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As a result, the authorization header is set.
Known issues
1) . When republishing SIP as RTMP to Wowza servers and when receiving a stream from Wowza via HLS, a spectator can see freezes, short time non-synchronous playback.
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disable_streaming_proxy=true |
2. When SIP call is redirecting to stream with SIP as Stream function, the audio only call stream does not play via WebRTC in a browser.
Solution: the audio only call stream should be played as audio only stream in a browser by explicitly constraints setting in player script when stream is created, for example
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session.createStream({constraints:{audio:true,video:false}).play(); |