[Galene] Re: Building a streaming gateway for Galène

[Juliusz Chroboczek <jch@irif.fr>]

> I don't really understand what you mean by "accurate NTP times". Streams
> from Firefox or GStreamer seem to use random NTP timestamps.

The NTP timestamp is used by the receiver to synchronise audio with video.
It does not need to be accurate, but the NTP timestamps of correlated
tracks (audio and video for a single video) should use the same clock,
otherwise the receiver won't be able to resynchronise them.  This is
particularly important for talking heads (lectures and conferences).

> When dumping Firefox WebRTC traffic, I also see "Generic RTP Feeback:
> NACK" packets, which confirms the fact that GStreamer is not doing NACK
> in my script. I should see if it's possible to enable it.

This will improve things a lot.  You'll need to keep a history of recently
sent packets, I don't know if gstreamer can do that automatically.

As to keyframes — you should send periodic keyframes, but you shouldn't do
it too often (Chrome sends one every 120s).  On the other hand, you should
send a keyframe fairly quickly (within 500ms or so) whenever you receive
a PLI (Picture Loss Indication) RTCP packet.

> I am still quite new to WebRTC, so be aware that I might have done
> something stupid with the GStreamer pipeline.

It's a complicated protocol stack, don't worry too much about getting all
the bits right straight away.

-- Juliusz