WebRTC generate invalid rtpmap for H264
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WebRTC generate invalid rtpmap for H264
 
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Hi all,
I found a pretty strange issue webrtcbin generates the following initial SDP offer with a negative clock rate: v=0 o=- 5716970113760455593 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle a=msid-semantic:WMS webrtc_sink m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:0ybqQZNeS9CznnGxJxW/BpKyapc+CXsL a=ice-pwd:ffvVDjwSJ7/+AAdh3MjUMMhz91xwZZGI a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 H264/-1784864736 a=rtcp-fb:96 nack pli a=mid:video0 a=fingerprint:sha-256 9B:11:48:BE:41:FF:CA:FE:62:B1:98:07:7B:E6:80:2A:56:9A:56:DC:A2:60:81:27:DE:EC:F5:76:C1:C4:39:51 Actually it is more like a random number it changes in every session setup. However as I see it should be set to 9000. My pipleine is pretty simple: nvarguscamerasrc sensor-id=0 sensor-mode=0 gainrange="1 16" ispdigitalgainrange="1 1" ! video/x-raw(memory:NVMM), width=(int)1920, height=(int)1080, format=(string)NV12, framerate=(fraction)30/1 ! nvvidconv ! nvv4l2h264enc bitrate=8000000 control-rate=1 insert-sps-pps=true ! rtph264pay mtu=1400 ! webrtcbin name=webrtc_sink latency=0 And on the application side Im handling the singling. Do you have any idea how to fix this? What I am missing here? Thanks, Peter _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Re: WebRTC generate invalid rtpmap for H264
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Hi, On Fri, 2021-04-09 at 19:22 +0200, Peter Biro wrote:
This is definitely a bug in webrtcbin, it shouldn't be generating an SDP with an invalid clock rate. Can you run it with "gst-launch-1.0 -v" to see wahtr caps the payloaders set on webrtcbin ? Do you set any codec preferences on webrtcbin ? Olivier
-- Olivier Crête _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Re: WebRTC generate invalid rtpmap for H264
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Hi,
Im not sure if if can run it from terminal (I dont know how to do the signalling that way) but I can setup debug logging if it that ok? I dont set any codec preferences on the webrtc. Strange thing this is only appears if I use H264, I just tried to run the very same pipeline with VP8 codec the SDP looks alright (but unfortunately I have to use H264 so this wont be a workaround for my use case): v=0 o=- 711117265011778833 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle a=msid-semantic:WMS webrtc_sink m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:Qfsq+80LvU2YdnPZzVjuA4/UrODXoKrN a=ice-pwd:HWOOU8kI1droNmky2tjnGekC1NquWNmH a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 nack pli a=ssrc:2035319666 msid:user1666440897@host-b40b4b48 webrtctransceiver0 a=ssrc:2035319666 cname:user1666440897@host-b40b4b48 a=mid:video0 a=fingerprint:sha-256 C2:C3:5D:14:8F:06:44:8A:0E:A8:76:4B:D8:46:5D:BF:31:73:F1:CB:32:F7:8B:F8:F5:D1:D9:58:1D:3C:5D:05 Thanks, Peter
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Re: WebRTC generate invalid rtpmap for H264
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Hi, Oh right, you're using webrtcbin.. Erm, you can enable the webrtcbin debug category and it should print the caps that it gets. Olivier On Fri, 2021-04-09 at 20:02 +0200, Peter Biro wrote:
-- Olivier Crête _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Re: WebRTC generate invalid rtpmap for H264
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ohh gosh my bad, user error.
So originally I had a 'producer' pipeline: nvarguscamerasrc sensor-id=<DCL_SENSOR_ID> sensor-mode=0 gainrange="1 16" ispdigitalgainrange="1 1" ! video/x-raw(memory:NVMM), width=(int)1920, height=(int)1080, format=(string)NV12, framerate=(fraction)20/1 ! nvvidconv ! textoverlay name=text_overlay ! video/x-raw,format=I420 ! nvvidconv ! nvv4l2h264enc bitrate=8000000 control-rate=1 insert-sps-pps=true ! rtph264pay mtu=1400 ! udpsink auto-multicast=true clients=<clinets> Which multicasted the RTP stream to clients and on the client side the WebRTC pipeline looked like this: udpsrc multicast-group=<source> auto-multicast=true port=<port> ! application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin name=webrtc_sink stun-server='<a href="stun://10.51.255.237:3478" class="">stun://10.51.255.237:3478' latency=0 But for testing purposes I tried to simplify the testcase and move the two parts into one pipeline, but somehow I did not applied the changes so the client still got the stream from the network.. My bad. Sorry for the confusion! After adding the clock-rate=90000 property on the client side: udpsrc multicast-group=<source> auto-multicast=true port=<port> ! application/x-rtp,media=video,encoding-name=H264,payload=96,clock-rate=90000 ! webrtcbin name=webrtc_sink stun-server='<a href="stun://10.51.255.237:3478" class="">stun://10.51.255.237:3478' latency=0 The SDP looks much better: v=0 o=- 8174947542390535236 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle a=msid-semantic:WMS webrtc_sink m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:9I6XsU/+eajaGQOZBYqjgcJp5/PiMRc9 a=ice-pwd:CJJO5TCzJSivnaaLU2b/ohif4OXj+fBj a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 H264/90000 a=rtcp-fb:96 nack pli a=mid:video0 a=fingerprint:sha-256 F9:13:FF:68:E4:83:59:A0:49:A3:5D:E1:BA:CD:15:0E:DE:95:14:0D:F5:F0:88:7A:63:28:D7:CD:0A:7E:C4:CD But chrome still cannot play WebRTC video on the frontend side, when I compare the two SDPs (the one from the merged into one pipeline and the one when the RTP stream is coming from the UDP sink) a=framerate:20 a=fmtp:96 packetization-mode=1;profile-level-id=424015;sprop-parameter-sets=Z0JAKJWgHgCJ+VA=,aM48gA== a=ssrc:1897942919 msid:user2169902196@host-6f4d07ad webrtctransceiver0 a=ssrc:1897942919 cname:user2169902196@host-6f4d07ad So tried to add them to the properties: udpsrc multicast-group=< source > auto-multicast=true port=<port> ! application/x-rtp,media=video,encoding-name=H264,payload=96,clock-rate=90000,framerate=20,packetization-mode=1,profile-level-id=424015,sprop-parameter-sets="Z0JAKJWgHgCJ+VA\=\,aM48gA\=\=" ! webrtcbin name=webrtc_sink stun-server='<a href="stun://10.51.255.237:3478" class="">stun://10.51.255.237:3478' latency=0 But only the 'sprop-parameter-sets' property finds its way to the SDP, so the final (still not working) SDP: v=0 o=- 8318672537835092582 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle a=msid-semantic:WMS webrtc_sink m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:wb9zs9ZzaiIUHHDrZSoJk7zqM7IEKUpD a=ice-pwd:6OaD73grf0TKP5l05/UZ2BUbNx33+SDU a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 H264/90000 a=rtcp-fb:96 nack pli a=fmtp:96 sprop-parameter-sets=Z0JAKJWgHgCJ+VA=,aM48gA== a=mid:video0 a=fingerprint:sha-256 71:00:FF:12:E9:E8:F0:9F:54:0B:5C:B7:52:9E:41:4C:79:A5:8A:A9:07:49:C8:BF:CC:4C:4D:5A:86:4A:06:FE How can I add the missing parameters to the SDP? Also Im not sure if I do any good by just inserting them (for me sprop-parameter-sets looks like a random thing) so Im not sure it profile-level-id and sprop-parameter-sets is a static thing or should be a variable. Thank you for the help! Bests, Peter Just for the reference the working SDP: v=0 o=- 7463942138372302405 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle a=msid-semantic:WMS webrtc_sink m=video 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:9oG0AHRZpwiykWEsKz1YRInr6Vbu+SYJ a=ice-pwd:QQYgqeYA2ckB2jCcCXKdmrcSS7Sb4z+O a=sendrecv a=rtcp-mux a=rtcp-rsize a=rtpmap:96 H264/90000 a=rtcp-fb:96 nack pli a=framerate:20 a=fmtp:96 packetization-mode=1;profile-level-id=424015;sprop-parameter-sets=Z0JAKJWgHgCJ+VA=,aM48gA== a=ssrc:1897942919 msid:user2169902196@host-6f4d07ad webrtctransceiver0 a=ssrc:1897942919 cname:user2169902196@host-6f4d07ad a=mid:video0 a=fingerprint:sha-256 72:5D:78:02:41:8B:FA:07:77:DB:EB:23:05:E9:3C:BF:D3:59:0C:01:75:24:9C:51:AE:33:9E:27:A3:98:B5:19
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