I've set up a remote callcenter routing calls through an IPSEC tunnel. Unfortunately I am having some issues with call quality (dropouts and gaps). I've investigated every option and I've finally realized that the internet connection in our remote site has unstable ping times (averages around 10-20ms but once every minute it can hit 100-150ms).
I am willing to trade some delay in voice in order to deal with this and I thought enabling jitter buffer would help.
I am using Freepbx 18.104.22.168 and asterisk 22.214.171.124
How can I enable Jitter Buffer?