Hi, in Asterisk how does the flow works with webRTC in a video call? I mean. the audio and video are sent in the same RTP packets or the audio is sent in a different packet than the video?
Is this an Asterisk question or an webRTC one?
They are separate packets, and this is really an RTP question.
Thank you for your response.
But I was wondering how can we guarantee the sync between both audio and video if they are sent through different packets?
RTP includes timestamps, and RTCP provides information to map timestamps to actual NTP values.
Ok, thank you for you answers and your time!
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