We are encountering problems with poor audio quality in the early stages of calls within our Asterisk certified-18.9-cert5 setup on a Debian 10 machine. We utilize PJSIP, and our SIP client JsSIP 3.10.0 connects to the Mini HTTP web server of Asterisk. In terms of codecs we have tried opus, alaw, g722.
All of this backup up on a 500Mbps network.
These audio issues occur specifically during the first moments (5-7 first seconds) of the call, after that everything is ok.
The issue is frequent but really random on agents and usually the problem is on the agent side (JsSIP) on an incoming call.
Any idea what may be wrong ?