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session.createStream({ name: streamName, display: remoteVideo }).on(STREAM_STATUS.PENDING, function (stream) { ... }).on(STREAM_STATUS.PLAYING, function (stream) { setStatus("#playStatus", stream.status()); onPlaying(stream); if (Browser.isSafariWebRTC() && Browser.isiOS() && Flashphoner.getMediaProviders()[0] === "WebRTC") { setTimeout(function () { muteVideo(); unmuteVideo(); }, 1500); } ... }).play(); |
Stereo audio playback in browser
When a stream captured from RTMP, RTSP or VOD source is plaing in browser, audio is usually transcoded to Opus codec. By default, Opus encoder is configured to play a speech and monophonic audio. Encoder bitrate should be raised to 60 kbps or higher to play stereo in
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browser
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opus.encoder.bitrate=60000 |
Chrome-base browsers
By default, Chrome browser plays WebRTC stream with stereo sound in Opus codec as mono due to engine bug. Since Web SDK build 0.5.28.2753.151 this can be worked around using the following playback constraint
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