[SR-Users] ice problems

[jaflong jaflong]

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.