(Continued from Building a Startup)
Our way wasn’t working. To put it mildly our “business grade” solution didn’t perform much better than Vonage. We became to exemplify VoIP - jittery calls, dropped calls, one way calls, etc, etc, etc. Most of this was because of the lack of quality ITSPs at that time. Either way our customers didn’t care. It was us. If we went to market with what we had the first time around we were going to loose.
The problem was the other predominant architecture at the time was “hosted”. Someone hosts a PBX for you and ships you some phones. You plug them in behind your router and magically you have a phone system. They weren’t doing much better. Sure, their sales looked good but even then it was becoming obvious customer churn was quite high. People didn’t like hosted either, and for good reason. Typically they have less control over the call than we do.
As I’ve eluded before I thought there was a better way. We needed to host the voice applications where it made the most “sense”. We were primarily using Asterisk and with a little creative provisioning, a kick-ass SIP proxy, and enough Asterisk machines we could build the perfect business PBX - even if that meant virtually none of it existed at the customer premise. Or maybe all of it did. That flexibility was key. After a lot of discussions, whiteboard sessions, and late nights everyone agreed. We needed a do-over.
So we got to work and slowly our new architecture began to take shape. We added a kick-ass SIP proxy (OpenSER). OpenSER would power the core routing between various Asterisk servers each meeting different needs - IVR/Auto Attendant, Conferencing, Voicemail, remote phones (for “hosted” phones/softphones), etc. The beauty was the SIP proxy could route between all of these different systems including the original AstLinux system at the customer premise. Customer needs to call voicemail? No problem - the AstLinux system at the CPE fires an INVITE off to the proxy and the proxy figures out where their voicemail server is. The call is connected and the media goes directly between the two endpoints. Same thing for calls between any two points on the network - AstLinux CPE to AstLinux CPE, PSTN to voicemail, IVR to conference.
This is a good time to take a break and acknowledge what really made this all possible - OpenSER. While it’s difficult to explain the exact history and family tree with any piece of SER software I can tell you one thing - this company would not be possible without it. There is no question in my mind. It’s now 2011 and whether you select Kamailio or OpenSIPS for your SIP project you will not be sorry. Even after five years you will not find a more capable, flexible, scalable piece of SIP server software. It was one of the best decisions we ever made.
Need to add another server to meet demand for IVR? No problem, bring another server online, add the IP to a table and presto - you’re now taking calls on your new IVR. Eventually a new IVR lead to several new IVRs, voicemail servers, conference systems, web portals, mail servers, various monitoring systems, etc.
What about infrastructure? Starting at our small scale, regional footprint, and focus on quality we began to buy our own PRIs and running them on a couple of Cisco AS5350XM gateways. This got us past our initial issues with questionable ITSPs. Bandwidth was becoming another problem... We had an excellent colocation provider that provided blended bandwidth but still we needed more control. Here came BGP, ARIN, AS numbers, a pair of Cisco 7206VXRs w/ G2s, iBGP, multiple upstream providers, etc.
At times I would wonder - whatever happened to spending my time worrying about cross compilers? Looking back I’m not sure which was worse - GNU autoconf cross-compiling hell or SIP interop, BGP, etc. It’s fairly safe to say I’m a sadomasochist either way.
Even with all of the pain, missteps, and work we finally had an architecture to take to market. It would be the architecture that would serve us well for several years. Of course there was more work to be done...
The problem was the other predominant architecture at the time was “hosted”. Someone hosts a PBX for you and ships you some phones. You plug them in behind your router and magically you have a phone system. They weren’t doing much better. Sure, their sales looked good but even then it was becoming obvious customer churn was quite high. People didn’t like hosted either, and for good reason. Typically they have less control over the call than we do.
As I’ve eluded before I thought there was a better way. We needed to host the voice applications where it made the most “sense”. We were primarily using Asterisk and with a little creative provisioning, a kick-ass SIP proxy, and enough Asterisk machines we could build the perfect business PBX - even if that meant virtually none of it existed at the customer premise. Or maybe all of it did. That flexibility was key. After a lot of discussions, whiteboard sessions, and late nights everyone agreed. We needed a do-over.
So we got to work and slowly our new architecture began to take shape. We added a kick-ass SIP proxy (OpenSER). OpenSER would power the core routing between various Asterisk servers each meeting different needs - IVR/Auto Attendant, Conferencing, Voicemail, remote phones (for “hosted” phones/softphones), etc. The beauty was the SIP proxy could route between all of these different systems including the original AstLinux system at the customer premise. Customer needs to call voicemail? No problem - the AstLinux system at the CPE fires an INVITE off to the proxy and the proxy figures out where their voicemail server is. The call is connected and the media goes directly between the two endpoints. Same thing for calls between any two points on the network - AstLinux CPE to AstLinux CPE, PSTN to voicemail, IVR to conference.
This is a good time to take a break and acknowledge what really made this all possible - OpenSER. While it’s difficult to explain the exact history and family tree with any piece of SER software I can tell you one thing - this company would not be possible without it. There is no question in my mind. It’s now 2011 and whether you select Kamailio or OpenSIPS for your SIP project you will not be sorry. Even after five years you will not find a more capable, flexible, scalable piece of SIP server software. It was one of the best decisions we ever made.
Need to add another server to meet demand for IVR? No problem, bring another server online, add the IP to a table and presto - you’re now taking calls on your new IVR. Eventually a new IVR lead to several new IVRs, voicemail servers, conference systems, web portals, mail servers, various monitoring systems, etc.
What about infrastructure? Starting at our small scale, regional footprint, and focus on quality we began to buy our own PRIs and running them on a couple of Cisco AS5350XM gateways. This got us past our initial issues with questionable ITSPs. Bandwidth was becoming another problem... We had an excellent colocation provider that provided blended bandwidth but still we needed more control. Here came BGP, ARIN, AS numbers, a pair of Cisco 7206VXRs w/ G2s, iBGP, multiple upstream providers, etc.
At times I would wonder - whatever happened to spending my time worrying about cross compilers? Looking back I’m not sure which was worse - GNU autoconf cross-compiling hell or SIP interop, BGP, etc. It’s fairly safe to say I’m a sadomasochist either way.
Even with all of the pain, missteps, and work we finally had an architecture to take to market. It would be the architecture that would serve us well for several years. Of course there was more work to be done...