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linuxwebrtcasterisksipjssip
JustLogin
JustLogin

Reputation: 1890

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?

Upvotes: 1

Views: 2652

Answers (2)

Chetan
Chetan

Reputation: 1468

I just encountered the same issue, for me it was codecs issue. I was allowing only G729 in sip.conf file and hence it was throwing 488 error. For now, I fixed it by setting allow=all (I would check later on which particular codec does it needs).

Upvotes: 1

JustLogin
JustLogin

Reputation: 1890

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.

Upvotes: 2

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