Asterisk Jitter

ad so that I can get to the real Asterisk freelancers quickly... I have 6 years of experience in asterisk and I can solve the issue quickly...

Hello! I'm VoIP Expert with more than 7 years experience in Asterisk and a2billing systems. In order to measure jitter in asterisk you can use rtcp debug and rtcp stats command. details - in pm High quality and post-project support guaranteed! Check my profile and let's start!

Hi, Jitter can not be measure from asterisk. Jitter is a pure network latency deviation. In that matter asterisk is only able to measure RTT in quite un proper way. Only way to measure is to capture actual RTP traffic and graph the delta between each two RTP packets. I am not sure who and how some can help you if you have big jitter over your network link. How much is it? What are you experiencing as an effect? Do you have a packet capture? Which codecs are you using? Do you use plc ?

Hi I am talented and expert VoIP and Collaboration engineer having years of experience on enterprise networks. I assumed that u have some issue with jitter. the only thing that u can optimize is buffering and to solve the jitter problem u shoudl improve the network quality. Regards.

so that I can get to the real Asterisk freelancers quickly.. hello i have much experience with asterisk and voip solutions, and can help you analyse your jilter traffic. well there are many ways to measure the kilter, one will be using tcpdump to fetch the output stream the analyse by comparing the RTP timestamps.

Hello -, I have read your requirement and think that I can help you in this. I have over 9 years of experience in Telecom Industry and also more than 5 years of experience implementing several features in Asterisk & other open source products. You can check the Jitter usage in a channel using "CHANNEL()' function. Watch out for 'txjitter' and 'rxjitter' Cheers, Kiran

I am expert on Asterisk PBX (Freepbx, Asterisknow, Elastix and other). I started worked with Asterisk in 2005. Now I support more than 100 Asterisk VoIP servers and VoIP-gates in the whole world.

Hi ! sip show channelstats :) can you tell us that you tried to do and customize and send part of the record with the problem