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.
.
.
.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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Is it because the programmer got a windows phone that it would be prioritised above android?
And is the second indiegogo found list what you want to get done first in general?
https://en.wikipedia.org/wiki/ZRTP
I have heard this encryption should be good and secure of NSA intrusion according to what this guy said in this talk https://www.youtube.com/watch?v=0SgGMj3Mf88 I don’t remember when he says it though
The programmer is mainly a Windows programmer. That’s why Windows phone is on the main list and Android is not.
Thanks for this nice app!
Can i put some wishes here:
– autentication per user
– ip whitelisting or ip blocking
– add FLAC codec (also used by comrex bric link) WAV quality less bandwith and even low latency
– ipv6 support
– Stereo support, but also the ability to force a incomming connection to be mono or mix left and right together
– more codec parameters to be influenced by the user.
Kind regards,
Dennis
I also vote for FLAC codec.
how exactly is this decentralized?
Not going through a central server. Doesn’t rely on a third party.
It’s decentralized because you and your budddy can download it and use it without relying on me in any way. Skype and other VoIPs rely on Skype’s servers or some other third party’s servers to connect.
MWD
Great stuff – one suggestion: during the live broadcast, each node records its high quality mic audio by itself to a separate disk file. At the conclusion of the broadcast, the nodes share their individual recording files with the master node, which mixes them into the final audio file. In this way, any network imperfections that happen during the live broadcast are eliminated in the final audio file (for use in later repeat broadcasts, or in a podcast). Could also be useful for off-the-air recordings of interviews over crappy network connections, where the final product is extremely high quality.
Gary:
http://feenphone.com/?p=2791
That is a planned feature, and would enable what you’re describing.
However, I’ve done a lot of double-ender podcasts. It’s a heck of a lot of work. Using FeenPhone as is, getting the levels right on the fly, I produce shows that are 99% as good as double-enders with about 1/7th the time and work.
Have you actually ever done one? It’s not as easy as it sounds. Since there’s no timecode, there is synch drift between machines. For a long show, it takes a lot of moving things around to approximate it being right. Adding timecode support to FeenPhone would be a wrong move in the direction of software bloat.
Besides, what you’re describing can already be done with Skype. We’re putting our limited time and effort into doing things that Skype does not already do.
Thank you for the kudos! Have you tried FeenPhone yet?
FYI, these are the only features we’ll likely be adding. These are much more important for making FeenPhone a future-proof killer app than any little bells and whistles like you’re describing that can already be done with FeenPhone (or Skype!), a cheap hardware mixer and a free copy of Audacity:
–IPv6 support.
–WASAPI support for Audio In / Lower latency in program.
–Recording.
–Encryption.
–Automatic port forwarding.
Peace,
MWD
Gary, here’s our motto:
“Great software is like aspirin: It does very few things, does them very well, has few side effects, isn’t expensive, has no patent and is easily available everywhere.”
— Beastlick Internet Policy Commission Outreach Team
http://bipcot.org/?page_id=29
I don’t see any way for skype to do what I suggest. What are you referring to?
If you’re suggesting recording each side of a call then sewing it together later, I used to do that all the time. You wear headphones, talk over Skype, but don’t record the Skype stream. Each person has a mic in front of them recording to a separate computer or hard disc recorder, Then one person FTPs their file to the other person, and the other person mixes them together in a multi-track program. Skype is just to have the conversation, the recording is done separately. It can actually be done with one mic on each end, with a mixer set up to do a mix minus. Neema Vedadi and I used to do this all the time. It’s called “a double-ender podcast.” It sounds very good and it’s a boat load of work.
If that’s not what you’re suggesting, you didn’t explain it well enough for me to grok it.
http://thepodcastersstudio.com/tps-ep-47-recording-a-double-ender-podcast/
http://www.summitsolutions.co.uk/blog/double-enders-using-skype-and-audacity-some-tips-and-a-few-things-to-watch-out-for/
Sounds really cool, call fidelity seems to be the last remaining hole in the game. Sorry to be ‘that guy’, but Mac? Iphones and macbooks ain’t exactly a small market segment if you’re looking to expand…
hey that guy,
re: Mac, we’re two guys, and not Mac programmers.
Get right on that and make it! It’s open source.
Thank you.
MWD
Did FLAC drop from the wishlist?
Preferred above OPUS for STL connections.
Dennis
Make it! Go ahead!
Dennis,
you are a programmer, right? Instead of telling us what to do over and over, why not help us make it happen?
Let me know and I’ll give you a list of what needs doing, plus you do whatever you think needs doing. How’s that?
Worms.
Sorry for asking what happened and why!
I do not tel, i asked kindly!
But hey goodbye
What happened is we have no money and limited resources and your request got knocked off the list in favor of things that improve the overall solidness of the program. And again, feel free to help us if you want.