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 mi
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)
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 pac