21 Dec
2012
21 Dec
'12
6:54 p.m.
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
22 Dec
22 Dec
12:13 a.m.
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(a)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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3:10 p.m.
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(a)crocodile-rcs.com> wrote:
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
one
--
Cheers
Ramaseshi
Mobile : 9949875064
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28 Dec
28 Dec
2:38 p.m.
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(a)crocodile-rcs.com> wrote:
configuration
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participants (2)
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Peter Dunkley -
Ramaseshi reddy kolli