Thanks ALexandru

   You have any route logic (script) for bellow operation

   1. On each registration user's proto is stored in redis database
2. When rtpengine is being called, Kamailio checks user's proto
    a) If user is WS and is incoming call, dispatch him to media relay with RTP/AVP flag
    b) If user is WS and is outgoing call (from media relay) send it to the endpoint with RTP/SAVPF flag
    c) If user is SIP and is incoming call, dispatch it to media-relay with RTP/AVP flag (some SIP clients also have SRTP turned on by default)
    d) If user is SIP and is outgoing call, send it to endpoint without any RTP flags (most sipphones ca recognize which traffic is incomig)

Because i am new for Kamailio ,And how i can store user's proto to redis 

On Sun, Jun 14, 2015 at 5:06 PM, Alexandru Covalschi <568691@gmail.com> wrote:
Hi, sorry, my previus answer wasn't clear enough - was writing it in a very sleepy mood :)

No, kamailio acts as a full proxy server for websocket and SIP. P2P is for caruzdias's configuration from github.
You can try following this http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket guide for editing your current configuration file to support WebRTC. But as I said, you can face some problems with NAT-traversal, so you may to create different routes for ws and simple SIP.
Also, if you use Asterisk - make sure your version doesn't have some problems with understanding SRTP handshake (RTP/SAVPF) - be sure that you have last stable version of your branch (my colleague spent 3 days to figure out that there was a bug in his version). However, even after update we couldn't perform a transparent proxy for SRTP, so I used rtpengine with such scheme:

1. On each registration user's proto is stored in redis database
2. When rtpengine is being called, Kamailio checks user's proto
    a) If user is WS and is incoming call, dispatch him to media relay with RTP/AVP flag
    b) If user is WS and is outgoing call (from media relay) send it to the endpoint with RTP/SAVPF flag
    c) If user is SIP and is incoming call, dispatch it to media-relay with RTP/AVP flag (some SIP clients also have SRTP turned on by default)
    d) If user is SIP and is outgoing call, send it to endpoint without any RTP flags (most sipphones ca recognize which traffic is incomig)

This configurations works well both with Asterisk and Freeswitch, but Freeswitch in my practice can provide more concurent calls for lesser cost.

2015-06-13 22:24 GMT+03:00 Murugan Pandian <manpower13.cse@gmail.com>:
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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--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/

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