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Just the following: The vital signs for voice are the following: Latency : Not exceeding 300ms round trip (max of 150ms one way) Packet loss: not exceeding 3%, preferably less that 1% (After 3% speech break up occurs) Jitter: max allowed is 20ms Bandwidth: G.711 (81.6 Kbps) average Bandwidth: G.729 (24.48 Kbps) average QoS for RTP = EF QoS for Signalling usually CS3 or EF
You do not mention what connectivity you have between the locations
From a Wireshark perspective I also had an experience where the call quality when listening to a call in Wireshark sounded worse then actual. This was caused when using the default setting of "Jitter" from the "Playback Timing" When I selected the "uninterrupted mode" from "playback timing" it matched and sounded the same as actual Hope this helps
