So I am still trying to get my VOIP over DSL as reliable as possible. I've enabled variable jitter buffers of up to 200mS on all inbound audio streams, figuring that a bit of delay is preferable to dropped packets. Generally this works very well, so even when my line quality is relatively poor, I have no problem hearing the party at the other end.
I've spent a bit of time talking to my VOIP provider to see if they also do or can implement variable buffers to help compensate for poor line quality in the other direction. Their first response was that they do not need buffers as they are connected directly to their service provider who in turn is connected directly to the PSTN. I pointed out that the buffers would be useful, not between them and their provider, but between them and ME. They said they'd pass my concern up to their technical department. That is where this has sat for a couple weeks now. Does anyone know much about VOIP providers? Do they tend to do much in the way of jitter buffers on the audio streams they receive from their customers? Are there any that are better than other? Perhaps letting their customers customize the buffer parameters themselves? I have times (particularly in the evening) when the jitter can be quite bad so a buffer of 200mS would be very useful. Other times there is little jitter so a buffer of 50mS would be sufficient. It would be great to know if my VOIP provider; first uses buffers at all; second, allows variable buffers as I have with my asterisk setup. cheers, darryl --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
