Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage

[Harald Alvestrand <[e-mail]>]

Thanks for the update!

It does seem reasonably clear now - clear enough that I found one point 
that I violently disagree with (if I'm interpreting it correctly). See 
below.

I don't have any negative comments on the rest of the section.

On 04/04/2014 11:24 AM, Magnus Westerlund wrote:
> WG,
> (As Author)
>
> Colin and I have been working on resolving terminology usage and in
> general giving the draft a polish over before the WG last call. One
> section that has gotten quite some attention from us is the below one.
> The changes are significantly from an editorial stand point. However,
> the intended content has not been intended to be changed, although
> clarified and better motivated. So please review it, we intended to
> include this in the upcoming draft update. Feedback appreciated.
>
> 5.1.  Conferencing Extensions and Topologies
>
>     RTP is a protocol that inherently supports group communication.
>     Groups can be implemented by having each endpoint sending its RTP
>     packet streams to an RTP middlebox that redistributes the traffic, by
>     using a mesh of unicast RTP packet streams between endpoints, or by
>     using an IP multicast group to distribute the RTP packet streams.
>     These topologies can be implemented in a number of ways as discussed
>     in [I-D.ietf-avtcore-rtp-topologies-update].
>
>     While the use of IP multicast groups is popular in IPTV systems, the
>     topologies based on RTP middleboxes are dominant in interactive video
>     conferencing environments.  Topologies based on a mesh of unicast
>     transport-layer flows to create a common RTP session have not seen
>     widespread deployment.  Accordingly, WebRTC implementations are not
>     expected to support topologies based on IP multicast groups.  WebRTC
>     implementations are also not expected to support mesh-based
>     topologies, such as a point-to-multipoint mesh configured as a single
>     RTP session (Topo-Mesh in the terminology of
>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>     multipoint mesh constructed using several RTP sessions, in the WebRTC
>     context using independent RTCPeerConnections can be expected to be
>     utilised by WebRTC applications.
>
>     WebRTC implementations of RTP endpoints implemented according to this
>     memo are expected to support all the topologies described in
>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>     and receive unicast RTP packet streams to some peer device, provided
>     that peer can participate in performing congestion control on the RTP
>     packet streams.  The peer device could be another RTP endpoint, or it
>     could be an RTP middlebox that redistributes the RTP packet streams
>     to other RTP endpoints.  This limitation means that some of the RTP
>     middlebox-based topologies are not suitable for use in the WebRTC
>     environment.  Specifically:
>
>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>        since they make the use of RTCP for congestion control and quality
>        of service reports problematic (see Section 3.6.2 of
>        [I-D.ietf-avtcore-rtp-topologies-update]).
>
>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>        terminating-MCU) SHOULD NOT be used since they break RTP loop
>        detection, and prevent receivers from identifying active senders
>        (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).

In my (strong) opinion: This recommendation is wrong.

All central conferencing units that use individual RTP sessions with 
clients fall into this category. This includes, I believe, every single 
multiuser video conferencing system based on WebRTC in existence (and 
there are many of them).

I believe the recommendation should be:

* Content modifying MCUs with RTCP termination 
(Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP 
loop detection and identification of active senders is the resposibility 
of the application; since the clients are isolated from each other at 
the RTP layer, RTP cannot assist with these functions.

>
>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>        SHOULD NOT be used because its safe use requires a point to
>        multipoint congestion control algorithm or RTP circuit breaker,
>        which has not yet been standardised.
>
>     The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
>     designed to be used with centralised conferencing, where an RTP
>     middlebox (e.g., a conference bridge) receives a participant's RTP
>     packet streams and distributes them to the other participants.  These
>     extensions are not necessary for interoperability; an RTP end-point
>     that does not implement these extensions will work correctly, but
>     might offer poor performance.  Support for the listed extensions will
>     greatly improve the quality of experience and, to provide a
>     reasonable baseline quality, some of these extensions are mandatory
>     to be supported by WebRTC end-points.
>
>     The RTCP conferencing extensions are defined in Extended RTP Profile
>     for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
>     AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
>     Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
>     usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
>
>
> Cheers
>
> Magnus Westerlund
>
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