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google-chrome - 需要哪些gstreamer管道设置来处理chrome的rtp流？

我正在使用webrtc将浏览器中的网络摄像头流式传输到服务器，该服务器可以进行以下设置:

使用Firefox和经过修改的echo-test html from janus gateway的

  • ，我将网络摄像头流发送到janus服务器
  • janus服务器正在使用改良的echotest插件运行，该插件仅将janus_videorecv_incoming_rtp()中的给定char * buf udp流传输到端口5060，仅用于测试目的(pretty much like this)
  • 以下gstreamer命令行实际上打开一个窗口，显示流式视频
    • GST_DEBUG=p*:5 gst-launch-1.0 -vvv udpsrc caps="application/x-rtp,media=video,clock-rate=90000,payload=96" port=5060 ! rtpvp8depay ! vp8dec ! autovideosink
    在修改后的echo测试javascript中，我从sdp答案中删除了几行，浏览器将收到如下内容:
    //jsep.sdp = jsep.sdp.replace(/a=rtcp-mux[^\s]*\s*/g, '');
jsep.sdp = jsep.sdp.replace(/a=rtpmap[^\s]*\s*red[^\s]*\s*/g, '');
jsep.sdp = jsep.sdp.replace(/a=rtpmap[^\s]*\s*ulpfec[^\s]*\s*/g, '');
jsep.sdp = jsep.sdp.replace(/a=fmtp[^\r\n]*\r*\n*/g, '');
jsep.sdp = jsep.sdp.replace(/a=rtcp-fb[^\s]*\s*goog-remb[^\s]*\s*/g, '');

    在下面，可以找到适用于上述gstreamer命令的修改后的firefox sdp答案
    但是，以同样的方式，修改后的sdp答案不适用于chrome
    我考虑过调整gstreamer帽中的有效载荷，但是32,33,96,100,120没有用

    所以问题是:如果使用 Chrome ，需要什么才能使其正常工作？

    我也尝试将janus的videoroom.c中的fir / pli请求添加为suggested here

    如果是chrome，则gstreamer输出为，命令仅在最后一行等待:
    0:00:00.025791492 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:219:gst_pipeline_init:<GstPipeline@0x1962180> set bus <bus2> on pipeline
Setting pipeline to PAUSED ...
0:00:00.029798090 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:282:reset_start_time:<pipeline0> reset start_time to 0
Pipeline is live and does not need PREROLL ...
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, payload=(int)96, encoding-name=(string)VP8-DRAFT-IETF-01
Setting pipeline to PLAYING ...
0:00:00.030045034 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:377:gst_pipeline_change_state:<pipeline0> selecting clock and base_time
0:00:00.030053523 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:398:gst_pipeline_change_state:<pipeline0> Need to update start_time
0:00:00.030058181 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:403:gst_pipeline_change_state:<pipeline0> Need to update clock.
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:src: caps = video/x-vp8, framerate=(fraction)0/1
/GstPipeline:pipeline0/GstVP8Dec:vp8dec0.GstPad:sink: caps = video/x-vp8, framerate=(fraction)0/1
0:00:00.030111345 22279      0x1954b90 DEBUG               pipeline gstpipeline.c:443:gst_pipeline_change_state:<pipeline0> start_time=0:00:00.000000000, now=33:52:04.529345754, base_time 33:52:04.529345754
/GstPipeline:pipeline0/GstRtpVP8Depay:rtpvp8depay0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, payload=(int)96, encoding-name=(string)VP8-DRAFT-IETF-01
New clock: GstSystemClock

    Chrome 答案:
    v=0
o=- 8913399741269897639 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS janus
m=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126
a=mid:audio
c=IN IP4 192.168.0.1
a=sendrecv
a=rtcp-mux
a=ice-ufrag:l0n9
a=ice-pwd:r1elT1Ew8lP3TNlzwAHUsC
a=ice-options:trickle
a=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44
a=setup:active
a=connection:new
a=rtpmap:111 opus/48000/2
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:600390024 cname:janusaudio
a=ssrc:600390024 msid:janus janusa0
a=ssrc:600390024 mslabel:janus
a=ssrc:600390024 label:janusa0
a=candidate:1 1 udp 2013266431 192.168.0.1 45728 typ host
m=video 1 RTP/SAVPF 100 116 117 96
a=mid:video
c=IN IP4 192.168.0.1
a=sendrecv
a=rtcp-mux
a=ice-ufrag:l0n9
a=ice-pwd:r1elT1Ew8lP3TNlzwAHUsC
a=ice-options:trickle
a=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44
a=setup:active
a=connection:new
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtpmap:96 rtx/90000
a=ssrc-group:FID 3188003624 3419969288
a=ssrc:677441062 cname:janusvideo
a=ssrc:677441062 msid:janus janusv0
a=ssrc:677441062 mslabel:janus
a=ssrc:677441062 label:janusv0
a=candidate:1 1 udp 2013266431 192.168.0.1 45728 typ host
m=application 0 DTLS/SCTP 0
c=IN IP4 192.168.0.1

    Firefox的答案:
    v=0
o=Mozilla-SIPUA-32.0.3 11426 0 IN IP4 127.0.0.1
s=SIP Call
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS janus
m=audio 1 RTP/SAVPF 109 0 8 101
a=mid:audio
c=IN IP4 192.168.0.1
a=sendrecv
a=rtcp-mux
a=ice-ufrag:BRBU
a=ice-pwd:2W4fGNr//HejhiC4UIabW6
a=ice-options:trickle
a=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44
a=setup:active
a=connection:new
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ssrc:3725983979 cname:janusaudio
a=ssrc:3725983979 msid:janus janusa0
a=ssrc:3725983979 mslabel:janus
a=ssrc:3725983979 label:janusa0
a=candidate:1 1 udp 2013266431 192.168.0.1 56574 typ host
m=video 1 RTP/SAVPF 120
a=mid:video
c=IN IP4 192.168.0.1
a=sendrecv
a=rtcp-mux
a=ice-ufrag:jZ5b
a=ice-pwd:dQQej9UIpPl5zuXBQNg3Nz
a=ice-options:trickle
a=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44
a=setup:active
a=connection:new
a=rtpmap:120 VP8/90000
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=ssrc:1425382999 cname:janusvideo
a=ssrc:1425382999 msid:janus janusv0
a=ssrc:1425382999 mslabel:janus
a=ssrc:1425382999 label:janusv0
a=candidate:2 1 udp 2013266431 192.168.0.1 39063 typ host
m=application 0 DTLS/SCTP 0
c=IN IP4 192.168.0.1

    更新:
    我修改了sdp-answer，使Firefox和chrome几乎都一样
    除了我只是从sdp-offer复制的“o =”和“s =”行之外v=0o=- 7589782217972865757 2 IN IP4 127.0.0.1s=-t=0 0a=group:BUNDLE audio videoa=msid-semantic: WMS janusm=audio 1 RTP/SAVPF 111a=mid:audioc=IN IP4 192.168.0.1a=sendrecva=rtcp-muxa=ice-ufrag:g0kZa=ice-pwd:d5oEody1jqIzDYUdf1fg6ta=ice-options:tricklea=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44a=setup:activea=connection:newa=rtpmap:111 opus/48000/2a=ssrc:1038736511 cname:janusaudioa=ssrc:1038736511 msid:janus janusa0a=ssrc:1038736511 mslabel:janusa=ssrc:1038736511 label:janusa0a=candidate:1 1 udp 2013266431 192.168.0.1 51232 typ hostm=video 1 RTP/SAVPF 100a=mid:videoc=IN IP4 192.168.0.1a=sendrecva=rtcp-muxa=ice-ufrag:g0kZa=ice-pwd:d5oEody1jqIzDYUdf1fg6ta=ice-options:tricklea=fingerprint:sha-256 C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44a=setup:activea=connection:newa=rtpmap:100 VP8/90000a=rtcp-fb:100 ccm fira=rtcp-fb:100 nacka=rtcp-fb:100 nack plia=rtcp-fb:100 goog-remba=ssrc:2455978689 cname:janusvideoa=ssrc:2455978689 msid:janus janusv0a=ssrc:2455978689 mslabel:janusa=ssrc:2455978689 label:janusv0a=candidate:1 1 udp 2013266431 192.168.0.1 51232 typ hostm=application 0 DTLS/SCTP 0c=IN IP4 192.168.0.1

    最佳答案

    我已经更新了包含the bidirectional streaming plugin的fork，向您展示了一个有效的示例(我已经在debian jessie上进行了测试)。

    这是我的插件更改指示

  • 确保从chrome
    • 请求关键帧之前，将gstreamer管道设置为接收
  • 在准备好webrtc媒体时请求关键帧(有关详细信息，请参见janus_bidirectional_streaming_setup_media函数)
  • 不要使用rtpbin gstreamer元素来处理传入流。由于某些原因，设置上限的方法实际上无法正常工作，并且管道会崩溃。如果您确实获得了rtp数据包并将其发送到端口，则以下管道可以正常工作:gst-launch-1.0 udpsrc port=<your listener> caps="application/x-rtp, clock-rate=90000, payload=100" ! rtpvp8depay ! vp8dec ! autovideosink sync=false async=false

    • 从理论上讲，直接将缓冲区推入插件内的appsrc也应该起作用。

    09-25 18:55
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