Guide Overview
This section provides detailed guides for various features of the webrtc-java library.
Media Basics
- Media Devices - Working with audio and video devices
- Bitrate and Framerate Constraints - Controlling media quality
- Send-only and Receive-only - Configure transceiver directions (send-only, receive-only or inactive)
- Media Files - Sending video and audio files instead of a camera and microphone
- Media Recording - Recording what a call sends or receives into a media file, without re-encoding
Audio
- Audio Device Selection - Selecting and configuring audio devices
- Audio Processing - Voice processing components
- Custom Audio Source - Using custom audio sources with WebRTC
- Headless Audio - Playout pull without touching real OS audio devices
- DTMF Sender - Sending DTMF tones in a call
Video
- Camera Capture - Capturing video from cameras
- Desktop Capture - Capturing and sharing screens and windows
- Custom Video Source - Using custom video sources with WebRTC
Data Communication
- Data Channels - Sending and receiving arbitrary data between peers
Networking and ICE
- Peer Connection Configuration - Candidate gathering, ICE checks, TURN, SRTP ciphers and other connection settings
- Port Allocator Config - Restrict ICE port ranges and control candidate gathering behavior
Monitoring and Debugging
- RTC Stats - Monitoring connection quality and performance
- Logging - Configuring and using the logging system
Advanced
- Field Trials - Enabling experimental features and tuning WebRTC internals
- Encoded Transforms - Reading and changing encoded frames, e.g. for end-to-end encryption
- Video Codecs - Adding video codecs implemented in Java, or limiting the built-in ones
- Audio Codecs - Limiting and ordering the built-in audio codecs
Additional Resources
For a complete API reference, check the JavaDoc.