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

[Bogdan-Andrei Iancu]

Hi,

So more or less, your problem is reduced to moving a leg of an existing 
call between several UA - like call is established between user A and B 
and during the call B user is moving (transparently) between several SIP UA.

If so, I think you can easily achieved by using the b2bua module - you 
can trigger the moving of a leg to a different location via an MI command.

Regards,
Bogdan

On 03/29/2011 04:37 PM, ALICOMPUTECH wrote:
> 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"