID in Q.931. Some operators of some countries
> send this number in international format; others send it in national
> format.
>
> Unless someone has a very good reason, I guess we can preclude national
> numbers from SDP, so we should remove the '+' sign from the syntax.

Or always include it. I guess if its always there it is redundant, but 
having it there would still remind people of what it is.

	Thanks,
	Paul

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<p class=3D"MsoNormal"><span lang=3D"EN-US">Hello all,<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">I have just submitted a new dra=
ft as bellow:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-=
family:=CB=CE=CC=E5;color:black">Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; &nbsp; draft-yue-mmusic-rtsp-substream-control-extension<br>
Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; 00<br>
Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &=
nbsp;&nbsp;RTSP Extension for Substream Control<br>
Creation date:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; 2011-10-21<br>
WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &=
nbsp;&nbsp;Individual Submission<br>
Number of pages: 11<br>
<br>
Abstract:<br>
&nbsp;&nbsp; This document defines extensions to RTSP 2.0 protocol, includi=
ng<br>
&nbsp;&nbsp; header </span><span lang=3D"EN-US" style=3D"font-size:10.0pt;c=
olor:black">=A1=B0</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;fon=
t-family:=CB=CE=CC=E5;color:black">Substream</span><span lang=3D"EN-US" sty=
le=3D"font-size:10.0pt;color:black">=A1=B1</span><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5;color:black">;,
 feature tag </span><span lang=3D"EN-US" style=3D"font-size:10.0pt;color:bl=
ack">=A1=B0</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-famil=
y:=CB=CE=CC=E5;color:black">Play.substream</span><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;color:black">=A1=B1</span><span lang=3D"EN-US" style=
=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5;color:black">;
 , and related new<br>
&nbsp;&nbsp; status codes.&nbsp; These extensions enables the playback cont=
rol of a<br>
&nbsp;&nbsp; media stream on substream basis.</span><span lang=3D"EN-US"><o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">You can refer it at <a href=3D"=
http://datatracker.ietf.org/doc/draft-yue-mmusic-rtsp-substream-control-ext=
ension/">
http://datatracker.ietf.org/doc/draft-yue-mmusic-rtsp-substream-control-ext=
ension/</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Your comments/suggestions are v=
ery much appreciated!!<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">BR,<o:p></o:p></span></p>
<p class=3D"MsoNormal" align=3D"left" style=3D"text-align:left"><span lang=
=3D"EN-US" style=3D"font-size:10.0pt;font-family:=BB=AA=CE=C4=CF=B8=BA=DA;c=
olor:black">Peiyu YUE(Roy)<br>
</span><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:=BB=AA=CE=
=C4=CF=B8=BA=DA;color:gray">Huawei Software Technologies Co., Ltd
</span><!--[if gte vml 1]><v:shapetype id=3D"_x0000_t75" coordsize=3D"21600=
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t-family:=CB=CE=CC=E5"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:12.0pt;font-=
family:=CB=CE=CC=E5"><br>
</span><span lang=3D"EN-US" style=3D"font-size:7.5pt;font-family:=BB=AA=CE=
=C4=CF=B8=BA=DA;color:gray">N5, NO. 101, Huawei Base<br>
Ruanjian Av, YuHua Dist<br>
Nanjing, 210012, P.R.C<br>
Tel: &#43;86-25-56622058<br>
Fax: &#43;86-25-56624081<br>
Mobile: &#43;86-13809030401<br>
Email: yuepeiyu@huawei.com<br>
http://www.huawei.com<br>
---------------------------------------------------------------------------=
----------------------------------------------------------<br>
</span><span style=3D"font-size:7.5pt;font-family:=BB=AA=CE=C4=CF=B8=BA=DA;=
color:gray">=B1=BE=D3=CA=BC=FE=BC=B0=C6=E4=B8=BD=BC=FE=BA=AC=D3=D0=BB=AA=CE=
=AA=B9=AB=CB=BE=B5=C4=B1=A3=C3=DC=D0=C5=CF=A2=A3=AC=BD=F6=CF=DE=D3=DA=B7=A2=
=CB=CD=B8=F8=C9=CF=C3=E6=B5=D8=D6=B7=D6=D0=C1=D0=B3=F6=B5=C4=B8=F6=C8=CB=BB=
=F2=C8=BA=D7=E9=A1=A3=BD=FB<span lang=3D"EN-US"><br>
</span>=D6=B9=C8=CE=BA=CE=C6=E4=CB=FB=C8=CB=D2=D4=C8=CE=BA=CE=D0=CE=CA=BD=
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=B1=BE=D3=CA=BC=FE=D6=D0<span lang=3D"EN-US"><br>
</span>=B5=C4=D0=C5=CF=A2=A1=A3=C8=E7=B9=FB=C4=FA=B4=ED=CA=D5=C1=CB=B1=BE=
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y use of the
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Date: Fri, 21 Oct 2011 02:19:54 +0000
From: Bert Greevenbosch <Bert.Greevenbosch@huawei.com>
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--Boundary_(ID_92JUmIYkuohpbtkmTrUQew)--

From miguel.a.garcia@ericsson.com  Fri Oct 21 00:14:41 2011
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Subject: [MMUSIC] Agenda requests for IETF 82 Taipei
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Hi,

If you want a timeslot for discussing a draft in Taipei, please ping the 
chairs (unless you have already done so).

I want to draw your attention to the fact that we are scheduled for 
Friday 9 AM, but for a two-hour time, and not 2.5-hours as we used to 
have. It seems that the schedule of the Friday morning at IETF meetings 
have changed the sessions time schemes.

/Miguel

-- 
Miguel A. Garcia
+34-91-339-3608
Ericsson Spain

From magnus.westerlund@ericsson.com  Fri Oct 21 02:25:29 2011
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-16
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Hi,

I have reviewed An "Extension to the Session Description Protocol (SDP)
for Media Loopback" whos WG last call ended yesterday. I hope this
comments will be addressed as I think there are some serious short
commings and also process steps that must be done.

1. This document contains media types. But it has not been reviewed by
the ietf-types@iana.org mailing list. This must be completed prior to
publication request.

2. I do CC the payload WG mailing list as this document really should be
reviewed there. I know the chairs have realized that need.

3. I think the authors and chairs should discuss how to handle the fact
that there is 6 authors on the front page. It is possible to ask for
exception on this, but might be simpler to move one of the authors to a
contributor section.

4. Title: I think the title is indicating that this is only an SDP
solution, but it clearly contains the considerations needed for the
RTP/RTCP part also. I think it should reflect that it is a solution for
RTP transported media loopback including SDP signalling.

5. Abstract: "maintaining voice/real-time Text/video quality,
reliability, and overall performance." The "voice/real-time"
construction appears to indicate that voice isn't real-time. I guess
something like: maintaining real-time voice/Text/video quality,
reliability, and overall performance. would be easier and clearer.

6. Section 3.
No reference to SDP offer answer. And the "Offering entity" is called
agent in SDP O/A terminology.

7. Section 3:
"it MAY do so as defined in section 5.4.1 of
    this specification."

Considering what is written in 5.4.1 I think that section can be moved
to section 3.

8. Section 4:
    An answering entity that is not compliant to this specification and
    which receives an offer with the "loopback" media attributes MAY
    ignore the attribute and treat the incoming offer as a normal
    request.

I would suggest reformulate this. I expect this text to say what is
expected to happen when the answering agent doesn't know what this
attribute is. Which is ignore the attribute, attempt to establish the
session without loopback and send back the answer without the attribute.

I would suggest to clarify that the offering agent should consider
terminating the establishment at this point as loopback was not
available, unless it might actual "ring" if this is an end-point at a user.

9. What about declarative SDP usage?

10. What is really the purpose of the sections 3 and 4. They seem to be
teasers for a very small part of the issue that is covered in detail in
section 5. Why not move the appropriate text in to the relevant subsections?

11. Section 5.3
"    Note: A loopback offer in a given media description MUST NOT
    contain the standard mode attributes sendonly, recvonly, sendrecv,
    or inactive. The loopback-mode attributes (loopback-source and
    loopback-mirror) replace the standard attributes."

This appears to be a strange MUST NOT which I think might encounter
issues. Considering that SIP proxies that looks into this SDP requires
the direction attribute to be present and adds it, what happens now.
And why isn't sendrecv and inactive possible values? sendrecv appears to
be requried for loopback to work, and inactive would put media transport
in an established session on hold. Isn't that reasonable?

12. Section 5.3:
I think one could clarify that the loopback source needs to include the
encapsulation or direct loopback payload formats when suggesting pckt
type loopback.

13. What happens if I  offer both media and pckt loopback at the same
time by including two loopback attributes?

14. Section 5.1:
    loopback-type          = loopback-choice [1*SP loopback-choice]
    loopback-choice        = loopback-type-pkt / loopback-type-media
    loopback-type-pkt      = "rtp-pkt-loopback"
    loopback-type-media    = "rtp-media-loopback"

The ABNF appears broken as it doesn't define the whole attribute. I
would suggest to add a line:

loopback-attr = "a=loopback:" loopback-type

15. Section 5.2 is missing ABNF.
I think it should make clear the attributes definition completely.

16. Section 5
Two new media attributes are defined: one indicates the type of
    loopback and the other indicates the mode of the loopback.

I would claim that this spec defines 3 attributes.

17. Section 5.4:

The loopback-mode attributes (loopbackThe
    port number and the address in the answer (m= line) indicate where
    the answerer would like to receive the media stream.

Strange sentence!

18. Section 5.4:

If an answerer wishes to accept the loopback request it MUST
    include both the loopback mode and loopback type attribute in the
    answer. When a stream is offered with the loopback-source
    attribute, the corresponding stream in the response MUST be
    loopback-mirror and vice versa, provided that answerer is capable
    of supporting the requested loopback-type.

This seems to be in partial conflict with 5.4.1. As my impression of the
above, is that if you don't want to accept, you don't include the
attributes. But 5.4.1 says that it should be included but the port set
to 0.

19. Section 5.4
I am missing clear rules for the cases when I might exclude payload
types from loopback-mode attributes compared to the m= lines. Are this
only the loopback payload types? What happens if I exclude a regular
media payload format?

20. Section 7.1: The packet expansion of the encapsulated format I think
needs better discussion. I think one needs to first recommend PMTU
discovery. Secondly I think it is likely worth discussing when the
loopback source should reduce the packet sizes to avoid fragmentation
versus the need to measure that size in the forward link and thus
causing fragmentation.

21. Section 6.
"An answering entity that is compliant to this specification and
    accepting a media with the loopback type rtp-media-loopback MUST
    transmit all received media back to the sender. "

I think this MUST, MUST have an exception for that the path from the
mirror to the source actually can support the bit-rate required.

22. Section 6.
Media loopback, if a loopback source would send more than one stream how
does the loopback mirror indicate which source is which in its return
stream? SDES CNAME could be copied but would be confusing for anyone
monitoring this stream as SSRCs from both end-points would appear to
have the same CNAME. The ssrc-grouping could possibly be used, but I
think one needs to consider the effect of that the SSRCs to be grouped
are not on the same end-point. I don't even know if my  SRCNAME proposal
could deal with this case well.

23. Section 7.1.1:
    Marker (M) bit: If the received RTP packet is looped back in
    multiple RTP packets, the M bit is set to 1 in the last packet,
    otherwise it is set to 0.

I don't like this definition as it means that complete non fragment
packets will have a m=0 the same as non last fragements. would it not be
better to define it so that all packets have m=1 except non-last fragments.

24.section 7.1.1.:
The RTP timestamp MUST use the same clock rate used by the
    loopback-source.

I think this needs to be defined as the same clock rate used by the
packet being encapsulated. Otherwise this becomes ambigous when the
loopback source have payload types with multiple rates.

25.  Section 7.1.2:

Payload strucutre. I think it is unclear. Which of the following options
do you really want.

Outer RTP header
(optional CSRC list or Header ext)
Receive timestamp
Inner RTP packet {
  - RTP header
  - (optional CSRC list or Header ext)
  - RTP payload
}

or

Outer RTP header
(optional CSRC list or Header ext)
RTP Payload header {
Receive timestamp
Inner RTP packet header fields as specified in 7.1.2
}
Inner RTP payload

Note that the second doesn't echo back the extension header

In general the text is not clear that the inner RTP payload should be
included.

26. Section 7.1.2:
You decide to overwrite the version field in the packet echoed back that
is likely fine, however, why are you overwriting the P and X bits? Why
not include the header extension in what is being echoed back so that
mirror doesn't have to remove them in case they are present.

27. Section 7.1.2: The fragmentation procedures are unclear.
I guess you want the RTP header part to be duplicated with each fragment
but that isn't clearly specified. But, will you include header
extensions also in the repeat if you include them?

28. Section 7.1.2:
    The
    Receive timestamp MUST be based on the same clock used by the
    loopback-source.

This is an impossibility. The two end-points can't have the same clock.
The can have a common clock reference like NTP. But, they are still not
the same clocks, but possibly synchronized well enough for your needs

What I guess you are after is that the clock rate should be the same as
for the packet encapsulated. The Timestamp in the outer header should
also be corrected.

I also think you should clarify that the receive timestamp and the
timestamp in the header needs to be from the same clock in the mirror.
Otherwise the time spent on the mirror for the packet is not useful. I
think it might also be worth actually writing up the measurements that
become possible by these two timestamps plus having the source logg
transmission and arrival times. The fact that one can derive both one
way delay and jitter for each loopback packet is kind of important.

However, to have correct one way delays the mirror clock needs to be
synchronized with the source clock, or at least the current offset.

29. Section 7.1.2:
"Any padding octets in the original packet MUST NOT be included in
    the loopback packet generated by a loopback-mirror."

What is the motivation behind this? If the reason I want to do packet
loopback is to find out if somebody is manipulating the packets between
the source and the mirror you need to include the whole packet as it is,
not throw away parts of it. What if padding is added or removed on path.
That might matter for the one doing the active measurement.

30. Section 7.2.1:
Sequence Number: The RTP sequence number SHOULD be generated by the
    loopback-mirror in the usual manner with a constant random offset.

What is the acceptable cases for when to break it? I assume the
intention with not saying MUST is that one can copy it from the
incomming packet. Thus when source -> mirror packet losses occur causing
wholes in the return path which isn't real loss on the mirror->source
path. It also causes third party monitor to report losses that aren't
real on this stream.

31. Section 7.2.1:
Timestamp: The RTP timestamp denotes the sampling instant for when
    the loopback-mirror is transmitting this packet to the
    loopback-source.  The RTP timestamp MUST be based on the same clock
    used by the loopback-source.  The initial value of the timestamp
    SHOULD be random for security reasons (see Section 5.1 of RFC 3550
    [RFC3550]).

One more case of not correct timestamp definition.

32. Section 7.2.1:
The usage of the CSRC and extension headers for this payload format
should be clarified. Shall the mirror copy these fields or not?

33. Section 8:
Why should a mirror do VoIP Metric Reports Block XR blocks for a video
stream. I think some clarification based on media type is in order here.

34. Section 9.
This section is thoroughly inadequate. The reason is that you have a
forward path that is coupled to the reverse path. If the loopback is to
work correctly the mirror needs to be able to send back what the source
sent to it. Thus it is the loopback source that needs to adapt the media
bit-rate so that both forward and reverse path can handle the flows. If
doesn't the mirror will need to reduce the bit-rate it transmits and
that can only happen in two ways, dropping packets or transcoding them
to a lower rate fulfilling the congestion control requirement. Neither
will fulfill the purpose of the loopback as I see it. This issue needs
discussion, including some considerations in how to actually accomplish
it with the tools we have available.

35. Section 10.1

What is the purpose of this example?

What I see it establish the fact that the end-points can take respecitve
roles of source and mirror. But it doesn't indicate any media which can
be looped back?

36. Section 10.1
Either of the o= lines are in error. Because they can't be the same when
it comes to the ID and version fields.

37. Section 11:
This needs to be very much fleshed out.

First of all the denial of service attack are actually several possible.

One is to use a loopback mirror as a slight amplification source and
primarily a indirection. This is accomplished by attacker spoofing
source address to the targets and sending media to the loopback mirror.

A much more hideous attack is the one where an attacker emits signalling
messages that results in that two mirrors are connected to each other.
As soon as a few packet are injected into that loop it will grow until
the resources are consumed. Any non lost packet arriving at either
mirror will be forwarded and grow 16 bytes per mirror bounce. It will
bounce between the mirrors until fragmentation occurs and then you have
two packets. This will continue until all CPU or bandwidth resources are
consume causing packet loss.

So I think you better explain how one in reality do authenticate the
signalling messages so that only approved initiators of loopback are
allowed. Because doing this without being certain that it is not a
malicous entity setting up the sessions are plain stupid and this should
not be specified if there is no solution to this issue.

I also think a consideration of the media plane security should be done.
Are there any privacy concerns.

38. I also don't see any discussion of the interaction between this an
the high layers. For example if this is implemented in a SIP end-point
is that supposed to ring? Are there any case where media will be allowed
to be played out when doing loopback? I don't think so, but it should be
stated.

39. Section 7.2:

If one has multiple media sources in an RTP session arriving at the
mirror. I expect the mirror to loopback both. However, in which way does
the loopback source determine which mirror stream is related to the one
being sent?

40. Section 7.2: In an RTP session where one uses more than a single
payload type for the source to mirror media, how does the source
determine which original payload type the payload had?

Is there need for solution the Retransmission payload foramt uses RFC 4588?

41. The media type registrations "rate" parameter I think needs to be
redefined. It needs to reference the rate of the payload types it
intends to mirror back.

Also, I don't think 8k Hz clock is typical enough for any of the media
type. Audio has a number of common ones beyond 8k. video uses most
commonly but not exclusively 90k. Text uses yet something else.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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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=
 Working Group of the IETF.

	Title           : SDP: Session Description Protocol
	Author(s)       : Mark Handley
                          Van Jacobson
                          Colin Perkins
                          Ali Begen
	Filename        : draft-ietf-mmusic-rfc4566bis-04.txt
	Pages           : 49
	Date            : 2011-10-24

   This memo defines the Session Description Protocol (SDP).  SDP is
   intended for describing multimedia sessions for the purposes of
   session announcement, session invitation, and other forms of
   multimedia session initiation.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc4566bis-04.txt

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Flemming Andreasen <fandreas@cisco.com>
Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On Oct 17, 2011, at 9:58 PM, Flemming Andreasen wrote:
[snip]
>  Again, to recap the concerns with the old approach:
> - We have a general and Standards Track NAT traversal solution in the for=
m of ICE so we should use it

And we are - the loopback draft explicitly says to do ICE if both sides can=
 do so.  It doesn't ignore ICE.  The question is if the loopback draft can =
also say what to do if ICE isn't supported. (and no, saying "well then go s=
upport ICE" isn't an option for the reasons explained previously in the ema=
il threads)


> - If there is a need for an ICE alternative, then we need to discuss that=
 more generally and come up with a proper solution
> - If latching is indeed needed as an alternative solution to ICE, then it=
 needs to be properly documented and the limitations of it spelled out clea=
rly. Furthermore, the loopback draft does not seem like the right place for=
 this.

You make it sound like latching is some new thing - it's not - it's been ar=
ound since long before ICE.  I don't disagree it's not documented in an RFC=
, though it is somewhat documented in draft-ietf-mmusic-media-path-middlebo=
xes.

But I'm not sure latching even needs to be documented for loopback-draft to=
 state something like "send packet out".  Something like what Dan Wing sugg=
ested here:
http://www.ietf.org/mail-archive/web/mmusic/current/msg08865.html

-hadriel


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-traffic-class-for-sdp-00.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=
 Working Group of the IETF.

	Title           : The Session Description Protocol (SDP) &#39;trafficclass=
&#39; Attribute
	Author(s)       : James Polk
                          Subha Dhesikan
                          Paul E. Jones
	Filename        : draft-ietf-mmusic-traffic-class-for-sdp-00.txt
	Pages           : 20
	Date            : 2011-10-24

   This document proposes a new Session Description Protocol (SDP)
   attribute to identify the traffic class a session is requesting
   in its offer/answer exchange.


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From HKaplan@acmepacket.com  Tue Oct 25 11:01:08 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-16
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On Oct 21, 2011, at 5:25 AM, Magnus Westerlund wrote:

> 11. Section 5.3
> "    Note: A loopback offer in a given media description MUST NOT
>    contain the standard mode attributes sendonly, recvonly, sendrecv,
>    or inactive. The loopback-mode attributes (loopback-source and
>    loopback-mirror) replace the standard attributes."
>=20
> This appears to be a strange MUST NOT which I think might encounter
> issues. Considering that SIP proxies that looks into this SDP requires
> the direction attribute to be present and adds it, what happens now.
> And why isn't sendrecv and inactive possible values? sendrecv appears to
> be requried for loopback to work, and inactive would put media transport
> in an established session on hold. Isn't that reasonable?

How about this:
Since media loopback requires bidirectional RTP, its normal direction mode =
is "sendrecv"; the "sendrecv" direction attribute MAY be encoded in SDP or =
not, as per section 5.1 of [RFC3264], since it is implied by default.  If e=
ither the loopback source or mirror wish to disable loopback use during a s=
ession, the direction attribute "inactive" MUST be used as per [RFC3264].  =
The direction mode attributes "recvonly" and "sendonly" are incompatible wi=
th the loopback mechanism and MUST NOT be indicated when generating an SDP =
Offer or Answer.  When receiving an SDP Offer or Answer, if "recvonly" or "=
sendonly" is indicated for loopback, the SDP-receiving agent MAY treat it a=
s a failure of the loopback negotiation or ignore the direction attribute.=
=20

[This last sentence is probably controversial?]

-hadriel


From fandreas@cisco.com  Wed Oct 26 08:23:03 2011
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Cc: Dan Wing <dwing@cisco.com>, mmusic <mmusic@ietf.org>, "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
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On 10/24/11 2:06 PM, Hadriel Kaplan wrote:
> On Oct 17, 2011, at 9:58 PM, Flemming Andreasen wrote:
> [snip]
>    
>>   Again, to recap the concerns with the old approach:
>> - We have a general and Standards Track NAT traversal solution in the form of ICE so we should use it
>>      
> And we are - the loopback draft explicitly says to do ICE if both sides can do so.  It doesn't ignore ICE.  The question is if the loopback draft can also say what to do if ICE isn't supported. (and no, saying "well then go support ICE" isn't an option for the reasons explained previously in the email threads)
>
>    
I'm not sure which previous part you are referring to since many 
different points have been made by different people.

However, can we change the above to say that loopback implementations 
MUST still support ICE, but then add text/procedures around what to do 
in case ICE procedures cannot be used per the points made by Emil in 
http://www.ietf.org/mail-archive/web/mmusic/current/msg08788.html, or 
are you objecting to the "MUST support ICE" part ?

>    
>> - If there is a need for an ICE alternative, then we need to discuss that more generally and come up with a proper solution
>> - If latching is indeed needed as an alternative solution to ICE, then it needs to be properly documented and the limitations of it spelled out clearly. Furthermore, the loopback draft does not seem like the right place for this.
>>      
> You make it sound like latching is some new thing - it's not - it's been around since long before ICE.  I don't disagree it's not documented in an RFC, though it is somewhat documented in draft-ietf-mmusic-media-path-middleboxes.
>    
We'll need more than that I think. As I've stated before, we have asked 
the working group on multiple occasions if there is interest in 
exploring alternatives to the ICE NAT traversal mechanisms (e.g. 
latching) and then get those discussed and documented properly. So far, 
we haven't seen any takers (and no, the loopback draft is not the right 
place to put it since it needs proper attention and discussion).

> But I'm not sure latching even needs to be documented for loopback-draft to state something like "send packet out".  Something like what Dan Wing suggested here:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg08865.html
>    
We are going in circles here. Dan's proposed text was:
<quote>

   "If the network is using media latching, the endpoint MUST emit
    a UDP packet (in order for media latching to work) towards the
    remote endpoint and it SHOULD send several packets to accommodate
    packet loss.  If the network is using ICE or media latching, the
    endpoint SHOULD emit packets every 15 seconds to accommodate
    stateful middleboxes (e.g., NATs) [RFC6263]."

</quote>

Again, media latching needs to be discussed and documented properly if 
you want to reference it, and it cannot be part of the loopback draft. 
Example questions that come to mind here:
- How does an endpoint know the "network" is using latching and hence 
whether to send this UDP packet or not ?
- Is the remote endpoint involved in this latching operation, and if so 
what does it need to do ?
- Under what circumstances should you use this media latching mechamism ?

We need something more solid to reference to move us forward, or some 
alternative concrete text proposal to the above that doesn't lead to 
open questions, e.g. along the lines of Emil's proposal, (however I'm 
not sure it goes to the extent that you are looking for in terms of 
removing ICE dependencies, but you tell me).

Thanks

-- Flemming


> -hadriel
>
>
>    

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On 10/24/11 2:06 PM, Hadriel Kaplan wrote:
<blockquote
 cite="mid:2B5668C4-72EB-4DFD-909C-847FF5AE2491@acmepacket.com"
 type="cite">
  <pre wrap="">
On Oct 17, 2011, at 9:58 PM, Flemming Andreasen wrote:
[snip]
  </pre>
  <blockquote type="cite">
    <pre wrap=""> Again, to recap the concerns with the old approach:
- We have a general and Standards Track NAT traversal solution in the form of ICE so we should use it
    </pre>
  </blockquote>
  <pre wrap="">
And we are - the loopback draft explicitly says to do ICE if both sides can do so.  It doesn't ignore ICE.  The question is if the loopback draft can also say what to do if ICE isn't supported. (and no, saying "well then go support ICE" isn't an option for the reasons explained previously in the email threads)

  </pre>
</blockquote>
I'm not sure which previous part you are referring to since many
different points have been made by different people. <br>
<br>
However, can we change the above to say that loopback implementations
MUST still support ICE, but then add text/procedures around what to do
in case ICE procedures cannot be used per the points made by Emil in
<a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/mmusic/current/msg08788.html">http://www.ietf.org/mail-archive/web/mmusic/current/msg08788.html</a>, or
are you objecting to the "MUST support ICE" part ? <br>
<br>
<blockquote
 cite="mid:2B5668C4-72EB-4DFD-909C-847FF5AE2491@acmepacket.com"
 type="cite">
  <pre wrap="">
  </pre>
  <blockquote type="cite">
    <pre wrap="">- If there is a need for an ICE alternative, then we need to discuss that more generally and come up with a proper solution
- If latching is indeed needed as an alternative solution to ICE, then it needs to be properly documented and the limitations of it spelled out clearly. Furthermore, the loopback draft does not seem like the right place for this.
    </pre>
  </blockquote>
  <pre wrap="">
You make it sound like latching is some new thing - it's not - it's been around since long before ICE.  I don't disagree it's not documented in an RFC, though it is somewhat documented in draft-ietf-mmusic-media-path-middleboxes.
  </pre>
</blockquote>
We'll need more than that I think. As I've stated before, we have asked
the working group on multiple occasions if there is interest in
exploring alternatives to the ICE NAT traversal mechanisms (e.g.
latching) and then get those discussed and documented properly. So far,
we haven't seen any takers (and no, the loopback draft is not the right
place to put it since it needs proper attention and discussion). <br>
<br>
<blockquote
 cite="mid:2B5668C4-72EB-4DFD-909C-847FF5AE2491@acmepacket.com"
 type="cite">
  <pre wrap="">
But I'm not sure latching even needs to be documented for loopback-draft to state something like "send packet out".  Something like what Dan Wing suggested here:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/mmusic/current/msg08865.html">http://www.ietf.org/mail-archive/web/mmusic/current/msg08865.html</a>
  </pre>
</blockquote>
We are going in circles here. Dan's proposed text was:<br>
&lt;quote&gt;<br>
<span class="Apple-style-span"
 style="color: rgb(0, 0, 0); font-family: Times; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; font-size: medium;">
<pre
 style="white-space: pre-wrap; word-wrap: break-word; width: 1322px;">  "If the network is using media latching, the endpoint MUST emit
   a UDP packet (in order for media latching to work) towards the 
   remote endpoint and it SHOULD send several packets to accommodate
   packet loss.  If the network is using ICE or media latching, the
   endpoint SHOULD emit packets every 15 seconds to accommodate
   stateful middleboxes (e.g., NATs) [RFC6263]."</pre>
</span>&lt;/quote&gt;<br>
<br>
Again, media latching needs to be discussed and documented properly if
you want to reference it, and it cannot be part of the loopback draft.
Example questions that come to mind here:<br>
- How does an endpoint know the "network" is using latching and hence
whether to send this UDP packet or not ? <br>
- Is the remote endpoint involved in this latching operation, and if so
what does it need to do ?&nbsp; <br>
- Under what circumstances should you use this media latching mechamism
? <br>
<br>
We need something more solid to reference to move us forward, or some
alternative concrete text proposal to the above that doesn't lead to
open questions, e.g. along the lines of Emil's proposal, (however I'm
not sure it goes to the extent that you are looking for in terms of
removing ICE dependencies, but you tell me).<br>
<br>
Thanks <br>
<br>
-- Flemming <br>
<br>
<br>
<blockquote
 cite="mid:2B5668C4-72EB-4DFD-909C-847FF5AE2491@acmepacket.com"
 type="cite">
  <pre wrap="">
-hadriel


  </pre>
</blockquote>
</body>
</html>

--------------020400040003020204030108--

From HKaplan@acmepacket.com  Wed Oct 26 20:36:11 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Flemming Andreasen <fandreas@cisco.com>
Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Cc: Dan Wing <dwing@cisco.com>, mmusic <mmusic@ietf.org>, "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Hi Flemming,
Sorry about the length of this email.
I think we are either talking past each other, or going in circles as you s=
aid.  :(
It would probably be better to discuss this in person in Taipei, but here's=
 an attempt at a reset:

1) My goal is to be enable a SIP-domain administrator to make loopback test=
-calls from a testing system (phone or server SIP UAC) to SIP clients; whet=
her those SIP clients be behind NATs or not.  To achieve that goal, I belie=
ve we need an RFC that we can reference the SIP client vendors to implement=
 against.

2) To make it useful, I believe such an RFC needs to:
	a) Be likely to be implemented: the easier it is to implement, the likelie=
r to be implemented.
	b) Be as close to normal calls as possible, both from a signaling and medi=
a-path perspective: if regular calls use ICE, media-loopback calls need to =
use ICE; if regular calls don't use ICE, media-loopback calls should not ei=
ther... because we don't want a different media-path used for them or else =
the purpose of media-loopback test calls is negated.
	c) Not require changing the deployed SIP and media-handling architecture, =
whatever it may be; media-loopback is a testing tool, not new service/reven=
ue, so people won't be willing to change their whole infrastructure just to=
 make a test call.

3) Current SIP-domain media-handling architectures differ, but we don't nee=
d to go define what all of them are or could be to enable media-loopback; a=
ll we have to do is point 2(b) above: keep them doing what they do for norm=
al calls, except in a loopback mode.  The only problem with that is that in=
 normal calls the SIP UA sends RTP without waiting to receive RTP, for a se=
ndrecv mode call.  Because of RFC 4787, we know that if the UA is behind a =
NAT then not sending some packets out first means the loopback packets won'=
t be let in. In order to make the loopback call appear as close to a normal=
 call as possible, therefore, we need to have the UA send some packets out =
first. (luckily we have RFC 4787 or we'd be having an argument about having=
 to publish that before publishing media-loopback! ;)

4) We don't need to describe latching to specify in a media-loopback RFC th=
at a SIP UA needs to mimic what is does in a normal call.  Al we have to sa=
y is: use ICE if you support ICE and the incoming SDP offer indicated ICE a=
s per RFC 5245, or just send some primer RTP packets if what you would do i=
n a normal call is send RTP without ICE.  Frankly, it won't hurt anything i=
f a SIP UAC *always* sends primer RTP packets in a loopback call, even if i=
t also does ICE.  Whether NAT traversal works by latching, or by some wires=
hark script, or by reading tea leaves doesn't matter.  We just need to foll=
ow 2(b): act like a normal call as much as possible.

5) My original email was to indicate my belief that the change from draft-1=
5 to draft-16 is a bad one: both because it makes the situation worse not b=
etter, and because I feel it was taken without real consensus in this WG.  =
I wouldn't care much about the lack of consensus for some editorial nit, bu=
t I think this is a big deployment-impacting change for a vote of 3 vs. 2 t=
o be called "rough consensus".

6) The current loopback draft-16 says this and only this with respect to ha=
ndling loopback calls and NATs:
Section 5.7:
    ICE/STUN/TURN provide a general solution to establishing media
    sessions between entities that are behind NATs. Loopback sessions
    that involve one or more end points behind NATs SHOULD use these
    general solutions wherever possible.

That's it, in the ENTIRE document.  That's approaching 'useless' on the Ric=
hter scale of RFC mandates.  How do you know if you're "behind NATs"?  If i=
t's a "SHOULD", what's the exception/alternative?  What if you're not in "w=
herever possible"?
The previous draft version-15, on the other hand, told you what to do if no=
t ICE. We can argue about whether what it said to do was good or not, but a=
t least it did something.=20

-hadriel


From miguel.a.garcia@ericsson.com  Thu Oct 27 00:13:00 2011
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Fwd: Protocol Action: 'IANA Registration of the 'image' Media Type for the Session Description Protocol (SDP)' to Proposed Standard (draft-salgueiro-mmusic-image-iana-registration-09.txt)
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Hi,

this document was developed through the AD-sponsored track. Just for your 
information, IANA registration of the media type "image" in SDP has been 
approved now.

/Miguel

-------- Original Message --------
Subject: Protocol Action: 'IANA Registration of the 'image' Media Type 
for the	Session Description Protocol (SDP)' to Proposed Standard 
(draft-salgueiro-mmusic-image-iana-registration-09.txt)
Date: Wed, 26 Oct 2011 21:56:51 +0200
From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce <ietf-announce@ietf.org>
CC: RFC Editor <rfc-editor@rfc-editor.org>

The IESG has approved the following document:
- 'IANA Registration of the 'image' Media Type for the Session
    Description Protocol (SDP)'
   (draft-salgueiro-mmusic-image-iana-registration-09.txt) as a Proposed
Standard

This document has been reviewed in the IETF but is not the product of an
IETF Working Group.

The IESG contact person is Gonzalo Camarillo.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-salgueiro-mmusic-image-iana-registration/




     Technical Summary

         This document describes the usage of the 'image' media type
   and registers it with IANA as a top-level media type for the
   Session Description Protocol (SDP).  This media type is
   primarily used by SDP to negotiate and establish T.38 media
   streams.

     Working Group Summary

     While this is not an official document of the MMUSIC working
         group, the document has been reviewed in the MMUSIC WG. It is
         widely supported by its members and external bodies. Has
         received substantial review, including an unofficial
         WGLC. There is consensus to proceed forward with the
         document. Since this document is registering a missing token
         in an existing registry, it was decided to expedite the
         document in the sake of simplicity and speedness.


     Document Quality

         There are implementations and RFCs documenting the usage of
         the "image" media type in SDP. This document adds it to the
         SDP parameters registry.

Personnel

   Miguel Angel Garcia is the document shepherd.
   Gonzalo Camarillo is the responsible AD.

RFC Editor's Note:

OLD:

   The 'image' media type for the Session Description Protocol
   is used to describes a media stream that represents a still
   image.  The media subtype further describes the specific
   format of the image.  Currently, the 'image' media type for
   SDP is used ubiquitously by the SIP control protocol to
   establish T.38 media streams.

NEW:

The 'image' media type for the Session Description Protocol
is used to describe a media stream whose content consists of
one or more separate images that require appropriate hardware to
display. The media subtype further describes the specific format
of the image. Currently, the 'image' media type for SDP is used
ubiquitously by the SIP control protocol to establish T.38 media streams.
_______________________________________________
IETF-Announce mailing list
IETF-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/ietf-announce

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Flemming Andreasen <fandreas@cisco.com>
Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Cc: "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>, "mmusic \(E-mail\)" <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Howdy,
Before you read the insanely long email I just sent, I have a proposed solu=
tion which may satisfy you, the authors, and the Working Group as a whole. =
 Or maybe this solution is what the loopback draft actually means and I'm j=
ust dumb as a stump.

As I understand it, your concern is we (1) don't have an RFC defining how N=
AT traversal works if not with ICE, (2) don't want to define a "primer" thi=
ngy just for this one loopback draft, and (3) are dizzy from of going aroun=
d in circles. :)

Well... there is an escape clause in ICE RFC 5245, section 5.1, for what to=
 do if there's an ICE mismatch or a received Offer doesn't indicate ICE.  I=
n theory an ICE-supporting UA must follow that behavior.  The rationale for=
 this is explained in section 18, with regards to ALGs and SBCs. =20

Note that when the other side doesn't do ICE or there is an ALG, section 5.=
1 instructs the UA to do section 10.  Per section 10, if the peer supports =
ICE at all (even if ICE is mismatched and not being used), the local UA sen=
ds STUN Binding Indications as the keepalives.  If the peer doesn't support=
 ICE at all, then what the local UA sends is its choice.  Examples the ICE =
RFC gives is Comfort Noise, or draft-ietf-avt-rtp-no-op (which I believe wa=
s written by someone named Flemming Andreasen :)

My point in all this is to propose the following high-level straw-man:

1) We use the STUN Binding Indications as our primer packet format, by refe=
rencing STUN RFC.
2) We put in text that a UA loopback-mirror MUST either support ICE, or at =
least section 10 of ICE.
3) We specify that the media-loopback indication in SDP means the loopback-=
source supports receiving and ignoring STUN Binding Indications, so that fo=
llowing Section 10 of RFC 5245 the mirror MUST send STUN not no-op or comfo=
rt noise.
4) We put some text in the loopback-source about sources MUST support ignor=
ing received STUN Binding Indications, not depend on getting them, etc.
5) We fix all of Magnus' comments and publish this as an RFC before our Sun=
 becomes a red giant and dies out.
6) Profit?

-hadriel


From magnus.westerlund@ericsson.com  Thu Oct 27 01:56:03 2011
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Cc: "mmusic \(E-mail\)" <mmusic@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-media-loopback-16
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On 2011-10-25 20:01, Hadriel Kaplan wrote:
> 
> On Oct 21, 2011, at 5:25 AM, Magnus Westerlund wrote:
> 
>> 11. Section 5.3 "    Note: A loopback offer in a given media
>> description MUST NOT contain the standard mode attributes sendonly,
>> recvonly, sendrecv, or inactive. The loopback-mode attributes
>> (loopback-source and loopback-mirror) replace the standard
>> attributes."
>> 
>> This appears to be a strange MUST NOT which I think might
>> encounter issues. Considering that SIP proxies that looks into this
>> SDP requires the direction attribute to be present and adds it,
>> what happens now. And why isn't sendrecv and inactive possible
>> values? sendrecv appears to be requried for loopback to work, and
>> inactive would put media transport in an established session on
>> hold. Isn't that reasonable?
> 
> How about this: Since media loopback requires bidirectional RTP, its
> normal direction mode is "sendrecv"; the "sendrecv" direction
> attribute MAY be encoded in SDP or not, as per section 5.1 of
> [RFC3264], since it is implied by default.  If either the loopback
> source or mirror wish to disable loopback use during a session, the
> direction attribute "inactive" MUST be used as per [RFC3264].  The
> direction mode attributes "recvonly" and "sendonly" are incompatible
> with the loopback mechanism and MUST NOT be indicated when generating
> an SDP Offer or Answer.  When receiving an SDP Offer or Answer, if
> "recvonly" or "sendonly" is indicated for loopback, the SDP-receiving
> agent MAY treat it as a failure of the loopback negotiation or ignore
> the direction attribute.
> 

That is much better. I think the only "controversial" from my
perspective is the "ignore" part. It is clearly an error.

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From internet-drafts@ietf.org  Thu Oct 27 04:39:31 2011
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rtsp-nat-evaluation-04.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=
 Working Group of the IETF.

	Title           : The Evaluation of Different Network Addres Translator (N=
AT) Traversal Techniques for Media Controlled by Real-time Streaming Protoc=
ol (RTSP)
	Author(s)       : Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-evaluation-04.txt
	Pages           : 37
	Date            : 2011-10-27

   This document describes several Network Address Translator (NAT)
   traversal techniques that was considered to be used by Real-time
   Streaming Protocol (RTSP).  Each technique includes a description on
   how it would be used, the security implications of using it and any
   other deployment considerations it has.  There are also disussions on
   how NAT traversal techniques relates to firewalls and how each
   technique can be applied in different use cases.  These findings
   where used when selecting the NAT traversal for RTSP 2.0 standardized
   in the MMUSIC WG.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rtsp-nat-evaluation-0=
4.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-rtsp-nat-evaluation-04=
.txt

From internet-drafts@ietf.org  Thu Oct 27 04:43:42 2011
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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=
 Working Group of the IETF.

	Title           : A Network Address Translator (NAT) Traversal mechanism f=
or media controlled by Real-Time Streaming Protocol (RTSP)
	Author(s)       : Jeff Goldberg
                          Magnus Westerlund
                          Thomas Zeng
	Filename        : draft-ietf-mmusic-rtsp-nat-11.txt
	Pages           : 29
	Date            : 2011-10-27

   This document defines a solution for Network Address Translation
   (NAT) traversal for datagram based media streams setup 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 extra RTSP extensions and
   procedures.


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From magnus.westerlund@ericsson.com  Thu Oct 27 04:50:15 2011
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WG,

No changes only to keep the document alive. Same is true for the nat
evaluation draft.

Cheers

Magnus

On 2011-10-27 13:43, 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)
> 	Author(s)       : Jeff Goldberg
>                           Magnus Westerlund
>                           Thomas Zeng
> 	Filename        : draft-ietf-mmusic-rtsp-nat-11.txt
> 	Pages           : 29
> 	Date            : 2011-10-27
> 
>    This document defines a solution for Network Address Translation
>    (NAT) traversal for datagram based media streams setup 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 extra RTSP extensions and
>    procedures.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rtsp-nat-11.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-rtsp-nat-11.txt
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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Färögatan 6                | Mobile +46 73 0949079
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----------------------------------------------------------------------


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On 10/27/11 4:55 AM, Magnus Westerlund wrote:
> On 2011-10-25 20:01, Hadriel Kaplan wrote:
>>
>> On Oct 21, 2011, at 5:25 AM, Magnus Westerlund wrote:
>>
>>> 11. Section 5.3 "    Note: A loopback offer in a given media
>>> description MUST NOT contain the standard mode attributes sendonly,
>>> recvonly, sendrecv, or inactive. The loopback-mode attributes
>>> (loopback-source and loopback-mirror) replace the standard
>>> attributes."
>>>
>>> This appears to be a strange MUST NOT which I think might
>>> encounter issues. Considering that SIP proxies that looks into this
>>> SDP requires the direction attribute to be present and adds it,
>>> what happens now. And why isn't sendrecv and inactive possible
>>> values? sendrecv appears to be requried for loopback to work, and
>>> inactive would put media transport in an established session on
>>> hold. Isn't that reasonable?
>>
>> How about this: Since media loopback requires bidirectional RTP, its
>> normal direction mode is "sendrecv"; the "sendrecv" direction
>> attribute MAY be encoded in SDP or not, as per section 5.1 of
>> [RFC3264], since it is implied by default.  If either the loopback
>> source or mirror wish to disable loopback use during a session, the
>> direction attribute "inactive" MUST be used as per [RFC3264].  The
>> direction mode attributes "recvonly" and "sendonly" are incompatible
>> with the loopback mechanism and MUST NOT be indicated when generating
>> an SDP Offer or Answer.  When receiving an SDP Offer or Answer, if
>> "recvonly" or "sendonly" is indicated for loopback, the SDP-receiving
>> agent MAY treat it as a failure of the loopback negotiation or ignore
>> the direction attribute.
>>
>
> That is much better. I think the only "controversial" from my
> perspective is the "ignore" part. It is clearly an error.

Yeah. Normally we don't specify the behavior in error cases.
Doing a behavior is recommended that allows things to proceed in a 
useful way, it is in effect "allowing" the error.

(Specifying *which* error to report, when there is a choice, seems 
reasonable.)

	Thanks,
	Paul

> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On 10/27/11 4:10 AM, Hadriel Kaplan wrote:
> Howdy,
> Before you read the insanely long email I just sent, I have a proposed solution which may satisfy you, the authors, and the Working Group as a whole.  Or maybe this solution is what the loopback draft actually means and I'm just dumb as a stump.
>    
I don't think either is really the case (or at least it's far from 
obvious :-)


> As I understand it, your concern is we (1) don't have an RFC defining how NAT traversal works if not with ICE, (2) don't want to define a "primer" thingy just for this one loopback draft, and (3) are dizzy from of going around in circles. :)
>
>    
Right, and
(4) If we are not happy with ICE as our IETF NAT traversal mechanism, 
for whatever reason, then we as WG need to be explicit about that and 
figure out a way of addressing it in general - not just in the form of 
and as part of some loopback specific discussion (or we'll have another 
one of these threads when the next draft comes along that needs to work 
behind NATs, but doesn't really want to do ICE).

> Well... there is an escape clause in ICE RFC 5245, section 5.1, for what to do if there's an ICE mismatch or a received Offer doesn't indicate ICE.  In theory an ICE-supporting UA must follow that behavior.  The rationale for this is explained in section 18, with regards to ALGs and SBCs.
>
>    
Right - I think this was Emil's point as well (further noting that even 
when ICE is supported, a media latching SBC may lead to these procedures 
being invoked).

> Note that when the other side doesn't do ICE or there is an ALG, section 5.1 instructs the UA to do section 10.  Per section 10, if the peer supports ICE at all (even if ICE is mismatched and not being used), the local UA sends STUN Binding Indications as the keepalives.  If the peer doesn't support ICE at all, then what the local UA sends is its choice.  Examples the ICE RFC gives is Comfort Noise, or draft-ietf-avt-rtp-no-op (which I believe was written by someone named Flemming Andreasen :)
>
>    
:-)
> My point in all this is to propose the following high-level straw-man:
>
> 1) We use the STUN Binding Indications as our primer packet format, by referencing STUN RFC.
>    
Any reason we can't reference the ICE RFC and make it a "SHOULD" 
implement ?
> 2) We put in text that a UA loopback-mirror MUST either support ICE, or at least section 10 of ICE.
>    
Ok - I think we can reference ICE in 1) then (right ?)

> 3) We specify that the media-loopback indication in SDP means the loopback-source supports receiving and ignoring STUN Binding Indications, so that following Section 10 of RFC 5245 the mirror MUST send STUN not no-op or comfort noise.
> 4) We put some text in the loopback-source about sources MUST support ignoring received STUN Binding Indications, not depend on getting them, etc.
>    
This seems like a way forward that would keep us consistent with current 
ICE procedures. We probably need to explore it in more detail (and I'd 
welcome more comments on this as well), but it seems promising. One 
question is the frequency with which these STUN Binding Requests would 
be sent ?


> 5) We fix all of Magnus' comments and publish this as an RFC before our Sun becomes a red giant and dies out.
> 6) Profit?
>    

Let's explore the details further and get more comments on this.

Thanks

-- Flemming

> -hadriel
>
>
>    

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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Hi Hadriel

Per your other subsequent e-mail 
(http://www.ietf.org/mail-archive/web/mmusic/current/msg08979.html), 
I'll defer on this one. If your other proposal ends up not moving 
forward, then please resurrect this one again.

Thanks

-- Flemming


On 10/26/11 11:36 PM, Hadriel Kaplan wrote:
> Hi Flemming,
> Sorry about the length of this email.
> I think we are either talking past each other, or going in circles as you said.  :(
> It would probably be better to discuss this in person in Taipei, but here's an attempt at a reset:
>
> 1) My goal is to be enable a SIP-domain administrator to make loopback test-calls from a testing system (phone or server SIP UAC) to SIP clients; whether those SIP clients be behind NATs or not.  To achieve that goal, I believe we need an RFC that we can reference the SIP client vendors to implement against.
>
> 2) To make it useful, I believe such an RFC needs to:
> 	a) Be likely to be implemented: the easier it is to implement, the likelier to be implemented.
> 	b) Be as close to normal calls as possible, both from a signaling and media-path perspective: if regular calls use ICE, media-loopback calls need to use ICE; if regular calls don't use ICE, media-loopback calls should not either... because we don't want a different media-path used for them or else the purpose of media-loopback test calls is negated.
> 	c) Not require changing the deployed SIP and media-handling architecture, whatever it may be; media-loopback is a testing tool, not new service/revenue, so people won't be willing to change their whole infrastructure just to make a test call.
>
> 3) Current SIP-domain media-handling architectures differ, but we don't need to go define what all of them are or could be to enable media-loopback; all we have to do is point 2(b) above: keep them doing what they do for normal calls, except in a loopback mode.  The only problem with that is that in normal calls the SIP UA sends RTP without waiting to receive RTP, for a sendrecv mode call.  Because of RFC 4787, we know that if the UA is behind a NAT then not sending some packets out first means the loopback packets won't be let in. In order to make the loopback call appear as close to a normal call as possible, therefore, we need to have the UA send some packets out first. (luckily we have RFC 4787 or we'd be having an argument about having to publish that before publishing media-loopback! ;)
>
> 4) We don't need to describe latching to specify in a media-loopback RFC that a SIP UA needs to mimic what is does in a normal call.  Al we have to say is: use ICE if you support ICE and the incoming SDP offer indicated ICE as per RFC 5245, or just send some primer RTP packets if what you would do in a normal call is send RTP without ICE.  Frankly, it won't hurt anything if a SIP UAC *always* sends primer RTP packets in a loopback call, even if it also does ICE.  Whether NAT traversal works by latching, or by some wireshark script, or by reading tea leaves doesn't matter.  We just need to follow 2(b): act like a normal call as much as possible.
>
> 5) My original email was to indicate my belief that the change from draft-15 to draft-16 is a bad one: both because it makes the situation worse not better, and because I feel it was taken without real consensus in this WG.  I wouldn't care much about the lack of consensus for some editorial nit, but I think this is a big deployment-impacting change for a vote of 3 vs. 2 to be called "rough consensus".
>
> 6) The current loopback draft-16 says this and only this with respect to handling loopback calls and NATs:
> Section 5.7:
>      ICE/STUN/TURN provide a general solution to establishing media
>      sessions between entities that are behind NATs. Loopback sessions
>      that involve one or more end points behind NATs SHOULD use these
>      general solutions wherever possible.
>
> That's it, in the ENTIRE document.  That's approaching 'useless' on the Richter scale of RFC mandates.  How do you know if you're "behind NATs"?  If it's a "SHOULD", what's the exception/alternative?  What if you're not in "wherever possible"?
> The previous draft version-15, on the other hand, told you what to do if not ICE. We can argue about whether what it said to do was good or not, but at least it did something.
>
> -hadriel
>
>
>    

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Thread-Topic: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On Oct 27, 2011, at 9:40 PM, Flemming Andreasen wrote:

>> As I understand it, your concern is we (1) don't have an RFC defining ho=
w NAT traversal works if not with ICE, (2) don't want to define a "primer" =
thingy just for this one loopback draft, and (3) are dizzy from of going ar=
ound in circles. :)
>>  =20
> Right, and
> (4) If we are not happy with ICE as our IETF NAT traversal mechanism, for=
 whatever reason, then we as WG need to be explicit about that and figure o=
ut a way of addressing it in general - not just in the form of and as part =
of some loopback specific discussion (or we'll have another one of these th=
reads when the next draft comes along that needs to work behind NATs, but d=
oesn't really want to do ICE).

While I and some others have noted on multiple occasions that ICE isn't the=
 utopian solution it's cracked up to be, it appears so far we're still in t=
he minority in the WG.

I did put in text in:
http://tools.ietf.org/html/draft-boucadair-mmusic-ccap-00
in Appendix A.2, of an example why ICE can't be used for some people - but =
that's sort of an obscure reference because the point of the draft is about=
 IPv4/v6 not NATs.  There're other ICE-problem examples too, like in the En=
terprise market where the IT guys want to force the media to go through the=
ir VPNs or stay inside and never reach the public Internet.

But essentially media-loopback is the only case I've seen where there's a N=
AT problem to solve that isn't auto-handled by just normal RTP behavior and=
 latching.  The IPv4/v6 case isn't a NAT traversal problem per se, nor a la=
tching one, but more of a need to indicate multiple address families in SDP=
, competing with the desire by some others in the WG that a connectivity ch=
eck type thing happens because IPv6 isn't stable.


>> My point in all this is to propose the following high-level straw-man:
>>=20
>> 1) We use the STUN Binding Indications as our primer packet format, by r=
eferencing STUN RFC.
>>  =20
> Any reason we can't reference the ICE RFC and make it a "SHOULD" implemen=
t ?

How about this: "A UAC compliant with this document SHOULD implement [RFC52=
45], but MAY only implement section 10 of [RFC5245] with the additional rul=
es noted in section X of this document if it does not use ICE for normal me=
dia sessions."

And section X would have something like this:
When a UAS loopback mirror receives an SDP Offer with media-loopback, if th=
e UAS implements [RFC5245] then it follows ICE procedures, including fallba=
ck to just sending STUN Binding Indications per Section 10 of ICE if the re=
ceived Offer did not indicate ICE or there is an ICE mismatch condition.  I=
f the UAS does not implement [RFC5245], then it MUST at least follow the pr=
ocedures of Section 10 of [RFC5245] and send STUN Binding Indications.  In =
all cases, even if ICE is not actively used, the existence of the media-loo=
pback indication in SDP indicates the far-end is capable of receiving and i=
gnoring STUN Binding Indications, and the local UAS MUST send such as per S=
ection 10 of [RFC5245] as if the SDP contained a candidate line.  This is t=
o ensure some initial packets gets sent to the loopback source, to create N=
AT bindings or open Firewall pinholes along the path, since RTP is not gene=
rated by the loopback mirror until RTP is received.

>> 3) We specify that the media-loopback indication in SDP means the loopba=
ck-source supports receiving and ignoring STUN Binding Indications, so that=
 following Section 10 of RFC 5245 the mirror MUST send STUN not no-op or co=
mfort noise.
>> 4) We put some text in the loopback-source about sources MUST support ig=
noring received STUN Binding Indications, not depend on getting them, etc.
>>  =20
> This seems like a way forward that would keep us consistent with current =
ICE procedures. We probably need to explore it in more detail (and I'd welc=
ome more comments on this as well), but it seems promising. One question is=
 the frequency with which these STUN Binding Requests would be sent ?

The frequency is in section 10, but I assume you mean when "inactive" mode =
is used in media-loopback?  That's a tough one.  In a normal call there wou=
ld be no RTP in that mode, and since the tester may want to test that mode =
to see if it is causing problems then I think the mirror shouldn't send STU=
N even then.  :(

-hadriel


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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=
 Working Group of the IETF.

	Title           : Real Time Streaming Protocol 2.0 (RTSP)
	Author(s)       : Henning Schulzrinne
                          Anup Rao
                          Rob Lanphier
                          Magnus Westerlund
                          Martin Stiemerling
	Filename        : draft-ietf-mmusic-rfc2326bis-28.txt
	Pages           : 297
	Date            : 2011-10-28

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

   The Real Time Streaming Protocol, or RTSP, is an application-level
   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).


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-28.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=
 Working Group of the IETF.

	Title           : Session Description Protocol (SDP) Extension For Setting=
 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Publ=
ic Switched Telephone Network (PSTN)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
	Filename        : draft-ietf-mmusic-sdp-cs-09.txt
	Pages           : 30
	Date            : 2011-10-31

   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).


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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-media-capabilities-12.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=
 Working Group of the IETF.

	Title           : SDP Media Mapabilities Negotiation
	Author(s)       : Robert R Gilman
                          Roni Even
                          Flemming Andreasen
	Filename        : draft-ietf-mmusic-sdp-media-capabilities-12.txt
	Pages           : 50
	Date            : 2011-10-31

   Session Description Protocol (SDP) capability negotiation provides a
   general framework for indicating and negotiating capabilities in SDP.
   The base framework defines only capabilities for negotiating
   transport protocols and attributes.  In this document, we extend the
   framework by defining media capabilities that can be used to
   negotiate media types and their associated parameters.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-media-capabilitie=
s-12.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-media-capabilities=
-12.txt

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Cc: "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>, "Elwell, John" <john.elwell@siemens.com>, "mmusic \(E-mail\)" <mmusic@ietf.org>, Dan Wing <dwing@cisco.com>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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On 10/27/11 11:36 PM, Hadriel Kaplan wrote:
> On Oct 27, 2011, at 9:40 PM, Flemming Andreasen wrote:
>
>   =20
>>> As I understand it, your concern is we (1) don't have an RFC defining=
 how NAT traversal works if not with ICE, (2) don't want to define a "pri=
mer" thingy just for this one loopback draft, and (3) are dizzy from of g=
oing around in circles. :)
>>>
>>>       =20
>> Right, and
>> (4) If we are not happy with ICE as our IETF NAT traversal mechanism, =
for whatever reason, then we as WG need to be explicit about that and fig=
ure out a way of addressing it in general - not just in the form of and a=
s part of some loopback specific discussion (or we'll have another one of=
 these threads when the next draft comes along that needs to work behind =
NATs, but doesn't really want to do ICE).
>>     =20
> While I and some others have noted on multiple occasions that ICE isn't=
 the utopian solution it's cracked up to be, it appears so far we're stil=
l in the minority in the WG.
>
> I did put in text in:
> http://tools.ietf.org/html/draft-boucadair-mmusic-ccap-00
> in Appendix A.2, of an example why ICE can't be used for some people - =
but that's sort of an obscure reference because the point of the draft is=
 about IPv4/v6 not NATs.
Would it make sense to include this as part of John Elwells ICE draft ?

>   There're other ICE-problem examples too, like in the Enterprise marke=
t where the IT guys want to force the media to go through their VPNs or s=
tay inside and never reach the public Internet.
>
> But essentially media-loopback is the only case I've seen where there's=
 a NAT problem to solve that isn't auto-handled by just normal RTP behavi=
or and latching.  The IPv4/v6 case isn't a NAT traversal problem per se, =
nor a latching one, but more of a need to indicate multiple address famil=
ies in SDP, competing with the desire by some others in the WG that a con=
nectivity check type thing happens because IPv6 isn't stable.
>
>
>   =20
>>> My point in all this is to propose the following high-level straw-man=
:
>>>
>>> 1) We use the STUN Binding Indications as our primer packet format, b=
y referencing STUN RFC.
>>>
>>>       =20
>> Any reason we can't reference the ICE RFC and make it a "SHOULD" imple=
ment ?
>>     =20
> How about this: "A UAC compliant with this document SHOULD implement [R=
FC5245], but MAY only implement section 10 of [RFC5245] with the addition=
al rules noted in section X of this document if it does not use ICE for n=
ormal media sessions."
>
> And section X would have something like this:
> When a UAS loopback mirror receives an SDP Offer with media-loopback, i=
f the UAS implements [RFC5245] then it follows ICE procedures, including =
fallback to just sending STUN Binding Indications per Section 10 of ICE i=
f the received Offer did not indicate ICE or there is an ICE mismatch con=
dition.  If the UAS does not implement [RFC5245], then it MUST at least f=
ollow the procedures of Section 10 of [RFC5245] and send STUN Binding Ind=
ications.  In all cases, even if ICE is not actively used, the existence =
of the media-loopback indication in SDP indicates the far-end is capable =
of receiving and ignoring STUN Binding Indications, and the local UAS MUS=
T send such as per Section 10 of [RFC5245] as if the SDP contained a cand=
idate line.  This is to ensure some initial packets gets sent to the loop=
back source, to create NAT bindings or open Firewall pinholes along the p=
ath, since RTP is not generated by the loopback mirror until RTP is recei=
ved.
>
>   =20

I'd like to hear more people commenting on this. My understanding of ICE =

Section 5.1/10 in this case is that it's for keep-alives and hence you=20
wouldn't send these messages initially (which is not to say that=20
couldn't be changed when used with loopback - the question is how it=20
differs from "real" ICE then).

>>> 3) We specify that the media-loopback indication in SDP means the loo=
pback-source supports receiving and ignoring STUN Binding Indications, so=
 that following Section 10 of RFC 5245 the mirror MUST send STUN not no-o=
p or comfort noise.
>>> 4) We put some text in the loopback-source about sources MUST support=
 ignoring received STUN Binding Indications, not depend on getting them, =
etc.
>>>
>>>       =20
>> This seems like a way forward that would keep us consistent with curre=
nt ICE procedures. We probably need to explore it in more detail (and I'd=
 welcome more comments on this as well), but it seems promising. One ques=
tion is the frequency with which these STUN Binding Requests would be sen=
t ?
>>     =20
> The frequency is in section 10, but I assume you mean when "inactive" m=
ode is used in media-loopback?  That's a tough one.  In a normal call the=
re would be no RTP in that mode, and since the tester may want to test th=
at mode to see if it is causing problems then I think the mirror shouldn'=
t send STUN even then.  :(
>   =20
I believe Section 10 would only apply to keep-alives in the scenario=20
being considered here (and hence inactive mode is fine). The issue, I=20
believe, is that you need these STUN binding packets sent initially to=20
punch a hole through the NAT, and I don't believe the current RFC 5245=20
by itself will give you that.

-- Flemming


> -hadriel
>
>
>   =20


From pkyzivat@alum.mit.edu  Mon Oct 31 12:30:28 2011
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Subject: Re: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-cs-09.txt
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Thanks. This clears up a lot.
In looking over the changes, I noted the following other things:

Syntax (in 5.2.3.1 and 5.6):

As written, this will allow individual correlation mechanisms to be 
repeated, possibly with differing values. E.g.

a=cs-correlation: dtmf:123 dtmf:456

I'm not sure, but I *doubt* you intend to allow that.
To address it, you have a couple of choices:

- fiddle the syntax to disallow it.
- put in textual rules to forbid it.

In section 5.2.3.2, you have:

    For example, the numbers +358-9-123-4567 and 09-123-4567
    could be considered as the same number.

This could be interpreted as indicating that the international E.164 
number might contain separator characters. Based on your syntax, the one 
in the cs-correlation attribute can't have them. I'm not sure if the one 
received as callerid info from the PSTN can have them or not.

In section 5.2.3.3:

The protocol discriminator must be specified *somewhere*.
Either you can specify a fixed one as part of this mechanism, or else 
you must allow it to be specified as part of the value.

In section 5.5.4:

"If either party wishes to drop and reestablish an existing call..."

This implies that one must use one o/a exchange to drop the existing 
call and another one to establish it again. That shouldn't be necessary. 
Sending the same sort of offer one would do to establish the initial 
session (including a=connection:new) should be fine.

	Thanks,
	Paul


On 10/31/11 8:06 AM, Simo.Veikkolainen@nokia.com wrote:
> Hi,
> We have submitted a new version of the sdp-cs draft (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-09.txt), addressing Paul's comments.
>
> Below is a list of changes in the -09 version:
>
> Number format
> -	Use the term âinternational E.164 numberâ throughout the document
> -	Require a â+â sign in all international E.164 numbers
>
> Added the following lines to addresstype specification in 5.2.1:
> â
>      When the<addrtype>  is PSTN, the connection address is defined
>      as follows:
>
>      o  an e164-address
>
>      When the<addrtype>  is "-", the connection address is defined
>      as follows:
>
>      o  the value "-", signifying that the address is unknown.
> OR
>      o  any syntactically valid value, which is to be ignored..
> â
>
> Syntax of cs-correlation changed in 5.2.3.1
>
> UUIE protocol discriminator
> -	Clarified that only the âUser informationâ part of UUIE is to be included as the value of uuie-mech, and used for comparing the incoming UUIE
> -	Added an OPEN ISSUE still here
>
> Moved the usage of âa=connection:existingâ to 5.5.4 Modifying the session.
>
> Added text in 5.5.4 on how connection reuse works.
>
> E.164 syntax in 5.6
> - we didnât include a reference to RFC3966 since we donât want to include the visual separator, and otherwise the benefit of having the RFC as reference would be small
>
> Changed the number in the examples. +15551234 ->  +35891234567 (I know this at least is a syntactically correct representation ď
>
> Things which did not make this version:
>
> TODO: Add example with CS audio/video stream.
> TODO: Security considerations needs more work based on Flemming's earlier comments (including caller's E164 number in SDP).
> TODO: Clarification on the mechanism works with 3PCC.
>
>
> Thanks
> Simo
>
>
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] On Behalf Of ext internet-drafts@ietf.org
> Sent: 31 October, 2011 14:03
> To: i-d-announce@ietf.org
> Cc: mmusic@ietf.org
> Subject: I-D Action: draft-ietf-mmusic-sdp-cs-09.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 Up Audio and Video Media Streams Over Circuit-Switched Bearers In The Public Switched Telephone Network (PSTN)
> 	Author(s)       : Miguel A. Garcia-Martin
>                            Simo Veikkolainen
> 	Filename        : draft-ietf-mmusic-sdp-cs-09.txt
> 	Pages           : 30
> 	Date            : 2011-10-31
>
>     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).
>
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-09.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-cs-09.txt
> _______________________________________________
> 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
> _______________________________________________
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> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Date: Tue, 01 Nov 2011 07:13:21 +0100
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Cc: Dan Wing <dwing@cisco.com>, "Elwell, John" <john.elwell@siemens.com>, "<draft-ietf-mmusic-media-loopback@tools.ietf.org>" <draft-ietf-mmusic-media-loopback@tools.ietf.org>, "mmusic \(E-mail\)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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Can rfc6263   RTP Keep-alive be referenced for the non-ICE case?

/Gunnar


Flemming Andreasen skrev 2011-10-31 20:26:
>
>
> On 10/27/11 11:36 PM, Hadriel Kaplan wrote:
>> On Oct 27, 2011, at 9:40 PM, Flemming Andreasen wrote:
>>
>>>> As I understand it, your concern is we (1) don't have an RFC 
>>>> defining how NAT traversal works if not with ICE, (2) don't want to 
>>>> define a "primer" thingy just for this one loopback draft, and (3) 
>>>> are dizzy from of going around in circles. :)
>>>>
>>> Right, and
>>> (4) If we are not happy with ICE as our IETF NAT traversal 
>>> mechanism, for whatever reason, then we as WG need to be explicit 
>>> about that and figure out a way of addressing it in general - not 
>>> just in the form of and as part of some loopback specific discussion 
>>> (or we'll have another one of these threads when the next draft 
>>> comes along that needs to work behind NATs, but doesn't really want 
>>> to do ICE).
>> While I and some others have noted on multiple occasions that ICE 
>> isn't the utopian solution it's cracked up to be, it appears so far 
>> we're still in the minority in the WG.
>>
>> I did put in text in:
>> http://tools.ietf.org/html/draft-boucadair-mmusic-ccap-00
>> in Appendix A.2, of an example why ICE can't be used for some people 
>> - but that's sort of an obscure reference because the point of the 
>> draft is about IPv4/v6 not NATs.
> Would it make sense to include this as part of John Elwells ICE draft ?
>
>>   There're other ICE-problem examples too, like in the Enterprise 
>> market where the IT guys want to force the media to go through their 
>> VPNs or stay inside and never reach the public Internet.
>>
>> But essentially media-loopback is the only case I've seen where 
>> there's a NAT problem to solve that isn't auto-handled by just normal 
>> RTP behavior and latching.  The IPv4/v6 case isn't a NAT traversal 
>> problem per se, nor a latching one, but more of a need to indicate 
>> multiple address families in SDP, competing with the desire by some 
>> others in the WG that a connectivity check type thing happens because 
>> IPv6 isn't stable.
>>
>>
>>>> My point in all this is to propose the following high-level straw-man:
>>>>
>>>> 1) We use the STUN Binding Indications as our primer packet format, 
>>>> by referencing STUN RFC.
>>>>
>>> Any reason we can't reference the ICE RFC and make it a "SHOULD" 
>>> implement ?
>> How about this: "A UAC compliant with this document SHOULD implement 
>> [RFC5245], but MAY only implement section 10 of [RFC5245] with the 
>> additional rules noted in section X of this document if it does not 
>> use ICE for normal media sessions."
>>
>> And section X would have something like this:
>> When a UAS loopback mirror receives an SDP Offer with media-loopback, 
>> if the UAS implements [RFC5245] then it follows ICE procedures, 
>> including fallback to just sending STUN Binding Indications per 
>> Section 10 of ICE if the received Offer did not indicate ICE or there 
>> is an ICE mismatch condition.  If the UAS does not implement 
>> [RFC5245], then it MUST at least follow the procedures of Section 10 
>> of [RFC5245] and send STUN Binding Indications.  In all cases, even 
>> if ICE is not actively used, the existence of the media-loopback 
>> indication in SDP indicates the far-end is capable of receiving and 
>> ignoring STUN Binding Indications, and the local UAS MUST send such 
>> as per Section 10 of [RFC5245] as if the SDP contained a candidate 
>> line.  This is to ensure some initial packets gets sent to the 
>> loopback source, to create NAT bindings or open Firewall pinholes 
>> along the path, since RTP is not generated by the loopback mirror 
>> until RTP is received.
>>
>
> I'd like to hear more people commenting on this. My understanding of 
> ICE Section 5.1/10 in this case is that it's for keep-alives and hence 
> you wouldn't send these messages initially (which is not to say that 
> couldn't be changed when used with loopback - the question is how it 
> differs from "real" ICE then).
>
>>>> 3) We specify that the media-loopback indication in SDP means the 
>>>> loopback-source supports receiving and ignoring STUN Binding 
>>>> Indications, so that following Section 10 of RFC 5245 the mirror 
>>>> MUST send STUN not no-op or comfort noise.
>>>> 4) We put some text in the loopback-source about sources MUST 
>>>> support ignoring received STUN Binding Indications, not depend on 
>>>> getting them, etc.
>>>>
>>> This seems like a way forward that would keep us consistent with 
>>> current ICE procedures. We probably need to explore it in more 
>>> detail (and I'd welcome more comments on this as well), but it seems 
>>> promising. One question is the frequency with which these STUN 
>>> Binding Requests would be sent ?
>> The frequency is in section 10, but I assume you mean when "inactive" 
>> mode is used in media-loopback?  That's a tough one.  In a normal 
>> call there would be no RTP in that mode, and since the tester may 
>> want to test that mode to see if it is causing problems then I think 
>> the mirror shouldn't send STUN even then.  :(
> I believe Section 10 would only apply to keep-alives in the scenario 
> being considered here (and hence inactive mode is fine). The issue, I 
> believe, is that you need these STUN binding packets sent initially to 
> punch a hole through the NAT, and I don't believe the current RFC 5245 
> by itself will give you that.
>
> -- Flemming
>
>
>> -hadriel
>>
>>
>
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