HI Alexandru,
i try to connect like this
!--Freeswitch(IVR,Callcenter,dialplan,sip auth)
Browser(chrome,firefox,opera)--(WS)--->Kamailio--->!
!--Freeswitch(IVR,Callcenter,dialplan,sip auth)
i understand Kamailio only handling signalling(using websocket) but stream goes to peer-to-peer ,But i need to play ivr and handle callcenter (freeswitch)
so here i try to kamailiio act proxy server
Any idea how i can achieve thid
On Sun, Jun 14, 2015 at 12:24 AM, Alexandru Covalschi 568691@gmail.com wrote:
Well, I performed that by creating a media relay consisting of 2 freeswitches and using rtpengine.
You just need to handle WebRTC by kamailio using kamailio websocket module: http://kamailio.org/docs/modules/4.3.x/modules/websocket.html caruzdias-es configuration helped me a lot in understanding how websockets work on Kamailio: https://github.com/caruizdiaz/kamailio-ws But be aware, this configuration is for peer2peer connections, not for dispatching!
Kamailio will send simple SIP packets to the media relay then.
Also I used different NAT-traversal mechanism for sip and ws traffic (different routes based on client's transport protocol). Also you'll maybe need to have different rtpengine flags for sip and ws - remember that WebRTC MUST have SRTP, but I had some issues in transfering the SRTP handshake in sipphone<-->kamailio<-->freeswitch scheme, so on webrtc connection my "incoming" rtpengine had RTP/AVP flag, and on outgoing webrtc it MUST have RTP/SAVP For usual SIP calls I also conveted everything to RTP/AVP.
So you'll need to know to which type of user - ws or tcp/udp you're calling to understand which type of RTP to send them.
2015-06-13 19:07 GMT+03:00 Murugan Pandian manpower13.cse@gmail.com:
it's posible dispatching websocket request?
I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can achieve more concurrent call(more then 1000 call)
On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov <abalashov@evaristesys.com
wrote:
That question is difficult to answer without some elaboration on your part as to what you want to achieve.
-- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Sent from my BlackBerry. *From: *Murugan Pandian *Sent: *Saturday, June 13, 2015 09:47 *To: *sr-users@lists.sip-router.org *Reply To: *Kamailio (SER) - Users Mailing List *Subject: *[SR-Users] SIP-over-Websocket Load Balancing
HI,
how to handle sip-over-websocket load balancing (WebRTC)
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
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
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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