Re: [MMUSIC] Multi-party transport in draft-hellstrom-mmusic-multi-party-rtt-00.txt

[Christer Holmberg <[e-mail]>]

Hi Gunnar,

I don’t think you need to describe the SDP in an e-mail – describe it in the draft instead :)

In addition, you need to describe the RTP considerations, regarding single SSRC etc.

I think it would be good to, for each alternative, have the following sub-sections.

SDP Considerations


  *   Here you describe how the SDP looks like

RTP Considerations


  *   Here you describe things like SSRC etc.

T.140 Considerations


  *   Here you describe whether some T.140 extensions etc are needed.


Much of the text already exists in the draft, but I think putting it into separate sub-sections would make it easier to read.

Thanks!

Regards,

Christer

From: Gunnar Hellstrom <[e-mail]>
Date: Wednesday, 18 December 2019 at 0.53
To: Christer Holmberg <[e-mail]>, "[e-mail]" <[e-mail]>
Subject: Multi-party transport in draft-hellstrom-mmusic-multi-party-rtt-00.txt


Hi Christer,
Den 2019-12-17 kl. 09:01, skrev Christer Holmberg:
Hi,

Before we decide on support indicators, we need to decide on how we are going to implement multi-party RTT.
Yes, these discussions at least need to go in parallel. Note that I switched subject now when the topic is multi-party RTT transport alternatives.


Section 4 specifies different alternatives. Are we going to specify all of those, or only a subset?

Related to Section 4, it would be nice with an SDP example for each alternative.

 The problem so far has been that it has been said that multi-party RTT can be arranged with regular RTP methods, and that has provided too wide freedom of choice.

So, I want to eventually change the draft to elaborate on either one common alternative for real-time transport for both SIP/RTP and WebRTC, or one each for these two environments. The other alternatives may be briefly mentioned in a background chapter or deleted.

So, let us go through the alternatives, and try to look at both chapter 4 and 5. That is both SIP/RTP and WebRTC.

1. Control code in the T.140 stream.

4.3<https://tools.ietf.org/html/draft-hellstrom-mmusic-multi-party-rtt-00#section-4.3>.  RTP Mixer indicating participants by a control code in the stream

and

5.2<https://tools.ietf.org/html/draft-hellstrom-mmusic-multi-party-rtt-00#section-5.2>.  RTT bridging in WebRTC with one common data channel

Both build on the same principle. The presentation protocol T.140 is extendable. New one letter  function codes can be defined and sent together with a parameter framed by the control codes called SOS and ST as defined in ISO 6429. So, we could define (or rather ask ITU-T SG26 to define) a function code "c" followed by a string uniquely identifying the source of the following text (or rather T140blocks). The data for each transmission request from the mixer should start with the SOS,c,source,ST code as soon as more than two parties are involved in the session. The receiving party need to scan incoming T140blocks and direct the received text to the proper area for presentation of text from each source in one of the suitable styles for RTT presentation (see chapter 9).

For 4.3, the RTP variant, I don't think we need to prescribe any more SDP than a regular one stream RTT plus an indication of multi-party RTT capability.

Example, combining this alternative with capability negotiation 7.3

Offer

 m=text 11000 RTP/AVP 100 98

      a=rtpmap:98 t140/1000

      a=rtpmap:100 red/1000

      a=fmtp:100 98/98/98

      a=rtt-mixer:t140-mixer



Answer of a capable party

 m=text 12000 RTP/AVP 100 98

      a=rtpmap:98 t140/1000

      a=rtpmap:100 red/1000

      a=fmtp:100 98/98/98

      a=rtt-mixer:t140-mixer

The parties can start exchanging multi-party text in one stream with T.140 source indications.



Answer of an uncapable party

 m=text 12000 RTP/AVP 100 98

      a=rtpmap:98 t140/1000

      a=rtpmap:100 red/1000

      a=fmtp:100 98/98/98

The mixer can start sending multi-party text prepared for presentation to a multi-party unaware UA according to chapter 10 and Appendix A, providing a lower functionality fixed view presentation.

This method can be combined with multi-party conferencing control for SIP as said in the beginning of chapter 7


"A method for detection of conference-awareness for centralized SIP conferencing in general is specified in RFC 4579<https://tools.ietf.org/html/rfc4579> [RFC4579<https://tools.ietf.org/html/rfc4579>]. The focus sends the "isfocus" feature tag in a SIP Contact header. This causes the conference-aware UA to subscribe to conference notifications from the focus. The focus then sends notifications to the UA about entering and disappearing conference participants and their media capabilities. The information is carried XML-formatted in a 'conference-info' block in the notification according to RFC<https://tools.ietf.org/html/rfc4575> 4575<https://tools.ietf.org/html/rfc4575>. The mechanism is described in detail in RFC 4575<https://tools.ietf.org/html/rfc4575> [RFC4575<https://tools.ietf.org/html/rfc4575>]."

But I think now that we do not need to make us so dependent on that mechanism. Any way to introduce and delete parties in the conference may be used.  The draft is currently too much tied to use of the RFC 4579 / RFC 4575 procedure.



For 5.2, the WebRTC case with T.140 source marked text, according to draft-ietf-mmusic-t140-usage-data-channel the sdp would be

 Offer:



       m=application 1400 UDP/DTLS/SCTP webrtc-datachannel

       c=IN IP6 2001:db8::3

       a=max-message-size:1000

       a=sctp-port 5000

       a=dcmap:2 label="ACME customer service";subprotocol="t140"

       a=dcsa:2 rtt-mixer:t140-mixer



   Answer from a capable party:



       m=application 2400 UDP/DTLS/SCTP webrtc-datachannel

       c=IN IP6 2001:db8::1

       a=max-message-size:1000

       a=sctp-port 6000

       a=dcmap:2 label="ACME customer service";subprotocol="t140"

       a=dcsa:2 rtt-mixer:t140-mixer

The parties can start exchanging multi-party text in one stream with T.140 source indications.

(and if the negotiation does not find multi-party capability for both parties, then the mixer can start sending according to chapter 10 and Appendix A.)

The exact form of the multi-party capability indication should go into the negotiation discussion belonging to chapter 7. (maybe it should be a new fmtp - parameter instead of a new sdp attribute if using sdp )

Pros:

Equal handling for SIP/RTP and WebRTC
Simple session and stream establishment.

Cons:

The mixer need to manipulate the T.140 stream.
In SIP/RTP, if more than 3 consecutive packets are lost, the source of the loss is not known, and an indication of possible loss needs to be inserted in some other place than in the stream that had loss.
The mixer needs to be allowed to decrypt and encrypt the stream.



I need to stop here and describe the other alternatives another day.

Thanks

Gunnar

Regards,

Christer

From: mmusic <[e-mail]><mailto:[e-mail]> on behalf of Gunnar Hellstrom <[e-mail]><mailto:[e-mail]>
Date: Sunday, 15 December 2019 at 22.25
To: "[e-mail]"<mailto:[e-mail]> <[e-mail]><mailto:[e-mail]>
Subject: [MMUSIC] Negotiation in draft-hellstrom-mmusic-multi-party-rtt-00.txt


Hi,

draft-hellstrom-mmusic-multi-party-rtt/<https://datatracker.ietf.org/doc/draft-hellstrom-mmusic-multi-party-rtt/> currently contains sets of alternative solutions for different aspects of multi-party real-time text (RTT).

I would like to discuss these alternatives aspect by aspect, and eventually reduce them to one solution per aspect. By aspects, I mean negotiation for SIP, negotiation for WebRTC, source identification and multi-party transport for SIP, source identification and multi-party transport for WebRTC and mixing for the multi-party unaware case.

The main requirements are described in section 3 of the draft. One requirement is that the solution should be implementable in a near future. Point-to-point RTT solutions based on SIP and RTP with RFC 4103 transport are implemented, and there are cases where upgrade to multi-party support is urgent. It is assumed that most multi-party cases will be based on one central bridge. The media coding is according to ITU-T T.140, that is Unicode text transmitted UTF-8 transformed, with some control codes from ISO 6429. Transmission takes place without any special action by the user other than creating text. Transmission is usually time sampled in about 300 ms samples.

I want to start with the negotiation aspect. Before the bridge can start transmitting RTT in a way suitable for good presentation of multi-party RTT, it must know that the receiving party is capable of separating text from the different participants. Also the UA may need an indication that it should include source information in a specific way for multi-party RTT to work..

Therefore, a negotiation method is needed. Chapter 7 contains the following negotiation alternatives.

7.1 Implicit negotiation by the bridge expressing RTT multi-party capability indications according to RFC 4575, and the UA showing multi-party awareness only if it supports multi-party RTT.

7.2 Define a new SIP media tag indicating RTT multi-party capability. Start using multi-party RTT after a successful capability exchange.

7.3 Define a new SDP attribute on media level, indicating RTT multi-party capability. Start using multi-party RTT after a successful capability exchange.

There may be other alternatives that I have not thought about.

Pros and cons for these methods are documented in chapter 7.

I hope we can discuss the negotiation methods and soon get to a conclusion.



Regards

Gunnar




Den 2019-11-01 kl. 00:43, skrev Gunnar Hellström:

Hi mmusic and avtcore,

I have replaced an old expired draft about real-time text multi-party handling. It touches both transport and negotiation topics, so I am sending this to both mmusic and avtcore, but I hope we can keep the discussion in mmusic. So, please respond with your comments to mmusic.

Right now, the draft is a discussion paper. I intend to convert it to a true BCP format after some discussions on the preferences I have proposed.

Regards

Gunnar Hellström


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


Title : Real-time text media handling in multi-party conferences
Author : Gunnar Hellstrom
Filename : draft-hellstrom-mmusic-multi-party-rtt-00.txt
Pages : 36
Date : 2019-10-31

Abstract:
This memo specifies methods for Real-Time Text (RTT) media handling
in multi-party calls. The main solution is to carry Real-Time text
by the RTP protocol in a time-sampled mode according to RFC 4103.
The main solution for centralized multi-party handling of real-time
text is achieved through a media control unit coordinating multiple
RTP text streams into one RTP session.

Identification for the streams are provided through the RTCP
messages. This mechanism enables the receiving application to
present the received real-time text medium in different ways
according to user preferences. Some presentation related features
are also described explaining suitable variations of transmission and
presentation of text.

Call control features are described for the SIP environment. A
number of alternative methods for providing the multi-party
negotiation, transmission and presentation are discussed and a
recommendation for the main one is provided. Two alternative methods
using a single RTP stream and source identification inline in the
text stream are also described, one of them being provided as a lower
functionality fallback method for endpoints with no multi-party
awareness for RTT.

Brief information is also provided for multi-party RTT in the WebRTC
environment.

EDITOR NOTE: A number of alternatives are specified for discussion.
A decision is needed which alternatives are preferred and then how
the preferred alternatives shall be emphasized.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-hellstrom-mmusic-multi-party-rtt/

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-hellstrom-mmusic-multi-party-rtt-00
https://datatracker.ietf.org/doc/html/draft-hellstrom-mmusic-multi-party-rtt-00


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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

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