So I have a freeswitch usecase that Im having trouble solving and would like the collective help of the community. I have a customer who has a commercial PBX on site ... ( Supports SIP ) However their network connectivity is quite bad at times, and there are no other options for connectivity. ( lets say they are a mine site or something remote like that ) SIP calls on this link sound pretty bad when they experience network issues ( mainly jitter & some loss ), BUT Skype calls sound pretty good. ( not perfect, but a LOT better than say G711, G729 or any other open standard VOIP Call ) What we considered doing is deploying a freeswitch box on their site, and one in a DC with reliable connectivity. have their PBX send G711 calls to the onsite freeswitch, and have freeswitch transcode this call to OPUS / SILK. send the OPUS / SILK over the bad network to the co-located freeswitch box, who will then trancode back to G711 and set the call to the PSTN network like normal. The intention here is to leverage the Packet Loss Concealment baked into OPUS / SILK as I imagine that is what gives skype its ability to hide the audible effects of loss. So far I have only set this up in a lab and used WANem to simulate loss, jitter, out of order packets. but ive found OPUS / SILK to be no better than G711 at covering loss. I even tried with a 2 sec jitter buffer on both FS instances ( on each end ), which made no difference. I was expecting the jitter buffer to add more delay to the call, but it didnt even do this ... maybe I misconfigured something in the jitter buffers. anyways.. I would like some help with this, has anyone tried doing anything similar to this and had success ? -- Sincerely Jay Binks