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.
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.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Biswas, Dipak -
Daniel-Constantin Mierla -
jaflong jaflong