From mmusic-admin@ietf.org  Fri Feb  1 03:43:34 2002
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Date: Fri, 01 Feb 2002 09:41:54 +0100
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Rob Lanphier <robla@real.com>
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Subject: Re: [MMUSIC] RTSP: Unclearities regarding pause with range (477423)
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Hi, Rob

I still dont agree with the resolution. I have objections against the following
part of a sentence:

"but if multiple range specifiers are present, npt/smpte takes precence over
time."

My interpretation for PLAY is that you always first look at the time=value. You
don't actually perform the request until your wall clock is equal or after the
time specifed in time=. Then when performing the PLAY you will look and perform
as the npt/smpte/utc specifies. So I would like to remove the part about
npt/smpte taking presendence as it changes the behaviour compared to PLAY. My
opinion is that they should use the same mechanism as it simplifies
implementation and also align behviour.

What is your opinion?


Rob Lanphier wrote:

> *  RESOLVED: Only a single range specifier (format "value-" where "value"
> is a time) is allowed for pause. time="execution time in UTC" may also be
> used, but if multiple range specifiers are present, npt/smpte takes
> precence over time. A new example will be provided.
>
> Rob
> ----

Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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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		: An Offer/Answer Model with SDP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-mmusic-sdp-offer-answer-00.txt
	Pages		: 24
	Date		: 31-Jan-02
	
This document defines a mechanism by which two entities can make use
of SDP to arrive at a common view of a multimedia session between
them. In the model, one participant offers the other a description of
the desired session from their perspective, and the other participant
answers with the desired session from their perspective. This
offer/answer model is most useful in unicast sessions where
information from both participants is needed for the complete view of
the session. The offer/answer model is used by protocols like the
Session Initiation Protocol (SIP).

A URL for this Internet-Draft is:
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From mmusic-admin@ietf.org  Fri Feb  1 12:08:41 2002
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Date: Fri, 1 Feb 2002 09:04:25 -0800 (PST)
From: Rob Lanphier <robla@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP: Unclearities regarding pause with range (477423)
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I don't have a strong opinion on this myself, but this is probably one I
should ask around about here at RN.  To be honest, in my haste to get the
email out last night, I overlooked that part of the discussion on the
list.

I'll get back to you next week on this one.

Rob

On Fri, 1 Feb 2002, Magnus Westerlund wrote:

> Hi, Rob
>
> I still dont agree with the resolution. I have objections against the following
> part of a sentence:
>
> "but if multiple range specifiers are present, npt/smpte takes precence over
> time."
>
> My interpretation for PLAY is that you always first look at the time=value. You
> don't actually perform the request until your wall clock is equal or after the
> time specifed in time=. Then when performing the PLAY you will look and perform
> as the npt/smpte/utc specifies. So I would like to remove the part about
> npt/smpte taking presendence as it changes the behaviour compared to PLAY. My
> opinion is that they should use the same mechanism as it simplifies
> implementation and also align behviour.
>
> What is your opinion?
>
>
> Rob Lanphier wrote:
>
> > *  RESOLVED: Only a single range specifier (format "value-" where "value"
> > is a time) is allowed for pause. time="execution time in UTC" may also be
> > used, but if multiple range specifiers are present, npt/smpte takes
> > precence over time. A new example will be provided.
> >
> > Rob
> > ----
>
> Regards
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research ERA/T/VA
> ----------------------------------------------------------------------
> Ericsson Radio Systems AB  | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>
>


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From mmusic-admin@ietf.org  Fri Feb  1 18:39:18 2002
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Date: Fri, 1 Feb 2002 15:39:17 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] RTSP Telecon Agenda - 06 Feb
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Below is the agenda for the RTSP telecon that will happen on Wednesday,
February 6 at 9 am PST (17:00 GMT).  Please RSVP no later than
end-of-business Monday (PST).

Agenda also available online:
http://rtsp.org/2002/telecon/agenda06Feb.html

Thanks
Rob
----
Rtsp Tc Agenda06 Feb2002

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

  1. Last Telecon
       1. Review Action Items/New Issues
       2. Discussed and Resolved
       3. Discussed and still pending
  2. This Telecon

Last Telecon

Minutes: http://rtsp.org/2002/telecon/minutes23Jan.html

Review Action Items/New Issues

   * ACTION ITEM (RL): submit proposal to eliminate queued play
   * ACTION ITEM (MW): continuous sequence number filing
   * ACTION ITEM (JS): Write up aggregate URL proposal.

Discussed and Resolved

   * http://rtsp.org/bug477421 - When to send response
        o RESOLVED: server allowed to send immediate response with no
          sequence number. server should NOT set sequence number
   * http://rtsp.org/bug477425 - Inconsistency between timeformats
        o RESOLVED: Extend this to other time formats, and probably clarify
          the semantics of this particular syntax. Should start at pause
          point or the beginning.
   * http://rtsp.org/bug477427 - Transport model requirements
        o RESOLVED: define header and response for non-persistent
          connections. Default is connection-oriented, and connectionless
          MAY be implemented. Look into "supported" header.

Discussed and still pending

   * http://rtsp.org/bug477423 - Unclearities regarding pause with range
        o RESOLVED: Only a single range specifier (format "value-" where
          "value" is a time) is allowed for pause. time="execution time in
          UTC" may also be used. A new example will be provided.
        o PENDING: If multiple range specifiers are present, we need to
          determine which takes precedence (npt/smpte or time?).
   * http://rtsp.org/bug477424 - How to queue timed plays?
        o PENDING: propose elimination of queued play to list. Separate
          email necessary. Postpone 477424 until queue play is resolved
   * http://rtsp.org/bug477428 - Priorites between aggregated & non cntrl
        o PENDING: await resolution of queued play. Aggregate control issue
          needs larger discussion

This Telecon

A little bit different format this time around. We've got two big,
interrelated topics, for which there are a series of subissues. I'd like to
come to some sort of consensus on the two issues

   * Mixing Aggregate and Non-aggregate Control - see the email thread. This
     is related to several bugs:
        o http://rtsp.org/bug477400 - Play without range while playing
        o http://rtsp.org/bug477428 - Priorites between aggregated & non
          cntrl
        o http://rtsp.org/bug483356 - media URL in Play on aggregated
          session
        o http://rtsp.org/bug493270 - PAUSE/Play on single media in
          aggregate
        o http://rtsp.org/bug506412 - RTSP Teardown functionallity
   * Eliminate queued play - see the email thread and
     http://rtsp.org/bug508211 . This relates to several issues
        o http://rtsp.org/bug477424 - How to queue timed plays?
        o http://rtsp.org/bug477432 - RTP info and queued plays
        o http://rtsp.org/bug493265 - Pause - Play behaviour clarification
        o http://rtsp.org/bug507882 - RTP timestamps when playing after
          pause


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From mmusic-admin@ietf.org  Sat Feb  2 12:01:00 2002
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Date: Sat, 02 Feb 2002 17:59:46 +0100
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Subject: [MMUSIC] WG Last Call for Offer/Answer
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Folks,

this is to announce Working Group Last Call on

        draft-ietf-mmusic-sdp-offer-answer-00.txt

for Proposed Standard.  Please note that this document is a
revision of draft-rosenberg-mmusic-offer-answer-00.txt.

The WG Last Call is to expire in two weeks, on 16 Feb 2002.

Please note that this document is referenced by SIP bis06
and thus is crucial to the 3G work.  So, please, review this
document quickly (it is not that long) and should you 
have any comments, please submit them asap to the MMUSIC
mailing list and the authors.

Thanks a lot,
Joerg
co-chair, MMUSIC

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Subject: [MMUSIC] Question About SDO Offer/Answer draft
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In section 4 "Protocol Operatioin":

It says that the agent Must Not generate a new offer if it has received an offer which it has not yet answered or rejected.

But for early media we use 180/PRACK to offer and answer the early media just before

the Invite/200 interaction is completed.

Is this method break the Offer/Answer law?


----- Original Message ----- 
From: Joerg Ott <jo@tzi.uni-bremen.de>
To: <mmusic@ietf.org>
Cc: <mankin@isi.edu>; <csp@isi.edu>; <jdrosen@dynamicsoft.com>
Sent: Sunday, February 03, 2002 12:59 AM
Subject: WG Last Call for Offer/Answer


> Folks,
> 
> this is to announce Working Group Last Call on
> 
>         draft-ietf-mmusic-sdp-offer-answer-00.txt
> 
> for Proposed Standard.  Please note that this document is a
> revision of draft-rosenberg-mmusic-offer-answer-00.txt.
> 
> The WG Last Call is to expire in two weeks, on 16 Feb 2002.
> 
> Please note that this document is referenced by SIP bis06
> and thus is crucial to the 3G work.  So, please, review this
> document quickly (it is not that long) and should you 
> have any comments, please submit them asap to the MMUSIC
> mailing list and the authors.
> 
> Thanks a lot,
> Joerg
> co-chair, MMUSIC
> 

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From mmusic-admin@ietf.org  Sun Feb  3 21:35:31 2002
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Peili Xu wrote:
> 
> In section 4 "Protocol Operatioin":
> 
> It says that the agent Must Not generate a new offer if it has received
> an offer which it has not yet answered or rejected.
> 
> But for early media we use 180/PRACK to offer and answer the early media
> just before
> 
> the Invite/200 interaction is completed.
> 
> Is this method break the Offer/Answer law?

No. It works like this:


INVITE (offer 1) -->
<-----183 (answer 1)-

<------200 OK (offer 2)--
-------ACK (answer 2)---->

that is, the 200 OK is a new offer, and then SDP is present in the ACK
with an answer. 

This is different from the "common" practice today for early media,
which has the 183 as an anwer, with a 200 OK that repeats the same SDP.
I believe there will largely be backwards compatibility. If the UAS does
the "common" thing when talking to an element compliant to bis/o-a, it
will send SDP in the ACK, but so long as its not different from the
offer in the INVITE, there won't be any issues. 

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [MMUSIC] RE: Last Call: Support for IPv6 in SDP to Proposed Standard
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	Hello Gonzales,

In message <3C59AA98.AEA5FAE5@lmf.ericsson.se> Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se> writes:
>Therefore, comments about the actual format are very welcome. Other
>comments are interesting for other documents, but I do not think the
>IESG has to be involved in such a discussion until those documents are
>done. 

	Would it be impossible (or too late) to align your spec with the
	updated format in sdp-new-04? It fixes the errors in your ABNF
	and it is mostly consistent with the syntax in RFC 2327.

	I have cut and pasted relevant ABNF from sdp-new-04 and RFC 2327
	(FQDN and TTL are from old SDP) below. It is not strictly
	compatible with RFC 2327 as it fixes a bug with
	multicast-address rule (which could have more than 4 octets in
	RFC 2327):

--8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<--
3. Syntax

   RFC2373 [1] gives an ABNF for the text representation of IPv6 addresses in 
   Appendix B. RFC2732 [3] covers the text representation of IPv6 addresses 
   when used within a URL. Using the ABNF described in these documents, the 
   following updated ABNF for SDP is proposed.

   connection-address =  multicast-address
                         | addr

   multicast-address =   addr "/" ttl [ "/" integer ]
        ;IPv4 multicast addresses must be in the range
        ;224.0.0.0 to 239.255.255.255
        ;IPv6 multicast addresses must begin with the byte FF
        ;or include an IPv4 multicast address

   ttl =                 decimal-uchar

   addr =                FQDN | unicast-address

   unicast-address =     IP4-address | IP6-address

   FQDN =                4*(alpha-numeric|"-"|".")
                         ;fully qualified domain name as specified in RFC1035

   ; IP4-address, IP6-address: From RFC 2373
   IP6-address =         hexpart [ ":" IP4-address ]
   IP4-address =         1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT

   hexpart =             hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ]
   hexseq =              hex4 *( ":" hex4)
   hex4 =                1*4HEXDIG

   uri =                 URI-reference ; defined in RFC2396/2732

   ; decimal-uchar, integer, and alpha-numeric are defined in RFC2327

-->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8--

	You could also discuss how TTL should be used with IPv6
	addresses. While TTL is nice thing and makes it easy to
    	recognize multicast addresses, it is not compatible with RFC2327
    	syntax. So, I propose something like following:

--8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<----8<--
4.2 IPv6 and Multicast Addresses

    The IPv6 multicast addresses have the following format:

   |   8    |  4 |  4 |                  112 bits                   |
   +------ -+----+----+---------------------------------------------+
   |11111111|flgs|scop|                  group ID                   |
   +--------+----+----+---------------------------------------------+

    A 8-bit prefix of all ones at the start of the address identitifies
    a multicast address. The text representation of the IPv6 multicast
    addresses begins always with a four-digit hex segment starting with
    "FF".

    Following the prefix are three reserved bits, initialized to zero,
    and a flag indicating how the address has been assigned. In text
    representation, the third hex-digit indicates if the multicast
    address is well-known (even digit) or transient (odd digit).

    The following four bits, scop, indicates the scope of the multicast
    address. The scop is the fourth hex-digit in the text
    representation. There is no TTL in IPv6, but the scope and group ID
    together identifies the multicast group.

    For multicast addresses mapped uniquely to IEEE 802 MAC addresses,
    the next 80 bits or 20 hex digits are reserved and must be zero. The
    last 32 bits identitifies the group within given scope.

   |   8    |  4 |  4 |          80 bits          |     32 bits     |
   +------ -+----+----+---------------------------+-----------------+
   |11111111|flgs|scop|   reserved must be zero   |    group ID     |
   +--------+----+----+---------------------------+-----------------+

4.2.1 Using TTL with IPv6 Multicast Address

    Conferences using an IPv6 multicast connection address SHOULD also
    have a time to live (TTL) value present in addition to the multicast
    address. The TTL and the IPv4 address together define the scope with
    which multicast packets sent in this conference will be sent. When
    an IPv4-compatible-IPv6 or IPv4-mapped-IPv6 address is used, TTL
    value MUST be in the range 0-255.

    When an IPv6 multicast connection address is used, a TTL value of 0
    SHOULD is included.

    The TTL for the session is appended to the address using a slash as
    a separator.  Examples are:

         c=IN IP6 FF1E::172A:1E24/0

         c=IN IP6 ::233.42.30.36/16
-->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8---->8--


						Pekka



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From mmusic-admin@ietf.org  Mon Feb  4 13:18:17 2002
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SUBJECT: Fw: Re: [MMUSIC] updated offer/answer draft
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Hi Folks,

It is specified in the section 6.1 of the new offer-answer draft that recvonly stream MAY indicate additional media formats, not listed in the offer, that the answerer is willing to receive with. Is it allowed in sendrecv streams?
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
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From mmusic-admin@ietf.org  Mon Feb  4 21:20:41 2002
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Date: Mon, 4 Feb 2002 18:21:59 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
To: Jonathan Sergent <sergent@eng.sun.com>
cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Mixing aggregate and non-aggregate control (long)
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Hi Jonathan,

I've floated your proposal around here at RN, and I think we agree with
most of what you lay out.  More inline.

On Fri, 25 Jan 2002, Jonathan Sergent wrote:
> (a) PLAY should only be allowed for playing everything that was SETUP.
>     It should be an error to PLAY a stream URL on a session where multiple
>     stream URLs have been SETUP.

Agreed.

> (b) PAUSE should then be modified so that only PAUSE of the URL that was
>     PLAYed is legal, to remove the other half of the "mute" functionality.

Also agreed.

> (b) New methods: MUTE and UNMUTE, to reintroduce mute functionality
>     -- Can be done as an extension rather than part of 2326
>     -- Either way, it should be optional

I think an extension is certainly the safer route to take.  If it's part
of a new RTSP draft, it might not be on the same interop timescale as the
rest of the draft.  Since it'd be an optional extension, it makes sense as
an extension.

>     -- Can be detected via OPTIONS (or by trying and letting it fail;
>        but the server shouldn't close the connection or session for an
>        unknown method) since they are new methods

Agreed.  That's part of what we like about this.

>     -- This could also be done with a pre-defined parameter instead of
>        a new method.

This seems new and different enough to justify a new method.  It's not
like method names are any more expensive than headers, and the semantics
are such that it is different.

>     -- Can only be used on a stream URL which has been SETUP for the
>        same session.  Error to use it on an aggregate URL.
>     -- "Mutedness" of a stream is not reset by doing PAUSE or PLAY
>     -- MUTE can be sent before PLAY and will persist

Agreed on all points.  One thing we might want to consider is whether or
not it is/can be an error to *not* SETUP one of the streams in an
aggregate session.  In my book, that's fine...the stream is just MUTEd by
default.  If we allow this, though, then we have to consider the
possibility of a SETUP after PLAY (and whether or not that's an implicit
UNMUTE).

>     -- Range header can be used to schedule MUTE and UNMUTE operations.
>        If queueing is to be performed, MUTE and UNMUTE requests are
>        queued the same way that PLAY requests are.

Hee hee....assuming we can agree on how PLAY requests get queued.  :)

In general, this is a good principle.

> (c) Deprecate the usage of PLAY with no range header on a playing stream
>     as a liveness ping.  Allow it to remain a no-op for a playing stream,
>     but suggest that clients use a SET_PARAMETER or an OPTIONS
>     which mentions the session identifier as a keep-alive.  Could introduce
>     a PING method, but I think this is unnecessary.

I agree with Magnus on this.  We could continue to "harmlessly"  overload
other methods, but that's the same trap that got us where we are.  It
seems that a PING would be so well understood as a method, and so easily
implemented, that we should just add it.

Where I disagree with Magnus is that I don't think we can require this
from clients and servers, but we can require the following:

1.  Implementations MUST not disconnect when rejecting this method (or any
well-formed new method)
2.  PING senders MUST be prepared to accept "405 Method not allowed", and
treat that as an acceptable liveness check.

> (d) Queued play should either be removed or be made optional.
>     If the server does not do queued play, then PLAY requests with
>     range headers are invalid while the server is playing and should
>     generate "method not valid in this state".

Agreed.

Rob



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Subject: [MMUSIC] Question about QoS parameters in SDPng
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Hello,

1) Draft-ietf-mmusic-sdpng-req-01.txt states that there is a requirement
from the IETF Multiparty Multimedia Session Control (MMUSIC) working
group to specify additional QoS parameters for SDPng. Which QoS
parameters did the authors have in mind? And how would they be used? Has
there been any work on the topic of QoS parameters in SDPng lately?
Personally I have some ideas about how it could work but I would like to
open the discussion to everyone.
 
2) Realistically speaking by when will the features for SDPng be
freezed? In the last version of the MMUSIC charter sent around on the
mailinglist I found 'Apr 02: Submit 'SDPng base spec' to IESG for
publication as Proposed Standard. Is this date feasible? If not, could
someone comment on a new feasible date? By when should the features be
frozen in order to get them included in the SDPng base spec?

Thanks,

Lieve

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From mmusic-admin@ietf.org  Tue Feb  5 10:44:58 2002
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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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        "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Mixing aggregate and non-aggregate control (long)
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Hi,

See comments below.

Rob Lanphier wrote:

> >     -- Can only be used on a stream URL which has been SETUP for the
> >        same session.  Error to use it on an aggregate URL.
> >     -- "Mutedness" of a stream is not reset by doing PAUSE or PLAY
> >     -- MUTE can be sent before PLAY and will persist
>
> Agreed on all points.  One thing we might want to consider is whether or
> not it is/can be an error to *not* SETUP one of the streams in an
> aggregate session.  In my book, that's fine...the stream is just MUTEd by
> default.  If we allow this, though, then we have to consider the
> possibility of a SETUP after PLAY (and whether or not that's an implicit
> UNMUTE).
>

My opinion is that we must allow to setup a single media from a presentation
containing multiple medias, as this is non-aggregate control. Some clients may
desire to do this in certain presentations. I believe disallowing doing single
stream control with play and pause in an aggregated session, i.e. a session with
two or more different medias setuped, still cleans up this a lot. Then
introducing an extension for MUTE and UNMUTE giving a that functionality. This
seems to provide all functionality that I can think of.

I do have a question regarding aggregated control. Is it allowed to setup two
medias coming from different presentations? I think this is a special case of
creating aggregates from medias in individual files and not in any container
files.

> > (c) Deprecate the usage of PLAY with no range header on a playing stream
> >     as a liveness ping.  Allow it to remain a no-op for a playing stream,
> >     but suggest that clients use a SET_PARAMETER or an OPTIONS
> >     which mentions the session identifier as a keep-alive.  Could introduce
> >     a PING method, but I think this is unnecessary.
>
> I agree with Magnus on this.  We could continue to "harmlessly"  overload
> other methods, but that's the same trap that got us where we are.  It
> seems that a PING would be so well understood as a method, and so easily
> implemented, that we should just add it.

> Where I disagree with Magnus is that I don't think we can require this
> from clients and servers, but we can require the following:
>
> 1.  Implementations MUST not disconnect when rejecting this method (or any
> well-formed new method)
> 2.  PING senders MUST be prepared to accept "405 Method not allowed", and
> treat that as an acceptable liveness check.
>

I believe that point 2 is in error. I agree that it must be ready to accept 405
or 501 Not Implemented. But it can't treat it as a an acceptable keep alive
mechanism. The session id probably never was interpreted and the timer not reset.
I can't see any reasons not to mandate PING for all implementations. When most
implementations support the new spec keep alive and livness checks can be done in
both directions (client<-->server). I do understand that a fallback mechanism
should be specified. In my opinion the livness check should be done in the
following way:

1. Try required method PING. If a old non upgraded implementation is used, on
error message try both 2.
2a. Try Options with session ID. I am uncertain if most implementations support
this? The problem is that you don't know if the session timer was reset.
2b. If SET_PARAMETER with session ID is implemented it should also work. But as
this is outside the minimal implementation it is uncertain if it will work.


Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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From mmusic-admin@ietf.org  Tue Feb  5 17:04:45 2002
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Roman Shpount wrote:
> 
> Hi Folks,
> 
> It is specified in the section 6.1 of the new offer-answer draft that
> recvonly stream MAY indicate additional media formats, not listed in the
> offer, that the answerer is willing to receive with. Is it allowed in
> sendrecv streams?

This is a good question.

In the sendrecv case, if the answerer adds a codec X, it cant ever send
with X, since it doesn't know if its supported by the other side. Thus,
its not actually being used as sendrecv, and therefore, does not belong
in the list as a valid sendrecv codec. That is different for recvonly. 

I could be convinced to remove this bit in the case of recvonly, to make
things simple and concise. The answer is always a subset, period.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Tue Feb  5 18:06:38 2002
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The reasoning behind this question is the same as the reasoning used to justify and support different payload type numbers in reply to the sendrecv offer. If we follow this logic, a called party gets an offer, which specifies what the calling party is willing to receive and replies with an answer that lists everything that called party is willing to receive for each of the offered streams. Those two offers must intersect as far as capabilities are concerned but both can include additional formats and use different payload types. Actually, what would be ideal is to have an ability to send two independent lists of formats that application is willing to send and to receive on the sendrecv stream. This way, for instance, application can specify its codec preferences for asymmetric connections.

Roman Shpount

------ Original message ------
>From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
>Subject: Re: Fw: Re: [MMUSIC] updated offer/answer draft
>Date: 05 Feb 02, 10:03 PM
>To: Roman Shpount <roman@atelo.com>
>Cc: mmusic@ietf.org

Roman Shpount wrote:
> 
> Hi Folks,
> 
> It is specified in the section 6.1 of the new offer-answer draft that
> recvonly stream MAY indicate additional media formats, not listed in the
> offer, that the answerer is willing to receive with. Is it allowed in
> sendrecv streams?

This is a good question.

In the sendrecv case, if the answerer adds a codec X, it cant ever send
with X, since it doesn't know if its supported by the other side. Thus,
its not actually being used as sendrecv, and therefore, does not belong
in the list as a valid sendrecv codec. That is different for recvonly. 

I could be convinced to remove this bit in the case of recvonly, to make
things simple and concise. The answer is always a subset, period.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com             
  PH:  (973) 952-5000
http://www.dynamicsoft.com






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From mmusic-admin@ietf.org  Tue Feb  5 21:14:21 2002
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Subject: [MMUSIC] Functionality missing in offer/answer
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Hi Folks,

From my point of view the following functionality is missing in offer/answer:

1. Ability to authorize media streams

SDP, due to its roots as a conference description protocol, does not specify any information about media stream origination address or any information that can be used to determine whether media was originated by a valid source. Since an end point or a firewall has no information to authorize media streams, deploying any offer/answer communication systems on public networks presents an unjustified security risk. Any party on the same network can scan a host for open UDP ports and send series of RTP packets that will interrupt all the RTP streams terminated by this host.

As far as I am aware current implementations are trying to avoid this problem by only accepting media from IP address specified as destination address for the media stream. This contradicts the current offer/answer draft and does not really solve the security problem, since origination IP address can be spoofed. The other solution is to store SDES and origination IP from the first media packets that pass probation and ignore all subsequent packets from the different IP or with different SDES. This solution is more secure, but contradicts current RTP specifications. 

Solution that I propose is to provide a way to specify SDES and/or origination IP in the offer/answer SDP. Unfortunately, this solution does not interoperate well with early media, since it will cause all the early media packets to be received but ignored.

2. Causing the other party of an already established session to send a new offer

This functionality is often needed when functionality similar to SIP REFER method is implemented using Back-to-Back SIP User Agent and re-INVITES. I believe this issue was brought up already during the discussion of the original offer/answer. Consider the following scenario:

1) a session is established through a Back-to-Back SIP User Agent between users A and B.

2) a Back-to-Back SIP User Agent needs to 

According to current specifications there is no reliable way to implement this scenario. If a new offer is received by a Back-to-Back SIP User Agent from C there is no way to ensure it will satisfy any of the requirements for the session description during session modification (not decreasing number of streams etc). On the other hand if the description of the session between A and B is used as offer to C it might be rejected due to the fact that media handling capabilities of A and C might not intersect (even though communications between C and B are possible). 

Additionally, knowledge of potential capabilities is insufficient to solve this problem, since the real offer (with valid destination ports and addresses) needs to be given to one of the parties and since this offer needs to compatible with the current session. 

The proposed solution to this problem is to cause one of the parties to send an offer that will be on one hand compatible with the current session, but on the other hand will include all the end point's desired media capabilities.  

___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com





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From mmusic-admin@ietf.org  Tue Feb  5 21:25:37 2002
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inline.

Roman Shpount wrote:
> 
> Hi Folks,
> 
> >From my point of view the following functionality is missing in
> offer/answer:
> 
> 1. Ability to authorize media streams
> 
> SDP, due to its roots as a conference description protocol, does not
> specify any information about media stream origination address or any
> information that can be used to determine whether media was originated
> by a valid source. Since an end point or a firewall has no information
> to authorize media streams, deploying any offer/answer communication
> systems on public networks presents an unjustified security risk. Any
> party on the same network can scan a host for open UDP ports and send
> series of RTP packets that will interrupt all the RTP streams terminated
> by this host.
> 
> As far as I am aware current implementations are trying to avoid this
> problem by only accepting media from IP address specified as destination
> address for the media stream. This contradicts the current offer/answer
> draft and does not really solve the security problem, since origination
> IP address can be spoofed. The other solution is to store SDES and
> origination IP from the first media packets that pass probation and
> ignore all subsequent packets from the different IP or with different
> SDES. This solution is more secure, but contradicts current RTP
> specifications.
> 
> Solution that I propose is to provide a way to specify SDES and/or
> origination IP in the offer/answer SDP. Unfortunately, this solution
> does not interoperate well with early media, since it will cause all the
> early media packets to be received but ignored.

That provides pretty much no security, since its totally spoofable.

The right way is to use SRTP. It will provide true cryptographic
authentication. You will be able to signal SRTP usage in SDP, and thus,
in offer/answer, and have proper security.


> 
> 2. Causing the other party of an already established session to send a
> new offer
> 
> This functionality is often needed when functionality similar to SIP
> REFER method is implemented using Back-to-Back SIP User Agent and
> re-INVITES. I believe this issue was brought up already during the
> discussion of the original offer/answer. Consider the following
> scenario:
> 
> 1) a session is established through a Back-to-Back SIP User Agent
> between users A and B.
> 
> 2) a Back-to-Back SIP User Agent needs to
> 
> According to current specifications there is no reliable way to
> implement this scenario. If a new offer is received by a Back-to-Back
> SIP User Agent from C there is no way to ensure it will satisfy any of
> the requirements for the session description during session modification
> (not decreasing number of streams etc). On the other hand if the
> description of the session between A and B is used as offer to C it
> might be rejected due to the fact that media handling capabilities of A
> and C might not intersect (even though communications between C and B
> are possible).
> 
> Additionally, knowledge of potential capabilities is insufficient to
> solve this problem, since the real offer (with valid destination ports
> and addresses) needs to be given to one of the parties and since this
> offer needs to compatible with the current session.
> 
> The proposed solution to this problem is to cause one of the parties to
> send an offer that will be on one hand compatible with the current
> session, but on the other hand will include all the end point's desired
> media capabilities.

This is covered thorougly in draft-ietf-sip-rfc2543bis-07. It has
nothing to do with offer answer. O/A doesn't tell you how or when you
send offers, only what the exchange needs to be when you do. O/A also
doesn't tell you how to formulate offers, only provides guidelines for
how you can. In bis-07, it explicilty mentions in Section 14.2, lines
2278 - 2285, which read:

A UAS providing an offer in a 2xx (because the INVITE did not contain an
offer) SHOULD construct the offer as if the UAS were making a brand new
call, subject to the constraints of sending an offer which 2279
updates an existing session, as described in [1] in the case of SDP.
Specifically, this means that it SHOULD 2280
include as many media formats and media types that the UA is willing to
support. The UAS MUST ensure 2281
that the session description overlaps with its previous session
description in media formats, transports, or 2282
other parameters that require support from the peer. This is to avoid
the need for the peer to reject the session 2283
description. If, however, it is unacceptable to the UAC, the UAC SHOULD
generate an answer with a valid 2284
session description, and then send a BYE to terminate the session. 2285


bis has always allowed one side to "prod" the other into making the
offer by sending an INVITE without SDP, and that text still exists in
bis-07.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb  6 11:53:35 2002
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Date: Wed, 06 Feb 2002 11:50:39 -0500
From: Aravind Narasimhan <aravind.narasimhan@sun.com>
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
CC: Jonathan Sergent <sergent@eng.sun.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Mixing aggregate and non-aggregate control (long)
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Please see my comments inline:

[snip]

>>Scenario #6:
>>Initial connection setup, followed by:
>>--------------------------------------------------------------------
>>C->S
>>PLAY rtsp://host/a/ RTSP/1.0
>>Cseq: 4
>>Session: ZZEfpZ0zxK5
>>Range: npt=0-30
>>
>>--------------------------------------------------------------------
>>S->C (time = t)
>>RTSP/1.0 200 OK
>>Cseq: 4
>>Session: ZZEfpZ0zxK5
>>RTP-Info: url=rtsp://host/a/b;seq=15914;rtptime=420914,
>>          url=rtsp://host/a/c;seq=16778;rtptime=1946900992
>>
>>--------------------------------------------------------------------
>>C->S (time = t + 1)
>>PLAY rtsp://host/a/ RTSP/1.0
>>Cseq: 5
>>Session: ZZEfpZ0zxK5
>>Range: npt=45-60
>>
>>--------------------------------------------------------------------
>>S->C
>>RTSP/1.0 200 OK
>>Cseq: 5
>>Session: ZZEfpZ0zxK5
>>RTP-Info: url=rtsp://host/a/b;seq=16664;rtptime=660914,
>>          url=rtsp://host/a/c;seq=18578;rtptime=1949600992
>>
>>--------------------------------------------------------------------
>>
>>This is normal queued play.  The server plays the presentation for the
>>first thirty seconds, then immediately jumps 15 seconds ahead and plays
>>the last fifteen seconds, then pauses.
>>
>
>I would like us to consider the language we are using. I would not call it
>pauses after it has played the last 15 seconds. Rather I would call it halt or
>stops. The reason is the use of the word pause implies that the client sent a
>pause, which obviously not is the case. 
>
I dont think you can use the word "halt".  Section 10.6 of the RFC uses 
halt in its explantion of the PAUSE functionality:
Section 10.6

   The PAUSE request causes the stream delivery to be interrupted
   (halted) temporarily. If the request URL names a stream, only
   playback and recording of that stream is halted. 

 The word is too overloaded.  The defintion of  a PLAY with range 
headers uses PAUSE in it explanation too.  
Section 10.5

    After playing the desired range, the presentation is automatically
    paused, as if a PAUSE request had been issued.

obviously "halt" and "pause" were meant to be used interchangeably in 
the RFC.
 I do think that clearer terminology is warrented in a newer rev. of the 
standard.

>And the difference is rather obvious if
>you would send another play without range after the stream has halted. If it has
>halted the play starts from the beginning of stream (npt=0-). If it was paused
>it would continue where it has paused.
>
I dont think that RFC draws this disctinction at present.

>>I think these changes are warranted to clean this up:
>>
>>(a) PLAY should only be allowed for playing everything that was SETUP.
>>    It should be an error to PLAY a stream URL on a session where multiple
>>    stream URLs have been SETUP.
>>(b) PAUSE should then be modified so that only PAUSE of the URL that was
>>    PLAYed is legal, to remove the other half of the "mute" functionality.
>>(b) New methods: MUTE and UNMUTE, to reintroduce mute functionality
>>    -- Can be done as an extension rather than part of 2326
>>    -- Either way, it should be optional
>>    -- Can be detected via OPTIONS (or by trying and letting it fail;
>>       but the server shouldn't close the connection or session for an
>>       unknown method) since they are new methods
>>    -- This could also be done with a pre-defined parameter instead of
>>       a new method.
>>    -- Can only be used on a stream URL which has been SETUP for the
>>       same session.  Error to use it on an aggregate URL.
>>    -- "Mutedness" of a stream is not reset by doing PAUSE or PLAY
>>    -- MUTE can be sent before PLAY and will persist
>>    -- Range header can be used to schedule MUTE and UNMUTE operations.
>>       If queueing is to be performed, MUTE and UNMUTE requests are
>>       queued the same way that PLAY requests are.
>>(c) Deprecate the usage of PLAY with no range header on a playing stream
>>    as a liveness ping.  Allow it to remain a no-op for a playing stream,
>>    but suggest that clients use a SET_PARAMETER or an OPTIONS
>>    which mentions the session identifier as a keep-alive.  Could introduce
>>    a PING method, but I think this is unnecessary.
>>(d) Queued play should either be removed or be made optional.
>>    If the server does not do queued play, then PLAY requests with
>>    range headers are invalid while the server is playing and should
>>    generate "method not valid in this state".
>>
>
>This behavior with the mute functionality looks fine. To my understanding the
>functionality which is removed is the possibility to change the synchronization
>between media, as this is possible with multiple single media sessions, i.e.
>non-aggregated control, I don't see a problem.
>
one thing that needs to be addressed is the question of muting all the 
streams in a session.  Likewise the question of muting a stream before 
playback has begun is important to consider.

>
>
>I would argue for a mandatory to implement method PING in both servers and
>clients. This prevents overloading of functionality which created this mess in
>the first place.
>
Agreed.

Aravind




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From mmusic-admin@ietf.org  Wed Feb  6 15:19:57 2002
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Roman Shpount wrote:
> 
> The reasoning behind this question is the same as the reasoning used to
> justify and support different payload type numbers in reply to the
> sendrecv offer. If we follow this logic, a called party gets an offer,
> which specifies what the calling party is willing to receive and replies
> with an answer that lists everything that called party is willing to
> receive for each of the offered streams. 

Thats not how it works. For sendrecv streams, the caller offers what it
can send and received, and same in the answer. If both sides just send
what they can receive, you can't guarantee overlap.

> Those two offers must intersect
> as far as capabilities are concerned but both can include additional
> formats and use different payload types. Actually, what would be ideal
> is to have an ability to send two independent lists of formats that
> application is willing to send and to receive on the sendrecv stream.
> This way, for instance, application can specify its codec preferences
> for asymmetric connections.

That is a known limitiation of SDP we cannot fix in this draft. In  a
sendrecv stream, you can only specify sendrecv codecs. If you want more
than that, you can use FID, and that works just fine within the
offer/answer framework.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From mmusic-admin@ietf.org  Wed Feb  6 16:25:38 2002
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Date: Wed, 6 Feb 2002 13:25:11 -0800
From: Jonathan Sergent <Jonathan.Sergent@Sun.COM>
To: Rob Lanphier <robla@real.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Mixing aggregate and non-aggregate control (long)
Message-ID: <20020206132511.C10833@vernors>
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Sorry it took so long for me to respond.  I have been out of reach
of email...

On Mon, Feb 04, 2002 at 06:21:59PM -0800, Rob Lanphier wrote:
> I think an extension is certainly the safer route to take.  If it's part
> of a new RTSP draft, it might not be on the same interop timescale as the
> rest of the draft.  Since it'd be an optional extension, it makes sense as
> an extension.

This sounds right.  I think will want to implement it, but we don't
have a client.  Is there someone else interested in making this work?

> Agreed on all points.  One thing we might want to consider is whether or
> not it is/can be an error to *not* SETUP one of the streams in an
> aggregate session.  In my book, that's fine...the stream is just MUTEd by
> default.  If we allow this, though, then we have to consider the
> possibility of a SETUP after PLAY (and whether or not that's an implicit
> UNMUTE).

I think that not doing a SETUP is different than doing a MUTE, because
the RTP session is never established, and RTCP reports cannot be
exchanged, and so on.

The RFC is unclear on what a SETUP means after PLAY if the SETUP is for
a URL that is not already part of the session.  (If it's for the same
URL, then it's just a request to change the transport parameters, of
course.)  I think we should forbid this, because it just adds more
complexity.  It would make it so that you could use SETUP and TEARDOWN
of individual streams to do approximately the same thing as MUTE and
UNMUTE.  (I would like to leave "There's More Than One Way To Do It" to
the Perl people and say that this is probably not desirable for network
protocols.)

Since the second and further SETUP requests can be pipelined (you can
send them all at once since you have the session ID after the first
one), I don't think there is a latency argument for this, either.

> >     -- Range header can be used to schedule MUTE and UNMUTE operations.
> >        If queueing is to be performed, MUTE and UNMUTE requests are
> >        queued the same way that PLAY requests are.
> 
> Hee hee....assuming we can agree on how PLAY requests get queued.  :)

Yeah.

I unfortunately missed the call today so I don't know if we came to any
conclusions on what to do about queued play yet.  We will need to
figure out a resolution for that before we could move forward with an
I-D on a MUTE/UNMUTE extension, because it would come out
differently...

> > (c) Deprecate the usage of PLAY with no range header on a playing stream
> >     as a liveness ping.  Allow it to remain a no-op for a playing stream,
> >     but suggest that clients use a SET_PARAMETER or an OPTIONS
> >     which mentions the session identifier as a keep-alive.  Could introduce
> >     a PING method, but I think this is unnecessary.
> 
> I agree with Magnus on this.  We could continue to "harmlessly"  overload
> other methods, but that's the same trap that got us where we are.  It
> seems that a PING would be so well understood as a method, and so easily
> implemented, that we should just add it.
> 
> Where I disagree with Magnus is that I don't think we can require this
> from clients and servers, but we can require the following:
> 
> 1.  Implementations MUST not disconnect when rejecting this method (or any
> well-formed new method)
> 2.  PING senders MUST be prepared to accept "405 Method not allowed", and
> treat that as an acceptable liveness check.

Can we add PING to the base specification and still go to draft standard?
I sympathize with the desire to avoid overloading, but I think it is a lot
easier to just keep the current use of OPTIONS.  I think we can agree that
using OPTIONS as a keep-alive is not going away, and I would prefer for us
to not introduce two ways of doing the same thing.

On (1) I agree 100%; on (2) I think it would be a 501 Not Implemented.
405 would imply that the server understood the PING but didn't allow it.
It seems like this would imply PING having failed rather than succeeded.
If we have PING, I think the client should be prepared to receive one of
200, 454, or 501.

I think 405 means that the method is not valid for the given URL, i.e.
if you try to send a web server a POST for an ordinary HTML file.  So
one final question about PING is what should be done with the URL
field.  (I wonder the same thing about OPTIONS, though; what should the
server do if you do an OPTIONS with a Session header on a URL that is
not part of that session?)


--jss.

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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

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When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-448521 ] URLs in Rtp-Info need to be quoted (noreply@sourceforge.net)
   2. [ rtspspec-Bugs-477404 ] Errors in table in chapter 12 (noreply@sourceforge.net)
   3. [ rtspspec-Bugs-511233 ] Timed PLAY requests has RTP-Info problem (noreply@sourceforge.net)
   4. [ rtspspec-Bugs-511686 ] Timed PLAY requests has RTP-Info problem (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 29 Jan 2002 17:54:15 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-448521 ] URLs in Rtp-Info need to be quoted

Bugs item #448521, was opened at 2001-08-06 13:01
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=448521&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Rob Lanphier (robla)
Summary: URLs in Rtp-Info need to be quoted

Initial Comment:
The Rtp-Info header contains URLs, but those URLs are 
not quoted, causing problems when the URLs contain 
semi-colon, equal signs and comma characters.

The best way to quote URLs are by using double-
quotes.  Using angle-brackets has also been proposed, 
but using double-quotes is more likely to be 
compatbile with existing implementations, because 
other RTSP/HTTP headers use double-quote characters 
in similar situations.
 

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 14:20

Message:
Logged In: YES 
user_id=437905

RESOLUTION: (Jan 9, 2002 telecon) 
Proposed solution was accepted.  Alternative use of 
semicolon and comma were considered  by classified 
as "dangerous characters" ans would break existing 
implementations.


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

Comment By: Henning Schulzrinne (hgschulz)
Date: 2002-01-09 09:20

Message:
Logged In: YES 
user_id=115926

URL escaping (%xx) doesn't work since you lose the
distinction between syntactically significant semicolons
(etc.) and just characters that were escaped in the original
URL.

Only comma and semicolon need to be escaped, since equal
signs are not delimiters.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 09:26

Message:
Logged In: YES 
user_id=3796

I should probably clarify the comments made yesterday.  This is a genuine issue, but
it needs to be weighed against backward-compatibility needs.  Would like input from
implementors on what the best solution is.

We would propose the following: 
*  We need to define the problem characters.  ";" and "=" are two, there may be others
*  URLs MUST not be quoted when no problem characters are present in the URL.  This will help maintain 
backwards-compatibilty with current implementations
*  URLs MUST be quoted when problem characters are present



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

Comment By: Rob Lanphier (robla)
Date: 2001-12-12 14:14

Message:
Logged In: YES 
user_id=3796

add quotes only when sending problem characters (";", "=", ",", need to think further)

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

Comment By: Rob Lanphier (robla)
Date: 2001-08-07 07:21

Message:
Logged In: YES 
user_id=3796

This is primarily to test if mail to confctrl is working

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=448521&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 29 Jan 2002 18:14:04 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477404 ] Errors in table in chapter 12

Bugs item #477404, was opened at 2001-11-01 23:22
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477404&group_id=23194

Category: None
>Group: Fix for DS - Large task
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Errors in table in chapter 12

Initial Comment:
 Table with RTSP headers page 45, contains the following errors?:
  - A session header is not required when using describe.
  - Proxy Authenticate is missing all values on what it is.
  - The transport header is specifed as required in 12.0 table, but
    the text in 12.39 says it MAY be included in requests and responses.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 14:41

Message:
Logged In: YES 
user_id=437905

RESOLUTION:  

The latest SIP draft has an equivalent table. It has been  
recommended that we evaluate this SIP table and unless 
someone objects and justifies the need to make a change 
let's adopt the SIP draft specification version. 

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

Comment By: Rob Lanphier (robla)
Date: 2002-01-22 19:47

Message:
Logged In: YES 
user_id=3796

Here's a discrepency between the initial CVS checkin and RFC 2326
  12  Header Field Definitions (Table 3):
   Change from:
   Content-Type         r      req.      entity
   To:
   Content-Type        e    req.   SET_PARAMETER, ANNOUNCE


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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 09:53

Message:
Logged In: YES 
user_id=3796

These are all good examples of what's wrong with this table.  We need to do an audit of the table, and we 
may restructure this table for clarity.  Suggestions for a new table format would be appreciated.

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477404&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 31 Jan 2002 07:37:54 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-511233 ] Timed PLAY requests has RTP-Info problem

Bugs item #511233, was opened at 2002-01-31 07:37
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=511233&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Timed PLAY requests has RTP-Info problem

Initial Comment:
In a PLAY with a range header, "time=" can be used to give activation time. This request shall be 
responded immediately to the client. In the response the server should give synchronization data 
like RTP TS which will be used when the PLAY is activated sometime in the future. It is hard to 
impossible for the server to produce such data. If no other PLAY request is done between request 
time and activation it is possible to give an accurate TS and sequnce number. But if another 
request is activated in the time between, the given RTP TS or sequnce number will not be correct. 

This problem is the same that queued plays has, see bugreport #477432, but due to different 
reasons.

One possible solution would be to introduce a SYNC method which a server can send at or slightly 
before activation of a queued or delayed play. This method would carry a RTP-info header and a 
range header, giving the information; what is playing and the synchronization information. 

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=511233&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Fri, 01 Feb 2002 05:17:27 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-511686 ] Timed PLAY requests has RTP-Info problem

Bugs item #511686, was opened at 2002-02-01 05:17
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=511686&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Timed PLAY requests has RTP-Info problem

Initial Comment:
In a PLAY with a range header, "time=" can be used to give activation time. This request shall be 
responded immediately to the client. In the response the server should give synchronization data 
like RTP TS which will be used when the PLAY is activated sometime in the future. It is hard to 
impossible for the server to produce such data. If no other PLAY request is done between request 
time and activation it is possible to give an accurate TS and sequnce number. But if another 
request is activated in the time between, the given RTP TS or sequnce number will not be correct. 

This problem is the same that queued plays has, see bugreport #477432, but due to different 
reasons.

One possible solution would be to introduce a SYNC method which a server can send at or slightly 
before activation of a queued or delayed play. This method would carry a RTP-info header and a 
range header, giving the information; what is playing and the synchronization information. 

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=511686&group_id=23194




End of Rtspspec-bugs Digest


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From mmusic-admin@ietf.org  Thu Feb  7 18:31:53 2002
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Date: Thu, 7 Feb 2002 15:33:24 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] RTSP: Minutes from Feb 6 Telecon
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Hi all,

Minutes from the last RTSP telecon are included below.  The hyperlinked
HTML version is available here:

http://rtsp.org/2002/telecon/minutes06Feb.html

Rob

  Rtsp Tc Minutes06 Feb2002

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

   1. Attendees <#line2>
   2. Last telecon: discussed and still pending <#line11>
         1. 477423 Unclearities regarding pause with range <#line12>
   3. This telecon <#line23>
         1. Mixing Aggregate and Non-aggregate Control <#line24>
               1. MUTE and UNMUTE discussion <#line30>
               2. 477400 Use of PLAY as aliveness check <#line62>
   4. Process for next draft <#line89>


  Attendees

    * Anne Jones (AJ)
    * Anders Klemets (AK)
    * Colin Perkins (CP)
    * Magnus Westerlund (MW)
    * Aravind Narasimhan (AN)
    * David Walker (DW)
    * Rob Lanphier (RL)



Agenda

http://rtsp.org/2002/telecon/agenda06Feb.html

(note: we only got down to http://rtsp.org/bug477400 - "Play without range
while playing")


Last telecon: discussed and still pending


    477423 Unclearities regarding pause with range

Bug report: http://rtsp.org/bug477423

Previous resolution: Only a single range specifier (format "value-"
where "value" is a time) is allowed for pause. time="execution time in
UTC" may also be used. A new example will be provided.

    * MW: any issues with my proposal? (see [[WWW]] Magnus' proposal
      <http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00148.html>)

    * RL: still discussion happening at RN
    * AN: agree with Magnus
    * RESOLVED: go with Magnus's proposal


  This telecon


    Mixing Aggregate and Non-aggregate Control

      see [[WWW]] the email thread
      <http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00115.html>


    * Point a) RESOLVED
    * First point b) RESOLVED


      MUTE and UNMUTE discussion

    * Second point b) MUTE and UNMUTE
          * AN: How about a MUTE that toggles
          * AK: How about SETUP and TEARDOWN
          * RL: I don't like SETUP and TEARDOWN as MUTE/UNMUTE
          * AN: spirit of orig RFC was that PAUSE doesn't change
            anything but whether packets are flowing
          * AK: SETUP/TEARDOWN may be appropriate when bandwidth
            reservation is involved
          * NEW ISSUE: should SETUP be allowed after a PLAY in the same
            session
          * NEW ISSUE: should TEARDOWN function on individual streams in
            a session
          * RL: back on MUTE/UNMUTE
          * AN: should consider MUTE as toggle
          * AK: client or server could become confused
          * MW: I like MUTE/UNMUTE
          * AK: can't add new methods to base spec
          * CP: depends on what path we want to go
          * RL: rather not
          * ...cycle back to proposed
          * MW: PING may need a separate thing
          * RL: MUTE/UNMUTE as separate spec
          * CP: depends on whether or not it's required
          * RL: I think it's optional
          * MW: really good functionality
          * RL: should it be required?
          * MW: should leave it as a pending issue
          * CP: write up as separate spec, and then discuss how it
            should go
          * AN: PAUSE is optional
          * AK: Can be done other ways. w/PAUSE
          * MW: Can't do this with agg control
          * AK: still, there are apps that don't need MUTE
          * RESOLVED: separate draft
          * ACTION ITEM: Call for authors on MUTE proposed. AN is
            tentatively signed up


      477400 Use of PLAY as aliveness check

    * Point C) (same as http://rtsp.org/bug477400)
          * AK: I don't think PING command is necessary. It breaks old
            implementations. 3GPP already uses OPTIONS anyway
          * MW: want to remove overloaded methods
          * RL: perhaps OPTIONS with option to switch to PING
          * MW: 3GPP might be able to get PING method
          * AK: wouldn't build something that relies on RTCP
          * MW: some cases, sure. PING should be 50-80 bytes, which is
            small enough
          * AK: What does PING with * mean?
          * RL: PING with * is a single hop mechanism. PING with host
            (and no resource) shoud be legal as well. That would be
            end-to-end
          * AN: perhaps session id in same way as PLAY
          * MW: need proposal
          * AK: aggregate session ids in PING method with comma sep list
          * MW: changing session header semantics is dangerous
          * RESOLVED: PING writeup will be done as part of next RTSP
            draft...continue discussion on mailing list
          * AK: Using PLAY with no range is still useful as a means of
            getting RTP-info
          * AN: we don't send it from our server
          * MW: state machine needs to be clarified
          * AN: play without range is start where left off
          * AN: need to clarify what PAUSE means
          * MW: will log issue on PLAY/PAUSE semantics
          * AK: need both Range header and RTP-info
          * AN: could start playback from pause point at the end of a range
          * MW: this is broken
          * AK: wouldn't mind seeing this go away
          * RESOLVED: deprecate PLAY w/o range as ping method


  Process for next draft

    * Prior to Feb 20 telecon: draft ready for group review
    * Feb 20: review on telecon
    * Prior to Feb 22: submit using standard I-D process.



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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-477404 ] Errors in table in chapter 12 (noreply@sourceforge.net)
   2. [ rtspspec-Bugs-513737 ] Unclarities when doing resetup (noreply@sourceforge.net)
   3. [ rtspspec-Bugs-513753 ] RTSP extensions and IANA (noreply@sourceforge.net)
   4. [ rtspspec-Bugs-513758 ] Proxy-Require clarification (noreply@sourceforge.net)
   5. [ rtspspec-Bugs-513787 ] Allow capability negotiations (noreply@sourceforge.net)
   6. [ rtspspec-Bugs-513875 ] Play when in state ready after playing (noreply@sourceforge.net)
   7. [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative (noreply@sourceforge.net)
   8. [ rtspspec-Bugs-513888 ] RTSP and NAT (noreply@sourceforge.net)
   9. [ rtspspec-Bugs-513891 ] Clarify use of Range in REDIRECT method (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 05:20:08 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477404 ] Errors in table in chapter 12

Bugs item #477404, was opened at 2001-11-01 23:22
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477404&group_id=23194

Category: None
Group: Fix for DS - Large task
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Errors in table in chapter 12

Initial Comment:
 Table with RTSP headers page 45, contains the following errors?:
  - A session header is not required when using describe.
  - Proxy Authenticate is missing all values on what it is.
  - The transport header is specifed as required in 12.0 table, but
    the text in 12.39 says it MAY be included in requests and responses.

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

>Comment By: Magnus Westerlund (magwes)
Date: 2002-02-06 05:20

Message:
Logged In: YES 
user_id=302620

The table is also missing the location header. 

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 14:41

Message:
Logged In: YES 
user_id=437905

RESOLUTION:  

The latest SIP draft has an equivalent table. It has been  
recommended that we evaluate this SIP table and unless 
someone objects and justifies the need to make a change 
let's adopt the SIP draft specification version. 

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

Comment By: Rob Lanphier (robla)
Date: 2002-01-22 19:47

Message:
Logged In: YES 
user_id=3796

Here's a discrepency between the initial CVS checkin and RFC 2326
  12  Header Field Definitions (Table 3):
   Change from:
   Content-Type         r      req.      entity
   To:
   Content-Type        e    req.   SET_PARAMETER, ANNOUNCE


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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 09:53

Message:
Logged In: YES 
user_id=3796

These are all good examples of what's wrong with this table.  We need to do an audit of the table, and we 
may restructure this table for clarity.  Suggestions for a new table format would be appreciated.

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477404&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 05:26:26 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513737 ] Unclarities when doing resetup

Bugs item #513737, was opened at 2002-02-06 05:26
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513737&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Unclarities when doing resetup

Initial Comment:
Setup is used to change transport parameters of a already setuped stream. What happends if the 
new SETUP contains errors? Except of responding with an error message, what happends to the 
session and transport state? It should be clarified in the specification. Which of the following 
alternatives are correct?

1. Keep the old setting and states as if the second SETUP never happend.
2. Invalidate the media in the session until a correct setup has happend.

To me case 1 seems more natural but will require more from the server. 

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513737&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 06:05:32 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513753 ] RTSP extensions and IANA

Bugs item #513753, was opened at 2002-02-06 06:05
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513753&group_id=23194

Category: None
Group: Fix for Draft Standard
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: RTSP extensions and IANA

Initial Comment:
Today the RFC 2326 talks about extensions to RTSP and also has a number of headers that uses 
parameters. In the next version the following clarifications should be done:
- Setup a IANA register for RTSP parameters.
- Rules how to register different types of parameters.
- What type of documents that are needed for extending RTSP. IANA registering of parameters.  
IETF RFC for new headers(?), methods and future protocol versions.
 

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513753&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 06:16:49 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513758 ] Proxy-Require clarification

Bugs item #513758, was opened at 2002-02-06 06:16
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513758&group_id=23194

Category: General
Group: Fix for DS - Typo/Thinko
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: Proxy-Require clarification

Initial Comment:
Today the proxy-require method says that the server should treat the proxy-require header 
identically as require. I don't understand how that mechanism work for paramters/functionallity that 
only the proxies should answer to. To avoid a negative acknowledgment the server must be aware 
of these parameters to not place them in the Unsupported header. I can't see that any proxy can 
modify the response as it will not know if the response comes from a proxy or a server. 

Can someone please clarify this?  

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--__--__--

Message: 5
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 07:33:19 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513787 ] Allow capability negotiations

Bugs item #513787, was opened at 2002-02-06 07:32
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513787&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: Allow capability negotiations

Initial Comment:
Today RTSP lacks the possibility to do any kind of capability/parameter negotiation or at least 
choices regarding media parameters. 

I would like to see possibilities for this in the future. My initial idea is to make a offer/answer model. 
This will require expanding the SETUP request to carry a message body. A message exchange could 
work like this.

1. C->S: Describe
2. S->C: Response with media description, SDP or in the future SDP-NG. This description contains 
multiple alternatives or in the case SDP-NG a complete description of the choices regarding media 
that the client can do.
3. Client selects its desired configuration.
4. C->S: Setup with message body with choosen configuration in SDP/SDP-NG.
5. S->C: Server either accepts or refuses the setup with the given configuration. Refusing should 
happend very rarely as server should not offer anything it can provide.

Such solution should try to keep the difference between media parameters and transport 
parameters. Transport parameters should be keept in the transport header to simplify processing in 
proxies and firewalls. Conclusion: only media related parameters should be negotiated with the 
offer/answer mechanism. 

To my knowledge the changes required for this is to allow a message body in SETUP requests. Also 
rules how and to what it may be used must be specified.



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--__--__--

Message: 6
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 10:43:33 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513875 ] Play when in state ready after playing

Bugs item #513875, was opened at 2002-02-06 10:43
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513875&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: Play when in state ready after playing

Initial Comment:
A client has been playing for a while and reaches the end of either media or range. At that time the 
state machine goes to state ready. When in state ready a play without range is issued. The 
question is what happends at this point:

1. The play starts from the begining of the media. 

2. If at end of media directly revert to ready, or if a play with range was the previous play 
continue at the end of that range. 

Both interpretations has been implemented. So please give your opinion on this, especially if you 
have arguments supporting one or the other, like usage cases.

 

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--__--__--

Message: 7
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 10:48:49 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513880 ] How to use Range when Scale is negative

Bugs item #513880, was opened at 2002-02-06 10:48
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513880&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Nobody/Anonymous (nobody)
Summary: How to use Range when Scale is negative

Initial Comment:
I think the spec could use an example that shows how 
to use the Range header when rewinding (Scale header 
is set to a negative value.)  Or, instead of an 
example, some explanatory text could be added.

The issue that I find a little bit confusing is how 
the server should specify the end of the range. The 
end of the range would be numerically smaller than the 
start of the range.

I think the following illustrates a valid exchange:

C->S:
PLAY rtsp://server/a RTSP/1.0
Scale: -1
Range: npt=45-

S->C:
RTSP/1.0 200 OK
Scale: -1
Range: npt=45-0

This is the type of example that I think would be nice 
to have in the spec.


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--__--__--

Message: 8
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 10:53:38 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513888 ] RTSP and NAT

Bugs item #513888, was opened at 2002-02-06 10:53
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513888&group_id=23194

Category: General
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: RTSP and NAT

Initial Comment:
RTSP today is not that NAT friendly. Can something be done to fix that problem or shouldn't at 
least a note about it be written in the new spec?

If you don't have a ALG can this problem be fixed?

Known NAT issues with RTSP is and possible solutions based on NAT and SIP work:
1. Receiver ports are specifed in transport header. Needs to either be translated by ALG or client 
must find out the ports that will be used on the public side of the NAT. Mechanism for this has 
been proposed for SIP.
2. Transport header does not support explicit specification of RTCP ports. This seem to be required 
if any NAT port discovery will be possible to use. 
3. Keep alive traffic needs to be sent over NAT translated ports when in pause to prevent binding 
timeout. Not an issue for RTCP port but also RTP ports needs to receive traffic. 



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--__--__--

Message: 9
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Wed, 06 Feb 2002 10:57:01 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513891 ] Clarify use of Range in REDIRECT method

Bugs item #513891, was opened at 2002-02-06 10:56
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513891&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Nobody/Anonymous (nobody)
Summary: Clarify use of Range in REDIRECT method

Initial Comment:
Section 10.10 says that the REDIRECT command can 
specify the Range header, to specify when the 
redirection should take place.  The example shows a 
Range header that specificies an absolute time (UTC 
time.)  Is it also possible to specify an NPT time for 
the REDIRECT command?

How should a client interpret the end of the range 
when the Range header appears in the REDIRECT command?

One way to interpret the end range time is to say that 
the client should connect back to the original server 
after that time.

So, for example, if ServerA sends this command:

REDIRECT rtsp://ServerB/foo RTSP/1.0
Range: npt=45-60

It could mean that the client should keep streaming 
from ServerA for time being, but when it reaches time 
NPT=45 in the presentation, then it should connect to 
ServerB and keep streaming from there for 15 seconds.  
After the time of the presentation reaches NPT=60, 
then the client will connect back to ServerA and 
continue streaming from there.

Somehow, I feel that this might not be exactly what 
the authors of the spec had in mind.
The spec needs to be clarified so that there is no 
ambiguity about how to interpret the Range header in 
this case. 

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End of Rtspspec-bugs Digest


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From mmusic-admin@ietf.org  Mon Feb 11 20:21:19 2002
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Date: Mon, 11 Feb 2002 20:19:52 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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Some comments below:

- p.2 says "The offer/answer model defined here is the mandatory
baseline mechanism used by the Session
Initiation Protocol (SIP) [3]". This requirement seems to belong in SIP,
not offer/answer.

- p. 3:    Definition of Answerer says "An agent capable of sending
and/or receiving media...". The ability to
send/receive media should not be included here, i.e. strike "capable of
sending and/or receiving media,"

- p. 3:    Definition of Offerer: strike "capable of sending and/or
receiving media,"

- p. 4:    "The agent receiving the offer MAY generate an answer, or it
MAY reject the offer.". At least in the unicast case, MUST you not
always generate an answer which may either accept or reject the offer ?

- p. 4:    "However, it MUST NOT generate a new offer if it has received
an offer which it has not yet answered or reject.". I'm not sure what
this buys you since each side may generate an offer simultanously and
hence you have to deal with the race anyway. (nits: "reject =>
rejected", and offers are either accepted or rejected, but always
answered)

p.4:    "The offer MAY contain zero or more media streams..." A fairly
empty statement (how do you not satisfy this ?).

Section 5.1, first paragraph:    You may want to elaborate on sendonly,
recvonly, etc as a session attribute versus as a media attribute for
specific media streams.

p. 6:    "However, for sendonly and sendrecv streams, the answer might
indicate different payload type numbers for the same codecs, in which
case, the offerer MUST send with the payload type numbers from the
answer." Should add a sentence that the answerer MUST still send using
the payload types from the offer.

p. 8, last paragraph:    Only recvonly and sendrecv streams are
discussed. For completeness, sendonly streams should be mentioned as
well. Also, consider moving this paragraph to right after the first
paragraph on p.6, where similar issues are discussed.

Section 5.2, first paragraph:    I'd suggest writing this from the
perspective of the offerer (like the rest of section 5), i.e. "answerer
can only send (receive)" => "offerer can only receive (send)".

p.9:    Should state explicitly whether sendonly and sendrecv streams in
an answer are allowed to contain any additional media formats or not.

p. 10:    "In that case, the answer MAY contain fmtp parameters, and
those MAY have the same values as those in the offer, or they MAY be
different." I agree with this and understand the difficulty in saying
something more specific, however it still leaves open the question of
how to negotiate fmtp parameters between an offer and an answer. Should
there be a sentence saying that people defining fmtp parameters should
define how they are negotiated in offer/answer ?

p. 10:    "The answerer MAY include a ptime" => "The answerer MAY
include a non-zero ptime"

p. 10:    "When sending media, it SHOULD use a packetization interval
equal to the value of the ptime attribute in the offer, if any was
present.". Since ptime is a media stream attribute, this can be
difficult to do if multiple media formats are specified. There should be
a rounding rule to account for that (e.g. use closest non-zero ptime if
you can't use the ptime specified).

p. 11:    "Similarly, the directionality information in the answer
(sendonly, recvonly, or sendrecv) MUST equal that of the offer." It may
be worth spelling it out more clearly what this means, since it differs
from the unicast.

p. 13:    "Otherwise, the media description for the removed stream
SHOULD be formatted identically to the corresponding stream in the
previous SDP."    Add ", except for the list of media formats."

p. 13:    "Removal of a media stream implies that media is no longer
sent for that stream.". Add sentence "Furthermore, when RTP was used for
the media stream, RTCP reports will no longer be sent or received."

Section 8.3.1, first paragraph:    "The offerer MUST NOT cease listening
for media ..". In practice, it will not always be possible to satisfy
this requirement, and hence it should rather be "SHOULD NOT". (Nit:
"listening for" should be "receiving" - twice in the paragraph).

Section 8.3.1, second paragraph:    "The answerer MUST NOT cease
listening for media on the old port until media arrives on the new
port.". Same comments (also "listening for" => "receiving").

Section 8.3.2, first paragraph:    "the mapping from a particular
dynamic payload type number to a particular codec MUST NOT change".
Needs to be clarified that it applies on a per media stream basis and
not across the entire session.

Section 8.4:    Section seems to be unicast specific. Both heading and
section text should reflect this.

Section 9:    Since the "capabilities" SDP is indistinguishable from an
"offer/answer" SDP, we should make it clear, that this is determined by
the context within which the SDP is provided (for example a SIP OPTIONS
as opposed to INVITE).

Section 9:    "The transport component of the "m=" line indicates the
preferred transport for that media type.". Not sure this is ideal. For
example, if I support both RTP and SRTP, do I only indicate one of them
(similar issue with  T.38 UDP or TCP) ?


Also, I have some strictly editorial nits which I'll send off-line.

-- Flemming


Joerg Ott wrote:

> Folks,
>
> this is to announce Working Group Last Call on
>
>         draft-ietf-mmusic-sdp-offer-answer-00.txt
>
> for Proposed Standard.  Please note that this document is a
> revision of draft-rosenberg-mmusic-offer-answer-00.txt.
>
> The WG Last Call is to expire in two weeks, on 16 Feb 2002.
>
> Please note that this document is referenced by SIP bis06
> and thus is crucial to the 3G work.  So, please, review this
> document quickly (it is not that long) and should you
> have any comments, please submit them asap to the MMUSIC
> mailing list and the authors.
>
> Thanks a lot,
> Joerg
> co-chair, MMUSIC
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Tue Feb 12 07:03:39 2002
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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: Session Description Protocol
	Author(s)	: M. Handley, V. Jacobson, C. Perkins
	Filename	: draft-ietf-mmusic-sdp-new-05.txt
	Pages		: 45
	Date		: 11-Feb-02
	
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-sdp-new-05.txt

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From mmusic-admin@ietf.org  Tue Feb 12 11:02:12 2002
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To: mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
In-Reply-To: Your message of "Tue, 12 Feb 2002 07:02:59 EST."
             <200202121202.HAA08627@ietf.org> 
Date: Tue, 12 Feb 2002 11:00:08 -0500
From: Colin Perkins <csp@isi.edu>
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--> Internet-Drafts@ietf.org writes:
>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)	: M. Handley, V. Jacobson, C. Perkins
>	Filename	: draft-ietf-mmusic-sdp-new-05.txt
>	Pages		: 45
>	Date		: 11-Feb-02
>	
>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-sdp-new-05.txt

Changes in this version are:
- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
  similarly for sendonly and inactive.  These only effect the RTP stream. 
- Fix ABNF to include a unicast-address element, allowing comedia to
  align with both this and RFC2327
- "s= " rather than "s=-"
- Fix ABNF to reflect that s= is mandatory
- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646

Any comments would be appreciated.

Colin

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From mmusic-admin@ietf.org  Tue Feb 12 18:39:07 2002
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Date: Tue, 12 Feb 2002 15:36:03 -0800
To: Colin Perkins <csp@isi.edu>
From: Rajesh Kumar <rkumar@cisco.com>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
Cc: mmusic@ietf.org
In-Reply-To: <200202121600.g1CG08H23917@chiron.nge.isi.edu>
References: <Your message of "Tue, 12 Feb 2002 07:02:59 EST." <200202121202.HAA08627@ietf.org>
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Colin,
I suggest that acceptance of "s=-" when there is no meaningful session name 
be retained for backwards compatibility.  After all, we're fixing an 
existing standard, and not creating a new one. Backward compatibility 
should be an important consideration. I believe "s=-" will satisfy the ABNF 
definition of the session name, and therefore there is no backward 
compatibility issue, right?

Thanks,
Rajesh

At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
>Changes in this version are:
>- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
>   similarly for sendonly and inactive.  These only effect the RTP stream.
>- Fix ABNF to include a unicast-address element, allowing comedia to
>   align with both this and RFC2327
>- "s= " rather than "s=-"
>- Fix ABNF to reflect that s= is mandatory
>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
>
>Any comments would be appreciated.
>
>Colin


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From mmusic-admin@ietf.org  Tue Feb 12 19:27:40 2002
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To: Rajesh Kumar <rkumar@cisco.com>
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
In-Reply-To: Your message of "Tue, 12 Feb 2002 15:36:03 PST."
             <4.3.2.7.2.20020212152724.020a1aa0@mira-sjc5-8.cisco.com> 
Date: Tue, 12 Feb 2002 19:27:04 -0500
From: Colin Perkins <csp@isi.edu>
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I changed it to "s= " to match offer/answer, and because I was told that
was what SIP clients expected. 

Colin



--> Rajesh Kumar writes:
>Colin,
>I suggest that acceptance of "s=-" when there is no meaningful session name 
>be retained for backwards compatibility.  After all, we're fixing an 
>existing standard, and not creating a new one. Backward compatibility 
>should be an important consideration. I believe "s=-" will satisfy the ABNF 
>definition of the session name, and therefore there is no backward 
>compatibility issue, right?
>
>Thanks,
>Rajesh
>
>At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
>>Changes in this version are:
>>- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
>>   similarly for sendonly and inactive.  These only effect the RTP stream.
>>- Fix ABNF to include a unicast-address element, allowing comedia to
>>   align with both this and RFC2327
>>- "s= " rather than "s=-"
>>- Fix ABNF to reflect that s= is mandatory
>>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
>>
>>Any comments would be appreciated.
>>
>>Colin

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From mmusic-admin@ietf.org  Tue Feb 12 19:44:02 2002
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To: Rajesh Kumar <rkumar@cisco.com>
cc: jswamina@cisco.com, smarch@cisco.com, joe Stone <joestone@cisco.com>,
        axagrawa@cisco.com, flemming Andreasen <fandreas@cisco.com>,
        bfoster@cisco.com, mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
In-Reply-To: Your message of "Tue, 12 Feb 2002 16:33:03 PST."
             <4.3.2.7.2.20020212162905.020a1ba0@mira-sjc5-8.cisco.com> 
Date: Tue, 12 Feb 2002 19:40:28 -0500
From: Colin Perkins <csp@isi.edu>
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I don't really care which is chosen, but it'd be good if we could have a
single definition. RFC 2327 doesn't specify and offer/answer uses "s= ".
I was following offer/answer, but maybe that conflicts with other use?

Colin



--> Rajesh Kumar writes:
>Some h.248/MGCP applications use the old definition. I know of several that 
>are out there in the field. Therefore, I do not know if we can or should 
>alter it, short of defining a new protocol which this draft does not 
>purport to do.
>
>Anyway, I believe s=- is still canonical since '-' is ISO 10646. So there 
>should be no backward compatibility problems.
>
>Rajesh
>
>At 07:27 PM 2/12/2002 -0500, you wrote:
>>I changed it to "s= " to match offer/answer, and because I was told that
>>was what SIP clients expected.
>>
>>Colin
>>
>>
>>
>>--> Rajesh Kumar writes:
>> >Colin,
>> >I suggest that acceptance of "s=-" when there is no meaningful session name
>> >be retained for backwards compatibility.  After all, we're fixing an
>> >existing standard, and not creating a new one. Backward compatibility
>> >should be an important consideration. I believe "s=-" will satisfy the ABNF
>> >definition of the session name, and therefore there is no backward
>> >compatibility issue, right?
>> >
>> >Thanks,
>> >Rajesh
>> >
>> >At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
>> >>Changes in this version are:
>> >>- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
>> >>   similarly for sendonly and inactive.  These only effect the RTP stream.
>> >>- Fix ABNF to include a unicast-address element, allowing comedia to
>> >>   align with both this and RFC2327
>> >>- "s= " rather than "s=-"
>> >>- Fix ABNF to reflect that s= is mandatory
>> >>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
>> >>
>> >>Any comments would be appreciated.
>> >>
>> >>Colin

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From mmusic-admin@ietf.org  Tue Feb 12 19:49:47 2002
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Date: Tue, 12 Feb 2002 16:46:27 -0800
To: Colin Perkins <csp@isi.edu>
From: Rajesh Kumar <rkumar@cisco.com>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
Cc: jswamina@cisco.com, smarch@cisco.com, joe Stone <joestone@cisco.com>,
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--=====================_833386246==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:
>I don't really care which is chosen, but it'd be good if we could have a
>single definition. RFC 2327 doesn't specify and offer/answer uses "s= ".
>I was following offer/answer, but maybe that conflicts with other use?

It does conflict, as I indicated. Since offer/answer is a draft, why not 
stick to s=- as stated in RFC 2327? What does the change buy for 
offer/answer?-Rajesh


>Colin
>
>
>
>--> Rajesh Kumar writes:
> >Some h.248/MGCP applications use the old definition. I know of several that
> >are out there in the field. Therefore, I do not know if we can or should
> >alter it, short of defining a new protocol which this draft does not
> >purport to do.
> >
> >Anyway, I believe s=- is still canonical since '-' is ISO 10646. So there
> >should be no backward compatibility problems.
> >
> >Rajesh
> >
> >At 07:27 PM 2/12/2002 -0500, you wrote:
> >>I changed it to "s= " to match offer/answer, and because I was told that
> >>was what SIP clients expected.
> >>
> >>Colin
> >>
> >>
> >>
> >>--> Rajesh Kumar writes:
> >> >Colin,
> >> >I suggest that acceptance of "s=-" when there is no meaningful 
> session name
> >> >be retained for backwards compatibility.  After all, we're fixing an
> >> >existing standard, and not creating a new one. Backward compatibility
> >> >should be an important consideration. I believe "s=-" will satisfy 
> the ABNF
> >> >definition of the session name, and therefore there is no backward
> >> >compatibility issue, right?
> >> >
> >> >Thanks,
> >> >Rajesh
> >> >
> >> >At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
> >> >>Changes in this version are:
> >> >>- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
> >> >>   similarly for sendonly and inactive.  These only effect the RTP 
> stream.
> >> >>- Fix ABNF to include a unicast-address element, allowing comedia to
> >> >>   align with both this and RFC2327
> >> >>- "s= " rather than "s=-"
> >> >>- Fix ABNF to reflect that s= is mandatory
> >> >>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
> >> >>
> >> >>Any comments would be appreciated.
> >> >>
> >> >>Colin

--=====================_833386246==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:<br>
<blockquote type=cite cite>I don't really care which is chosen, but it'd
be good if we could have a<br>
single definition. RFC 2327 doesn't specify and offer/answer uses
&quot;s= &quot;.<br>
I was following offer/answer, but maybe that conflicts with other
use?</blockquote><font face="Comic Sans MS"><br>
It does conflict, as I indicated. Since offer/answer is a draft, why not
stick to s=- as stated in RFC 2327? What does the change buy for
offer/answer?-Rajesh<br>
<br>
<br>
</font><blockquote type=cite cite>Colin<br>
<br>
<br>
<br>
--&gt; Rajesh Kumar writes:<br>
&gt;Some h.248/MGCP applications use the old definition. I know of
several that <br>
&gt;are out there in the field. Therefore, I do not know if we can or
should <br>
&gt;alter it, short of defining a new protocol which this draft does not
<br>
&gt;purport to do.<br>
&gt;<br>
&gt;Anyway, I believe s=- is still canonical since '-' is ISO 10646. So
there <br>
&gt;should be no backward compatibility problems.<br>
&gt;<br>
&gt;Rajesh<br>
&gt;<br>
&gt;At 07:27 PM 2/12/2002 -0500, you wrote:<br>
&gt;&gt;I changed it to &quot;s= &quot; to match offer/answer, and
because I was told that<br>
&gt;&gt;was what SIP clients expected.<br>
&gt;&gt;<br>
&gt;&gt;Colin<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;--&gt; Rajesh Kumar writes:<br>
&gt;&gt; &gt;Colin,<br>
&gt;&gt; &gt;I suggest that acceptance of &quot;s=-&quot; when there is
no meaningful session name<br>
&gt;&gt; &gt;be retained for backwards compatibility.&nbsp; After all,
we're fixing an<br>
&gt;&gt; &gt;existing standard, and not creating a new one. Backward
compatibility<br>
&gt;&gt; &gt;should be an important consideration. I believe
&quot;s=-&quot; will satisfy the ABNF<br>
&gt;&gt; &gt;definition of the session name, and therefore there is no
backward<br>
&gt;&gt; &gt;compatibility issue, right?<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;Thanks,<br>
&gt;&gt; &gt;Rajesh<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:<br>
&gt;&gt; &gt;&gt;Changes in this version are:<br>
&gt;&gt; &gt;&gt;- Clarify that a=recvonly does NOT mean that you don't
send RTCP, and<br>
&gt;&gt; &gt;&gt;&nbsp;&nbsp; similarly for sendonly and inactive.&nbsp;
These only effect the RTP stream.<br>
&gt;&gt; &gt;&gt;- Fix ABNF to include a unicast-address element,
allowing comedia to<br>
&gt;&gt; &gt;&gt;&nbsp;&nbsp; align with both this and RFC2327<br>
&gt;&gt; &gt;&gt;- &quot;s= &quot; rather than &quot;s=-&quot;<br>
&gt;&gt; &gt;&gt;- Fix ABNF to reflect that s= is mandatory<br>
&gt;&gt; &gt;&gt;- Clarify that s= is non-empty, and the contents SHOULD
be ISO 10646<br>
&gt;&gt; &gt;&gt;<br>
&gt;&gt; &gt;&gt;Any comments would be appreciated.<br>
&gt;&gt; &gt;&gt;<br>
&gt;&gt; &gt;&gt;Colin </blockquote></html>

--=====================_833386246==_.ALT--


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From mmusic-admin@ietf.org  Tue Feb 12 19:59:27 2002
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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt
To: rkumar@cisco.com (Rajesh Kumar)
Date: Wed, 13 Feb 2002 01:58:03 +0100 (MET)
Cc: csp@isi.edu (Colin Perkins), jswamina@cisco.com, smarch@cisco.com,
        joestone@cisco.com (joe Stone), axagrawa@cisco.com,
        fandreas@cisco.com (flemming Andreasen), bfoster@cisco.com,
        mmusic@ietf.org
In-Reply-To: <4.3.2.7.2.20020212164517.0210ca88@mira-sjc5-8.cisco.com> from "Rajesh Kumar" at Feb 12, 2002 04:46:27 PM
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Offer/answer follows the idea of referencing other documents.  Quite a
few implementations seem to use it this way.  And offer/answer will
be referenced by SIP.

MEGACO has created its own "profile" of SDP anyway.  And MEGACO does
not become less SDP compliant than it currently is.  Nor will existing
implementations break.

BTW: "s=-" is not stated in RFC2327.  At least I can't find anything like
that.

Joerg

> 
> --=====================_833386246==_.ALT
> Content-Type: text/plain; charset="us-ascii"; format=flowed
> 
> At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:
> >I don't really care which is chosen, but it'd be good if we could have a
> >single definition. RFC 2327 doesn't specify and offer/answer uses "s= ".
> >I was following offer/answer, but maybe that conflicts with other use?
> 
> It does conflict, as I indicated. Since offer/answer is a draft, why not 
> stick to s=- as stated in RFC 2327? What does the change buy for 
> offer/answer?-Rajesh
> 
> 
> >Colin
> >
> >
> >
> >--> Rajesh Kumar writes:
> > >Some h.248/MGCP applications use the old definition. I know of several that
> > >are out there in the field. Therefore, I do not know if we can or should
> > >alter it, short of defining a new protocol which this draft does not
> > >purport to do.
> > >
> > >Anyway, I believe s=- is still canonical since '-' is ISO 10646. So there
> > >should be no backward compatibility problems.
> > >
> > >Rajesh
> > >
> > >At 07:27 PM 2/12/2002 -0500, you wrote:
> > >>I changed it to "s= " to match offer/answer, and because I was told that
> > >>was what SIP clients expected.
> > >>
> > >>Colin
> > >>
> > >>
> > >>
> > >>--> Rajesh Kumar writes:
> > >> >Colin,
> > >> >I suggest that acceptance of "s=-" when there is no meaningful 
> > session name
> > >> >be retained for backwards compatibility.  After all, we're fixing an
> > >> >existing standard, and not creating a new one. Backward compatibility
> > >> >should be an important consideration. I believe "s=-" will satisfy 
> > the ABNF
> > >> >definition of the session name, and therefore there is no backward
> > >> >compatibility issue, right?
> > >> >
> > >> >Thanks,
> > >> >Rajesh
> > >> >
> > >> >At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
> > >> >>Changes in this version are:
> > >> >>- Clarify that a=recvonly does NOT mean that you don't send RTCP, and
> > >> >>   similarly for sendonly and inactive.  These only effect the RTP 
> > stream.
> > >> >>- Fix ABNF to include a unicast-address element, allowing comedia to
> > >> >>   align with both this and RFC2327
> > >> >>- "s= " rather than "s=-"
> > >> >>- Fix ABNF to reflect that s= is mandatory
> > >> >>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
> > >> >>
> > >> >>Any comments would be appreciated.
> > >> >>
> > >> >>Colin
> 
> --=====================_833386246==_.ALT
> Content-Type: text/html; charset="us-ascii"
> 
> <html>
> At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:<br>
> <blockquote type=cite cite>I don't really care which is chosen, but it'd
> be good if we could have a<br>
> single definition. RFC 2327 doesn't specify and offer/answer uses
> &quot;s= &quot;.<br>
> I was following offer/answer, but maybe that conflicts with other
> use?</blockquote><font face="Comic Sans MS"><br>
> It does conflict, as I indicated. Since offer/answer is a draft, why not
> stick to s=- as stated in RFC 2327? What does the change buy for
> offer/answer?-Rajesh<br>
> <br>
> <br>
> </font><blockquote type=cite cite>Colin<br>
> <br>
> <br>
> <br>
> --&gt; Rajesh Kumar writes:<br>
> &gt;Some h.248/MGCP applications use the old definition. I know of
> several that <br>
> &gt;are out there in the field. Therefore, I do not know if we can or
> should <br>
> &gt;alter it, short of defining a new protocol which this draft does not
> <br>
> &gt;purport to do.<br>
> &gt;<br>
> &gt;Anyway, I believe s=- is still canonical since '-' is ISO 10646. So
> there <br>
> &gt;should be no backward compatibility problems.<br>
> &gt;<br>
> &gt;Rajesh<br>
> &gt;<br>
> &gt;At 07:27 PM 2/12/2002 -0500, you wrote:<br>
> &gt;&gt;I changed it to &quot;s= &quot; to match offer/answer, and
> because I was told that<br>
> &gt;&gt;was what SIP clients expected.<br>
> &gt;&gt;<br>
> &gt;&gt;Colin<br>
> &gt;&gt;<br>
> &gt;&gt;<br>
> &gt;&gt;<br>
> &gt;&gt;--&gt; Rajesh Kumar writes:<br>
> &gt;&gt; &gt;Colin,<br>
> &gt;&gt; &gt;I suggest that acceptance of &quot;s=-&quot; when there is
> no meaningful session name<br>
> &gt;&gt; &gt;be retained for backwards compatibility.&nbsp; After all,
> we're fixing an<br>
> &gt;&gt; &gt;existing standard, and not creating a new one. Backward
> compatibility<br>
> &gt;&gt; &gt;should be an important consideration. I believe
> &quot;s=-&quot; will satisfy the ABNF<br>
> &gt;&gt; &gt;definition of the session name, and therefore there is no
> backward<br>
> &gt;&gt; &gt;compatibility issue, right?<br>
> &gt;&gt; &gt;<br>
> &gt;&gt; &gt;Thanks,<br>
> &gt;&gt; &gt;Rajesh<br>
> &gt;&gt; &gt;<br>
> &gt;&gt; &gt;At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:<br>
> &gt;&gt; &gt;&gt;Changes in this version are:<br>
> &gt;&gt; &gt;&gt;- Clarify that a=recvonly does NOT mean that you don't
> send RTCP, and<br>
> &gt;&gt; &gt;&gt;&nbsp;&nbsp; similarly for sendonly and inactive.&nbsp;
> These only effect the RTP stream.<br>
> &gt;&gt; &gt;&gt;- Fix ABNF to include a unicast-address element,
> allowing comedia to<br>
> &gt;&gt; &gt;&gt;&nbsp;&nbsp; align with both this and RFC2327<br>
> &gt;&gt; &gt;&gt;- &quot;s= &quot; rather than &quot;s=-&quot;<br>
> &gt;&gt; &gt;&gt;- Fix ABNF to reflect that s= is mandatory<br>
> &gt;&gt; &gt;&gt;- Clarify that s= is non-empty, and the contents SHOULD
> be ISO 10646<br>
> &gt;&gt; &gt;&gt;<br>
> &gt;&gt; &gt;&gt;Any comments would be appreciated.<br>
> &gt;&gt; &gt;&gt;<br>
> &gt;&gt; &gt;&gt;Colin </blockquote></html>
> 
> --=====================_833386246==_.ALT--
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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From mmusic-admin@ietf.org  Tue Feb 12 20:07:28 2002
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Date: Tue, 12 Feb 2002 17:04:32 -0800
To: Joerg Ott <jo@Informatik.Uni-Bremen.DE>
From: Rajesh Kumar <rkumar@cisco.com>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt
Cc: mmusic@ietf.org
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I guess I was looking at username which RFC 2327 mentions can be dummied 
out to a "-".

Notwithstanding, I do know several implementations that use a "s=-". The 
question is: Is "-" an illegal session name? That is all I need to know.

Thanks,

Rajesh


At 01:58 AM 2/13/2002 +0100, Joerg Ott wrote:
>Offer/answer follows the idea of referencing other documents.  Quite a
>few implementations seem to use it this way.  And offer/answer will
>be referenced by SIP.
>
>MEGACO has created its own "profile" of SDP anyway.  And MEGACO does
>not become less SDP compliant than it currently is.  Nor will existing
>implementations break.
>
>BTW: "s=-" is not stated in RFC2327.  At least I can't find anything like
>that.
>
>Joerg
>
> >
> > --=====================_833386246==_.ALT
> > Content-Type: text/plain; charset="us-ascii"; format=flowed
> >
> > At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:
> > >I don't really care which is chosen, but it'd be good if we could have a
> > >single definition. RFC 2327 doesn't specify and offer/answer uses "s= ".
> > >I was following offer/answer, but maybe that conflicts with other use?
> >
> > It does conflict, as I indicated. Since offer/answer is a draft, why not
> > stick to s=- as stated in RFC 2327? What does the change buy for
> > offer/answer?-Rajesh
> >
> >
> > >Colin
> > >
> > >
> > >
> > >--> Rajesh Kumar writes:
> > > >Some h.248/MGCP applications use the old definition. I know of 
> several that
> > > >are out there in the field. Therefore, I do not know if we can or should
> > > >alter it, short of defining a new protocol which this draft does not
> > > >purport to do.
> > > >
> > > >Anyway, I believe s=- is still canonical since '-' is ISO 10646. So 
> there
> > > >should be no backward compatibility problems.
> > > >
> > > >Rajesh
> > > >
> > > >At 07:27 PM 2/12/2002 -0500, you wrote:
> > > >>I changed it to "s= " to match offer/answer, and because I was told 
> that
> > > >>was what SIP clients expected.
> > > >>
> > > >>Colin
> > > >>
> > > >>
> > > >>
> > > >>--> Rajesh Kumar writes:
> > > >> >Colin,
> > > >> >I suggest that acceptance of "s=-" when there is no meaningful
> > > session name
> > > >> >be retained for backwards compatibility.  After all, we're fixing an
> > > >> >existing standard, and not creating a new one. Backward compatibility
> > > >> >should be an important consideration. I believe "s=-" will satisfy
> > > the ABNF
> > > >> >definition of the session name, and therefore there is no backward
> > > >> >compatibility issue, right?
> > > >> >
> > > >> >Thanks,
> > > >> >Rajesh
> > > >> >
> > > >> >At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:
> > > >> >>Changes in this version are:
> > > >> >>- Clarify that a=recvonly does NOT mean that you don't send 
> RTCP, and
> > > >> >>   similarly for sendonly and inactive.  These only effect the RTP
> > > stream.
> > > >> >>- Fix ABNF to include a unicast-address element, allowing comedia to
> > > >> >>   align with both this and RFC2327
> > > >> >>- "s= " rather than "s=-"
> > > >> >>- Fix ABNF to reflect that s= is mandatory
> > > >> >>- Clarify that s= is non-empty, and the contents SHOULD be ISO 10646
> > > >> >>
> > > >> >>Any comments would be appreciated.
> > > >> >>
> > > >> >>Colin
> >
> > --=====================_833386246==_.ALT
> > Content-Type: text/html; charset="us-ascii"
> >
> > <html>
> > At 07:40 PM 2/12/2002 -0500, Colin Perkins wrote:<br>
> > <blockquote type=cite cite>I don't really care which is chosen, but it'd
> > be good if we could have a<br>
> > single definition. RFC 2327 doesn't specify and offer/answer uses
> > &quot;s= &quot;.<br>
> > I was following offer/answer, but maybe that conflicts with other
> > use?</blockquote><font face="Comic Sans MS"><br>
> > It does conflict, as I indicated. Since offer/answer is a draft, why not
> > stick to s=- as stated in RFC 2327? What does the change buy for
> > offer/answer?-Rajesh<br>
> > <br>
> > <br>
> > </font><blockquote type=cite cite>Colin<br>
> > <br>
> > <br>
> > <br>
> > --&gt; Rajesh Kumar writes:<br>
> > &gt;Some h.248/MGCP applications use the old definition. I know of
> > several that <br>
> > &gt;are out there in the field. Therefore, I do not know if we can or
> > should <br>
> > &gt;alter it, short of defining a new protocol which this draft does not
> > <br>
> > &gt;purport to do.<br>
> > &gt;<br>
> > &gt;Anyway, I believe s=- is still canonical since '-' is ISO 10646. So
> > there <br>
> > &gt;should be no backward compatibility problems.<br>
> > &gt;<br>
> > &gt;Rajesh<br>
> > &gt;<br>
> > &gt;At 07:27 PM 2/12/2002 -0500, you wrote:<br>
> > &gt;&gt;I changed it to &quot;s= &quot; to match offer/answer, and
> > because I was told that<br>
> > &gt;&gt;was what SIP clients expected.<br>
> > &gt;&gt;<br>
> > &gt;&gt;Colin<br>
> > &gt;&gt;<br>
> > &gt;&gt;<br>
> > &gt;&gt;<br>
> > &gt;&gt;--&gt; Rajesh Kumar writes:<br>
> > &gt;&gt; &gt;Colin,<br>
> > &gt;&gt; &gt;I suggest that acceptance of &quot;s=-&quot; when there is
> > no meaningful session name<br>
> > &gt;&gt; &gt;be retained for backwards compatibility.&nbsp; After all,
> > we're fixing an<br>
> > &gt;&gt; &gt;existing standard, and not creating a new one. Backward
> > compatibility<br>
> > &gt;&gt; &gt;should be an important consideration. I believe
> > &quot;s=-&quot; will satisfy the ABNF<br>
> > &gt;&gt; &gt;definition of the session name, and therefore there is no
> > backward<br>
> > &gt;&gt; &gt;compatibility issue, right?<br>
> > &gt;&gt; &gt;<br>
> > &gt;&gt; &gt;Thanks,<br>
> > &gt;&gt; &gt;Rajesh<br>
> > &gt;&gt; &gt;<br>
> > &gt;&gt; &gt;At 11:00 AM 2/12/2002 -0500, Colin Perkins wrote:<br>
> > &gt;&gt; &gt;&gt;Changes in this version are:<br>
> > &gt;&gt; &gt;&gt;- Clarify that a=recvonly does NOT mean that you don't
> > send RTCP, and<br>
> > &gt;&gt; &gt;&gt;&nbsp;&nbsp; similarly for sendonly and inactive.&nbsp;
> > These only effect the RTP stream.<br>
> > &gt;&gt; &gt;&gt;- Fix ABNF to include a unicast-address element,
> > allowing comedia to<br>
> > &gt;&gt; &gt;&gt;&nbsp;&nbsp; align with both this and RFC2327<br>
> > &gt;&gt; &gt;&gt;- &quot;s= &quot; rather than &quot;s=-&quot;<br>
> > &gt;&gt; &gt;&gt;- Fix ABNF to reflect that s= is mandatory<br>
> > &gt;&gt; &gt;&gt;- Clarify that s= is non-empty, and the contents SHOULD
> > be ISO 10646<br>
> > &gt;&gt; &gt;&gt;<br>
> > &gt;&gt; &gt;&gt;Any comments would be appreciated.<br>
> > &gt;&gt; &gt;&gt;<br>
> > &gt;&gt; &gt;&gt;Colin </blockquote></html>
> >
> > --=====================_833386246==_.ALT--
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic


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From mmusic-admin@ietf.org  Tue Feb 12 21:19:49 2002
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To: Rajesh Kumar <rkumar@cisco.com>
cc: Joerg Ott <jo@Informatik.Uni-Bremen.DE>, mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt 
In-Reply-To: Your message of "Tue, 12 Feb 2002 17:04:32 PST."
             <4.3.2.7.2.20020212170241.0210bc00@mira-sjc5-8.cisco.com> 
Date: Tue, 12 Feb 2002 21:19:07 -0500
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--> Rajesh Kumar writes:
>I guess I was looking at username which RFC 2327 mentions can be dummied 
>out to a "-".
>
>Notwithstanding, I do know several implementations that use a "s=-". The 
>question is: Is "-" an illegal session name? That is all I need to know.

No, it's not illegal. The "s= " is just a recommendation if there's nothing
better to use; you can use whatever you like for a session name.

Colin

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From mmusic-admin@ietf.org  Tue Feb 12 22:49:26 2002
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Date: Tue, 12 Feb 2002 22:48:12 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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CC: Rajesh Kumar <rkumar@cisco.com>, Joerg Ott <jo@informatik.uni-bremen.de>,
        mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-05.txt
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I don't understand the problem. A valid SDP implementation should accept
anything. Whether its " " or "-" is irrelevant. As Colin says, these are
just suggestions in any case.

-Jonathan R.

Colin Perkins wrote:
> 
> --> Rajesh Kumar writes:
> >I guess I was looking at username which RFC 2327 mentions can be
> dummied
> >out to a "-".
> >
> >Notwithstanding, I do know several implementations that use a "s=-".
> The
> >question is: Is "-" an illegal session name? That is all I need to
> know.
> 
> No, it's not illegal. The "s= " is just a recommendation if there's
> nothing
> better to use; you can use whatever you like for a session name.
> 
> Colin
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Tue Feb 12 23:07:18 2002
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Subject: [MMUSIC] Using payload types not specified in answer in offer/answer
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It is unclear from the specification whether answerer is allowed to send media with payload types not specified in the response to sendonly or sendrecv offer. It clearly states that the offerer should be prepared to receive those payload types before the answer arrives, but is it safe to assume that those payload types will not be used if they are not specified in the answer once the answer was received?___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
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From mmusic-admin@ietf.org  Wed Feb 13 11:04:38 2002
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Roman Shpount wrote:
> 
> It is unclear from the specification whether answerer is allowed to send
> media with payload types not specified in the response to sendonly or
> sendrecv offer. 

This doesn't make sense to me. If the offer is sendonly, the answer is
recvonly, so the answerer cannot send media. 

In the case of sendrecv, it should only be sending with codecs in its
answer. This is in the document:

It
{\MUST} be prepared to send and receive media for any sendrecv streams
in the answer, and {\MAY} send media immediately. 

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb 13 12:27:37 2002
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My mistake. What I wanted to say was if the offer was recvonly or sendrecv. For example if the offer is:

   v=0
   o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
   s=
   c=IN IP4 host.anywhere.com
   t=0 0
   m=audio 49170 RTP/AVP 0 110
   a=rtpmap:0 PCMU/8000 
   a=rtpmap:110 telephone-events 
   a=recvonly

And if the answer is

   v=0
   o=bob 2890844730 2890844730 IN IP4 host.example.com
   s=
   c=IN IP4 host.example.com
   t=0 0
   m=audio 47920 RTP/AVP 0
   a=rtpmap:0 PCMU/8000 
   a=sendonly

Is the answerer allowed to send AVT tones with a RTP payload type 110?

___________________________________
Roman Shpount

------ Original message ------
>From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
>Subject: Re: Using payload types not specified in answer in offer/answer
>Date: 13 Feb 02, 04:03 PM
>To: Roman Shpount <roman@atelo.com>
>Cc: mmusic@ietf.org

Roman Shpount wrote:
> 
> It is unclear from the specification whether answerer is allowed to send
> media with payload types not specified in the response to sendonly or
> sendrecv offer. 

This doesn't make sense to me. If the offer is sendonly, the answer is
recvonly, so the answerer cannot send media. 

In the case of sendrecv, it should only be sending with codecs in its
answer. This is in the document:

It
{\MUST} be prepared to send and receive media for any sendrecv streams
in the answer, and {\MAY} send media immediately. 

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com






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From mmusic-admin@ietf.org  Wed Feb 13 13:28:13 2002
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Roman Shpount wrote:
> 
> My mistake. What I wanted to say was if the offer was recvonly or
> sendrecv. For example if the offer is:
> 
>    v=0
>    o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
>    s=
>    c=IN IP4 host.anywhere.com
>    t=0 0
>    m=audio 49170 RTP/AVP 0 110
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:110 telephone-events
>    a=recvonly
> 
> And if the answer is
> 
>    v=0
>    o=bob 2890844730 2890844730 IN IP4 host.example.com
>    s=
>    c=IN IP4 host.example.com
>    t=0 0
>    m=audio 47920 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    a=sendonly
> 
> Is the answerer allowed to send AVT tones with a RTP payload type 110?

No. 

But, if it wants to, why would it delete it from the anwer? Just put it
in.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb 13 13:56:19 2002
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Roman Shpount wrote:
> 
> My mistake. What I wanted to say was if the offer was recvonly or
> sendrecv. For example if the offer is:
> 
>    v=0
>    o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
>    s=
>    c=IN IP4 host.anywhere.com
>    t=0 0
>    m=audio 49170 RTP/AVP 0 110
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:110 telephone-events
>    a=recvonly
> 
> And if the answer is
> 
>    v=0
>    o=bob 2890844730 2890844730 IN IP4 host.example.com
>    s=
>    c=IN IP4 host.example.com
>    t=0 0
>    m=audio 47920 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    a=sendonly
> 
> Is the answerer allowed to send AVT tones with a RTP payload type 110?

Jonathan D. Rosenberg wrote:

>No. 
>
>But, if it wants to, why would it delete it from the anwer? Just put it
>in.

Is it stated somewhere in the offer/answer? 

Again, it probably makes little sence for recvonly, but it is quite common for sendrecv as well. So if we modify above example for the sendrecv scenario.
 The offer would be:

    v=0
    o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
    s=
    c=IN IP4 host.anywhere.com
    t=0 0
    m=audio 49170 RTP/AVP 0 110
    a=rtpmap:0 PCMU/8000
    a=rtpmap:110 telephone-events
 
 And if the answer would be:
 
    v=0
    o=bob 2890844730 2890844730 IN IP4 host.example.com
    s=
    c=IN IP4 host.example.com
    t=0 0
    m=audio 47920 RTP/AVP 0
    a=rtpmap:0 PCMU/8000

Would any party on the call be allowed to use avt tones? If it does or does not, where is it specified?
___________________________________
Roman Shpount





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From mmusic-admin@ietf.org  Wed Feb 13 14:42:03 2002
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Subject: Re: [MMUSIC] Re: Using payload types not specified in answer in 
 offer/answer
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Jonathan Rosenberg wrote:

> Roman Shpount wrote:
> >
> > My mistake. What I wanted to say was if the offer was recvonly or
> > sendrecv. For example if the offer is:
> >
> >    v=0
> >    o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
> >    s=
> >    c=IN IP4 host.anywhere.com
> >    t=0 0
> >    m=audio 49170 RTP/AVP 0 110
> >    a=rtpmap:0 PCMU/8000
> >    a=rtpmap:110 telephone-events
> >    a=recvonly
> >
> > And if the answer is
> >
> >    v=0
> >    o=bob 2890844730 2890844730 IN IP4 host.example.com
> >    s=
> >    c=IN IP4 host.example.com
> >    t=0 0
> >    m=audio 47920 RTP/AVP 0
> >    a=rtpmap:0 PCMU/8000
> >    a=sendonly
> >
> > Is the answerer allowed to send AVT tones with a RTP payload type 110?
>
> No.
>

No ? I believe the answer is yes. Haven't we always said, that the SDP
indicates the set of media formats I can receive (assuming proper direction
attributes), and I want to change that, I need to send a new SDP ?

Where in offer/answer is this restriction described ? Also, if it you still
mean "no", then I don't understand why you want to allow the answer to
contain media formats not specified in the offer ? If asymmetry is allowed
in one direction, it should be allowed in both.


>
> But, if it wants to, why would it delete it from the anwer?

Same reason as the answerer might want to add a media format not present in
the offer:    asymmetry (e.g. I can send RFC 2833 to you, but I can't
receive it).

-- Flemming


--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Wed Feb 13 15:15:53 2002
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Flemming Andreasen wrote:
> 
> > > Is the answerer allowed to send AVT tones with a RTP payload type
> 110?
> >
> > No.
> >
> 
> No ? I believe the answer is yes. Haven't we always said, that the SDP
> indicates the set of media formats I can receive (assuming proper
> direction
> attributes), and I want to change that, I need to send a new SDP ?

It indicates what you can receive for recvonly, indicates what you send
for sendonly, and indicates what you will both send and receive for
sendrecv. I thought you had argued strongly for that view at IETF 52...

> 
> Where in offer/answer is this restriction described ? Also, if it you
> still
> mean "no", then I don't understand why you want to allow the answer to
> contain media formats not specified in the offer ? If asymmetry is
> allowed
> in one direction, it should be allowed in both.

In the case of sendrecv, subsetting is the only thing that makes sense.
If A indicates in an offer that it can send and receive with codecs
1,2,3, and B indicates in its answer that it can send and receive with
2,3,4, the presence of 4 is useless. B can't send with it, since it
doesn't know whether A can receive it. 

Now, consider the case once more of sendrecv, where A sends 1,2,3 in the
offer, and B sends 1,2 in the answer. Can B send with codec 3? No, it
can't. It can't because A can't send with it, and thus the codec would
effectively be unidirectional. This is a birdirectional stream. So, in
the case of sendrecv, B needs to be prepared to send and receive with
the codecs in the answer, and this is what the text says:

--
Once the answerer has sent the answer, it {\MUST} be prepared to
receive media for any recvonly streams described by that answer. It
{\MUST} be prepared to send and receive media for any sendrecv streams
in the answer, and {\MAY} send media immediately. 
--

Now, in the case of a recvonly offer, A could indicate the ability to
receive 1,2, and 3. In the answer, lets say B returns 1,2 as sendonly.
But, then it sends using 3 anyway. If the answerer does not go with
whats in the answer, the answer provides zero information. The point of
the answer in this case is to help the offerer restrict the set of
codecs it needs to be prepared to receive. I do admit that I find no
text at the moment which says that the answerer in a sendonly answer
should send only with the codecs in the answer. However, that has been
my interprtation.

Let me ask the converse - what is gained by having the answerer be able
to send with things not in the answer? Why not just include them in the
answer?


> 
> >
> > But, if it wants to, why would it delete it from the anwer?
> 
> Same reason as the answerer might want to add a media format not present
> in
> the offer:    asymmetry (e.g. I can send RFC 2833 to you, but I can't
> receive it).

Remember, we only support asymmetric codecs in asymmetric stream types.
In the case you describe, you are talking about a recvonly offer and a
sendonly answer. The spec says that the offer lists the codecs the
offerer can receive. If its not there, what is gained by adding it in
the answer? The answerer can't send it anyway, since it doesn't know if
the offerer supports it?

We did previously allow addition of recvonly codecs by the answerer in a
sendrecv stream, but agreed at IETF 52 to remove this hack in favor of
FID, which is the proper way to support such things. I recall you argued
in favor of that, even.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
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Jonathan Rosenberg wrote:

> Flemming Andreasen wrote:
> >
> > > > Is the answerer allowed to send AVT tones with a RTP payload type
> > 110?
> > >
> > > No.
> > >
> >
> > No ? I believe the answer is yes. Haven't we always said, that the SDP
> > indicates the set of media formats I can receive (assuming proper
> > direction
> > attributes), and I want to change that, I need to send a new SDP ?
>
> It indicates what you can receive for recvonly, indicates what you send
> for sendonly, and indicates what you will both send and receive for
> sendrecv. I thought you had argued strongly for that view at IETF 52...
>

Yes - that's what I meant.


>
> >
> > Where in offer/answer is this restriction described ? Also, if it you
> > still
> > mean "no", then I don't understand why you want to allow the answer to
> > contain media formats not specified in the offer ? If asymmetry is
> > allowed
> > in one direction, it should be allowed in both.
>
> In the case of sendrecv, subsetting is the only thing that makes sense.
> If A indicates in an offer that it can send and receive with codecs
> 1,2,3, and B indicates in its answer that it can send and receive with
> 2,3,4, the presence of 4 is useless. B can't send with it, since it
> doesn't know whether A can receive it.

Correct, but A can send with it.

>
>
> Now, consider the case once more of sendrecv, where A sends 1,2,3 in the
> offer, and B sends 1,2 in the answer. Can B send with codec 3? No, it
> can't. It can't because A can't send with it, and thus the codec would
> effectively be unidirectional.

I don't buy that argument since you could equally well have A sending with 1
and B sending with 2.


> This is a birdirectional stream. So, in
> the case of sendrecv, B needs to be prepared to send and receive with
> the codecs in the answer, and this is what the text says:
>
> --
> Once the answerer has sent the answer, it {\MUST} be prepared to
> receive media for any recvonly streams described by that answer. It
> {\MUST} be prepared to send and receive media for any sendrecv streams
> in the answer, and {\MAY} send media immediately.
> --
>

If the above was the intended implication, then it's not entirely clear.


>
> Now, in the case of a recvonly offer, A could indicate the ability to
> receive 1,2, and 3. In the answer, lets say B returns 1,2 as sendonly.
> But, then it sends using 3 anyway. If the answerer does not go with
> whats in the answer, the answer provides zero information. The point of
> the answer in this case is to help the offerer restrict the set of
> codecs it needs to be prepared to receive.

I thought it was so offerer can figure out what he can actually send with
(ignoring some of the unidirectional stream cases for a momemt).


> I do admit that I find no
> text at the moment which says that the answerer in a sendonly answer
> should send only with the codecs in the answer. However, that has been
> my interprtation.

I have always understood, that once I hand out an SDP indicating the ability
to receive codec 1,2, and 3, then I have to be able to receive those codecs
until I send out another SDP. Of course you can say, that this is because SDP
never defined the negotiation that offer/answer does, but this seems to
introduce an unnecesary deviation from "normal" SDP usage.


>
>
> Let me ask the converse - what is gained by having the answerer be able
> to send with things not in the answer? Why not just include them in the
> answer?

It could for example be that I can send RFC 2833 but not receive it.
Admittedly, SDP is not really good at expressing such unidirectional
constraints, and we probably shouldn't get hung up on it (however I thought
this was one of your concerns). My real concern here is that it seems to
deviate from current SDP behavior.


>
>
> >
> > >
> > > But, if it wants to, why would it delete it from the anwer?
> >
> > Same reason as the answerer might want to add a media format not present
> > in
> > the offer:    asymmetry (e.g. I can send RFC 2833 to you, but I can't
> > receive it).
>
> Remember, we only support asymmetric codecs in asymmetric stream types.

We do ? If I offer 2 or more codecs and you answer with two or more codecs,
then we can have asymmetric codec usage.


>
> In the case you describe, you are talking about a recvonly offer and a
> sendonly answer. The spec says that the offer lists the codecs the
> offerer can receive. If its not there, what is gained by adding it in
> the answer? The answerer can't send it anyway, since it doesn't know if
> the offerer supports it?
>

However, I thought it would apply to bidirectional streams as well. For
example, if RFC 2833 is not offered , but the answerer is able to accept it,
he may want to indicate so in the answer (and vice versa).


>
> We did previously allow addition of recvonly codecs by the answerer in a
> sendrecv stream, but agreed at IETF 52 to remove this hack in favor of
> FID, which is the proper way to support such things. I recall you argued
> in favor of that, even.

I still do, however that doesn't mean you can't add a sendrecv codec to a
sendrecv stream, even though you may only be able to send or receive with it.

-- Flemming

--
Flemming Andreasen
Cisco Systems



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Jonathan D. Rosenberg wrote:

> Let me ask the converse - what is gained by having the answerer be able
> to send with things not in the answer? Why not just include them in the
> answer?

This is primarily a question of interpretation. I know at least one gateway implementation which is capable of sending, but not receiving avt tones. So this gateway replies to a sendrecv offer, which contains avt tones and G.711, with an answer that contains only G.711, but sends out avt tones anyway. I am looking for the language that states whether this behavior is legal or not.
___________________________________
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Flemming,

Thanks for your comments. Responses inline.

Flemming Andreasen wrote:
> 
> Some comments below:
> 
> - p.2 says "The offer/answer model defined here is the mandatory
> baseline mechanism used by the Session
> Initiation Protocol (SIP) [3]". This requirement seems to belong in SIP,
> not offer/answer.

Its not a normative statement; it merely provides a context for the
reader of this document. I think its valuable to note that this
mechanism is what SIP uses.

> 
> - p. 3:    Definition of Answerer says "An agent capable of sending
> and/or receiving media...". The ability to
> send/receive media should not be included here, i.e. strike "capable of
> sending and/or receiving media,"

I got a similar comment in private email. I removed the text.

> 
> - p. 3:    Definition of Offerer: strike "capable of sending and/or
> receiving media,"

Done.

> 
> - p. 4:    "The agent receiving the offer MAY generate an answer, or it
> MAY reject the offer.". At least in the unicast case, MUST you not
> always generate an answer which may either accept or reject the offer ?

No. You can reject the offer outright. In the case of SIP, that would
mean sending
a 488. There is no answer in the 488.

> 
> - p. 4:    "However, it MUST NOT generate a new offer if it has received
> an offer which it has not yet answered or reject.". I'm not sure what
> this buys you since each side may generate an offer simultanously and
> hence you have to deal with the race anyway. (nits: "reject =>
> rejected", and offers are either accepted or rejected, but always
> answered)

The nit has been fixed already. 

I have also already added text saying that the higher layer protocol
needs to provide a away to resolve the glare condition of both sending
offers simultaneously. It is always best to prevent those kinds of glare
conditions from happening, since they do take time to resolve (can be
seconds, at least, for the sip glare resolution mechanisms). Thus, the
text about not sending an offer when you have one outstanding is a good
one, I think. 

> 
> p.4:    "The offer MAY contain zero or more media streams..." A fairly
> empty statement (how do you not satisfy this ?).

Good point. The sentence is needed to preface the following sentence,
which define a meaning for zero media streams. So, I've reworded the
first sentence to be non-normative. The first two sentences now read:

The offer will contain zero or more media streams (each media stream
is described by an m line and its associated attributes). Zero media
streams implies that the offerer wishes to communicate, but that the
streams for the session will be added at a later time through a
modified offer. 

> 
> Section 5.1, first paragraph:    You may want to elaborate on sendonly,
> recvonly, etc as a session attribute versus as a media attribute for
> specific media streams.

Excellent point. The paragraph now reads:

If the offerer wishes to only send media on a stream to its peer, it
{\MUST} mark the stream as sendonly with the a=sendonly attribute. We
refer to a stream as being marked with a certain direction if a
direction attribute was present as either a media stream attribute or
a session attribute. If the offerer wishes to only receive media from
its peer, it {\MUST} mark the stream as recvonly. 
<snip>


> 
> p. 6:    "However, for sendonly and sendrecv streams, the answer might
> indicate different payload type numbers for the same codecs, in which
> case, the offerer MUST send with the payload type numbers from the
> answer." Should add a sentence that the answerer MUST still send using
> the payload types from the offer.

Yes, thats a good point. That belongs farther down, in the section on
answerer behavior. The bottom of that paragraph now reads:

<snip>
It {\SHOULD} send media using a banwdith no
higher than the value of the bandwidth attribute in the offer, if any
was
present. The answerer {\SHOULD} send using the most preferred codec in
the offer supported by the answerer. In the case of RTP, it {\MUST}
use the payload type numbers from the offer, even if they differ from
those in the answer.


> 
> p. 8, last paragraph:    Only recvonly and sendrecv streams are
> discussed. For completeness, sendonly streams should be mentioned as
> well. Also, consider moving this paragraph to right after the first
> paragraph on p.6, where similar issues are discussed.

Actually its page 7. I added a sentence on sendonly. 

I like the paragraph where it is, since it descsribes the processing
that the offerer follows after sending the offer, but before the answer.
The first paragraph on p. 6 describes the general meaning of the
direction attributes and m lines.

> 
> Section 5.2, first paragraph:    I'd suggest writing this from the
> perspective of the offerer (like the rest of section 5), i.e. "answerer
> can only send (receive)" => "offerer can only receive (send)".

Actually, I've been trying to get my arms around exactly what the
meaning is for multicast. The text thats there now, or your text, are
both wrong I think; they take a unicast view of the SDP. Really, for
multicast, it should be like the SAP usage. If a stream is recvonly,
then all participants, including the offerer and answerer, will only
receive on it. This is already reflected in how the answer is generated
(the directionalities are not flipped). So, I have reworded this section
to read:

If a session description contains a multicast media stream which is
listed as receive (send) only, it means that the participants,
including the offerer and answerer, can only receive (send) on that
stream. This differs from the unicast view, where the directionality
refers to the flow of media between offerer and answerer.




> 
> p.9:    Should state explicitly whether sendonly and sendrecv streams in
> an answer are allowed to contain any additional media formats or not.

We're discussing that one on the list... I've logged it in bugzilla as
an open issue.


> 
> p. 10:    "In that case, the answer MAY contain fmtp parameters, and
> those MAY have the same values as those in the offer, or they MAY be
> different." I agree with this and understand the difficulty in saying
> something more specific, however it still leaves open the question of
> how to negotiate fmtp parameters between an offer and an answer. Should
> there be a sentence saying that people defining fmtp parameters should
> define how they are negotiated in offer/answer ?

I was debating that. I didn't put it in because I thought it was too
burdensome for drafts that define them, but its probably a good idea.
I'll add it. Speak now if you have a problem with that.... Here is the
sentence I tacked on here:

SDP extensions
that define new parameters {\SHOULD} specify the proper interpretation
in offer/answer. 

we can debate whether this is SHOULD or should; a similar debate is
going on in bis about whether such statements even belong in bis.
However, I think normative makes sense, and unlike bis, where we have
another document which defines rules for extensions
(draft-ietf-sip-guidelines), there isn't anywhere else to put it.

> 
> p. 10:    "The answerer MAY include a ptime" => "The answerer MAY
> include a non-zero ptime"

I didn't include that restriction for the offer, and you had no comment
on it there. Do I interpret this to mean you want this restriction only
for the answer, or do you really want it for both? It had never even
occurred to me to use a zero ptime, I don't even know what that would
mean....


> 
> p. 10:    "When sending media, it SHOULD use a packetization interval
> equal to the value of the ptime attribute in the offer, if any was
> present.". Since ptime is a media stream attribute, this can be
> difficult to do if multiple media formats are specified. There should be
> a rounding rule to account for that (e.g. use closest non-zero ptime if
> you can't use the ptime specified).

This is really a general SDP issue, not specific to O/A. Do we want to
discuss it here? Probably sdp-new should clarify this.

> 
> p. 11:    "Similarly, the directionality information in the answer
> (sendonly, recvonly, or sendrecv) MUST equal that of the offer." It may
> be worth spelling it out more clearly what this means, since it differs
> from the unicast.

I mention it in the text on constructing the offer, see above response.

> 
> p. 13:    "Otherwise, the media description for the removed stream
> SHOULD be formatted identically to the corresponding stream in the
> previous SDP."    Add ", except for the list of media formats."

I rewrote this altogether, since the stream really isn't formatted at
all identically to the corresponding stream in the previous SDP. It now
reads:

A stream that is offered with a port of zero {\MUST} be marked with
port zero in the answer. Like the offer, the answer {\MAY} omit all
attributes present previously, and {\MAY} list just a single media
format from amongst those in the offer. 



> 
> p. 13:    "Removal of a media stream implies that media is no longer
> sent for that stream.". Add sentence "Furthermore, when RTP was used for
> the media stream, RTCP reports will no longer be sent or received."

Added:

In the case of RTP, RTCP transmission also ceases.


> 
> Section 8.3.1, first paragraph:    "The offerer MUST NOT cease listening
> for media ..". In practice, it will not always be possible to satisfy
> this requirement, and hence it should rather be "SHOULD NOT". (Nit:
> "listening for" should be "receiving" - twice in the paragraph).

On MUST NOT -> SHOULD NOT; OK, I made that change. I also added a note
that said if you don't, there could be media loss during the transition.
I don;'t want to encourage people to break the SHOULD. 

Regarding listening for -> receiving, I disagree. Listening is not the
same as receiving; you can be listening, but receive nothing. Listening
is the ability to receive. In this case, it truly is the ability to
receive, not the actual act of receiving which is mandated.

> 
> Section 8.3.1, second paragraph:    "The answerer MUST NOT cease
> listening for media on the old port until media arrives on the new
> port.". Same comments (also "listening for" => "receiving").

Same as above.

> 
> Section 8.3.2, first paragraph:    "the mapping from a particular
> dynamic payload type number to a particular codec MUST NOT change".
> Needs to be clarified that it applies on a per media stream basis and
> not across the entire session.

OK. I clarified.

> 
> Section 8.4:    Section seems to be unicast specific. Both heading and
> section text should reflect this.

I changed the section heading to "putting a unicast stream on hold". I
see no need to repeat that within the body text.

> 
> Section 9:    Since the "capabilities" SDP is indistinguishable from an
> "offer/answer" SDP, we should make it clear, that this is determined by
> the context within which the SDP is provided (for example a SIP OPTIONS
> as opposed to INVITE).

Good point. Added:

Since SDP has no way to indicate that the message is for
the purpose of capability indication, this is determined from the
context of the higher layer protocol.


> 
> Section 9:    "The transport component of the "m=" line indicates the
> preferred transport for that media type.". Not sure this is ideal. For
> example, if I support both RTP and SRTP, do I only indicate one of them
> (similar issue with  T.38 UDP or TCP) ?

Yes. Its not really preferred, its "the one". If you want to indicate
support for both SRTP and RTP, you would pick one in the offer, include
simcaps indicating the other. Similar in the answer. THe offerer could
then send an updated offer switching to SRTP. This is discussed in the
unify draft.

> 
> Also, I have some strictly editorial nits which I'll send off-line.

OK, thanks. I appreciate your comments.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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inline. Let us resolve this quickly, only a few days remain in wglc:

Flemming Andreasen wrote:
> 

> > In the case of sendrecv, subsetting is the only thing that makes
> sense.
> > If A indicates in an offer that it can send and receive with codecs
> > 1,2,3, and B indicates in its answer that it can send and receive with
> > 2,3,4, the presence of 4 is useless. B can't send with it, since it
> > doesn't know whether A can receive it.
> 
> Correct, but A can send with it.

So, if A can send with it, but B can't, how is this a sendrecv codec,
practically speaking? 

> > Now, consider the case once more of sendrecv, where A sends 1,2,3 in
> the
> > offer, and B sends 1,2 in the answer. Can B send with codec 3? No, it
> > can't. It can't because A can't send with it, and thus the codec would
> > effectively be unidirectional.
> 
> I don't buy that argument since you could equally well have A sending
> with 1
> and B sending with 2.

True, although that is not the recommended behavior. 

> > This is a birdirectional stream. So, in
> > the case of sendrecv, B needs to be prepared to send and receive with
> > the codecs in the answer, and this is what the text says:
> >
> > --
> > Once the answerer has sent the answer, it {\MUST} be prepared to
> > receive media for any recvonly streams described by that answer. It
> > {\MUST} be prepared to send and receive media for any sendrecv streams
> > in the answer, and {\MAY} send media immediately.
> > --
> >
> 
> If the above was the intended implication, then it's not entirely clear.

Apparently not....

> >
> > Now, in the case of a recvonly offer, A could indicate the ability to
> > receive 1,2, and 3. In the answer, lets say B returns 1,2 as sendonly.
> > But, then it sends using 3 anyway. If the answerer does not go with
> > whats in the answer, the answer provides zero information. The point
> of
> > the answer in this case is to help the offerer restrict the set of
> > codecs it needs to be prepared to receive.
> 
> I thought it was so offerer can figure out what he can actually send
> with
> (ignoring some of the unidirectional stream cases for a momemt).

You've lost me; we are talking explicitly about the unidirectional cases
here.

> 
> > I do admit that I find no
> > text at the moment which says that the answerer in a sendonly answer
> > should send only with the codecs in the answer. However, that has been
> > my interprtation.
> 
> I have always understood, that once I hand out an SDP indicating the
> ability
> to receive codec 1,2, and 3, then I have to be able to receive those
> codecs
> until I send out another SDP. 

It is certainly true that you need to be able to receive them once you
send them, to handle media clipping. It had been my implicit assumption,
arguably not well thought thruough, that the answer would restrict what
the element needs to be prepared to receive. I suppose that, since it
needs to be able to receive them initially anyway, is there much gained
by learning that the set of things to receive is less than it needs to
initially?

One example is if A can support codecs 1,2,3,4,5, and it sends an offer
to B which really only supports 1, and thats what is in the answer. Now,
A would still need to be preapred to reeive 1-5 based on your proposal,
even though it effectively knows that B is never going to use anythign
but 1. I had assumed that the point of the offer/answer exchange was to
do a one-pass subsetting of the codecs in use. I am sure we have
discussed this in the past in the sip wg.


> Of course you can say, that this is
> because SDP
> never defined the negotiation that offer/answer does, but this seems to
> introduce an unnecesary deviation from "normal" SDP usage.

I can see that it is a bit odd.

I suppose that, if the offerer really wants to know a more restricted
set, it can always send an updated offer with fewer codecs, ala the
1-of-N mechanism in the examples in the draft. But, isn't it better to
avoid that if you know that the answer defines a subset that is
acceptable?


> 
> >
> >
> > Let me ask the converse - what is gained by having the answerer be
> able
> > to send with things not in the answer? Why not just include them in
> the
> > answer?
> 
> It could for example be that I can send RFC 2833 but not receive it.
> Admittedly, SDP is not really good at expressing such unidirectional
> constraints, and we probably shouldn't get hung up on it (however I
> thought
> this was one of your concerns). My real concern here is that it seems to
> deviate from current SDP behavior.

I guess I am confused on this, because we had text along those lines
before, and we agreed at ietf 52 to remove it. Specifically, we agreed
that, in a sendrecv answer, you cannot add codecs which are recvonly,
they have to be sendrecv. From the above, it sounds like you are backing
down from that.

> > > Same reason as the answerer might want to add a media format not
> present
> > > in
> > > the offer:    asymmetry (e.g. I can send RFC 2833 to you, but I
> can't
> > > receive it).
> >
> > Remember, we only support asymmetric codecs in asymmetric stream
> types.
> 
> We do ? If I offer 2 or more codecs and you answer with two or more
> codecs,
> then we can have asymmetric codec usage.

See above comment; I thought that we agreed that a sendrecv stream had
to list sendrecv codecs. 

> 
> >
> > In the case you describe, you are talking about a recvonly offer and a
> > sendonly answer. The spec says that the offer lists the codecs the
> > offerer can receive. If its not there, what is gained by adding it in
> > the answer? The answerer can't send it anyway, since it doesn't know
> if
> > the offerer supports it?
> >
> 
> However, I thought it would apply to bidirectional streams as well. For
> example, if RFC 2833 is not offered , but the answerer is able to accept
> it,
> he may want to indicate so in the answer (and vice versa).

As above, I removed text that explicitly allowed this based on iETF 52
consensus, where we agreed fid was better for it.

> 
> >
> > We did previously allow addition of recvonly codecs by the answerer in
> a
> > sendrecv stream, but agreed at IETF 52 to remove this hack in favor of
> > FID, which is the proper way to support such things. I recall you
> argued
> > in favor of that, even.
> 
> I still do, however that doesn't mean you can't add a sendrecv codec to
> a
> sendrecv stream, even though you may only be able to send or receive
> with it.

Perhaps I am being pedantic, but if I am only able to send or receive
with it, isn't it not a sendrecv codec?



Anyway, I am willing to be flexible here, but to me this is a reasonably
big change, and so I'd like to here from other folks.....

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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Hi Folks,

There are enough SIP implementations on the market that would require treating the set of codecs in the offer and the answer as two independent sets. We can decide whatever we want in the standard but interoperability with existing solutions will force people to implement this anyway. Furthermore, the behavior in which offer and answer are treated separately is more generic and includes all possible scenarios.

So what I offer is that if offer for sendonly or sendrecv offer contains codecs A, B, C, the answer can contain A, B, D. Additionally answer can use different payload types for A and B. The only requirements are that the offer and the answer contain at least one codec of the same type and the answerer must use the payload types from the offer. 
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
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Hi Jonathan.

Thanks for the responses - I've snipped the ones we are in sync on and/or are
being discussed elsewhere. Whatever is left here is really minor.

Jonathan Rosenberg wrote:

> >
> > - p. 4:    "The agent receiving the offer MAY generate an answer, or it
> > MAY reject the offer.". At least in the unicast case, MUST you not
> > always generate an answer which may either accept or reject the offer ?
>
> No. You can reject the offer outright. In the case of SIP, that would
> mean sending
> a 488. There is no answer in the 488.

OK, however an answer may also reject an offer by rejecting the streams. No
big deal.


>
> >
> > p. 10:    "The answerer MAY include a ptime" => "The answerer MAY
> > include a non-zero ptime"
>
> I didn't include that restriction for the offer, and you had no comment
> on it there. Do I interpret this to mean you want this restriction only
> for the answer, or do you really want it for both?

Should apply to both.

> It had never even
> occurred to me to use a zero ptime, I don't even know what that would
> mean....

Me neither, but then again I don't think it makes any sense to have bandwidth
attributes with a value of zero, but I'll leave that one alone ;-)


>
>
> >
> > p. 10:    "When sending media, it SHOULD use a packetization interval
> > equal to the value of the ptime attribute in the offer, if any was
> > present.". Since ptime is a media stream attribute, this can be
> > difficult to do if multiple media formats are specified. There should be
> > a rounding rule to account for that (e.g. use closest non-zero ptime if
> > you can't use the ptime specified).
>
> This is really a general SDP issue, not specific to O/A. Do we want to
> discuss it here? Probably sdp-new should clarify this.
>

Good point (only minor issue then is that we are referencing RFC 2327
throughout, and not sdp-new).

>
> >
> > p. 11:    "Similarly, the directionality information in the answer
> > (sendonly, recvonly, or sendrecv) MUST equal that of the offer." It may
> > be worth spelling it out more clearly what this means, since it differs
> > from the unicast.
>
> I mention it in the text on constructing the offer, see above response.

I still think it's worth spelling it out, since the less than careful reader
may not pick up on it. Not a big deal.


>
>
> >
> > p. 13:    "Removal of a media stream implies that media is no longer
> > sent for that stream.". Add sentence "Furthermore, when RTP was used for
> > the media stream, RTCP reports will no longer be sent or received."
>
> Added:
>
> In the case of RTP, RTCP transmission also ceases.

Not only that, you also will not accept any incoming RTCP messages, right
(transmission only implies sending) ?


>
> >
> > Section 8.4:    Section seems to be unicast specific. Both heading and
> > section text should reflect this.
>
> I changed the section heading to "putting a unicast stream on hold". I
> see no need to repeat that within the body text.
>

I like being explict but OK. How about at least adding "unicast" in the first
paragraph, i.e. "one or more media streams" => "one or more unicast media
streams".


> >
> > Section 9:    "The transport component of the "m=" line indicates the
> > preferred transport for that media type.". Not sure this is ideal. For
> > example, if I support both RTP and SRTP, do I only indicate one of them
> > (similar issue with  T.38 UDP or TCP) ?
>
> Yes. Its not really preferred, its "the one". If you want to indicate
> support for both SRTP and RTP, you would pick one in the offer, include
> simcaps indicating the other. Similar in the answer. THe offerer could
> then send an updated offer switching to SRTP. This is discussed in the
> unify draft.

Never mind - I got confused on OPTIONS for a second.


-- Flemming

--
Flemming Andreasen
Cisco Systems



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Jonathan Rosenberg wrote:

> inline. Let us resolve this quickly, only a few days remain in wglc:
>
> Flemming Andreasen wrote:
> >
>
> > > In the case of sendrecv, subsetting is the only thing that makes
> > sense.
> > > If A indicates in an offer that it can send and receive with codecs
> > > 1,2,3, and B indicates in its answer that it can send and receive with
> > > 2,3,4, the presence of 4 is useless. B can't send with it, since it
> > > doesn't know whether A can receive it.
> >
> > Correct, but A can send with it.
>
> So, if A can send with it, but B can't, how is this a sendrecv codec,
> practically speaking?

It is a sendrecv codec inasmuch, that if B sends an offer with this codec (as
sendrecv), it knows that A can actually support it.


>
> > >
> > > Now, in the case of a recvonly offer, A could indicate the ability to
> > > receive 1,2, and 3. In the answer, lets say B returns 1,2 as sendonly.
> > > But, then it sends using 3 anyway. If the answerer does not go with
> > > whats in the answer, the answer provides zero information. The point
> > of
> > > the answer in this case is to help the offerer restrict the set of
> > > codecs it needs to be prepared to receive.
> >
> > I thought it was so offerer can figure out what he can actually send
> > with
> > (ignoring some of the unidirectional stream cases for a momemt).
>
> You've lost me; we are talking explicitly about the unidirectional cases
> here.
>

I thought you were referring to offer/answer in general. The purpose of an
offer/answer should be the same for unidirectional and bidirectional media
streams. The question in general is, if A sends an offer indicating
willingness to receive codecs 1,2, and 3, and B sends an answer omitting one
or more of these codecs, is A still obligated to receive all three codecs. I
don't see this as a unidirectional stream issue only.

>
> >
> > > I do admit that I find no
> > > text at the moment which says that the answerer in a sendonly answer
> > > should send only with the codecs in the answer. However, that has been
> > > my interprtation.
> >
> > I have always understood, that once I hand out an SDP indicating the
> > ability
> > to receive codec 1,2, and 3, then I have to be able to receive those
> > codecs
> > until I send out another SDP.
>
> It is certainly true that you need to be able to receive them once you
> send them, to handle media clipping. It had been my implicit assumption,
> arguably not well thought thruough, that the answer would restrict what
> the element needs to be prepared to receive. I suppose that, since it
> needs to be able to receive them initially anyway, is there much gained
> by learning that the set of things to receive is less than it needs to
> initially?
>

There is, but in the process, we loose some abilities too. Since we could
effectively achieve the same by issuing a new offer with the "answer" set of
codecs, I'm not convinced this is a good trade-off.


>
> One example is if A can support codecs 1,2,3,4,5, and it sends an offer
> to B which really only supports 1, and thats what is in the answer. Now,
> A would still need to be preapred to reeive 1-5 based on your proposal,
> even though it effectively knows that B is never going to use anythign
> but 1.

That's where we differ - I don't think A can make that assumption.


> I had assumed that the point of the offer/answer exchange was to
> do a one-pass subsetting of the codecs in use. I am sure we have

> discussed this in the past in the sip wg.
>

>
> > Of course you can say, that this is
> > because SDP
> > never defined the negotiation that offer/answer does, but this seems to
> > introduce an unnecesary deviation from "normal" SDP usage.
>
> I can see that it is a bit odd.
>
> I suppose that, if the offerer really wants to know a more restricted
> set, it can always send an updated offer with fewer codecs, ala the
> 1-of-N mechanism in the examples in the draft. But, isn't it better to
> avoid that if you know that the answer defines a subset that is
> acceptable?

We loose something in the process as described above, and hence I'm not
convinced. Also, I am concerned about deviating from "normal" SDP usage (the
way I've understood it so far at least).

> >
> > >
> > >
> > > Let me ask the converse - what is gained by having the answerer be
> > able
> > > to send with things not in the answer? Why not just include them in
> > the
> > > answer?
> >
> > It could for example be that I can send RFC 2833 but not receive it.
> > Admittedly, SDP is not really good at expressing such unidirectional
> > constraints, and we probably shouldn't get hung up on it (however I
> > thought
> > this was one of your concerns). My real concern here is that it seems to
> > deviate from current SDP behavior.
>
> I guess I am confused on this, because we had text along those lines
> before, and we agreed at ietf 52 to remove it. Specifically, we agreed
> that, in a sendrecv answer, you cannot add codecs which are recvonly,
> they have to be sendrecv. From the above, it sounds like you are backing
> down from that.

No - I'm still saying that if an agent lists a media formats listed in a
sendrecv stream then the agent must support that media format bidirectionally
(regardless of whether the context implies that it can only be used
unidirectionally). In the above example, if the (sendrecv) offer contains RFC
2833, but the answer does not, the answerer might still be able to send with
RFC 2833, but not receive it (which is why he didn't include it in the
answer).

>
>
> > > > Same reason as the answerer might want to add a media format not
> > present
> > > > in
> > > > the offer:    asymmetry (e.g. I can send RFC 2833 to you, but I
> > can't
> > > > receive it).
> > >
> > > Remember, we only support asymmetric codecs in asymmetric stream
> > types.
> >
> > We do ? If I offer 2 or more codecs and you answer with two or more
> > codecs,
> > then we can have asymmetric codec usage.
>
> See above comment; I thought that we agreed that a sendrecv stream had
> to list sendrecv codecs.
>

We still do, however whether the context in which is supplied may still lead
to it only being usable in one direction. If the context changes, then maybe
it can be used bidirectionally.


>
> >
> > >
> > > In the case you describe, you are talking about a recvonly offer and a
> > > sendonly answer. The spec says that the offer lists the codecs the
> > > offerer can receive. If its not there, what is gained by adding it in
> > > the answer? The answerer can't send it anyway, since it doesn't know
> > if
> > > the offerer supports it?
> > >
> >
> > However, I thought it would apply to bidirectional streams as well. For
> > example, if RFC 2833 is not offered , but the answerer is able to accept
> > it,
> > he may want to indicate so in the answer (and vice versa).
>
> As above, I removed text that explicitly allowed this based on iETF 52
> consensus, where we agreed fid was better for it.
>

I believe that was different inasmuch as that was referring to unidirectional
capabilities on bidirectional streams. If I recall correctly, the fid-draft
would allow the offerer to indicate unidirectional capabilities, and have the
answerer respond accordingly. However, the answerer cannot add media streams
to indicate unidirectional capabilities in response to bidirectional media
stream.

What I am talking about is bidirectional capabilities on bidirectional
streams, however the context may imply that the flow of a particular media
format can only occur in one direction (e.g. because only the offer or the
answer contained that media format).


> > >
> > > We did previously allow addition of recvonly codecs by the answerer in
> > a
> > > sendrecv stream, but agreed at IETF 52 to remove this hack in favor of
> > > FID, which is the proper way to support such things. I recall you
> > argued
> > > in favor of that, even.
> >
> > I still do, however that doesn't mean you can't add a sendrecv codec to
> > a
> > sendrecv stream, even though you may only be able to send or receive
> > with it.
>
> Perhaps I am being pedantic, but if I am only able to send or receive
> with it, isn't it not a sendrecv codec?

It may only be able to be used unidirectionally in the currect context,
however if the context changes, we know that it is supported bidirectionally
(3pcc for example).

>
> Anyway, I am willing to be flexible here, but to me this is a reasonably
> big change,

Maybe from the current text, but the discussions we had on the previous
version of the draft at least was intended to convey the above.

-- Flemming


> and so I'd like to here from other folks.....

>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Thu Feb 14 11:04:52 2002
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Flemming Andreasen wrote:
> 
> > > - p. 4:    "The agent receiving the offer MAY generate an answer, or
> it
> > > MAY reject the offer.". At least in the unicast case, MUST you not
> > > always generate an answer which may either accept or reject the
> offer ?
> >
> > No. You can reject the offer outright. In the case of SIP, that would
> > mean sending
> > a 488. There is no answer in the 488.
> 
> OK, however an answer may also reject an offer by rejecting the streams.
> No
> big deal.

Right, of course. However, there is still in answer in that case. So,
the two cases still remain - MAY generate an answer, or MAY reject it.

> 
> >
> > >
> > > p. 10:    "The answerer MAY include a ptime" => "The answerer MAY
> > > include a non-zero ptime"
> >
> > I didn't include that restriction for the offer, and you had no
> comment
> > on it there. Do I interpret this to mean you want this restriction
> only
> > for the answer, or do you really want it for both?
> 
> Should apply to both.

I'll add nonzero, but I suspect most folks wouldn't have even thought to
put zero in there.


> 
> >
> >
> > >
> > > p. 10:    "When sending media, it SHOULD use a packetization
> interval
> > > equal to the value of the ptime attribute in the offer, if any was
> > > present.". Since ptime is a media stream attribute, this can be
> > > difficult to do if multiple media formats are specified. There
> should be
> > > a rounding rule to account for that (e.g. use closest non-zero ptime
> if
> > > you can't use the ptime specified).
> >
> > This is really a general SDP issue, not specific to O/A. Do we want to
> > discuss it here? Probably sdp-new should clarify this.
> >
> 
> Good point (only minor issue then is that we are referencing RFC 2327
> throughout, and not sdp-new).

I'd rather not discuss this in O/A, even if we only refer to 2327. In
reality, people can and will look at sdp-new. What we do need to do is
make sure that the text on this goes into sdp-new...

> 
> >
> > >
> > > p. 11:    "Similarly, the directionality information in the answer
> > > (sendonly, recvonly, or sendrecv) MUST equal that of the offer." It
> may
> > > be worth spelling it out more clearly what this means, since it
> differs
> > > from the unicast.
> >
> > I mention it in the text on constructing the offer, see above
> response.
> 
> I still think it's worth spelling it out, since the less than careful
> reader
> may not pick up on it. Not a big deal.

OK. I added this:

This is because all
participants in a multicast session need to have equivalent views of
the parameters of the session, and underlying assumption of the
multicast bias of RFC 2327.


> 
> >
> >
> > >
> > > p. 13:    "Removal of a media stream implies that media is no longer
> > > sent for that stream.". Add sentence "Furthermore, when RTP was used
> for
> > > the media stream, RTCP reports will no longer be sent or received."
> >
> > Added:
> >
> > In the case of RTP, RTCP transmission also ceases.
> 
> Not only that, you also will not accept any incoming RTCP messages,
> right
> (transmission only implies sending) ?

True, and then I need to say the same for the RTP itself, since the text
also only says to stop sending. I'll change. It now reads:

Removal of a media stream implies that media is no longer sent for
that stream, and any media that is received is discarded. In the case
of RTP, RTCP transmission also ceases, as does processing of any
received RTCP packets.


> 
> >
> > >
> > > Section 8.4:    Section seems to be unicast specific. Both heading
> and
> > > section text should reflect this.
> >
> > I changed the section heading to "putting a unicast stream on hold". I
> > see no need to repeat that within the body text.
> >
> 
> I like being explict but OK. How about at least adding "unicast" in the
> first
> paragraph, i.e. "one or more media streams" => "one or more unicast
> media
> streams".

Done.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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Roman Shpount wrote:
> 
> Jonathan D. Rosenberg wrote:
> 
> > Let me ask the converse - what is gained by having the answerer be
> able
> > to send with things not in the answer? Why not just include them in
> the
> > answer?
> 
> This is primarily a question of interpretation. I know at least one
> gateway implementation which is capable of sending, but not receiving
> avt tones. So this gateway replies to a sendrecv offer, which contains
> avt tones and G.711, with an answer that contains only G.711, but sends
> out avt tones anyway. I am looking for the language that states whether
> this behavior is legal or not.

I had text that allowed exactly this in the previous version, for
exactly this reason. Consensus at the last IETF was to remove it. The
reason was, that in a sendrecv stream, the codecs you list have to be
sendrecv codecs. I think that all of the confusion here is about the
interpretation of what it means to be a "sendrecv" codec... please see
my response to Flemming.

Just as a note, we have a DROP DEAD date for completion of this document
by monday, so let us not let this thread die! So far so good, but I just
want to be sure folks are aware of the time pressures we are under.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Thu Feb 14 11:11:24 2002
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Looks good.

Thanks a lot

            Flemming

Jonathan Rosenberg wrote:

> Flemming Andreasen wrote:
> >
> > > > - p. 4:    "The agent receiving the offer MAY generate an answer, or
> > it
> > > > MAY reject the offer.". At least in the unicast case, MUST you not
> > > > always generate an answer which may either accept or reject the
> > offer ?
> > >
> > > No. You can reject the offer outright. In the case of SIP, that would
> > > mean sending
> > > a 488. There is no answer in the 488.
> >
> > OK, however an answer may also reject an offer by rejecting the streams.
> > No
> > big deal.
>
> Right, of course. However, there is still in answer in that case. So,
> the two cases still remain - MAY generate an answer, or MAY reject it.
>
> >
> > >
> > > >
> > > > p. 10:    "The answerer MAY include a ptime" => "The answerer MAY
> > > > include a non-zero ptime"
> > >
> > > I didn't include that restriction for the offer, and you had no
> > comment
> > > on it there. Do I interpret this to mean you want this restriction
> > only
> > > for the answer, or do you really want it for both?
> >
> > Should apply to both.
>
> I'll add nonzero, but I suspect most folks wouldn't have even thought to
> put zero in there.
>
> >
> > >
> > >
> > > >
> > > > p. 10:    "When sending media, it SHOULD use a packetization
> > interval
> > > > equal to the value of the ptime attribute in the offer, if any was
> > > > present.". Since ptime is a media stream attribute, this can be
> > > > difficult to do if multiple media formats are specified. There
> > should be
> > > > a rounding rule to account for that (e.g. use closest non-zero ptime
> > if
> > > > you can't use the ptime specified).
> > >
> > > This is really a general SDP issue, not specific to O/A. Do we want to
> > > discuss it here? Probably sdp-new should clarify this.
> > >
> >
> > Good point (only minor issue then is that we are referencing RFC 2327
> > throughout, and not sdp-new).
>
> I'd rather not discuss this in O/A, even if we only refer to 2327. In
> reality, people can and will look at sdp-new. What we do need to do is
> make sure that the text on this goes into sdp-new...
>
> >
> > >
> > > >
> > > > p. 11:    "Similarly, the directionality information in the answer
> > > > (sendonly, recvonly, or sendrecv) MUST equal that of the offer." It
> > may
> > > > be worth spelling it out more clearly what this means, since it
> > differs
> > > > from the unicast.
> > >
> > > I mention it in the text on constructing the offer, see above
> > response.
> >
> > I still think it's worth spelling it out, since the less than careful
> > reader
> > may not pick up on it. Not a big deal.
>
> OK. I added this:
>
> This is because all
> participants in a multicast session need to have equivalent views of
> the parameters of the session, and underlying assumption of the
> multicast bias of RFC 2327.
>
> >
> > >
> > >
> > > >
> > > > p. 13:    "Removal of a media stream implies that media is no longer
> > > > sent for that stream.". Add sentence "Furthermore, when RTP was used
> > for
> > > > the media stream, RTCP reports will no longer be sent or received."
> > >
> > > Added:
> > >
> > > In the case of RTP, RTCP transmission also ceases.
> >
> > Not only that, you also will not accept any incoming RTCP messages,
> > right
> > (transmission only implies sending) ?
>
> True, and then I need to say the same for the RTP itself, since the text
> also only says to stop sending. I'll change. It now reads:
>
> Removal of a media stream implies that media is no longer sent for
> that stream, and any media that is received is discarded. In the case
> of RTP, RTCP transmission also ceases, as does processing of any
> received RTCP packets.
>
> >
> > >
> > > >
> > > > Section 8.4:    Section seems to be unicast specific. Both heading
> > and
> > > > section text should reflect this.
> > >
> > > I changed the section heading to "putting a unicast stream on hold". I
> > > see no need to repeat that within the body text.
> > >
> >
> > I like being explict but OK. How about at least adding "unicast" in the
> > first
> > paragraph, i.e. "one or more media streams" => "one or more unicast
> > media
> > streams".
>
> Done.
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Thu Feb 14 12:36:06 2002
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Jonathan D. Rosenberg wrote:

> Just as a note, we have a DROP DEAD date for completion of this document
> by monday, so let us not let this thread die! So far so good, but I just
> want to be sure folks are aware of the time pressures we are under.

I understand the time pressure. All I want is to have a consistent behavior specified by the standard. The original rfc2543 use to specify in section B.2:

   If the media stream is listed as both send and receive, the callee 
   lists those codecs it is capable of sending or receiving among the 
   ones listed by the caller in the INVITE. 

Starting from rfc2543bis3 it it changed to (Section B.2.1):

   If multiple codecs are listed, it means that the offerer is capable 
   of making use of any of those codecs during the call. In other words,
   the answerer MAY change codecs in the middle of the call, without
   sending a SIP message, to make use of any of those listed.

Once again, there are enough rfc2543bis3 compliant implementations out there to justify not changing the offer/answer behavior once more. The later model can easily be implemented, since the support for it is already in place for early media. In any case, we should at least specify that the offerer MUST accept all the codecs from the original offer, regardless whether the answerer MAY or SHOULD not send using any codecs not specified in the answer.
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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From mmusic-admin@ietf.org  Thu Feb 14 13:39:21 2002
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CC: Roman Shpount <roman@atelo.com>, mmusic@ietf.org,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [MMUSIC] Re: Using payload types not specified in answer in  
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Flemming Andreasen wrote:
> 

> I thought you were referring to offer/answer in general. The purpose of
> an
> offer/answer should be the same for unidirectional and bidirectional
> media
> streams. The question in general is, if A sends an offer indicating
> willingness to receive codecs 1,2, and 3, and B sends an answer omitting
> one
> or more of these codecs, is A still obligated to receive all three
> codecs. I
> don't see this as a unidirectional stream issue only.

That is the question, indeed. 

So, I sent private emails to a bunch of folks who I suspect have an
opinion, but who haven't been following the threads (I fear that the
huge volume that has been going on on the lists, and the fast pace, have
pushed a lot of people out of touch). All of the replies I got (about 5)
indicated that peoples interpretations had been the same as yours and
Roman's, with one person waffling a bit.

So, in the interests of progress and rough consensus, I will declare
myself in the apparent minority. I see and understand the points
Flemming is making; I don't think either side is technicall incorrect,
it is merely an issue of interpretation. I will change the text to
indicate that the offerer needs to be prepared to always receive any of
them, and that the answerer can send with ones not present in the answer
(but won't receive, obviously).

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Thu Feb 14 13:54:04 2002
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CC: Roman Shpount <roman@atelo.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Re: Using payload types not specified in answer in  
 offer/answer
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Sounds good - that addresses all of my comments then.

Thanks a lot for all your hard work on this.

-- Flemming


Jonathan Rosenberg wrote:

> Flemming Andreasen wrote:
> >
>
> > I thought you were referring to offer/answer in general. The purpose of
> > an
> > offer/answer should be the same for unidirectional and bidirectional
> > media
> > streams. The question in general is, if A sends an offer indicating
> > willingness to receive codecs 1,2, and 3, and B sends an answer omitting
> > one
> > or more of these codecs, is A still obligated to receive all three
> > codecs. I
> > don't see this as a unidirectional stream issue only.
>
> That is the question, indeed.
>
> So, I sent private emails to a bunch of folks who I suspect have an
> opinion, but who haven't been following the threads (I fear that the
> huge volume that has been going on on the lists, and the fast pace, have
> pushed a lot of people out of touch). All of the replies I got (about 5)
> indicated that peoples interpretations had been the same as yours and
> Roman's, with one person waffling a bit.
>
> So, in the interests of progress and rough consensus, I will declare
> myself in the apparent minority. I see and understand the points
> Flemming is making; I don't think either side is technicall incorrect,
> it is merely an issue of interpretation. I will change the text to
> indicate that the offerer needs to be prepared to always receive any of
> them, and that the answerer can send with ones not present in the answer
> (but won't receive, obviously).
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Thu Feb 14 15:51:10 2002
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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Flemming Andreasen'" <fandreas@cisco.com>
Cc: "'Roman Shpount'" <roman@atelo.com>, <mmusic@ietf.org>
Subject: RE: [MMUSIC] Re: Using payload types not specified in answer in   offer/answer
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I was trying to find a place to jump in (with all the traffic and its pace
:) but I see I'm late now.  I'll say my 2 cents anyway...

I always thought not getting an affirmative indication of what a sender will
send was a deficiency in the protocol.  For a multi-session platform that
wants to optimize its resources, this behavior is not desirable if its goal
is to try to find one format the other end supports.  In order to minimize
resource usage (i.e., not allocate DSP resources for formats that will
probably not be used in the session), I can only "advertise" what I'm
willing to reserve.  Which will be a single encoding format in almost all
cases.  I liked where Jonathan was going, but I understand consensus...

If I look at the case where a gateway can send rfc2833 and an endpoint wants
it, I would prefer to see the gateway "advertise" support with the sendonly
attribute rather than it saying nothing.

Regards,
Bert

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: Thursday, February 14, 2002 1:38 PM
> To: Flemming Andreasen
> Cc: Roman Shpount; mmusic@ietf.org; Jonathan Rosenberg
> Subject: Re: [MMUSIC] Re: Using payload types not specified
> in answer in
> offer/answer
>
>
>
>
> Flemming Andreasen wrote:
> >
>
> > I thought you were referring to offer/answer in general.
> The purpose of
> > an
> > offer/answer should be the same for unidirectional and bidirectional
> > media
> > streams. The question in general is, if A sends an offer indicating
> > willingness to receive codecs 1,2, and 3, and B sends an
> answer omitting
> > one
> > or more of these codecs, is A still obligated to receive all three
> > codecs. I
> > don't see this as a unidirectional stream issue only.
>
> That is the question, indeed.
>
> So, I sent private emails to a bunch of folks who I suspect have an
> opinion, but who haven't been following the threads (I fear that the
> huge volume that has been going on on the lists, and the fast
> pace, have
> pushed a lot of people out of touch). All of the replies I
> got (about 5)
> indicated that peoples interpretations had been the same as yours and
> Roman's, with one person waffling a bit.
>
> So, in the interests of progress and rough consensus, I will declare
> myself in the apparent minority. I see and understand the points
> Flemming is making; I don't think either side is technicall incorrect,
> it is merely an issue of interpretation. I will change the text to
> indicate that the offerer needs to be prepared to always
> receive any of
> them, and that the answerer can send with ones not present in
> the answer
> (but won't receive, obviously).
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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From mmusic-admin@ietf.org  Thu Feb 14 16:08:11 2002
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Bert Culpepper wrote:

> I was trying to find a place to jump in (with all the traffic and its pace
> :) but I see I'm late now.  I'll say my 2 cents anyway...
>
> I always thought not getting an affirmative indication of what a sender will
> send was a deficiency in the protocol.  For a multi-session platform that
> wants to optimize its resources, this behavior is not desirable if its goal
> is to try to find one format the other end supports.  In order to minimize
> resource usage (i.e., not allocate DSP resources for formats that will
> probably not be used in the session), I can only "advertise" what I'm
> willing to reserve.  Which will be a single encoding format in almost all
> cases.  I liked where Jonathan was going, but I understand consensus...
>

Also note, that you can achieve almost the same by issuing an updated
offer/answer with the the single codec, if that's what you want.

>
> If I look at the case where a gateway can send rfc2833 and an endpoint wants
> it, I would prefer to see the gateway "advertise" support with the sendonly
> attribute rather than it saying nothing.

There's only so much we can do within the existing semantics of SDP, however you
can still use the fid-draft to provide unidirectional media formats in an offer
- you just can't add it to the answer if it the offer didn't contain it.

-- Flemming

--
Flemming Andreasen
Cisco Systems



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Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Roman Shpount'" <roman@atelo.com>, <mmusic@ietf.org>
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>
> Also note, that you can achieve almost the same by issuing an updated
> offer/answer with the the single codec, if that's what you want.

I shouldn't be in so much of a hurry to jump in...You're right, with an
extra message exchange I can satisfy both goals.  This will work for me.

Thanks,
Bert



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I got some more responses in private, and also noodled over this some
more. I still have a queasy feeling.

The question is really fundamental. What is the "answer" in an O/A
exchange? Is it a declaration, as is the offer, which are not much
related, or is it a negotiation to arrive at agreed upon parameters? One
way to figure it out is to look at other aspects of the SDP negotiation.

Consider the a=sendrecv, a=sendonly, a=recvonly attributes. If A sends
an offer, with the stream as sendrecv, and the answer (from B) says
recvonly, here is the question. Can B now send to A? If you believe that
the directional attributes are a shared, negotiated view of the session,
which has been my interpetation, the answer is NO. This stream is from A
to B, period. If you believe that B always can do whatever is declared
to it in the offer, the answer is yes - A can receive, so B should be
able to send.

Really, the offer/answer model has some attributes which are merely
declarative (like ptime), and others which are shared and negotiated
(directionality). Therefore, the argument that SDP is always declarative
holds no ground, since I think (hope) we can agree that there are most
definitely attributes which are shared and negotiated. So, are, or are
not, codecs part of this? If the goal, in general, is for a common codec
to be used in each direction, then I think the view has to be that they
are shared. While we could just reduce the set later with an additional
offer, as Flemming has pointed out, the value that the O/A exchange
brings to the table is the ability to not need to do that. 

Thanks,
Jonathan R.

Bert Culpepper wrote:
> 
> >
> > Also note, that you can achieve almost the same by issuing an updated
> > offer/answer with the the single codec, if that's what you want.
> 
> I shouldn't be in so much of a hurry to jump in...You're right, with an
> extra message exchange I can satisfy both goals.  This will work for me.
> 
> Thanks,
> Bert

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Fri Feb 15 03:24:56 2002
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I would definitely agree with Jonathan in his point of view about codec negotiation if we had a luxury of defining a new standard. Unfortunately what is being done here (at least from my point of view) is just an attempt to ensure interoperability of existing applications. My suggestion for this standard would be something along the lines:

  Offerer MUST accept all the codecs from the original offer. Answerer 
  SHOULD NOT send using any codecs not specified in the answer.

This way we have implementations compatible with both rfc2543 and rfc2543bis3. It is a compromise that I think should always work.

___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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From mmusic-admin@ietf.org  Fri Feb 15 04:30:08 2002
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Hi all,

Suppose the offerer is dual stack and does not know which version of 
IP is supported by the answerer.  The answerer may support both or 
only one IP version, either v4 or v6.

In this case, the offerer may use the offer that includes two m lines 
(one IPv4 and the other IPv6) grouped using FID.  Both lines contains 
the same codec(s).  What the offerer expects is that the answerer 
selects only one of the two m lines, i.e. the answerer accepts 
one m line with an appropriate port, and explicitly rejects 
the other m line with port=0 in the corresponding m lines in the answer.

However, the answerer may accept both of them, and if so, it means that 
two RTP sessions at the same time, as the draft-ietf-mmusic-fid-05 says, 
'Remember that if several "m" lines grouped together using FID semantics 
contain the same codec the media agent MUST send media over several 
RTP sessions at the same time' in 7.4.1.

How can the offerer prevent it?
Do we need a new mechanism to show "alternative" m lines to let 
the answer know that it must select only one of them?

Regards,
Hiroshi Waki

NTT Network Service Systems Laboratories
tel +81 422 59 3404  fax +81 422 59 3494
e-mail  waki.hiroshi@lab.ntt.co.jp




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From mmusic-admin@ietf.org  Fri Feb 15 10:03:42 2002
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Jonathan Rosenberg wrote:

> Really, the offer/answer model has some attributes which are merely
> declarative (like ptime), and others which are shared and negotiated
> (directionality). Therefore, the argument that SDP is always declarative
> holds no ground, since I think (hope) we can agree that there are most
> definitely attributes which are shared and negotiated. So, are, or are
> not, codecs part of this? If the goal, in general, is for a common codec
> to be used in each direction, then I think the view has to be that they
> are shared. While we could just reduce the set later with an additional
> offer, as Flemming has pointed out, the value that the O/A exchange
> brings to the table is the ability to not need to do that.

Actually, it doesn't. With the present O/A text, we already have to support
all codecs in the offer until we get the answer. Furthermore, since the
offerer has no control over the answer, it is quite possible that the answer
contains more than one of the offered codec, and hence we are back where we
started. The only way to get down to one codec is a new offer/answer exchange.
Thus, the above can at most be seen as a possible optimization between
offerers and answerers that both agree to pick just one codec. Since it
doesn't really solve the problem then, we loose some functionality in the
process, and we depart with what at least a lot of people had understood
(based on your earlier e-mail), we should stick to yesterday's consensus.

-- Flemming


>
>
> Thanks,
> Jonathan R.
>
> Bert Culpepper wrote:
> >
> > >
> > > Also note, that you can achieve almost the same by issuing an updated
> > > offer/answer with the the single codec, if that's what you want.
> >
> > I shouldn't be in so much of a hurry to jump in...You're right, with an
> > extra message exchange I can satisfy both goals.  This will work for me.
> >
> > Thanks,
> > Bert
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 10:08:11 2002
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Roman Shpount wrote:

> I would definitely agree with Jonathan in his point of view about codec negotiation if we had a luxury of defining a new standard. Unfortunately what is being done here (at least from my point of view) is just an attempt to ensure interoperability of existing applications. My suggestion for this standard would be something along the lines:
>
>   Offerer MUST accept all the codecs from the original offer. Answerer
>   SHOULD NOT send using any codecs not specified in the answer.
>

"SHOULD NOT" is not to be taken lightly, and hence this effectively says that you can only use the codecs in the answer. As I indicated in another e-mail to Jonathan, there's little to be gained from this restriction, but things to be lost, and hence we should continue with the practice of allowing the answerer to send with offered codecs not
contained in the answer.

-- Flemming



>
> This way we have implementations compatible with both rfc2543 and rfc2543bis3. It is a compromise that I think should always work.
>
> ___________________________________
> Roman Shpount, VP of Technology
> aTelo, Inc.
> 2107 Wilson Blvd., Suite 301
> Arlington, VA 22201
> PCN: 703-621-4040 for voice / fax
> www.atelo.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 10:19:34 2002
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Hiroshi Waki wrote:
> 
> Hi all,
> 
> Suppose the offerer is dual stack and does not know which version of
> IP is supported by the answerer.  The answerer may support both or
> only one IP version, either v4 or v6.
> 
> In this case, the offerer may use the offer that includes two m lines
> (one IPv4 and the other IPv6) grouped using FID.  Both lines contains
> the same codec(s).  What the offerer expects is that the answerer
> selects only one of the two m lines, i.e. the answerer accepts
> one m line with an appropriate port, and explicitly rejects
> the other m line with port=0 in the corresponding m lines in the answer.
> 
> However, the answerer may accept both of them, and if so, it means that
> two RTP sessions at the same time, as the draft-ietf-mmusic-fid-05 says,
> 
> 'Remember that if several "m" lines grouped together using FID semantics
> 
> contain the same codec the media agent MUST send media over several
> RTP sessions at the same time' in 7.4.1.
> 
> How can the offerer prevent it?
> Do we need a new mechanism to show "alternative" m lines to let
> the answer know that it must select only one of them?

You can always send an updated offer with the v4 stream disabled. In
fact, you can avoid fid altogether, and send the offer with both v4 and
v6, and once you get the answer, send an updated offer with just one of
them immediately (if needed).

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From: "Sanjoy Sen"<sanjoy@nortelnetworks.com>
To: "'Flemming Andreasen'" <fandreas@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: Bert Culpepper <bert.culpepper@intervoice-brite.com>,
        "'Roman Shpount'"
	 <roman@atelo.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] Re: Using payload types not specified in answer in  
	 offer/answer
Date: Fri, 15 Feb 2002 11:22:16 -0600
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Doesn't the new consensus mean that we're back to the original requirement
that the client (offerer) needs to load multiple DSP's and be prepared to
receive on any of them (at least until the second offer/answer is made)? Am
I missing something? We should have a mechanism to avoid this for lesser
clients that can only use one DSP resource at a time. What happened to the
a=pickone attribute?

Sanjoy

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Friday, February 15, 2002 9:03 AM
> To: Jonathan Rosenberg
> Cc: Bert Culpepper; 'Roman Shpount'; mmusic@ietf.org
> Subject: Re: [MMUSIC] Re: Using payload types not specified 
> in answer in
> offer/answer
> 
> 
> 
> 
> Jonathan Rosenberg wrote:
> 
> > Really, the offer/answer model has some attributes which are merely
> > declarative (like ptime), and others which are shared and negotiated
> > (directionality). Therefore, the argument that SDP is 
> always declarative
> > holds no ground, since I think (hope) we can agree that 
> there are most
> > definitely attributes which are shared and negotiated. So, 
> are, or are
> > not, codecs part of this? If the goal, in general, is for a 
> common codec
> > to be used in each direction, then I think the view has to 
> be that they
> > are shared. While we could just reduce the set later with 
> an additional
> > offer, as Flemming has pointed out, the value that the O/A exchange
> > brings to the table is the ability to not need to do that.
> 
> Actually, it doesn't. With the present O/A text, we already 
> have to support
> all codecs in the offer until we get the answer. Furthermore, 
> since the
> offerer has no control over the answer, it is quite possible 
> that the answer
> contains more than one of the offered codec, and hence we are 
> back where we
> started. The only way to get down to one codec is a new 
> offer/answer exchange.
> Thus, the above can at most be seen as a possible optimization between
> offerers and answerers that both agree to pick just one 
> codec. Since it
> doesn't really solve the problem then, we loose some 
> functionality in the
> process, and we depart with what at least a lot of people had 
> understood
> (based on your earlier e-mail), we should stick to 
> yesterday's consensus.
> 
> -- Flemming
> 
> 
> >
> >
> > Thanks,
> > Jonathan R.
> >
> > Bert Culpepper wrote:
> > >
> > > >
> > > > Also note, that you can achieve almost the same by 
> issuing an updated
> > > > offer/answer with the the single codec, if that's what you want.
> > >
> > > I shouldn't be in so much of a hurry to jump in...You're 
> right, with an
> > > extra message exchange I can satisfy both goals.  This 
> will work for me.
> > >
> > > Thanks,
> > > Bert
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com
> 
> --
> Flemming Andreasen
> Cisco Systems
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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<TITLE>RE: [MMUSIC] Re: Using payload types not specified in answer in  =
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<P><FONT SIZE=3D2>Doesn't the new consensus mean that we're back to the =
original requirement that the client (offerer) needs to load multiple =
DSP's and be prepared to receive on any of them (at least until the =
second offer/answer is made)? Am I missing something? We should have a =
mechanism to avoid this for lesser clients that can only use one DSP =
resource at a time. What happened to the a=3Dpickone =
attribute?</FONT></P>

<P><FONT SIZE=3D2>Sanjoy</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Flemming Andreasen [<A =
HREF=3D"mailto:fandreas@cisco.com">mailto:fandreas@cisco.com</A>]</FONT>=

<BR><FONT SIZE=3D2>&gt; Sent: Friday, February 15, 2002 9:03 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Jonathan Rosenberg</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: Bert Culpepper; 'Roman Shpount'; =
mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [MMUSIC] Re: Using payload types =
not specified </FONT>
<BR><FONT SIZE=3D2>&gt; in answer in</FONT>
<BR><FONT SIZE=3D2>&gt; offer/answer</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan Rosenberg wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Really, the offer/answer model has some =
attributes which are merely</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; declarative (like ptime), and others which =
are shared and negotiated</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (directionality). Therefore, the argument =
that SDP is </FONT>
<BR><FONT SIZE=3D2>&gt; always declarative</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; holds no ground, since I think (hope) we =
can agree that </FONT>
<BR><FONT SIZE=3D2>&gt; there are most</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; definitely attributes which are shared and =
negotiated. So, </FONT>
<BR><FONT SIZE=3D2>&gt; are, or are</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; not, codecs part of this? If the goal, in =
general, is for a </FONT>
<BR><FONT SIZE=3D2>&gt; common codec</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to be used in each direction, then I think =
the view has to </FONT>
<BR><FONT SIZE=3D2>&gt; be that they</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; are shared. While we could just reduce the =
set later with </FONT>
<BR><FONT SIZE=3D2>&gt; an additional</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; offer, as Flemming has pointed out, the =
value that the O/A exchange</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; brings to the table is the ability to not =
need to do that.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Actually, it doesn't. With the present O/A =
text, we already </FONT>
<BR><FONT SIZE=3D2>&gt; have to support</FONT>
<BR><FONT SIZE=3D2>&gt; all codecs in the offer until we get the =
answer. Furthermore, </FONT>
<BR><FONT SIZE=3D2>&gt; since the</FONT>
<BR><FONT SIZE=3D2>&gt; offerer has no control over the answer, it is =
quite possible </FONT>
<BR><FONT SIZE=3D2>&gt; that the answer</FONT>
<BR><FONT SIZE=3D2>&gt; contains more than one of the offered codec, =
and hence we are </FONT>
<BR><FONT SIZE=3D2>&gt; back where we</FONT>
<BR><FONT SIZE=3D2>&gt; started. The only way to get down to one codec =
is a new </FONT>
<BR><FONT SIZE=3D2>&gt; offer/answer exchange.</FONT>
<BR><FONT SIZE=3D2>&gt; Thus, the above can at most be seen as a =
possible optimization between</FONT>
<BR><FONT SIZE=3D2>&gt; offerers and answerers that both agree to pick =
just one </FONT>
<BR><FONT SIZE=3D2>&gt; codec. Since it</FONT>
<BR><FONT SIZE=3D2>&gt; doesn't really solve the problem then, we loose =
some </FONT>
<BR><FONT SIZE=3D2>&gt; functionality in the</FONT>
<BR><FONT SIZE=3D2>&gt; process, and we depart with what at least a lot =
of people had </FONT>
<BR><FONT SIZE=3D2>&gt; understood</FONT>
<BR><FONT SIZE=3D2>&gt; (based on your earlier e-mail), we should stick =
to </FONT>
<BR><FONT SIZE=3D2>&gt; yesterday's consensus.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- Flemming</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Bert Culpepper wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; &gt; Also note, that you can achieve =
almost the same by </FONT>
<BR><FONT SIZE=3D2>&gt; issuing an updated</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; &gt; offer/answer with the the single =
codec, if that's what you want.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; I shouldn't be in so much of a hurry =
to jump in...You're </FONT>
<BR><FONT SIZE=3D2>&gt; right, with an</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; extra message exchange I can satisfy =
both goals.&nbsp; This </FONT>
<BR><FONT SIZE=3D2>&gt; will work for me.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Bert</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
72 Eagle Rock Avenue</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; East Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX: (973) =
952-5050</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; PH:&nbsp; (973) 952-5000</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; Flemming Andreasen</FONT>
<BR><FONT SIZE=3D2>&gt; Cisco Systems</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; mmusic mailing list</FONT>
<BR><FONT SIZE=3D2>&gt; mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/mmusic" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/mmusic</A></FON=
T>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From mmusic-admin@ietf.org  Fri Feb 15 12:52:18 2002
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To: Sanjoy Sen <sanjoy@nortelnetworks.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Bert Culpepper <bert.culpepper@intervoice-brite.com>,
        "'Roman Shpount'" <roman@atelo.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Re: Using payload types not specified in answer in  
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Sanjoy Sen wrote:

>
>
> Doesn't the new consensus mean that we're back to the original
> requirement that the client (offerer) needs to load multiple DSP's and
> be prepared to receive on any of them (at least until the second
> offer/answer is made)?

Actually, we never departed from that.


> Am I missing something? We should have a mechanism to avoid this for
> lesser clients that can only use one DSP resource at a time. What
> happened to the a=pickone attribute?

There is basically no way of doing this in a backwards compatible way.
You can come fairly close with the current text, additional offer/answer
exchanges and/or new attributes, but at the end of the day, we can't
change the way SDP works. The level of expressiveness you are looking
for is provided by SDPng.

-- Flemming

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 13:18:34 2002
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From: Roman Shpount <roman@atelo.com>
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Subject: [MMUSIC] Re: Using payload types not specified in answer in   offer/answer
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Roman Shpount wrote:

> I would definitely agree with Jonathan in his point of view about codec
> negotiation if we had a luxury of defining a new standard. Unfortunately 
> what is being done here (at least from my point of view) is just an 
> attempt to ensure interoperability of existing applications. My 
> suggestion for this standard would be something along the lines:
>
>   Offerer MUST accept all the codecs from the original offer. Answerer
>   SHOULD NOT send using any codecs not specified in the answer.
>

Flemming Andreasen wrote:

> "SHOULD NOT" is not to be taken lightly, and hence this effectively says
> that you can only use the codecs in the answer. As I indicated in 
> another e-mail to Jonathan, there's little to be gained from this 
> restriction, but things to be lost, and hence we should continue with 
> the practice of allowing the answerer to send with offered codecs not
> contained in the answer.

There are two problems that I have with answerers sending with codecs not in the answer:
1. Interoperability with original RFC2543
2. Interoperability with current 1 of N codecs selection

Let me explain the second one. The offerer is capable of sending and receiving G.711, G.723 and telephone-events with only G.711 or G.723 supported at the same time. The answerer is capable of sending and receiving G.711 and G.723, but can only send telephone-event. The following message exchange takes place:

Initial offer:

   v=0
   o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
   s=
   c=IN IP4 host.anywhere.com
   t=0 0
   m=audio 6400 RTP/AVP 0 4 110
   a=rtpmap:0 PCMU/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:110 telephone-events 
   a=inactive

Initial answer:

   v=0
   o=bob 2890844730 2890844731 IN IP4 host.example.com
   s=
   c=IN IP4 host.example.com
   t=0 0
   m=audio 6400 RTP/AVP 0 4
   a=rtpmap:0 PCMU/8000
   a=rtpmap:4 G723/8000
   a=inactive

When Alice generates updated offer it has no way of knowing that Bob intends to use telephon
d offer:

   v=0
   o=alice 2890844526 2890844527 IN IP4 host.anywhere.com
   s=
   c=IN IP4 host.example.com
   t=0 0
   m=audio 6400 RTP/AVP 4
   a=rtpmap:4 G723/8000
   a=sendrecv

Final answer:

   v=0
   o=bob 2890844730 2890844732 IN IP4 host.example.com
   s=
   c=IN IP4 host.example.com
   t=0 0
   m=audio 6400 RTP/AVP 4
   a=rtpmap:4 G723/8000
   a=sendrecv

Thus after this exchange telephone-events are not used by either party. Thus I am not sure that we achieve anything by allowing the answerer to send with payloads not included in the answer. 

The solution for the problems that we are trying to solve (assymetric use of codecs) is in either SDPng or expanding SDP. I am convinced that expanding SDP can be done in backwards-compatible way (for instance by adding a direction attribute on the per codec basis), but the discussion of this is probably outside the topic of this thread.
  
____________________________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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Date: Fri, 15 Feb 2002 13:34:21 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Flemming Andreasen <fandreas@cisco.com>
CC: Sanjoy Sen <sanjoy@nortelnetworks.com>,
        Bert Culpepper <bert.culpepper@intervoice-brite.com>,
        "'Roman Shpount'" <roman@atelo.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Re: Using payload types not specified in answer in  
 offer/answer
References: <3C6D49C9.DD75C57C@cisco.com>
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Flemming Andreasen wrote:
> 

> > Am I missing something? We should have a mechanism to avoid this for
> > lesser clients that can only use one DSP resource at a time. What
> > happened to the a=pickone attribute?
> 
> There is basically no way of doing this in a backwards compatible way.
> You can come fairly close with the current text, additional offer/answer
> exchanges and/or new attributes, but at the end of the day, we can't
> change the way SDP works. The level of expressiveness you are looking
> for is provided by SDPng.

Or, with another O/A exchange to reduce to a single codec. The a=pickone
never went forward, that is not the issue here as Flemming indicates.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Fri Feb 15 14:21:48 2002
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Roman Shpount wrote:

> Roman Shpount wrote:
>
> > I would definitely agree with Jonathan in his point of view about codec
> > negotiation if we had a luxury of defining a new standard. Unfortunately
> > what is being done here (at least from my point of view) is just an
> > attempt to ensure interoperability of existing applications. My
> > suggestion for this standard would be something along the lines:
> >
> >   Offerer MUST accept all the codecs from the original offer. Answerer
> >   SHOULD NOT send using any codecs not specified in the answer.
> >
>
> Flemming Andreasen wrote:
>
> > "SHOULD NOT" is not to be taken lightly, and hence this effectively says
> > that you can only use the codecs in the answer. As I indicated in
> > another e-mail to Jonathan, there's little to be gained from this
> > restriction, but things to be lost, and hence we should continue with
> > the practice of allowing the answerer to send with offered codecs not
> > contained in the answer.
>
> There are two problems that I have with answerers sending with codecs not in the answer:
> 1. Interoperability with original RFC2543

Not sure this is a real issue, as RFC 2543 was not entirely clear on this and, as indicated earlier, the understanding a lot of people had is to allow the answerer to send with any offered codec, regardless of the answer.


>
> 2. Interoperability with current 1 of N codecs selection

>
> Let me explain the second one. The offerer is capable of sending and receiving G.711, G.723 and telephone-events with only G.711 or G.723 supported at the same time. The answerer is capable of sending and receiving G.711 and G.723, but can only send telephone-event. The following message exchange takes place:
>
> Initial offer:
>
>    v=0
>    o=alice 2890844526 2890844526 IN IP4 host.anywhere.com
>    s=
>    c=IN IP4 host.anywhere.com
>    t=0 0
>    m=audio 6400 RTP/AVP 0 4 110
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:4 G723/8000
>    a=rtpmap:110 telephone-events
>    a=inactive
>
> Initial answer:
>
>    v=0
>    o=bob 2890844730 2890844731 IN IP4 host.example.com
>    s=
>    c=IN IP4 host.example.com
>    t=0 0
>    m=audio 6400 RTP/AVP 0 4
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:4 G723/8000
>    a=inactive
>
> When Alice generates updated offer it has no way of knowing that Bob intends to use telephone-events, so it does not include it in updated offer:
>
>    v=0
>    o=alice 2890844526 2890844527 IN IP4 host.anywhere.com
>    s=
>    c=IN IP4 host.example.com
>    t=0 0
>    m=audio 6400 RTP/AVP 4
>    a=rtpmap:4 G723/8000
>    a=sendrecv
>
> Final answer:
>
>    v=0
>    o=bob 2890844730 2890844732 IN IP4 host.example.com
>    s=
>    c=IN IP4 host.example.com
>    t=0 0
>    m=audio 6400 RTP/AVP 4
>    a=rtpmap:4 G723/8000
>    a=sendrecv
>
> Thus after this exchange telephone-events are not used by either party. Thus I am not sure that we achieve anything by allowing the answerer to send with payloads not included in the answer.

Well, nothing prevents Alice from including telephone-events in the offer again, especially if she knows that Bob could actually still use it, so this really doesn't show much. Also, if you didn't use the "inactive" offer/answer intially, but instead did "sendrecv", it would still work.


>
> The solution for the problems that we are trying to solve (assymetric use of codecs)

I'm not trying to solve that problem, as it would entail much more than we have on the table here. The question is simply which of two interpretations/modes of operation to go with. I've provided arguments on this in several e-mails already.



> is in either SDPng or expanding SDP. I am convinced that expanding SDP can be done in backwards-compatible way (for instance by adding a direction attribute on the per codec basis), but the discussion of this is probably outside the topic of this thread.

I agree it's a separate discussion :-)

-- Flemming


--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 14:58:30 2002
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Roman Shpount wrote:

> There are two problems that I have with answerers sending with codecs
> not in the answer:
> 1. Interoperability with original RFC2543

Flemming Andreasen wrote:

> Not sure this is a real issue, as RFC 2543 was not entirely clear on
> this and, as indicated earlier, the understanding a lot of people had is
> to allow the answerer to send with any offered codec, regardless of the
> answer.

The problem is that there are also enough people that interpreted RFC 2543 the other way (not accepting codecs not specified in the answer). So, in order to interoperate with products from those people the answerer should not send media with codecs not included in the answer.  
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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From mmusic-admin@ietf.org  Fri Feb 15 16:00:01 2002
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Roman Shpount wrote:

> Roman Shpount wrote:
>
> > There are two problems that I have with answerers sending with codecs
> > not in the answer:
> > 1. Interoperability with original RFC2543
>
> Flemming Andreasen wrote:
>
> > Not sure this is a real issue, as RFC 2543 was not entirely clear on
> > this and, as indicated earlier, the understanding a lot of people had is
> > to allow the answerer to send with any offered codec, regardless of the
> > answer.
>
> The problem is that there are also enough people that interpreted RFC 2543 the other way (not accepting codecs not specified in the answer). So, in order to interoperate with products from those people the answerer should not send media with codecs not included in the answer.

We obviously can't support both interpretations, however could we put in a note about it being a potential problem for some older implementations and hence at least alert people to it ?

-- Flemming


>
> ___________________________________
> Roman Shpount, VP of Technology
> aTelo, Inc.
> 2107 Wilson Blvd., Suite 301
> Arlington, VA 22201
> PCN: 703-621-4040 for voice / fax
> www.atelo.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 16:26:03 2002
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Flemming Andreasen wrote:

> 
> Roman Shpount wrote:
> 
> > Roman Shpount wrote:
> >
> > > There are two problems that I have with answerers sending 
> with codecs
> > > not in the answer:
> > > 1. Interoperability with original RFC2543
> >
> > Flemming Andreasen wrote:
> >
> > > Not sure this is a real issue, as RFC 2543 was not 
> entirely clear on
> > > this and, as indicated earlier, the understanding a lot 
> of people had is
> > > to allow the answerer to send with any offered codec, 
> regardless of the
> > > answer.
> >
> > The problem is that there are also enough people that 
> interpreted RFC 2543 the other way (not accepting codecs not 
> specified in the answer). So, in order to interoperate with 
> products from those people the answerer should not send media 
> with codecs not included in the answer.

I had the same understanding as Flemming describes (it solves 
clipping issues and supports sending early media).  However, as 
I said earlier I'd rather see positive indications of what a 
sender will send (this doesn't affect clipping/early media 
capabilities).  But again, this is not what I got from reading 
rfc2543 and rfc2327.  I think the interpretation you describe 
is incorrect (although I'd prefer your interpretation).

> 
> We obviously can't support both interpretations, however 
> could we put in a note about it being a potential problem for 
> some older implementations and hence at least alert people to it ?
> 

I'd like to see some mention of it written down someplace.  That 
way I wouldn't have to argue my interpretation, I could point 
to a I-D/rfc :)

Regards,
Bert

> -- Flemming
> 
> 
> >
> > ___________________________________
> > Roman Shpount, VP of Technology
> > aTelo, Inc.
> > 2107 Wilson Blvd., Suite 301
> > Arlington, VA 22201
> > PCN: 703-621-4040 for voice / fax
> > www.atelo.com
> 
> --
> Flemming Andreasen
> Cisco Systems
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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From mmusic-admin@ietf.org  Fri Feb 15 16:33:03 2002
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Subject: [MMUSIC] summarizing discussion on codec placement in answer
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Folks,

I wanted to take a moment to summarize where we were in regards to our
long thread on interpretation of the codecs in the answer. The purpose
of the summarization is to put all the points on the table, and
hopefully invite some further comment from folks outside the few thread
participants in the few days which remain.

ISSUE:

A sends an offer to B. That offer, for example, is for a single sendrecv
media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
and lists codecs 1 and 2. Now, the question is this: can B still send to
A using codec 3? Related to that, can B add codecs in the answer, which
A can then send with?

Arguments for YES:

1. A needs to be prepared to receive with codec 3 once it sends the
offer anyway.
2. If A wants to restrict the set based on the answer, it can always
generate a new offer with a reduced set of codecs.
3. SDP has always been interpreted as indicating your own capabilities,
and is therefore an "advertisement" that sticks until updated.
4. There are useful cases for adding codecs in the answer, particularly,
support for rfc2833. Thus, lets say A is an end user which can send rfc
2833, but not receive it. So, when it sends an INVITE, it doesn't
include rfc2833 in the INVITE. The server that responds is a media
server, which supports rfc2833. So, it includes it in the answer. Now, A
can actually send with it during the middle of the session, and it will
never receive it (since it didn't advertise it). This provides a useful
function that would otherwise require FID.

Arguments for NO:

1. The purpose of the O/A model is to agree on shared views of certain
session parameters, such as the directionality (send/recv/sendrecv).
Codecs fit into that category as well, since you are already required to
indicate, in the answer, a set that partially overlaps with the offer.
Furthermore, the goal with codecs is to have the same in both
directions, generally, so that this would speak to them being a shared
state that is negotiated.

2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If
you go for YES, there may be interop issues. The converse is not true,
of course.

3. It is very useful to narrow down the codec sets in the answer, and we
should not always require an extra exchange of O/A just to do that.



I took a private poll amongst some implementors. The initial results
showed that most assumed the YES case. Some more mails since then
indicate its a bit more split, although I think somewhat in favor of YES
still.

I am somewhat on the fence; I think NO is more correct based on
offer/answer, although YES would be the SDP purist view. What I really
want is more opinions on this one....

THanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Fri Feb 15 16:42:05 2002
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Roman Shpount wrote:

> There are two problems that I have with answerers sending with codecs
> not in the answer:
> 1. Interoperability with original RFC2543

Flemming Andreasen wrote:

> Not sure this is a real issue, as RFC 2543 was not entirely clear on
> this and, as indicated earlier, the understanding a lot of people had is
> to allow the answerer to send with any offered codec, regardless of the
> answer.

Roman Shpount wrote:

> The problem is that there are also enough people that interpreted RFC
> 2543 the other way (not accepting codecs not specified in the answer).
> So, in order to interoperate with products from those people the 
> answerer should not send media with codecs not included in the answer.

Flemming Andreasen wrote:

> We obviously can't support both interpretations, however could we put in
> a note about it being a potential problem for some older implementations
> and hence at least alert people to it ?

Why can't we? If a device accepts everything in its offer and sends only with codecs specified in the answer it will interoperate with devices that are implemented according to either interpretation. This is a general "accept as much as possible, generate only standard compliant" type principal. This is why I proposed "SHOULD NOT send with codecs not specified in answer" in the standard.
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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Roman Shpount wrote:

>
> Flemming Andreasen wrote:
>
> > We obviously can't support both interpretations, however could we put in
> > a note about it being a potential problem for some older implementations
> > and hence at least alert people to it ?
>
> Why can't we? If a device accepts everything in its offer and sends only with codecs specified in the answer it will interoperate with devices that are implemented according to either interpretation. This is a general "accept as much as possible, generate only standard compliant" type principal. This is why I proposed "SHOULD NOT send with codecs not specified in answer" in the standard.

Well, that would be picking one interpretation - not supporting both.

-- Flemming


>
> ___________________________________
> Roman Shpount, VP of Technology
> aTelo, Inc.
> 2107 Wilson Blvd., Suite 301
> Arlington, VA 22201
> PCN: 703-621-4040 for voice / fax
> www.atelo.com

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 16:59:24 2002
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Jonathan Rosenberg wrote:

>
> 2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If
> you go for YES, there may be interop issues. The converse is not true,
> of course.

On the contrary. Consider the offerer sending codec 1, 2, 3 based on the
"NO" interpretation, and the answerer sending back codec 1, 2 based on the
"YES" interpretation. Regardless of which interpretation you pick, you can
get into trouble when trying to interoperate with the other.

-- Flemming

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Fri Feb 15 17:29:37 2002
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Subject: RE: [MMUSIC] summarizing discussion on codec placement in answer
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I think your YES points are currently allowed by rfc2543 and rfc2327.
However, I'd like the changes as suggested by your NO points.  But I see
these as changes and likely to require changes in some implementations.
(Again, I'd like to deprecate sending codecs not "declared" by the sender.)

Regards,
Bert

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org]On Behalf Of
> Jonathan Rosenberg
> Sent: Friday, February 15, 2002 4:32 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] summarizing discussion on codec placement in answer
>
>
> Folks,
>
> I wanted to take a moment to summarize where we were in regards to our
> long thread on interpretation of the codecs in the answer. The purpose
> of the summarization is to put all the points on the table, and
> hopefully invite some further comment from folks outside the
> few thread
> participants in the few days which remain.
>
> ISSUE:
>
> A sends an offer to B. That offer, for example, is for a
> single sendrecv
> media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> and lists codecs 1 and 2. Now, the question is this: can B
> still send to
> A using codec 3?
> Related to that, can B add codecs in the
> answer, which
> A can then send with?
>
> Arguments for YES:
>
> 1. A needs to be prepared to receive with codec 3 once it sends the
> offer anyway.
> 2. If A wants to restrict the set based on the answer, it can always
> generate a new offer with a reduced set of codecs.
> 3. SDP has always been interpreted as indicating your own
> capabilities,
> and is therefore an "advertisement" that sticks until updated.
> 4. There are useful cases for adding codecs in the answer,
> particularly,
> support for rfc2833. Thus, lets say A is an end user which
> can send rfc
> 2833, but not receive it. So, when it sends an INVITE, it doesn't
> include rfc2833 in the INVITE. The server that responds is a media
> server, which supports rfc2833. So, it includes it in the
> answer. Now, A
> can actually send with it during the middle of the session,
> and it will
> never receive it (since it didn't advertise it). This
> provides a useful
> function that would otherwise require FID.
>
> Arguments for NO:
>
> 1. The purpose of the O/A model is to agree on shared views of certain
> session parameters, such as the directionality (send/recv/sendrecv).
> Codecs fit into that category as well, since you are already
> required to
> indicate, in the answer, a set that partially overlaps with the offer.
> Furthermore, the goal with codecs is to have the same in both
> directions, generally, so that this would speak to them being a shared
> state that is negotiated.
>
> 2. Many people have interpreted the SDP/bis/rfc2543 drafts
> this way. If
> you go for YES, there may be interop issues. The converse is not true,
> of course.
>
> 3. It is very useful to narrow down the codec sets in the
> answer, and we
> should not always require an extra exchange of O/A just to do that.
>
>
>
> I took a private poll amongst some implementors. The initial results
> showed that most assumed the YES case. Some more mails since then
> indicate its a bit more split, although I think somewhat in
> favor of YES
> still.
>
> I am somewhat on the fence; I think NO is more correct based on
> offer/answer, although YES would be the SDP purist view. What I really
> want is more opinions on this one....
>
> THanks,
> Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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From: Takumi Ohba <ohba.takumi@lab.ntt.co.jp>
Subject: Re: [MMUSIC] Answerers alternative selection of m lines(one
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Hi,

At 10:17 02/02/15 -0500, Jonathan Rosenberg wrote:
> You can always send an updated offer with the v4 stream disabled. In
> fact, you can avoid fid altogether, and send the offer with both v4 and
> v6, and once you get the answer, send an updated offer with just one of
> them immediately (if needed).

Establishing two RTP sessions is not the offerer's initial intention.
Sending an updated offer will cause increasing setup delay of the call that 
matches to the offerer's intention.
In addition, I'm not sure that all implementations of dual stack can handle 
both V4 and V6 at one time.

Is there a mechanism that has been already discussed in this WG and be able 
to show the offerer's intention for the answerer to select only one of multi
ple (or at least two) choices?



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Takumi Ohba wrote:
> 
> Hi,
> 
> At 10:17 02/02/15 -0500, Jonathan Rosenberg wrote:
> > You can always send an updated offer with the v4 stream disabled. In
> > fact, you can avoid fid altogether, and send the offer with both v4
> and
> > v6, and once you get the answer, send an updated offer with just one
> of
> > them immediately (if needed).
> 
> Establishing two RTP sessions is not the offerer's initial intention.
> Sending an updated offer will cause increasing setup delay of the call
> that
> matches to the offerer's intention.

No; the call can proceed while the second pass occurs. I agree its not
ideal, but a lot of problems can be solved by multi-pass offer-answers
if needed.

> In addition, I'm not sure that all implementations of dual stack can
> handle
> both V4 and V6 at one time.

There is no need to actually support both at the same time, only two
advertise two of them, and then pick one once the answer comes.

> 
> Is there a mechanism that has been already discussed in this WG and be
> able
> to show the offerer's intention for the answerer to select only one of
> multi
> ple (or at least two) choices?

Not that I am aware of. Doesn't mean its not a good idea; we've
discussed this kind of thing in the context of nat, where a caller can
use stun to obtain a public address. THen, it sends its invite with both
the public and private addresses, and wants the callee to pick just one.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From mmusic-admin@ietf.org  Mon Feb 18 04:17:28 2002
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Date: Mon, 18 Feb 2002 01:11:20 -0800 (PST)
From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] RTSP: Agenda for Feb 20 telecon
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Hi all,

Here's the agenda for this Wednesday, 9AM PST (-8 GMT):
http://rtsp.org/2002/telecon/agenda20Feb.html

It's also available below.

Rob

                           Rtsp Tc Agenda20 Feb2002
     _________________________________________________________________

    1. [1]Last Telecon

    1. [2]Review Action Items/New Issues
    2. [3]Resolved last telecon

     [4]This Telecon

                                 Last Telecon

   Minutes: [5]http://rtsp.org/2002/telecon/minutes06Feb.html

Review Action Items/New Issues

     * ACTION ITEM (RL/HS): Submit new draft of RTSP before February 22
       deadline
     * ACTION ITEM: Call for authors on MUTE proposed. AN is tentatively
       signed up

Resolved last telecon

     * [6]http://rtsp.org/bug477423 - Unclearities regarding pause with
       range
          + RESOLVED: go with Magnus's proposal (see [7]Magnus' proposal)
     * [8]http://rtsp.org/bug518997 - Mixing Aggregate aand Non-aggregate
       Control
          + see [9]the email thread
          + Point a) RESOLVED
          + First point b) RESOLVED
          + Second point b) MUTE and UNMUTE
               o RESOLVED: separate draft
          + Point C) (same as [10]http://rtsp.org/bug477400)
               o RESOLVED: PING writeup will be done as part of
                  next RTSP draft...continue discussion on mailing list
               o RESOLVED: deprecate PLAY w/o range as ping method

                                 This Telecon

   We'll try to slog through the issues from last time around. We've got
   two big, interrelated topics, for which there are a series of
   subissues. I'd like to come to some sort of consensus on the two
   issues

     * Mixing Aggregate and Non-aggregate Control - see [11]the email
       thread. This is related to several bugs:
          + [12]http://rtsp.org/bug477428 - Priorites between aggregated
            & non cntrl
          + [13]http://rtsp.org/bug483356 - media URL in Play on
            aggregated session
          + [14]http://rtsp.org/bug493270 - PAUSE/Play on single media in
            aggregate
          + [15]http://rtsp.org/bug506412 - RTSP Teardown functionallity
     * Eliminate queued play - see [16]the email thread and
       [17]http://rtsp.org/bug508211 . This relates to several issues
          + [18]http://rtsp.org/bug477424 - How to queue timed plays?
          + [19]http://rtsp.org/bug477432 - RTP info and queued plays
          + [20]http://rtsp.org/bug493265 - Pause - Play behaviour
            clarification
          + [21]http://rtsp.org/bug507882 - RTP timestamps when playing
            after pause

References

   1. http://rtsp.org/2002/telecon/agenda20Feb.html?template=plain#line3
   2. http://rtsp.org/2002/telecon/agenda20Feb.html?template=plain#line7
   3. http://rtsp.org/2002/telecon/agenda20Feb.html?template=plain#line12
   4. http://rtsp.org/2002/telecon/agenda20Feb.html?template=plain#line26
   5. http://rtsp.org/2002/telecon/minutes06Feb.html
   6. http://rtsp.org/bug477423
   7. http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00148.html
   8. http://rtsp.org/bug518997
   9. http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00115.html
  10. http://rtsp.org/bug477400
  11. http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00115.html
  12. http://rtsp.org/bug477428
  13. http://rtsp.org/bug483356
  14. http://rtsp.org/bug493270
  15. http://rtsp.org/bug506412
  16. http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00105.html
  17. http://rtsp.org/bug508211
  18. http://rtsp.org/bug477424
  19. http://rtsp.org/bug477432
  20. http://rtsp.org/bug493265
  21. http://rtsp.org/bug507882



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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		: Support for IPv6 in SDP
	Author(s)	: S. Olson
	Filename	: draft-ietf-mmusic-sdp-ipv6-02.txt
	Pages		: 5
	Date		: 15-Feb-02
	
This document describes the use of IPv6 addresses [1] in conjunction 
with the Session Description Protocol (SDP) [2]. Specifically, this 
document clarifies existing text in SDP with regards to the syntax of 
IPv6 addresses.

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From mmusic-admin@ietf.org  Mon Feb 18 17:03:32 2002
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Date: Mon, 18 Feb 2002 14:01:54 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] RADPLAY extentions for SDP
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Hi all,

Just wanted to let everyone know about a new draft defining extensions
to SDP which help out in the process of RTP archiving, delivery and
playback.  Here's the abstract:

    This document defines RTP Archiving, Delivery, and Playback
    (RADPLAY) extensions for the Session Description Protocol
    (SDP, [RFC 2327]).  These extensions provide a standard means
    of archiving and delivering RTP streams via HTTP, using SDP
    and the commonly used 'rtpdump' file format.  As a result of
    its design, it also provides a way of delivering generic RTP
    streams over HTTP, thus making it a suitable replacement for
    approaches which rely on network-unfriendly mechanisms such
    as 'HTTP tunneling'.

Give it a look, and lemme know what you think.  I'd like to consider this
as a submission to move down the standards track, so if people generally
think this is a good idea, then we can work out what the strategy is for
moving this work forward.

Thanks
Rob

---------- Forwarded message ----------
Date: Mon, 18 Feb 2002 11:10:46 -0500
From: "Internet-Drafts@ietf.org" <Internet-Drafts@ietf.org>
To: IETF-Announce:  ;
Subject: I-D ACTION:draft-lanphier-radplay-00.txt

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


	Title		: RTP Archiving, Delivery, and Playback (RADPLAY)
                          Extensions for SDP
	Author(s)	: R. Lanphier
	Filename	: draft-lanphier-radplay-00.txt
	Pages		: 10
	Date		: 15-Feb-02

This document defines RTP Archiving, Delivery, and Playback
(RADPLAY) extensions for the Session Description Protocol
(SDP, [RFC 2327]).  These extensions provide a standard means
of archiving and delivering RTP streams via HTTP, using SDP
and the commonly used 'rtpdump' file format.  As a result of
its design, it also provides a way of delivering generic RTP
streams over HTTP, thus making it a suitable replacement for
approaches which rely on network-unfriendly mechanisms such
as 'HTTP tunneling'.

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From mmusic-admin@ietf.org  Mon Feb 18 17:24:30 2002
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From: Rob Lanphier <robla@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] RADPLAY comment ... (fwd)
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Forwarded with permission from the author....
---------- Forwarded message ----------
Date: Mon, 18 Feb 2002 13:43:20 -0800
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
To: "robla@real.com" <robla@real.com>
Subject: RADPLAY comment ...


Hi Rob,

        Quick comment on RADPLAY -- I think of all of those bits devoted
to resiliency inside of each payload type (not via one of the generic
resiliency schemes like RFC2733, but the extra codec-specific stuff
that's added for resilience), all going to waste when downloaded via
HTTP via TCP. I guess the packetization I'm working on is extreme in
the regard:

http://www.ietf.org/internet-drafts/draft-ietf-avt-mwpp-midi-rtp-00.txt

	where there's two versions (resilient and non-resilient) with
a large relative bandwidth cost for resiliency (absolute cost isn't
bad, though, about voice codec range).

	Maybe this would all be "outside" RADPLAY -- people would be
expected to convert formats from resilient to non-resilient to optimize
bandwidth as part of the file preparation process -- but maybe there's
a reason why RADPLAY needs to be aware of the issue too ...

							--john lazzaro

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------



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From mmusic-admin@ietf.org  Mon Feb 18 18:16:08 2002
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From: "Sanjoy Sen"<sanjoy@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] summarizing discussion on codec placement in answer
Date: Mon, 18 Feb 2002 17:15:04 -0600
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Jonathan,

   We at Nortel (that includes at least four of us including a codec expert)
support the second option, i.e., "NO". The primary reason is the #3 reason
that you've - we would like to have the possibility of narrowing down to one
codec without the overhead of an extra roundtrip (O/A exchange). Also, its
not clear to some of us how would you negotiate the attributes that're
associated with the media session (e.g., security parameters). Example,
suppose Offerer sends 3 codecs - 2,3,4 and the answerer responds with 3,4
and adds an a=keymgmt-data: attribute for this media, should the offerer be
ready to accept 2 with the security keys applied?

Thanks,
Sanjoy


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, February 15, 2002 3:32 PM
> To: mmusic@ietf.org
> Subject: [MMUSIC] summarizing discussion on codec placement in answer
> 
> 
> Folks,
> 
> I wanted to take a moment to summarize where we were in regards to our
> long thread on interpretation of the codecs in the answer. The purpose
> of the summarization is to put all the points on the table, and
> hopefully invite some further comment from folks outside the 
> few thread
> participants in the few days which remain.
> 
> ISSUE:
> 
> A sends an offer to B. That offer, for example, is for a 
> single sendrecv
> media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> and lists codecs 1 and 2. Now, the question is this: can B 
> still send to
> A using codec 3? Related to that, can B add codecs in the 
> answer, which
> A can then send with?
> 
> Arguments for YES:
> 
> 1. A needs to be prepared to receive with codec 3 once it sends the
> offer anyway.
> 2. If A wants to restrict the set based on the answer, it can always
> generate a new offer with a reduced set of codecs.
> 3. SDP has always been interpreted as indicating your own 
> capabilities,
> and is therefore an "advertisement" that sticks until updated.
> 4. There are useful cases for adding codecs in the answer, 
> particularly,
> support for rfc2833. Thus, lets say A is an end user which 
> can send rfc
> 2833, but not receive it. So, when it sends an INVITE, it doesn't
> include rfc2833 in the INVITE. The server that responds is a media
> server, which supports rfc2833. So, it includes it in the 
> answer. Now, A
> can actually send with it during the middle of the session, 
> and it will
> never receive it (since it didn't advertise it). This 
> provides a useful
> function that would otherwise require FID.
> 
> Arguments for NO:
> 
> 1. The purpose of the O/A model is to agree on shared views of certain
> session parameters, such as the directionality (send/recv/sendrecv).
> Codecs fit into that category as well, since you are already 
> required to
> indicate, in the answer, a set that partially overlaps with the offer.
> Furthermore, the goal with codecs is to have the same in both
> directions, generally, so that this would speak to them being a shared
> state that is negotiated.
> 
> 2. Many people have interpreted the SDP/bis/rfc2543 drafts 
> this way. If
> you go for YES, there may be interop issues. The converse is not true,
> of course.
> 
> 3. It is very useful to narrow down the codec sets in the 
> answer, and we
> should not always require an extra exchange of O/A just to do that.
> 
> 
> 
> I took a private poll amongst some implementors. The initial results
> showed that most assumed the YES case. Some more mails since then
> indicate its a bit more split, although I think somewhat in 
> favor of YES
> still.
> 
> I am somewhat on the fence; I think NO is more correct based on
> offer/answer, although YES would be the SDP purist view. What I really
> want is more opinions on this one....
> 
> THanks,
> Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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<TITLE>RE: [MMUSIC] summarizing discussion on codec placement in =
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<BODY>

<P><FONT SIZE=3D2>Jonathan,</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; We at Nortel (that includes at least =
four of us including a codec expert) support the second option, i.e., =
&quot;NO&quot;. The primary reason is the #3 reason that you've - we =
would like to have the possibility of narrowing down to one codec =
without the overhead of an extra roundtrip (O/A exchange). Also, its =
not clear to some of us how would you negotiate the attributes that're =
associated with the media session (e.g., security parameters). Example, =
suppose Offerer sends 3 codecs - 2,3,4 and the answerer responds with =
3,4 and adds an a=3Dkeymgmt-data: attribute for this media, should the =
offerer be ready to accept 2 with the security keys applied?</FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Sanjoy</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Friday, February 15, 2002 3:32 PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [MMUSIC] summarizing discussion on =
codec placement in answer</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I wanted to take a moment to summarize where we =
were in regards to our</FONT>
<BR><FONT SIZE=3D2>&gt; long thread on interpretation of the codecs in =
the answer. The purpose</FONT>
<BR><FONT SIZE=3D2>&gt; of the summarization is to put all the points =
on the table, and</FONT>
<BR><FONT SIZE=3D2>&gt; hopefully invite some further comment from =
folks outside the </FONT>
<BR><FONT SIZE=3D2>&gt; few thread</FONT>
<BR><FONT SIZE=3D2>&gt; participants in the few days which =
remain.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; ISSUE:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A sends an offer to B. That offer, for example, =
is for a </FONT>
<BR><FONT SIZE=3D2>&gt; single sendrecv</FONT>
<BR><FONT SIZE=3D2>&gt; media stream, which lists codecs 1,2 and 3. The =
answerer, B, responds,</FONT>
<BR><FONT SIZE=3D2>&gt; and lists codecs 1 and 2. Now, the question is =
this: can B </FONT>
<BR><FONT SIZE=3D2>&gt; still send to</FONT>
<BR><FONT SIZE=3D2>&gt; A using codec 3? Related to that, can B add =
codecs in the </FONT>
<BR><FONT SIZE=3D2>&gt; answer, which</FONT>
<BR><FONT SIZE=3D2>&gt; A can then send with?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Arguments for YES:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1. A needs to be prepared to receive with codec =
3 once it sends the</FONT>
<BR><FONT SIZE=3D2>&gt; offer anyway.</FONT>
<BR><FONT SIZE=3D2>&gt; 2. If A wants to restrict the set based on the =
answer, it can always</FONT>
<BR><FONT SIZE=3D2>&gt; generate a new offer with a reduced set of =
codecs.</FONT>
<BR><FONT SIZE=3D2>&gt; 3. SDP has always been interpreted as =
indicating your own </FONT>
<BR><FONT SIZE=3D2>&gt; capabilities,</FONT>
<BR><FONT SIZE=3D2>&gt; and is therefore an &quot;advertisement&quot; =
that sticks until updated.</FONT>
<BR><FONT SIZE=3D2>&gt; 4. There are useful cases for adding codecs in =
the answer, </FONT>
<BR><FONT SIZE=3D2>&gt; particularly,</FONT>
<BR><FONT SIZE=3D2>&gt; support for rfc2833. Thus, lets say A is an end =
user which </FONT>
<BR><FONT SIZE=3D2>&gt; can send rfc</FONT>
<BR><FONT SIZE=3D2>&gt; 2833, but not receive it. So, when it sends an =
INVITE, it doesn't</FONT>
<BR><FONT SIZE=3D2>&gt; include rfc2833 in the INVITE. The server that =
responds is a media</FONT>
<BR><FONT SIZE=3D2>&gt; server, which supports rfc2833. So, it includes =
it in the </FONT>
<BR><FONT SIZE=3D2>&gt; answer. Now, A</FONT>
<BR><FONT SIZE=3D2>&gt; can actually send with it during the middle of =
the session, </FONT>
<BR><FONT SIZE=3D2>&gt; and it will</FONT>
<BR><FONT SIZE=3D2>&gt; never receive it (since it didn't advertise =
it). This </FONT>
<BR><FONT SIZE=3D2>&gt; provides a useful</FONT>
<BR><FONT SIZE=3D2>&gt; function that would otherwise require =
FID.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Arguments for NO:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 1. The purpose of the O/A model is to agree on =
shared views of certain</FONT>
<BR><FONT SIZE=3D2>&gt; session parameters, such as the directionality =
(send/recv/sendrecv).</FONT>
<BR><FONT SIZE=3D2>&gt; Codecs fit into that category as well, since =
you are already </FONT>
<BR><FONT SIZE=3D2>&gt; required to</FONT>
<BR><FONT SIZE=3D2>&gt; indicate, in the answer, a set that partially =
overlaps with the offer.</FONT>
<BR><FONT SIZE=3D2>&gt; Furthermore, the goal with codecs is to have =
the same in both</FONT>
<BR><FONT SIZE=3D2>&gt; directions, generally, so that this would speak =
to them being a shared</FONT>
<BR><FONT SIZE=3D2>&gt; state that is negotiated.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 2. Many people have interpreted the =
SDP/bis/rfc2543 drafts </FONT>
<BR><FONT SIZE=3D2>&gt; this way. If</FONT>
<BR><FONT SIZE=3D2>&gt; you go for YES, there may be interop issues. =
The converse is not true,</FONT>
<BR><FONT SIZE=3D2>&gt; of course.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; 3. It is very useful to narrow down the codec =
sets in the </FONT>
<BR><FONT SIZE=3D2>&gt; answer, and we</FONT>
<BR><FONT SIZE=3D2>&gt; should not always require an extra exchange of =
O/A just to do that.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I took a private poll amongst some =
implementors. The initial results</FONT>
<BR><FONT SIZE=3D2>&gt; showed that most assumed the YES case. Some =
more mails since then</FONT>
<BR><FONT SIZE=3D2>&gt; indicate its a bit more split, although I think =
somewhat in </FONT>
<BR><FONT SIZE=3D2>&gt; favor of YES</FONT>
<BR><FONT SIZE=3D2>&gt; still.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I am somewhat on the fence; I think NO is more =
correct based on</FONT>
<BR><FONT SIZE=3D2>&gt; offer/answer, although YES would be the SDP =
purist view. What I really</FONT>
<BR><FONT SIZE=3D2>&gt; want is more opinions on this one....</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; THanks,</FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
72 Eagle Rock Avenue</FONT>
<BR><FONT SIZE=3D2>&gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; East Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX: (973) =
952-5050</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; PH:&nbsp; (973) 952-5000</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; _______________________________________________<=
/FONT>
<BR><FONT SIZE=3D2>&gt; mmusic mailing list</FONT>
<BR><FONT SIZE=3D2>&gt; mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/mmusic" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/mmusic</A></FON=
T>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From mmusic-admin@ietf.org  Mon Feb 18 19:57:40 2002
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Sanjoy Sen wrote:

> Also, its not clear to some of us how would you negotiate the 
> attributes that're associated with the media session (e.g., security 
> parameters). Example, suppose Offerer sends 3 codecs - 2,3,4 and the 
> answerer responds with 3,4 and adds an a=keymgmt-data: attribute for
> this media, should the offerer be ready to accept 2 with the security 
> keys applied?

I have a related question: How would the same security parameter negotiation work with early media?
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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Message-ID: <000f01c1b8ef$f0b9fda0$81190b0a@huawei.com.cn>
From: "Peili Xu" <xupeili@huawei.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <mmusic@ietf.org>
References: <003c01c1ad1e$91b273a0$db100b0a@huawei.com.cn> <3C5DF33A.783BB466@dynamicsoft.com>
Date: Tue, 19 Feb 2002 10:48:41 +0800
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Subject: [MMUSIC] Re: Question About SDO Offer/Answer draft
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But in bis07 line 2036 it reads:
"Once the UAS has sent or received an answer to the initial offer, it MUST NOT generate subsequent 
offers in any responses to the INVITE. Since only the UAC can send PRACK, this means the a UAS 
based on this specification alone can never generate subsequent offers. "

This means that the UAS cannot start a second offer use 200 and get answer in ACK??

There is a conflict here.


----- Original Message ----- 
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: Peili Xu <xupeili@huawei.com>
Cc: <mmusic@ietf.org>
Sent: Monday, February 04, 2002 10:34 AM
Subject: Re: Question About SDO Offer/Answer draft


> 
> 
> Peili Xu wrote:
> > 
> > In section 4 "Protocol Operatioin":
> > 
> > It says that the agent Must Not generate a new offer if it has received
> > an offer which it has not yet answered or rejected.
> > 
> > But for early media we use 180/PRACK to offer and answer the early media
> > just before
> > 
> > the Invite/200 interaction is completed.
> > 
> > Is this method break the Offer/Answer law?
> 
> No. It works like this:
> 
> 
> INVITE (offer 1) -->
> <-----183 (answer 1)-
> 
> <------200 OK (offer 2)--
> -------ACK (answer 2)---->
> 
> that is, the 200 OK is a new offer, and then SDP is present in the ACK
> with an answer. 
> 
> This is different from the "common" practice today for early media,
> which has the 183 as an anwer, with a 200 OK that repeats the same SDP.
> I believe there will largely be backwards compatibility. If the UAS does
> the "common" thing when talking to an element compliant to bis/o-a, it
> will send SDP in the ACK, but so long as its not different from the
> offer in the INVITE, there won't be any issues. 
> 
> -Jonathan R.
> 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com

_______________________________________________
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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-477421 ] When to send response (noreply@sourceforge.net)
   2. [ rtspspec-Bugs-477423 ] Unclearities regarding pause with range (noreply@sourceforge.net)
   3. [ rtspspec-Bugs-477424 ] How to queue timed plays? (noreply@sourceforge.net)
   4. [ rtspspec-Bugs-477425 ] Inconsistency between timeformats (noreply@sourceforge.net)
   5. [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl (noreply@sourceforge.net)
   6. [ rtspspec-Bugs-477421 ] When to send response (noreply@sourceforge.net)
   7. [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl (noreply@sourceforge.net)
   8. [ rtspspec-Bugs-477425 ] Inconsistency between timeformats (noreply@sourceforge.net)
   9. [ rtspspec-Bugs-477423 ] Unclearities regarding pause with range (noreply@sourceforge.net)
  10. [ rtspspec-Bugs-477427 ] Transport model requirements (noreply@sourceforge.net)
  11. [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl (noreply@sourceforge.net)
  12. [ rtspspec-Bugs-518997 ] Mixing aggregate and non-aggregate (noreply@sourceforge.net)
  13. [ rtspspec-Bugs-513758 ] Proxy-Require clarification (noreply@sourceforge.net)
  14. [ rtspspec-Bugs-448521 ] URLs in Rtp-Info need to be quoted (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 17:10:43 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477421 ] When to send response

Bugs item #477421, was opened at 2001-11-01 23:43
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: When to send response

Initial Comment:
Server can't respond to a bad request if no sequence number is present in the request. Resulting in 
that the client does not know that the server has received it and has issues. Please clarify when a 
server is allowed to send response to bad request.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:10

Message:
Logged In: YES 
user_id=437905

RESOLVED: The Server is allowed to send an immediate 
response with no sequence number. As a general rule the 
server should NOT set sequence numbers. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:01

Message:
Logged In: YES 
user_id=3796

This one is going to be up to the server.  It can send "Connection: close" if the state machine is too 
messed up (though see #492546 for another discussion).  We can't overspecify this, though.

Please suggest text to insert in the specification.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 17:12:15 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477423 ] Unclearities regarding pause with range

Bugs item #477423, was opened at 2001-11-01 23:45
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477423&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Unclearities regarding pause with range

Initial Comment:
How shall the pause point be specifed? Is the only correct way to do it "(start time)- "? The use of 
range with time? It seem to be unecessary as play with range works instead.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:12

Message:
Logged In: YES 
user_id=437905

RESOLVED: Only a single range specifier
(format "value-" where "value" is a time)
is allowed for pause. time="execution time in UTC" may 
also be used, but if multiple range specifiers are 
present, npt/smpte takes precence over time. A new example 
will be provided.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:03

Message:
Logged In: YES 
user_id=3796

Does need clarification in specification, but probably warrents discussion.

Recommendation: end time in range is ignored

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477423&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 17:44:00 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477424 ] How to queue timed plays?

Bugs item #477424, was opened at 2001-11-01 23:47
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477424&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: How to queue timed plays?

Initial Comment:
When you queue items that has a activation time, does that override the order of the events that 
are queued? Play(1) has time A, Play(2) has time B, time A are after time B. 
Assumption on how it work: 
First Play(2) activated at time B, then at time B Play(1) tries to start, if already playing, it 
is queued as a normal untimed event.



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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:44

Message:
Logged In: YES 
user_id=437905

RESOLVED: Propose elimination of queued play to list. 
Separate email necessary. 

RESOLVED: postpone 477424 until queue play is resolved.

ACTION ITEM (RL): submit proposal to eliminate queued play.
ACTION ITEM (MW): continuous sequence number filing.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:04

Message:
Logged In: YES 
user_id=3796

Interpretation is correct.  Do you (or anyone in the working group) have a proposal for text to drop into the 
spec?

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477424&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 17:44:40 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477425 ] Inconsistency between timeformats

Bugs item #477425, was opened at 2001-11-01 23:48
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477425&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Inconsistency between timeformats

Initial Comment:
NPT time in range is allowed to be specifed as "-nptime". No other timeformat allows that, why?

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:44

Message:
Logged In: YES 
user_id=437905

RESOLVED: Extend this to other time formats, and probably 
clarify the semantics of this particular syntax. Should 
start at pause point or the beginning.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-01-03 01:07

Message:
Logged In: YES 
user_id=302620

I belive we need to to consider how a play with a range "-time" after a pause should be treated. This is 
related to #493265. If #493265 alternative 2 is the agreed behaviour for that issue, the -time range could 
be used to restart a paused media with a given end point. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:06

Message:
Logged In: YES 
user_id=3796

We're pretty sure this is a thinko, and we think that this technique should be extended to the other time 
formats.  Anyone in the working group think differerently?

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 07:44

Message:
Logged In: YES 
user_id=3796

It's clear we need to extend this to other time formats, and probably clarify the semantics of this particular 
syntax.  As to "why?", it was probably just an oversight; we focused on npt for most of our work.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477425&group_id=23194


--__--__--

Message: 5
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 17:45:21 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl

Bugs item #477428, was opened at 2001-11-01 23:51
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477428&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Priorites between aggregated & non cntrl

Initial Comment:
A single media stream is playing. Then an aggregated control which include this stream is issued a 
play. How should this be interpreted? I use that it is queue and waits until all single streams under 
that aggregated control finish their play.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:45

Message:
Logged In: YES 
user_id=437905

RESOLVED: Await resolution of queued play. Aggregate 
control issue needs larger discussion.

ACTION ITEM: Write up aggregate URL proposal.

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477428&group_id=23194


--__--__--

Message: 6
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 18:02:58 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477421 ] When to send response

Bugs item #477421, was opened at 2001-11-01 23:43
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: When to send response

Initial Comment:
Server can't respond to a bad request if no sequence number is present in the request. Resulting in 
that the client does not know that the server has received it and has issues. Please clarify when a 
server is allowed to send response to bad request.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:10

Message:
Logged In: YES 
user_id=437905

RESOLVED: The Server is allowed to send an immediate 
response with no sequence number. As a general rule the 
server should NOT set sequence numbers. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:01

Message:
Logged In: YES 
user_id=3796

This one is going to be up to the server.  It can send "Connection: close" if the state machine is too 
messed up (though see #492546 for another discussion).  We can't overspecify this, though.

Please suggest text to insert in the specification.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194


--__--__--

Message: 7
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Tue, 12 Feb 2002 18:03:17 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl

Bugs item #477428, was opened at 2001-11-01 23:51
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477428&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Priorites between aggregated & non cntrl

Initial Comment:
A single media stream is playing. Then an aggregated control which include this stream is issued a 
play. How should this be interpreted? I use that it is queue and waits until all single streams under 
that aggregated control finish their play.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:45

Message:
Logged In: YES 
user_id=437905

RESOLVED: Await resolution of queued play. Aggregate 
control issue needs larger discussion.

ACTION ITEM: Write up aggregate URL proposal.

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477428&group_id=23194


--__--__--

Message: 8
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 14 Feb 2002 14:40:40 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477425 ] Inconsistency between timeformats

Bugs item #477425, was opened at 2001-11-01 23:48
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477425&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Inconsistency between timeformats

Initial Comment:
NPT time in range is allowed to be specifed as "-nptime". No other timeformat allows that, why?

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:44

Message:
Logged In: YES 
user_id=437905

RESOLVED: Extend this to other time formats, and probably 
clarify the semantics of this particular syntax. Should 
start at pause point or the beginning.

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

Comment By: Magnus Westerlund (magwes)
Date: 2002-01-03 01:07

Message:
Logged In: YES 
user_id=302620

I belive we need to to consider how a play with a range "-time" after a pause should be treated. This is 
related to #493265. If #493265 alternative 2 is the agreed behaviour for that issue, the -time range could 
be used to restart a paused media with a given end point. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:06

Message:
Logged In: YES 
user_id=3796

We're pretty sure this is a thinko, and we think that this technique should be extended to the other time 
formats.  Anyone in the working group think differerently?

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 07:44

Message:
Logged In: YES 
user_id=3796

It's clear we need to extend this to other time formats, and probably clarify the semantics of this particular 
syntax.  As to "why?", it was probably just an oversight; we focused on npt for most of our work.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477425&group_id=23194


--__--__--

Message: 9
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 14 Feb 2002 14:41:27 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477423 ] Unclearities regarding pause with range

Bugs item #477423, was opened at 2001-11-01 23:45
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477423&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Unclearities regarding pause with range

Initial Comment:
How shall the pause point be specifed? Is the only correct way to do it "(start time)- "? The use of 
range with time? It seem to be unecessary as play with range works instead.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:12

Message:
Logged In: YES 
user_id=437905

RESOLVED: Only a single range specifier
(format "value-" where "value" is a time)
is allowed for pause. time="execution time in UTC" may 
also be used, but if multiple range specifiers are 
present, npt/smpte takes precence over time. A new example 
will be provided.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:03

Message:
Logged In: YES 
user_id=3796

Does need clarification in specification, but probably warrents discussion.

Recommendation: end time in range is ignored

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477423&group_id=23194


--__--__--

Message: 10
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 14 Feb 2002 14:40:59 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477427 ] Transport model requirements

Bugs item #477427, was opened at 2001-11-01 23:49
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477427&group_id=23194

Category: None
>Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Transport model requirements

Initial Comment:
Which connection models are required. Is non persitent TCP connections a MUST support, or only 
RECOMMENDED?

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:07

Message:
Logged In: YES 
user_id=3796

Good question.  This is related to #492546

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477427&group_id=23194


--__--__--

Message: 11
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 14 Feb 2002 14:42:13 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477428 ] Priorites between aggregated & non cntrl

Bugs item #477428, was opened at 2001-11-01 23:51
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477428&group_id=23194

Category: None
>Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Priorites between aggregated & non cntrl

Initial Comment:
A single media stream is playing. Then an aggregated control which include this stream is issued a 
play. How should this be interpreted? I use that it is queue and waits until all single streams under 
that aggregated control finish their play.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:45

Message:
Logged In: YES 
user_id=437905

RESOLVED: Await resolution of queued play. Aggregate 
control issue needs larger discussion.

ACTION ITEM: Write up aggregate URL proposal.

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

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--__--__--

Message: 12
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 18 Feb 2002 00:31:06 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-518997 ] Mixing aggregate and non-aggregate

Bugs item #518997, was opened at 2002-02-18 00:30
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=518997&group_id=23194

Category: General
Group: Fix - Discussed/Resolved
Status: Open
Resolution: None
Priority: 5
Submitted By: Rob Lanphier (robla)
Assigned to: Rob Lanphier (robla)
Summary: Mixing aggregate and non-aggregate

Initial Comment:
This is a tracking issue for the email here:
http://www1.ietf.org/mail-archive/working-groups/mmusic/current/msg00115.html

 * Point a) RESOLVED
 * First point b) RESOLVED
 * Second point b) MUTE and UNMUTE
   * RESOLVED: separate draft
   * ACTION ITEM: Call for authors on MUTE proposed. AN
is tentatively signed up
 * Point C) (same as http://rtsp.org/bug477400)
   * RESOLVED: PING writeup will be done as part of
next RTSP draft...continue discussion on mailing list
   * RESOLVED: deprecate PLAY w/o range as ping method

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--__--__--

Message: 13
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 18 Feb 2002 15:33:25 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-513758 ] Proxy-Require clarification

Bugs item #513758, was opened at 2002-02-06 06:16
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=513758&group_id=23194

Category: General
>Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Rob Lanphier (robla)
Summary: Proxy-Require clarification

Initial Comment:
Today the proxy-require method says that the server should treat the proxy-require header 
identically as require. I don't understand how that mechanism work for paramters/functionallity that 
only the proxies should answer to. To avoid a negative acknowledgment the server must be aware 
of these parameters to not place them in the Unsupported header. I can't see that any proxy can 
modify the response as it will not know if the response comes from a proxy or a server. 

Can someone please clarify this?  

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--__--__--

Message: 14
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 18 Feb 2002 16:36:49 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-448521 ] URLs in Rtp-Info need to be quoted

Bugs item #448521, was opened at 2001-08-06 13:01
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=448521&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Rob Lanphier (robla)
Summary: URLs in Rtp-Info need to be quoted

Initial Comment:
The Rtp-Info header contains URLs, but those URLs are 
not quoted, causing problems when the URLs contain 
semi-colon, equal signs and comma characters.

The best way to quote URLs are by using double-
quotes.  Using angle-brackets has also been proposed, 
but using double-quotes is more likely to be 
compatbile with existing implementations, because 
other RTSP/HTTP headers use double-quote characters 
in similar situations.
 

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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-18 16:36

Message:
Logged In: YES 
user_id=3796

This has been fixed in the version in CVS

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 14:20

Message:
Logged In: YES 
user_id=437905

RESOLUTION: (Jan 9, 2002 telecon) 
Proposed solution was accepted.  Alternative use of 
semicolon and comma were considered  by classified 
as "dangerous characters" ans would break existing 
implementations.


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

Comment By: Henning Schulzrinne (hgschulz)
Date: 2002-01-09 09:20

Message:
Logged In: YES 
user_id=115926

URL escaping (%xx) doesn't work since you lose the
distinction between syntactically significant semicolons
(etc.) and just characters that were escaped in the original
URL.

Only comma and semicolon need to be escaped, since equal
signs are not delimiters.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 09:26

Message:
Logged In: YES 
user_id=3796

I should probably clarify the comments made yesterday.  This is a genuine issue, but
it needs to be weighed against backward-compatibility needs.  Would like input from
implementors on what the best solution is.

We would propose the following: 
*  We need to define the problem characters.  ";" and "=" are two, there may be others
*  URLs MUST not be quoted when no problem characters are present in the URL.  This will help maintain 
backwards-compatibilty with current implementations
*  URLs MUST be quoted when problem characters are present



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

Comment By: Rob Lanphier (robla)
Date: 2001-12-12 14:14

Message:
Logged In: YES 
user_id=3796

add quotes only when sending problem characters (";", "=", ",", need to think further)

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

Comment By: Rob Lanphier (robla)
Date: 2001-08-07 07:21

Message:
Logged In: YES 
user_id=3796

This is primarily to test if mail to confctrl is working

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=448521&group_id=23194




End of Rtspspec-bugs Digest


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From mmusic-admin@ietf.org  Tue Feb 19 05:18:25 2002
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From: "Even, Roni" <roni.even@polycom.co.il>
To: mmusic@ietf.org
Subject: RE: [MMUSIC] summarizing discussion on codec placement in answer
Date: Tue, 19 Feb 2002 12:17:31 +0200
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Hi,
I would vote for the No. My reasons are that I would like to distinguish
between symmetric codecs and non symmetric codecs. I would use the sendrecv
if I want symmetric codecs and use recvonly and sendonly if I want non
symmetric codecs. The reasons for the different modes is that I may be an
MCU that needs the video streams to be symmetric or a terminal that cannot
decode one video algorithm and encode another. Point 3 in the no is very
important.

Roni Even
Polycom


-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Friday, February 15, 2002 11:32 PM
To: mmusic@ietf.org
Subject: [MMUSIC] summarizing discussion on codec placement in answer


Folks,

I wanted to take a moment to summarize where we were in regards to our
long thread on interpretation of the codecs in the answer. The purpose
of the summarization is to put all the points on the table, and
hopefully invite some further comment from folks outside the few thread
participants in the few days which remain.

ISSUE:

A sends an offer to B. That offer, for example, is for a single sendrecv
media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
and lists codecs 1 and 2. Now, the question is this: can B still send to
A using codec 3? Related to that, can B add codecs in the answer, which
A can then send with?

Arguments for YES:

1. A needs to be prepared to receive with codec 3 once it sends the
offer anyway.
2. If A wants to restrict the set based on the answer, it can always
generate a new offer with a reduced set of codecs.
3. SDP has always been interpreted as indicating your own capabilities,
and is therefore an "advertisement" that sticks until updated.
4. There are useful cases for adding codecs in the answer, particularly,
support for rfc2833. Thus, lets say A is an end user which can send rfc
2833, but not receive it. So, when it sends an INVITE, it doesn't
include rfc2833 in the INVITE. The server that responds is a media
server, which supports rfc2833. So, it includes it in the answer. Now, A
can actually send with it during the middle of the session, and it will
never receive it (since it didn't advertise it). This provides a useful
function that would otherwise require FID.

Arguments for NO:

1. The purpose of the O/A model is to agree on shared views of certain
session parameters, such as the directionality (send/recv/sendrecv).
Codecs fit into that category as well, since you are already required to
indicate, in the answer, a set that partially overlaps with the offer.
Furthermore, the goal with codecs is to have the same in both
directions, generally, so that this would speak to them being a shared
state that is negotiated.

2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If
you go for YES, there may be interop issues. The converse is not true,
of course.

3. It is very useful to narrow down the codec sets in the answer, and we
should not always require an extra exchange of O/A just to do that.



I took a private poll amongst some implementors. The initial results
showed that most assumed the YES case. Some more mails since then
indicate its a bit more split, although I think somewhat in favor of YES
still.

I am somewhat on the fence; I think NO is more correct based on
offer/answer, although YES would be the SDP purist view. What I really
want is more opinions on this one....

THanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Tue Feb 19 09:31:23 2002
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Date: Tue, 19 Feb 2002 17:02:00 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Peili Xu <xupeili@huawei.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Re: Question About SDO Offer/Answer draft
References: <003c01c1ad1e$91b273a0$db100b0a@huawei.com.cn> <3C5DF33A.783BB466@dynamicsoft.com> <000f01c1b8ef$f0b9fda0$81190b0a@huawei.com.cn>
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Hello,

Peili Xu wrote:
> 
> But in bis07 line 2036 it reads:
> "Once the UAS has sent or received an answer to the initial offer, it MUST NOT generate subsequent
> offers in any responses to the INVITE. Since only the UAC can send PRACK, this means the a UAS
> based on this specification alone can never generate subsequent offers. "
> 
> This means that the UAS cannot start a second offer use 200 and get answer in ACK??
> 

If at the same time that the UAS is sending a new offer in the 200 OK
the UAC is sending an UPDATE, we will have a glare condition. But the
UAC has no way of rejecting the offer in the 200 OK (SIP does not have a
NACK method). That's why this is disallowed.

Regards,

Gonzalo

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From: "Peter Barany"<pbarany@nortelnetworks.com>
To: "'Roman Shpount'" <roman@atelo.com>,
        "Sanjoy Sen"<sanjoy@nortelnetworks.com>, jdrosen@dynamicsoft.com,
        mmusic@ietf.org
Cc: "'msec@securemulticast.org'" <msec@securemulticast.org>
Subject: RE: [MMUSIC] RE: summarizing discussion on codec placement in ans
	wer
Date: Tue, 19 Feb 2002 10:29:14 -0600
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Hi,

If we assume that we are talking about SRTP (draft-ietf-avt-srtp-02.txt)
with the key exchange mechanism/SDP extensions being defined in
draft-ietf-msec-mikey-00.txt and draft-ietf-mmusic-kmgmt-ext-01.txt,
respectively, then there should be no ambiguity with early media solution(s)
provided that the new proposed SDP attribute "a=keymgmt-data:" contains
based-64 encoded MIKEY payloads in the UPDATE request method contains all of
the parameters (e.g., SSRC,etc.) required to create a cryptographic
context(s) (...and then provided that the response code to the UPDATE
request method accepts the codec(s) offered to it).

The potential problem with the previous e-mail regarding offer/answer issue
related to a conflict regarding the sending of "a=keymgmt-data:" in SDP
with, for example, 3 cryptographic context(s) (3 SSRCs) but only two
codec(s). See below. Thanks.

Regards,

Pete

-----Original Message-----
From: Roman Shpount [mailto:roman@atelo.com]
Sent: Monday, February 18, 2002 6:49 PM
To: Sen, Sanjoy [NGC:B692:EXCH]; jdrosen@dynamicsoft.com;
mmusic@ietf.org
Subject: [MMUSIC] RE: summarizing discussion on codec placement in
answer


Sanjoy Sen wrote:

> Also, its not clear to some of us how would you negotiate the 
> attributes that're associated with the media session (e.g., security 
> parameters). Example, suppose Offerer sends 3 codecs - 2,3,4 and the 
> answerer responds with 3,4 and adds an a=keymgmt-data: attribute for
> this media, should the offerer be ready to accept 2 with the security 
> keys applied?

I have a related question: How would the same security parameter negotiation
work with early media?
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com






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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: [MMUSIC] RE: summarizing discussion on codec placement in =
answer</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>If we assume that we are talking about SRTP =
(draft-ietf-avt-srtp-02.txt) with the key exchange mechanism/SDP =
extensions being defined in draft-ietf-msec-mikey-00.txt and =
draft-ietf-mmusic-kmgmt-ext-01.txt, respectively, then there should be =
no ambiguity with early media solution(s) provided that the new =
proposed SDP attribute &quot;a=3Dkeymgmt-data:&quot; contains based-64 =
encoded MIKEY payloads in the UPDATE request method contains all of the =
parameters (e.g., SSRC,etc.) required to create a cryptographic =
context(s) (...and then provided that the response code to the UPDATE =
request method accepts the codec(s) offered to it).</FONT></P>

<P><FONT SIZE=3D2>The potential problem with the previous e-mail =
regarding offer/answer issue related to a conflict regarding the =
sending of &quot;a=3Dkeymgmt-data:&quot; in SDP with, for example, 3 =
cryptographic context(s) (3 SSRCs) but only two codec(s). See below. =
Thanks.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Pete</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Roman Shpount [<A =
HREF=3D"mailto:roman@atelo.com">mailto:roman@atelo.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, February 18, 2002 6:49 PM</FONT>
<BR><FONT SIZE=3D2>To: Sen, Sanjoy [NGC:B692:EXCH]; =
jdrosen@dynamicsoft.com;</FONT>
<BR><FONT SIZE=3D2>mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [MMUSIC] RE: summarizing discussion on =
codec placement in</FONT>
<BR><FONT SIZE=3D2>answer</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Sanjoy Sen wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Also, its not clear to some of us how would you =
negotiate the </FONT>
<BR><FONT SIZE=3D2>&gt; attributes that're associated with the media =
session (e.g., security </FONT>
<BR><FONT SIZE=3D2>&gt; parameters). Example, suppose Offerer sends 3 =
codecs - 2,3,4 and the </FONT>
<BR><FONT SIZE=3D2>&gt; answerer responds with 3,4 and adds an =
a=3Dkeymgmt-data: attribute for</FONT>
<BR><FONT SIZE=3D2>&gt; this media, should the offerer be ready to =
accept 2 with the security </FONT>
<BR><FONT SIZE=3D2>&gt; keys applied?</FONT>
</P>

<P><FONT SIZE=3D2>I have a related question: How would the same =
security parameter negotiation work with early media?</FONT>
<BR><FONT SIZE=3D2>___________________________________</FONT>
<BR><FONT SIZE=3D2>Roman Shpount, VP of Technology</FONT>
<BR><FONT SIZE=3D2>aTelo, Inc. </FONT>
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<BR><FONT SIZE=3D2>www.atelo.com</FONT>
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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-461083 ] Body w/o Content-Length clarification (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 18 Feb 2002 16:52:22 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-461083 ] Body w/o Content-Length clarification

Bugs item #461083, was opened at 2001-09-12 18:50
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=461083&group_id=23194

Category: General
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Jason Lango (jasonlango)
Assigned to: Rob Lanphier (robla)
Summary: Body w/o Content-Length clarification

Initial Comment:
RFC 2326 says: 

4.4 Message Length

   When a message body is included with a message, the
length of that body is determined by one of the
following (in order of precedence):
 
   1.     Any response message which MUST NOT include a
message body
          (such as the 1xx, 204, and 304 responses) is
always terminated
          by the first empty line after the header
fields, regardless of
          the entity-header fields present in the
message. (Note: An
          empty line consists of only CRLF.)

   2.     If a Content-Length header field (section
12.14) is present,
          its value in bytes represents the length of
the message-body.
          If this header field is not present, a value
of zero is
          assumed.
 
   3.     By the server closing the connection.
(Closing the connection
          cannot be used to indicate the end of a
request body, since
          that would leave no possibility for the
server to send back a
          response.)
 
.. so it is saying that closing the connection is Ok to
indicate the length.

.. but then in section 12.14, it also says that the
Content-Length header MUST be sent.

The two sections should probably be made consistent.

Currently a popular RTSP server implementation (QTSS)
sends
"Connection: Close" followed by a message body for
certain
RTSP error status responses.

It might be worth requiring 'Connection: Close' in item
3 of
section 4.4 to reduce the ambiguity of whether or not a
compatible implementation may expect a response to
contain
a message body.

Alternatively it might be worth -not- saying that
message length
may be determined by the server closing the connection.


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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-18 16:52

Message:
Logged In: YES 
user_id=3796

Eliminated item #3 from the list.

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 14:28

Message:
Logged In: YES 
user_id=437905

RESOLUTION: (Jan 9, 2002 teleconference)

Eliminate implicit body-length via connection: close.  
COntactt David SInger on this issues and encourage f 
future participation.  Verify resolution is consistent 
with SIP specificaiton.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 09:32

Message:
Logged In: YES 
user_id=3796

This is a potentially gnarly issue.  We agree that this is a problem that needs fixing.

We propose that we eliminate the option of implied Content-Length if possible (need 
to check with implementors on this....Darwin server in particular).  If we can't do 
that, require Connection: close, and change header definition.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=461083&group_id=23194




End of Rtspspec-bugs Digest


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From mmusic-admin@ietf.org  Wed Feb 20 01:47:48 2002
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Date: Wed, 20 Feb 2002 01:43:24 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Roman Shpount wrote:
> 
> Sanjoy Sen wrote:
> 
> > Also, its not clear to some of us how would you negotiate the
> > attributes that're associated with the media session (e.g., security
> > parameters). Example, suppose Offerer sends 3 codecs - 2,3,4 and the
> > answerer responds with 3,4 and adds an a=keymgmt-data: attribute for
> > this media, should the offerer be ready to accept 2 with the security
> > keys applied?

That just doesn't work.

My understanding of the a-keymngmt-data attribute is that it is a tunnel
for mikey. MIKEY, as I understand it, requires a two pass exchange.
Thus,
you would have to have this attribute in the offer and the answer for
this
to work.

The a=keymngmt-data attribute is hardly a normal session attribute, for
this reason. As far as other attributes are concerned, some may be
negoiated (like the directional ones), and others are merely declarative
(like ptime) such that there is no relationship between the values in
the offer and the answer. I think the current o/a doc points this out.
There is no general way to know which case it is for a particular
attribute.

> 
> I have a related question: How would the same security parameter
> negotiation work with early media?

No different! Thats the entire idea behind the unify proposal. With
unify, early media is just an O/A exchange that finishes before the sip
invite transaction does. Since mikey will work by embedding in any O/A
exchange, as far as I can tell, the fact that the exchange is before the
sip transaction is completed is totally irrelevant. Please do read the
unify draft:

http://search.ietf.org/internet-drafts/draft-rosenberg-sip-unify-00.txt

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb 20 01:51:27 2002
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Peili Xu wrote:
> 
> But in bis07 line 2036 it reads:
> "Once the UAS has sent or received an answer to the initial offer, it
> MUST NOT generate subsequent
> offers in any responses to the INVITE. Since only the UAC can send
> PRACK, this means the a UAS
> based on this specification alone can never generate subsequent offers.
> "
> 
> This means that the UAS cannot start a second offer use 200 and get
> answer in ACK??

Right, for the reasons Gonzalo has outlined. 

> 
> There is a conflict here.

Indeed. I suspect my response, which you quote below, was sent before I
had fully worked it all out, or that I was just plain confused. Either
way, my apologies.

-Jonathan R.

> 
> ----- Original Message -----
> From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> To: Peili Xu <xupeili@huawei.com>
> Cc: <mmusic@ietf.org>
> Sent: Monday, February 04, 2002 10:34 AM
> Subject: Re: Question About SDO Offer/Answer draft
> 
> >
> >
> > Peili Xu wrote:
> > >
> > > In section 4 "Protocol Operatioin":
> > >
> > > It says that the agent Must Not generate a new offer if it has
> received
> > > an offer which it has not yet answered or rejected.
> > >
> > > But for early media we use 180/PRACK to offer and answer the early
> media
> > > just before
> > >
> > > the Invite/200 interaction is completed.
> > >
> > > Is this method break the Offer/Answer law?
> >
> > No. It works like this:
> >
> >
> > INVITE (offer 1) -->
> > <-----183 (answer 1)-
> >
> > <------200 OK (offer 2)--
> > -------ACK (answer 2)---->
> >
> > that is, the 200 OK is a new offer, and then SDP is present in the ACK
> > with an answer.
> >
> > This is different from the "common" practice today for early media,
> > which has the 183 as an anwer, with a 200 OK that repeats the same
> SDP.
> > I believe there will largely be backwards compatibility. If the UAS
> does
> > the "common" thing when talking to an element compliant to bis/o-a, it
> > will send SDP in the ACK, but so long as its not different from the
> > offer in the INVITE, there won't be any issues.
> >
> > -Jonathan R.
> >
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb 20 10:50:35 2002
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Message-ID: <3C73BF9A.5AFE15A7@era.ericsson.se>
Date: Wed, 20 Feb 2002 16:24:11 +0100
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Subject: [MMUSIC] RTSP state machine
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Hi all RTSP interested people,

Here comes an updated RTSP state machine which tries to clarify the
different transactions possible and their differences in functionality.
It is based on the current proposal for handling aggregated and
non-aggregated control. PAUSE and PLAY can't be used on individual
media. If that is allowed there will be a number of extra versions of
transitions with requirements on the different media state.

It reflects my understanding of the currently proposed functionality and
I hope that intressted people take the time to at least read through it.
Please comment if you have questions or found errors or omitted things.
The goal is that this state machine will help give a common view and
understanding.

This state machine does not contain RECORD which needs to be added.

State transitions not present are not allowed. Some RTSP methods like
options, and  SET_PARAMETER does not effect these states.


States          Explanation
-----------------------------------------------------------------------
Init            Initial state no session exist
Ready-na        Non aggregated session in Ready state
Playing-na      Non aggregated session in Playing state
Paused-na       Non aggregated session in Paused state
Ready-a         Aggregated session in Ready state
Playing-a       Aggregated session in Playing
Paused-a        Aggregated session in Pause

The reason for separating aggregated and non aggregated states are the
following:
- Play and Pause may only be done on aggregate URL and not media URLs
  in aggregated control. In non-aggregated the two are the same.


State variables
---------------
Nr-media    The number of media which are currently ready.

State: Init
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP                           Ready-na        Nr-media = 1


State: Ready-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media = 2
SETUP           Present URL     Ready-na        Change transport param.
TEARDOWN                        Init            Ok
PLAY            No range        Playing-na      from beginning of media
PLAY            Range           Playing-na      according to range
Timeout                         Init

State: Playing-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY                            Playing-na      Queue play
PAUSE           No range        Paused-na       Set pause-point
PAUSE           Range > now     Playing-na      Set pause-point
Pause point reached             Paused-na
End of media                    Ready-na
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP     Present URL, Allowed? Playing-na      Change transport param.
TEARDOWN        Allowed?        Init


State: Paused-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY            No range        Playing-na      Play from pause point
PLAY            Range           Playing-na      according to range
Timeout                         Init
SETUP           New URL         Paused-a        Nr-media +=1
SETUP           Present URL     Ready-na        Change transport param.
TEARDOWN                        Init

State: Ready-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media +=1
SETUP           Present URL     Ready-a         Change transport param.
TEARDOWN        Nr-media >2     Ready-a         Nr-media -=1
TEARDOWN        Nr-media == 2   Ready-na        Nr-media = 1
Timeout                         Init
PLAY            No range,AggURL Playing-a       from beginning of media
PLAY            Range, Agg URL  Playing-a       according to range


State: Playing-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY                            Playing-a       Queue play
PAUSE           No range        Paused-a        Set pause-point
PAUSE           Range > now     Playing-a       Set pause-point
Pause point reached             Paused-a
End of media    >=1 Media plays Playing-a
End of media    All media ended Ready-a
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP      Present URL,Allowed? Ready-na        Change transport param.
TEARDOWN        Allowed? Nr-m>2 Playing-a       Nr-media -=1
TEARDOWN        Allowed? Nr-m=2 Playing-na      Nr-media = 1

State: Paused-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY            No range        Playing-a       Play from pause point
PLAY            Range           Playing-a       according to range
Timeout                         Init
SETUP           New URL         Paused-a        Nr-media +=1
SETUP           Present URL     Ready-na        Change transport param.
TEARDOWN        Nr-media >2     Paused-a        Nr-media -=1
TEARDOWN        Nr-media == 2   Paused-na       Nr-media = 1



Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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From mmusic-admin@ietf.org  Wed Feb 20 13:57:58 2002
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Hi again!

I made some errors in the table and I guess there may be some more still
in there. Anyway, here is a new version.

RTSP state machine ver. 0.2

This state machine does not contain RECORD which needs to be added.
It is based on the proposal that in aggregated control, PAUSE and PLAY
can't be used on individual medias. If that is allowed there will be
a number of extra transitions with requirements on the
different medias state. Does not include queued play either.

There is discussion if teardown on media URL should end up in a middle
state between Init and Ready-na. This state would contain no media but
keep all session related state and keep its ID. The only way to remove
such a state will be the teardown with * or a session timeout.


States          Explanation
-----------------------------------------------------------------------
Init            Initial state no session exist
Ready-na        Non aggregated session in Ready state
Playing-na      Non aggregated session in Playing state
Paused-na       Non aggregated session in Paused state
Ready-a         Aggregated session in Ready state
Playing-a       Aggregated session in Playing
Paused-a        Aggregated session in Pause

The reason for separating aggregated and non aggregated states are the
following:
- Play and Pause may only be done on aggregate URL and not media URLs
  in aggregated control. In non-aggregated the two are the same.
-


State variables
---------------
Nr-media

State: Init
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP                           Ready-na        Nr-media = 1


State: Ready-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media = 2
SETUP           Present URL     Ready-na        Change transport param.
Teardown        *|media URL     Init            Ok
PLAY            No range        Playing-na      from beginning of media
PLAY            Range           Playing-na      according to range
Timeout                         Init

State: Playing-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PAUSE           No range        Paused-na       Set pause point
PAUSE           Range > now     Playing-na      Set pause point
Pause-point reached             Paused-na
End of media                    Ready-na
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP     Present URL, Allowed? Playing-na      Change transport param.
TEARDOWN        Allowed?        Init


State: Paused-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY            No range        Playing-na      Play from pause point
PLAY            Range           Playing-na      according to range
Timeout                         Init
SETUP           New URL         Paused-a        Nr-media +=1
SETUP           Present URL     Ready-na        Change transport param.
TEARDOWN        *|media URL     Init

State: Ready-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media +=1
SETUP          Present mediaURL Ready-a         Change transport param.
TEARDOWN        * | Pres URL    Init            No session hdr.
TEARDOWN        Nr-media >2     Ready-a         Nr-media -=1, sess hdr
TEARDOWN        Nr-media == 2   Ready-na        Nr-media = 1, sess hdr
Timeout                         Init
PLAY            No range,AggURL Playing-a       from beginning of media
PLAY            Range, Agg URL  Playing-a       according to range


State: Playing-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PAUSE           No range        Paused-a        Set pause point
PAUSE           Range > now     Playing-a       Set point to pause at
Pause-point reached             Paused-a        Set pause point
End of media    >=1 Media plays Playing-a
End of media    All media ended Ready-a
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP      Present URL,Allowed? Playing-a       Change transport param
TEARDOWN        *|pres URL      Init            No session hdr allowed?
TEARDOWN        Allowed? Nr-m>2 Playing-a       Nr-media -=1, sess hdr
TEARDOWN        Allowed? Nr-m=2 Playing-na      Nr-media = 1, sess hdr


State: Paused-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PLAY            No range        Playing-a       Play from pause point
PLAY            Range           Playing-a       according to range
Timeout                         Init
SETUP           New URL         Paused-a        Nr-media +=1
SETUP           Present URL     Paused-a        Change transport param.
TEARDOWN        *|pres URL      Init
TEARDOWN        Nr-media >2     Paused-a        Nr-media -=1
TEARDOWN        Nr-media == 2   Paused-na       Nr-media = 1





Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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From mmusic-admin@ietf.org  Wed Feb 20 20:29:29 2002
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Subject: [MMUSIC] Re: [Sip-implementors] updated offer/answer draft
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Sorry to be slow commenting - I am backed up on my reading.

I have one possible nit. You have incorporated a:inactive as the
replacement for c=0.0.0.0 for hold, and indicated that use of c=0.0.0.0
is deprecated though it still must be supported by recipients.

I think it is a bit strong to say that it is deprecated. There are still
cases where it is *needed*. In particular, a B2BUA may need to offer on
hold to solicit an answer, with the intent to make a new offer later
after another offer/answer exchange with another party. In this case, it
may not desire or be capable of receiving RTCP, or of providing an
address/port to do so. The C=0.0.0.0 provides an effective solution in
this case. 

So, rather than being deprecated for all use, I think the use of
c=0.0.0.0 ought to be be more narrowly restricted.

	Paul

Jonathan Rosenberg wrote:
> 
> Folks,
> 
> I have just submitted an update to the offer/answer model draft. Until
> it appears in the archives, you can pick it up at:
> 
> http://www.jdrosen.net/papers/draft-ietf-mmusic-sdp-offer-answer-00.txt
> 
> This draft includes all of the changes agreed to since IETF 52 and on
> the list since then. There are no open issues that I am aware of.
> 
> Perhaps the most substantial change is the allowance of different
> payload type numbers in each direction, as per list discussion on
> mmusic. Other changes are noted in the changes section.
> 
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

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From mmusic-admin@ietf.org  Thu Feb 21 03:34:40 2002
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Well, technically, working group last call has ended, and I am about to
submit this draft. But, since this was easy, I incorporated your
comment. The text now reads:

RFC 2543 specified that placing a user on hold was accomplished by
setting the connection address to 0.0.0.0. Its usage for putting a
call on hold is no longer recommended, since it doesn't allow for RTCP
to be used with held streams, doesn't work with IPv6, and breaks with
connection oriented media. However, it can be useful in an initial
offer when the offerer knows it wants to use a particular set of media
streams and formats, but doesn't know the addresses and ports at the
time of the offer. An agent {\MUST} be capable of receiving SDP with a
connection address of 0.0.0.0, in which case it means that neither RTP
nor RTCP should be sent to the peer.


Thanks,
Jonathan R.

Paul Kyzivat wrote:
> 
> Sorry to be slow commenting - I am backed up on my reading.
> 
> I have one possible nit. You have incorporated a:inactive as the
> replacement for c=0.0.0.0 for hold, and indicated that use of c=0.0.0.0
> is deprecated though it still must be supported by recipients.
> 
> I think it is a bit strong to say that it is deprecated. There are still
> cases where it is *needed*. In particular, a B2BUA may need to offer on
> hold to solicit an answer, with the intent to make a new offer later
> after another offer/answer exchange with another party. In this case, it
> may not desire or be capable of receiving RTCP, or of providing an
> address/port to do so. The C=0.0.0.0 provides an effective solution in
> this case.
> 
> So, rather than being deprecated for all use, I think the use of
> c=0.0.0.0 ought to be be more narrowly restricted.
> 
>         Paul
> 
> Jonathan Rosenberg wrote:
> >
> > Folks,
> >
> > I have just submitted an update to the offer/answer model draft. Until
> > it appears in the archives, you can pick it up at:
> >
> >
> http://www.jdrosen.net/papers/draft-ietf-mmusic-sdp-offer-answer-00.txt
> >
> > This draft includes all of the changes agreed to since IETF 52 and on
> > the list since then. There are no open issues that I am aware of.
> >
> > Perhaps the most substantial change is the allowance of different
> > payload type numbers in each direction, as per list discussion on
> > mmusic. Other changes are noted in the changes section.
> >
> > Thanks,
> > Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@cs.columbia.edu
> > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Thu Feb 21 03:59:37 2002
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Folks,

we need to reach closure on this issue fairly shortly.
Having gone through the mailing list discussion, we believe
that we can narrow down Jonathan's summary a bit further
and come up with a concrete proposal to beat up:

Restating the issue:

> A sends an offer to B. That offer, for example, is for a single sendrecv
> media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> and lists codecs 1 and 2. Now, the question is this: can B still send to
> A using codec 3? Related to that, can B add codecs in the answer, which
> A can then send with?

Let us keep in mind that we are not designing offer/answer for telephony
but that telephony is rather one (maybe not unimportant) special case of
multimedia communications.  Let us also keep in mind that we should
optimize for the more common cases but at the same time not lose
generality.  And let us not forget that SIP is just *one* customer of
offer/answer.

Let us assume for a moment that we follow the "NO" proposal (and assume
further that this approach does not break interoperability/backward
compatility):

1. This idea allows us to narrow down codecs in an efficient fashion.

   This helps people who do not want to require their implementations
   to listen for multiple codecs simultaneously (e.g. because of
   limited DSP resources or difficulties in implementing MCUs).

   The counter-argument is that such device need to be able to support
   multiple codecs in the beginning anyway because of the race condition
   between the answer and the first RTP packet.  However, this can be
   circumvent this by simply waiting for the answer before accepting
   media at the risk of clipping.  But people may be able to architected
   their network in a way that it does not delay the answer unncessarily
   and thus avoid clipping in the first place).  Alternatively, they may
   guess which type of codec to expect or only load the decoder upon
   arrival of the first packet (assuming this is not going to change) at
   the expense of some additional delay, etc.

   Looks like there are sufficient alternatives that clipping can be
   considered a non-issue.  (Please note that if you are doing security
   handshakes and QoS preconditions or similar stuff, you are not going
   to worry about this type of clipping anyway -- but rather about post-
   dial delay.)

2. Expanding the codec set can be done by means of an extra offer in
   the opposite direction.  This is an extra exchange but we would assume
   this to be the less common case (and thus not optimize for it).

   The only application scenario for which this is really needed people
   have mentioned so far is DTMF -- more specifically, combining a
   sendrecv and an sendonly/recvonly media stream.  Please share ideas
   for others with us.

   We do acknowledge that this scenario is important and need support.
   But we also judge that this is not the common case and need not be
   optimized for at all cost if a cleaner solution exists.

   This cleaner solution can be based upon FID.  In the simplest case,
   A uses FID from the very beginning:

   A ----  INVITE   ----> B
       a=group:FID 1 2
       m=audio pt=0
       a=mid:1
       m=audio pt=97
       a=rtpmap:97 telephone-events
       a=mid:2
       a=sendonly

   and B can then decide whether or not to reject telephone events.

   However, A may or may not understand FID and may or may not indicate
   its send capabilities as described above.  Then we get the following
   simple exchange for A calling B:

   A ---- INVITE ----> B
          m=audio pt=0
      
   A <--- 200 OK ----- B
          m=audio pt=0
   A ----  ACK   ----> B

   Now B can indicate its receive caps using FID doing a re-INVITE.

   A <--- INVITE ----- B
          a=group:FID 1 2
          m=audio pt=0
          a=mid:1
          m=audio pt=97
          a=rtpmap:97 telephone-events
          a=mid:2
          a=recvonly

   ...

   Again, we can properly express the intended semantics.  If A does not
   support FID it ignores the attributes thus yielding the following
   in A's perception:

   A <--- INVITE ----- B
          m=audio pt=0
          m=audio pt=97
          a=rtpmap:97 telephone-events

   ...

   This is the alternative that can be pursued without FID (and, at the
   same time serves as fallback).  According to the rules defined in
   offer/answer for local processing of multiple streams of the same
   type with one local source or sink, this should just work.  The sender
   copies the input to all streams if the data can be mapped (i.e. only
   events go across the second media session), the receiver mixes the
   the incoming data streams locally as appropriate.  In a gateway this
   may sometimes mean to locally generate DTMF tones and mix them with
   audio signals, sometimes mean to locally sink the DTMF and process it.

   Jonathan, is this interpretation correct?  Is there need for clarifying
   text on this issue regarding offer/answer?

   The major issue to discuss here is whether or not there is an extra RTT
   required for this signaling (or whether other SIP methods can be used
   to achieve this more quickly) -- and whether the additional delay
   (whatever this may be) is an issue for DTMF.

   Thoughts?

Is this a viable approach ahead?

Note that we are not religious about this proposal -- but it seems to be a
reasonable way ahead taking into account some general design principles.
There may be other ways and if better ones are found, such proposals are
welcome.  But they need to be found NOW.

A small disclaimer in the end: probably we should be taking the more purist
SDP view (as Jonathan calles the "YES" approach) -- but since SDP is used 
in a way it was not supposed to be anyway the NO method may not cause too
much *extra* headache.

Colin and Joerg

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Date: Thu, 21 Feb 2002 04:02:01 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Well, the time has arrived for me to make a judgement call, and put some
text into offer/answer.

I think a sensible approach is a compromise. I phrased the question
thusly:

> ISSUE:
> 
> A sends an offer to B. That offer, for example, is for a single sendrecv
> media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> and lists codecs 1 and 2. Now, the question is this: can B still send to
> A using codec 3? Related to that, can B add codecs in the answer, which
> A can then send with?

There are two questions in there. I had assumed that both needed the
same answer, but they don't. Most of the advocates of NO want it because
of the first question (narrowing down the set). Most of the advocates of
YES wanted it because of the second question. So, let us do both. NO to
first, YES to second. This is actually consistent; the consistency is
that the answer contains the final set, whether it be a larger or
smaller one.

Specifically, for a sendrecv stream, the answerer MUST NOT send using
any media formats not present in both the offer and answer. It MUST be
prepared to receive with any codecs in the answer. The offerer MUST NOT
send using any media formats not present in the answer, and it MUST be
prepared to receive using any media formats in the offer which were also
in the answer.

Similarly, for an sendonly offer, the answerer can add media formats to
the answer (or put fewer).

For a recvonly offer, the answerer cannot add any formats (it makes no
sense).

This is not set in stone; there is the opportunity to modify this during
IESG last call, but it seems a reasonable compromise position.

Thanks,
Jonathan R.


Jonathan Rosenberg wrote:
> 
> Folks,
> 
> I wanted to take a moment to summarize where we were in regards to our
> long thread on interpretation of the codecs in the answer. The purpose
> of the summarization is to put all the points on the table, and
> hopefully invite some further comment from folks outside the few thread
> participants in the few days which remain.
> 
> ISSUE:
> 
> A sends an offer to B. That offer, for example, is for a single sendrecv
> media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> and lists codecs 1 and 2. Now, the question is this: can B still send to
> A using codec 3? Related to that, can B add codecs in the answer, which
> A can then send with?
> 
> Arguments for YES:
> 
> 1. A needs to be prepared to receive with codec 3 once it sends the
> offer anyway.
> 2. If A wants to restrict the set based on the answer, it can always
> generate a new offer with a reduced set of codecs.
> 3. SDP has always been interpreted as indicating your own capabilities,
> and is therefore an "advertisement" that sticks until updated.
> 4. There are useful cases for adding codecs in the answer, particularly,
> support for rfc2833. Thus, lets say A is an end user which can send rfc
> 2833, but not receive it. So, when it sends an INVITE, it doesn't
> include rfc2833 in the INVITE. The server that responds is a media
> server, which supports rfc2833. So, it includes it in the answer. Now, A
> can actually send with it during the middle of the session, and it will
> never receive it (since it didn't advertise it). This provides a useful
> function that would otherwise require FID.
> 
> Arguments for NO:
> 
> 1. The purpose of the O/A model is to agree on shared views of certain
> session parameters, such as the directionality (send/recv/sendrecv).
> Codecs fit into that category as well, since you are already required to
> indicate, in the answer, a set that partially overlaps with the offer.
> Furthermore, the goal with codecs is to have the same in both
> directions, generally, so that this would speak to them being a shared
> state that is negotiated.
> 
> 2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If
> you go for YES, there may be interop issues. The converse is not true,
> of course.
> 
> 3. It is very useful to narrow down the codec sets in the answer, and we
> should not always require an extra exchange of O/A just to do that.
> 
> I took a private poll amongst some implementors. The initial results
> showed that most assumed the YES case. Some more mails since then
> indicate its a bit more split, although I think somewhat in favor of YES
> still.
> 
> I am somewhat on the fence; I think NO is more correct based on
> offer/answer, although YES would be the SDP purist view. What I really
> want is more opinions on this one....
> 
> THanks,
> Jonathan R.
> 
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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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		: RTCP attribute in SDP
	Author(s)	: C. Huitema
	Filename	: draft-ietf-mmusic-sdp4nat-01.txt
	Pages		: 7
	Date		: 20-Feb-02
	
The session description protocol (SDP) is used to describe the 
parameters of media streams used in multimedia sessions. When a 
session requires multiple ports, SDP assumes that these port have 
consecutive numbers. However, when the session crosses a network 
address translation device that also uses port mapping, the ordering 
of ports can be destroyed by the translation. To handle this, we 
propose an extension attribute to SDP.

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	Title		: Key Management Extensions for SDP and RTSP
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-02.txt
	Pages		: 11
	Date		: 20-Feb-02
	
This document defines general extensions for SDP and RTSP to carry
the security information needed by a key management protocol, in
order to secure the media. These extensions are presented as a
framework, to be used by one or more key management protocols. As
such, its use is meaningful only when it is completed by the key
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	Title		: Requirements for Distributed Control of ASR and TTS 
                          Resources
	Author(s)	: E. Burger, D. Oran
	Filename	: draft-burger-mrcp-reqts-00.txt
	Pages		: 8
	Date		: 20-Feb-02
	
This document outlines the needs and requirements for a protocol to 
control distributed speech processing of audio streams.  By speech 
processing, this document specifically means automatic speech 
recognition and text-to-speech.  Other IETF protocols, such as SIP 
and RTSP, address rendezvous and control for generalized media 
streams.  However, speech processing presents additional 
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From mmusic-admin@ietf.org  Thu Feb 21 08:50:49 2002
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Date: Thu, 21 Feb 2002 08:49:42 -0500
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Organization: dynamicsoft
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Folks,

I've just submitted an update to the offer/answer draft. Until it
appears in the archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-sdp-offer-answer-01.txt

This is based on comments received during working group last call, and
is the version which will go into IESG last call.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
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Folks,

MMUSIC is scheduled to meet once for a two hour slot in Minneapolis.
The current schedule is

	Monday, 18 March 2002, 1530 - 1730
	(but this may still change)

We are now starting to put together the agenda.  Please direct slot
requests to csp@isi.edu and jo@ipdialog.com.

Please note that agenda slots are meant for discussion of items on the
WG charter or at least candidates for discussion in the working group.
Please also note that meeting time is not meant for presentation of
ideas but rather for discussing and solving issues.

Please also take note of the IETF statement regarding IPR to be found at

	http://www.ietf.org/meetings/notewell.html

The draft agenda and all related documents will be posted soon as

	http://www.dmn.tzi.org/ietf/mmusic

where you can also find past agendas, Internet Drafts, meeting minutes,
etc.

Thanks,
Joerg


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From: "William Janning"<billj@nortelnetworks.com>
To: "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "'jdrosen@dynamicsoft.com'"
	 <jdrosen@dynamicsoft.com>
Cc: "Sanjoy Sen"<sanjoy@nortelnetworks.com>,
        "Mary Barnes"<mbarnes@nortelnetworks.com>
Subject: RE: [VoIP-list] FW: [MMUSIC] summarizing discussion on codec plac
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I got this but am not currently subscribed to mmusic. I guess I should join.
My reply to this discussion thread follows...

-----Original Message-----
From: Janning, William [NGC:B601:EXCH] 
Sent: Wednesday, February 20, 2002 2:39 PM
To: Sen, Sanjoy [NGC:B692:EXCH]; Audet, Francois [SC2:4K02:EXCH]; McGrath,
Tim [NGC:B682:EXCH]
Cc: List, VOIP [NCRTP:9999:UNIX]; Barnes, Mary [NGC:B601:EXCH]; Orton, Scott
[NGC:B633:EXCH]; Parameswar, Sriram [NGC:B692:EXCH]; Bharatia, Jayshree
[RICH2:2H13:EXCH]; Hogg, Chris [MDN05:MA01:EXCH]; Watson, Mark
[MDN05:EP10:EXCH]; Stucker, Brian [NGC:B635:EXCH]; Taylor, Tom-PT
[CAR:B800:EXCH]
Subject: RE: [VoIP-list] FW: [MMUSIC] summarizing discussion on codec plac
ement in answer


NO is definitely the answer. 
 
When "A" sends a request with SDP, it is saying "I CAN do 1,2,3". When "B"
responds, it is saying "I WILL do 1, 2, or 3". In reality it makes no sense
for "B" to respond with "I WILL do 1 and 2" as practically speaking this
would mean B could dynamically switch between receiving codec 1 or 2 at the
whim of A, or that B would be able to properly mix 1 & 2 to the transducer.
If this is negotiation, then let us CHOOSE.
 
There is an assumption in codec selection that the codec "B" elects to
receive is also the one it is sending (so it must pick from "A"'s initial
list). This is implied in the SDP protocol because a single media
description only negotiates a single codec - there is not provision to
identify the send codec from the receive codec. This fact and the above
practicality are why NO is the answer with even additional restrictions (B
should never reply with multiple codecs). 
 
Supporting the codec capability preference for a particular session for
clients is completely related to bandwidth management and perceived voice
quality - though mostly bandwidth management. If everybody could do G711 or
G722, then they would. Period. So if I say I will receive G729, then there's
no way I'm going to send G711 or even G723. Likewise I can't say I'll
receive G711 (because I have higher download bandwidth) but send G729
(because my upstream is limited). "A" would have no clue I'm doing so
because all I replied in my response was G711. 
 
There is no such thing as a "send only" or "receive only" codec - codecs
always support both encoding/decoding. However if I did want to specify
unique send and receive codecs, then the use of two media descriptions could
be used as such:
 
v=0
o=user 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user
t=104100432 0
m=audio 9900 RTP/AVP 0
c=IN IP4 192.168.0.1
a=rtpmap:0 PCMU/8000
a=ptime:30
a=recvonly
m=audio 9902 RTP/AVP 4 18
c=IN IP4 192.168.0.1
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=ptime:10
a=sendonly

 
I would argue that the attributes "sendonly, recvonly, sendrecv" are
MANDATORY in messages and reflect the "I WILL" versus "I CAN", though the
difference is also identified by "response" versus "request" message. If the
originator wanted to tell the far-end that it was receiving already in the
request, then it MUST use the recvonly attribute (though this is a security
risk - since the originator doesn't know who the far-end is supposed to be,
anybody can be sending the originator packets which they'd happily play
until the real far-end answers (assuming of course there was no early media
established like on a 183). 
 
In order to really handle multiple m= lines with the same media type,
something like s= lines should also be per media description (or another
field added to the m= line to uniquely identify the media description). This
would eliminate m= ordering requirements and facilitate better renegotiation
management.
 
---------------------------------------------------


Arguments for YES: 

1. A needs to be prepared to receive with codec 3 once it sends the 
offer anyway. 

It is a security risk for A to receive RTP before it knows about the
far-end. "A' should not receive anything until it receives an SDP response.
A should not "play" received RTP unless it comes from the address indicated
in the far-end's media description (one reason why c= lines should always be
per media description, not session description). 


2. If A wants to restrict the set based on the answer, it can always 
generate a new offer with a reduced set of codecs. 

B can do likewise. Once the Offer-Answer completes, SDP is set until another
Offer-Answer is initiated by either side.


3. SDP has always been interpreted as indicating your own capabilities, 
and is therefore an "advertisement" that sticks until updated. 

We must clearly delineate when we are broadcasting our capabilities versus
when we are choosing for session establishment. At some point in time we
have to CHOOSE. Advertising your capabilities should be done via some other
mechanism like OPTIONS.


4. There are useful cases for adding codecs in the answer, particularly, 
support for rfc2833. Thus, lets say A is an end user which can send rfc 
2833, but not receive it. So, when it sends an INVITE, it doesn't 
include rfc2833 in the INVITE. The server that responds is a media 
server, which supports rfc2833. So, it includes it in the answer. Now, A 
can actually send with it during the middle of the session, and it will 
never receive it (since it didn't advertise it). This provides a useful 
function that would otherwise require FID. 

Either : a) secondary codec's are uniquely flagged in the media description
for a specific purpose (like 2833), b) multiple m= lines are used and the
response is allowed to negotiate in capabilities which would then be
responded to in the ACK. For instance:

Request :



v=0
o=user1 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user1
t=104100432 0
m=audio 9900 RTP/AVP 4 18 0
c=IN IP4 192.168.0.1
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:30

Response :

v=0
o=user2 13723311241683545 13723311241683545 IN IP4 192.168.0.2
s=session1
p=+user2
t=104100452 0
m=audio 9800 RTP/AVP 0
c=IN IP4 192.168.0.2
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
m=audio 9802 RTP/AVP 99
c=IN IP4 192.168.0.2
a=rtpmap:99 RFC2833/8000
a=ptime:10
a=recvonly

ACK :  (Really just necessary to confirm the change

v=0
o=user1 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user1
t=104100432 0
m=audio 9900 RTP/AVP 0
c=IN IP4 192.168.0.1
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
m=audio 9802 RTP/AVP 99
c=IN IP4 0.0.0.0
a=rtpmap:99 RFC2833/8000
a=ptime:10
a=sendonly
  

The latter choice is good in that it can be used for other things as well
(e.g. the term party could negotiate in video without having to generate
another Offer/Answer exchange) . This would also support the conference
server case where it would like to negotiate a separate return stream (e.g.
G729 in, G711 out).
 
William Janning
Nortel Networks 
billj@nortelnetworks.com <mailto:billj@nortelnetworks.com> 
sip:billj@ <mailto:sip:billj@sipfx .com> sipfx .com



-----Original Message-----
From: Sen, Sanjoy [NGC:B692:EXCH] 
Sent: Monday, February 18, 2002 11:52 AM
Subject: [VoIP-list] FW: [MMUSIC] summarizing discussion on codec placement
in answer
Importance: High



Do we've any opinion on the interpretation of the ANSWER in the SDP
Offer/Answer model (please read Jonathan's email below)? I would
particularly appreciate some feedback from our codec experts.

Sanjoy 


-----Original Message----- 
From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com
<mailto:jdrosen@dynamicsoft.com> ] 
Sent: Friday, February 15, 2002 3:32 PM 
To: mmusic@ietf.org 
Subject: [MMUSIC] summarizing discussion on codec placement in answer 


Folks, 

I wanted to take a moment to summarize where we were in regards to our 
long thread on interpretation of the codecs in the answer. The purpose 
of the summarization is to put all the points on the table, and 
hopefully invite some further comment from folks outside the few thread 
participants in the few days which remain. 

ISSUE: 

A sends an offer to B. That offer, for example, is for a single sendrecv 
media stream, which lists codecs 1,2 and 3. The answerer, B, responds, 
and lists codecs 1 and 2. Now, the question is this: can B still send to 
A using codec 3? Related to that, can B add codecs in the answer, which 
A can then send with? 

Arguments for YES: 

1. A needs to be prepared to receive with codec 3 once it sends the 
offer anyway. 
2. If A wants to restrict the set based on the answer, it can always 
generate a new offer with a reduced set of codecs. 
3. SDP has always been interpreted as indicating your own capabilities, 
and is therefore an "advertisement" that sticks until updated. 
4. There are useful cases for adding codecs in the answer, particularly, 
support for rfc2833. Thus, lets say A is an end user which can send rfc 
2833, but not receive it. So, when it sends an INVITE, it doesn't 
include rfc2833 in the INVITE. The server that responds is a media 
server, which supports rfc2833. So, it includes it in the answer. Now, A 
can actually send with it during the middle of the session, and it will 
never receive it (since it didn't advertise it). This provides a useful 
function that would otherwise require FID. 

Arguments for NO: 

1. The purpose of the O/A model is to agree on shared views of certain 
session parameters, such as the directionality (send/recv/sendrecv). 
Codecs fit into that category as well, since you are already required to 
indicate, in the answer, a set that partially overlaps with the offer. 
Furthermore, the goal with codecs is to have the same in both 
directions, generally, so that this would speak to them being a shared 
state that is negotiated. 

2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If 
you go for YES, there may be interop issues. The converse is not true, 
of course. 

3. It is very useful to narrow down the codec sets in the answer, and we 
should not always require an extra exchange of O/A just to do that. 



I took a private poll amongst some implementors. The initial results 
showed that most assumed the YES case. Some more mails since then 
indicate its a bit more split, although I think somewhat in favor of YES 
still. 

I am somewhat on the fence; I think NO is more correct based on 
offer/answer, although YES would be the SDP purist view. What I really 
want is more opinions on this one.... 

THanks, 
Jonathan R. 

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue 
Chief Scientist                         First Floor 
dynamicsoft                             East Hanover, NJ 07936 
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050 
http://www.jdrosen.net <http://www.jdrosen.net>                   PH:  (973)
952-5000 
http://www.dynamicsoft.com <http://www.dynamicsoft.com>  

_______________________________________________ 
mmusic mailing list 
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https://www1.ietf.org/mailman/listinfo/mmusic
<https://www1.ietf.org/mailman/listinfo/mmusic>  


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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=230570716-21022002>I got 
this but am not currently subscribed to mmusic. I guess I should join. My reply 
to this discussion thread follows...</SPAN></FONT></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Janning, William 
  [NGC:B601:EXCH] <BR><B>Sent:</B> Wednesday, February 20, 2002 2:39 
  PM<BR><B>To:</B> Sen, Sanjoy [NGC:B692:EXCH]; Audet, Francois [SC2:4K02:EXCH]; 
  McGrath, Tim [NGC:B682:EXCH]<BR><B>Cc:</B> List, VOIP [NCRTP:9999:UNIX]; 
  Barnes, Mary [NGC:B601:EXCH]; Orton, Scott [NGC:B633:EXCH]; Parameswar, Sriram 
  [NGC:B692:EXCH]; Bharatia, Jayshree [RICH2:2H13:EXCH]; Hogg, Chris 
  [MDN05:MA01:EXCH]; Watson, Mark [MDN05:EP10:EXCH]; Stucker, Brian 
  [NGC:B635:EXCH]; Taylor, Tom-PT [CAR:B800:EXCH]<BR><B>Subject:</B> RE: 
  [VoIP-list] FW: [MMUSIC] summarizing discussion on codec plac ement in 
  answer<BR><BR></DIV></FONT>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=723571419-20022002>NO 
  is definitely the answer. </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=723571419-20022002>When 
  "A" sends a request with SDP, it is saying "I CAN do 1,2,3". When "B" 
  responds, it is saying "I WILL do 1, 2, or 3". In reality it makes no sense 
  for "B" to respond with "I WILL do 1 and 2" as practically speaking this would 
  mean B could dynamically switch between receiving codec 1 or 2 at the whim of 
  A, or that B would be able to properly mix 1 &amp; 2 to the transducer. If 
  this is negotiation, then let us CHOOSE.</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002>There is an assumption in codec selection that the 
  codec&nbsp;"B"&nbsp;elects to receive is also the one it is sending (so it 
  must pick from "A"'s initial list). This is implied in the SDP protocol 
  because a single media description only negotiates a single codec - there is 
  not provision to identify the send codec from the receive codec. This fact and 
  the above practicality are why NO is the answer with even additional 
  restrictions (B&nbsp;should never reply with multiple codecs). 
  </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002>Supporting the codec capability preference for a 
  particular session for clients is completely related to bandwidth management 
  and perceived voice quality - though mostly bandwidth management. If everybody 
  could do G711 or G722, then they would. Period.&nbsp;So if I say I will 
  receive G729, then there's no way I'm going to send G711 or even G723. 
  Likewise I can't say I'll receive G711 (because I have higher download 
  bandwidth) but send G729 (because my upstream is limited). "A" would have no 
  clue I'm doing so because&nbsp;all I replied in my response was&nbsp;G711. 
  </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT><FONT color=#0000ff face=Arial 
  size=2><SPAN class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002>There is no such thing as a "send only" or "receive 
  only" codec - codecs always support both encoding/decoding. However if I did 
  want to specify unique send and receive codecs, then the use of two media 
  descriptions could be used as such:</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002>v=0<BR>o=user 
  13723311241683544 13723311241683544 IN IP4 192.168.0</SPAN></FONT><FONT 
  face=Arial size=2><SPAN class=723571419-20022002>.1</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=723571419-20022002>s=session1</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=723571419-20022002>p=+user</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002>t=104100432 
  0<BR><STRONG>m=audio </STRONG>9900 RTP/AVP 0<BR>c=IN IP4 192.168.0<FONT 
  face=Arial size=2><SPAN 
  class=723571419-20022002>.1</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><FONT face=Arial 
  size=2><SPAN class=723571419-20022002></SPAN></FONT>a=rtpmap:0 
  PCMU/8000<BR>a=ptime:30</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=723571419-20022002><STRONG>a=recvonly<BR>m=audio </STRONG>9902 RTP/AVP 4 
  18</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002>c=IN IP4 
  192.168.0<FONT face=Arial size=2><SPAN 
  class=723571419-20022002>.1</SPAN></FONT></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><FONT face=Arial 
  size=2><SPAN class=723571419-20022002></SPAN></FONT>a=rtpmap:4 
  G723/8000<BR>a=rtpmap:18 G729/8000<BR>a=ptime:10</DIV>
  <DIV><FONT face=Arial size=2><STRONG><SPAN 
  class=723571419-20022002>a=sendonly<BR></SPAN></STRONG></FONT></DIV></SPAN></FONT>
  <DIV>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=723571419-20022002>I 
  would argue that the attributes "sendonly, recvonly, sendrecv" are MANDATORY 
  in messages and reflect the "I WILL" versus "I CAN", though the difference is 
  also identified by "response" versus "request" message. If the originator 
  wanted to tell the far-end that it was receiving already in the request, then 
  it MUST use the recvonly attribute (though this is a security risk&nbsp;- 
  since the originator doesn't know who the far-end is supposed to be, anybody 
  can be sending the originator packets which they'd happily play until the real 
  far-end answers (assuming of course there was no early media established like 
  on a 183). </SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT><FONT color=#0000ff face=Arial 
  size=2><SPAN class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=723571419-20022002>In 
  order to really handle multiple m= lines with the same media type, something 
  like s= lines should also be per media description (or another field added to 
  the m= line to uniquely identify the media description). This would eliminate 
  m= ordering requirements and facilitate&nbsp;better renegotiation 
  management.</SPAN></FONT></DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
  class=723571419-20022002>---------------------------------------------------</SPAN></FONT></DIV><FONT 
  color=#0000ff><SPAN class=723571419-20022002>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <P><FONT face=Arial><FONT color=#000000 size=2>Arguments for YES: 
    </FONT></FONT></P>
    <P><FONT face=Arial><FONT color=#000000 size=2>1. A needs to be prepared to 
    receive with codec 3 once it sends the <BR>offer anyway. 
  </FONT></FONT></P></BLOCKQUOTE>
  <P><FONT face=Arial size=2><SPAN class=723571419-20022002>It is a security 
  risk for A to receive RTP before it knows about the far-end. "A' should not 
  receive anything until it receives an SDP response. A should not "play" 
  received RTP unless it comes from the address indicated in the far-end's media 
  description (one reason why c= lines should always be per media description, 
  not session description). </SPAN></FONT></P>
  <P><FONT face=Arial size=2><SPAN class=723571419-20022002></SPAN></FONT><FONT 
  color=#000000><BR><FONT face=Arial size=2>2. If A wants to restrict the set 
  based on the answer, it can always <BR>generate a new offer with a reduced set 
  of codecs. </FONT></FONT></P>
  <P><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
  class=723571419-20022002><FONT color=#0000ff>B can do likewise. Once the 
  Offer-Answer completes, SDP is set until another Offer-Answer is initiated by 
  either side.</FONT></SPAN></FONT></FONT></FONT></P>
  <P><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
  class=723571419-20022002></SPAN><BR>3. SDP has always been interpreted as 
  indicating your own capabilities, <BR>and is therefore an "advertisement" that 
  sticks until updated. </FONT></FONT></FONT></P>
  <P><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
  class=723571419-20022002><FONT color=#0000ff>We must clearly delineate when we 
  are broadcasting our capabilities versus when we are choosing for session 
  establishment. At some point in time we have to CHOOSE. Advertising your 
  capabilities should be done via some other mechanism like 
  OPTIONS.</FONT></SPAN></FONT></FONT></FONT></P>
  <P><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
  class=723571419-20022002></SPAN><BR>4. There are useful cases for adding 
  codecs in the answer, particularly, <BR>support for rfc2833. Thus, lets say A 
  is an end user which can send rfc <BR>2833, but not receive it. So, when it 
  sends an INVITE, it doesn't <BR>include rfc2833 in the INVITE. The server that 
  responds is a media <BR>server, which supports rfc2833. So, it includes it in 
  the answer. Now, A <BR>can actually send with it during the middle of the 
  session, and it will <BR>never receive it (since it didn't advertise it). This 
  provides a useful <BR>function that would otherwise require FID. 
  </FONT></FONT></FONT></P>
  <P><FONT face=Arial size=2><SPAN class=723571419-20022002>Either : a) 
  secondary codec's are uniquely flagged in the media description for a specific 
  purpose (like 2833), b) multiple m= lines are used and the response is allowed 
  to negotiate in capabilities which would then be responded to in the ACK. For 
  instance:</SPAN></FONT></P>
  <P><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>Request :</SPAN></FONT></P><SPAN 
  class=723571419-20022002>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <DIV><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
    class=723571419-20022002>v=0<BR>o=user1 13723311241683544 13723311241683544 
    IN IP4 192.168.0</SPAN><SPAN 
    class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></DIV>
    <DIV><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>s=session1</SPAN></FONT></DIV>
    <DIV><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>p=+user1</SPAN></FONT></DIV>
    <DIV><FONT color=#000000><SPAN class=723571419-20022002><FONT face=Arial 
    size=2>t=104100432 0<BR></FONT><SPAN class=723571419-20022002><SPAN 
    class=723571419-20022002></SPAN><FONT size=2><FONT 
    face=Arial><STRONG>m=audio 9900 RTP/AVP 4 18 0<BR></STRONG>c=IN IP4 
    192.168.0<SPAN 
    class=723571419-20022002>.1</SPAN></FONT></FONT></SPAN></SPAN></FONT></DIV>
    <DIV><FONT color=#000000><SPAN class=723571419-20022002><SPAN 
    class=723571419-20022002><SPAN class=723571419-20022002></SPAN><FONT 
    face=Arial size=2>a=rtpmap:4 G723/8000<BR>a=rtpmap:18 
    G729/8000<BR></FONT></SPAN></SPAN></FONT><SPAN 
    class=723571419-20022002><SPAN class=723571419-20022002></SPAN><FONT 
    color=#000000 face=Arial size=2>a=rtpmap:0 
    PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV></BLOCKQUOTE>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>Response :</SPAN></FONT><SPAN 
  class=723571419-20022002><SPAN class=723571419-20022002></DIV>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <DIV><FONT color=#000000><FONT size=2><FONT face=Arial><SPAN 
    class=723571419-20022002>v=0<BR>o=user2 13723311241683545 13723311241683545 
    IN IP4 192.168.0</SPAN><SPAN 
    class=723571419-20022002>.2</SPAN></FONT></FONT></FONT></DIV>
    <DIV><SPAN class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
    size=2><SPAN class=723571419-20022002>s=session1</SPAN></FONT></DIV>
    <DIV><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>p=+user2</SPAN></FONT></DIV>
    <DIV><FONT color=#000000><FONT size=2><FONT face=Arial><SPAN 
    class=723571419-20022002>t=104100452 0<BR><STRONG>m=audio 9800 RTP/AVP 
    0<BR></STRONG>c=IN IP4 192.168.0<SPAN 
    class=723571419-20022002>.2</SPAN></SPAN></FONT></FONT></FONT></DIV>
    <DIV><SPAN class=723571419-20022002><SPAN 
    class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
    size=2>a=rtpmap:0 PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV>
    <DIV><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>a=sendrecv</SPAN></FONT></DIV>
    <DIV><SPAN class=723571419-20022002></SPAN><FONT color=#000000><FONT 
    size=2><FONT face=Arial><STRONG>m=audio 980<SPAN 
    class=723571419-20022002>2</SPAN> RTP/AVP&nbsp;<SPAN 
    class=723571419-20022002>99</SPAN><BR></STRONG>c=IN IP4 192.168.0<SPAN 
    class=723571419-20022002>.2</SPAN></FONT></FONT></FONT></DIV>
    <DIV><SPAN class=723571419-20022002><SPAN 
    class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
    size=2>a=rtpmap:99 RFC2833/8000<BR>a=ptime:10</FONT></SPAN></DIV>
    <DIV><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>a=recvonly</SPAN></FONT></DIV></BLOCKQUOTE>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN class=723571419-20022002>ACK 
  :&nbsp; (Really just necessary to confirm the change</SPAN></FONT><SPAN 
  class=723571419-20022002><SPAN class=723571419-20022002></DIV>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <DIV style="MARGIN-RIGHT: 0px"><FONT size=2><FONT color=#000000><FONT 
    face=Arial><SPAN class=723571419-20022002>v=0<BR>o=user1 13723311241683544 
    13723311241683544 IN IP4 192.168.0</SPAN><SPAN 
    class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
    color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>s=session1</SPAN></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>p=+user1</SPAN></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><FONT 
    face=Arial><FONT color=#000000><FONT size=2>t=104100432 
    0<BR><STRONG></FONT></FONT></FONT><FONT color=#000000 face=Arial 
    size=2>m=audio 9900 RTP/AVP 0<BR></FONT></STRONG><FONT size=2><FONT 
    color=#000000><FONT face=Arial>c=IN IP4 192.168.0<SPAN 
    class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></SPAN></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><SPAN 
    class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
    size=2>a=rtpmap:0 PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>a=sendrecv</SPAN></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
    size=2><FONT color=#000000><FONT face=Arial><STRONG>m=audio 980<SPAN 
    class=723571419-20022002>2</SPAN> RTP/AVP&nbsp;<SPAN 
    class=723571419-20022002>99</SPAN><BR></STRONG>c=IN IP4 <SPAN 
    class=723571419-20022002>0.0.0.0</SPAN></FONT></FONT></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><FONT 
    color=#000000 face=Arial size=2>a=rtpmap:99 
    RFC2833/8000<BR>a=ptime:10</FONT></SPAN></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
    class=723571419-20022002>a=sendonly</SPAN></FONT></DIV>
    <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
    color=#000000><FONT face=Arial 
  size=2>&nbsp;&nbsp;</FONT></FONT></DIV></BLOCKQUOTE>
  <DIV></SPAN></SPAN><FONT size=2><FONT face=Arial><SPAN 
  class=723571419-20022002>The latter choice is good in that it can be used 
  for&nbsp;</SPAN><SPAN class=723571419-20022002>other things 
  a</SPAN></FONT><SPAN class=723571419-20022002><FONT face=Arial>s well (e.g. 
  the term</FONT></SPAN><SPAN class=723571419-20022002><FONT face=Arial> party 
  could negotiate in video without having to generate another 
  </FONT></SPAN><SPAN class=723571419-20022002><FONT face=Arial>Offer/Answer 
  exchange)</FONT></SPAN></FONT>&nbsp;<FONT face=Arial size=2><SPAN 
  class=723571419-20022002>. This would also support the conference server case 
  where it would like to negotiate a separate return stream (e.g. G729 in, G711 
  out).</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><STRONG>William 
  Janning</STRONG></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><SPAN 
  class=230570716-21022002>Nortel Networks&nbsp;</SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><SPAN 
  class=230570716-21022002><A 
  href="mailto:billj@nortelnetworks.com">billj@nortelnetworks.com</A></SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=723571419-20022002><A 
  href="mailto:sip:billj@sipfx&nbsp;.com">sip:billj@<SPAN 
  class=230570716-21022002>sipfx&nbsp;</SPAN>.com</A></SPAN></FONT></SPAN></SPAN></SPAN></DIV></SPAN></FONT>
  <BLOCKQUOTE 
  style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"><FONT 
    face=Tahoma size=2>
    <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
      <DIV align=left class=OutlookMessageHeader dir=ltr>-----Original 
      Message-----<BR><B>From:</B> Sen, Sanjoy [NGC:B692:EXCH] <BR><B>Sent:</B> 
      Monday, February 18, 2002 11:52 AM<BR><B>Subject:</B> [VoIP-list] FW: 
      [MMUSIC] summarizing discussion on codec placement in 
      answer<BR><B>Importance:</B> High<BR><BR></DIV></FONT>
      <P><FONT size=2>Do we've any opinion on the interpretation of the ANSWER 
      in the SDP Offer/Answer model (please read Jonathan's email below)? I 
      would particularly appreciate some feedback from our codec 
      experts.</FONT></P>
      <P><FONT size=2>Sanjoy</FONT> </P><BR>
      <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
      Jonathan Rosenberg [<A 
      href="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT> 
      <BR><FONT size=2>Sent: Friday, February 15, 2002 3:32 PM</FONT> <BR><FONT 
      size=2>To: mmusic@ietf.org</FONT> <BR><FONT size=2>Subject: [MMUSIC] 
      summarizing discussion on codec placement in answer</FONT> </P><BR>
      <P><FONT size=2>Folks,</FONT> </P>
      <P><FONT size=2>I wanted to take a moment to summarize where we were in 
      regards to our</FONT> <BR><FONT size=2>long thread on interpretation of 
      the codecs in the answer. The purpose</FONT> <BR><FONT size=2>of the 
      summarization is to put all the points on the table, and</FONT> <BR><FONT 
      size=2>hopefully invite some further comment from folks outside the few 
      thread</FONT> <BR><FONT size=2>participants in the few days which 
      remain.</FONT> </P>
      <P><FONT size=2>ISSUE:</FONT> </P>
      <P><FONT size=2>A sends an offer to B. That offer, for example, is for a 
      single sendrecv</FONT> <BR><FONT size=2>media stream, which lists codecs 
      1,2 and 3. The answerer, B, responds,</FONT> <BR><FONT size=2>and lists 
      codecs 1 and 2. Now, the question is this: can B still send to</FONT> 
      <BR><FONT size=2>A using codec 3? Related to that, can B add codecs in the 
      answer, which</FONT> <BR><FONT size=2>A can then send with?</FONT> </P>
      <P><FONT size=2>Arguments for YES:</FONT> </P>
      <P><FONT size=2>1. A needs to be prepared to receive with codec 3 once it 
      sends the</FONT> <BR><FONT size=2>offer anyway.</FONT> <BR><FONT size=2>2. 
      If A wants to restrict the set based on the answer, it can always</FONT> 
      <BR><FONT size=2>generate a new offer with a reduced set of codecs.</FONT> 
      <BR><FONT size=2>3. SDP has always been interpreted as indicating your own 
      capabilities,</FONT> <BR><FONT size=2>and is therefore an "advertisement" 
      that sticks until updated.</FONT> <BR><FONT size=2>4. There are useful 
      cases for adding codecs in the answer, particularly,</FONT> <BR><FONT 
      size=2>support for rfc2833. Thus, lets say A is an end user which can send 
      rfc</FONT> <BR><FONT size=2>2833, but not receive it. So, when it sends an 
      INVITE, it doesn't</FONT> <BR><FONT size=2>include rfc2833 in the INVITE. 
      The server that responds is a media</FONT> <BR><FONT size=2>server, which 
      supports rfc2833. So, it includes it in the answer. Now, A</FONT> 
      <BR><FONT size=2>can actually send with it during the middle of the 
      session, and it will</FONT> <BR><FONT size=2>never receive it (since it 
      didn't advertise it). This provides a useful</FONT> <BR><FONT 
      size=2>function that would otherwise require FID.</FONT> </P>
      <P><FONT size=2>Arguments for NO:</FONT> </P>
      <P><FONT size=2>1. The purpose of the O/A model is to agree on shared 
      views of certain</FONT> <BR><FONT size=2>session parameters, such as the 
      directionality (send/recv/sendrecv).</FONT> <BR><FONT size=2>Codecs fit 
      into that category as well, since you are already required to</FONT> 
      <BR><FONT size=2>indicate, in the answer, a set that partially overlaps 
      with the offer.</FONT> <BR><FONT size=2>Furthermore, the goal with codecs 
      is to have the same in both</FONT> <BR><FONT size=2>directions, generally, 
      so that this would speak to them being a shared</FONT> <BR><FONT 
      size=2>state that is negotiated.</FONT> </P>
      <P><FONT size=2>2. Many people have interpreted the SDP/bis/rfc2543 drafts 
      this way. If</FONT> <BR><FONT size=2>you go for YES, there may be interop 
      issues. The converse is not true,</FONT> <BR><FONT size=2>of 
      course.</FONT> </P>
      <P><FONT size=2>3. It is very useful to narrow down the codec sets in the 
      answer, and we</FONT> <BR><FONT size=2>should not always require an extra 
      exchange of O/A just to do that.</FONT> </P><BR><BR>
      <P><FONT size=2>I took a private poll amongst some implementors. The 
      initial results</FONT> <BR><FONT size=2>showed that most assumed the YES 
      case. Some more mails since then</FONT> <BR><FONT size=2>indicate its a 
      bit more split, although I think somewhat in favor of YES</FONT> <BR><FONT 
      size=2>still.</FONT> </P>
      <P><FONT size=2>I am somewhat on the fence; I think NO is more correct 
      based on</FONT> <BR><FONT size=2>offer/answer, although YES would be the 
      SDP purist view. What I really</FONT> <BR><FONT size=2>want is more 
      opinions on this one....</FONT> </P>
      <P><FONT size=2>THanks,</FONT> <BR><FONT size=2>Jonathan R.</FONT> </P>
      <P><FONT size=2>-- </FONT><BR><FONT size=2>Jonathan D. Rosenberg, 
      Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 72 
      Eagle Rock Avenue</FONT> <BR><FONT size=2>Chief 
      Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
      First Floor</FONT> <BR><FONT 
      size=2>dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
      East Hanover, NJ 07936</FONT> <BR><FONT 
      size=2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
      FAX: (973) 952-5050</FONT> <BR><FONT size=2><A 
      href="http://www.jdrosen.net" 
      target=_blank>http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
      PH:&nbsp; (973) 952-5000</FONT> <BR><FONT size=2><A 
      href="http://www.dynamicsoft.com" 
      target=_blank>http://www.dynamicsoft.com</A></FONT> </P>
      <P><FONT size=2>_______________________________________________</FONT> 
      <BR><FONT size=2>mmusic mailing list</FONT> <BR><FONT 
      size=2>mmusic@ietf.org</FONT> <BR><FONT size=2><A 
      href="https://www1.ietf.org/mailman/listinfo/mmusic" 
      target=_blank>https://www1.ietf.org/mailman/listinfo/mmusic</A></FONT> 
    </P></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Date: Thu, 21 Feb 2002 22:17:43 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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FYI, I had a conversation with Joerg early yesterday morning about this,
and the proposal which I sent out is the one reflected in
offer-answer-01.

Thanks,
Jonathan R.

Joerg Ott wrote:
> 
> Folks,
> 
> we need to reach closure on this issue fairly shortly.
> Having gone through the mailing list discussion, we believe
> that we can narrow down Jonathan's summary a bit further
> and come up with a concrete proposal to beat up:
> 
> Restating the issue:
> 
> > A sends an offer to B. That offer, for example, is for a single
> sendrecv
> > media stream, which lists codecs 1,2 and 3. The answerer, B, responds,
> > and lists codecs 1 and 2. Now, the question is this: can B still send
> to
> > A using codec 3? Related to that, can B add codecs in the answer,
> which
> > A can then send with?
> 
> Let us keep in mind that we are not designing offer/answer for telephony
> but that telephony is rather one (maybe not unimportant) special case of
> multimedia communications.  Let us also keep in mind that we should
> optimize for the more common cases but at the same time not lose
> generality.  And let us not forget that SIP is just *one* customer of
> offer/answer.
> 
> Let us assume for a moment that we follow the "NO" proposal (and assume
> further that this approach does not break interoperability/backward
> compatility):
> 
> 1. This idea allows us to narrow down codecs in an efficient fashion.
> 
>    This helps people who do not want to require their implementations
>    to listen for multiple codecs simultaneously (e.g. because of
>    limited DSP resources or difficulties in implementing MCUs).
> 
>    The counter-argument is that such device need to be able to support
>    multiple codecs in the beginning anyway because of the race condition
>    between the answer and the first RTP packet.  However, this can be
>    circumvent this by simply waiting for the answer before accepting
>    media at the risk of clipping.  But people may be able to architected
>    their network in a way that it does not delay the answer unncessarily
>    and thus avoid clipping in the first place).  Alternatively, they may
>    guess which type of codec to expect or only load the decoder upon
>    arrival of the first packet (assuming this is not going to change) at
>    the expense of some additional delay, etc.
> 
>    Looks like there are sufficient alternatives that clipping can be
>    considered a non-issue.  (Please note that if you are doing security
>    handshakes and QoS preconditions or similar stuff, you are not going
>    to worry about this type of clipping anyway -- but rather about post-
>    dial delay.)
> 
> 2. Expanding the codec set can be done by means of an extra offer in
>    the opposite direction.  This is an extra exchange but we would
> assume
>    this to be the less common case (and thus not optimize for it).
> 
>    The only application scenario for which this is really needed people
>    have mentioned so far is DTMF -- more specifically, combining a
>    sendrecv and an sendonly/recvonly media stream.  Please share ideas
>    for others with us.
> 
>    We do acknowledge that this scenario is important and need support.
>    But we also judge that this is not the common case and need not be
>    optimized for at all cost if a cleaner solution exists.
> 
>    This cleaner solution can be based upon FID.  In the simplest case,
>    A uses FID from the very beginning:
> 
>    A ----  INVITE   ----> B
>        a=group:FID 1 2
>        m=audio pt=0
>        a=mid:1
>        m=audio pt=97
>        a=rtpmap:97 telephone-events
>        a=mid:2
>        a=sendonly
> 
>    and B can then decide whether or not to reject telephone events.
> 
>    However, A may or may not understand FID and may or may not indicate
>    its send capabilities as described above.  Then we get the following
>    simple exchange for A calling B:
> 
>    A ---- INVITE ----> B
>           m=audio pt=0
> 
>    A <--- 200 OK ----- B
>           m=audio pt=0
>    A ----  ACK   ----> B
> 
>    Now B can indicate its receive caps using FID doing a re-INVITE.
> 
>    A <--- INVITE ----- B
>           a=group:FID 1 2
>           m=audio pt=0
>           a=mid:1
>           m=audio pt=97
>           a=rtpmap:97 telephone-events
>           a=mid:2
>           a=recvonly
> 
>    ...
> 
>    Again, we can properly express the intended semantics.  If A does not
>    support FID it ignores the attributes thus yielding the following
>    in A's perception:
> 
>    A <--- INVITE ----- B
>           m=audio pt=0
>           m=audio pt=97
>           a=rtpmap:97 telephone-events
> 
>    ...
> 
>    This is the alternative that can be pursued without FID (and, at the
>    same time serves as fallback).  According to the rules defined in
>    offer/answer for local processing of multiple streams of the same
>    type with one local source or sink, this should just work.  The
> sender
>    copies the input to all streams if the data can be mapped (i.e. only
>    events go across the second media session), the receiver mixes the
>    the incoming data streams locally as appropriate.  In a gateway this
>    may sometimes mean to locally generate DTMF tones and mix them with
>    audio signals, sometimes mean to locally sink the DTMF and process
> it.
> 
>    Jonathan, is this interpretation correct?  Is there need for
> clarifying
>    text on this issue regarding offer/answer?
> 
>    The major issue to discuss here is whether or not there is an extra
> RTT
>    required for this signaling (or whether other SIP methods can be used
>    to achieve this more quickly) -- and whether the additional delay
>    (whatever this may be) is an issue for DTMF.
> 
>    Thoughts?
> 
> Is this a viable approach ahead?
> 
> Note that we are not religious about this proposal -- but it seems to be
> a
> reasonable way ahead taking into account some general design principles.
> There may be other ways and if better ones are found, such proposals are
> welcome.  But they need to be found NOW.
> 
> A small disclaimer in the end: probably we should be taking the more
> purist
> SDP view (as Jonathan calles the "YES" approach) -- but since SDP is
> used
> in a way it was not supposed to be anyway the NO method may not cause
> too
> much *extra* headache.
> 
> Colin and Joerg

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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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		: An Offer/Answer Model with SDP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-mmusic-sdp-offer-answer-01.txt
	Pages		: 26
	Date		: 21-Feb-02
	
This document defines a mechanism by which two entities can make use
of SDP to arrive at a common view of a multimedia session between
them. In the model, one participant offers the other a description of
the desired session from their perspective, and the other participant
answers with the desired session from their perspective. This
offer/answer model is most useful in unicast sessions where
information from both participants is needed for the complete view of
the session. The offer/answer model is used by protocols like the
Session Initiation Protocol (SIP).

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From mmusic-admin@ietf.org  Fri Feb 22 11:15:22 2002
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The IESG has received a request from the Session Initiation Protocol
Working Group and the Multiparty Multimedia Session Control Working
Group to consider publication of the following Internet-Drafts as
Proposed Standards:


 o SIP: Session Initiation Protocol
	<draft-ietf-sip-rfc2543bis-08.txt>

 o Reliability of Provisional Responses in SIP 
	<draft-ietf-sip-100rel-05.txt> 

 o SIP: Locating SIP Servers
	<draft-ietf-sip-srv-05.txt> 

 o SIP-Specific Event Notification
	<draft-ietf-sip-events-03.txt> 

 o An Offer/Answer Model with SDP
	<draft-ietf-mmusic-sdp-offer-answer-01.txt>

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by March 8, 2002.

Files can be obtained via
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http://www.ietf.org/internet-drafts/draft-ietf-sip-events-03.txt
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-offer-answer-01.txt

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From mmusic-admin@ietf.org  Fri Feb 22 14:02:38 2002
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        Jonathan Rosemberg <jdrosen@dynamicsoft.com>, wtm@research.att.com
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Subject: [MMUSIC] Removing security preconditions from manyfolks
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Hi,

we are thinking of removing security preconditions from manyfolks. Right
now manyfolks handles QoS and security preconditions. However, with
MIKEY, this feature seems useless to establish secure channels.

If somebody thinks that removing security preconditions from manyfolks
is a bad idea, please speak up.

Regards,

Gonzalo
-- 
Gonzalo Camarillo                    Phone :   +1 212 939 71 71
Columbia University                  Mobile:  +358 40 702 35 35
472 Computer Science Building        Fax   :  +358  9 299 30 52
1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
New York, NY 10027                   
USA                              Gonzalo.Camarillo@ericsson.com

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From mmusic-admin@ietf.org  Fri Feb 22 16:35:55 2002
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Subject: [MMUSIC] Using 0.0.0.0 for placing a call on hold
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I think, we should specify, that the port for the stream which was put on hold using "0.0.0.0" address should be an arbitrary non-zero value in order for it not to be confused with removed streams. Additionally, note about using "0.0.0.0" address in initial offer should probably belong to 5.1 and not 8.4.
___________________________________
Roman Shpount, VP of Technology
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From mmusic-admin@ietf.org  Sat Feb 23 09:56:30 2002
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From: "William Janning"<billj@nortelnetworks.com>
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Subject: RE:[MMUSIC] Mulit-codec Response Offer/Answer Issue
Date: Sat, 23 Feb 2002 08:26:26 -0600
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This is re: Jonathan Rosenberg's question on codec selection. This may be a
double post (I'm not sure it got through the first time)
 
-----------------------------------------------
 
NO is definitely the answer. 
 
When "A" sends a request with SDP, it is saying "I CAN do 1,2,3". When "B"
responds, it is saying "I WILL do 1, 2, or 3". In reality it makes no sense
for "B" to respond with "I WILL do 1 and 2" as practically speaking this
would mean B could dynamically switch between receiving codec 1 or 2 at the
whim of A, or that B would be able to properly mix 1 & 2 to the transducer.
If this is negotiation, then let us CHOOSE.
 
There is an assumption in codec selection that the codec "B" elects to
receive is also the one it is sending (so it must pick from "A"'s initial
list). This is implied in the SDP protocol because a single media
description only negotiates a single codec - there is not provision to
identify the send codec from the receive codec. This fact and the above
practicality are why NO is the answer with even additional restrictions (B
should never reply with multiple codecs). 
 
Supporting the codec capability preference for a particular session for
clients is completely related to bandwidth management and perceived voice
quality - though mostly bandwidth management. If everybody could do G711 or
G722, then they would. Period. So if I say I will receive G729, then there's
no way I'm going to send G711 or even G723. Likewise I can't say I'll
receive G711 (because I have higher download bandwidth) but send G729
(because my upstream is limited). "A" would have no clue I'm doing so
because all I replied in my response was G711. 
 
There is no such thing as a "send only" or "receive only" codec - codecs
always support both encoding/decoding. However if I did want to specify
unique send and receive codecs, then the use of two media descriptions could
be used as such:
 
v=0
o=user 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user
t=104100432 0
m=audio 9900 RTP/AVP 0
c=IN IP4 192.168.0.1
a=rtpmap:0 PCMU/8000
a=ptime:30
a=recvonly
m=audio 9902 RTP/AVP 4 18
c=IN IP4 192.168.0.1
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=ptime:10
a=sendonly

 
I would argue that the attributes "sendonly, recvonly, sendrecv" are
MANDATORY in messages and reflect the "I WILL" versus "I CAN", though the
difference is also identified by "response" versus "request" message. If the
originator wanted to tell the far-end that it was receiving already in the
request, then it MUST use the recvonly attribute (though this is a security
risk - since the originator doesn't know who the far-end is supposed to be,
anybody can be sending the originator packets which they'd happily play
until the real far-end answers (assuming of course there was no early media
established like on a 183). 
 
In order to really handle multiple m= lines with the same media type,
something like s= lines should also be per media description (or another
field added to the m= line to uniquely identify the media description). This
would eliminate m= ordering requirements and facilitate better renegotiation
management. Maybe FID helps here (I need to understand the purpose of FID
better). 
 
---------------------------------------------------

Arguments for YES: 

1. A needs to be prepared to receive with codec 3 once it sends the 
offer anyway. 

It is a security risk for A to receive RTP before it knows about the
far-end. "A' should not receive anything until it receives an SDP response.
A should not "play" received RTP unless it comes from the address indicated
in the far-end's media description (one reason why c= lines should always be
per media description, not session description). 


2. If A wants to restrict the set based on the answer, it can always 
generate a new offer with a reduced set of codecs. 

B can do likewise. Once the Offer-Answer completes, SDP is set until another
Offer-Answer is initiated by either side.


3. SDP has always been interpreted as indicating your own capabilities, 
and is therefore an "advertisement" that sticks until updated. 

We must clearly delineate when we are broadcasting our capabilities versus
when we are choosing for session establishment. At some point in time we
have to CHOOSE. Advertising your capabilities should be done via some other
mechanism like OPTIONS.


4. There are useful cases for adding codecs in the answer, particularly, 
support for rfc2833. Thus, lets say A is an end user which can send rfc 
2833, but not receive it. So, when it sends an INVITE, it doesn't 
include rfc2833 in the INVITE. The server that responds is a media 
server, which supports rfc2833. So, it includes it in the answer. Now, A 
can actually send with it during the middle of the session, and it will 
never receive it (since it didn't advertise it). This provides a useful 
function that would otherwise require FID. 

Either : a) secondary codec's are uniquely flagged in the media description
for a specific purpose (like 2833), b) multiple m= lines are used and the
response is allowed to negotiate in capabilities which would then be
responded to in the ACK. For instance:

Request :

v=0
o=user1 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user1
t=104100432 0
m=audio 9900 RTP/AVP 4 18 0
c=IN IP4 192.168.0.1
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=ptime:30

Response :

v=0
o=user2 13723311241683545 13723311241683545 IN IP4 192.168.0.2
s=session1
p=+user2
t=104100452 0
m=audio 9800 RTP/AVP 0
c=IN IP4 192.168.0.2
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
m=audio 9802 RTP/AVP 99
c=IN IP4 192.168.0.2
a=rtpmap:99 RFC2833/8000
a=ptime:10
a=recvonly

ACK :  (Really just necessary to confirm the change

v=0
o=user1 13723311241683544 13723311241683544 IN IP4 192.168.0.1
s=session1
p=+user1
t=104100432 0
m=audio 9900 RTP/AVP 0
c=IN IP4 192.168.0.1
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
m=audio 9802 RTP/AVP 99
c=IN IP4 0.0.0.0
a=rtpmap:99 RFC2833/8000
a=ptime:10
a=sendonly
  

The latter choice is good in that it can be used for other things as well
(e.g. the term party could negotiate in video without having to generate
another Offer/Answer exchange) . This would also support the conference
server case where it would like to negotiate a separate return stream (e.g.
G729 in, G711 out).
 
William Janning
Nortel Networks 
billj@nortelnetworks.com <mailto:billj@nortelnetworks.com> 
sip:billj@ <mailto:sip:billj@sipfx .com> sipfx .com



-----Original Message-----
From: Sen, Sanjoy [NGC:B692:EXCH] 
Sent: Monday, February 18, 2002 11:52 AM
Subject: FW: [MMUSIC] summarizing discussion on codec placement in answer
Importance: High


Do we've any opinion on the interpretation of the ANSWER in the SDP
Offer/Answer model (please read Jonathan's email below)? I would
particularly appreciate some feedback from our codec experts.

Sanjoy 


-----Original Message----- 
From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com
<mailto:jdrosen@dynamicsoft.com> ] 
Sent: Friday, February 15, 2002 3:32 PM 
To: mmusic@ietf.org 
Subject: [MMUSIC] summarizing discussion on codec placement in answer 


Folks, 

I wanted to take a moment to summarize where we were in regards to our 
long thread on interpretation of the codecs in the answer. The purpose 
of the summarization is to put all the points on the table, and 
hopefully invite some further comment from folks outside the few thread 
participants in the few days which remain. 

ISSUE: 

A sends an offer to B. That offer, for example, is for a single sendrecv 
media stream, which lists codecs 1,2 and 3. The answerer, B, responds, 
and lists codecs 1 and 2. Now, the question is this: can B still send to 
A using codec 3? Related to that, can B add codecs in the answer, which 
A can then send with? 

Arguments for YES: 

1. A needs to be prepared to receive with codec 3 once it sends the 
offer anyway. 
2. If A wants to restrict the set based on the answer, it can always 
generate a new offer with a reduced set of codecs. 
3. SDP has always been interpreted as indicating your own capabilities, 
and is therefore an "advertisement" that sticks until updated. 
4. There are useful cases for adding codecs in the answer, particularly, 
support for rfc2833. Thus, lets say A is an end user which can send rfc 
2833, but not receive it. So, when it sends an INVITE, it doesn't 
include rfc2833 in the INVITE. The server that responds is a media 
server, which supports rfc2833. So, it includes it in the answer. Now, A 
can actually send with it during the middle of the session, and it will 
never receive it (since it didn't advertise it). This provides a useful 
function that would otherwise require FID. 

Arguments for NO: 

1. The purpose of the O/A model is to agree on shared views of certain 
session parameters, such as the directionality (send/recv/sendrecv). 
Codecs fit into that category as well, since you are already required to 
indicate, in the answer, a set that partially overlaps with the offer. 
Furthermore, the goal with codecs is to have the same in both 
directions, generally, so that this would speak to them being a shared 
state that is negotiated. 

2. Many people have interpreted the SDP/bis/rfc2543 drafts this way. If 
you go for YES, there may be interop issues. The converse is not true, 
of course. 

3. It is very useful to narrow down the codec sets in the answer, and we 
should not always require an extra exchange of O/A just to do that. 



I took a private poll amongst some implementors. The initial results 
showed that most assumed the YES case. Some more mails since then 
indicate its a bit more split, although I think somewhat in favor of YES 
still. 

I am somewhat on the fence; I think NO is more correct based on 
offer/answer, although YES would be the SDP purist view. What I really 
want is more opinions on this one.... 

THanks, 
Jonathan R. 

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue 
Chief Scientist                         First Floor 
dynamicsoft                             East Hanover, NJ 07936 
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050 
http://www.jdrosen.net <http://www.jdrosen.net>                   PH:  (973)
952-5000 
http://www.dynamicsoft.com <http://www.dynamicsoft.com>  

_______________________________________________ 
mmusic mailing list 
mmusic@ietf.org 
https://www1.ietf.org/mailman/listinfo/mmusic
<https://www1.ietf.org/mailman/listinfo/mmusic>  


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<DIV align=left class=OutlookMessageHeader dir=ltr 
style="MARGIN-RIGHT: 0px"><FONT face=Tahoma size=2><SPAN 
class=939372714-23022002>This is re: Jonathan Rosenberg's question on codec 
selection. This may be a double post (I'm not sure it got through the first 
time)</SPAN></FONT></DIV>
<DIV align=left class=OutlookMessageHeader dir=ltr 
style="MARGIN-RIGHT: 0px"><FONT face=Tahoma size=2><SPAN 
class=939372714-23022002></SPAN></FONT>&nbsp;</DIV>
<DIV align=left class=OutlookMessageHeader dir=ltr 
style="MARGIN-RIGHT: 0px"><FONT face=Tahoma size=2><SPAN 
class=939372714-23022002>-----------------------------------------------</SPAN></FONT></DIV>
<DIV align=left class=OutlookMessageHeader dir=ltr 
style="MARGIN-RIGHT: 0px"><FONT face=Tahoma size=2><SPAN 
class=939372714-23022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>NO is definitely the answer. </SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>When "A" sends a request with SDP, it is saying "I CAN 
do 1,2,3". When "B" responds, it is saying "I WILL do 1, 2, or 3". In reality it 
makes no sense for "B" to respond with "I WILL do 1 and 2" as practically 
speaking this would mean B could dynamically switch between receiving codec 1 or 
2 at the whim of A, or that B would be able to properly mix 1 &amp; 2 to the 
transducer. If this is negotiation, then let us CHOOSE.</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>There is an assumption in codec selection that the 
codec&nbsp;"B"&nbsp;elects to receive is also the one it is sending (so it must 
pick from "A"'s initial list). This is implied in the SDP protocol because a 
single media description only negotiates a single codec - there is not provision 
to identify the send codec from the receive codec. This fact and the above 
practicality are why NO is the answer with even additional restrictions 
(B&nbsp;should never reply with multiple codecs). </SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>Supporting the codec capability preference for a 
particular session for clients is completely related to bandwidth management and 
perceived voice quality - though mostly bandwidth management. If everybody could 
do G711 or G722, then they would. Period.&nbsp;So if I say I will receive G729, 
then there's no way I'm going to send G711 or even G723. Likewise I can't say 
I'll receive G711 (because I have higher download bandwidth) but send G729 
(because my upstream is limited). "A" would have no clue I'm doing so 
because&nbsp;all I replied in my response was&nbsp;G711. </SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT><FONT color=#0000ff face=Arial 
size=2><SPAN class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>There is no such thing as a "send only" or "receive 
only" codec - codecs always support both encoding/decoding. However if I did 
want to specify unique send and receive codecs, then the use of two media 
descriptions could be used as such:</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>v=0<BR>o=user 13723311241683544 13723311241683544 IN 
IP4 192.168.0</SPAN></FONT><FONT face=Arial size=2><SPAN 
class=723571419-20022002>.1</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>s=session1</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>p=+user</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>t=104100432 0<BR><STRONG>m=audio </STRONG>9900 RTP/AVP 
0<BR>c=IN IP4 192.168.0<FONT face=Arial size=2><SPAN 
class=723571419-20022002>.1</SPAN></FONT></SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><FONT face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>a=rtpmap:0 
PCMU/8000<BR>a=ptime:30</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><STRONG>a=recvonly<BR>m=audio </STRONG>9902 RTP/AVP 4 
18</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>c=IN IP4 192.168.0<FONT face=Arial size=2><SPAN 
class=723571419-20022002>.1</SPAN></FONT></SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><FONT face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>a=rtpmap:4 G723/8000<BR>a=rtpmap:18 
G729/8000<BR>a=ptime:10</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><STRONG><SPAN 
class=723571419-20022002>a=sendonly<BR></SPAN></STRONG></FONT></DIV></SPAN></FONT>
<DIV style="MARGIN-RIGHT: 0px">&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>I would argue that the attributes "sendonly, recvonly, 
sendrecv" are MANDATORY in messages and reflect the "I WILL" versus "I CAN", 
though the difference is also identified by "response" versus "request" message. 
If the originator wanted to tell the far-end that it was receiving already in 
the request, then it MUST use the recvonly attribute (though this is a security 
risk&nbsp;- since the originator doesn't know who the far-end is supposed to be, 
anybody can be sending the originator packets which they'd happily play until 
the real far-end answers (assuming of course there was no early media 
established like on a 183). </SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT><FONT color=#0000ff face=Arial 
size=2><SPAN class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT size=2><FONT color=#0000ff><FONT 
face=Arial><SPAN class=723571419-20022002>In order to really handle multiple m= 
lines with the same media type, something like s= lines should also be per media 
description (or another field added to the m= line to uniquely identify the 
media description). This would eliminate m= ordering requirements and 
facilitate&nbsp;better renegotiation management<SPAN class=939372714-23022002>. 
Maybe FID helps here (I need to understand the purpose of FID 
better).&nbsp;</SPAN></SPAN></FONT></FONT></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#0000ff face=Arial size=2><SPAN 
class=723571419-20022002>---------------------------------------------------</SPAN></FONT></DIV><FONT 
color=#0000ff><SPAN class=723571419-20022002>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <P><FONT face=Arial><FONT color=#000000 size=2>Arguments for YES: 
  </FONT></FONT></P>
  <P><FONT face=Arial><FONT color=#000000 size=2>1. A needs to be prepared to 
  receive with codec 3 once it sends the <BR>offer anyway. 
</FONT></FONT></P></BLOCKQUOTE>
<P style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>It is a security risk for A to receive RTP before it 
knows about the far-end. "A' should not receive anything until it receives an 
SDP response. A should not "play" received RTP unless it comes from the address 
indicated in the far-end's media description (one reason why c= lines should 
always be per media description, not session description). </SPAN></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT><FONT color=#000000><BR><FONT face=Arial 
size=2>2. If A wants to restrict the set based on the answer, it can always 
<BR>generate a new offer with a reduced set of codecs. </FONT></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT size=2><FONT face=Arial><FONT 
color=#000000><SPAN class=723571419-20022002><FONT color=#0000ff>B can do 
likewise. Once the Offer-Answer completes, SDP is set until another Offer-Answer 
is initiated by either side.</FONT></SPAN></FONT></FONT></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT size=2><FONT face=Arial><FONT 
color=#000000><SPAN class=723571419-20022002></SPAN><BR>3. SDP has always been 
interpreted as indicating your own capabilities, <BR>and is therefore an 
"advertisement" that sticks until updated. </FONT></FONT></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT size=2><FONT face=Arial><FONT 
color=#000000><SPAN class=723571419-20022002><FONT color=#0000ff>We must clearly 
delineate when we are broadcasting our capabilities versus when we are choosing 
for session establishment. At some point in time we have to CHOOSE. Advertising 
your capabilities should be done via some other mechanism like 
OPTIONS.</FONT></SPAN></FONT></FONT></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT size=2><FONT face=Arial><FONT 
color=#000000><SPAN class=723571419-20022002></SPAN><BR>4. There are useful 
cases for adding codecs in the answer, particularly, <BR>support for rfc2833. 
Thus, lets say A is an end user which can send rfc <BR>2833, but not receive it. 
So, when it sends an INVITE, it doesn't <BR>include rfc2833 in the INVITE. The 
server that responds is a media <BR>server, which supports rfc2833. So, it 
includes it in the answer. Now, A <BR>can actually send with it during the 
middle of the session, and it will <BR>never receive it (since it didn't 
advertise it). This provides a useful <BR>function that would otherwise require 
FID. </FONT></FONT></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002>Either : a) secondary codec's are uniquely flagged in 
the media description for a specific purpose (like 2833), b) multiple m= lines 
are used and the response is allowed to negotiate in capabilities which would 
then be responded to in the ACK. For instance:</SPAN></FONT></P>
<P style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
class=723571419-20022002>Request :</SPAN></FONT></P><SPAN 
class=723571419-20022002>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV><FONT size=2><FONT face=Arial><FONT color=#000000><SPAN 
  class=723571419-20022002>v=0<BR>o=user1 13723311241683544 13723311241683544 IN 
  IP4 192.168.0</SPAN><SPAN 
  class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></DIV>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>s=session1</SPAN></FONT></DIV>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>p=+user1</SPAN></FONT></DIV>
  <DIV><FONT color=#000000><SPAN class=723571419-20022002><FONT face=Arial 
  size=2>t=104100432 0<BR></FONT><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002></SPAN><FONT size=2><FONT face=Arial><STRONG>m=audio 
  9900 RTP/AVP 4 18 0<BR></STRONG>c=IN IP4 192.168.0<SPAN 
  class=723571419-20022002>.1</SPAN></FONT></FONT></SPAN></SPAN></FONT></DIV>
  <DIV><FONT color=#000000><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002><SPAN class=723571419-20022002></SPAN><FONT 
  face=Arial size=2>a=rtpmap:4 G723/8000<BR>a=rtpmap:18 
  G729/8000<BR></FONT></SPAN></SPAN></FONT><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
  size=2>a=rtpmap:0 PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV></BLOCKQUOTE>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
class=723571419-20022002>Response :</SPAN></FONT><SPAN 
class=723571419-20022002><SPAN class=723571419-20022002></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV><FONT color=#000000><FONT size=2><FONT face=Arial><SPAN 
  class=723571419-20022002>v=0<BR>o=user2 13723311241683545 13723311241683545 IN 
  IP4 192.168.0</SPAN><SPAN 
  class=723571419-20022002>.2</SPAN></FONT></FONT></FONT></DIV>
  <DIV><SPAN class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
  size=2><SPAN class=723571419-20022002>s=session1</SPAN></FONT></DIV>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>p=+user2</SPAN></FONT></DIV>
  <DIV><FONT color=#000000><FONT size=2><FONT face=Arial><SPAN 
  class=723571419-20022002>t=104100452 0<BR><STRONG>m=audio 9800 RTP/AVP 
  0<BR></STRONG>c=IN IP4 192.168.0<SPAN 
  class=723571419-20022002>.2</SPAN></SPAN></FONT></FONT></FONT></DIV>
  <DIV><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
  size=2>a=rtpmap:0 PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>a=sendrecv</SPAN></FONT></DIV>
  <DIV><SPAN class=723571419-20022002></SPAN><FONT color=#000000><FONT 
  size=2><FONT face=Arial><STRONG>m=audio 980<SPAN 
  class=723571419-20022002>2</SPAN> RTP/AVP&nbsp;<SPAN 
  class=723571419-20022002>99</SPAN><BR></STRONG>c=IN IP4 192.168.0<SPAN 
  class=723571419-20022002>.2</SPAN></FONT></FONT></FONT></DIV>
  <DIV><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
  size=2>a=rtpmap:99 RFC2833/8000<BR>a=ptime:10</FONT></SPAN></DIV>
  <DIV><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>a=recvonly</SPAN></FONT></DIV></BLOCKQUOTE>
<DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
class=723571419-20022002>ACK :&nbsp; (Really just necessary to confirm the 
change</SPAN></FONT><SPAN class=723571419-20022002><SPAN 
class=723571419-20022002></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV style="MARGIN-RIGHT: 0px"><FONT size=2><FONT color=#000000><FONT 
  face=Arial><SPAN class=723571419-20022002>v=0<BR>o=user1 13723311241683544 
  13723311241683544 IN IP4 192.168.0</SPAN><SPAN 
  class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
  color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>s=session1</SPAN></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>p=+user1</SPAN></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><FONT 
  face=Arial><FONT color=#000000><FONT size=2>t=104100432 
  0<BR><STRONG></FONT></FONT></FONT><FONT color=#000000 face=Arial 
  size=2>m=audio 9900 RTP/AVP 0<BR></FONT></STRONG><FONT size=2><FONT 
  color=#000000><FONT face=Arial>c=IN IP4 192.168.0<SPAN 
  class=723571419-20022002>.1</SPAN></FONT></FONT></FONT></SPAN></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><SPAN 
  class=723571419-20022002></SPAN><FONT color=#000000 face=Arial 
  size=2>a=rtpmap:0 PCMU/8000<BR>a=ptime:30</FONT></SPAN></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>a=sendrecv</SPAN></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
  size=2><FONT color=#000000><FONT face=Arial><STRONG>m=audio 980<SPAN 
  class=723571419-20022002>2</SPAN> RTP/AVP&nbsp;<SPAN 
  class=723571419-20022002>99</SPAN><BR></STRONG>c=IN IP4 <SPAN 
  class=723571419-20022002>0.0.0.0</SPAN></FONT></FONT></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002><FONT 
  color=#000000 face=Arial size=2>a=rtpmap:99 
  RFC2833/8000<BR>a=ptime:10</FONT></SPAN></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><FONT color=#000000 face=Arial size=2><SPAN 
  class=723571419-20022002>a=sendonly</SPAN></FONT></DIV>
  <DIV style="MARGIN-RIGHT: 0px"><SPAN class=723571419-20022002></SPAN><FONT 
  color=#000000><FONT face=Arial 
size=2>&nbsp;&nbsp;</FONT></FONT></DIV></BLOCKQUOTE>
<DIV style="MARGIN-RIGHT: 0px"></SPAN></SPAN><FONT size=2><FONT face=Arial><SPAN 
class=723571419-20022002>The latter choice is good in that it can be used 
for&nbsp;</SPAN><SPAN class=723571419-20022002>other things a</SPAN></FONT><SPAN 
class=723571419-20022002><FONT face=Arial>s well (e.g. the 
term</FONT></SPAN><SPAN class=723571419-20022002><FONT face=Arial> party could 
negotiate in video without having to generate another </FONT></SPAN><SPAN 
class=723571419-20022002><FONT face=Arial>Offer/Answer 
exchange)</FONT></SPAN></FONT>&nbsp;<FONT face=Arial size=2><SPAN 
class=723571419-20022002>. This would also support the conference server case 
where it would like to negotiate a separate return stream (e.g. G729 in, G711 
out).</SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002></SPAN></FONT>&nbsp;</DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><STRONG>William Janning</STRONG></SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><SPAN class=230570716-21022002>Nortel 
Networks&nbsp;</SPAN></SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><SPAN class=230570716-21022002><A 
href="mailto:billj@nortelnetworks.com">billj@nortelnetworks.com</A></SPAN></SPAN></FONT></DIV>
<DIV style="MARGIN-RIGHT: 0px"><FONT face=Arial size=2><SPAN 
class=723571419-20022002><A 
href="mailto:sip:billj@sipfx&nbsp;.com">sip:billj@<SPAN 
class=230570716-21022002>sipfx&nbsp;</SPAN>.com</A></SPAN></FONT></SPAN></SPAN></SPAN></DIV></SPAN></FONT>
<BLOCKQUOTE 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"><FONT 
  face=Tahoma size=2>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <DIV align=left class=OutlookMessageHeader dir=ltr>-----Original 
    Message-----<BR><B>From:</B> Sen, Sanjoy [NGC:B692:EXCH] <BR><B>Sent:</B> 
    Monday, February 18, 2002 11:52 AM<BR><B>Subject:</B> FW: [MMUSIC] 
    summarizing discussion on codec placement in answer<BR><B>Importance:</B> 
    High<BR></DIV></FONT>
    <P><FONT size=2>Do we've any opinion on the interpretation of the ANSWER in 
    the SDP Offer/Answer model (please read Jonathan's email below)? I would 
    particularly appreciate some feedback from our codec experts.</FONT></P>
    <P><FONT size=2>Sanjoy</FONT> </P><BR>
    <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
    Jonathan Rosenberg [<A 
    href="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT> 
    <BR><FONT size=2>Sent: Friday, February 15, 2002 3:32 PM</FONT> <BR><FONT 
    size=2>To: mmusic@ietf.org</FONT> <BR><FONT size=2>Subject: [MMUSIC] 
    summarizing discussion on codec placement in answer</FONT> </P><BR>
    <P><FONT size=2>Folks,</FONT> </P>
    <P><FONT size=2>I wanted to take a moment to summarize where we were in 
    regards to our</FONT> <BR><FONT size=2>long thread on interpretation of the 
    codecs in the answer. The purpose</FONT> <BR><FONT size=2>of the 
    summarization is to put all the points on the table, and</FONT> <BR><FONT 
    size=2>hopefully invite some further comment from folks outside the few 
    thread</FONT> <BR><FONT size=2>participants in the few days which 
    remain.</FONT> </P>
    <P><FONT size=2>ISSUE:</FONT> </P>
    <P><FONT size=2>A sends an offer to B. That offer, for example, is for a 
    single sendrecv</FONT> <BR><FONT size=2>media stream, which lists codecs 1,2 
    and 3. The answerer, B, responds,</FONT> <BR><FONT size=2>and lists codecs 1 
    and 2. Now, the question is this: can B still send to</FONT> <BR><FONT 
    size=2>A using codec 3? Related to that, can B add codecs in the answer, 
    which</FONT> <BR><FONT size=2>A can then send with?</FONT> </P>
    <P><FONT size=2>Arguments for YES:</FONT> </P>
    <P><FONT size=2>1. A needs to be prepared to receive with codec 3 once it 
    sends the</FONT> <BR><FONT size=2>offer anyway.</FONT> <BR><FONT size=2>2. 
    If A wants to restrict the set based on the answer, it can always</FONT> 
    <BR><FONT size=2>generate a new offer with a reduced set of codecs.</FONT> 
    <BR><FONT size=2>3. SDP has always been interpreted as indicating your own 
    capabilities,</FONT> <BR><FONT size=2>and is therefore an "advertisement" 
    that sticks until updated.</FONT> <BR><FONT size=2>4. There are useful cases 
    for adding codecs in the answer, particularly,</FONT> <BR><FONT 
    size=2>support for rfc2833. Thus, lets say A is an end user which can send 
    rfc</FONT> <BR><FONT size=2>2833, but not receive it. So, when it sends an 
    INVITE, it doesn't</FONT> <BR><FONT size=2>include rfc2833 in the INVITE. 
    The server that responds is a media</FONT> <BR><FONT size=2>server, which 
    supports rfc2833. So, it includes it in the answer. Now, A</FONT> <BR><FONT 
    size=2>can actually send with it during the middle of the session, and it 
    will</FONT> <BR><FONT size=2>never receive it (since it didn't advertise 
    it). This provides a useful</FONT> <BR><FONT size=2>function that would 
    otherwise require FID.</FONT> </P>
    <P><FONT size=2>Arguments for NO:</FONT> </P>
    <P><FONT size=2>1. The purpose of the O/A model is to agree on shared views 
    of certain</FONT> <BR><FONT size=2>session parameters, such as the 
    directionality (send/recv/sendrecv).</FONT> <BR><FONT size=2>Codecs fit into 
    that category as well, since you are already required to</FONT> <BR><FONT 
    size=2>indicate, in the answer, a set that partially overlaps with the 
    offer.</FONT> <BR><FONT size=2>Furthermore, the goal with codecs is to have 
    the same in both</FONT> <BR><FONT size=2>directions, generally, so that this 
    would speak to them being a shared</FONT> <BR><FONT size=2>state that is 
    negotiated.</FONT> </P>
    <P><FONT size=2>2. Many people have interpreted the SDP/bis/rfc2543 drafts 
    this way. If</FONT> <BR><FONT size=2>you go for YES, there may be interop 
    issues. The converse is not true,</FONT> <BR><FONT size=2>of course.</FONT> 
    </P>
    <P><FONT size=2>3. It is very useful to narrow down the codec sets in the 
    answer, and we</FONT> <BR><FONT size=2>should not always require an extra 
    exchange of O/A just to do that.</FONT> </P><BR><BR>
    <P><FONT size=2>I took a private poll amongst some implementors. The initial 
    results</FONT> <BR><FONT size=2>showed that most assumed the YES case. Some 
    more mails since then</FONT> <BR><FONT size=2>indicate its a bit more split, 
    although I think somewhat in favor of YES</FONT> <BR><FONT 
    size=2>still.</FONT> </P>
    <P><FONT size=2>I am somewhat on the fence; I think NO is more correct based 
    on</FONT> <BR><FONT size=2>offer/answer, although YES would be the SDP 
    purist view. What I really</FONT> <BR><FONT size=2>want is more opinions on 
    this one....</FONT> </P>
    <P><FONT size=2>THanks,</FONT> <BR><FONT size=2>Jonathan R.</FONT> </P>
    <P><FONT size=2>-- </FONT><BR><FONT size=2>Jonathan D. Rosenberg, 
    Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 72 
    Eagle Rock Avenue</FONT> <BR><FONT size=2>Chief 
    Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    First Floor</FONT> <BR><FONT 
    size=2>dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    East Hanover, NJ 07936</FONT> <BR><FONT 
    size=2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    FAX: (973) 952-5050</FONT> <BR><FONT size=2><A href="http://www.jdrosen.net" 
    target=_blank>http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    PH:&nbsp; (973) 952-5000</FONT> <BR><FONT size=2><A 
    href="http://www.dynamicsoft.com" 
    target=_blank>http://www.dynamicsoft.com</A></FONT> </P>
    <P><FONT size=2>_______________________________________________</FONT> 
    <BR><FONT size=2>mmusic mailing list</FONT> <BR><FONT 
    size=2>mmusic@ietf.org</FONT> <BR><FONT size=2><A 
    href="https://www1.ietf.org/mailman/listinfo/mmusic" 
    target=_blank>https://www1.ietf.org/mailman/listinfo/mmusic</A></FONT> 
  </P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-477407 ] Transport BNF, ; and unclearities (noreply@sourceforge.net)
   2. [ rtspspec-Bugs-477413 ] Transport BNF: mode (noreply@sourceforge.net)
   3. [ rtspspec-Bugs-477416 ] BNF error section 3.6 NPT (noreply@sourceforge.net)
   4. [ rtspspec-Bugs-477421 ] When to send response (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 21 Feb 2002 16:44:32 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477407 ] Transport BNF, ; and unclearities

Bugs item #477407, was opened at 2001-11-01 23:26
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477407&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Transport BNF, ; and unclearities

Initial Comment:
Page 61, Transport BNF: Should not multicast/unicast be proceeded by a ";"? Is multiple entries of 
the same type allowed? Are there any requirement on having multicast/unicast first?



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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-21 16:44

Message:
Logged In: YES 
user_id=3796

Language inserted:
"A transport-spec transport option may only contain one of
any given parameter within it.  Parameters may be given in
any order.  Additionally, it may only contain the unicast or
multicast transport parameter."


Example has been fleshed out as follows:

"Below is a usage example, showing a client advertising the
capability to handle multicast or unicast, preferring
multicast.  Since this is a unicast-only stream, the server
responds with the proper transport parameters for unicast.

  C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
        CSeq: 302
        Transport: RTP/AVP;multicast;mode="PLAY",
           RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

  S->C: RTSP/1.0 200 OK
        CSeq: 302
        Date: 23 Jan 1997 15:35:06 GMT
        Session: 47112344
        Transport: RTP/AVP;unicast;client_port=3456-3457;
            server_port=6256-6257;mode="PLAY"



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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 16:46

Message:
Logged In: YES 
user_id=437905

RESOLUTION:

BNF will require semicolon, no mandated ordering, and 
example will be added.

Question was raised if multiple entries of the same type 
should be allowed?  This question was referred to a new 
issue.

Question was raied if interleave parameters should require 
paramters?  This question was referred to a new issue.

 

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

Comment By: Henning Schulzrinne (hgschulz)
Date: 2002-01-09 09:59

Message:
Logged In: YES 
user_id=115926

BNF will be fixed to have semicolon. Ordering doesn't
matter. Say before header section that parameter order
doesn't matter. Also state that parameters can't be
replicated (;port=1;port=2) within the same header instance.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 19:34

Message:
Logged In: YES 
user_id=3796

Agreed it needs fix.  We'll need to sort out the right BNF for this, in such a way that it doesn't break
backwards compatibility

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477407&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 21 Feb 2002 17:40:48 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477413 ] Transport BNF: mode

Bugs item #477413, was opened at 2001-11-01 23:32
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477413&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: Transport BNF: mode

Initial Comment:
Transport ABNF, mode line, The "=" lacks " around it. There are also unclearities if you could 
actually have mode=""PLAY","RECORD"" ?


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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-21 17:40

Message:
Logged In: YES 
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New BNF:
parameter          = ";" ( "unicast" | "multicast" )
                ...
                   | ";" "mode" "=" mode-spec
mode-spec          =  <"> 1#mode <"> | mode
mode               =  "PLAY" | "RECORD" | token


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

Comment By: Rob Lanphier (robla)
Date: 2002-01-29 18:06

Message:
Logged In: YES 
user_id=3796

RESOLVED (Jan 9 telecon: http://rtsp.org/2002/telecon/jan09.html

Rework BNF, fixing multiple problems not limited to allow outside
quotes, but not inside quotes. Quotes are optional 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 19:45

Message:
Logged In: YES 
user_id=3796

First issue is a thinko, and just needs to be fixed.  The BNF is pretty sloppy the way that it is, and needs to 
be fixed

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477413&group_id=23194


--__--__--

Message: 3
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 21 Feb 2002 18:06:24 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477416 ] BNF error section 3.6 NPT

Bugs item #477416, was opened at 2001-11-01 23:37
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477416&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: BNF error section 3.6 NPT

Initial Comment:
BNF definition:npt-range, sec 3.6 lacks "npt=" in the begining of the definition. Compare with 
smpte-range or utc-range. 

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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-21 18:06

Message:
Logged In: YES 
user_id=3796

Fixed in CVS:
http://cvs.sourceforge.net/cgi-bin/viewcvs.cgi/rtspspec/parameters.tex.diff?r1=1.4&r2=1.5

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

Comment By: David W. Walker (davidwwalker)
Date: 2002-01-28 16:49

Message:
Logged In: YES 
user_id=437905

RESOLUTION:

Require npt=, but have backwards-compatibility note in 
specificaton.

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 19:53

Message:
Logged In: YES 
user_id=3796

Hmmm...do implementors predominantly implement from the BNF, or from the numerous examples?



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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477416&group_id=23194


--__--__--

Message: 4
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Thu, 21 Feb 2002 19:00:27 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-477421 ] When to send response

Bugs item #477421, was opened at 2001-11-01 23:43
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194

Category: None
>Group: Fixed in CVS/Need WG Approval
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Nobody/Anonymous (nobody)
Summary: When to send response

Initial Comment:
Server can't respond to a bad request if no sequence number is present in the request. Resulting in 
that the client does not know that the server has received it and has issues. Please clarify when a 
server is allowed to send response to bad request.

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

>Comment By: Rob Lanphier (robla)
Date: 2002-02-21 19:00

Message:
Logged In: YES 
user_id=3796

Added the following section as a new 11.3.1

11.3.1  400 Bad Request

The request could not be understood by the server due to
malformed syntax. The client SHOULD NOT repeat the request
without modifications [H10.4.1]. If the request does not
have a CSeq header, the server MUST not include a CSeq in
the response.



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

Comment By: David W. Walker (davidwwalker)
Date: 2002-02-12 17:10

Message:
Logged In: YES 
user_id=437905

RESOLVED: The Server is allowed to send an immediate 
response with no sequence number. As a general rule the 
server should NOT set sequence numbers. 

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

Comment By: Rob Lanphier (robla)
Date: 2001-12-13 10:01

Message:
Logged In: YES 
user_id=3796

This one is going to be up to the server.  It can send "Connection: close" if the state machine is too 
messed up (though see #492546 for another discussion).  We can't overspecify this, though.

Please suggest text to insert in the specification.


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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=477421&group_id=23194




End of Rtspspec-bugs Digest


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Date: Mon, 25 Feb 2002 00:41:47 -0500
From: Flemming Andreasen <fandreas@cisco.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
CC: sip <sip@ietf.org>, mmusic <mmusic@ietf.org>,
        "Fredrik Lindholm (ERA)" <Fredrik.Lindholm@era.ericsson.se>,
        Jonathan Rosemberg <jdrosen@dynamicsoft.com>, wtm@research.att.com
References: <3C769D54.C3B4923A@lmf.ericsson.se>
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Subject: [MMUSIC] Re: [Sip] Removing security preconditions from manyfolks
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Not all media streams use RTP or SRTP, and there are other security
mechanism than MIKEY. While I don't feel strongly about keeping the
"security" precondition, I don't see much gained from removing it. Perhaps
more importantly, since you are going to break the existing manyfolks-03
draft mechanism anyway, it would make sense to make the preconditions
extensible (with suitable IANA registration procedures of course).

-- Flemming


Gonzalo Camarillo wrote:

> Hi,
>
> we are thinking of removing security preconditions from manyfolks. Right
> now manyfolks handles QoS and security preconditions. However, with
> MIKEY, this feature seems useless to establish secure channels.
>
> If somebody thinks that removing security preconditions from manyfolks
> is a bad idea, please speak up.
>
> Regards,
>
> Gonzalo
> --
> Gonzalo Camarillo                    Phone :   +1 212 939 71 71
> Columbia University                  Mobile:  +358 40 702 35 35
> 472 Computer Science Building        Fax   :  +358  9 299 30 52
> 1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
> New York, NY 10027
> USA                              Gonzalo.Camarillo@ericsson.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

--
Flemming Andreasen
Cisco Systems



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From mmusic-admin@ietf.org  Mon Feb 25 12:10:51 2002
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Subject: [MMUSIC] new draft on QoS extensions for SDPng
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Hi all,

We submitted recently a new draft proposing QoS extensions for SDPng.
We would very much appreciate the groups feeling and comments on this
draft.

The draft is available at:
http://search.ietf.org/internet-drafts/draft-bos-mmusic-sdpng-qos-00.txt

Abstract:
This document describes a framework that allows end-users/applications
to inform each other and negotiate about the QoS characteristics of the
media components in a session, prior to its establishment. The QoS
negotiation is modeled after the existing codec negotiation and as such
becomes an integral part of the existing SDPng Offer/Answer model. The
QoS information is exchanged at session set-up time using two new types
of SDPng extensions between the end-users sites. Apart from the end
hosts also some authorized intermediate proxies controlling the session
set-up MAY participate to the QoS negotiation.

Secondly this document proposes SDPng extensions which allow the
end-user to request an ordered list of preferred QoS levels per media
stream during the QoS negotiation. The first type of extensions, called
TI (Traffic Information), characterizes the traffic type of the bearer
associated with the media stream. TI is typically related to bandwidth
and packet size. The second type of extensions, called SI (Sensitivity
Information), characterizes the sensitivity level of the service
requested by the end-user with respect to the QoS information carried in
the first type of SDPng extensions. SI is typically related to delay,
jitter and packet loss.

Regards,
    Suresh





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From mmusic-admin@ietf.org  Mon Feb 25 17:15:43 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Flemming Andreasen'" <fandreas@cisco.com>,
        Gonzalo Camarillo
	 <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [MMUSIC] RE: [Sip] Removing security preconditions from manyfolks
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One of the reasons why you need a special preconditions flag (and the rest
of the manyfolks apparatus) for QoS is because you use a separate protocol
(like RSVP) to negotiate the reserved path end-to-end. Nothing about SIP or
SDP or RTP would otherwise tell you that reservation of the path has been
expected or achieved. IMHO, this would be unwise, and probably infeasible,
for a security mechanism. The use of sRTP e2e will be obvious to the
respective user agents, for example, and the parameters of the sRTP session
should be correlated easily to information in SDP or SIP so the UAs can have
verifiable expectations about the session's security. I think all e2e media
security mechanisms will probably need to have these sorts of properties
(regardless of whether or not you use RTP, secure media will still be
secured, and that will always be obvious to the user agents). Therefore, I
think that a separate preconditions flag for security should always be
superfluous for any useful security mechanism. I think we can ditch it.

That much said, I don't necessarily oppose the IANA registration suggested
by Flemming below, although I think as a motivation for it we'd need to
understand, generally speaking, what sort of other preconditions people
might find desirable.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]
> Sent: Sunday, February 24, 2002 9:42 PM
> To: Gonzalo Camarillo
> Cc: sip; mmusic; Fredrik Lindholm (ERA); Jonathan Rosemberg;
> wtm@research.att.com
> Subject: Re: [Sip] Removing security preconditions from manyfolks
> 
> 
> Not all media streams use RTP or SRTP, and there are other security
> mechanism than MIKEY. While I don't feel strongly about keeping the
> "security" precondition, I don't see much gained from removing it. Perhaps
> more importantly, since you are going to break the existing manyfolks-03
> draft mechanism anyway, it would make sense to make the preconditions
> extensible (with suitable IANA registration procedures of course).
> 
> -- Flemming
> 
> 
> Gonzalo Camarillo wrote:
> 
> > Hi,
> >
> > we are thinking of removing security preconditions from 
> manyfolks. Right
> > now manyfolks handles QoS and security preconditions. However, with
> > MIKEY, this feature seems useless to establish secure channels.
> >
> > If somebody thinks that removing security preconditions 
> from manyfolks
> > is a bad idea, please speak up.
> >
> > Regards,
> >
> > Gonzalo
> > --
> > Gonzalo Camarillo                    Phone :   +1 212 939 71 71
> > Columbia University                  Mobile:  +358 40 702 35 35
> > 472 Computer Science Building        Fax   :  +358  9 299 30 52
> > 1214 Amsterdam Ave., Mail Code 0401  http://www.hut.fi/~gonzalo
> > New York, NY 10027
> > USA                              Gonzalo.Camarillo@ericsson.com
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> --
> Flemming Andreasen
> Cisco Systems
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From mmusic-admin@ietf.org  Mon Feb 25 20:48:54 2002
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Date: Mon, 25 Feb 2002 17:47:38 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
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Subject: [MMUSIC] Minutes from February 20 telecon
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Hi all,

Here are the minutes from last week's RTSP telecon:

http://rtsp.org/2002/telecon/minutes20Feb.html

They're also included below.

Enjoy!

Rob
----

  Rtsp Tc Minutes20 Feb2002

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

   1. Administrivia <#line3>
         1. Attendees <#line5>
         2. Meeting in Minneapolis <#line22>
   2. Issue discussion <#line28>
         1. 477428 <#line30>
         2. 483356 <#line41>
         3. 493270 <#line83>
         4. 506412 <#line94>


  Administrivia


    Attendees

    * Milko Boic
    * Magnus Westerlund
    * Rob Lanphier
    * Jonathan Sergent
    * Anders Klemets
    * Colin Perkins
    * Joerg Ott
    * Sean Sheedy
    * Aravind Narasimhan
    * Tom Marshall
    * Thomas Zeng
    * Jeff Ayars

Agenda: http://rtsp.org/2002/telecon/agenda20Feb.html


    Meeting in Minneapolis

MMUSIC meets on Monday.

BOF - Tentatively Tuesday, Rob to look into room (lunch)


  Issue discussion


    477428

    * http://rtsp.org/bug477428
    * JS: I think it's not allowed
    * MW: Did you see my state machine?
    * RL: (notes not many have grokked it)
    * JS: Not legal per previous resolution
    * MW: agreed

RESOLUTION: not allowed, as per point a of email


    483356

http://rtsp.org/bug483356

    * JS: Use of "*" doesn't seem right given proxies
    * TM: proxy will use session
    * JA: Use agg url, short of that, use "base" url, short of that, use
      session
    * JS: some servers allow arbitrary aggregation
    * JA: seems pretty cavalier for a client to assume that this is legal
    * JS: if agg control, then the URL doesn't necessarily have any
      semantics
    * JS: how does one validate that URL?
    * TM: can play message be sent with multiple session ids
    * JS: session not defined as multi-value feature
    * JA: perfectly happy without relying on play URL
    * JA: don't like "*"
    * JS: Could specifically say that servers shouldn't try to validate URL
    * RL: url strictly informative
    * JS: server SHOULD not validate url
    * MW: we should disallow the media URL
    * RL: what does a server allow and what not?
    * JS: can say client should use presentation URL
    * AK: presentation URL from first session created will be what the
      play URL is
    * RL: how about a bunch of shoulds (including recommendation for
      logging)
    * JA: can't trust PLAY url anyway
    * TZ: it's ok to use the first SDP
    * JS: are server writers out ahead of client writers?
    * JA: yes
    * TZ: multiple urls in play request?
    * JS: not allowed by grammer
    * JA: how about host/sessionid
    * JS: client aggregation is not the norm, and we shouldn't optimize
      for it
    * TZ: seem to be shutting down future opportunites
    * JS: you want to report presentation url
    * AN: we should consider adding a method called COMPOSE or having an
      initial SETUP to negotiate the presentation URL with the server
      for a given session, especially if the session is going to
      aggregate stream URLs from multiple sources.
    * RL: possible resolution? client SHOULD send the presentation URL
      as defined by session description; server MAY refuse play request
      if presentation URL doesn't match
    * RL: how about a client aggregation require tag?
    * AK: session header is optional
    * TM: (reads spec)
    * JS: have informative note explaining why it's not a must; explain
      that only when not possible, not when not convenient
    * RESOLVED: client SHOULD send the presentation URL as defined by
      session description; server MAY refuse play request if
      presentation URL doesn't match. Have informative note explaining
      why it's not a must; explain that only when not possible, not when
      not convenient
    * ACTION ITEM (no owner yet): put together feature tag proposal for
      client agg


    493270

    * http://rtsp.org/bug493270
    * RESOLVED: this is solved in MUTE/UNMUTE proposal
    * TZ: name MUTE has audio connotations
    * JS: means that when resumes (UNMUTE) that synch is maintained
    * JA: waht about PAUSE
    * JS: PAUSE should be same as PLAY
    * TM: does server have to fail
    * RL: no....just has the option
    * JS: it can be bw compat


    506412

    * http://rtsp.org/bug506412
    * JS: allows TEARDOWN on individual streams
    * MW: tear down all media session goes away
    * JS: it only invalidates the session id if the presentation url is
      used in TEARDOWN
    * MW: could disallow in PLAY state
    * JS: what is that useful for?
    * JA: language track switching
    * AJ: language tracks useful
    * JS: spec not clear if setup new url while playing
    * MW: state machine is currently insufficient anyway
    * JA: because we have MUTE/UNMUTE, and because we have mixing of
      agg/no agg control, it's hard to get back into ready state
    * JA: disallow in PLAY state, but allow in READY state
    * RL: perhaps we need an option tag
    * RL: server may refuse
    * RESOLUTION: client MAY send TEARDOWN of stream URL. Server MAY
      refuse. Need proposal for option negotiation. When Server refuses,
      MUST return "method not allowed in this state" with proper Allow
      header
    * MW: need to clarify when session goes away
    * AK: but then client may not know how to cope
    * JS: must use presentation url
    * TZ: Treat TEARDOWN as method to relinquish the session ID
    * JS: should we depart from current RFC for this
    * AK: client may not expect session to go away
    * JS: client should be prepared to not get a session id back, and
      that's the signal that it's gone
    * TZ: how about having the * for teardown
    * RL: how about allow * or presentation url for session teardown
    * RESOLVED: allow * or presentation url for session teardown
    * OPEN ISSUE: per media stream URLs, what does teardown mean?



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From mmusic-admin@ietf.org  Tue Feb 26 15:39:44 2002
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Subject: FW: [MMUSIC] I-D ACTION:draft-ietf-mmusic-kmgmt-ext-02.txt
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Hi,

I have a question/comment reqarding the I-D in e-mail below.

Section 2.1 states that "where safe and byte-string are as defined in SDP
[SDP]" regarding usage of the new SDP attributes:

keymgmt-data   = "keymgmt-data: " byte-string
key-extra-auth = "keymgmt-auth:" byte-string

In RFC 2327 "SDP: Session Description Protocol", "byte-string" is defined
as:

text =                   byte-string
                         ;default is to interpret this as IS0-10646 UTF8
                         ;ISO 8859-1 requires a "a=charset:ISO-8859-1"
                         ;session-level attribute to be used

Since this data can be encrypted, shouldn't the parameter be "base64 encoded
data" instead (perhaps I am missing something)? See Section 6.0 of
<draft-ietf-msec-mikey-00.txt> "MIKEY: Multimedia Internet KEYing" where a
specific example is given (I cut and paste from that I-D below):

     a=keymgmt-data:<base64 encoded data>

Also, I'm not sure about the usage of "safe" in the SDPbis I-D (see
<draft-ietf-mmusic-sdp-new-05.txt> "SDP" Session Description Protocol". 

Regards,

Pete

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Thursday, February 21, 2002 6:02 AM
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D ACTION:draft-ietf-mmusic-kmgmt-ext-02.txt


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

	Title		: Key Management Extensions for SDP and RTSP
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-02.txt
	Pages		: 11
	Date		: 20-Feb-02
	
This document defines general extensions for SDP and RTSP to carry
the security information needed by a key management protocol, in
order to secure the media. These extensions are presented as a
framework, to be used by one or more key management protocols. As
such, its use is meaningful only when it is completed by the key
management protocol in use.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-02.txt

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------_=_NextPart_001_01C1BF05.955A2860
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>FW: [MMUSIC] I-D =
ACTION:draft-ietf-mmusic-kmgmt-ext-02.txt</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>I have a question/comment reqarding the I-D in e-mail =
below.</FONT>
</P>

<P><FONT SIZE=3D2>Section 2.1 states that &quot;where safe and =
byte-string are as defined in SDP [SDP]&quot; regarding usage of the =
new SDP attributes:</FONT></P>

<P><FONT SIZE=3D2>keymgmt-data&nbsp;&nbsp; =3D &quot;keymgmt-data: =
&quot; byte-string</FONT>
<BR><FONT SIZE=3D2>key-extra-auth =3D &quot;keymgmt-auth:&quot; =
byte-string</FONT>
</P>

<P><FONT SIZE=3D2>In RFC 2327 &quot;SDP: Session Description =
Protocol&quot;, &quot;byte-string&quot; is defined as:</FONT>
</P>

<P><FONT SIZE=3D2>text =
=3D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; byte-string</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; ;default is to interpret this as IS0-10646 UTF8</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; ;ISO 8859-1 requires a =
&quot;a=3Dcharset:ISO-8859-1&quot;</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; ;session-level attribute to be used</FONT>
</P>

<P><FONT SIZE=3D2>Since this data can be encrypted, shouldn't the =
parameter be &quot;base64 encoded data&quot; instead (perhaps I am =
missing something)? See Section 6.0 of =
&lt;draft-ietf-msec-mikey-00.txt&gt; &quot;MIKEY: Multimedia Internet =
KEYing&quot; where a specific example is given (I cut and paste from =
that I-D below):</FONT></P>

<P><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp; a=3Dkeymgmt-data:&lt;base64 =
encoded data&gt;</FONT>
</P>

<P><FONT SIZE=3D2>Also, I'm not sure about the usage of =
&quot;safe&quot; in the SDPbis I-D (see =
&lt;draft-ietf-mmusic-sdp-new-05.txt&gt; &quot;SDP&quot; Session =
Description Protocol&quot;. </FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Pete</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Internet-Drafts@ietf.org [<A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Thursday, February 21, 2002 6:02 AM</FONT>
<BR><FONT SIZE=3D2>Cc: mmusic@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [MMUSIC] I-D =
ACTION:draft-ietf-mmusic-kmgmt-ext-02.txt</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>A New Internet-Draft is available from the on-line =
Internet-Drafts directories.</FONT>
<BR><FONT SIZE=3D2>This draft is a work item of the Multiparty =
Multimedia Session Control Working Group of the IETF.</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
Key Management Extensions for SDP and RTSP</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : J. Arkko et =
al.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-ietf-mmusic-kmgmt-ext-02.txt</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
11</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 20-Feb-02</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>This document defines general extensions for SDP and =
RTSP to carry</FONT>
<BR><FONT SIZE=3D2>the security information needed by a key management =
protocol, in</FONT>
<BR><FONT SIZE=3D2>order to secure the media. These extensions are =
presented as a</FONT>
<BR><FONT SIZE=3D2>framework, to be used by one or more key management =
protocols. As</FONT>
<BR><FONT SIZE=3D2>such, its use is meaningful only when it is =
completed by the key</FONT>
<BR><FONT SIZE=3D2>management protocol in use.</FONT>
</P>

<P><FONT SIZE=3D2>A URL for this Internet-Draft is:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-=
02.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-mmusic-=
kmgmt-ext-02.txt</A></FONT>
</P>

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<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>NOTE:&nbsp;&nbsp; The mail server at ietf.org can =
return the document in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>MIME-encoded form by using the &quot;mpack&quot; =
utility.&nbsp; To use this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>feature, =
insert the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>command.&nbsp; To decode the response(s), you will need =
&quot;munpack&quot; or</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>a =
MIME-compliant mail reader.&nbsp; Different MIME-compliant mail =
readers</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>exhibit =
different behavior, especially when dealing with</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;multipart&quot; MIME messages (i.e. documents which have =
been split</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>up into =
multiple messages), so check your local documentation on</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>how to =
manipulate these messages.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>Below is the data which will enable a MIME compliant =
mail reader</FONT>
<BR><FONT SIZE=3D2>implementation to automatically retrieve the ASCII =
version of the</FONT>
<BR><FONT SIZE=3D2>Internet-Draft.</FONT>
</P>

<P><FONT FACE=3D"Arial" SIZE=3D2 COLOR=3D"#000000"></FONT>&nbsp;

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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

http://sf.net/projects/rtspspec

When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-522602 ] Should Allow S->C  "Teardown" (noreply@sourceforge.net)
   2. [ rtspspec-Feature Requests-522607 ] Need END_OF_STREAM request (noreply@sourceforge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 25 Feb 2002 11:38:22 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-522602 ] Should Allow S->C  "Teardown"

Bugs item #522602, was opened at 2002-02-25 11:38
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=522602&group_id=23194

Category: General
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Rob Lanphier (robla)
Summary: Should Allow S->C  "Teardown" 

Initial Comment:
Table 2 in section 10 of RFC2326 seem to disallow S->C 
Teardown. And section 10.7 is vague on whether S->C 
teardown is OK.

This should be relaxed, because Server might have to 
use TEARDOWN to tell client a live session is over.

 

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

You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377744&aid=522602&group_id=23194


--__--__--

Message: 2
To: noreply@sourceforge.net
From: noreply@sourceforge.net
Date: Mon, 25 Feb 2002 11:49:56 -0800
Subject: [rtspspec-bugs] [ rtspspec-Feature Requests-522607 ] Need END_OF_STREAM request

Feature Requests item #522607, was opened at 2002-02-25 11:49
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377747&aid=522607&group_id=23194

Category: None
Group: None
Status: Open
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Nobody/Anonymous (nobody)
Summary: Need END_OF_STREAM request

Initial Comment:
Currently there is no way for Server to inform Client 
the completion of a presentation (VOD or Live) without 
issuing a TEARDOWN and thus release reources. What if 
the player wants to use the same resources (pertinent 
to its sessionID) to watch another clip? It would be 
wasteful to teardown and then acquire another 
sessionID immediately.

Would like to see a END_OF_STREAM method added on the 
S->C direction. E.g.:

END_OF_STREAM <p_URI> or <track_URL> RTSP/1.0
SESSION: <session_ID>
REASON: <text describing reason of EOS>




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From mmusic-admin@ietf.org  Tue Feb 26 19:58:47 2002
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Date: Tue, 26 Feb 2002 16:56:49 -0800 (Pacific Standard Time)
From: Rob Lanphier <robla@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP state machine
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Hi Magnus,

This is really good work.  Still sifting through the implications here,
but one quick question to kick off the conversation:  could the "ready-na"
state and "paused-na" state be merged (as well as "ready-a" and
"paused-a")?  I'm not sure there's a useful distinction to be made between
these two states.

Rob


On Wed, 20 Feb 2002, Magnus Westerlund wrote:

> Hi again!
>
> I made some errors in the table and I guess there may be some more still
> in there. Anyway, here is a new version.
>
> RTSP state machine ver. 0.2
>
> This state machine does not contain RECORD which needs to be added.
> It is based on the proposal that in aggregated control, PAUSE and PLAY
> can't be used on individual medias. If that is allowed there will be
> a number of extra transitions with requirements on the
> different medias state. Does not include queued play either.
>
> There is discussion if teardown on media URL should end up in a middle
> state between Init and Ready-na. This state would contain no media but
> keep all session related state and keep its ID. The only way to remove
> such a state will be the teardown with * or a session timeout.
>
>
> States          Explanation
> -----------------------------------------------------------------------
> Init            Initial state no session exist
> Ready-na        Non aggregated session in Ready state
> Playing-na      Non aggregated session in Playing state
> Paused-na       Non aggregated session in Paused state
> Ready-a         Aggregated session in Ready state
> Playing-a       Aggregated session in Playing
> Paused-a        Aggregated session in Pause
>
> The reason for separating aggregated and non aggregated states are the
> following:
> - Play and Pause may only be done on aggregate URL and not media URLs
>   in aggregated control. In non-aggregated the two are the same.
> -
>
>
> State variables
> ---------------
> Nr-media
>
> State: Init
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> SETUP                           Ready-na        Nr-media = 1
>
>
> State: Ready-na
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> SETUP           New URL         Ready-a         Nr-media = 2
> SETUP           Present URL     Ready-na        Change transport param.
> Teardown        *|media URL     Init            Ok
> PLAY            No range        Playing-na      from beginning of media
> PLAY            Range           Playing-na      according to range
> Timeout                         Init
>
> State: Playing-na
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> PAUSE           No range        Paused-na       Set pause point
> PAUSE           Range > now     Playing-na      Set pause point
> Pause-point reached             Paused-na
> End of media                    Ready-na
> Timeout                         Init
> SETUP          New URL,Allowed? Playing-a       Nr-media +=1
> SETUP     Present URL, Allowed? Playing-na      Change transport param.
> TEARDOWN        Allowed?        Init
>
>
> State: Paused-na
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> PLAY            No range        Playing-na      Play from pause point
> PLAY            Range           Playing-na      according to range
> Timeout                         Init
> SETUP           New URL         Paused-a        Nr-media +=1
> SETUP           Present URL     Ready-na        Change transport param.
> TEARDOWN        *|media URL     Init
>
> State: Ready-a
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> SETUP           New URL         Ready-a         Nr-media +=1
> SETUP          Present mediaURL Ready-a         Change transport param.
> TEARDOWN        * | Pres URL    Init            No session hdr.
> TEARDOWN        Nr-media >2     Ready-a         Nr-media -=1, sess hdr
> TEARDOWN        Nr-media == 2   Ready-na        Nr-media = 1, sess hdr
> Timeout                         Init
> PLAY            No range,AggURL Playing-a       from beginning of media
> PLAY            Range, Agg URL  Playing-a       according to range
>
>
> State: Playing-a
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> PAUSE           No range        Paused-a        Set pause point
> PAUSE           Range > now     Playing-a       Set point to pause at
> Pause-point reached             Paused-a        Set pause point
> End of media    >=1 Media plays Playing-a
> End of media    All media ended Ready-a
> Timeout                         Init
> SETUP          New URL,Allowed? Playing-a       Nr-media +=1
> SETUP      Present URL,Allowed? Playing-a       Change transport param
> TEARDOWN        *|pres URL      Init            No session hdr allowed?
> TEARDOWN        Allowed? Nr-m>2 Playing-a       Nr-media -=1, sess hdr
> TEARDOWN        Allowed? Nr-m=2 Playing-na      Nr-media = 1, sess hdr
>
>
> State: Paused-a
> Action          Prerequisite    New State       Response
> -----------------------------------------------------------------------
> PLAY            No range        Playing-a       Play from pause point
> PLAY            Range           Playing-a       according to range
> Timeout                         Init
> SETUP           New URL         Paused-a        Nr-media +=1
> SETUP           Present URL     Paused-a        Change transport param.
> TEARDOWN        *|pres URL      Init
> TEARDOWN        Nr-media >2     Paused-a        Nr-media -=1
> TEARDOWN        Nr-media == 2   Paused-na       Nr-media = 1
>
>
>
>
>
> Cheers
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research ERA/T/VA
> ----------------------------------------------------------------------
> Ericsson Radio Systems AB  | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>
>



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From mmusic-admin@ietf.org  Tue Feb 26 22:52:38 2002
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Roman Shpount wrote:
> 
> I think, we should specify, that the port for the stream which was put
> on hold using "0.0.0.0" address should be an arbitrary non-zero value in
> order for it not to be confused with removed streams. 

0.0.0.0 is not recommended for on hold, but I otherwise agree. I added a
sentence to that effect.

Additionally, note
> about using "0.0.0.0" address in initial offer should probably belong to
> 5.1 and not 8.4.

The idea is not to special case it, and since its usage is not
recommended, I would prefer to leave it where it sits.

Thanks,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From mmusic-admin@ietf.org  Wed Feb 27 04:48:19 2002
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Date: Wed, 27 Feb 2002 10:47:50 +0100
From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Subject: Re: [MMUSIC] RTSP state machine
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Rob,

In my opinion they have a useful distinction. The different state represent if
the point of pausing is valid or not. This is part of the state and it helps
clarify behavior regarding pause and play without range. When in ready-na and
receiving a play without range it changes state to playing-na and plays from
the beginning. When in pause-na and receives the same play it will play from
the pause point stored upon entering the pause state.

We have had a discussion on this during one of the telecoms. What was raised
then was the following. When a play without range is issued in state ready-na
it starts from the beginning. When the server reaches the end of the media,
what shall it do. My interpretation which is represented in the state machine
is that it returns to ready-na, when doing play without range again it will
start from the beginning. Another interpretation is that it stays at the end
point and any play without range will be accepted but return to ready state
directly as it is at the end. This behavior requires a explicit rewind of the
media with a play with range 0- to start from the beginning.

This is logged as issue #513875 at source forge.

Regards

Magnus


Rob Lanphier wrote:

> Hi Magnus,
>
> This is really good work.  Still sifting through the implications here,
> but one quick question to kick off the conversation:  could the "ready-na"
> state and "paused-na" state be merged (as well as "ready-a" and
> "paused-a")?  I'm not sure there's a useful distinction to be made between
> these two states.
>
> Rob
>
> On Wed, 20 Feb 2002, Magnus Westerlund wrote:
>
> > Hi again!
> >
> > I made some errors in the table and I guess there may be some more still
> > in there. Anyway, here is a new version.
> >
> > RTSP state machine ver. 0.2
> >
> > This state machine does not contain RECORD which needs to be added.
> > It is based on the proposal that in aggregated control, PAUSE and PLAY
> > can't be used on individual medias. If that is allowed there will be
> > a number of extra transitions with requirements on the
> > different medias state. Does not include queued play either.
> >
> > There is discussion if teardown on media URL should end up in a middle
> > state between Init and Ready-na. This state would contain no media but
> > keep all session related state and keep its ID. The only way to remove
> > such a state will be the teardown with * or a session timeout.
> >
> >
> > States          Explanation
> > -----------------------------------------------------------------------
> > Init            Initial state no session exist
> > Ready-na        Non aggregated session in Ready state
> > Playing-na      Non aggregated session in Playing state
> > Paused-na       Non aggregated session in Paused state
> > Ready-a         Aggregated session in Ready state
> > Playing-a       Aggregated session in Playing
> > Paused-a        Aggregated session in Pause
> >
> > The reason for separating aggregated and non aggregated states are the
> > following:
> > - Play and Pause may only be done on aggregate URL and not media URLs
> >   in aggregated control. In non-aggregated the two are the same.
> > -
> >
> >
> > State variables
> > ---------------
> > Nr-media
> >
> > State: Init
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > SETUP                           Ready-na        Nr-media = 1
> >
> >
> > State: Ready-na
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > SETUP           New URL         Ready-a         Nr-media = 2
> > SETUP           Present URL     Ready-na        Change transport param.
> > Teardown        *|media URL     Init            Ok
> > PLAY            No range        Playing-na      from beginning of media
> > PLAY            Range           Playing-na      according to range
> > Timeout                         Init
> >
> > State: Playing-na
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > PAUSE           No range        Paused-na       Set pause point
> > PAUSE           Range > now     Playing-na      Set pause point
> > Pause-point reached             Paused-na
> > End of media                    Ready-na
> > Timeout                         Init
> > SETUP          New URL,Allowed? Playing-a       Nr-media +=1
> > SETUP     Present URL, Allowed? Playing-na      Change transport param.
> > TEARDOWN        Allowed?        Init
> >
> >
> > State: Paused-na
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > PLAY            No range        Playing-na      Play from pause point
> > PLAY            Range           Playing-na      according to range
> > Timeout                         Init
> > SETUP           New URL         Paused-a        Nr-media +=1
> > SETUP           Present URL     Ready-na        Change transport param.
> > TEARDOWN        *|media URL     Init
> >
> > State: Ready-a
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > SETUP           New URL         Ready-a         Nr-media +=1
> > SETUP          Present mediaURL Ready-a         Change transport param.
> > TEARDOWN        * | Pres URL    Init            No session hdr.
> > TEARDOWN        Nr-media >2     Ready-a         Nr-media -=1, sess hdr
> > TEARDOWN        Nr-media == 2   Ready-na        Nr-media = 1, sess hdr
> > Timeout                         Init
> > PLAY            No range,AggURL Playing-a       from beginning of media
> > PLAY            Range, Agg URL  Playing-a       according to range
> >
> >
> > State: Playing-a
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > PAUSE           No range        Paused-a        Set pause point
> > PAUSE           Range > now     Playing-a       Set point to pause at
> > Pause-point reached             Paused-a        Set pause point
> > End of media    >=1 Media plays Playing-a
> > End of media    All media ended Ready-a
> > Timeout                         Init
> > SETUP          New URL,Allowed? Playing-a       Nr-media +=1
> > SETUP      Present URL,Allowed? Playing-a       Change transport param
> > TEARDOWN        *|pres URL      Init            No session hdr allowed?
> > TEARDOWN        Allowed? Nr-m>2 Playing-a       Nr-media -=1, sess hdr
> > TEARDOWN        Allowed? Nr-m=2 Playing-na      Nr-media = 1, sess hdr
> >
> >
> > State: Paused-a
> > Action          Prerequisite    New State       Response
> > -----------------------------------------------------------------------
> > PLAY            No range        Playing-a       Play from pause point
> > PLAY            Range           Playing-a       according to range
> > Timeout                         Init
> > SETUP           New URL         Paused-a        Nr-media +=1
> > SETUP           Present URL     Paused-a        Change transport param.
> > TEARDOWN        *|pres URL      Init
> > TEARDOWN        Nr-media >2     Paused-a        Nr-media -=1
> > TEARDOWN        Nr-media == 2   Paused-na       Nr-media = 1
> >
> >
> >
> >

--

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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From mmusic-admin@ietf.org  Wed Feb 27 08:13:28 2002
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> Hi, 
Hi,

(comments inline)

>I have a question/comment reqarding the I-D in e-mail below. 
> Section 2.1 states that "where safe and byte-string are as 
> defined in SDP [SDP]" regarding usage of the new SDP attributes:
>
> keymgmt-data   = "keymgmt-data: " byte-string 
> key-extra-auth = "keymgmt-auth:" byte-string 
> In RFC 2327 "SDP: Session Description Protocol", 
> "byte-string" is defined as: 
>
> text =                   byte-string 
>                         ;default is to interpret this as IS0-10646 UTF8 
>                         ;ISO 8859-1 requires a "a=charset:ISO-8859-1" 
>                         ;session-level attribute to be used 
>
Almost ;-) 
   byte-string =         1*(0x01..0x09|0x0b|0x0c|0x0e..0xff)
                         ;any byte except NUL, CR or LF
anyway...

> Since this data can be encrypted, shouldn't the parameter be 
> "base64 encoded data" instead (perhaps I am missing something)? 

I agree that this will be the best way to do it in most solutions. 
But as this is a framework, I don't think it is good to restrict 
the encoding of the field too much. Therefore, the encoding of the 
field is uniquely defined by the key management protocol used 
(I guess we will need to clarify this). E.g. if MIKEY is used, then 
the "keymgmt-data" field will be base64 encoded as it is stated 
that in MIKEY.

> See Section 6.0 of <draft-ietf-msec-mikey-00.txt> "MIKEY: 
> Multimedia Internet KEYing" where a specific example is given 
> (I cut and paste from that I-D below):
>     a=keymgmt-data:<base64 encoded data> 
> Also, I'm not sure about the usage of "safe" in the SDPbis I-D 
> (see <draft-ietf-mmusic-sdp-new-05.txt> "SDP" Session Description 
> Protocol". 
>
That's a good one. It just disapeared :-)  


Thanks,
Fredrik


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From mmusic-admin@ietf.org  Wed Feb 27 15:16:27 2002
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On Wed, Feb 27, 2002 at 10:47:50AM +0100, Magnus Westerlund wrote:
> Rob,
> 
> In my opinion they have a useful distinction. The different state represent if
> the point of pausing is valid or not. This is part of the state and it helps
> clarify behavior regarding pause and play without range. When in ready-na and
> receiving a play without range it changes state to playing-na and plays from
> the beginning. When in pause-na and receives the same play it will play from
> the pause point stored upon entering the pause state.

I don't see the distinction.  After a successful setup, is the clip not
paused at time zero?

> 
> We have had a discussion on this during one of the telecoms. What was raised
> then was the following. When a play without range is issued in state ready-na
> it starts from the beginning. When the server reaches the end of the media,
> what shall it do. My interpretation which is represented in the state machine
> is that it returns to ready-na, when doing play without range again it will
> start from the beginning. Another interpretation is that it stays at the end
> point and any play without range will be accepted but return to ready state
> directly as it is at the end. This behavior requires a explicit rewind of the
> media with a play with range 0- to start from the beginning.

I don't like the idea of the server rewinding the clip without being
requested by the client.

I would prefer that the end-of-clip functionality match the rest of the
protocol, in that the client initiates all of the server's state
transitions.  When the server reaches the end of the clip, why can't it stay
in the playing state until the client sends a pause or teardown?

If the server makes the state transition on its own, the client runs the
possibility that a pause message very close to the end of the clip could
reach the server after the clip has finished and the server is back in the
ready/paused state.  The expected behaviour doesn't seem to be defined in
this case, for either the old or the proposed state machine.

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From mmusic-admin@ietf.org  Wed Feb 27 18:25:16 2002
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To: "Fredrik Lindholm (ERA)" <Fredrik.Lindholm@era.ericsson.se>
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Subject: SDP-new ABNF (was Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-kmgmt-ext-02.txt)
In-Reply-To: Your message of "Wed, 27 Feb 2002 09:19:37 +0100."
             <0DAEDF148988D411BB980008C7E65D2E04FE40A3@esealnt416> 
Date: Wed, 27 Feb 2002 18:18:23 -0500
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--> "Fredrik Lindholm (ERA)" writes:
...
>> See Section 6.0 of <draft-ietf-msec-mikey-00.txt> "MIKEY: 
>> Multimedia Internet KEYing" where a specific example is given 
>> (I cut and paste from that I-D below):
>>     a=keymgmt-data:<base64 encoded data> 
>> Also, I'm not sure about the usage of "safe" in the SDPbis I-D 
>> (see <draft-ietf-mmusic-sdp-new-05.txt> "SDP" Session Description 
>> Protocol". 
>>
>That's a good one. It just disapeared :-)  

It disappeared because it was inconsistent. RFC 2327 says (section 6):

	SDP session descriptions are entirely textual using the ISO 10646
	character set in UTF-8 encoding. SDP field names and attributes
	names use only the US-ASCII subset of UTF-8, but textual fields 
	and attribute values may use the full ISO 10646 character set.   

and

	Text records such as the session name and information are bytes
	strings which may contain any byte with the exceptions of 0x00
	(Nul), 0x0a (ASCII newline) and 0x0d (ASCII carriage return).  The
	sequence CRLF (0x0d0a) is used to end a record, although parsers
	should be tolerant and also accept records terminated with a single
	newline character.  By default these byte strings contain ISO-10646
	characters in UTF-8 encoding, but this default may be changed using
	the `charset' attribute.

and in o= restricts the username field to not containing spaces (i.e. the
username can be any UTF-8 character except NUL, CR, LF and space). This
conflicted with the definition of "safe" in the ABNF, which was much more
restrictive. The new ABNF makes username a non-ws-string, which actually
matches the text.

Colin

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Date: Wed, 27 Feb 2002 17:02:07 -0800
From: Jonathan Sergent <Jonathan.Sergent@Sun.COM>
To: Tom Marshall <tmarshall@real.com>
Cc: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        Rob Lanphier <robla@real.com>, IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP state machine
Message-ID: <20020227170207.C18053@vernors>
References: <Pine.WNT.4.43.0202261408340.281-100000@robla350.dev.prognet.com> <3C7CAB46.A3239E06@era.ericsson.se> <20020227200947.GA10045@real.com>
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On Wed, Feb 27, 2002 at 12:09:47PM -0800, Tom Marshall wrote:
> If the server makes the state transition on its own, the client runs the
> possibility that a pause message very close to the end of the clip could
> reach the server after the clip has finished and the server is back in the
> ready/paused state.  The expected behaviour doesn't seem to be defined in
> this case, for either the old or the proposed state machine.

Thank you for agreeing with me (I have pointed this out before but not
really gotten anyone to think it is as ugly as I do).  This has
actually turned up as a bug for us, because we used to refuse PAUSE
while paused.

The other way around this is to let the server transition in state but
to make sure that requests received close to the end of the stream have
the same effect as requests received immediately after the end of the
stream.  Using PLAY with no range header as a "ping" makes this not
true.

I think the cleanest thing is what you suggest, that the client should
need to send a PAUSE if it expects the server to be in the paused
state.

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Date: Wed, 27 Feb 2002 17:04:37 -0800
From: Jonathan Sergent <Jonathan.Sergent@Sun.COM>
To: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP state machine
Message-ID: <20020227170437.D18053@vernors>
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On Wed, Feb 27, 2002 at 10:47:50AM +0100, Magnus Westerlund wrote:
> We have had a discussion on this during one of the telecoms. What was raised
> then was the following. When a play without range is issued in state ready-na
> it starts from the beginning. When the server reaches the end of the media,
> what shall it do. My interpretation which is represented in the state machine
> is that it returns to ready-na, when doing play without range again it will
> start from the beginning. Another interpretation is that it stays at the end
> point and any play without range will be accepted but return to ready state
> directly as it is at the end. This behavior requires a explicit rewind of the
> media with a play with range 0- to start from the beginning.

A more interesting case to look at is if the initial PLAY request has a
closed range in the Range header (i.e. Range: npt=0-15).  After playing
has stopped due to the closed range, the client sends a PLAY with no
range header.  Does the stream restart from the beginning, or from
npt=15?


--jss.

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From mmusic-admin@ietf.org  Thu Feb 28 07:08:35 2002
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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		: Real Time Streaming Protocol (RTSP)
	Author(s)	: H. Schulzrinne et al.
	Filename	: draft-ietf-mmusic-rfc2326bis-00.txt
	Pages		: 4
	Date		: 27-Feb-02
	
This memorandum is a revision of RFC 2326, which is currently a
Proposed Standard.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rfc2326bis-00.txt

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title		: SDPng Transition
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	Filename	: draft-ietf-mmusic-sdpng-trans-00.txt
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	Date		: 27-Feb-02
	
The Session Description Protocol (SDP) is today widely used in the
Internet to announce as well as negotiate multimedia sessions and
exchange capabilities.  Having originally been designed for session
announcements only, as opposed to announcements and capabilities
negotiation announcements, native SDP lacks numerous features to be
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From mmusic-admin@ietf.org  Thu Feb 28 11:54:54 2002
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Hi Rob, Tom and Jonathan,

I have reconsidered about removing pause state and I can agree that it is
unnecessary. I attach an updated version of the state machine without the paused
state. Instead I replaced it with setting of the pause point or as I have called it
in the state machine resume point (RP). I think that is a better name for it.

I do believe we have the following  issues regarding pause, play and reaching end
of media or range.

- When a presentation reaches end of media or range. Shall it automatically return
to ready state?

- When a presentation reaches end of media. What shall the resume point be set to?

- When reaching end of a range, should the resume point be set to 0 or end of
range?

- Should we add a end of clip/range notification.

I will start with the last issue. I do believe such a mechanism is need for more
than this reason to signal the state transition. It will also be good for the
buffer handling, retransmission, adaptation, and notification to user. So I do
agree that we should add one

Then I do believe you are suggesting a control model like a old tape player, I have
always thought of it in terms of CD player control. All my CD players return to the
beginning of the disc when reaching end. But one thing we really do need to
consider is backwards compatibility. Will an RFC 2326 implemented client be able to
use a server that would requires a pause before each play after the first? I would
guess that it would get into trouble. Also how should this work relative to queued
play? Even if we remove as suggested I belive it will be difficult to combine this
model with that. Therefore I would like to have automatic return to state READY.

I think that the resume point should be reset to 0.0 when reaching end of media.
Actually I don't care that much about this behavior except that it should be
backwards compatible. If it does not return without a explicit play 0- can this
break any client?

Regarding reaching end of range I have no strong opinion on what to do as long as
it is resolved. The state machine says to set it to 0.0 for moment which is
harmonized with end of media behavior.

By removing play without range while playing and also queued plays we remove the
functionality that has potentials for messing up at end of streams.

Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se


--------------C9A7BA0DAB2AC7DC54CA3924
Content-Type: text/plain; charset=us-ascii;
 name="state-machine_v0.3_nopause.txt"
Content-Disposition: inline;
 filename="state-machine_v0.3_nopause.txt"
Content-Transfer-Encoding: 7bit

RTSP state machine ver. 0.3

This state machine does not contain RECORD which needs to be added.
It is based on the proposal that in aggregated control, PAUSE and PLAY
can't be used on individual medias. If that is allowed there will be
a number of extra transitions with requirements on the 
different medias state. Does not include queued play either.

There is discussion if teardown on media URL should end up in a middle
state between Init and Ready-na. This state would contain no media but
keep all session related state and keep its ID. The only way to remove 
such a state will be the teardown with * or a session timeout. 

I now have removed the paused state.


States          Explanation
-----------------------------------------------------------------------
Init            Initial state no session exist
Ready-na        Non aggregated session in Ready state
Playing-na      Non aggregated session in Playing state
Paused-na       Non aggregated session in Paused state
Ready-a         Aggregated session in Ready state
Playing-a       Aggregated session in Playing
Paused-a        Aggregated session in Pause

The reason for separating aggregated and non aggregated states are the 
following:
- Play and Pause may only be done on aggregate URL and not media URLs 
  in aggregated control. In non-aggregated the two are the same.
- 


State variables
---------------
Nr-media	 The number of medias currently setuped in this session
RP = Resume point  The point that a play request without range resumes.


State: Init
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP                           Ready-na        Nr-media = 1, RP=0.0


State: Ready-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media = 2
SETUP           Present URL     Ready-na        Change transport param.
Teardown        *|media URL     Init            Ok
PLAY            No range        Playing-na      Play from RP
PLAY            Range           Playing-na      according to range
Timeout                         Init            

State: Playing-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PAUSE           No range        Ready-na       Set RP to present point
PAUSE           Range > now     Playing-na      Set RP to given point 
Pause-point reached             Ready-na	
End of media                    Ready-na	Set RP = 0.0
End of range			Ready-na	Set RP = 0.0?
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP     Present URL, Allowed? Playing-na      Change transport param.
TEARDOWN        Allowed?        Init    


State: Ready-a  
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media +=1
SETUP          Present mediaURL Ready-a         Change transport param.
TEARDOWN        * | Pres URL    Init            No session hdr.
TEARDOWN        Nr-media >2     Ready-a         Nr-media -=1, sess hdr
TEARDOWN        Nr-media == 2   Ready-na        Nr-media = 1, sess hdr
Timeout                         Init
PLAY            No range,AggURL Playing-a       From RP
PLAY            Range, Agg URL  Playing-a       according to range


State: Playing-a
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
PAUSE           No range        Ready-a        Set RP to present pos.
PAUSE           Range > now     Playing-a       Set RP to given point 
Pause-point reached             Ready-a        
End of range			Ready-a		Set RP = 0.0?
End of media    >=1 Media plays Playing-a	
End of media    All media ended Ready-a		Set RP to 0.0
Timeout                         Init
SETUP          New URL,Allowed? Playing-a       Nr-media +=1
SETUP      Present URL,Allowed? Playing-a       Change transport param
TEARDOWN        *|pres URL      Init            No session hdr allowed?
TEARDOWN        Allowed? Nr-m>2 Playing-a       Nr-media -=1, sess hdr
TEARDOWN        Allowed? Nr-m=2 Playing-na      Nr-media = 1, sess hdr





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From mmusic-admin@ietf.org  Thu Feb 28 12:07:14 2002
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Hi,

I have directly answered your questions despite the fact that they are present in my
other email today.

> >
> > We have had a discussion on this during one of the telecoms. What was raised
> > then was the following. When a play without range is issued in state ready-na
> > it starts from the beginning. When the server reaches the end of the media,
> > what shall it do. My interpretation which is represented in the state machine
> > is that it returns to ready-na, when doing play without range again it will
> > start from the beginning. Another interpretation is that it stays at the end
> > point and any play without range will be accepted but return to ready state
> > directly as it is at the end. This behavior requires a explicit rewind of the
> > media with a play with range 0- to start from the beginning.
>
> I don't like the idea of the server rewinding the clip without being
> requested by the client.
>
> I would prefer that the end-of-clip functionality match the rest of the
> protocol, in that the client initiates all of the server's state
> transitions.  When the server reaches the end of the clip, why can't it stay
> in the playing state until the client sends a pause or teardown?


The biggest problem I have with this is backwards compatibility. The proposed
behavior requires that you pipeline all play request with a pause the plat to change
to ready state. And I don't believe all clients out there has this behavior.

>
> If the server makes the state transition on its own, the client runs the
> possibility that a pause message very close to the end of the clip could
> reach the server after the clip has finished and the server is back in the
> ready/paused state.  The expected behaviour doesn't seem to be defined in
> this case, for either the old or the proposed state machine.

In the new state machine there is no need. Doing play without range will be
decapricated due to the fact that it could cause restarting of stream. If we also
remove queued plays the client will receive a error message "not allowed in this
state" if the is in playing state.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



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From mmusic-admin@ietf.org  Thu Feb 28 13:52:12 2002
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: mmusic@ietf.org
Subject: RE: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rfc2326bis-00.txt
Date: Thu, 28 Feb 2002 10:48:43 -0800
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Dear RTSP Authors:

In this rfc2326bis draft, it appears TEARDOWN from S->C is not allowed (no
change
from RFC2326). 

Is it a mistake or by intention?

Thanks,
thomas

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Thursday, February 28, 2002 4:08 AM
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rfc2326bis-00.txt


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

	Title		: Real Time Streaming Protocol (RTSP)
	Author(s)	: H. Schulzrinne et al.
	Filename	: draft-ietf-mmusic-rfc2326bis-00.txt
	Pages		: 4
	Date		: 27-Feb-02
	
This memorandum is a revision of RFC 2326, which is currently a
Proposed Standard.

A URL for this Internet-Draft is:
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Subject: [MMUSIC] Unicast address for media in RTSP
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Hi Folks,

What is the reason why media in unicast session is transmitted to the source of RTSP request? Why can't RTSP client specify different 
address for unicast session in SDP? Ability to send media to a different destination will be extremely useful to implement applications such as music on hold in SIP call from RTSP source.

___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com





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From mmusic-admin@ietf.org  Thu Feb 28 14:57:47 2002
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From: Rob Lanphier <robla@real.com>
To: Roman Shpount <roman@atelo.com>
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Subject: Re: [MMUSIC] Unicast address for media in RTSP
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On 28 Feb 2002, Roman Shpount wrote:
> What is the reason why media in unicast session is transmitted to the
> source of RTSP request? Why can't RTSP client specify different address
> for unicast session in SDP? Ability to send media to a different
> destination will be extremely useful to implement applications such as
> music on hold in SIP call from RTSP source.

I'm not sure what you mean here.  The client can send a Transport: line
that contains a destination parameter, and the server can accept that
line.  As the spec says, this should be used with caution, and many
implementations may forbid anonymous use of this, because of the
denial-of-service possibilities associated with this feature.

Rob


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From mmusic-admin@ietf.org  Thu Feb 28 15:48:39 2002
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On 28 Feb 2002, Roman Shpount wrote:
> What is the reason why media in unicast session is transmitted to the
> source of RTSP request? Why can't RTSP client specify different address
> for unicast session in SDP? Ability to send media to a different
> destination will be extremely useful to implement applications such as
> music on hold in SIP call from RTSP source.

Rob Lanphier wrote:
> I'm not sure what you mean here.  The client can send a Transport: line
> that contains a destination parameter, and the server can accept that
> line.  As the spec says, this should be used with caution, and many
> implementations may forbid anonymous use of this, because of the
> denial-of-service possibilities associated with this feature.

The latest draft bis as well as the original RFC specify in section 1.6:

 Unicast: The media is transmitted to the source of the RTSP
          request, with the port number chosen by the client.
          Alternatively, the media is transmitted on the same
          reliable stream as RTSP.

This clearly states that the media is always send to the source of the RTSP request and not to the destination specified in the request. This actually contradicts to the section 12.40 of the bis. 
___________________________________
Roman Shpount, VP of Technology
aTelo, Inc. 
2107 Wilson Blvd., Suite 301 
Arlington, VA 22201 
PCN: 703-621-4040 for voice / fax
www.atelo.com







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From mmusic-admin@ietf.org  Thu Feb 28 17:52:34 2002
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Subject: RE: [MMUSIC] RTSP state machine
Date: Thu, 28 Feb 2002 14:49:22 -0800
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Dear Magnus:

A couple of comments on your state machine document:

1. For ready state, you have:


State: Ready-na
Action          Prerequisite    New State       Response
-----------------------------------------------------------------------
SETUP           New URL         Ready-a         Nr-media = 2
SETUP           Present URL     Ready-na        Change transport param.
Teardown        *|media URL     Init            Ok


I think in last telecon we agreed that TEARDOWN with "Presentation URL"
releases the session ID and takes the session 
to INIT state.

BTW, I don't believe TEARDOWN on "media URL", even if  Nr-media = 0 as a
result, should release
the session ID.  Suggest the addition of "Ready-No-Track" state?

2. On TIMEOUT handling:

If the server times out on a client, I would suggest that server send a
TEARDOWN to client (if TEARDOWN is allowed in the S->C direction, which is a
separate  thread of discussion).

3. On end of stream/range notification:

I agree with your proposal. Please also see the feature request I submited
to sourceforge below. Incidentally, Rob L. suggested that TEARDOWN in the
S->C direction be used to serve as notification before END_OF_STREAM  is
added to RTSP.

Best regards,
thomas

---------------cut here ---------------

Feature Requests item #522607, was opened at 2002-02-25 11:49
You can respond by visiting: 
http://sourceforge.net/tracker/?func=detail&atid=377747&aid=522607&group_id=
23194

Category: None
Group: None
Status: Open
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Nobody/Anonymous (nobody)
Summary: Need END_OF_STREAM request

Initial Comment:
Currently there is no way for Server to inform Client 
the completion of a presentation (VOD or Live) without 
issuing a TEARDOWN and thus release reources. What if 
the player wants to use the same resources (pertinent 
to its sessionID) to watch another clip? It would be 
wasteful to teardown and then acquire another 
sessionID immediately.

Would like to see a END_OF_STREAM method added on the 
S->C direction. E.g.:

END_OF_STREAM <p_URI> or <track_URL> RTSP/1.0
SESSION: <session_ID>
REASON: <text describing reason of EOS>




-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
Sent: Thursday, February 28, 2002 8:51 AM
Cc: IETF MMUSIC WG
Subject: Re: [MMUSIC] RTSP state machine


Hi Rob, Tom and Jonathan,

I have reconsidered about removing pause state and I can agree that it is
unnecessary. I attach an updated version of the state machine without the
paused
state. Instead I replaced it with setting of the pause point or as I have
called it
in the state machine resume point (RP). I think that is a better name for
it.

I do believe we have the following  issues regarding pause, play and
reaching end
of media or range.

- When a presentation reaches end of media or range. Shall it automatically
return
to ready state?

- When a presentation reaches end of media. What shall the resume point be
set to?

- When reaching end of a range, should the resume point be set to 0 or end
of
range?

- Should we add a end of clip/range notification.

I will start with the last issue. I do believe such a mechanism is need for
more
than this reason to signal the state transition. It will also be good for
the
buffer handling, retransmission, adaptation, and notification to user. So I
do
agree that we should add one

Then I do believe you are suggesting a control model like a old tape player,
I have
always thought of it in terms of CD player control. All my CD players return
to the
beginning of the disc when reaching end. But one thing we really do need to
consider is backwards compatibility. Will an RFC 2326 implemented client be
able to
use a server that would requires a pause before each play after the first? I
would
guess that it would get into trouble. Also how should this work relative to
queued
play? Even if we remove as suggested I belive it will be difficult to
combine this
model with that. Therefore I would like to have automatic return to state
READY.

I think that the resume point should be reset to 0.0 when reaching end of
media.
Actually I don't care that much about this behavior except that it should be
backwards compatible. If it does not return without a explicit play 0- can
this
break any client?

Regarding reaching end of range I have no strong opinion on what to do as
long as
it is resolved. The state machine says to set it to 0.0 for moment which is
harmonized with end of media behavior.

By removing play without range while playing and also queued plays we remove
the
functionality that has potentials for messing up at end of streams.

Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/T/VA
----------------------------------------------------------------------
Ericsson Radio Systems AB  | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se


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From mmusic-admin@ietf.org  Thu Feb 28 22:02:52 2002
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Date: Wed, 27 Feb 2002 19:38:40 -0800
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Subject: [MMUSIC] Fwd: I-D ACTION:draft-sergent-rtsp-mute-00.txt
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This is an internet-draft for MUTE and UNMUTE proposal that we discussed 
on the RTSP teleconference.  Note the open issues list at the end.  
(Please also note that I got the snail mail and telephone contact 
information wrong due to an editing error.  Oops.)  
Feedback/comments/questions from all are welcome.

Begin forwarded message:
> From: Internet-Drafts@ietf.org
> Date: Wed Feb 27, 2002  04:24:16 AM US/Pacific
> To: IETF-Announce: ;
> Subject: I-D ACTION:draft-sergent-rtsp-mute-00.txt
> Reply-To: Internet-Drafts@ietf.org
>
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
>
>
> 	Title		: MUTE and UNMUTE extension to RTSP
> 	Author(s)	: A. Narasimhan, J. Sergent
> 	Filename	: draft-sergent-rtsp-mute-00.txt
> 	Pages		: 8
> 	Date		: 26-Feb-02
> 	
> RFC 2326 [1] overloads the functionality of the PLAY and PAUSE
> methods to provide rudimentary support for 'muting' individual
> streams in an (Real Time Streaming Protocol) presentation.  The
> current specification has several inconsistencies.  In addition, we
> believe that this functionality should be optional.  This document
> specifies an extension to RTSP consisting of new MUTE and UNMUTE
> methods, providing equivalent functionality in a consistent, optional
> manner.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sergent-rtsp-mute-00.txt
>
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