Asterisk, Freeswitch, YATE, sipXecs . . . what to choose?

It seems that recently a lot of open source projects have emerged with excellent pedigree. In a nutshell, Mighty Asterisk has got some competition.

FreeSwitch has recently released their first non-beta version (which is already 1.0.1), and their community is exploding. FreeSwitch has performance capabilities that are very impressive, and if you are a developer of complex or carrier-grade voice applications, you owe it to yourself to give FreeSwitch a careful look.

YATE has been around for almost 5 years, and is well-regarded for it’s rock-solid stability. It has fantastic performance capabilities as well, and is solidly carrier-grade. The YATE community is a bit lean, and the documentation is a bit too light, but this powerful engine should not be ignored.

sipXecs evolved out of Pingtel, who recognized the importance of open source telecom and started SIPFoundry. These folks are serious about SIP, and cannot be ignored. Many of the SIPFoundry folks contribute to the IETF SIP standard, so they are well-respected by the community. Recently, Nortel has created a product that uses this technology, which demonstrates the professionalism and technical accomplishment of this product. Also, Amazon uses this for their PBX, so you know it scales and is reliable.

Open source telecom has grown up, and it’s going to be exciting to see how these projects mature and evolve. One thing is certain, all this innovation in the open source telecom space is certain to benefit both solution providers, and–more importantly–customers.