Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config:
<param name="use-jb-lookahead" value="1"/> .
The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan:
<action application="set" data="jitterbuffer_msec=2p:25p:"/>
And then enable PLC :
<action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards,
Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orgJoshAny help with this would be appreciated! Thanks in advance.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.The problem that I am trying to solve is this...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?1) What is the best codec to use when there is going to be jitter and some packetloss?Hello,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.
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