[whatwg] PeerConnection feedback: RTP SSRC and data over RTP

Fri Apr 29 01:06:00 PDT 2011

Ian Hickson skrev 2011-04-22 00:27:

> On Mon, 11 Apr 2011, Justin Uberti wrote:
>>> On Tue, 29 Mar 2011, Harald Alvestrand wrote:
>>>
>>>> multiplexing of multiple data streams on the same channel using
>>>> SSRC,
>>>
>>> I don't follow. What benefit would that have?
>>
>> If you are in a conference that has 10 participants, you don't want to
>> have to set up a new transport for each participant. Instead, SSRC
>> provides an excellent way to multiplex multiple media streams over a
>> single RTP session (and network transport).
> 
> Could you elaborate on this? I have tried finding information on how SSRC
> works but cannot find anything useful. Can you point to the relevant parts
> of the RFCs that describe this mechanism maybe? I'm having trouble
> understanding how it works even for audio/video streams, let alone whether
> it would actually be appropriate for data.
> 
> 

Lets first disregard the question of data over RTP which I will comet to
below.

I am sorry that RTP on a general level is not more easily understood
without some in depth reading. I have an Internet draft which is
targeted towards people intended to write RTP payload formats it
contains an RTP overview initially, I would read from start until end of
section 3, one can skip 3.1.1.

https://datatracker.ietf.org/doc/draft-ietf-payload-rtp-howto/?include_text=1

RFC 3550 is designed with the following layers of multiplexing:

Different purposes of the media streams, like audio or video are
supposed to be multiplexed using different RTP sessions.

Within an RTP session each media source, i.e. a camera or mixed audio
stream, or microphone is its own media stream thus have its own SSRC.
The RTP sequence number and timestamp space are scoped by the SSRC. The
SSRC is also used by receivers only end-points so that they have an
identity when providing feedback.

In the below examples it becomes relevant to take the topologies the RTP
can be used into account as discussed in RFC 5117.

Some examples:

1. Peer to peer with multiple sources at one end.

A connects directly with B. A has two video cameras which he both wants
to have active at the same time. Because one is showing him and the
other is showing something he wants to show to B.

This would look like:

A: SSRC1, SSRC2 <-- Video RTP session -----> B:SSRC3

2. Audio Conference using transport relaying bridge.
So in this session you have 4 participants A, B, C ,D each having one
audio stream. There is also a transport relay (R) which will forward
what it receives from one participant to all the others.

A:SSRC1 ------         -------- B:SSRC2
             |         |
             |         |
          +--------------+
          |     Relay    |
          +--------------+
             |         |
             |         |
C:SSRC3 ------         -------- D:SSRC4

So in this case C will receive RTP flows from A with SSRC1, B with SSRC2
and D with SSRC4 all over the unicast UDP flow that it has with the
relay. C will send its RTP flow and RTCP feedback on all of the received
flows. Each of A to D decides locally if they send their audio stream or
not.

If one wants to transport optimize this one can change the relay into an
RTP mixer that then needs to actively mix the audio content. Whats is
more efficient and possible depends both on the number of session
participants and the availability of mixers.

3. RTP retransmission using SSRC multi-plexing
RTP retransmission is defined in RFC 4588 and is an RTP robustification
mechanism that allows the receiver to request retransmission and the
sender to retransmit the most important packets and only the ones that
will be received within some boundaries of time that make them still
useful. It really intended for semi-reliability in sessions with not
super strict delay requirements.

In this case any RTP packet that needs to be retransmitted needs to be
sent over a new SSRC as transmitting the same packet twice with the same
SSRC and sequence number destroys the transport feedback in RTCP one
uses a encapsulating RTP payload format and a new SSRC either in second
RTP session or in this case in the same RTP session. There are good
arguments why both models makes sense for different applications.

I hope this gives some insight into how SSRCs are used and I think
reading the chapter 2 in RFC 3550 gives some insight in the design
choices. Having a clear mind on that RTP is multi-party protocol and
review the topologies that might occur in RFC 5117 gives understanding
of that angle.

Data over RTP
-------------
When it comes to data over RTP I am hesitant to it. RTP has a model of
real-time constraints are much more important that reliability. It is
also designed around Application Level Framing, where the higher layer
creating RTP payloads is capable of creating data fragments that to as
high degree as possible is usable on its own. As currently discussed on
the RTCWEB at ietf.org mailing list there is clearly things beyond audio
and video that makes sense to define RTP payload formats for. We already
have some mouse pointer format, and real-time text chat formats for
showing characters as you type them. There is clearly more things that
has the properties that match RTP, for example some gaming updates where
the previous data is completely replaced by the next such as in
Real-time Shooters.

However, data formats that wants higher reliability or has less time
constrained requirements gets certain amount of baggage from RTP and
still needs extensions that likely are more suitable to another
protocol. Which protocol(s) are the question.

Cheers

Magnus Westerlund

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