Hi,

I am using Asterisk 13.7.0 with PJSIP.

I set up Asterisk for use with WebRTC SIP clients. After I managed to get video working, I noticed, that the calling party receives no video till 90s (or so) have passed. After 90s both parties receive video perfectly.

I am suspecting that this is due to the time needed for the DTLS
handshake between Asterisk and the caller. Since Asterisk first establishes a full connection to the callee, the callee already begins sending data, while Asterisk is still performing the DTLS handshake with the caller. As a consequence the caller misses the first RTCP Full Intraframe Request (FIR) and the received video stream cannot be rendered till the next FIR 90s later arrives.

Am I right or is this nonsense?
Is this a known issue? I couldn’t find anything about this. Is there a fix available?

Thanks in advance!

Simon