Currently I have a system that makes use of FreeSWITCH for outbound calls via SIP External with flowroute and works well, but some users complain about the quality of the call. The system is running the call using a lua script, in which you create two sessions (one for each user), and which are within the same script bridge and record the call, once both have established a connection. G711 codec is used.
Users who complain say that sometimes the audio is very low or stutters. The strange thing is that when you listen to recordings of these calls both people listen very well.
Have been testing the user and normally is poorly listening of the leg 2 of the call. Because of this and the characteristics of the system that I mention, I suspect that when bridging the communication is that the audio fails, or low quality. But I have not found anything conclusive.
I write to ask if anyone knows why this behavior can be given during calls, and because it is perceived in his recordings.