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 ja
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 w
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 Descr
