Search jitter in sample sip.conf. Everything is well documented there.
Regards,
Qasim
On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf muyous...@gmail.comwrote:
I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in
server. I am having some issue in audio quality. I want to enable jitter
buffer on asterisk but don't know, how to do. Any one can help me.
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