[OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

[ALICOMPUTECH]

Hi
  Bogdan
        thanks for the prompt and quick reply
                                             i will be using Multi Criteria Decision Theory (MCDT) to take the handoff decision between base stations during a call

the possible scenario might be

e.g. if the Signal strength is not good enough in an OpenBTS cell and there is jitter above a predefined threshold value and and some other parameters involved (measured via dedicated OpenBTS python scripts) are crossing the threshold values then i will use (MCDT) to take the handoff decision. Remember that the endpoints are emulated as SIP User Agents(clients) using SIP extensions

sorry in advance if i once again did not describe my problem properly

Best Regards 

Bye

----- Original Message -----
From: "Bogdan-Andrei Iancu" <bogdan at opensips.org>
To: "ALICOMPUTECH" <alicomputech at yahoo.com>, "OpenSIPS users mailling list" <users at lists.opensips.org>
Sent: Tuesday, March 29, 2011 2:25:50 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna
Subject: Re: [OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

Hi,

First of all OpenSIPS is a sip server so it works only with SIP.

Secondly, by default opensips is SIP proxy, so it cannot do handover. 
But using the Back2Back User agent module, you may be able to play with 
the ongoing calls and move them between different termination points.

I can help you more if you could describe the handover scenario you need.

Regards,
Bogdan

ALICOMPUTECH wrote:
> Hello
>       Everyone
>                I want to replace the Asterisk (being used as a SIP Server for registration, authentication and call routing) with OpenSIPS in OpenBTS project, as i am planning to have an Asterisk cluster for dedicated services and OpenSIPS will be forwarding the SIP calls to the cluster.
>
> OpenBTS implements GSM Um air interface and emulate the Mobile handsets as the SIP endpoint and these handsets can be used as SIP extensions in a SIP-capable server.
>
> I need to know the handoff and/or handover support in OpenSIPS as i am a newbie to this wonderful open source solution.
>
> If there is any pointer and/or previously handoff/handover work done please share, it will then ease my work
>
> thanks in advance
>
> Best Regards
>
> Bye
>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and "know-how"