webrtcbin create-answer returns unknown SDP type

[Matthew Waters]

Please keep the list in CC so others searching for the same issue might
find a solution.

PyGObject is not gst-python.  gst-python provides some convenience API
on top of PyGObject and may be required in this case.

That log is expected for your case (no existing transceiver or pad).

The value for the enum is different from your last run.  This points to
uninitialized/random memory being accessed.  You would need to figure
out if that's in the python translation layer or not.

Cheers
-Matt

On 5/7/22 22:41, Alexei Golovko wrote:
> Here is the log with GST_DEBUG=3,*webrtc*:7 —
> https://pastebin.com/RshtnqQx
> I use PyGObject 3.42.1.
>
> I see in logs following lines:
> 2022-07-05 15:15:57.949 jord ERROR| 0:00:01.214992971     9
> 0x55e2cbd19980 DEBUG              webrtcbin
> gstwebrtcbin.c:1874:_find_codec_preferences:<webrtctransceiver0> Could
> not find caps for mline 0
> 2022-07-05 15:15:57.949 jord ERROR| 0:00:01.215007126     9
> 0x55e2cbd19980 TRACE              webrtcbin
> gstwebrtcbin.c:4071:_create_answer_task:<wrb> trying to compare
> application/x-rtp, media=(string)video, payload=(int)96,
> clock-rate=(int)90000, encoding-name=(string)H264,
> profile-level-id=(string)42e01f, packetization-mode=(string)1,
> rtcp-fb-nack=(boolean)true, rtcp-fb-nack-pli=(boolean)true,
> rtcp-fb-goog-remb=(boolean)true; application/x-rtp,
> media=(string)video, payload=(int)97, clock-rate=(int)90000,
> encoding-name=(string)RTX, apt=(string)96 and (NULL)
>
> Maybe they are related to the problem, though I am not sure.
>
> вт, 5 июл. 2022 г. в 15:12, Matthew Waters <ystreet00 at gmail.com>:
>
>     Hi,
>
>     That SDP doesn't actually exist as a valid value.  You can enable
>     GStreamer debugging with GST_DEBUG=3,*webrtc*:7 and that may show
>     something interesting.
>
>     Otherwise, one has to figure out if the python bindings are doing
>     something weird here.  I assume you have a relevant gst-python
>     installation available and in use.
>
>     Cheers
>     -Matt
>
>     On 5/7/22 02:44, Alexei Golovko via gstreamer-devel wrote:
>     > Hi,
>     > I try to use webrtcbin with Janus.
>     > I've found examples like
>     >
>     https://github.com/centricular/gstwebrtc-demos/blob/master/janus/janusvideoroom.py,
>     >
>     https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/tests/examples/webrtc/webrtcrenego.c,
>     >
>     https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst-rust/src/main.rs,
>     >
>     https://gitlab.freedesktop.org/gstreamer/gst-examples/-/blob/master/webrtc/sendrecv/gst/webrtc_sendrecv.py.
>     >
>     > They tend to create an offer locally, send it to Janus, receive an
>     > answer and set it as a remote description. I would like to mimic
>     > behaviour, which we used to use with python's aiortc: get an offer
>     > from Janus, generate an answer and send it back.
>     >
>     > I set SDP offer from Janus with set-remote-description, then emit
>     > create-answer; but I get an answer WebRTCSessionDescription with an
>     > empty SDP and a very strange WebRTCSDPType=-43384639
>     (non-existing enum?).
>     >
>     > What am I doing wrong and how to fix this?
>     >
>     > The python script used is here: https://pastebin.com/WGPcF64m
>     > Module dvr_processor.dvr_processor.janus contains communication with
>     > Janus: https://pastebin.com/KAhTv8Lq
>     > Logs are here: https://pastebin.com/QVegDhRZ
>     >
>     > It is gstreamer 1.20.1 on ubuntu-22.04 based image.
>     >
>     > Regards,
>     > Alexei Golovko.
>
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