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"<[email protected]>
To: "ALICOMPUTECH"<[email protected]>, "OpenSIPS users mailling
list"<[email protected]>
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
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Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and "know-how"
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