Re: [asterisk-users] Minimalize jitter in VoIP calls

2010-04-03 Thread 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.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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Re: [asterisk-users] RTCP How to stop

2010-04-03 Thread nakaji 
Thank you for your infomation. 
It is regrettable that it is not easily revokable.


 There is no configuration option for doing this; RTCP is a mandatory
 part of an RTP implementation that intends to be compliant with the
 RFCs. If you want to disable it, you'd have to modify the source code.
 
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nakaji


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[asterisk-users] asterisk 1.6.1/1.6.2 binary packages

2010-04-03 Thread Pablo Ruiz 
Hello,

Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
packages at packages.asterisk.org?

Greets.
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Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages

2010-04-03 Thread bruce bruce 
RPMs for CentOS already exist. Though, I agree with better
notification/documentation for these and the keeping up with the updates.

On Sat, Apr 3, 2010 at 8:14 AM, Pablo Ruiz pablo.r...@gmail.com wrote:

 Hello,

 Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
 packages at packages.asterisk.org?

 Greets.

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