[SR-Users] ice problems

[Biswas, Dipak]

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 at 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
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