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Hi Josh,<br>
<br>
Use FS master, it has a new audio Jitter Buffer.<br>
<br>
Then you can use Opus with this setting in the codec config:<br>
<param name="use-jb-lookahead" value="1"/> . <br>
The other side must be sending FEC so you could profit from it.<br>
<br>
<br>
You can enable the Jitter Buffer in the dial plan:<br>
<action application="set" data="jitterbuffer_msec=2p:25p:"/><br>
And then enable PLC :<br>
<action application="set" data="rtp_jitter_buffer_plc=true"/><br>
<br>
Regards,<br>
Dragos<br>
<br>
<br>
On 11/12/2015 07:51 AM, Josh Richesin wrote:
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cite="mid:CADedEFyXCe73H=xPVWzwecgQc9XjJ=sEJoPFepQN+0uQnC_WaA@mail.gmail.com"
type="cite">
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<div>Hello,<br>
<br>
</div>
I am running FreeSwitch for our small Wireless ISP. Obviously, as a
wireless ISP and ITSP, we have a lot of jitter and some packet loss.
Have a few questions for this setup.<br>
<br>
</div>
1) What is the best codec to use when there is going to be jitter and
some packetloss?<br>
<br>
</div>
2) The ATA's that we are using have adaptive jitter buffers, should I
enable jitterbuffer on Freeswitch as well? If so, should it be via
sofia profiles?<br>
<br>
</div>
The problem that I am trying to solve is this...<br>
<br>
</div>
Our voice customers on our freeswitch have a lot of issues with "calls
breaking up" When they use another voice provider, like say vonage or
something running off of a metaswitch, they do not have the same
issues. <br>
<br>
</div>
Any help with this would be appreciated! Thanks in advance.<br>
<br>
</div>
Josh<br>
</div>
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