<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div dir="ltr"></div><div dir="ltr"><div dir="ltr" style="-webkit-text-size-adjust: auto;">So I’ve checked rtpjitterbuffer source code and it doesn’t set SSRC, but neither does anyone (rtprtxreceive, rtpssrcdemux etc). Is this a bug?</div><div dir="ltr" style="-webkit-text-size-adjust: auto;"> https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c#L4136 </div></div><div dir="ltr"><blockquote type="cite">19 авг. 2022 г., в 13:13, David Grinberg <green.grinya@gmail.com> написал(а): </blockquote></div><blockquote type="cite"><div dir="ltr">Setting profiles and payload map didn’t help :( But I’ve found out that GstRTPRetransmissionRequest does not contain SSRC. For now a trivial fix is to manually set it inside some probe, but it would be nice to know why that happens, my guess is that it is rtpssrcdemux (which i dont use) who sets ssrc <blockquote type="cite">19 авг. 2022 г., в 00:28, Olivier Crête <olivier.crete@collabora.com> написал(а): </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">Hi, </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">It seems like you are completely right that it's not used. </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">Your code also seems to be missing the "payload-type-map" property on </blockquote><blockquote type="cite">the rtxrtpreceive element, but that shouldn't prevent the requests from </blockquote><blockquote type="cite">being sent. </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">I think you also need to set the rtp-profile property on rtpsession to </blockquote><blockquote type="cite">avfp or savpf. </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">Olivier </blockquote><blockquote type="cite"> </blockquote><blockquote type="cite"><blockquote type="cite">On Thu, 2022-08-18 at 23:43 +0400, David Grinberg via gstreamer-devel </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">wrote: </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">Hi, I’ve already asked my question here - </blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1386 </blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">Suggestion of adding rtcp-fb-nack=(boolean)true didn’t work, also </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">there isn’t any reference to rtcp-fb-nack in rtpsession code (though </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">it is used by webrtcbin). </blockquote></blockquote><blockquote type="cite"><blockquote type="cite">The problem is that find_source fails and I don’t really know why </blockquote></blockquote><blockquote type="cite"> </blockquote><blockquote type="cite">-- </blockquote><blockquote type="cite">Olivier Crête </blockquote><blockquote type="cite">olivier.crete@collabora.com </blockquote></div></blockquote></body></html>