The setup:
Plantronics USB CS50 Headset > USB Port of "Thin Client" (Running Win Pro) > (Wireless) RDP to Remote Desktop Server > Eyebeam Softphone Client > VOIP Server > Caller (cell phone or another in-office extension)
During a live phone call, the incoming voice (From the caller played through the USB Headset on the remote client computer) is all broken up and barely intelligible.
The outgoing voice recorded from the remote client computer and sent to the caller is flawless.
Any other audio playback is flawless as well. For example, the caller can call and leave a voicemail on the VOIP server and then the user can play the voicemail back through the headset on the remote client and it's clear as day.
Is there anything I can do to improve the audio playback sound quality during a live call?
Note: I am currently using "Play on this computer" versus "Play on remote computer" as the server does not have a sound card installed.
Your setup is suboptimal -- I'd even go so far as to say that the setup you described is actively hostile to VoIP communications.
A few key items: 1. You're sending video and audio both ways over the RDP session. 2. You're going over a wireless network.
While this often works fine it's generally not a great idea, especially on common WiFi (not dedicated to VoIP) - lots of traffic/noise can delay or lose voice packets. 3. If you're not the only person on the RDP server the shared workload can really mess you up.
VoIP generally doesn't play well with virtualization... 4. SIP is UDP - Packets that arrive out of order get dropped.
All the stuff above can lead to dropped/out-of-order packets.
Things to consider:
The solution to this problem was to utilize USB over Ethernet software.