Subject: [MMUSIC] Negotiation in draft-hellstrom-mmusic-multi-party-rtt-00.txt

From: Gunnar Hellström <[email-masked]>
Date: Sun, 15 Dec 2019 20:24:36 +0000

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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Ämne:   I-D Action: draft-hellstrom-mmusic-multi-party-rtt-00.txt
Datum:  Thu, 31 Oct 2019 16:33:35 -0700
Från:   [email-masked]<mailto:[email-masked]>
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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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Gunnar Hellström
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