[Galene] Re: Congestion control and WebRTC [was: Logging]

[Juliusz Chroboczek <jch@irif.fr>]

> So in this instance a new flow happens when a new user joins and their
> video flow has to be established to every peer?

When a user joins, nothing much happens: remember that Galène is meant to
be usable for lectures with hundred of students.  When a user clicks
« Ready », a stream with up to two tracks is established in the client->server
direction.  Galène then determines the set of clients that have expressed
interest in this flow (through the "request" message), and establishes n -
1 streams in the server->client direction.

In the bundle case (which is what we currently do), each of the flows is
an RTP session (a UDP flow).  In the non-bundle case, each of the flows is
one or two RTP sessions, one per track.

>> We do not currently implement the delay-based controller, which causes
>> collapse if the sender is on a low-rate bufferbloated network.  That is
>> why Galène's client limits the rate to 700kbit/s by default (in the
>> « Send » entry in the side menu).

> Right, but the browsers do?

They do, and Galène provides them with all the data they need to do their
job.  So currently you have state-of-the-art congestion control in the
server->client direction, but only basic loss-based congestion control in
the client->server direction.  Which is good enough for lecturing (you
usually try to give your lecture over an uncongested link) but sometimes
suboptimal during departmental meetings (some people need to lock
themselves in the attic in order to get away from their children).

-- Juliusz