How does WebRTC handle audio and video? How to control media stream quality?

WebRTC handles audio and video through the following mechanisms:

1. Media Capture:

   • Uses getUserMedia() API to access the user's camera and microphone
   • Can specify media constraints such as resolution, frame rate, audio sampling rate, etc.

2. Media Processing:

   • Audio Processing: Includes echo cancellation (AEC), noise suppression (NS), automatic gain control (AGC), etc.
   • Video Processing: Includes video encoding, decoding, adaptive bitrate adjustment, etc.

3. Media Transmission:

   • Uses SRTP (Secure Real-time Transport Protocol) to encrypt media data transmission
   • Uses RTP (Real-time Transport Protocol) to encapsulate media data
   • Supports DTLS-SRTP key negotiation to ensure media transmission security

Methods to control media stream quality:

1. Media Constraints:

   • Set constraints in getUserMedia(), such as { video: { width: 1280, height: 720, frameRate: 30 } }
   • Set direction and preferred codecs in RTCPeerConnection.addTransceiver()

2. Bandwidth Management:

   • Use RTCPeerConnection.setParameters() to adjust bandwidth limits
   • Set total bandwidth through the b=AS field in SDP
   • Use RTCPeerConnection.getStats() to monitor bandwidth usage

3. Adaptive Bitrate:

   • WebRTC has a built-in adaptive bitrate (ARQ) mechanism that automatically adjusts bitrate based on network conditions
   • Can monitor network conditions through RTCRemoteInboundRtpStreamStats
   • Can implement custom bandwidth estimation algorithms to optimize video quality

4. Network Adaptation:

   • Uses NACK (Negative Acknowledgment) and FEC (Forward Error Correction) to improve transmission reliability
   • Smoothes network jitter through Jitter Buffer
   • Reduces video resolution or frame rate when network conditions are poor, prioritizing audio quality