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I'm streaming a webcam from a browser using webrtc to a server where the following setup works:

  1. using firefox and a modified echo-test html from janus gateway I send the webcam stream to a janus server
  2. the janus server is running using a modified echotest plugin which simply udp-streams the given char *buf in janus_videorecv_incoming_rtp() to port 5060, just for testing purpose (pretty much like this)
  3. the following gstreamer command line actually opens a window showing the streaming video

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

in the modified echo test javascript i remove a few lines from the sdp answer the browser will receive like the following:

//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, '');

below, one can find the modified firefox sdp answer which works for above gstreamer command but the, in the same way, modified sdp answer doesnt work in case of chrome i thought about adjusting the payload in the gstreamer caps, but 32,33,96,100,120 didnt work

so the question is: what is needed in case of chrome to get this to work?

i also tried adding fir/pli requests like in videoroom.c from janus as suggested here

the gstreamer output in case of chrome is, where the command just keeps waiting at the last line:

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 answer:

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 answer:

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

UPDATE: i modified the sdp-answer so both firefox and chrome get nearly the same except for the "o=" and "s=" lines which i just copy from the sdp-offer v=0 o=- 7589782217972865757 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 a=mid:audio c=IN IP4 192.168.0.1 a=sendrecv a=rtcp-mux a=ice-ufrag:g0kZ a=ice-pwd:d5oEody1jqIzDYUdf1fg6t 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=ssrc:1038736511 cname:janusaudio a=ssrc:1038736511 msid:janus janusa0 a=ssrc:1038736511 mslabel:janus a=ssrc:1038736511 label:janusa0 a=candidate:1 1 udp 2013266431 192.168.0.1 51232 typ host m=video 1 RTP/SAVPF 100 a=mid:video c=IN IP4 192.168.0.1 a=sendrecv a=rtcp-mux a=ice-ufrag:g0kZ a=ice-pwd:d5oEody1jqIzDYUdf1fg6t 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=rtcp-fb:100 goog-remb a=ssrc:2455978689 cname:janusvideo a=ssrc:2455978689 msid:janus janusv0 a=ssrc:2455978689 mslabel:janus a=ssrc:2455978689 label:janusv0 a=candidate:1 1 udp 2013266431 192.168.0.1 51232 typ host m=application 0 DTLS/SCTP 0 c=IN IP4 192.168.0.1

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3 回答 3

1

WebRTC 使用 DTLS-SRTP 强制加密（Chrome 仍然支持非标准和显式 MUST-NOT-IMPLMENT SDES 密钥）。

您不能只向 webrtc 提供 RTP 流；它必须是带有初始 DTLS 连接的 DTLS-SRTP 流。

人们已经将 node.js 连接到 webrtc 浏览器，所以我想你需要的所有机器都在那里。

于 2014-09-27T05:34:01.970 回答
0

我已经更新了包含双向流插件的fork，向您展示了一个有效的示例（我在 debian jessie 上进行了测试）。

这是我对您的插件更改的指示

  1. 在您从 chrome 请求关键帧之前，请确保您的 gstreamer 管道设置为接收
  2. 当 webrtc 媒体准备好时请求你的关键帧（详见janus_bidirectional_streaming_setup_media函数）
  3. 不要使用rtpbingstreamer 元素来处理传入的流。由于某种原因，它设置上限的方式并没有真正起作用，并且管道会崩溃。如果您确实获得了 rtp 数据包并且能够将它们发送到端口，那么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，直接将缓冲区推送到插件内的appsrc 应该也可以。

于 2014-09-29T16:53:54.543 回答
0

Kurento 媒体服务器 (KMS) 是一个完全编写在 GStreamer 之上的 WebRTC 媒体服务器。KMS 提供了一个 WebRtcEndpoint，实现了向/从 Web 浏览器发送/接收 WebRTC 流所需的所有协议和算法。KMS 公开基于媒体元素和媒体管道的 API，这些 API 转换为 GStreamer 媒体管道。一般来说，您在 GStreamer 上拥有的所有功能也可以在 KMS 中使用。您可以在http://www.kurento.org中查看 KMS 。

免责声明：我是 Kurento 开发团队的一员。

于 2014-09-27T15:29:27.417 回答