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
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)


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--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/

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