Re: [AVTCORE] draft-ietf-avtcore-multi-party-rtt-mix Issue 1: transport

Brian Rosen <br@brianrosen.net> Fri, 15 May 2020 15:03 UTC

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From: Brian Rosen <br@brianrosen.net>
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Date: Fri, 15 May 2020 11:03:35 -0400
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Subject: Re: [AVTCORE] draft-ietf-avtcore-multi-party-rtt-mix Issue 1: transport
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I think we have to consider who has to do what.

If we are requiring all implementations to change because of other multi-party issues, then I think we should us an actual reliable protocol, and not just a “repeat enough times that the probability it gets there is high enough.  

If we aren’t asking all implementations to change for multi-party, but only the mixer, then I think that we’re sticking with T.140, 

We’re in the latter case, right?  The point of this work is don’t change the endpoints, only the conference bridge.

Brian

> On May 14, 2020, at 11:01 AM, Gunnar Hellström <gunnar.hellstrom@ghaccess.se> wrote:
> 
> I have concluded that only two of the discussed transports are realistic.
> 
> Comments below
> 
> Den 2020-05-11 kl. 12:22, skrev Gunnar Hellström:
>> In a recent e-mail, I listed 9 issues to act on in draft-ietf-avtcore-multi-party-rtt-mix-00
>> 
>> I want to deal with them one by one or in small groups. Here is number 1:
>> 
>> 1. Consider rapidly if there is any more reliable transport that is feasible to move to.
>>> (e.g. Comedia RFC 4145 and RFC 4572, or the recently approved WebRTC t140 data channel draft-ietf-mmusic-t140-data-channel-usage, or use of SAVPF with NAK and retransmission RFC 4588) 
>> 
>> It may look strange with this issue after many months as an individual draft. But I want to touch it anyway before we move on in one fixed direction.
>> 
>> T.140 and its RTP transport (RFC 2793 - later RFC 4103) were created 1998 - 2000 as the third real-time medium for human conversations beside voice and video. The idea was to give equal opportunities to persons wanting to communicate by text as the ones who use voice or video. That means real-time transmission while text is created and accepting some rare dropouts just as we do with voice and video. However, users are nowadays used to text messaging where it is customary to accept a delay and get the text complete in most cases, rather than to have loss. That user experience might be expected from real-time text as well. I do not have any strong user indications that this is the case, it is just my own thinking.
>> 
>> The reason to bring this up now, is that we seem to need to introduce the multi-party mixed format at least as a new text media subtype, text/rex instead of text/red. Then we are anyway introducing signaling complexity of similar kind that another transport will do.
>> 
>> Are any of the initially mentioned more reliable transports realistic and easily implemented in the target implementation environments: NG emergency services, 3GPP IMS MTSI, IETF RUM, and plain SIP multimedia? Or are there any other not mentioned?
>> 
>> When considering this, we should have in mind that the proposed transport should be with security so that we do not need to introduce more options to negotiate between.
>> 
>> And we shall also keep in mind that NAT traversal needs to be supported as well as multi-party-signaling through the SIP central conferencing model RFC 4353.
>> 
>> Another complexity is that current regulation requires RFC 4103 and it would be best that the finaly specified multi-party solution can be perceived as an extension to RFC 4103.
>> 
>> What can be said about the options?
>> 
>>  1. Comedia RFC 4145 and RFC 4572. Makes use of TLS for transport, so it is secured. Should use RFC 6544 ICE for TCP for NAT traversal. Requires specification of how to arrange the streams and code the sources in the multi-party environment. I do not know how well these RFCs are supported in the target environments. Seems to increase complexity.
> --Increases complexity - not selected
>> 
>> 2. draft-ietf-mmusic-t140-usage-data-channel. Has security, NAT traversal and possibility to code multi-party source. Has good opportunity for being supported in endpoint devices, because all of them are expected to support WebRTC. Maybe less supported in traditional SIP bridges.
> --A realistic solution. The base is already approved and is for a popular environment. Multi-party is briefly mentioned but should probably be a bit further specified. Should however not be the only solution. The RTP based solution is also needed.
>> 
>> 3. SAVPF with NACK and RFC 4588 retransmission. I assume this can be combined with OSRTP RFC 8643 for security negotiation. When the immediate or early feedback option can be used, this method can likely be used without redundancy to achieve a reliability enhancement. That will not work well over networks with high latency. Further study needed if redundancy or FEC is needed as complement for high latency networks. Easy to achieve up to 5 simultaneously sending users.
> --Increases complexity - not selected
>> 
>> 4. (Not mentioned in the introduction above) Use RFC 4103 plus one of the RTP based methods for multi-party source indication but just increase redundancy to one original and three (instead of two) redundant generations. Can easily be done if reliability increase is really a concern. Has low overhead. Easily applicable to OSRTP security, SIP conferencing model and ICE NAT traversal.
> --Easily done by local recommendations if 3 generations redundancy (including the original) would not be felt sufficient somewhere.
>> 
>> 
>> 5. Accept reliability that is quite good as it is with RTP with one original and two redundant generations in the RFC 2198 - style ( with one of the additional methods discussed for increasing switching performance)
>> 
> --Realistic and regarded sufficient. By move to a mixer method allowing 300 ms transmission interval, the protection against burtsy packet loss is quite good. Continue on this track.
> 
> 
> The conclusion is reflected in version -01 of the draft, just published.
> 
> Regards
> 
> Gunnar
> 
> 
>> Comments please so we can take a rapid decision and move on with one solution.
>> 
>> Regards
>> 
>> 
>> Gunnar
>> 
>> 
>> 
>> 
> -- 
> Gunnar Hellström
> GHAccess
> gunnar.hellstrom@ghaccess.se <mailto:gunnar.hellstrom@ghaccess.se>
> 
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