sr-users May 2015

sr-users@lists.kamailio.org

80 participants
114 discussions

WebSocket data in TCP connection
by mayamatakeshi 04 May '15

04 May '15

Hello, about module websocket, is it possible to associate a value with the TCP connection and have this value readable when handling SIP requests on that connection?

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Regd: Message Routing
by suryakiiran vanam 04 May '15

04 May '15

Hi Team, I am using Jitsi client with Kamailio. I was able to IM between the clients. How can I control the Message routing in Kamailio script. Please help me with any link. Regards, Surya.

2 1

WebRTC video calls
by Ivan Vujisic 04 May '15

04 May '15

I made successful audio calls from browser to browser using Asterisk 13.1 and SIPML5 browser phone. Asterisk can't manage WebRTC video calls due to lack of codec negotiation module, but I also faced RTP ports NAT traversal issue. To my understanding Kamailio is capable to resolve this. Can anybody confirm that he made successful browser to browser video calls with Kamailio sip proxy / registrar in front of Asterisk PBX. Also, any link to good tutorial or doc pages will be appreciated. Best Regards, Ivan Vujisic

2 1

UAC Module
by Ali Jibran 02 May '15

02 May '15

Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too. I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk? Basically this is what I'm trying to workout FS---->kamailio---->trunk. Any help will be much appreciated. Thanks. AJ

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sr-users May 2015