Digging into VoIP

Well I've recently started to ramp up the testing of VoIP technologies. I decided my immediate goals are:

1. Buy a SIP/IAX capable phone to test our servers and communication to Nexxia's soft phone software.
2. Buy an account from a PSTN provider (so we can dial out to real world phones).
3. Buy a DID (incoming phone #) so the real world can dial into Nexxia and the SIP/IAX hard phone.
4. Test between all those to evaluate reliability of the server, the software and the quality of the call audio.

I've basically now have all that in place except #3 as I'm having a bit of difficulty with Globotech's instructions.

I'm pretty happy with the reliability so far. Calls seem to go through every time. So far I'm a bit dissapointed with the call quality though. I need to do further testing of audio codecs though.

Latency doesn't seem to be too much of an issue but the audio does get choppy at times so I'm not sure whats causing that. Be it the connection between me and our provider, or if its the codecs used on the SIP/IAX hard phone, or the codecs used in Nexxia.

I hope to setup some tests so Sean and myself can evaluate it further. Perhaps get some others involved for further feedback.

Thanks go out to Matt O'Gorman for helping out with the Asterisk configuration.