VoIP Mailing List Archives

[simon.hohberg at mcs-d...]

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

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