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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
