We have an existing VoIP platform which has various features and which has evolved over a few years. It has registered SIP phones, relaying to configured SIP endpoints, multiple "also ring" targets with delays and all sorts. Works well. I coded it all on linux from scratch.
We are working on moving to a new platform, whilst retaining the features of the existing system. This is going to be fun. I blogged on code wearing out before - well this code is a tad dusty to say the least.
The new system uses FireBrick VoIP platform, which, from a VoIP point of view, is completely new code. It is a lot better than my previous linux code. It works well and we have been using it for what?, a year, in the office.
The newest code works by using RADIUS to make call routing decisions. This is really good from a redundancy and scalability basis as it allows the call servers(s) to talk to multiple RADIUS servers to find the answer. If there is a problem with one then within hundreds milliseconds they are trying another. There is even an RFC covering the SIP based authentication logic (http auth).
The FireBrick code is pretty much ready, but I expect changes when I am using it "in anger" on the new system. The plan is to mimic the current operation of "speechless" by providing suitable RADIUS replies to the new "voiceless" system. As I type I am already pondering a few minor tweaks.
Once I have this code sorted over the next few days, we can get traillists on board. Even so, we are missing a key feature which is call recording, so that is the next job on the FireBrick side. I am very keen to keep something of a unique feature in that we make stereo a-law call recordings.
There are a lot of advantages to moving to a new system. The separation of control and authentication logic from the actual call routing will help reliability and scalability. The new system should easily allow multiple servers at the VoIP level, RADIUS level, and even for call recording and voicemail. The newer FireBrick SIP code is nicer and we are hoping we can even make it work with some styles of NAT (yuck).
The challenge is the current "speechless" code is all integrates VoIP/SIP and routing decisions. I have to understand my code, and make new code for my RADiUS servers. It is not too bad, but the code is a tad old and grey, so will be a fun challenge making it new and shiny.
Nose back to the grindstone over the weekend. That is, assuming the whole idea of a VoIP service is not screwed up by OFCOM.
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