[OpenSIPS-Users] use case for settlement free peering

A G 28rhills at gmail.com
Mon Oct 19 20:16:36 CEST 2009


Greetings:

I'm looking for advice on a project/proof of concept I'm working on.
I would like to create a settlement-free peering fabric for voice
traffic between and among some peer institutions in my area.  Because
this is more of a side-project for cost cutting measure, I'm primarily
looking at open source software, though commercial product
recommendations would be helpful as well.

The organizations I would like to connect have their own PBXs with
large blocks of numbers (whole NPA-NXXs), with no number portability
in or out.  I imagine both at the individual PBXs and peering fabric,
the number routing would be static.  To put another way, we would
manually configure which connections the block of telephone numbers is
reachable at.

Here is the required ASCII art diagram   :)


                    +-------+
                    |  PBX  |
                    +-------+
                        |
                    +-------+
                    |  SBC  |
                    +-------+
                        |
                        |
+---+  +---+         .--------.            +---+  +---+
| P |  | S |        /          \           | S |  | P |
| B |--| B |------ (     ????   )----------| B |--| B |
| X |  | C |        \          /           | C |  | X |
+---+  +---+        `---------'            +---+  +---+
                        |
                        |
                    +-------+
                    |  SBC  |
                    +-------+
                        |
                    +-------+
                    |  PBX  |
                    +-------+




For scalability reasons, a full mesh of connections between and among
the SBCs is not an attractive option.

Here's what I think I need:
Basic SIP routing
TCP, TLS, and UDP support

What would be nice to have:
IPv6
CDR

What is probably not needed:
User agent client registration, presence, IM, voice mail

I see there are several different open source voice projects.
Do you think this is an appropriate use for OpenSIPS?
I'm seeking comments on what you would use for this situation.
Are there any existing  projects along these lines?
Is there one project that is better than another for this application?

Thank you



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