Codec & voice quality


I’ve got a dual question:

First : is there a way to add commercial or open source external codec and sip stack to asterisk ? If so, how and is there someone who made tests to compare performance (CPU, voice quality, bandwitdh) ?

Second: I’m facing quite a facinating but also disapointing issue. I have a stong asymetry in quality. Calls I passing over my asterisk system are quite poor to the called side while very good on the caller side (using a TE210P card). I used zttool and ztmonitor and zttest and tried several different configuration for Polycom and GrandStream phone and other softphones, the problem remains… What is happening is that the called side is experiencing a low volume and some kind of white noise (often defined as "you’re calling from the sea side, aren’t you?) and voice is not as sharp as expected. I already played with txgain values and some codec properties but for now I have no idea.

Phones I’m using have VAD deactivated.

Thank you for your time and help.