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session.createStream({constraints:{audio:true,video:false}}).play(); |
3. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect
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Solution: according to the standard, 'SIP Login' and 'SIP Authentification Authentication name' should not contain any of unescaped spaces and special symbols and should not be enclosed in angle brackets '<>'.
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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
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rtp_in_buffer=true |