Versions Compared

Key

  • This line was added.
  • This line was removed.
  • Formatting was changed.

...

Code Block
languagejs
themeRDark
sipLogin='Ralf_C12441'
sipAuthenticationName='Ralf_C'
sipPassword='demo'
sipVisibleName='Ralf C'

4. 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

Code Block
themeRDark
rtp_in_buffer=true