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Date: Thu, 08 Jan 2009 16:12:26 +0800
From: Richard Wu <richard@astri.org>
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Cc: Fu Siu Ming Claudio <claudiof@astri.org>
Subject: [MMUSIC] Popularity of supporting RTCP extended report (RFC3611) in
 voip phone/video phone
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Hi All,
    I would like to know the popularity of supporting RFC3611 in current 
voip phone/video phone. Thanks in advance.

Regards,
Richard

Senior Engineer
ASTRI

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From mmusic-bounces@ietf.org  Thu Jan  8 00:18:21 2009
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Cc: Fu Siu Ming Claudio <claudiof@astri.org>, mmusic@ietf.org, avt@ietf.org
Subject: Re: [MMUSIC] Popularity of supporting RTCP extended report
 (RFC3611) in voip phone/video phone
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Hi,

you should address this question to the AVT group.
(CC'ed).

Joerg

Richard Wu wrote:
> Hi All,
>    I would like to know the popularity of supporting RFC3611 in current 
> voip phone/video phone. Thanks in advance.
> 
> Regards,
> Richard
> 
> Senior Engineer
> ASTRI
> 
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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> and notify the sender. Any use, disclosure, copying, or distribution of
> this message and/or any attachments is strictly prohibited.
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> 
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From mmusic-bounces@ietf.org  Thu Jan  8 00:33:48 2009
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Thread-Topic: RTSP From header issue
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From: "Ingemar Johansson S" <ingemar.s.johansson@ericsson.com>
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
Subject: [MMUSIC] RTSP From header issue
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Hi

Going through the syntax of the parts that handles URI's I found a possible=
 issue with the syntax of the From header. =

1) In section 16.23 in RTSP 2.0 there is only a reference section 14.22 in =
RFC2616 which contains a rather simple definition.
       From   =3D "From" ":" mailbox
Section 20.2.3 in RTSP 2.0 however contains a much more elaborate definitio=
n with extensions (from-param). =

Isn't it better to specify something in section 16.23 like "The From header=
 is defined in in [H14.22] but is extended in this specification" ?

2) Looking at the syntax I see a risk that occasional ";" in the URI can cl=
ash with URI's that precedes each optional from-param, making parsing compl=
icated.
The ABNF for the From header is
   From           =3D  "From" HCOLON from-spec
   from-spec      =3D  ( name-addr / addr-spec ) *( SEMI from-param )
   name-addr      =3D  [ display-name ] LAQUOT addr-spec RAQUOT
   addr-spec      =3D  RTSP-URI / absolute-URI
   absolute-URI   =3D  < As defined in RFC 3986>
   display-name   =3D  *(token LWS) / quoted-string
   from-param     =3D  tag-param / generic-param
   tag-param      =3D  "tag" EQUAL token

Here I would suggest to change the addr-spec definition to =

   addr-spec      =3D  DQ ( RTSP-URI / absolute-URI ) DQ
As a comparision, Accept-Credentials, which has some similarities with the =
>From header, use DQ around the URI.

3) In general. The idea to use DQ around URI's in some cases and in some ca=
ses not may cause some headache for the implementor. Sure, one should read =
the ABNF but I would say that it would be more consistent to always use DQ =
around URI's in headers regardless if they are needed or not.

/Ingemar
 =


******************************************* =

Ingemar Johansson =

Senior Research Engineer, IETF "nethead" =

EAB/TVP - Multimedia Technologies =

Ericsson Research Ericsson AB =

Box 920 S-971 28 Lule=E5, Sweden =

Tel: +46 (0)10 7143042 =

ECN: 852-43042 =

ECC: 852-19042
Mobile: +46 (0)730 783289 =

Visit http://labs.ericsson.com !
******************************************* =

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On Thursday 08 January 2009 10:33:25 ext Ingemar Johansson S, you wrote:
> Hi
>
> Going through the syntax of the parts that handles URI's I found a possib=
le
> issue with the syntax of the From header. 1) In section 16.23 in RTSP 2.0
> there is only a reference section 14.22 in RFC2616 which contains a rather
> simple definition. From   =3D "From" ":" mailbox
> Section 20.2.3 in RTSP 2.0 however contains a much more elaborate
> definition with extensions (from-param). Isn't it better to specify
> something in section 16.23 like "The From header is defined in in [H14.22]
> but is extended in this specification" ?

Is anyone using the From: in HTTP or RTSP anyhow? I guess not.

-- =

R=E9mi Denis-Courmont
Maemo Software, Nokia Devices R&D
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Joerg Ott'" <jo@netlab.tkk.fi>, "'Richard Wu'" <richard@astri.org>
References: <4965B56A.6060203@astri.org> <4965B6C2.9060709@netlab.tkk.fi>
In-Reply-To: <4965B6C2.9060709@netlab.tkk.fi>
Date: Thu, 8 Jan 2009 11:24:20 -0500
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Cc: 'Fu Siu Ming Claudio' <claudiof@astri.org>, mmusic@ietf.org, avt@ietf.org
Subject: Re: [MMUSIC] [AVT] Popularity of supporting RTCP extended report
	(RFC3611) in voip phone/video phone
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Richard,

You might be asking the wrong question.  I personally do not care how widely
implemented it is: it ought to be implemented more widely.  Heck, even RTCP
is not as widely implemented (or at least used) as it ought to be.  

RCTP XR can help to provide important statistics that can be used to improve
call quality and diagnosing network issues.  For example, H.323 has the
wherewithal to report the RTCP XR data to the Gatekeeper at the end of the
call.  The Gatekeeper can then feed that to back-end call processing systems
that can alert operators to network issues and to routing engines that use
that information to route calls to avoid problematic network links or
devices.

Paul

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Joerg Ott
> Sent: Thursday, January 08, 2009 3:18 AM
> To: Richard Wu
> Cc: Fu Siu Ming Claudio; mmusic@ietf.org; avt@ietf.org
> Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended
> report (RFC3611) in voip phone/video phone
> 
> Hi,
> 
> you should address this question to the AVT group.
> (CC'ed).
> 
> Joerg
> 
> Richard Wu wrote:
> > Hi All,
> >    I would like to know the popularity of supporting RFC3611 in
> current
> > voip phone/video phone. Thanks in advance.
> >
> > Regards,
> > Richard
> >
> > Senior Engineer
> > ASTRI
> >
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> > This message (including any attachments) is for the named
> > addressee(s)'s use only. It may contain sensitive, confidential,
> > private proprietary or legally privileged information intended for a
> > specific individual and purpose, and is protected by law. If you are
> > not the intended recipient, please immediately delete it and all
> copies
> > of it from your system, destroy any hard copies of it
> > and notify the sender. Any use, disclosure, copying, or distribution
> of
> > this message and/or any attachments is strictly prohibited.
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


_______________________________________________
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From mmusic-bounces@ietf.org  Thu Jan  8 17:29:45 2009
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	<005e01c971ad$90290ca0$b07b25e0$@com>
From: "Jaehwan Kim" <jaehwan@vidiator.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "Joerg Ott" <jo@netlab.tkk.fi>,
	"Richard Wu" <richard@astri.org>
Cc: Fu Siu Ming Claudio <claudiof@astri.org>, mmusic@ietf.org, avt@ietf.org
Subject: Re: [MMUSIC] [AVT] Popularity of supporting RTCP extended
	report(RFC3611) in voip phone/video phone
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Hi,
If video phone means 3GPP compliant streaming solution...
RTCP XR is a part of 3GPP PSS specification.
I know one phone manufacturer supports it and some streaming servers
support it as well. 

Cheers,
Jaehwan

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Paul E. Jones
Sent: Friday, January 09, 2009 1:24 AM
To: 'Joerg Ott'; 'Richard Wu'
Cc: 'Fu Siu Ming Claudio'; mmusic@ietf.org; avt@ietf.org
Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended
report(RFC3611) in voip phone/video phone

Richard,

You might be asking the wrong question.  I personally do not care how
widely
implemented it is: it ought to be implemented more widely.  Heck, even
RTCP
is not as widely implemented (or at least used) as it ought to be.  

RCTP XR can help to provide important statistics that can be used to
improve
call quality and diagnosing network issues.  For example, H.323 has the
wherewithal to report the RTCP XR data to the Gatekeeper at the end of
the
call.  The Gatekeeper can then feed that to back-end call processing
systems
that can alert operators to network issues and to routing engines that
use
that information to route calls to avoid problematic network links or
devices.

Paul

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Joerg Ott
> Sent: Thursday, January 08, 2009 3:18 AM
> To: Richard Wu
> Cc: Fu Siu Ming Claudio; mmusic@ietf.org; avt@ietf.org
> Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended
> report (RFC3611) in voip phone/video phone
> 
> Hi,
> 
> you should address this question to the AVT group.
> (CC'ed).
> 
> Joerg
> 
> Richard Wu wrote:
> > Hi All,
> >    I would like to know the popularity of supporting RFC3611 in
> current
> > voip phone/video phone. Thanks in advance.
> >
> > Regards,
> > Richard
> >
> > Senior Engineer
> > ASTRI
> >
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> > This message (including any attachments) is for the named
> > addressee(s)'s use only. It may contain sensitive, confidential,
> > private proprietary or legally privileged information intended for a
> > specific individual and purpose, and is protected by law. If you are
> > not the intended recipient, please immediately delete it and all
> copies
> > of it from your system, destroy any hard copies of it
> > and notify the sender. Any use, disclosure, copying, or distribution
> of
> > this message and/or any attachments is strictly prohibited.
> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 


_______________________________________________
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--NextPart

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 media capabilities Negotiation
	Author(s)       : R. Gilman, et al.
	Filename        : draft-ietf-mmusic-sdp-media-capabilities-06.txt
	Pages           : 45
	Date            : 2009-01-09

Session Description Protocol (SDP) capability negotiation provides a
general framework for indicating and negotiating capabilities in SDP.
The base framework defines only capabilities for negotiating
transport protocols and attributes.  In this document, we extend the
framework by defining media capabilities that can be used to
negotiate media types and their associated parameters.  This
extension is designed to map easily to existing and future SDP media
attributes, but not encodings or formatting.

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	Title           : The SDP (Session Description Protocol) Grouping Framework
	Author(s)       : G. Camarillo
	Filename        : draft-ietf-mmusic-rfc3388bis-02.txt
	Pages           : 20
	Date            : 2009-01-13

In this specification, we define a framework to group "m" lines in
SDP (Session Description Protocol) for different purposes.  This
framework uses the "group" and "mid" SDP attributes, both of which
are defined in this specification.  Additionally, we specify how to
use the framework for two different purposes: for lip synchronization
and for receiving a media flow consisting of several media streams on
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Folks,

we are issuing Working Group Last Call for
draft-ietf-mmusic-rfc3388bis-02.txt for Proposed Standard.

Please review the document and send comments to the MMUSIC
mailing list and/or the authors.

The WGLC is to expire on 31 January 2009.

Joerg

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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [MMUSIC] RTSP 2.0 status
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Hi,

We authors had the goal to provide a new draft version before new year.
This has not happened due to one major reason, namely RFC 5378.

If you missed the intensive discussion on this subject on ietf@ietf.org
list, the issue is that you as author when submitting a document using
the new IPR rules needs to grand additional rights to the IETF trust.
However, for documents that has text contributions from people beyond
they authors, the author needs to ensure that these rights are provided
from these contributors also. This can be problematic as the copyright
of ones work commonly belong to our employers.

We have been working on obtaining these rights for all that has directly
contributed text to RTSP. This has been quite successful and are to my
knowledge currently missing only the rights from Real Networks. However,
the sad story is that this will not be sufficient to submit the draft.
RTSP 2.0 contains numerous paragraphs of text that is copied from HTTP
(RFC 2068 or RFC 2616) and then adapted for RTSP. The problem with HTTP
is that is has a very long list of contributors and the chance of
success to acquire those rights in a timely manor are slim. Martin and I
have discussed the alternatives and we need this text in the document
for RTSP 2.0 to be understandable.

Fortunately there are some hope, the trust are working on providing a
work around until an updated version of RFC 5378 can be written.
Hopefully this workaround will be available before draft submission
cut-off for San Francisco. Thus allowing us to submit the draft for your
review and discussion in San Francisco.

We where also slightly behind on closing the open issues and we are
continuing to work on these to ensure that we have a draft in good shape
when we can submit it.

Best Regards

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
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From mmusic-bounces@ietf.org  Thu Jan 15 06:42:32 2009
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	Fri,  9 Jan 2009 10:35:43 -0500 (EST)
From: Alan Clark <alan.d.clark@telchemy.com>
To: Jaehwan Kim <jaehwan@vidiator.com>,
	"Paul E. Jones" <paulej@packetizer.com>, Joerg Ott <jo@netlab.tkk.fi>, 
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Cc: Fu Siu Ming Claudio <claudiof@astri.org>, mmusic@ietf.org, avt@ietf.org
Subject: Re: [MMUSIC] [AVT] Popularity of supporting RTCP
 extended	report(RFC3611) in voip phone/video phone
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Independently of any mobile phone applications, there are known to be at least 
50 million devices (in reality closer to 200 million) and over 100 equipment 
vendors that have the capability of reporting RTCP XR, and a fair number that 
also use the SIP RTCP Summary Reporting draft.  These include IP phones from 
two tier 1 telecom equipment providers, softphones, residential and trunking 
gateways, and MTA's (cable industry).  Also Texas Instruments incorporated 
support for RTCP XR into all of their VoIP solutions.

In addition to the RTCP XR VoIP metrics being adopted as part of H.323, they 
were also incorporated into H.248 (Megaco) and into PacketCable 1.5.  

Regards

Alan Clark
Telchemy


---- Original message ----
>Date: Fri, 9 Jan 2009 10:28:01 +0900
>From: "Jaehwan Kim" <jaehwan@vidiator.com>  
>Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended 
report(RFC3611) in voip phone/video phone  
>To: "Paul E. Jones" <paulej@packetizer.com>, "Joerg Ott" <jo@netlab.tkk.fi>, 
"Richard Wu" <richard@astri.org>
>Cc: Fu Siu Ming Claudio <claudiof@astri.org>, mmusic@ietf.org, avt@ietf.org
>
>Hi,
>If video phone means 3GPP compliant streaming solution...
>RTCP XR is a part of 3GPP PSS specification.
>I know one phone manufacturer supports it and some streaming servers
>support it as well. 
>
>Cheers,
>Jaehwan
>
>-----Original Message-----
>From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>Paul E. Jones
>Sent: Friday, January 09, 2009 1:24 AM
>To: 'Joerg Ott'; 'Richard Wu'
>Cc: 'Fu Siu Ming Claudio'; mmusic@ietf.org; avt@ietf.org
>Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended
>report(RFC3611) in voip phone/video phone
>
>Richard,
>
>You might be asking the wrong question.  I personally do not care how
>widely
>implemented it is: it ought to be implemented more widely.  Heck, even
>RTCP
>is not as widely implemented (or at least used) as it ought to be.  
>
>RCTP XR can help to provide important statistics that can be used to
>improve
>call quality and diagnosing network issues.  For example, H.323 has the
>wherewithal to report the RTCP XR data to the Gatekeeper at the end of
>the
>call.  The Gatekeeper can then feed that to back-end call processing
>systems
>that can alert operators to network issues and to routing engines that
>use
>that information to route calls to avoid problematic network links or
>devices.
>
>Paul
>
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
>> Joerg Ott
>> Sent: Thursday, January 08, 2009 3:18 AM
>> To: Richard Wu
>> Cc: Fu Siu Ming Claudio; mmusic@ietf.org; avt@ietf.org
>> Subject: Re: [AVT] [MMUSIC] Popularity of supporting RTCP extended
>> report (RFC3611) in voip phone/video phone
>> 
>> Hi,
>> 
>> you should address this question to the AVT group.
>> (CC'ed).
>> 
>> Joerg
>> 
>> Richard Wu wrote:
>> > Hi All,
>> >    I would like to know the popularity of supporting RFC3611 in
>> current
>> > voip phone/video phone. Thanks in advance.
>> >
>> > Regards,
>> > Richard
>> >
>> > Senior Engineer
>> > ASTRI
>> >
>> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>> > This message (including any attachments) is for the named
>> > addressee(s)'s use only. It may contain sensitive, confidential,
>> > private proprietary or legally privileged information intended for a
>> > specific individual and purpose, and is protected by law. If you are
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>> > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>> >
>> > _______________________________________________
>> > mmusic mailing list
>> > mmusic@ietf.org
>> > https://www.ietf.org/mailman/listinfo/mmusic
>> >
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>> 
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt

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Subject: [MMUSIC] I-D Action:draft-ietf-mmusic-rfc4756bis-00.txt
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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           : Forward Error Correction Grouping Semantics in Session Description Protocol
	Author(s)       : A. Begen
	Filename        : draft-ietf-mmusic-rfc4756bis-00.txt
	Pages           : 10
	Date            : 2009-01-21

The Session Description Protocol (SDP) supports grouping media lines.
SDP also has semantics defined for grouping the associated source and
Forward Error Correction (FEC)-based repair flows.  However, the
semantics that were defined in RFC 4756 generally fail to provide the
specific grouping relationships between the source and repair flows
when there are more than one source and/or repair flows in the same
group.  Furthermore, the existing semantics also do not support
additive repair flows.  This document addresses these issues by
introducing new FEC grouping semantics.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc4756bis-00.txt

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Errors-To: mmusic-bounces@ietf.org We have just submitted a new version of the file transfer document. This 
version includes a few new paragraphs to satisfy Richard Barnes' comments 
during the security directorate review. This paragraphs provide 
recommendations that verify that the properties of a received file match 
the properties declared in the SDP. Also some reference to authentication 
and authorization in SIP and MSRP are included. If you are interested in 
a more detail level of changes, you can ran the diff with version -09 
when this new version shows up in tools.ietf.org.

/Miguel



A new version of I-D, draft-ietf-mmusic-file-transfer-mech-10.txt has 
been successfuly submitted by Miguel Garcia and posted to the IETF 
repository.

Filename:	 draft-ietf-mmusic-file-transfer-mech
Revision:	 10
Title:		 A Session Description Protocol (SDP) Offer/Answer Mechanism to 
Enable File Transfer
Creation_date:	 2009-01-22
WG ID:		 mmusic
Number_of_pages: 49

Abstract:
This document provides a mechanism to negotiate the transfer of one
or more files between two endpoints by using the Session Description
Protocol (SDP) offer/answer model specified in RFC 3264.  SDP is
extended to describe the attributes of the files to be transferred.
The offerer can either describe the files it wants to send, or the
files it would like to receive.  The answerer can either accept or
reject the offer separately for each individual file.  The transfer
of one or more files is initiated after a successful negotiation.
The Message Session Relay Protocol (MSRP) is defined as the default
mechanism to actually carry the files between the endpoints.  The
conventions on how to use MSRP for file transfer are also provided in
this document.
 



The IETF Secretariat.


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.


	Title           : A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer
	Author(s)       : M. Garcia, et al.
	Filename        : draft-ietf-mmusic-file-transfer-mech-10.txt
	Pages           : 49
	Date            : 2009-01-22

This document provides a mechanism to negotiate the transfer of one
or more files between two endpoints by using the Session Description
Protocol (SDP) offer/answer model specified in RFC 3264.  SDP is
extended to describe the attributes of the files to be transferred.
The offerer can either describe the files it wants to send, or the
files it would like to receive.  The answerer can either accept or
reject the offer separately for each individual file.  The transfer
of one or more files is initiated after a successful negotiation.
The Message Session Relay Protocol (MSRP) is defined as the default
mechanism to actually carry the files between the endpoints.  The
conventions on how to use MSRP for file transfer are also provided in
this document.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-file-transfer-mech-10.txt

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From mmusic-bounces@ietf.org  Mon Jan 26 06:57:07 2009
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Subject: [MMUSIC] Last Call: draft-ietf-mmusic-decoding-dependency (Signaling media decoding dependency in Session Description Protocol (SDP)) to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session 
Control WG (mmusic) to consider the following document:

- 'Signaling media decoding dependency in Session Description Protocol 
   (SDP) '
   <draft-ietf-mmusic-decoding-dependency-05.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2009-02-09. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-decoding-dependency-05.txt


IESG discussion can be tracked via
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Subject: [MMUSIC] Last Call: draft-ietf-mmusic-sdp-source-attributes (Source-Specific Media Attributes in the Session Description Protocol (SDP)) to Proposed Standard
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The IESG has received a request from the Multiparty Multimedia Session 
Control WG (mmusic) to consider the following document:

- 'Source-Specific Media Attributes in the Session Description Protocol 
   (SDP) '
   <draft-ietf-mmusic-sdp-source-attributes-02.txt> as a Proposed Standard


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ietf@ietf.org mailing lists by 2009-02-09. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
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The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-source-attributes-02.txt


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From mmusic-bounces@ietf.org  Mon Jan 26 08:04:31 2009
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From: Adrian Georgescu <ag@ag-projects.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] SIP desktop sharing with draft-garcia-mmusic-sdp-collaboration
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Hello,

We have successfully implemented desktop sharing application using a  
variation of draft-garcia-mmusic-sdp-collaboration namely using an  
MSRP session instead of pure TCP. Here is the relevant negotiated SDP:

INVITE sip:2233361709@sip2sip.info SIP/2.0
Record-Route: <sip:81.23.228.129;lr=on;ftag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2;did=919.7c6dbad4 
 >
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK7ec3.24ae5eb1.0
Via: SIP/2.0/UDP  
192.168.1.6 
:51891;received=80.101.96.20;rport=51891;branch=z9hG4bKPjcZPkzRl- 
dkYHaxeakmQLu8i5TDxIARL6
Max-Forwards: 68
From: "Adrian G." <sip:31208005169@ag- 
projects.com>;tag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2
To: sip:2233361709@sip2sip.info
Contact: <sip:y2CTtqem1v@80.101.96.20:51891;transport=udp>
Call-ID: M64kmnwHHuWtiVQl.QaP0svHi.rhI0-N
CSeq: 13415 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,  
MESSAGE
Supported: 100rel
User-Agent: sip2sip-0.4.0-pjsip-1.0.1-r2427
Content-Type: application/sdp
Content-Length: 281
Thor-Scope: authenticate
Thor-Origin: sip:80.101.96.20:51891

v=0
o=- 3441959219 3441959219 IN IP4 192.168.1.6
s=
c=IN IP4 192.168.1.6
t=0 0
m=application 12345 TCP/TLS/MSRP/RFB *
a=path:msrps://192.168.1.6:12345/Oj0BcOEf4sim;tcp
a=accept-types:message/cpim application/x-rfb
a=accept-wrapped-types:application/x-rfb
a=setup:active


SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK4ec3.220d6d27.0
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK4ec3.cb13ac42.0
Via: SIP/2.0/UDP  
192.168.1.6 
: 
51891 
;rport=51891;received=80.101.96.20;branch=z9hG4bKPjpf7mDBJxwkWa5hlptKm- 
Bozt9IGYm01Y
Record-Route: <sip:81.23.228.150;lr;ftag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2;did=919.8a3da7b5 
 >
Record-Route: <sip:85.17.186.7;lr;ftag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2;did=919.98f40fd4 
 >
Record-Route: <sip:81.23.228.129;lr;ftag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2;did=919.8c6dbad4 
 >
Call-ID: M64kmnwHHuWtiVQl.QaP0svHi.rhI0-N
From: "Adrian G." <sip:31208005169@ag- 
projects.com>;tag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2
To: <sip:2233361709@sip2sip.info>;tag=Qv2lE9YdxHGSmy1O9b534j8jhdKIoErw
CSeq: 13416 INVITE
Contact: <sip:o43D6SEA1I@82.95.141.127:34206>
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,  
MESSAGE
Supported: 100rel
User-Agent: ag-projects/sipclient-0.4.0-pjsip-1.0.1-r2427
Content-Type: application/sdp
Content-Length: 393

v=0
o=- 3441959219 3441959220 IN IP4 192.168.178.33
s=
c=IN IP4 192.168.178.33
t=0 0
m=application 54990 TCP/TLS/MSRP/RFB *
a=path:msrps://node03.dns-hosting.info:2855/6dm+f 
+uC6mgbryTjbHVGEDEyMzI5NzA0MjAuMDU0OjgyLjk1LjE0MS4xMjc=;tcp msrps:// 
192.168.178.33:54990/WSucWUOgCOKg;tcp
a=accept-types:message/cpim application/x-rfb
a=accept-wrapped-types:application/x-rfb
a=setup:passive

The complete SIP session negotiation (click on each message to expand  
its content) is here:

https://secure.dns-hosting.info/CDRTool/sip_trace.phtml?cdr_source=sip_trace_thor&callid=M64kmnwHHuWtiVQl.QaP0svHi.rhI0-N&fromtag=IjiXUJuaiN03As3HGxf0osjkibdnUtd2&totag=Qv2lE9YdxHGSmy1O9b534j8jhdKIoErw&proxyIP=85.17.186.7&public=1

More information and the software is available at:

http://lists.ag-projects.com/mailman/listinfo/sipclient

I am wondering if it makes sense to mention the MSRP/TLS  transport as  
a valid option for negotiating/transporting in the draft.

Kind regards,
Adrian Georgescu



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Thread-Topic: new draft on media routing with ALGs
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From: "Ejzak, Richard P \(Richard\)" <ejzak@alcatel-lucent.com>
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Subject: [MMUSIC] new draft on media routing with ALGs
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Hello,
Please comment on the following draft, which was submitted to improve
media routing when interconnecting SIP networks using ALGs.  Work on
this draft was triggered by a 3GPP Release 9 work item on "optimal media
routing", which is documented in 3GPP TR 23.894.
=20
http://www.ietf.org/internet-drafts/draft-ejzak-mmusic-bg-bypass-00.txt
=20
This is the abstract from the draft:
=20
"This document describes an extension to the Session Description
Protocol (SDP) that can be used by systems of cooperating networks using
Application Level Gateways (ALG) to insert border gateways performing as
Network Address Port Translators (NAPT) between their IP realms to
identify when border gateways can be bypassed for more efficient media
flow.  This extension can be used by networks based on a protocol using
the SDP offer/answer model, such as the IP Multimedia Subsystem (IMS) of
the Third Generation Partnership Project (3GPP), which is based on the
Session Initiation Protocol (SIP).  ALGs using this extension can
determine within a single SDP offer/answer transaction when the
insertion of a new border gateway would cause the media path to re-enter
an IP realm visited elsewhere within the media path, and to bypass one
or more border gateways that would otherwise be included in the media
path.  This extension also works with hosted NAPT traversal schemes to
establish a direct media path between endpoints within the same IP
realm.  Optional procedures provide additional means to improve media
flow."
=20
BR, Richard Ejzak
Alcatel-Lucent

------_=_NextPart_001_01C98008.90B6EE0B
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D251043322-26012009>Hello,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D251043322-26012009>Please =
comment on=20
the following draft, which was submitted to improve media routing when=20
interconnecting SIP networks using ALGs.&nbsp; Work on this draft was =
triggered=20
by a 3GPP Release 9 work item on "optimal media routing", which is =
documented in=20
3GPP TR 23.894.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D251043322-26012009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D251043322-26012009><A=20
title=3Dhttp://www.ietf.org/internet-drafts/draft-ejzak-mmusic-bg-bypass-=
00.txt=20
href=3D"http://www.ietf.org/internet-drafts/draft-ejzak-mmusic-bg-bypass-=
00.txt">http://www.ietf.org/internet-drafts/draft-ejzak-mmusic-bg-bypass-=
00.txt</A></SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D251043322-26012009></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D251043322-26012009>This =
is the abstract=20
from the draft:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D251043322-26012009></SPAN></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D251043322-26012009>
<DIV><FONT face=3DArial><FONT size=3D2><SPAN =
class=3D210294521-05012009>"This document=20
describes an extension to the Session Description </SPAN><SPAN=20
class=3D210294521-05012009>Protocol (SDP) that can be used by systems of =

cooperating networks </SPAN><SPAN class=3D210294521-05012009>using =
Application=20
Level Gateways (ALG) to insert border gateways </SPAN><SPAN=20
class=3D210294521-05012009>performing as Network Address Port =
Translators (NAPT)=20
between their </SPAN><SPAN class=3D210294521-05012009>IP realms to =
identify when=20
border gateways can be bypassed for more </SPAN><SPAN=20
class=3D210294521-05012009>efficient media flow.&nbsp; This extension =
can be used=20
by networks based </SPAN><SPAN class=3D210294521-05012009>on a protocol =
using the=20
SDP offer/answer model, such as the IP </SPAN><SPAN=20
class=3D210294521-05012009>Multimedia Subsystem (IMS) of the Third =
Generation=20
Partnership </SPAN><SPAN class=3D210294521-05012009>Project (3GPP), =
which is based=20
on the Session Initiation Protocol </SPAN><SPAN=20
class=3D210294521-05012009>(SIP).&nbsp; ALGs using this extension can =
determine=20
within a single SDP </SPAN><SPAN class=3D210294521-05012009>offer/answer =

transaction when the insertion of a new border gateway </SPAN><SPAN=20
class=3D210294521-05012009>would cause the media path to re-enter an IP =
realm=20
visited elsewhere&nbsp;within the media path, and to bypass one or more =
border=20
gateways </SPAN><SPAN class=3D210294521-05012009>that would otherwise be =
included=20
in the media path.&nbsp; This extension </SPAN><SPAN=20
class=3D210294521-05012009>also works with hosted NAPT traversal schemes =
to=20
establish a direct </SPAN><SPAN class=3D210294521-05012009>media path =
between=20
endpoints within the same IP realm.&nbsp; Optional </SPAN><SPAN=20
class=3D210294521-05012009>procedures provide additional means to =
improve media=20
flow."</SPAN></FONT></FONT></DIV>
<DIV><SPAN class=3D210294521-05012009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D210294521-05012009><SPAN =
class=3D251043322-26012009><FONT=20
face=3DArial size=3D2>BR, Richard Ejzak</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D210294521-05012009><SPAN =
class=3D251043322-26012009><FONT=20
face=3DArial=20
size=3D2>Alcatel-Lucent</FONT></SPAN></SPAN></DIV></SPAN></DIV></BODY></H=
TML>

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