[karoo-users] Upper Registration and Invite routing with local RTP

[Doug Reid]

Hi Joegen

Thanks and please let me know when this is available one thing Im struggling with is the "AOR" ability I don't see any reference to this in the cookbook and need to understand how
to interact with the UA list that appears in the AOR for example sending an Invite from a trunk to the UA that is in a upper registration mode and then creating the local RTP stream.
Have you got any documentation on this?

Thanks
Doug
From: Joegen E. Baclor [mailto:joegen at ossapp.com]
Sent: 30 August 2012 01:05 PM
To: Doug Reid
Cc: karoo-users at ossapp.com; Phillip Smith
Subject: Re: [karoo-users] Upper Registration and Invite routing with local RTP

Hi Doug,

I will update the cookbook to also include a failover recipe.  I will give you an update when it is available.  For the meantime, make sure that all the UA involved can handle 3xx responses.  Karoo uses standard redirects for failover.

Joegen

On 08/30/2012 04:23 PM, Doug Reid wrote:
Hi Joegen

I have a requirement to setup a specific situation where we would need similar functionalities of the old OSBC but with more capability. There is a large project that will generate large sales volumes if I can achieve this.

I need to use the B2BUA with Upper Registration Mode  that keeps an AOR of relayed users, these AOR entries must be able to be interacted with.

I need to connect a local UA on port  5050 through the SBC and the REGISTER must relay to an Upper Registration IP address.

I need to then connect a trunk to a either Asterisk, Freeswitch or a gateway such as Audiocdes/NET/Vegastream.

>From a routing perspective I need to have the following routing ability that can be altered when required:


UA calls UA:

*         A local call can be made from UA to UA in the following codecs G711a/u, G729, AMR, GSM

*         If the UA is not present in the AOR then the next route will apply i.e the INVITE is sent to the ITSP or IP PBX with codec support for G711a/u, G729, AMR, GSM

UA calls IP PBX

*         A local UA calls a predefined an extension dial plan i.e. 3XXX the INVITE is sent to the IP PBX to initiate a local RTP call

*         If the IP PBX rejects the INVITE there should be a ability for a alternate route to be sent for example to the ITSP or other routes

IP PBX calls UA

*         If the IP PBX calls a UA on the SIP trunk the INVITE should initially search for the UA in the AOR and try initiate a call.

*         If the UA does not exist then the INVITE must be redirected to the ITSP or other route.

As a last requirement if the Upper Registration route fails all above should still function except the route to the ITSP (like a SAS ability)

Please let me know if you can assist?



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Regards
Doug





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