Amazing that they use Yate (which is unfortunately not that widely known).
Years ago I've examined different VoIP engines for a project:
While Asterisk is one of the most widely used VoIP engines the source code and the whole architecture (at least few years ago) was ugly as hell. Instead of designing the whole system for concurrency there were some pretty ugly workarounds. In addition, it was trivial to crash the system with a little bit of SIP fuzzing (and some of those crashes may have been exploitable).
Yate on the other hand - while not having as many features as Asterisk - was just beautiful. A really nice architecture and extremely good readable C++ source code. In fact, it was a real joy to study the source.
Unfortunately I didn't really had a chance to study Freeswitch, as it was pretty new at this time and therefore not an option. From what I've seen it seems better architected and more stable than Asterisk. The main difference to Yate is that Yate only has few external dependencies, while Freeswitch tries to utilize as many external libraries as possible (e.g., instead of implementing SIP itself it uses an external library).
If I'd had to choose a VoIP engine today (and if a simple SIP proxy wouldn't be sufficient), Yate would be pretty much at the top of the list.
And the team behind it is really cool - met them on the 2.0 launch party and they were extremely friendly. They're honest about the project priorities too. It's basically a "what feature do you want to sponsor" project. (and I mean it in a good way)
Years ago I've examined different VoIP engines for a project:
While Asterisk is one of the most widely used VoIP engines the source code and the whole architecture (at least few years ago) was ugly as hell. Instead of designing the whole system for concurrency there were some pretty ugly workarounds. In addition, it was trivial to crash the system with a little bit of SIP fuzzing (and some of those crashes may have been exploitable).
Yate on the other hand - while not having as many features as Asterisk - was just beautiful. A really nice architecture and extremely good readable C++ source code. In fact, it was a real joy to study the source.
Unfortunately I didn't really had a chance to study Freeswitch, as it was pretty new at this time and therefore not an option. From what I've seen it seems better architected and more stable than Asterisk. The main difference to Yate is that Yate only has few external dependencies, while Freeswitch tries to utilize as many external libraries as possible (e.g., instead of implementing SIP itself it uses an external library).
If I'd had to choose a VoIP engine today (and if a simple SIP proxy wouldn't be sufficient), Yate would be pretty much at the top of the list.