Hello Taug'rs, I've had my Asterisk server set up and running for a couple of months now, and everything but sound quality is great. I've done everything I can to improve the sound quality. I'm using the QoS bits in the IP packet which helps my own router give priority to the packets from my Asterisk box, but I'm pretty sure the DSLAM ignores. I've implemented an adaptive jitter buffer of up to 200mS which did a remarkable job of improving the sound quality of incoming audio. Even at times when a computer estimate of MOS has a value of 3. It does however (predictably) have no effect on outgoing audio quality. I've enquired with my VOIP provider about what kind of jitter buffer is implementing on my packets before they hit the PSTN, but have never received a straight answer. My suspicion is that in fact they do not control this and it is somewhere further down the line. If in fact they have a standard 30-50mS fixed buffer, I'm pretty sure that would explain my poor sound. Having a delay in sound is far preferable to having a lot of dropped outgoing packets.
Does anybody know any VOIP providers around who provide or allow long adaptive buffers? I'm pretty sure this is the only way I'll get an acceptable VOIP solution. cheers, darryl --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
