On Fri, May 8, 2020 at 8:58 AM Erol Toker <e...@truly.co> wrote: > 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) >
There are specific actions which can cause a notification of a source change to bubble up. Inbound SSRC change being one, and I think a few others (been years since I looked at that stuff). The chan_pjsip module may not implement handling of those, so the SSRC doesn't change. An RTP capture on both sides would show anything out of the ordinary, and bumping up the debug level (core set debug 3 may do it) may show why. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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