while its considered to keep packet loss below 1% for voip audio calls. Is there any way in freepbx we can do to keep the call connected on say upto 5 % packet loss ?
In CUCM realm we have noticed no call drops at least even on 5 % packet loss. however on freepbx as soon as the packet loss nears 1% the call drops.
What reason is FreePBX (Asterisk) giving for dropping the call? I can’t think of anything that would cause it to drop a call at 1% loss, if the loss was evenly distributed. As far a I know, it would need to lose several consecutive seconds of media, or multiple retransmissions of the same session refresh re-INVITE to drop the call.
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