As time passes, I changed my mind. I would suggest people integrate WebRTC based on the PeerConnection API, not the rtc_media API. Because rtc_media API is not stable, which means it maybe removed someday. When you upgrade WebRTC, you may have to change lots of integration code.
WebRTC native demo on Windows without signaling, just showing how to make use of webrtc video/audio engine.
The reason is that I can only get Javascript sample code by Google: https://github.com/webrtc/samples. But sometimes people wants to make use of WebRTC native api to set up a video/audio real-time commucation Application.
There are two native demo in the WebRTC source code: peerconnection_client for Windows, AppRTCMobile for Android/iOS.
But it's not simple enough to show how to use WebRTC video/audio engine API, because both demo set up based on libjingle_peerconnection API, not rtc_media.
Take peerconnection_client as example:
WebRTC M59 commit id: 61fe801ad874104a2d461083f53caee4c19c51b6
| Plan | Status |
|---|---|
| Video capture | done |
| video render | done |
| video codec | |
| udp transport | |
| audio |