Enabling jitter buffer: Difference between revisions
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jbforce=no | jbforce=no | ||
jbmaxsize=1000 | jbmaxsize=1000 | ||
jbimpl=adaptive | |||
</pre> | </pre> | ||
Latest revision as of 14:11, 13 July 2016
Jitter buffer is not enabled by default, but in case of network problems, it can be enabled to improve the call quality despite a n higher call latency.
To enable it you need to edit /etc/asterisk/sip.conf, adding one of more of the following commands:
jbenable = yes|no : Enables the use of a jitterbuffer on the receiving side of a SIP channel. jbforce = yes|no : Forces the use of a jitterbuffer on the receive side of a SIP channel. jbmaxsize = Number : Max length of the jitterbuffer in milliseconds. jbimpl = fixed|adaptive: Jitterbuffer implementation, used on the receiving side of a SIP channel. Two implementations are currently available - "fixed" (with size always equals to jbmaxsize) and "adaptive" (with variable size).
A typical activation includes:
jbenable=yes jbforce=no jbmaxsize=1000 jbimpl=adaptive
Once the jitter buffer has been activated in sip.conf, you need to reload the sip module using:
asterisk -rx 'sip reload'