[asterisk-users] Minimalize jitter in VoIP calls

[John]

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.kellens at 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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