[Asterisk-Users] How to increase the performance?

[Rich Adamson]

> >>Do you need any more facts?
> > 
> > Sure, getting closer...
> > 
> > Help us understand what "calls over sip" means. From what device to what
> > device when the call is bad (need to understand the path that you're
> > talking about including any transcoding going on (if any), what type
> > of sip phone, is the sip connection local or through the dsl, and the
> > other end of this 'bad call' where is it?
> 
> Okay. The call goes to or from a phone connected to the Phonejack Lite. 
> The Asterisk Server (Version 1.0.1) converts from signed linear 
> (Phonejack) to ULAW/ALAW/GSM. Then it is transferred over ADSL to 
> sipgate.de (ping time 100ms) (most sip-calls are incoming calls from the 
> pstn to the sipgate.de-gateway)
> 
> > Current version of * or what?
> 
> No. It's the 1.0.1 since there are no newer versions available for 
> debian woody.
> 
> > Where ever this sip connection goes that is you're referring to, are
> > there any CLI errors or have you tried to use a packet sniffer to
> > see what's going on?
> 
> I haven't used any packet sniffer by now. The sound problems are 
> occuring in 10-30% of the time. So I guess this can only be a 
> performance or traffic problem. But since the sound problems are 
> occuring in both directions I guess its a performance problem.

For sip calls, asterisk essentially obtains its timing from the remote
sip device. If packets are dropped or missing for whatever reason
between the sip device and asterisk, audio will be impacted in both
directions (usually).

Given the small bandwidth noted for your dsl connection, it is entirely
possible that is the problem. But, you can verify that fairly easily
by using a packet sniffer (eg, Ethereal) and looking at a few timestamps
within the rtp packets. The timestamps should be consistently increasing
by exactly the same amount (for both directions).