[VoiceOps] SIP routing engine - growing pains
scott at sberkman.net
Fri Oct 30 17:15:42 EDT 2009
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
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
Subject: [VoiceOps] SIP routing engine - growing pains
Trying to pick the collective brain here!
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
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
Suggestions and/or recommendations?
Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net
Impulse Internet Service - http://www.impulse.net/
Your local telephone and internet company - 805 884-6323 - WB6RDV
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