It's important to highlight the benefits of this solution. In comparing with other submissions, it covers all of them:
1. Competitional quality of audio with the possibility of further improvements (in fact, not all variables were tuned up yet).
2. The codebase is good and 90% ready to deploy into production (Android build, multiple endpoints, data saving mode, TCP endpoints, proxy and so on are supported).
3. Possibility to launch video calls in weeks (thanks to WebRTC it's almost ready).
4. Possibility to launch group calls in weeks (thanks to WebRTC it's almost ready too).

I've investigated the problem and prepared the solution:
1. You need to apply the small patch for org_telegram_messenger_voip_TgVoip.cpp: https://www.dropbox.com/s/y4wsic3489eu294/jni.patch?dl=1
2. You have to include webrtc/src/out/Android/lib.java/sdk/android/libwebrtc.jar into the final APK; it's also available here: https://www.dropbox.com/s/ru35xeaw87kplmo/libwebrtc.jar?dl=1.

The correct libtgvoip.so in debug mode for the arm platform is available here: https://www.dropbox.com/s/9njgat11iavo061/libtgvoip.so?dl=1

Успешно справились со всеми заданиями.
Большой объем измненений.
Проактивные доработки: множественные рефлекторы и т д.
Библиотека собралась под Android с небольшим исправлением, успешно работает.

When we build Android version and launch it, the app crashes. Here is stack trace. Can you check if you see any reason for it?

http://paste.debian.net/1140295/

#issue10972 Here is the log with debug enabled:
http://paste.debian.net/hidden/995e0685/

It seems that check fails here:

JVM* JVM::GetInstance() {
RTC_DCHECK(g_jvm);
return g_jvm;
}

Do you see any easy fix for this?