webrtcbin doesn't retransmit because rtprtxsend has no payload type map

[Michiel Konstapel]

Thanks Matthew!

On 16-03-2022 01:49, Matthew Waters via gstreamer-devel wrote:
> Hi,
>
> On 16/3/22 03:09, Michiel Konstapel via gstreamer-devel wrote:
>>
>> Aha - it does appear to work if I set "bundle-policy" to 
>> "max-bundle". At least, then I see it setting up the pt map, and I 
>> see it creating/pushing rtx buffers. It doesn't seem to immediately 
>> work wonders for combating packet loss, though, but I do think it 
>> runs a little smoother.
>>
>> So RTX doesn't work with the default bundle-policy, is that correct?
>>
>
> Unknown.  If you're talking with a browser, then you can pretty much 
> always use max-bundle so the other modes don't get too much testing.  
> The whole bundle-policy thing is also up for potential removal at some 
> point in the WebRTC specification.

I figured I'd stick with default values (like bundle-policy) if I don't 
understand the impact. This is purely streaming to browsers, so I'll use 
max-bundle then. Should I create a ticket for the bug impacting the 
default policy?

>>> Looks like this is the call from 
>>> gstwebrtcbin.c:on_rtpbin_request_aux_sender, but that only assigns 
>>> stream->rtxsend AFTER the call to _set_rtx_ptmap_from_stream. 
>
> This was an issue that may have only been fixed in latest git.

I'm looking at master now, but it looks like 
<https://github.com/GStreamer/gst-plugins-bad/blob/master/ext/webrtc/gstwebrtcbin.c#L6117> 
this code path is unchanged.

By the way - do you happen to know if nack/rtx actually works when 
streaming to a browser? I figured retransmits would easily hide 1% 
packet loss. I see the browser sending nacks, and I see rtprtxsend 
pushing out buffers, but I see no visual improvement. Is there any 
statistic I can get from the browser that shows whether the retransmits 
are arriving (at all, and on time?) Would VP8 or VP9 cope with packet 
loss better than H.264?

Cheers,
Michiel
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