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Hello list,<BR>
<BR>
I have set the tos-settings in sip.conf as recommended at <A HREF="http://www.voip-info.org/wiki/view/Asterisk+sip+tos">http://www.voip-info.org/wiki/view/Asterisk+sip+tos</A> :<BR>
<BR>
sip.conf tos_sip <B>cs3</B> <BR>
sip.conf tos_audio <B>ef</B><BR>
<BR>
But there is still jitter and audio delay. What other measures can I take ??<BR>
<BR>
Zoiper softphone --> D-Link router --> ADSL (ISP) --> Asterisk-server --> ITSP --> rest of the world<BR>
<BR>
The same TOS-settings for sip and audio are set in the Zoiper softphone.<BR>
On the workstation there is some minimal web browsing, no hardcore downloading or file transfer.<BR>
<BR>
Kind regards.<BR>
<BR>
<BR>
On Tue, 2010-03-23 at 17:21 +0100, jonas kellens wrote:<BR>
<BLOCKQUOTE TYPE=CITE>
Hello list,<BR>
<BR>
what can I do to minimalize the jitter in SIP-calls at server level ?<BR>
<BR>
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.<BR>
<BR>
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 ??<BR>
<BR>
<BR>
Kind regards,<BR>
<BR>
Jonas.
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