I'm familiar with Freeswitch, not Asterisk. So, I don't know my comment will be applicable there.

But, could you explain your signaling path a little. Is websocket being handled by Asterisk or somebody else in between. In my case, there is Kamailio in between FS and webRTC client. So, Freeswitch was modifying the SDP to non-webRTC, so called webRTC client rejected the call. I had to set FS to media proxy mode to stop it from modifying SDP.

Thanks,

Dipak

On Mon, Feb 10, 2014 at 8:00 AM, jaflong jaflong <jaflong@yandex.com> wrote:

I am having problems with calls from webrtc to kamailio forwarded to Asterisk

These are snippet of the debug logs

Asterisk

CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0

Jssip

Cause: Bad Media Description
Origin: remote

Searching on google I get some indication this is to do with ice config?
Please can some one suggest if this is so.

In my scenerio the webrt clients will only call to the asterisk server (and not to other user agent).
Considersing this I think maybe can do without ice.

Is it possbile to disable ice.

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Thanks,

Dipak