As mentioned you really have to look at the over all RTP and the negotiation that is done on the WebRTC side. It may be taking time for ICE or DTLS to finish, which is needed in order to have media flow.
As far as my understanding ICE gatherings/negotiations happen before the call is connected, correct me if I am wrong. But I am seeing choping in audio after answering the call.
Also, We have tested with other trunk provider (Twilio) as suggested by ambiorixg12. We didn’t see any choping there.
There are two ways to understand it. 1) SIP.conf asterisk IP sould not be 0.0.0.0 , make it fixed IP. 2) check RX and RX at system level for check message queue of server.
I have checked RTP packets with timings. I am seeing RTP packet flow immediately when the dialed number answers the call, with seconds level accuracy.
This bug is not consistent, we are observing in few calls. Can this be because the network is poor on one of the entities (Dialed side/ Asterisk / SIP truck providers / voip or PSTN). Since media will flow on UDP protocols, I am assuming UDP packets might dropped due to poor network at one of the entities. Or Is this a Asterisk system level issue, Has anyone faced this issue.