Bad audio in the beggining of the call

Asterisk Asterisk WebRTC
1

Hello,

Having a setup as follows:

  • Asterisk 18.20.2 with PJSIP
  • Debian 10
  • JsSIP

We are experiencing some bad audio with the calls, usually and very frequently in the beginning of the calls (first 10 seconds).

After some analysis we are suspecting is due to DTLS.
This enforced also by the fact that we are now receiving a warning (we were not receiving this warning with lower asterisk 18.X versions):

WARNING[24207][C-00001223] res_rtp_asterisk.c: 1704879756.12880: DTLS packet from 10.xxx.x.xxxx:55248 dropped. ICE not completed yet.

On JsSIP we have tried to not use a STUN server, since our setup is internal, but also with the google STUN and its the same.

Any help on this ?

2

I think you need to monitor your RTP traffic. Connection buildup and signalling has nothing to do with your audio problems.

3

It could be a jitter issue. Check this: Jitter Buffer Operation and Use in Asterisk ⋆ Asterisk
and the settings in the hardware (if any).

4

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