<div dir="ltr"><div>I have installed python3-gi package from apt repository, version 3.42.0-3build1.</div><div>Yes, value for the enum changes each run.</div><div>Could you give advice on how to check if this is a python binding problem or lower-level one?<br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">вт, 5 июл. 2022 г. в 16:02, Matthew Waters <<a href="mailto:ystreet00@gmail.com">ystreet00@gmail.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
Please keep the list in CC so others searching for the same issue
might find a solution.<br>
<br>
PyGObject is not gst-python. gst-python provides some convenience
API on top of PyGObject and may be required in this case.<br>
<br>
That log is expected for your case (no existing transceiver or pad).<br>
<br>
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.<br>
<br>
Cheers<br>
-Matt<br>
<br>
<div>On 5/7/22 22:41, Alexei Golovko wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>Here is the log with GST_DEBUG=3,*webrtc*:7 — <a href="https://pastebin.com/RshtnqQx" target="_blank">https://pastebin.com/RshtnqQx</a><br>
</div>
<div>I use PyGObject 3.42.1.</div>
<div><br>
</div>
<div>I see in logs following lines:</div>
<div>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<br>
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)</div>
<div><br>
</div>
<div>Maybe they are related to the problem, though I am not
sure.<br>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">вт, 5 июл. 2022 г. в 15:12,
Matthew Waters <<a href="mailto:ystreet00@gmail.com" target="_blank">ystreet00@gmail.com</a>>:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi,<br>
<br>
That SDP doesn't actually exist as a valid value. You can
enable<br>
GStreamer debugging with GST_DEBUG=3,*webrtc*:7 and that may
show<br>
something interesting.<br>
<br>
Otherwise, one has to figure out if the python bindings are
doing<br>
something weird here. I assume you have a relevant gst-python<br>
installation available and in use.<br>
<br>
Cheers<br>
-Matt<br>
<br>
On 5/7/22 02:44, Alexei Golovko via gstreamer-devel wrote:<br>
> Hi,<br>
> I try to use webrtcbin with Janus.<br>
> I've found examples like<br>
> <a href="https://github.com/centricular/gstwebrtc-demos/blob/master/janus/janusvideoroom.py" rel="noreferrer" target="_blank">https://github.com/centricular/gstwebrtc-demos/blob/master/janus/janusvideoroom.py</a>,<br>
> <a href="https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/tests/examples/webrtc/webrtcrenego.c" rel="noreferrer" target="_blank">https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/blob/master/tests/examples/webrtc/webrtcrenego.c</a>,<br>
> <a href="https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst-rust/src/main.rs" rel="noreferrer" target="_blank">https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/gst-rust/src/main.rs</a>,<br>
> <a href="https://gitlab.freedesktop.org/gstreamer/gst-examples/-/blob/master/webrtc/sendrecv/gst/webrtc_sendrecv.py" rel="noreferrer" target="_blank">https://gitlab.freedesktop.org/gstreamer/gst-examples/-/blob/master/webrtc/sendrecv/gst/webrtc_sendrecv.py</a>.<br>
><br>
> They tend to create an offer locally, send it to Janus,
receive an<br>
> answer and set it as a remote description. I would like
to mimic<br>
> behaviour, which we used to use with python's aiortc: get
an offer<br>
> from Janus, generate an answer and send it back.<br>
><br>
> I set SDP offer from Janus with set-remote-description,
then emit<br>
> create-answer; but I get an answer
WebRTCSessionDescription with an<br>
> empty SDP and a very strange WebRTCSDPType=-43384639
(non-existing enum?).<br>
><br>
> What am I doing wrong and how to fix this?<br>
><br>
> The python script used is here: <a href="https://pastebin.com/WGPcF64m" rel="noreferrer" target="_blank">https://pastebin.com/WGPcF64m</a><br>
> Module dvr_processor.dvr_processor.janus contains
communication with<br>
> Janus: <a href="https://pastebin.com/KAhTv8Lq" rel="noreferrer" target="_blank">https://pastebin.com/KAhTv8Lq</a><br>
> Logs are here: <a href="https://pastebin.com/QVegDhRZ" rel="noreferrer" target="_blank">https://pastebin.com/QVegDhRZ</a><br>
><br>
> It is gstreamer 1.20.1 on ubuntu-22.04 based image.<br>
><br>
> Regards,<br>
> Alexei Golovko.<br>
<br>
</blockquote>
</div>
</blockquote>
<br>
</div>
</blockquote></div>