<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>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.<br><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">De :</span></b> Mark Michelson <mmichelson@digium.com><br><b><span style="font-weight: bold;">À :</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Cc :</span></b> asterisk-users-bounces@lists.digium.com<br><b><span style="font-weight: bold;">Envoyé le :</span></b> Lundi, 15 Décembre
2008, 23h06mn 08s<br><b><span style="font-weight: bold;">Objet :</span></b> Re: [asterisk-users] tcpdum<br></font><br>michel freiha wrote:<br>> Dear Sir,<br>> <br>> What I'm interested to is to know how much time the rtp packets takes <br>> from the time it access the asterisk server,to when it'll leave<br>> Is this function or variable exist anywhere?<br>> <br><br>If you want statistics on RTP packets, then you should look into RTCP reporting. <br>A simple facility for looking at this information would be the Asterisk CLI <br>commands "rtcp stats on" and "rtcp debug" assuming that you are running Asterisk <br>1.4. If you are using Asterisk trunk, the commands are "rtcp set stats on" and <br>"rtcp set debug on". You may also be able to filter the RTCP packets in a <br>program like wireshark and analyze them there as well.<br><br>Mark Michelson<br><br>_______________________________________________<br>-- Bandwidth and Colocation
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