Guillermo Salas M a écrit :

Hi,

I need to have some SIP extentions on remote places where the latency
from my asterisk box with public ip is 1~1.5 seconds.

What codec will work fine on this sceneary? I'm planning to use iLBC, is
a good choice?
There are basically three parameters I can think of when speaking of voice over ip quality:

1 - Lag. In your case, a ping from your Asterisk box is 1 to 1.5 ms. Changing codecs is not going to help you here.

2 - Jitter. In your case, if the ping does vary between 1 and 1.5, that's 500ms "ping jitter", which is high. You might want to have a large jitter buffer to compensate for it. But this increases lag even more...

3 - Packet drop. iLBC is meant to cope "better" with packet drop than other codecs, although in my experience any codec with too much packet drop will sound dreadful.

If you have the bandwith and no packet loss, I would recommend that you bump up the jitter and stick with ulaw. While there might be a lot of lag - "half duplex" kind of conversations... - the audio should remain clear.

If you are having packet loss on top of this, you might want to try iLBC...

At any rate, nothing is going to replace trying out some settings for yourself...

BTW: How come the latency is so high? The worst I've seen so far was a link varying between 600 and 1200ms and the quality varied from "good enough" to "pretty horrible"...

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
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