<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div class="h5"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
OpenSIPs/Kamillo/SER or whatever it's called today can do most of this out<br>
of the box. I think it's the best option to match your needs if you have<br>
the time and engineering to figure it out.<br></blockquote></div></div></blockquote><div><br>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.<br>
<br>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.<br>
<br>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.<br>
<br>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.<br>
<br>HTH,<br>Randal<br></div></div>