I would agree with Scott strongly. Scott Berkman wrote:
OpenSIPs/Kamillo/SER or whatever it's called today can do most of this out of the box. I think it's the best option to match your needs if you have the time and engineering to figure it out.
From there you can go to the Acme/Covergence SBC platforms, the lower end Covergence will run in a VM or on your own hardware, but can handle most of what you are looking for.
Only thing on the list this doesn't include is the T.38 fax stuff, but since all of these would require separate media gateways anyway, that's the only place you need the T.38 supported, SER or the SBC would just pass the SDP along.
-Scott
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jay Hennigan Sent: Friday, October 30, 2009 4:21 PM To: VoiceOps Subject: [VoiceOps] SIP routing engine - growing pains
Trying to pick the collective brain here!
Scenario:
We are a growing network service provider with ISP roots. We got into VoIP by selling outsourced hosted PBX services. Merged with a company doing early Asterisk deployment of hosted VoIP.
Now doing trunk replacement with Adtran TA9xx at customer premise.
PSTN connections are PRI trunks to a wholesale provider routed through ah Atlas 550 to feed our Asterisk and Cisco 5350 SIP gateways, SIP trunking and some offnet via outsources hosted PBX provider. Recently added local ENUM database server feeding the 5350s as dial-peers got unwieldy.
Our ultimate goal is to migrate from the outsourced hosted provider to our own Broadsoft platform or equivalent but we're not yet at critical mass to make that financially viable. We're presently peaking at about 150 to 200 simultaneous calls busy hour through our own fabric plus about that on the hosted PBX provider offnet.
What I'm looking for:
SIP routing engine and/or softswitch but don't need feature server. Needs to do at a minimum:
* SIP routing based on destination number or pattern. * Registrar for remote Adtran TA9xx at customer sites. * Interface via SIP with 5350s for calls using PRI to PSTN. * Fax compatibility T.38/G.711 passthrough. * Support multiple SIP carriers inbound and outbound. * CDR generation. * Robust enough for solid performance - failover pair or load-balanced * Ability to grow to handle ~1000 simultaneous calls
Nice to have:
* LCRE based on time of day or destination pattern. * Geographic redundancy, located in multiple POPs. * NAT traversal B2BUA type functionality * IPv6 or at least a path to IPv6 functionality * Transcoding
Suggestions and/or recommendations?
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