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Supported codecs

WebRTC video:

• H264
• VP8
• VP9 (since WCS build 5.2.1908 and SFU SDK build 2.0.268)

WebRTC audio:

• Opus

Implementation basics

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codecs_exclude_sfu=alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,flv,mpv,vp8,h265
webrtc_cc_min_bitrate=1000000
profiles=42e01f,640028

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1. Open SFU client example in browser, for example https://demo.flashphoner.com:8888/client2/sfu/examples/client/main.html, enter server URL, room name, pin code and user name, then click Enter

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If participant publishes more than one stream from camera, only the first published stream will be available at WCS side.

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TURN support

1. A stream captured from a screen window simulcast publishing will crash Chrome browser tab on minimizing this window

A standard RTCPeerConnection object is used in browser to publish and play audio and video tracks, so this object should be configured properly to relay a media traffic via TURN server. For example, all the streams are published directly to  WCS instance in SFU Two Way Streaming example:

code

pc = new RTCPeerConnection();
...

The code should be changed as follows to use a TURN server, for example, internal WCS TURN server:

let connectionConfig = {
    iceServers: [
        {
            urls: 'turn:wcs:3478?transport=tcp',
            credential: 'coM77EMrV7Cwhyan',
            username: 'flashphoner'
        }
    ],
    iceTransportPolicy: "relay"
};
pc = new RTCPeerConnection(connectionConfig);
...

Where:

• wcs - WCS server address;
• flashphoner - WCS internal TURN server default username;
• coM77EMrV7Cwhyan - WCS internal TURN server default password

In this case all the media traffic will pass through the WCS internal TURN server. This feature may be also used to wrap WebRTC traffic to TCP if the client has a bad channel, because WCS does not support TCP transport for SFU streams.

VP9 support

Since WCS build 5.2.1908 and SFU SDK build 2.0.268, VP9 codec is supported for SFU mediatraffic publishing and playback

codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,vp9,flv,mpv

VP9 must be excluded for the cases when it is not supported by other party (SIP calls for instance)

codecs_exclude_sip=mpeg4-generic,flv,mpv,h265,vp9
codecs_exclude_sip_rtmp=opus,g729,g722,mpeg4-generic,vp8,vp9,mpv,h265

It is necessary to exclude H264 and VP8 codecs for SFU to enable VP9

codecs_exclude_sfu=alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h265,flv,mpv,vp8,h264,h265

VP9 codec allows to use SVC (Scalable Video Coding) at browser side by setting scalabilityMode in encoding profile

{
  "room": {
    ...
  },
  "media": {
    "audio": {
      "tracks": [
      ]
    },
    "video": {
      "tracks": [{
        "source": "camera",
        "width": 1280,
        "height": 720,
        "codec": "vp9",
        "encodings": [
          {
            "rid": "nonsense",
            "active": true,
            "scalabilityMode": "L1T3"
          }
        ]
      }]
    }
  }
}

In this example, a video track will be published in one resolution (L1) with 3 FPS layers (T3). A full scalability modes list is available here.

Known limits

1. It is not recommended to use Simulcast (using a number of encoding profiles) and SVC simultaneously because this may not be properly supported in a browser.

2. Only one spatial layer (L1) is available for screen sharing, other settings are ignored. The limit is hardcoded at WebRTC library level and applied in any browser.

Known issues

1. A stream captured from a screen window simulcast publishing will crash Chrome browser tab on minimizing this window

Symptoms: when stream is capturing from active screen window, Chrome tab crashes if this window is minimized by user

Solution: there is the Chromium bug, a stream capturing from a screen window should be publihed in only one quality (no simulcast) until this bug is fixed (in Chrome build 98.0.4736.0)