linuxwebrtcasterisksipjssip

Asterisk instantly terminates WebRTC (JSSIP) call


I'm running Asterisk 11.2.2 with SRTP and STUN support under Calculate Linux (Gentoo-based distribution).

When I try to call from one WebRTC instance to another, using JSSIP, the call passes, but if i answer it on another instance, the call suddenly terminates. Using Asterisk debug mode, i can catch 488 error (Not acceptable here).

If I use one SIP phone (Ekiga) instance instead of WebRTC, then I can call JSSIP from it, and everything works fine. Nevertheless, I can't call Ekiga from JSSIP, and this makes me confused.

Can you advise me, what have I do to localize this bug?


Solution

  • The problem was in my Asterisk: it had some WebRTC issues in 11.2.2 version. Upgrading to 11.4.0 makes everything works fine.