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 <568...@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....@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 <abalas...@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
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>>
>>
>
>
> --
> 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
>
>
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