Hi,
I have enabled websocket support in kamailio and registered with test123 one SIP end point and test123 Websocket client.
Now i am trying to call from another test456 sip end point to test123 then call some times comes to both end points and some times does not come one end point.
I appreciate any help in this issue.
Thanks in advance.
Thanks Ram
Can you make a successful call between test456 and each of the endpoints when only one of them is registered?
Given that few, if any, ordinary SIP clients support the RTP/SAVPF profile mandated by WebRTC I would expect at least some of these calls to fail because of that. If the client stacks are any good these failures will be 488s.
Regards,
Peter
On 21 Dec 2012, at 16:54, Ramaseshi reddy kolli ramaseshireddy@gmail.com wrote:
Hi,
I have enabled websocket support in kamailio and registered with test123 one SIP end point and test123 Websocket client.
Now i am trying to call from another test456 sip end point to test123 then call some times comes to both end points and some times does not come one end point.
I appreciate any help in this issue.
Thanks in advance.
Thanks Ram
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Hi,
OK. Well I do know that parallel forking works with WebSocket. So I suspect you have a problem with your kamailio.cfg.
Does parallel forking work with that configuration when none of the clients connect over WebSocket?
It will also help if you can provide information on your Kamailio configuration and any log messages that Kamailio is generating.
Regards,
Peter
When only one of them is registered i can make calls.
On Sat, Dec 22, 2012 at 3:43 AM, Peter Dunkley < peter.dunkley@crocodile-rcs.com> wrote:
one
-- Cheers
Ramaseshi
Mobile : 9949875064 _______________________________________________ sr-dev mailing list sr-dev@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
Hi,
Parallel forking works when none of clients connect over WenSocket.
I think i should explain my scenario in a better way.
when one web socket and one sip end point registered. when i make call to kamailio using ICE and webrtc RTP profile RTP/SAVPF in SDP (using some ICE attributes), then call is forked to two end points (webrtc and sip end point) and call successfully goes to webrtc client and SIP end point.
Webrtc client rings and can answer the call as well. But SIP end point returns 488 not acceptable here because of RTP profile RTP/SAVPF so it fails.
Now the question is there any way i can modify SDP RTP profile when call is forked to normal sip end point.
Thanks in advance.
On Sat, Dec 22, 2012 at 6:40 PM, Peter Dunkley < peter.dunkley@crocodile-rcs.com> wrote:
configuration