I guess ice is implemented at web browser level, not sure there is an
option to control it from java script.
On the other hand, you should run asterisk in debug mode and see what it
is printing. It doesn't seem an issue related to kamailio at all.
Cheers,
Daniel
On 10/02/14 14:00, jaflong jaflong 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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