[VoiceOps] SIP routing engine - growing pains

[randal k]

>
> 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.
>>
>
We started out our voip platform on OpenSIPS + Cisco, and have been pretty
pleased with the results. The only thing that is a huge pain is mixing our
PSTN upstreams - we have PRIs from the LEC and we have SIP trunks from a LD
provider, which, because of their own individual peculiarities, require lots
of SIP handholding to make fax, MoIP, etc work reliably. We are very
thankful that our prem gear is overwhelmingly Cisco, because consistency
makes life better.

Our biggest hurdles have been tremendous amounts of development time on
OpenSIPS to make it do what we want it to do, and interoperability. Interop
- both with our vendors and with random customer PBXs (don't get me start on
mismatched NSE/NTE) - has been a major time sink.

That said, the bang for the buck is out of this world. Because it's Linux
everywhere, load balancing and fail-over is a no-brainer. Need 1000 more
calls *right now*? Image whatever server is laying around, make 2-3 LB/DNS
changes, and pow, done. Scaling couldn't be easier, imo.

That said, we are getting our feet with with OpenSIPS 1.6 B2BUA-style
service -- it doesn't proxy RTP streams (can if you want to use MediaProxy
addon), but does let it hold up multiple sip dialogs for one call, and also
does topo hiding. This turns osips into basically a SBC, without RTP. As
long as our RTP interop stays clean, it should be a killer solution.

HTH,
Randal
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