[OpenSIPS-Users] Transfer issue

[Alex Balashov]

SEMS is another option for a B2BUA, though all its prebuilt B2BUA apps  
are not pure B2BUA but incorporate some sort of initial media-based  
announcement premise.  But you can use its Python or C++ API to create  
one.

Another option is YATE?

--
Sent from mobile device

On Oct 23, 2009, at 8:07 PM, Iñaki Baz Castillo <ibc at aliax.net> wrote:

> El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
>> I follow this, but -and please correct me where I'm  
>> misunderstanding- isn't
>> the point of using a Proxy versus a B2BUA is that it's more  
>> lightweight
>> and scalable?
>
> Of course, but if you want PBX features you need a B2BUA. Also,  
> trasferring a
> call from a SIP PSTN provider does require *always* a B2BUA (in fact  
> the user
> doesn't transfer the provider, but the B2BUA which will accept such  
> REFER).
>
>
>
>> It seems with this setup every call ends up routed through
>> Asterisk on the initial INVITE, simply so that when a REFER  
>> potentially
>> comes later in the dialog, it can handle it.
>
> If you need a proxy use a proxy. If you need a PBX you need a PBX/ 
> B2BUA. If
> you need a very scalable system with high load you can scale it with  
> a proxy
> in front of various PBX's (of course, most probably the PBXs won't  
> share
> dialog inforamtion between them...).
>
>
>> Our setup has been initially
>> engineered for thousands of concurrent calls, and we're hoping to  
>> avoid
>> having a dozen Asterisk machines :)
>
> What you are looking for is the dream all want: a scalable SIP B2BUA  
> (no media
> handling), so a cluster of these B2BUA's would be located behind a  
> proxy
> (which does load balancing and failover). And it would be greater if  
> the B2BUA
> share information (about current dialogs and so) in some way  
> (memcache? common
> database?...).
>
> You could implement it with SipServlets (see Sailin SIP application  
> server or
> others), or FreeSwitch which allows calls without handling the  
> media...
> Of course, Asterisk is not the most suitable solution: it involves  
> media
> handling ("canreinvite" is a hack), it has a very poor SIP stack...  
> and
> basically it's designed to be a single PBX box.
>
>
>> Our goal is to design something like:
>>
>> UA <--> Opensips <--> Our PSTN gateways
>>
>> That could dynamically turn into, on the fly:
>>
>> UA <--> Opensips <--> B2BUA <--> Our PSTN Gateway
>>
>> We haven't found a clean way of doing this, unfortunately...  It  
>> may just
>> end up being a hack on Opensips 1.6 if that might work.
>
> You shouldn't expect that OpenSIPS could behave as a PBX because  
> it's a proxy.
> And what you need is a PBX, you MUST assume it. A proxy is very  
> scalable, fast
> and so, but it will NEVER provide PBX features (as transferring a  
> call coming
> from a PSTN provider to other user).
> IMHO OpenSIPS b2bua module will be useful for some limited tasks  
> (topology
> hidding and others) but not for replacing a PBX.
>
>
>
> Regards.
>
>
>
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
>
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