WebRTC inside libtgvoip protocol
Because of the source code of libtgvoip is difficult to read, it would be wasting time if I tried to improve it. Instead, I've rewritten libtgvoip with a strong OOP approach and tried to save compatibility with old protocols. Instead of using OPUS codec inside with tuned parameters on-the-fly, I replaced it by WebRTC stream. The task is very ambitious, so not all parts were completed and work well.

The strengths:
- Telegram relay servers are used with the original encryption algorithm;
- possibility to reimplement old GrVP protocol to maintain backward compatibility;
- MediaEngineWebrtc class as a standalone solution of WebRTC Native API;
- very simple build step (which builds WebRTC with required options under the hood);
- strong OOP approach.

Works bad or doesn't work:
- only the last added endpoint is used: endpoint switching is not implemented yet;
- tcp relay is called "obfuscated" and didn't tested at all;
- (fixed!) udp relay is not delivering all packets to the peer (probably I missed some details in implementation);
- (fixed!) init stage of call flow is skipped due to the previous point;
- there is a lot of WebRTC logs in the stderr which are not switching off;
- call statistic is missing.

How to build
(fixed!) Just run `cmake . && make`. It should automatically download and compile all dependencies, including WebRTC! If something goes wrong, let me know.

How to run
Small modification into [tgvoipcall](https://github.com/telegramdesktop/tgvoip-test-suite/blob/master/src/tgvoipcall/main.cpp) is required: delete the line `rtc::LogMessage::SetLogToStderr(false);`. After that use libtgvoip.so as libtgvoip.so.0 for tgvoipcall.

To make the solution better, I fixed the following problems of original submission:
- build steps (of downloading and compile all WebRTC dependencies);
- udp relay is not delivering all packets to the peer (probably I missed some details in implementation);
- init stage of call flow is skipped due to the previous point;
- missed README and deb-packages.txt files.

https://www.dropbox.com/s/e8ta2x3bazlyml6/result.zip?dl=1

A promising implementation of an ambitious task.
Compatibility with current relay servers is a plus.
Huge room for improvement in terms of audio quality.

We expect the contestant to take part in round 3. Suggested priorities:
1. Focus on improving audio quality on different connections.
2. Build library for Android (real device I/O etc).

no readme :(

There are no README and build instructions 
Even more it was builded with wrong dependencies:
$ ldd ./tgvoipcall 
...
libtgvoip.so.0 => not found
libopus.so.0 => not found
...

"You should build" - I should not, sorry

This is not an issue, just a request. If there were any specific network conditions which you used to optimize the initally provided library, you are welcome to share test sequences as well as your results. e.g.:

->loss(12)->rateControl('32kbit')->networkType('3g')
Stable version: 3.0
My submission: 3.3

->loss(3)->rateControl('64kbit')->after(3)->loss(20)->rateControl('8kbit')->after(3)->rateControl('64kbit')->networkType('3g')
Stable version: 3.2
My submission: 3.4

We will re-check your results and may use the sequences you provide to test other submissions.

First of all, do not try to reach me for a private messages.

Then,
https://drive.google.com/open?id=1bO2O5rhPLlU7YESRKjTRArL62vV_4TuR
Look at the 2nd example. tgvoiprate esimates it in 1,24... While Dejavu's confidence is 242 (it's about the same as the original sample).
What could you say?