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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();