I have an issue with the asterisk jitterbuffers and would like an insight into them. I have configured buffers on my asterisk 1.6.2.9 system as follows SIP -> Local -> [Jitter Buffer] SIP. The problem that I am running across is that the jitter buffer doesn’t actually kick in until an RTP packet arrives that has the marker bit set. Does anyone know how to make the system initialize and use the buffer on stream creation, instead of some random time into the call(I have seen anywhere from 12 to 37 seconds before the jitter buffer kicks in and starts working).
Thanks