Check mark 
means it has been implemented already.
Do next. More complex to implement, but are the most important for making FeenPhone a “must have”, and for making it future proof:
–BETTER ERROR CORRECTION to deal with non-perfect networks.
–IPv6 support.
–WASAPI on audio in.
–Automatic port forwarding.
–Ability to raise volume of input to 150%
–More robust password authentication.
–Recording (more info HERE)
–Hot Swapping mics and headphones
–a couple small issues: cursor should go straight into text fields to enter Server. And highlighting “Change me” should delete that whole placeholder, not just part of it. Change BTC address to website with updatable address.
–ANDROID SUPPORT for easily doing remotes.
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PROOF OF CONCEPT:
P2P HQ mono true-duplex audio direct from one computer to another via Opus codec.
No noise reduction or echo cancellation, by choice.
Solid networking built from the ground up.
Extensible for other protocols.
Drop-down menus to pick audio input, audio output, buffer settings and codec settings.
Text chat for troubleshooting.
Saves settings (including remote IP) on exit.
Buffer meter
BASIC REQUIREMENTS:
Low-latency
No creep in latency.
Roll separate server module into program.
Multi-protocol support (TCP, UDP, Telnet)
Master volume VU meter
Additional audio format / codec: PCM, G.722. (were later removed from interface, didn’t sound as good as Opus)
Minimal clicking and drop-outs.
Doesn’t crash.
Works almost flawlessly for two hours of solid two-way talk.
Ability to add third person to conversation.
More variations in Opus settings.
Default to best buffer settings on all three sliders.
Input VU meter for each host
Volume sliders
Default to Opus settings: Sample rate 24 kHz, Bit rate: 32 Kb/s, Super Wide Band (12 kHz) but variable.
Remove codec settings that don’t test well or aren’t needed.
Decide proper license.
Make buffer reset button a little bigger, and easier to see.
Label buffer reset “Buff Dump” (radio broadcasters will confuse simply “Dump” with a cuss-dump button).
Headphone icon on Audio Out section
Hide little-used options, make accessible with Advanced Options button
About Tab with: Readme, manual, credits, License, donate link, BTC address and web links.
Basic icons and minimalist branding.
ADDITIONAL GOALS FOR 1.0 RELEASE TO PUBLIC:
Extensive testing completed, bugs worked out.
Works flawlessly for two hours of solid two-way talk.
Well-written, easy to understand manual.
GOAL LIST FOR FULL SECOND ROUND OF FUNDING:
Create installer, package for public use.
Code added on GitHub
Make volume slider on Audio Out work more smoothly.
Switchable auto-answer function (with password protection).
Input / Output level sliders.
–Change “Run Server” button to a tick box, or make it change color when pressed.
–Add a “Phone book” to save different IP connections and add name or label.
–Cough button
–Audio alerts on incoming call (DJ’s “FeenPhone, FeenPhone” sample.)
–Add call clock (for each caller, in case the third person comes in after the second person.)
–Hide WASAPI option for Audio In drop down.
–Passwords not obscured in interface.
–WASAPI support for Audio In / Lower latency in program.
–Fix issue with passwords, not logging on perfect every time.
–One-direction only option.
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.OTHER FEATURES TO CONSIDER ADDING
SIP support
Port randomizing option (within a range).
–Ability to mute and block persons in conversation.
Ability to block IPs.
–Advanced Option: Toggable gentle EQ boost at 12hz. And/or maybe a 4-band EQ with sliders.
IP address display, and ability to copy IP address to the clipboard.
See what else we can move to advanced options.
–Auto-reconnect if the connection drops.
Add ability to save different configurations, and name them so you call them up easily.
Option to auto-adjust settings for best sound and lowest latency depending on network.
IP ACL
RADIUS support
Ability to use stereo transmission.
Ability to roll over to a second pre-set IP if first connection is lost.
Mix Minus (this would be a holy grail, it would make this a killer app by replacing audio hardware).
Automation interface with relay closures
PAN left and right: for recording two-host shows for later mixing.
General overall improvements in all functionalities.
WebRTC support (for callers calling in to a radio show or podcast).
Pro interface and website.
No clicking or drop outs.
Windows Phone version.
WISH LIST OF FEATURES ADDED BY OTHERS, VIA OPEN-SOURCING:
Linux version.
Mac version.
Android version.
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