Hi Everyone, We're routing calls through Asterisk (dialing in via sip and then dialing out via SIP).
We've noticed a curious behavior in chan_sip that doesn't persist with chan_pjsip. When examining the packet capture, we're seeing the SSRC changing constantly on the call. At first it happens over a variable interval (15s 6s etc) but eventually it ends up changing exactly every 1000ms. Every time the SSRC changes, it causes a very minor but noticeable gap in audio. The fact that it's changing on this exact interval makes me think there is an explicit setting somewhere, or it's intentional behavior in the code. We do NOT see this behavior with chan_pjsip, all other things being equal. Does anyone know what might be driving this difference in behavior, and what we can do about it? We're on Ast 13 (same behavior on our current version, and when we upgrade to latest) Thank you!
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