Hi, Forgot to mention that I am using Asterisk version 1.4.15 running on RHEL 3.0 server.
Regards, Mayur _____ From: Mayur [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 26, 2007 7:03 PM To: 'asterisk-users@lists.digium.com' Subject: SIP Channel jitter buffer issue Hi, I have a SIP client which is registered to asterisk. Asterisk is registered to a SIP trunk and also handles the media. Now since my client has some issues in its RTP Tx, which seems to have some amount of jitter (mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and max delta is 85 ms), to over come that I have enabled jitter buffer in the SIP channel by setting sip.conf parameters jenable=yes, jbforce=yes, jbmaxsize=200 and jbimpl=fixed. However on setting these parameters I am unable to hear on the trunk side. From the jitter logs as given below, I can see audio frames being dropped: JB_PUT {now=1130}: Dropped frame with ts=21125 and len=20 JB_GET {now=1130}: now < next=2121 JB_GET {now=1142}: now < next=2121 JB_GET {now=1163}: now < next=2121 JB_PUT {now=1181}: Dropped frame with ts=21132 and len=20 JB_GET {now=1181}: now < next=2121 JB_GET {now=1183}: now < next=2121 JB_PUT {now=1185}: Dropped frame with ts=21132 and len=20 JB_GET {now=1185}: now < next=2121 I have tried increasing the jitter buffer from 200 to 1000 ms but with same result. Am I missing anything here? How can I determine what is causing asterisk to drop the audio frames? Regards, Mayur
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