[asterisk-users] SIP Channel jitter buffer issue

[Mayur]

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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