Random bad audio quality first seconds of a call


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 ?

You should contact Sangoma for support with your certified versions. If you want support from the opensource community you should use 18.20.0 (or 20 or 21). Certified versions are only intended for people with support contracts, and are only certified for certain uses, e.g., if you had used chan_sip, that usage would not be certified. Iā€™m not sure if use on Debian is certified.

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