[SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

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 at 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 at 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 at 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)
>>
>>
>> _______________________________________________
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>>
>>
>
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>


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