[Asterisk-Users] How to increase the performance?

[Jim Van Meggelen]

asterisk-users-bounces at lists.digium.com wrote:
> Rich Adamson schrieb:
>>> In the past I had problems with the audio over sip. Then I tried the
>>> "-p" Option and increased the memory. Now it is better but not
>>> perfect. 
>>> 
>>> Are there any more possibilities to increase it more? By now I'm
>>> using a P-II/333.
>>> 
>>> Could a completely hand optimized kernel (I use 2.6.) help a bit?
>> 
>> There's no way to answer your question with any degree of reasonable
>> truth as you haven't mentioned they type of phones, type of pstn
>> interface, which codecs, etc, etc.
> 
> Okay. My server has got:
> 
> - One Phonejack Lite
> - One X100P Clone
> - 256mb Memory
> - P II/333
> - Linux 2.6.5
> - Debian Woody
> - Asterisk 1.0.1
> - Codecs: GSM, ulaw, alaw
> - ADSL 1000kBit/s Downstream, 128kBit/s Upstream

That upstream bandwitch will need to be managed carefully. If you're
using G.711, one channel would be using roughly 80kbit of your upstream.
Who has the most quality complaints: you, or the people you are talking
to?

> Calls from or to the pstn are completely okay. Calls over SIP aren't.
> Calls over IAX couldn't be tested at the moment.

Can you make a SIP connection directly to the box? No LAN, no WAN, just
a crossover cable between your SIP phone (soft or hard) and your
Asterisk system? That'll give us some idea of whether the problem is
network or server-based.

Jim.

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