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From: <mohamed.boucadair@orange.com>
To: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Date: Thu, 1 Dec 2011 15:05:14 +0100
Thread-Topic: [MMUSIC] Do We Need an ICE Alternative ?
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Subject: Re: [MMUSIC] Do We Need an ICE Alternative ?
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Dear Flemming, all,

As an input to this discussion, I just submitted an I-D identifying some IP=
v6 use cases which motivate the need to define a new SDP attribute. These u=
se cases cover both multicast and unicast.

The I-D is available at: http://tools.ietf.org/html/draft-boucadair-mmusic-=
ipv6-use-cases-00.

Comments and questions are welcome.

Cheers,
Med
=20

> -----Message d'origine-----
> De : mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]=20
> De la part de Flemming Andreasen
> Envoy=E9 : mardi 15 novembre 2011 13:49
> =C0 : mmusic
> Objet : [MMUSIC] Do We Need an ICE Alternative ?
>=20
> Greetings
>=20
> Over the past several years, the MMUSIC WG has developed and=20
> standardized ICE (RFC 5245) as the mechanism to enable NAT=20
> traversal and=20
> IPv4/IPv6 transition for Offer/Answer based protocols (such as SIP).=20
> Alternative NAT traversal mechanisms exist, and others have been=20
> proposed, however they have either been limited to specific=20
> deployment=20
> topologies (e.g. symmetric RTP or STUN by itself) or required=20
> some level=20
> of network support (e.g. an ALG of some sort). Similarly, for=20
> IPv4/IPv6=20
> transition, RFC 4091/4092 (ANAT) defined an alternative solution that=20
> was deprecated following MMUSIC and SIPPING WG discssion and=20
> consensus=20
> at IETF68 (in 2007):
>=20
>      http://www.ietf.org/proceedings/68/minutes/mmusic.txt
>      http://www.ietf.org/proceedings/68/minutes/sipping.txt
>=20
> Several years have passed and we have been getting more=20
> implementation=20
> and deployment experience with ICE. However, we have also seen=20
> suggestions for alternative solutions, with recent examples including=20
> the ALTC draft for IPv4/IPv6 transition and the latching=20
> mechanism for=20
> NAT traversal:
>=20
>      Link to ALTC draft:
>      https://datatracker.ietf.org/doc/draft-boucadair-mmusic-altc/
>=20
>      Link to latching mechanism (Section 5.7 in old version=20
> of loopback=20
> draft):
>      http://tools.ietf.org/id/draft-ietf-mmusic-media-loopback-15.txt
>=20
>=20
> Alternatives, that generally have a more limited scope than ICE, keep=20
> popping up every now and then, and it seems that at least some set of=20
> people are not happy with the current ICE solution. So, we=20
> would like to=20
> understand from the working group if there are issues with=20
> ICE and if we=20
> need to explore alternative solutions for NAT traversal and/or for=20
> IPv4/IPv6 transition.
>=20
> Our goal as a standards organization is to enable interoperable=20
> products, and having multiple different standards for the=20
> same problem=20
> or multiple different standards for specific instances of a general=20
> problem is not conducive to that. If we have to revisit one of the=20
> standards for a given problem space, we ought to be careful and=20
> understand what are the issues that we have with that standard.=20
> Furthermore, if we are to come with alternative solutions for that=20
> problem space, we have to ensure a clear understanding of what has=20
> changed since the last time we created the standard, what are the new=20
> set of requirements to satisfy, and which of the old requirements no=20
> longer apply.
>=20
> So, we would like to sense the WG for opinions in this matter
>=20
> Thanks
>=20
>      Flemming and Miguel (MMUSIC chairs)
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> =

From miguel.a.garcia@ericsson.com  Thu Dec  1 10:01:53 2011
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To: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc4566bis-04.txt
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Hi Shinji,

my replies (as an individual). I am also copying Ali, since he is editing 
the draft.

On 30/11/2011 10:34, OKUMURA Shinji wrote:
> Hi,
>
> I have some questions concerning draft-ietf-mmusic-rfc4566bis-04.
>
> [Page 25]
>      "If the<proto>  sub-field is "udp" the<fmt>  sub-fields MUST
>      reference a media type describing the format under the "audio",
>      "video", "text", "application", or "message" top-level media
>      types.  The media type registration SHOULD define the packet
>      format for use with UDP transport."
>
> Q1. In this document, there are some "media type",
>      "top-level media content types", "top-level content type"
>      and "top-level media types".
>
>      Which do these mean "MIME media type" or "SDP media type"?
>      Or do the both of two have the same meaning?
>      I think these are ambiguous expressions.


I think in the past the original term was MIME type, but since it was not 
only used for e-mail, that old term has been nowadays being replaced by 
"media types".

A media type is composed of a "top-level media type", such as "audio", or 
"video", and a "subtype", such as "G729" or "iLBC". Sometimes, the 
"top-level media type" is also known as "Content type", or more simply, 
"type".

I agree with you that the draft should be coherent with the terminology. 
I suggest to avoid the word "MIME". And I suggest to pick one term for 
the "top-level media type", I don't have a strong preference for one.


>
> Q2. As for the "audio", "video", "text", "application", or "message"
>      top-level media types, where is the definition of the packet
>      format for use with UDP transport?
>      Why is it necessary for new media types?

I believe "audio" and "video" usually use the RTP/AVP as the transport 
protocol. So, these days we don't send "audio" or "video" directly over 
UDP, something that will imply to define the packet format. Instead, we 
send audio and video over RTP, following the audio video profile defined 
in RFC 3551.

Other media types, such as "message", may use other transport protocols. 
This is the case when MSRP is used as a transport protocol.

This is necessary for any media type, it is not enough to say that you 
are going to send the media over UDP, but you also need to define the 
packet format.

>
> [Page 36]
> 8.2.3.  Media Formats ("fmt")
>      "For the "udp" protocol, new formats SHOULD be registered.  Use of an
>      existing media subtype for the format is encouraged.  If no media
>      subtype exists, it is RECOMMENDED that a suitable one be registered
>      through the IETF process [RFC4288] by production of, or reference to,
>      a standards-track RFC that defines the transport protocol for the
>      format."
>
> Q3. When a new media format(such as t38) is registered,
>      why should we define a new format for "udp" protocol?
>      IMHO all media format don't need the "udp" protocol.

I understand the sentence a bit different. I think the sentence wants to 
say that if you are going to write "udp" in the <proto> field of the "m=" 
line, then you need to register the format, it is hasn't been done it 
yet. So, it does not say that every format should use UDP, but the other 
way round.

BR,

       Miguel




>
> Regards,
> Shinji
>
> internet-drafts@ietf.org
> Mon, 24 Oct 2011 10:57:30 -0700
>> 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			: 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
>>
>> 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-rfc4566bis-04.txt
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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

From simon.perreault@viagenie.ca  Thu Dec  1 10:18:33 2011
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] Do We Need an ICE Alternative ?
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On 2011-12-01 09:05, mohamed.boucadair@orange.com wrote:
> As an input to this discussion, I just submitted an I-D identifying some IPv6
> use cases which motivate the need to define a new SDP attribute. These use
> cases cover both multicast and unicast.
>
> The I-D is available at:
> http://tools.ietf.org/html/draft-boucadair-mmusic-ipv6-use-cases-00.

VERY interesting read, thank you.

I think one important thing missing is one small paragraph for each use case 
starting with "ICE(-lite) does not solve the problem because ..."

This information is necessary to conclude that an ICE alternative would be needed.

Thanks,
Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

From iesg-secretary@ietf.org  Thu Dec  1 10:33:38 2011
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Subject: [MMUSIC] Protocol Action: 'TCP Candidates with Interactive Connectivity	Establishment (ICE)' to Proposed Standard	(draft-ietf-mmusic-ice-tcp-16.txt)
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The IESG has approved the following document:
- 'TCP Candidates with Interactive Connectivity Establishment (ICE)'
  (draft-ietf-mmusic-ice-tcp-16.txt) as a Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Robert Sparks.

A URL of this Internet Draft is:
http://datatracker.ietf.org/doc/draft-ietf-mmusic-ice-tcp/




Technical Summary

Interactive Connectivity Establishment (ICE) defines a mechanism for
NAT traversal for multimedia communication protocols based on the
offer/answer model of session negotiation. ICE works by providing a
set of candidate transport addresses for each media stream, which are
then validated with peer-to-peer connectivity checks based on Session
Traversal Utilities for NAT (STUN). ICE provides a general framework
for describing candidates, but only defines UDP-based transport
protocols. This specification extends ICE to TCP-based media,
including the ability to offer a mix of TCP and UDP-based candidates
for a single stream.

Working Group Summary

This document is a product of the MMUSIC WG. The document has been in
progress for a while with significant Working Group interest,
contribution and review. There are no controversial issues.

Document Quality

The document has received significant review and the quality is good.
The chairs are aware of multiple implementations of the mechanism. 

Personnel

  Flemming Andreasen is the document's shepherd.
  Gonzalo Camarillo is the responsible AD.

From mohamed.boucadair@orange.com  Thu Dec  1 11:00:15 2011
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From: <mohamed.boucadair@orange.com>
To: Simon Perreault <simon.perreault@viagenie.ca>
Date: Thu, 1 Dec 2011 20:00:11 +0100
Thread-Topic: [MMUSIC] Do We Need an ICE Alternative ?
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: [MMUSIC] RE :  Do We Need an ICE Alternative ?
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Hi Simon,

Thank you for reading the draft.

For sure, I can add some text for that purpose: the main argument is ICE-li=
te MUST respond  to STUN messages. Would it be sufficient to add a pointer =
to http://tools.ietf.org/html/draft-hutton-mmusic-icemicrolite-02 which exp=
lains why a lite "ICE-lite" is needed?=20

I will need then to add some text to explain why micro-lite is not sufficie=
nt:
* There is no need to adopt the ICE synthax since there is no plan to have =
ICE in those deployments
* micro -lite uses some dummy values ufrag and password but this is not com=
pliant with RFC5245 (The ice-ufrag and ice-pwd attributes MUST be chosen ra=
ndomly)
* the SDP offer is lengthy: risk for fragmented SIP packets

Cheers,
Med

________________________________________
De : Simon Perreault [simon.perreault@viagenie.ca]
Date d'envoi : jeudi 1 d=E9cembre 2011 19:18
=C0 : BOUCADAIR Mohamed OLNC/NAD/TIP
Cc : Flemming Andreasen; mmusic
Objet : Re: [MMUSIC] Do We Need an ICE Alternative ?

On 2011-12-01 09:05, mohamed.boucadair@orange.com wrote:
> As an input to this discussion, I just submitted an I-D identifying some =
IPv6
> use cases which motivate the need to define a new SDP attribute. These us=
e
> cases cover both multicast and unicast.
>
> The I-D is available at:
> http://tools.ietf.org/html/draft-boucadair-mmusic-ipv6-use-cases-00.

VERY interesting read, thank you.

I think one important thing missing is one small paragraph for each use cas=
e
starting with "ICE(-lite) does not solve the problem because ..."

This information is necessary to conclude that an ICE alternative would be =
needed.

Thanks,
Simon
--
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
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On 2011-12-01 14:00, mohamed.boucadair@orange.com wrote:
> the main argument is ICE-lite
> MUST respond  to STUN messages.

Right, this is what I expected.

Why is it a problem that ICE-lite MUST respond to STUN messages?

It has been stated before that the SHA-1 computation is too heavy. Is this the 
problem you're facing?

> Would it be sufficient to add a pointer to
> http://tools.ietf.org/html/draft-hutton-mmusic-icemicrolite-02 which explains
> why a lite "ICE-lite" is needed?

Well, it doesn't exactly answer the question. It says:

    However support for ICE or ICE-
    Lite, which is a subset of ICE, is over complex for scenarios in
    which a transition mechanism is needed without the need to use ICE
    for NAT traversal and connectivity checking.

ICE-Lite too complex? Really? What part is complex?

ICE-Lite is basically ANAT++ bundled with a STUN server.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
STUN/TURN server               --> http://numb.viagenie.ca

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From: <mohamed.boucadair@orange.com>
To: Simon Perreault <simon.perreault@viagenie.ca>
Date: Fri, 2 Dec 2011 09:13:01 +0100
Thread-Topic: RE : [MMUSIC] Do We Need an ICE Alternative ?
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Cc: Flemming Andreasen <fandreas@cisco.com>, mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] RE :  Do We Need an ICE Alternative ?
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Re-,

What is needed for the use cases described in the I-D is an ANAT++ which is=
 simple, backward compatible, does not increase the length of SDP bodies an=
d does not require connectivity checks at all (some implementers already re=
ported performance issues related to SHA-1 computation, issues with timers,=
 issue with some services such as Fax, interaction between RTP and ICE, etc=
.). This mechanism must avoid session failure when legacy service elements =
are involved.

BTW, for the multicast case, if more than one receiver are ICE-enabled, wha=
t would be their behaviour?

Cheers,
Med=20

> -----Message d'origine-----
> De : Simon Perreault [mailto:simon.perreault@viagenie.ca]=20
> Envoy=E9 : jeudi 1 d=E9cembre 2011 20:15
> =C0 : BOUCADAIR Mohamed OLNC/NAD/TIP
> Cc : Flemming Andreasen; mmusic
> Objet : Re: RE : [MMUSIC] Do We Need an ICE Alternative ?
>=20
> On 2011-12-01 14:00, mohamed.boucadair@orange.com wrote:
> > the main argument is ICE-lite
> > MUST respond  to STUN messages.
>=20
> Right, this is what I expected.
>=20
> Why is it a problem that ICE-lite MUST respond to STUN messages?
>=20
> It has been stated before that the SHA-1 computation is too=20
> heavy. Is this the=20
> problem you're facing?
>=20
> > Would it be sufficient to add a pointer to
> >=20
> http://tools.ietf.org/html/draft-hutton-mmusic-icemicrolite-02
>  which explains
> > why a lite "ICE-lite" is needed?
>=20
> Well, it doesn't exactly answer the question. It says:
>=20
>     However support for ICE or ICE-
>     Lite, which is a subset of ICE, is over complex for scenarios in
>     which a transition mechanism is needed without the need to use ICE
>     for NAT traversal and connectivity checking.
>=20
> ICE-Lite too complex? Really? What part is complex?
>=20
> ICE-Lite is basically ANAT++ bundled with a STUN server.
>=20
> Simon
> --=20
> DTN made easy, lean, and smart --> http://postellation.viagenie.ca
> NAT64/DNS64 open-source        --> http://ecdysis.viagenie.ca
> STUN/TURN server               --> http://numb.viagenie.ca
> =

From shinji.okumura@softfront.jp  Fri Dec  2 02:10:17 2011
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Hi Miguel,

thank you so much for your reply.


"Miguel A. Garcia" <Miguel.A.Garcia@ericsson.com>
Thu, 1 Dec 2011 19:01:46 +0100
>Hi Shinji,
>
>my replies (as an individual). I am also copying Ali, since he is editing 
>the draft.
>
>On 30/11/2011 10:34, OKUMURA Shinji wrote:
>> Hi,
>>
>> I have some questions concerning draft-ietf-mmusic-rfc4566bis-04.
>>
>> [Page 25]
>>      "If the<proto>  sub-field is "udp" the<fmt>  sub-fields MUST
>>      reference a media type describing the format under the "audio",
>>      "video", "text", "application", or "message" top-level media
>>      types.  The media type registration SHOULD define the packet
>>      format for use with UDP transport."
>>
>> Q1. In this document, there are some "media type",
>>      "top-level media content types", "top-level content type"
>>      and "top-level media types".
>>
>>      Which do these mean "MIME media type" or "SDP media type"?
>>      Or do the both of two have the same meaning?
>>      I think these are ambiguous expressions.
>
>
>I think in the past the original term was MIME type, but since it was not 
>only used for e-mail, that old term has been nowadays being replaced by 
>"media types".
>
>A media type is composed of a "top-level media type", such as "audio", or 
>"video", and a "subtype", such as "G729" or "iLBC". Sometimes, the 
>"top-level media type" is also known as "Content type", or more simply, 
>"type".
>
>I agree with you that the draft should be coherent with the terminology. 
>I suggest to avoid the word "MIME". And I suggest to pick one term for 
>the "top-level media type", I don't have a strong preference for one.

I think it may not be necessary to avoid the word "MIME".
IIUC some of "media type" in this draft mean top-level "MIME"
media type that is registered with IANA.

>> Q2. As for the "audio", "video", "text", "application", or "message"
>>      top-level media types, where is the definition of the packet
>>      format for use with UDP transport?
>>      Why is it necessary for new media types?
>
>I believe "audio" and "video" usually use the RTP/AVP as the transport 
>protocol. So, these days we don't send "audio" or "video" directly over 
>UDP, something that will imply to define the packet format. Instead, we 
>send audio and video over RTP, following the audio video profile defined 
>in RFC 3551.
>
>Other media types, such as "message", may use other transport protocols. 
>This is the case when MSRP is used as a transport protocol.

therefore, as for the "audio", "video", "text", "application",
or "message", the packet format for the <proto> sub-field "udp"
is not defined.

As a consequence I think the 1st sentence is needless.

>This is necessary for any media type, it is not enough to say that you 
>are going to send the media over UDP, but you also need to define the 
>packet format.

When we register new <fmt> sub-fields for the <proto> sub-field
"udp", it is necessary to define the packet format.
But, when we register only SDP media type without "proto" nor
"fmt", it is meaningless and impossible to define the packet
format for use with <proto> "udp".

As a consequence I think the 2nd sentence is needless.

Basically the packet format depends on both of <proto> and <fmt>.
And there is no reason for giving special treatment only to "udp".

>> [Page 36]
>> 8.2.3.  Media Formats ("fmt")
>>      "For the "udp" protocol, new formats SHOULD be registered.  Use of an
>>      existing media subtype for the format is encouraged.  If no media
>>      subtype exists, it is RECOMMENDED that a suitable one be registered
>>      through the IETF process [RFC4288] by production of, or reference to,
>>      a standards-track RFC that defines the transport protocol for the
>>      format."
>>
>> Q3. When a new media format(such as t38) is registered,
>>      why should we define a new format for "udp" protocol?
>>      IMHO all media format don't need the "udp" protocol.
>
>I understand the sentence a bit different. I think the sentence wants to 
>say that if you are going to write "udp" in the <proto> field of the "m=" 
>line, then you need to register the format, it is hasn't been done it 
>yet. So, it does not say that every format should use UDP, but the other 
>way round.

When new <fmt> media format is used over the <fmt> field "udp", 
it is necessary. I see.

Regards,
Shinji

>BR,
>
>       Miguel
>>
>> internet-drafts@ietf.org
>> Mon, 24 Oct 2011 10:57:30 -0700
>>> 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			: 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
>>>
>>> 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-rfc4566bis-04.txt
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>-- 
>Miguel A. Garcia
>+34-91-339-3608
>Ericsson Spain

From gsalguei@cisco.com  Sun Dec  4 17:26:20 2011
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To: Muthu ArulMozhi Perumal (mperumal) <mperumal@cisco.com>
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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-media-path-middleboxes-03
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Muthu -=20

Please forgive the delayed response, but I have been buried under work =
with some of my new responsibilities at work.  I am getting ready to =
release the new version of this draft and had a quick question about =
your one major comment.


On Mar 29, 2011, at 9:21 AM, Muthu ArulMozhi Perumal (mperumal) wrote:

> Overall this draft looks to be in a good shape.
> =20
> I've one major comment. Similar to REC #2 in section 7, it would be =
desirable to have a recommendation that the middlebox present on the =
media path allow at least a nominal amount of traffic to be exchanged =
between endpoints for a small duration of time (say, one RTT) after the =
middlebox receives a message from theMIDCOM agent that indicates the =
media session being terminated. This will ensure that any transit =
signaling packets on the media path exchanged during the session =
termination, like RTCP Bye packets, pass through the middle box.
> =20

I think I understand the intent of this recommendation and think it =
might be useful to address specifically.  Where did you think this text =
should be added ?

Regards,

Gonzalo





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<html><head><base href=3D"x-msg://71/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; ">Muthu -&nbsp;<div><br></div><div>Please forgive the =
delayed response, but I have been buried under work with some of my new =
responsibilities at work. &nbsp;I am getting ready to release the new =
version of this draft and had a quick question about your one major =
comment.</div><div><br></div><div><br><div><div>On Mar 29, 2011, at 9:21 =
AM, Muthu ArulMozhi Perumal (mperumal) wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Calibri; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"Section1" =
style=3D"page: Section1; "><div style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; "><span style=3D"font-size: 10pt; =
font-family: 'Courier New'; ">Overall this draft looks to be in a good =
shape.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><span style=3D"font-size: =
10pt; font-family: 'Courier New'; "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 10pt; font-family: 'Courier =
New'; ">I've one major comment. Similar to<span =
class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"SpellE">REC</span><span =
class=3D"Apple-converted-space">&nbsp;</span>#2 in section 7, it would =
be desirable to have a recommendation that the<span =
class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"SpellE">middlebox</span><span =
class=3D"Apple-converted-space">&nbsp;</span>present on the media path =
allow at least a nominal amount of traffic to be exchanged between =
endpoints for a small duration of time (say, one<span =
class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"SpellE">RTT</span>) after the<span =
class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"SpellE">middlebox</span><span =
class=3D"Apple-converted-space">&nbsp;</span>receives a message from =
the<span class=3D"SpellE">MIDCOM</span><span =
class=3D"Apple-converted-space">&nbsp;</span>agent that indicates the =
media session being terminated. This will ensure that any transit =
signaling packets on the media path exchanged during the session =
termination, like<span class=3D"Apple-converted-space">&nbsp;</span><span =
class=3D"SpellE">RTCP</span><span =
class=3D"Apple-converted-space">&nbsp;</span>Bye packets, pass through =
the middle box.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><span style=3D"font-size: =
10pt; font-family: 'Courier New'; =
"><o:p>&nbsp;</o:p></span></div></div></div></span></blockquote><div><br><=
/div>I think I understand the intent of this recommendation and think it =
might be useful to address specifically. &nbsp;Where did you think this =
text should be added =
?</div><div><br></div><div>Regards,</div><div><br></div><div>Gonzalo</div>=
<div><br><br></div><div><br></div><br></div></body></html>=

--Apple-Mail-2--1003689832--

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Thread-Topic: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Jochen Issing" <iss@ieee.org>, <payload@ietf.org>, <mmusic@ietf.org>
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Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Jochen,

I am not that familiar with the details of the variation modes. If you
want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
HE-AACv2, it seems you could include multiple payload types in the
m-line, and then specify the corresponding fmtp parameters for each. Is
there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
within a single payload type?

Charles

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
Behalf Of Jochen Issing
> Sent: Wednesday, November 30, 2011 7:18 AM
> To: payload@ietf.org; mmusic@ietf.org
> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>=20
> Dear experts,
>=20
> I need to double-apologize before posting our findings:
> 1) for cross-posting, as I do not know, where this issue has its exact
home, which is hopefully not a
> response killer.
> 2) for the long mail, even though I tried to keep it short.
>=20
> Now, here's the actual request:
>=20
> We are referring to RFC3640 SDP fields only, as this is most often the
stumbling block. Let's use the
> following example:
>=20
>    m=3Daudio 49170 RTP/AVP 96
>    a=3Drtpmap:96 mpeg4-generic/48000/2
>    a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
config=3D1190; sizeLength=3D13;
> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>=20
> The fields of interest are in the fmtp line: 'profile-level-id' and
'config'.
> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
types of sub codecs, which are
> identified using the Audio Object Type (AOT). The AOT and some
additional (codec internal) information
> is stored in the Audio Specific Config (ASC), which is conveyed in the
'config' field.
> The 'profile-level-id' points to a profile with a certain level. The
profile comprises a set of one or
> more AOTs and the level restricts the stream to a maximum sample rate
and/or number channels. The
> profile-level-ids, as specified in ISO 14496-3, are partly
hierarchical and most often ambiguous.
>=20
> Suppose the classic SIP use case that two clients want to exchange
their codec capabilities.
> The first client specifies some payload types in the SDP part of SIP
INVITE which can differ in their
> profile-level-id and/or config.
> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
signal these codecs/AOTs he could
> either indicate all formats with the same profile-level-id (e.g., 48),
which includes all these AOTs
> or he could specify profile-level-ids, that are most 'narrow' to the
actual specified AOT or he could
> even mix this principle.
> On the other hand, the ASC might use implicit signaling, explicit
backwards-compatible, or explicit
> non-backwards-compatible.
> Bobs main problem would be to determine, whether the profile-level-id
was chosen to really identify
> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
if the profile-level-id was
> just brawling about the encoders feature.
>=20
> To clean up this mess of opportunities, we would like to specify a
non-ambiguous way to utilize the
> profile-level-id and config fields appropriately and provide a guide
on how to accomplish clean and
> easier signalization and thus improve interoperability.
>=20
>=20
> But first of all we would like to make sure, that you agree to this
problem statement or do we miss
> something here?
> Comments are as always highly appreciated.
>=20
> Best Regards,
> --
> jochen
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From gunnar.hellstrom@omnitor.se  Mon Dec  5 13:19:09 2011
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Subject: Re: [MMUSIC] WGLC for draft-ietf-mmusic-media-loopback-16
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This is a multi-part message in MIME format.
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I rechecked the proposal to use RFC 6263 to create and maintain NAT 
bindings for loopback if ICE is not used.

It is apparent from the introduction of RFC 6263 that it is intended to 
both create initial NAT mapping and maintain it.

Here is a sentence about the intention of RFC 6263:
".....This introduces a derived requirement for applications engaged in 
a multimedia session involving NAT traversal: they need to generate a 
minimum of flow activity in order to create NAT mappings and maintain 
them. When applied to applications using the Real-time Transport 
Protocol (RTP) [RFC3550 <./rfc3550>], the RTP media stream packets 
themselves normally fulfill this requirement. However, there exist some 
cases where RTP does not generate the minimum required flow activity. "

The text continues indicating that RFC 6263 solves these situations.

So I think a reference to RFC 6263 would safely resolve the issue about 
the missing "priming" mechanism raisede by Hadriel on 5, October.


Gunnar
--------------------------------------------------------------------------

Flemming Andreasen skrev 2011-11-02 21:54:
> That might be another option (multiplex RTP and RTCP with RTCP as the 
> keep-alive mechanism), however I think you still have the issue of 
> sending the intial packets to punch a hole.
>
> -- Flemming
>
> On 11/1/11 2:13 AM, Gunnar Hellström wrote:
>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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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    I rechecked the proposal to use RFC 6263 to create and maintain NAT
    bindings for loopback if ICE is not used.<br>
    <br>
    It is apparent from the introduction of RFC 6263 that it is intended
    to both create initial NAT mapping and maintain it. <br>
    <br>
    Here is a sentence about the intention of RFC 6263:<br>
    ".....This introduces a derived requirement for applications engaged
    in a multimedia session involving NAT traversal: they need to
    generate a minimum of flow activity in order to create NAT mappings
    and maintain them. When applied to applications using the Real-time
    Transport Protocol (RTP) [<a title="&quot;RTP: A Transport Protocol
      for Real-Time Applications&quot;" href="./rfc3550">RFC3550</a>],
    the RTP media stream packets themselves normally fulfill this
    requirement. However, there exist some cases where RTP does not
    generate the minimum required flow activity.
    "<br>
    <br>
    The text continues indicating that RFC 6263 solves these situations.<br>
    <br>
    So I think a reference to RFC 6263 would safely resolve the issue
    about the missing "priming" mechanism raisede by Hadriel on 5,
    October.<br>
    <br>
    <br>
    Gunnar<br>
--------------------------------------------------------------------------<br>
    <br>
    Flemming Andreasen skrev 2011-11-02 21:54:
    <blockquote cite="mid:4EB1ADFD.4030501@cisco.com" type="cite">That
      might be another option (multiplex RTP and RTCP with RTCP as the
      keep-alive mechanism), however I think you still have the issue of
      sending the intial packets to punch a hole.
      <br>
      <br>
      -- Flemming
      <br>
      <br>
      On 11/1/11 2:13 AM, Gunnar Hellstr&ouml;m wrote:
      <br>
      <blockquote type="cite">Can rfc6263&nbsp;&nbsp; RTP Keep-alive be referenced
        for the non-ICE case?
        <br>
        <br>
        /Gunnar
        <br>
        <br>
        <br>
        Flemming Andreasen skrev 2011-10-31 20:26:
        <br>
        <blockquote type="cite">
          <br>
          <br>
          On 10/27/11 11:36 PM, Hadriel Kaplan wrote:
          <br>
          <blockquote type="cite">On Oct 27, 2011, at 9:40 PM, Flemming
            Andreasen wrote:
            <br>
            <br>
            <blockquote type="cite">
              <blockquote type="cite">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. :)
                <br>
                <br>
              </blockquote>
              Right, and
              <br>
              (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).
              <br>
            </blockquote>
            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.
            <br>
            <br>
            I did put in text in:
            <br>
            <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-boucadair-mmusic-ccap-00">http://tools.ietf.org/html/draft-boucadair-mmusic-ccap-00</a>
            <br>
            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.
            <br>
          </blockquote>
          Would it make sense to include this as part of John Elwells
          ICE draft ?
          <br>
          <br>
          <blockquote type="cite">&nbsp; 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.
            <br>
            <br>
            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.&nbsp; 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.
            <br>
            <br>
            <br>
            <blockquote type="cite">
              <blockquote type="cite">My point in all this is to propose
                the following high-level straw-man:
                <br>
                <br>
                1) We use the STUN Binding Indications as our primer
                packet format, by referencing STUN RFC.
                <br>
                <br>
              </blockquote>
              Any reason we can't reference the ICE RFC and make it a
              "SHOULD" implement ?
              <br>
            </blockquote>
            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."
            <br>
            <br>
            And section X would have something like this:
            <br>
            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.&nbsp; 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.&nbsp;
            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.&nbsp; 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.
            <br>
            <br>
          </blockquote>
          <br>
          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).
          <br>
          <br>
          <blockquote type="cite">
            <blockquote type="cite">
              <blockquote type="cite">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.
                <br>
                4) We put some text in the loopback-source about sources
                MUST support ignoring received STUN Binding Indications,
                not depend on getting them, etc.
                <br>
                <br>
              </blockquote>
              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 ?
              <br>
            </blockquote>
            The frequency is in section 10, but I assume you mean when
            "inactive" mode is used in media-loopback?&nbsp; That's a tough
            one.&nbsp; 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.&nbsp; :(
            <br>
          </blockquote>
          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.
          <br>
          <br>
          -- Flemming
          <br>
          <br>
          <br>
          <blockquote type="cite">-hadriel
            <br>
            <br>
            <br>
          </blockquote>
          <br>
          _______________________________________________
          <br>
          mmusic mailing list
          <br>
          <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
          <br>
          <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
          <br>
        </blockquote>
        <br>
      </blockquote>
      _______________________________________________
      <br>
      mmusic mailing list
      <br>
      <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
      <br>
      <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
      <br>
    </blockquote>
  </body>
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From iss@IEEE.ORG  Mon Dec  5 14:50:55 2011
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To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Cc: mmusic@ietf.org, payload@ietf.org
Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Charles,

thanks for your response! However, the issue we identified is somewhat =
different:

The problem already takes place with a single fmtp line:
For example, one client indicates that it can (decode and) encode an AAC =
stream using
the config field in fmtp. The other side, receiving this information, =
then must
check its decoder, if it can decode the ASC and needs to setup its =
encoder,
so that it uses the very same ASC.

The problem occurs at the encoder of the caller (the receiver of the =
first SDP):
It might not be possible with reasonable effort to get an encoder =
creating the
same ASC as given in SDP (encoders are in general not initialized using =
an ASC).

I hope, this explained our problem much clearer.
Best,

jochen

On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:

> Hi Jochen,
>=20
> I am not that familiar with the details of the variation modes. If you
> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. =
and
> HE-AACv2, it seems you could include multiple payload types in the
> m-line, and then specify the corresponding fmtp parameters for each. =
Is
> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
> within a single payload type?
>=20
> Charles
>=20
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Jochen Issing
>> Sent: Wednesday, November 30, 2011 7:18 AM
>> To: payload@ietf.org; mmusic@ietf.org
>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>=20
>> Dear experts,
>>=20
>> I need to double-apologize before posting our findings:
>> 1) for cross-posting, as I do not know, where this issue has its =
exact
> home, which is hopefully not a
>> response killer.
>> 2) for the long mail, even though I tried to keep it short.
>>=20
>> Now, here's the actual request:
>>=20
>> We are referring to RFC3640 SDP fields only, as this is most often =
the
> stumbling block. Let's use the
>> following example:
>>=20
>>   m=3Daudio 49170 RTP/AVP 96
>>   a=3Drtpmap:96 mpeg4-generic/48000/2
>>   a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
> config=3D1190; sizeLength=3D13;
>> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>>=20
>> The fields of interest are in the fmtp line: 'profile-level-id' and
> 'config'.
>> In contrast to other codecs, e.g. G.XXX, AAC integrates many =
different
> types of sub codecs, which are
>> identified using the Audio Object Type (AOT). The AOT and some
> additional (codec internal) information
>> is stored in the Audio Specific Config (ASC), which is conveyed in =
the
> 'config' field.
>> The 'profile-level-id' points to a profile with a certain level. The
> profile comprises a set of one or
>> more AOTs and the level restricts the stream to a maximum sample rate
> and/or number channels. The
>> profile-level-ids, as specified in ISO 14496-3, are partly
> hierarchical and most often ambiguous.
>>=20
>> Suppose the classic SIP use case that two clients want to exchange
> their codec capabilities.
>> The first client specifies some payload types in the SDP part of SIP
> INVITE which can differ in their
>> profile-level-id and/or config.
>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
> signal these codecs/AOTs he could
>> either indicate all formats with the same profile-level-id (e.g., =
48),
> which includes all these AOTs
>> or he could specify profile-level-ids, that are most 'narrow' to the
> actual specified AOT or he could
>> even mix this principle.
>> On the other hand, the ASC might use implicit signaling, explicit
> backwards-compatible, or explicit
>> non-backwards-compatible.
>> Bobs main problem would be to determine, whether the profile-level-id
> was chosen to really identify
>> the necessary (and applied) AAC AOTs or if he could rely on the ASC =
or
> if the profile-level-id was
>> just brawling about the encoders feature.
>>=20
>> To clean up this mess of opportunities, we would like to specify a
> non-ambiguous way to utilize the
>> profile-level-id and config fields appropriately and provide a guide
> on how to accomplish clean and
>> easier signalization and thus improve interoperability.
>>=20
>>=20
>> But first of all we would like to make sure, that you agree to this
> problem statement or do we miss
>> something here?
>> Comments are as always highly appreciated.
>>=20
>> Best Regards,
>> --
>> jochen
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic

--=20
jochen


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From: "Severa, Michael J (Mike)" <mike.severa@alcatel-lucent.com>
To: Jochen Issing <iss@IEEE.ORG>, "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
Date: Mon, 5 Dec 2011 17:28:12 -0600
Thread-Topic: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "payload@ietf.org" <payload@ietf.org>
Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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My personal feeling is that the ASC need not be identical in syntax, just i=
n semantics (one of the unfortunate aspects of AAC-related signaling is tha=
t there are a number of ways to signal the same bitstream). Another unfortu=
nate aspect is that one of the options is to signal implicitly, which essen=
tially means dont signal at all, rather 'detect' at decode time. That is ob=
viously not useful for a capabilities exchange step.=20

I think there might also be other open questions surrounding ambiguities in=
 signaling AAC-LC/HE-AAC that should be worked through to generate a best o=
r acceptable practices approach.=20

Two of the ones that jump out at me are:
- the question of what timescale from the rtpmap line means when you intend=
 to be using HE-AAC in a 3640 environment (3GPP has explicitly indicated th=
is should be sampling frequency of the underlying, 'core' AAC-LC portion of=
 an HE-AAC stream. I assume this is the same semantic in 3640 (though I did=
nt notice it specified explicitly)). This question is probably only relevan=
t in cases where the goal is to assume HE-AAC (something the multiple fmtp =
line suggestion might be implying).=20
- how HE-AAC can be detected in implicit mode during negotiation (again, 3G=
PP added an SBR-Enabled flag to the SDP to allow HE-AAC to be detected ahea=
d of time even with implicit signaling, im not sure there is an analogous s=
olution for 3640).=20

Both of these questions are probably irrelevant if implicit signaling is di=
sallowed for HE-AAC (though in the first case that would require parsing th=
e ASC during negotiation). Using multiple fmtp lines might be one way to 'd=
isable' the implicit signaling requirement (since the backwards compatibili=
ty concern associated with implicit signaling can be dealt with by an expli=
cit AAC-LC-only ftmp (and rtpmap)line).=20

Sorry if that's a tangent.=20

thanks,=20

Mike

________________________________________
From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] On Behalf Of Jochen=
 Issing [iss@IEEE.ORG]
Sent: Monday, December 05, 2011 2:50 PM
To: Charles Eckel (eckelcu)
Cc: mmusic@ietf.org; payload@ietf.org
Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues

Hi Charles,

thanks for your response! However, the issue we identified is somewhat diff=
erent:

The problem already takes place with a single fmtp line:
For example, one client indicates that it can (decode and) encode an AAC st=
ream using
the config field in fmtp. The other side, receiving this information, then =
must
check its decoder, if it can decode the ASC and needs to setup its encoder,
so that it uses the very same ASC.

The problem occurs at the encoder of the caller (the receiver of the first =
SDP):
It might not be possible with reasonable effort to get an encoder creating =
the
same ASC as given in SDP (encoders are in general not initialized using an =
ASC).

I hope, this explained our problem much clearer.
Best,

jochen

On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:

> Hi Jochen,
>
> I am not that familiar with the details of the variation modes. If you
> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
> HE-AACv2, it seems you could include multiple payload types in the
> m-line, and then specify the corresponding fmtp parameters for each. Is
> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
> within a single payload type?
>
> Charles
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Jochen Issing
>> Sent: Wednesday, November 30, 2011 7:18 AM
>> To: payload@ietf.org; mmusic@ietf.org
>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>
>> Dear experts,
>>
>> I need to double-apologize before posting our findings:
>> 1) for cross-posting, as I do not know, where this issue has its exact
> home, which is hopefully not a
>> response killer.
>> 2) for the long mail, even though I tried to keep it short.
>>
>> Now, here's the actual request:
>>
>> We are referring to RFC3640 SDP fields only, as this is most often the
> stumbling block. Let's use the
>> following example:
>>
>>   m=3Daudio 49170 RTP/AVP 96
>>   a=3Drtpmap:96 mpeg4-generic/48000/2
>>   a=3Dfmtp:96 streamtype=3D5; profile-level-id=3D15; mode=3DAAC-hbr;
> config=3D1190; sizeLength=3D13;
>> indexLength=3D3; indexDeltaLength=3D3; constantDuration=3D1024
>>
>> The fields of interest are in the fmtp line: 'profile-level-id' and
> 'config'.
>> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
> types of sub codecs, which are
>> identified using the Audio Object Type (AOT). The AOT and some
> additional (codec internal) information
>> is stored in the Audio Specific Config (ASC), which is conveyed in the
> 'config' field.
>> The 'profile-level-id' points to a profile with a certain level. The
> profile comprises a set of one or
>> more AOTs and the level restricts the stream to a maximum sample rate
> and/or number channels. The
>> profile-level-ids, as specified in ISO 14496-3, are partly
> hierarchical and most often ambiguous.
>>
>> Suppose the classic SIP use case that two clients want to exchange
> their codec capabilities.
>> The first client specifies some payload types in the SDP part of SIP
> INVITE which can differ in their
>> profile-level-id and/or config.
>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
> signal these codecs/AOTs he could
>> either indicate all formats with the same profile-level-id (e.g., 48),
> which includes all these AOTs
>> or he could specify profile-level-ids, that are most 'narrow' to the
> actual specified AOT or he could
>> even mix this principle.
>> On the other hand, the ASC might use implicit signaling, explicit
> backwards-compatible, or explicit
>> non-backwards-compatible.
>> Bobs main problem would be to determine, whether the profile-level-id
> was chosen to really identify
>> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
> if the profile-level-id was
>> just brawling about the encoders feature.
>>
>> To clean up this mess of opportunities, we would like to specify a
> non-ambiguous way to utilize the
>> profile-level-id and config fields appropriately and provide a guide
> on how to accomplish clean and
>> easier signalization and thus improve interoperability.
>>
>>
>> But first of all we would like to make sure, that you agree to this
> problem statement or do we miss
>> something here?
>> Comments are as always highly appreciated.
>>
>> Best Regards,
>> --
>> jochen
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic

--
jochen

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

From iss@ieee.org  Mon Dec  5 15:50:14 2011
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Cc: "payload@ietf.org" <payload@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
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Hi Mike,

as far as I understand the AAC codecs, ASCs of the same AOT (and 
channels and sample rate) are not necessarily 'compatible' to each 
other, especially when it comes to more recent codecs, like AAC-ELD, 
where critical information might be included in the ASC. Thus, for a 
uniform usage of ASCs, it should be bit-exact. I'm afraid the caller 
needs to add a payload type if he does not manage to reproduce the ASC 
indicated by the callee, which would be then used in one direction only.

Regarding the signalization of SBR and PS (HE-AAC/HE-AACv2) you are 
right, that's another issue. However, with presence of SIP/SDP and 
offer/answer functionality, implicit and backwards-compatible 
signalization do not make sense at all IMHO. The legacy receiver can 
simply disregard the unsupported codecs. Anyhow, this is neither stated 
anywhere.

Best,
jochen

Severa, Michael J (Mike) wrote:
> My personal feeling is that the ASC need not be identical in syntax, just in semantics (one of the unfortunate aspects of AAC-related signaling is that there are a number of ways to signal the same bitstream). Another unfortunate aspect is that one of the options is to signal implicitly, which essentially means dont signal at all, rather 'detect' at decode time. That is obviously not useful for a capabilities exchange step.
>
> I think there might also be other open questions surrounding ambiguities in signaling AAC-LC/HE-AAC that should be worked through to generate a best or acceptable practices approach.
>
> Two of the ones that jump out at me are:
> - the question of what timescale from the rtpmap line means when you intend to be using HE-AAC in a 3640 environment (3GPP has explicitly indicated this should be sampling frequency of the underlying, 'core' AAC-LC portion of an HE-AAC stream. I assume this is the same semantic in 3640 (though I didnt notice it specified explicitly)). This question is probably only relevant in cases where the goal is to assume HE-AAC (something the multiple fmtp line suggestion might be implying).
> - how HE-AAC can be detected in implicit mode during negotiation (again, 3GPP added an SBR-Enabled flag to the SDP to allow HE-AAC to be detected ahead of time even with implicit signaling, im not sure there is an analogous solution for 3640).
>
> Both of these questions are probably irrelevant if implicit signaling is disallowed for HE-AAC (though in the first case that would require parsing the ASC during negotiation). Using multiple fmtp lines might be one way to 'disable' the implicit signaling requirement (since the backwards compatibility concern associated with implicit signaling can be dealt with by an explicit AAC-LC-only ftmp (and rtpmap)line).
>
> Sorry if that's a tangent.
>
> thanks,
>
> Mike
>
> ________________________________________
> From: mmusic-bounces@ietf.org [mmusic-bounces@ietf.org] On Behalf Of Jochen Issing [iss@IEEE.ORG]
> Sent: Monday, December 05, 2011 2:50 PM
> To: Charles Eckel (eckelcu)
> Cc: mmusic@ietf.org; payload@ietf.org
> Subject: Re: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>
> Hi Charles,
>
> thanks for your response! However, the issue we identified is somewhat different:
>
> The problem already takes place with a single fmtp line:
> For example, one client indicates that it can (decode and) encode an AAC stream using
> the config field in fmtp. The other side, receiving this information, then must
> check its decoder, if it can decode the ASC and needs to setup its encoder,
> so that it uses the very same ASC.
>
> The problem occurs at the encoder of the caller (the receiver of the first SDP):
> It might not be possible with reasonable effort to get an encoder creating the
> same ASC as given in SDP (encoders are in general not initialized using an ASC).
>
> I hope, this explained our problem much clearer.
> Best,
>
> jochen
>
> On 05.12.2011, at 20:02, Charles Eckel (eckelcu) wrote:
>
>> Hi Jochen,
>>
>> I am not that familiar with the details of the variation modes. If you
>> want to specify different fmtp parameters for AAC-LC vs. HE-AAC vs. and
>> HE-AACv2, it seems you could include multiple payload types in the
>> m-line, and then specify the corresponding fmtp parameters for each. Is
>> there a need to support, for example, AAC-LC, HE-AACC, and HE-AACv2
>> within a single payload type?
>>
>> Charles
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Jochen Issing
>>> Sent: Wednesday, November 30, 2011 7:18 AM
>>> To: payload@ietf.org; mmusic@ietf.org
>>> Subject: [MMUSIC] RFC3640 (AAC) and SIP offer/answer issues
>>>
>>> Dear experts,
>>>
>>> I need to double-apologize before posting our findings:
>>> 1) for cross-posting, as I do not know, where this issue has its exact
>> home, which is hopefully not a
>>> response killer.
>>> 2) for the long mail, even though I tried to keep it short.
>>>
>>> Now, here's the actual request:
>>>
>>> We are referring to RFC3640 SDP fields only, as this is most often the
>> stumbling block. Let's use the
>>> following example:
>>>
>>>    m=audio 49170 RTP/AVP 96
>>>    a=rtpmap:96 mpeg4-generic/48000/2
>>>    a=fmtp:96 streamtype=5; profile-level-id=15; mode=AAC-hbr;
>> config=1190; sizeLength=13;
>>> indexLength=3; indexDeltaLength=3; constantDuration=1024
>>>
>>> The fields of interest are in the fmtp line: 'profile-level-id' and
>> 'config'.
>>> In contrast to other codecs, e.g. G.XXX, AAC integrates many different
>> types of sub codecs, which are
>>> identified using the Audio Object Type (AOT). The AOT and some
>> additional (codec internal) information
>>> is stored in the Audio Specific Config (ASC), which is conveyed in the
>> 'config' field.
>>> The 'profile-level-id' points to a profile with a certain level. The
>> profile comprises a set of one or
>>> more AOTs and the level restricts the stream to a maximum sample rate
>> and/or number channels. The
>>> profile-level-ids, as specified in ISO 14496-3, are partly
>> hierarchical and most often ambiguous.
>>> Suppose the classic SIP use case that two clients want to exchange
>> their codec capabilities.
>>> The first client specifies some payload types in the SDP part of SIP
>> INVITE which can differ in their
>>> profile-level-id and/or config.
>>> For instance, client bob supports AAC-LC, HE-AAC and HE-AACv2. To
>> signal these codecs/AOTs he could
>>> either indicate all formats with the same profile-level-id (e.g., 48),
>> which includes all these AOTs
>>> or he could specify profile-level-ids, that are most 'narrow' to the
>> actual specified AOT or he could
>>> even mix this principle.
>>> On the other hand, the ASC might use implicit signaling, explicit
>> backwards-compatible, or explicit
>>> non-backwards-compatible.
>>> Bobs main problem would be to determine, whether the profile-level-id
>> was chosen to really identify
>>> the necessary (and applied) AAC AOTs or if he could rely on the ASC or
>> if the profile-level-id was
>>> just brawling about the encoders feature.
>>>
>>> To clean up this mess of opportunities, we would like to specify a
>> non-ambiguous way to utilize the
>>> profile-level-id and config fields appropriately and provide a guide
>> on how to accomplish clean and
>>> easier signalization and thus improve interoperability.
>>>
>>>
>>> But first of all we would like to make sure, that you agree to this
>> problem statement or do we miss
>>> something here?
>>> Comments are as always highly appreciated.
>>>
>>> Best Regards,
>>> --
>>> jochen
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>
> --
> jochen
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

-- 
jochen

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Subject: [MMUSIC] Please help me on solving following Issues related to RTP based IMS VOIP Session.
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Content-Type: text/plain; charset=ISO-8859-1

Hi,

I am facing following Issues in RTP based IMS VOIP Session:

a) Previous VT/VOIP Call Media Session RTP packets flowing into current
VT/VOIP Call Media Session due to Network Congestion/Delay.

b) Media Packets received have unknown SSRC than the other Media packets
due to eavesdropper in the network.

How to ignore Unknown SSRC Packets/Previous Call Media Packets from playing?

-- 
*Thanks & Regards,
R.Ramesh Kumar
*

--14dae9340c47cfc64404b36c394e
Content-Type: text/html; charset=ISO-8859-1

<p>Hi,</p>
<p>I am facing following Issues in RTP based IMS VOIP Session:</p>
<p>a) Previous VT/VOIP Call Media Session RTP packets flowing into current VT/VOIP Call Media Session due to Network Congestion/Delay.</p>
<p>b) Media Packets received have unknown SSRC than the other Media packets due to eavesdropper in the network.</p>
<p>How to ignore Unknown SSRC Packets/Previous Call Media Packets from playing?<br clear="all"><br>-- <br><font style="BACKGROUND-COLOR: #ffffff" color="#000000"><em>Thanks &amp; Regards,<br>R.Ramesh Kumar<br></em></font><br>
</p>

--14dae9340c47cfc64404b36c394e--

From ciphor@starv.tv  Tue Dec  6 22:54:57 2011
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Subject: Re: [MMUSIC] Please help me on solving following Issues related to RTP based IMS VOIP Session.
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Just drop the packets with unknown SSRC before the audio/video decoder.

2011/12/6 ramesh kumar <ramesh3184@gmail.com>

> Hi,
>
> I am facing following Issues in RTP based IMS VOIP Session:
>
> a) Previous VT/VOIP Call Media Session RTP packets flowing into current
> VT/VOIP Call Media Session due to Network Congestion/Delay.
>
> b) Media Packets received have unknown SSRC than the other Media packets
> due to eavesdropper in the network.
>
> How to ignore Unknown SSRC Packets/Previous Call Media Packets from
> playing?
>
> --
> *Thanks & Regards,
> R.Ramesh Kumar
> *
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--bcaec548a063633bb904b37b070f
Content-Type: text/html; charset=ISO-8859-1

Just drop the packets with unknown SSRC before the audio/video decoder.<br><br><div class="gmail_quote">2011/12/6 ramesh kumar <span dir="ltr">&lt;<a href="mailto:ramesh3184@gmail.com">ramesh3184@gmail.com</a>&gt;</span><br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><p>Hi,</p>
<p>I am facing following Issues in RTP based IMS VOIP Session:</p>
<p>a) Previous VT/VOIP Call Media Session RTP packets flowing into current VT/VOIP Call Media Session due to Network Congestion/Delay.</p>
<p>b) Media Packets received have unknown SSRC than the other Media packets due to eavesdropper in the network.</p>
<p>How to ignore Unknown SSRC Packets/Previous Call Media Packets from playing?<br clear="all"><font color="#888888"><br>-- <br><font style="BACKGROUND-COLOR:#ffffff" color="#000000"><em>Thanks &amp; Regards,<br>R.Ramesh Kumar<br>

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

--bcaec548a063633bb904b37b070f--

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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: Incoming liaison statement from MMUSIC
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Hi all,
mmusic has received an incoming liaison from MPEG, indicating that they have
started on "codec independent media description code points".  My
understanding of this work is that these code points may be represented in
SDP.
My personal viewpoint is that this may well be sound architectural work, but
it may be better conducted here in mmusic, at least when it comes to SDP
representations and the related IANA work.  However, I know nothing about
this work beyond what's written in the statement, which is why I used "soft"
language.  Perhaps some of you can shed light into what's really going on.
The assorted avt WGs are probably also affected, as some of the proposed
code points sound like they are being described in payload formats, RTP
extensions and perhaps even in xrblock.  Let me suggest to use only the
music list for discussions, as only mmusic is being addressed in the
statement.
As it is short, I have attached the statement in word format to this email.
It will also appear shortly in the IETF liaison database.
Regards,
Stephan



--B_3406090145_2760267
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div>mmusic has=
 received an incoming liaison from MPEG, indicating that they have started o=
n "codec independent media description code points". &nbsp;My understanding =
of this work is that these code points may be represented in SDP.</div><div>=
My personal viewpoint is that this may well be sound architectural work, but=
 it may be better conducted here in mmusic, at least when it comes to SDP re=
presentations and the related IANA work. &nbsp;However,&nbsp;I know nothing =
about this work beyond what's written in the statement, which is why I used =
"soft" language. &nbsp;Perhaps some of you can shed light into what's really=
 going on.</div><div>The assorted avt WGs are probably also affected, as som=
e of the proposed code points sound like they are being described in payload=
 formats, RTP extensions and perhaps even in xrblock. &nbsp;Let me suggest t=
o use only the music list for discussions, as only mmusic is being addressed=
 in the statement.</div><div>As it is short, I have attached the statement i=
n word format to this email. &nbsp;It will also appear shortly in the IETF l=
iaison database.</div><div>Regards,</div><div>Stephan</div></body></html>

--B_3406090145_2760267--


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--B_3406090145_2752175--



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Subject: Re: [MMUSIC] [AVTCORE] Incoming liaison statement from MMUSIC
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Hi Stephan

I can explain a bit.  We noticed in systems work at MPEG two things
a) we wanted to say something about the signal that was being compressed (audio or video) that was nothing to do with the compression, and we ended up having to 'pick' a codec document and point at it, because the codes we wanted were always documented with codecs.
b) the codes are also copied from spec. to spec. in many cases (so they happen to have values having the same meanings); in a few cases, each codec has 'invented its own' and the codes needlessly differ from spec. to spec.

Examples: the 'classic three' values used to describe video color: color_primaries, transfer_characteristics, matrix_coefficients.  These are copied from spec. to spec., with the more recent specs having more values. Recently there was a mistake noticed in an equation, and I think it had to be corrected in several places.  Another example is 'frame packing' (stereo packed into a mono video stream); the specs actually don't always align on the definitions here.

The variation of semantics for some points could be a problem; different values for the same semantics is a nuisance, e.g. when wanting to be codec-independent or when transcoding. Different numbers of values for the same code in different specs is a nuisance, and raises questions, and so on.

The basic idea is to collect a whole bunch of these, document them in one place, and remove the definitions from a whole bunch of specs.  So, for example, in a video spec. one might be able to say "this 8-bit field carries a color_primaries value as defined in [[this new spec]]; if absent, the default value is 23".  Correspondingly, in say SDP, one might be able to say "if this field is 0, the video is mono; otherwise, it carries a frame_packing values as defined in [[this new spec]]".

Makes sense?


On Dec 7, 2011, at 8:08 , Stephan Wenger wrote:

> Hi all,
> mmusic has received an incoming liaison from MPEG, indicating that they have started on "codec independent media description code points".  My understanding of this work is that these code points may be represented in SDP.
> My personal viewpoint is that this may well be sound architectural work, but it may be better conducted here in mmusic, at least when it comes to SDP representations and the related IANA work.  However, I know nothing about this work beyond what's written in the statement, which is why I used "soft" language.  Perhaps some of you can shed light into what's really going on.
> The assorted avt WGs are probably also affected, as some of the proposed code points sound like they are being described in payload formats, RTP extensions and perhaps even in xrblock.  Let me suggest to use only the music list for discussions, as only mmusic is being addressed in the statement.
> As it is short, I have attached the statement in word format to this email.  It will also appear shortly in the IETF liaison database.
> Regards,
> Stephan
> <29n124561.doc>_______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt

David Singer
Multimedia and Software Standards, Apple Inc.


--Boundary_(ID_CtRC7gm2oU3HeqUBm5UV3A)
Content-type: text/html; CHARSET=US-ASCII
Content-transfer-encoding: quoted-printable

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi =
Stephan<div><br></div><div>I can explain a bit. &nbsp;We noticed in =
systems work at MPEG two things</div><div>a) we wanted to say something =
about the signal that was being compressed (audio or video) that was =
nothing to do with the compression, and we ended up having to 'pick' a =
codec document and point at it, because the codes we wanted were always =
documented with codecs.</div><div>b) the codes are also copied from =
spec. to spec. in many cases (so they happen to have values having the =
same meanings); in a few cases, each codec has 'invented its own' and =
the codes needlessly differ from spec. to =
spec.</div><div><br></div><div>Examples: the 'classic three' values used =
to describe video color: color_primaries, transfer_characteristics, =
matrix_coefficients. &nbsp;These are copied from spec. to spec., with =
the more recent specs having more values. Recently there was a mistake =
noticed in an equation, and I think it had to be corrected in several =
places. &nbsp;Another example is 'frame packing' (stereo packed into a =
mono video stream); the specs actually don't always align on the =
definitions here.</div><div><br></div><div>The variation of semantics =
for some points could be a problem; different values for the same =
semantics is a nuisance, e.g. when wanting to be codec-independent or =
when transcoding. Different numbers of values for the same code in =
different specs is a nuisance, and raises questions, and so =
on.</div><div><br></div><div>The basic idea is to collect a whole bunch =
of these, document them in one place, and remove the definitions from a =
whole bunch of specs. &nbsp;So, for example, in a video spec. one might =
be able to say "this 8-bit field carries a color_primaries value as =
defined in [[this new spec]]; if absent, the default value is 23". =
&nbsp;Correspondingly, in say SDP, one might be able to say "if this =
field is 0, the video is mono; otherwise, it carries a frame_packing =
values as defined in [[this new spec]]".</div><div><br></div><div>Makes =
sense?</div><div><br></div><div><br><div><div>On Dec 7, 2011, at 8:08 , =
Stephan Wenger wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: =
14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div>mmusic =
has received an incoming liaison from MPEG, indicating that they have =
started on "codec independent media description code points". &nbsp;My =
understanding of this work is that these code points may be represented =
in SDP.</div><div>My personal viewpoint is that this may well be sound =
architectural work, but it may be better conducted here in mmusic, at =
least when it comes to SDP representations and the related IANA work. =
&nbsp;However,&nbsp;I know nothing about this work beyond what's written =
in the statement, which is why I used "soft" language. &nbsp;Perhaps =
some of you can shed light into what's really going on.</div><div>The =
assorted avt WGs are probably also affected, as some of the proposed =
code points sound like they are being described in payload formats, RTP =
extensions and perhaps even in xrblock. &nbsp;Let me suggest to use only =
the music list for discussions, as only mmusic is being addressed in the =
statement.</div><div>As it is short, I have attached the statement in =
word format to this email. &nbsp;It will also appear shortly in the IETF =
liaison database.</div><div>Regards,</div><div>Stephan</div></div>
=
<span>&lt;29n124561.doc&gt;</span>________________________________________=
_______<br>Audio/Video Transport Core Maintenance<br><a =
href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>https://www.ietf.org/mail=
man/listinfo/avt<br></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: medium; 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; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; 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; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div><div>David Singer</div><div>Multimedia and =
Software Standards, Apple =
Inc.</div></div></div></span></div></span></span>
</div>
<br></div></body></html>=

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From ciphor@starv.tv  Wed Dec  7 18:55:03 2011
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Please help me on solving following Issues related to RTP based IMS VOIP Session.
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The "Transport" header in the SETUP response may have "ssrc=3D" parameter.
You can use that parameter to detect unknown ssrc.

regards
Chen

2011/12/7 ramesh kumar <ramesh3184@gmail.com>

> Hi,
> How can we detect UnKnown SSRC to drop it?
>
> Thanks & Regards,
> Ramesh.
> On 7 December 2011 12:24, =E8=8C=83=E6=99=A8 <ciphor@starv.tv> wrote:
>
>> Just drop the packets with unknown SSRC before the audio/video decoder.
>>
>> 2011/12/6 ramesh kumar <ramesh3184@gmail.com>
>>
>>>  Hi,
>>>
>>> I am facing following Issues in RTP based IMS VOIP Session:
>>>
>>> a) Previous VT/VOIP Call Media Session RTP packets flowing into current
>>> VT/VOIP Call Media Session due to Network Congestion/Delay.
>>>
>>> b) Media Packets received have unknown SSRC than the other Media packet=
s
>>> due to eavesdropper in the network.
>>>
>>> How to ignore Unknown SSRC Packets/Previous Call Media Packets from
>>> playing?
>>>
>>> --
>>> *Thanks & Regards,
>>> R.Ramesh Kumar
>>> *
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>>
>>
>
>
> --
> *Thanks & Regards,
> R.Ramesh Kumar
> *
>

--bcaec548a06365ceeb04b38bcbed
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

The &quot;Transport&quot; header in the SETUP response may have &quot;ssrc=
=3D&quot; parameter.<div>You can use that parameter to detect unknown ssrc.=
</div><div><br></div><div>regards</div><div>Chen<br><br><div class=3D"gmail=
_quote">

2011/12/7 ramesh kumar <span dir=3D"ltr">&lt;<a href=3D"mailto:ramesh3184@g=
mail.com">ramesh3184@gmail.com</a>&gt;</span><br><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex;">

<div>Hi,</div>
<div>How can we detect UnKnown SSRC to drop it?</div>
<div>=C2=A0</div>
<div>Thanks &amp; Regards,</div>
<div>Ramesh.<br></div><div class=3D"HOEnZb"><div class=3D"h5">
<div class=3D"gmail_quote">On 7 December 2011 12:24, =E8=8C=83=E6=99=A8 <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:ciphor@starv.tv" target=3D"_blank">cip=
hor@starv.tv</a>&gt;</span> wrote:<br>
<blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PA=
DDING-LEFT:1ex" class=3D"gmail_quote">Just drop the packets with unknown SS=
RC before the audio/video decoder.<br><br>
<div class=3D"gmail_quote">2011/12/6 ramesh kumar <span dir=3D"ltr">&lt;<a =
href=3D"mailto:ramesh3184@gmail.com" target=3D"_blank">ramesh3184@gmail.com=
</a>&gt;</span><br>
<blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PA=
DDING-LEFT:1ex" class=3D"gmail_quote">
<div>
<div>
<p>Hi,</p>
<p>I am facing following Issues in RTP based IMS VOIP Session:</p>
<p>a) Previous VT/VOIP Call Media Session RTP packets flowing into current =
VT/VOIP Call Media Session due to Network Congestion/Delay.</p>
<p>b) Media Packets received have unknown SSRC than the other Media packets=
 due to eavesdropper in the network.</p>
<p>How to ignore Unknown SSRC Packets/Previous Call Media Packets from play=
ing?<br clear=3D"all"><font color=3D"#888888"><br>-- <br><font style=3D"BAC=
KGROUND-COLOR:#ffffff" color=3D"#000000"><em>Thanks &amp; Regards,<br>R.Ram=
esh Kumar<br>


</em></font><br></font></p><br></div></div>________________________________=
_______________<br>mmusic mailing list<br><a href=3D"mailto:mmusic@ietf.org=
" target=3D"_blank">mmusic@ietf.org</a><br><a href=3D"https://www.ietf.org/=
mailman/listinfo/mmusic" target=3D"_blank">https://www.ietf.org/mailman/lis=
tinfo/mmusic</a><br>


<br></blockquote></div><br></blockquote></div><br><br clear=3D"all"><br>-- =
<br><font style=3D"BACKGROUND-COLOR:#ffffff" color=3D"#000000"><em>Thanks &=
amp; Regards,<br>R.Ramesh Kumar<br></em></font><br>
</div></div></blockquote></div><br></div>

--bcaec548a06365ceeb04b38bcbed--

From miguel.a.garcia@ericsson.com  Sun Dec 11 23:50:47 2011
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The minutes of the IETF 82 MMUSIC meeting are now available at:

http://www.ietf.org/proceedings/82/minutes/mmusic.htm

Please read them, and if you have comments, questions, or changes to be
made, let the chairs know it.

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

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Subject: [MMUSIC] Need clarification on SSRC usage in Consecutive RTP Sessions
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--e89a8f83a0e1f851c704b40abe81
Content-Type: text/plain; charset=ISO-8859-1

Hi All,

I have a question on SSRC usage in RTP:

As per RFC 3550,

 The SSRC identifier is a
      randomly chosen value meant to be globally unique within a
      particular RTP session.  A participant need not
      use the same SSRC identifier for all the RTP sessions in a
      multimedia session.

Question:  Is there any possibility of SSRC Identifier be same across
consecutive RTP Sessions ? Does Standard emphasis on this ?

-- 
*Thanks & Regards,
R.Ramesh Kumar
*

--e89a8f83a0e1f851c704b40abe81
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Hi All,</div>
<div>=A0</div>
<div>I have a question on SSRC usage in RTP:</div>
<div>=A0</div>
<div>As per RFC 3550,</div>
<div>=A0</div>
<div>=A0The SSRC identifier is a<br>=A0=A0=A0=A0=A0 randomly chosen value m=
eant to be globally unique within a<br>=A0=A0=A0=A0=A0 particular RTP sessi=
on.=A0 A participant need not<br>=A0=A0=A0=A0=A0 use the same SSRC identifi=
er for all the RTP sessions in a<br>
=A0=A0=A0=A0=A0 multimedia session.</div>
<div>=A0</div>
<div>Question:=A0=A0Is there any possibility of=A0SSRC Identifier be same a=
cross consecutive RTP Sessions ? Does Standard emphasis on this ? <br clear=
=3D"all"><br>-- <br><font style=3D"BACKGROUND-COLOR:#ffffff" color=3D"#0000=
00"><em>Thanks &amp; Regards,<br>
R.Ramesh Kumar<br></em></font><br></div>

--e89a8f83a0e1f851c704b40abe81--

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Subject: [MMUSIC] Fwd: RFC 6466 on IANA Registration of the 'image' Media Type for the Session Description Protocol (SDP)
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FYI

-------- Original Message --------
Subject: RFC 6466 on IANA Registration of the 'image' Media Type for the 
Session Description Protocol (SDP)
Date: Mon, 19 Dec 2011 17:19:14 +0100
From: rfc-editor@rfc-editor.org <rfc-editor@rfc-editor.org>
To: ietf-announce@ietf.org <ietf-announce@ietf.org>, 
rfc-dist@rfc-editor.org <rfc-dist@rfc-editor.org>
CC: rfc-editor@rfc-editor.org <rfc-editor@rfc-editor.org>


A new Request for Comments is now available in online RFC libraries.


         RFC 6466

         Title:      IANA Registration of the 'image'
                     Media Type for the Session Description
                     Protocol (SDP)
         Author:     G. Salgueiro
         Status:     Standards Track
         Stream:     IETF
         Date:       December 2011
         Mailbox:    gsalguei@cisco.com
         Pages:      6
         Characters: 11660
         Updates/Obsoletes/SeeAlso:   None

         I-D Tag:    draft-salgueiro-mmusic-image-iana-registration-09.txt

         URL:        http://www.rfc-editor.org/rfc/rfc6466.txt

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.  [STANDARDS-TRACK]

This is now a Proposed Standard Protocol.

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

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From petithug@acm.org  Tue Dec 20 09:16:55 2011
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

The example at the bottom of page 5 looks incorrect:

"v=0
 o=- 25678 753849 IN IP4 128.96.41.1
 s=
 c=IN IP4 128.96.41.1
 t=0 0
 m=audio 3456 RTP/AVP 18 96
 a=rtpmap:96 telephone-event
 a=fmtp:96 0-15,32-35
 a=sqn: 0
 a=cdsc: 1 audio RTP/AVP 0 18 96
 a=cpar: a=fmtp:96 0-16,32-35
 a=cdsc: 4 image udptl t38
 a=cdsc: 5 image tcp t38"

AFAIK the clock rate is mandatory, so the a=rtpmap line should be:

a=rtpmap:96 telephone-event/1000

If I do not receive emails stating that I am wrong in the next few days, I will
fill an errata.

Thanks.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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The rate parameter for telephone events is OPTIONAL (RFC 4733, ex 2833).
See =A7 7.1.2/RFC 4733.
The concept as such, as well as the default value, makes perfectly sense (e=
.g., also from an historical point of narrowband voice codecs).

Thus, RFC 4733/2833 takes precedence over RFC 4566/2327.

If you want to enhance the RFCs, then that relation could be added.

-Albrecht
ALCATEL-LUCENT
=20

> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Marc Petit-Huguenin
> Sent: Dienstag, 20. Dezember 2011 18:17
> To: mmusic@ietf.org
> Subject: [MMUSIC] Error in RFC 3407?
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>=20
> The example at the bottom of page 5 looks incorrect:
>=20
> "v=3D0
>  o=3D- 25678 753849 IN IP4 128.96.41.1
>  s=3D
>  c=3DIN IP4 128.96.41.1
>  t=3D0 0
>  m=3Daudio 3456 RTP/AVP 18 96
>  a=3Drtpmap:96 telephone-event
>  a=3Dfmtp:96 0-15,32-35
>  a=3Dsqn: 0
>  a=3Dcdsc: 1 audio RTP/AVP 0 18 96
>  a=3Dcpar: a=3Dfmtp:96 0-16,32-35
>  a=3Dcdsc: 4 image udptl t38
>  a=3Dcdsc: 5 image tcp t38"
>=20
> AFAIK the clock rate is mandatory, so the a=3Drtpmap line should be:
>=20
> a=3Drtpmap:96 telephone-event/1000
>=20
> If I do not receive emails stating that I am wrong in the=20
> next few days, I will fill an errata.
>=20
> Thanks.
>=20
> - --
> Marc Petit-Huguenin
> Personal email: marc@petit-huguenin.org
> Professional email: petithug@acm.org
> Blog: http://blog.marc.petit-huguenin.org
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.11 (GNU/Linux)
>=20
> iQIcBAEBCAAGBQJO8ML5AAoJECnERZXWan7E/nEP/2Bg12DvPQcFrh3QdiSTZ9qT
> miLRM48XZ6LHwR+WubsGkSOesPtrwtfd7zgqHwt+lh415izps58gYtHD9yt9ozMx
> cFoWAZpFMcOd5lVlSDSU+83l9I5TSAO/+W+HFFlNIwa9/VKJnVvBRjVolHxPj1Yb
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> z382+a8VwDUmv3aWAmpVmX3jjMhAr1/pngiCvEV0i7zGbMtYjGt8uiAQiyTBCgRa
> 8Oc4DkMfdLDDyUta0eTlBO/6+p5AZ+9ulEl4crn/gtwqjUUEoNOhHoNOJhMa+GlM
> FJ6D9f5NMxaBysC1xY7RmwmrERMBuKhL/tTRvxg/D+7T0zj8SHK8XS4Cdd95qV6t
> AsARa42AXgap/w5t0GGoBYBNjEo9TtlhsGvJKgepxEYMaf5hRCs5TadrMrEUCK3G
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> 0J9YY4iGmjTcYm0C9gqF
> =3DdpRU
> -----END PGP SIGNATURE-----
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> =

From petithug@acm.org  Tue Dec 20 09:58:28 2011
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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [MMUSIC] Error in RFC 3407?
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

On 12/20/2011 09:50 AM, Schwarz, Albrecht (Albrecht) wrote:
> The rate parameter for telephone events is OPTIONAL (RFC 4733, ex 2833).

In the MIME type it is optional with adefault value of 8000, but the rate
parameter in rtpmap is not optional (section 6 of RFC 4566):

"a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding
         parameters>]"

> See Â§ 7.1.2/RFC 4733.
> The concept as such, as well as the default value, makes perfectly sense (e.g., also from an historical point of narrowband voice codecs).
> 
> Thus, RFC 4733/2833 takes precedence over RFC 4566/2327.
> 
> If you want to enhance the RFCs, then that relation could be added.
> 
> -Albrecht
> ALCATEL-LUCENT
>  
> 
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org 
>> [mailto:mmusic-bounces@ietf.org] On Behalf Of Marc Petit-Huguenin
>> Sent: Dienstag, 20. Dezember 2011 18:17
>> To: mmusic@ietf.org
>> Subject: [MMUSIC] Error in RFC 3407?
>>
> The example at the bottom of page 5 looks incorrect:
> 
> "v=0
>  o=- 25678 753849 IN IP4 128.96.41.1
>  s=
>  c=IN IP4 128.96.41.1
>  t=0 0
>  m=audio 3456 RTP/AVP 18 96
>  a=rtpmap:96 telephone-event
>  a=fmtp:96 0-15,32-35
>  a=sqn: 0
>  a=cdsc: 1 audio RTP/AVP 0 18 96
>  a=cpar: a=fmtp:96 0-16,32-35
>  a=cdsc: 4 image udptl t38
>  a=cdsc: 5 image tcp t38"
> 
> AFAIK the clock rate is mandatory, so the a=rtpmap line should be:
> 
> a=rtpmap:96 telephone-event/1000
> 
> If I do not receive emails stating that I am wrong in the 
> next few days, I will fill an errata.
> 
> Thanks.
> 

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Tue Dec 20 16:33:47 2011
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-----BEGIN PGP SIGNED MESSAGE-----
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Section 15.3 states that:

'"ice-mismatch" is a media-level attribute only.[...]'

But section 21.1.4 (and IANA) declares this attribute as a session level attribute.

It seems logical to have ice-mismatch as a media level attribute, but I guess
that something more than an errata is needed to fix the wrong IANA declaration.

What is the process in this case?

Thanks.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Tue Dec 20 17:00:26 2011
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This document was not presented in Taipei for lack of time, but it was discussed
in the Quebec City meeting.  As the minutes show, at the time there was
discussions about the problem been perhaps larger than just for ICE.  I did an
analysis of all the session level only attributes currently registered by IANA
and my conclusion is that only ICE has this problem.  The analysis for each
attribute was added in the document as an appendix.

Also, as explained in my previous email, it seems that ice-mismatch was
incorrectly registered as a session level attribute.  This document could be a
good place to fix this if a standard track document is needed for that.

Changelog:

o  Added reference to RFC 5245 in the abstract (idnits).
o  Changed author affiliation.
o  Added analysis of all IANA registered session level only
   attributes.
o  Added sections for ice-lite and ice-mismatch.


[1] http://www.ietf.org/proceedings/81/minutes/mmusic.txt

- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-mmusic-ice-attributes-level-02.txt
Date: Tue, 20 Dec 2011 16:48:19 -0800
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org


A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : Media level ice-options SDP attribute
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-petithuguenin-mmusic-ice-attributes-level-02.txt
	Pages           : 14
	Date            : 2011-12-20

   This document normatively updates RFC 5245 by redefining the ice-
   options SDP attribute as a session-level and media-level attribute.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-mmusic-ice-attributes-level-02.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-petithuguenin-mmusic-ice-attributes-level-02.txt

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From: "Mostafa, Adel (Adel)" <amostafa@avaya.com>
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Date: Fri, 23 Dec 2011 17:43:18 -0500
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First draft for negotiating multi-stream Continuous Presence Video in SIP, =
please let me know if any comments.


Regards,
Adel Mostafa|Unified Communications Architect, P.Eng.|Avaya R&D|Canada|Offi=
ce: +1.613.967.5514




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<div>First draft for negotiating multi-stream Continuous Presence Video in =
SIP, please let me know if any comments.</div>
<div>&nbsp;</div>
<div> </div>
<div>Regards,</div>
<div><font face=3D"Century Gothic, sans-serif" size=3D"2" color=3D"#999999"=
><b>Adel Mostafa</b><font color=3D"#FF0033">|</font><b>Unified Communicatio=
ns Architect, P.Eng.</b><font color=3D"#FF0033">|</font><b>Avaya R&amp;D</b=
><font color=3D"#FF0033">|</font><b>Canada</b><font color=3D"#FF0033">|</fo=
nt><b>Office:
&#43;1.613.967.5514</b><font face=3D"Calibri, sans-serif">&nbsp;&nbsp;</fon=
t></font></div>
<div>&nbsp;</div>
<div>&nbsp;</div>
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