Hi all, I use webrtcbin for video and audio streaming (see my pipeline below). The issue that when webrtcbin generates SDP offers in 60% cases a:ssrc parameter is missed for video stream. In 40% cases it's generated. It's important because Android client does not create remote video track if ssrc is missed. So how to force webrtcbin to generate ssrc all the time? Here is my pipeline: #define WEBRTC_PIPELINE "webrtcbin name=webrtc " STUN_SERVER_PROP "=" STUN_1 " " STUN_SERVER_PROP "=" STUN_2 " " STUN_SERVER_PROP "=" STUN_3 " " \ STUN_SERVER_PROP "=" STUN_4 " " STUN_SERVER_PROP "=" STUN_5 " " TURN_SERVER_PROP "=" TURN_1 " " \ "v536videosrc sys-init=false push_mode=true device=1 channel=2 encoder=2 format=H264 width=640 height=480 ! video/x-h264, stream-format=byte-stream, alignment=au, profile=baseline ! rtph264pay rtp-h264aggregate-mode=2 ! capsfilter caps=" RTP_CAPS_H264 "96 ! queue leaky=downstream ! webrtc. " \ "alsasrc ! queue leaky=downstream ! audioconvert ! opusenc ! rtpopuspay ! capsfilter caps=" RTP_CAPS_OPUS "97 ! webrtc. " _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
You need to wait until webrtcbin's sink pads get caps from upstream. You can connect to the notify::caps signal on both sink pads and verify caps are there before generating the offer. On Thu, Feb 4, 2021 at 9:38 AM Vladimir Tyutin <[hidden email]> wrote:
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Thanks Trey, It seems you are right! Because if I set delay in 1 sec ssrc is added all the time. Thanks for your advice I will implement it On Thu, Feb 4, 2021 at 6:46 PM Trey Hutcheson <[hidden email]> wrote:
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