[asterisk-users] Minimalize jitter in VoIP calls

[jonas kellens]

Hello list,

I have set the tos-settings in sip.conf as recommended at
http://www.voip-info.org/wiki/view/Asterisk+sip+tos :

sip.conf tos_sip cs3 
sip.conf tos_audio ef

But there is still jitter and audio delay. What other measures can I
take ??

Zoiper softphone --> D-Link router --> ADSL (ISP) --> Asterisk-server
--> ITSP --> rest of the world

The same TOS-settings for sip and audio are set in the Zoiper softphone.
On the workstation there is some minimal web browsing, no hardcore
downloading or file transfer.

Kind regards.


On Tue, 2010-03-23 at 17:21 +0100, jonas kellens wrote:

> Hello list,
> 
> what can I do to minimalize the jitter in SIP-calls at server level ?
> 
> If at local network level, there is a VoIP-router and their is a
> physical network dedicated to IP-phones, but there is still jitter.
> 
> When using a Hosted Asterisk server, which settings on the
> Asterisk-server can minimalize the jitter between the VoIP-router and
> the Asterisk-server on the public internet ??
> 
> 
> Kind regards,
> 
> Jonas.
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