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  • parameters of your SIP account the call will be made from;
  • the stream name to republish the call to (the toStream parameter), for example, call_stream1$stream1;
  • the name of your second SIP account the call will be made to.

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rtp_session_init_always=true

2. Freezes may occur, audio may be out of sync with video when republishing a SIP call stream as RTMP

Symptoms: freezes and audio/video out of sync are observed while playing an RTMP stream republished by /push/startup REST query from a SIP call

Solution:

a) in WCS builds before 5.2.1541 add the delay to audio/video generator start

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generate_av_start_delay=1000

b) update WCS to build 5.2.1541 where the issue was fixed

3. RTP traffic buffering should be enabled in some cases when republishing SIP as Stream or SIP as RTMP

Symptoms: audio and video may be out of sync when playing a SIP call stream

Solution: update WCS  to build 5.2.1910 and enable RTP traffic buffering

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rtp_in_buffer=true