Hello,
We are running a small communal network primarily based on Ubiquiti wifi wireless equipment. Some of our clients have been complaining about poor audio quality on their VOIP calls.
From the wireshark traces I have noticed that some packets are being buffered, causing a gap in communication (up to 300 ms) followed by followed by all packets immediately after one another.
In the RTP player I saw three options for Playback timing - Jitter buffer, RTP timestamp and uninterrupted mode. Which one should be used to give the most representative behavior of a VOIP phone? (I am using an Asterisk server, installed at the client, which plays a continuous playback)
