[asterisk-users] Re : tcpdum

[Olfa Echi]

I think to reslove latency of communication, you should disable media server option on asterisk, so that RTP packets are exchanged only between the Two SIP clients.





________________________________
De : Mark Michelson <mmichelson at digium.com>
À : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Cc : asterisk-users-bounces at lists.digium.com
Envoyé le : Lundi, 15 Décembre 2008, 23h06mn 08s
Objet : Re: [asterisk-users] tcpdum

michel freiha wrote:
> Dear Sir,
> 
> What I'm interested to is to know how much time the rtp packets takes 
> from the time it access the asterisk server,to when it'll leave
> Is this function or variable exist anywhere?
> 

If you want statistics on RTP packets, then you should look into RTCP reporting. 
A simple facility for looking at this information would be the Asterisk CLI 
commands "rtcp stats on" and "rtcp debug" assuming that you are running Asterisk 
1.4. If you are using Asterisk trunk, the commands are "rtcp set stats on" and 
"rtcp set debug on". You may also be able to filter the RTCP packets in a 
program like wireshark and analyze them there as well.

Mark Michelson

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