[Galene] Re: Building a streaming gateway for Galène

[Alexandre IOOSS <erdnaxe@crans.org>]

On 4/6/21 2:57 PM, Juliusz Chroboczek wrote:
> I'll have a look at your code as soon as I have some time.  In the
> meantime, there are three things that you need to do:
> 
> 1. send periodic RTCP Sender Reports with accurate NTP times;
> 2. react to RTCP PLI packets by inserting a keyframe;
> 3. react to RTCP NACK packets by resending the packets listed in the NACK.
> 
> (1) is essential for proper lipsynch.  (2) is what allows a stream to
> recover after a catastrophic loss event.  (3) is what allows a stream to
> recover rapidly after moderate amounts of loss.

When dumping traffic in Wireshark, I see only RTCP "Sender report" and
"Receiver report" packets. Sender reports are periodic, and sent by
groups of 2 (video then audio):

```
Real-time Transport Control Protocol (Sender Report)
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 0000 = Reception report count: 0
    Packet type: Sender Report (200)
    Length: 6 (28 bytes)
    Sender SSRC: 0x9ce1015b (2631991643)
    Timestamp, MSW: 2797058655 (0xa6b7ba5f)
    Timestamp, LSW: 1448315617 (0x56538ae1)
    [MSW and LSW as NTP timestamp: Aug 20, 1988 08:44:15.337212257 UTC]
    RTP timestamp: 866227088
    Sender's packet count: 2194185357
    Sender's octet count: 2612991028
    [RTCP frame length check: OK - 28 bytes]

Real-time Transport Control Protocol (Sender Report)
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 0000 = Reception report count: 0
    Packet type: Sender Report (200)
    Length: 6 (28 bytes)
    Sender SSRC: 0x3a724a90 (980568720)
    Timestamp, MSW: 4079490799 (0xf32816ef)
    Timestamp, LSW: 2870575121 (0xab198011)
    [MSW and LSW as NTP timestamp: Apr 10, 2029 07:53:19.668357853 UTC]
    RTP timestamp: 2166723788
    Sender's packet count: 468654353
    Sender's octet count: 2838781192
    [RTCP frame length check: OK - 28 bytes]
```

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

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.

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

Thank you for pointing out this could be a potential problem,

-- 
Alexandre Iooss