Audio and video webrtc

Max

Administrator
Staff member
Hello.
You can set bitrate using constraints as shown on this page when publishing any WebRTC stream, for example:
Code:
    session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false,
        rtmpUrl: rtmpUrl,
        constraints: {audio: {bitrate: 96}, video: {minBitrate: 3000, maxBitrate: 7000}}
 

Max

Administrator
Staff member
We tested this code on latest public build, and it works. So collect logs and dumps as described here, including Chrome browser console output, and send to support@flashphoner.com, we will check.
 

Max

Administrator
Staff member
Please let know WCS server and Web SDK versions you are using.
Try modifying the demo Two-way Streaming (WCS_HOME/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html) to use the above code.
 
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