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. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- 01295 712110 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users