Off the top of my head
- use a decent ISP
- make sure your ADSL link isn't heavily contended
- use the latest stable firmware in your router
- set QOS for outgoing packets on your router (whether your ISP takes
any notice is another matter)
- Set QOS on your router to reserve bandwidth for VOIP packets
- Trace the source of the jitter with tshark and wireshark- is it ITSP
to server (or vice versa) or server to you (or vice versa)
- make sure the server isn't overloaded, or on a poor network link, or
geographically very distant with high latency
- try a change of codec e.g. G729
- try using a hardware phone, or a different softphone e.g. xlite/ linphone
- google asterisk jitter buffer
etc. etc. etc.!
John
On 30 March 2010 15:11, jonas kellens jonas.kell...@telenet.be wrote:
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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