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Video SDK Ā· WebRTC 2.0

Production WebRTC Infrastructure
for Real-Time Video Applications

Build low-latency duplex communication, multi-peer media routing, and secure signaling workflows using WebRTC 2.0 architecture in Medialooks Video SDK.

WebRTC 2.0 extends standard WebRTC into a production-ready media infrastructure layer with signaling control, server-side processing, and broadcast-oriented features.

Explore Video SDK View Architecture Overview
WebRTC 2.0
Channel-based media model
Mix-Minus
Per-participant audio routing
GPU
Server-side media processing
.NET / C++
SDK language support
Context

WebRTC Alone Is Not Enough
for Production Video Systems

WebRTC solves real-time peer connectivity in browsers, but production-grade video systems require more than peer-to-peer transport.

WebRTC 2.0 addresses these requirements by extending WebRTC into a structured media infrastructure layer.

Typical production requirements
  • Multi-peer routing and session control
  • Secure signaling infrastructure
  • Media mixing and processing
  • Protocol interoperability (WebRTC, SRT, RTSP, RTMP)
  • GPU-based video processing pipelines
  • Monitoring and channel management
  • Deployment in controlled environments
WebRTC 2.0 addresses these requirements by extending WebRTC into a structured media infrastructure layer.
WebRTC 2.0 Architecture

WebRTC 2.0 System Architecture

WebRTC 2.0 introduces a channel-based media model built around signaling, media transport, and dynamic peer management.

šŸ”
Signaling Server
Session discovery, authentication, channel coordination
IMCChannelSGN
ā†•
šŸ“”
Channel Layer
Communication sessions ā€” signaling + media transport
IMCChannel
ā†•
āš”
Connection Engine
Dynamically creates or joins channels based on runtime state
MConnectClass
ā†•
šŸŽ¬
Media Layer
GPU-accelerated processing, mixing, and routing of audio/video streams
GPU Pipeline
Signaling
Signaling Server
Handles session discovery, authentication, and channel coordination. Replaces legacy SimpleWebRTC logic.
Channel
Channel Layer
Represents communication sessions between peers.
IMCChannelSGN

IMCChannel
Engine
Connection Engine
MConnectClass dynamically creates or joins channels based on runtime state.
Media
Media Layer
GPU-accelerated processing, mixing, and routing of audio and video streams.
What's New in WebRTC 2.0

What's New in WebRTC 2.0

šŸ”§
Updated WebRTC Core
Built on a modern WebRTC implementation with improved connection stability and performance.
šŸ—ļø
New Signaling Infrastructure
Dedicated signaling server with updated web clients optimized for desktop and mobile environments.
šŸ”„
Backward Compatibility
Existing WebRTC 1.0 applications continue to work with minimal modifications.
āš™ļø
New API Capabilities
Expanded API surface for production deployments:
  • Many-to-many duplex communication
  • Improved connection logic and reduced TURN dependency
  • Mix-minus audio processing
  • Runtime peer enumeration and metadata access
Mix-Minus Audio Processing

Professional Audio Routing with Mix-Minus

Mix-minus is an audio mixing approach where each participant receives a mix of all other participants ā€” excluding their own audio.

Mix-minus = per-participant audio mix of "everyone else, except me"
Conferencing systems Broadcast production workflows Remote contribution Live interactive sessions
Step 01
Peer Discovery & Signaling
Peers connect to the signaling server to discover other participants and establish sessions. Signaling is used only for session setup.
Step 02
Media Transmission
Once connected, audio and video streams are transmitted to the WebRTC 2.0 media layer for processing.
Step 03
Per-Peer Audio Mixing
Each participant receives a unique audio mix that explicitly excludes their own stream ā€” eliminating self-echo.
Step 04
Result
Each participant receives a clean, echo-free audio mix. Every peer hears all others, but never themselves.
Mix-Minus Configuration
// Enable mix-minus for channel current_channel.PropsSet("mix_minus.enabled", "true"); // Optional tuning current_channel.PropsSet("mix_minus.src_channels", "all"); current_channel.PropsSet("mix_minus.delay_msec", "300"); current_channel.PropsSet("mix_minus.level_db", "-10");
Per-participant audio result
A
Peer A receives B + C + D (A excluded)
B
Peer B receives A + C + D (B excluded)
C
Peer C receives A + B + D (C excluded)
D
Peer D receives A + B + C (D excluded)
Signaling Infrastructure

Production Signaling Server Architecture

Server Capabilities
  • HTTPS signaling
  • Token-based authentication
  • Peer metadata exchange
  • Channel lifecycle management
  • TURN integration
  • Docker deployment
Channel Creation Logic
server.Enum_Create(URL + "_enum", "username=test_user password=12345", " ", null, out server_manager); if (members.Count <= 0) { current_channel = CreateChannel(channel_url); } else { current_channel = ConnectToChannel(channel_url); } // Secure channel example Channel_Create(_url, "secured=true", " ", callback, out var channel); // Auth failure handling attributes.AttributesStringSet("succeded", "false"); attributes.AttributesStringSet("reason", "not_authorized");
TURN & Network Resilience

TURN Server Integration

WebRTC 2.0 supports TURN for NAT traversal in restricted networks. Configuration is propagated automatically from the signaling server to clients.

TURN is used when direct P2P is not possible due to
  • Symmetric NAT
  • Corporate firewalls
  • Restrictive networks
TURN Configuration
{ "turn_server": [ "user@password:server-ip:3478" ] }
Peer Management & Metadata

Runtime Peer Enumeration

WebRTC 2.0 allows dynamic inspection of connected peers at runtime ā€” enabling channel-aware routing decisions and real-time session management.

āš  Peers marked as "enum" are excluded from media processing.
Peer Enumeration API
server_manager.VTSG_EnumGetCount( null, out var count, null); server_manager.VTSG_EnumGetByIndex( "", i, out var channel, out var channel_props); channel_props.AttributesStringGet( "id", out var id);
Media Processing Pipeline

Server-Side Media Processing

WebRTC 2.0 supports GPU-based media workflows for server-side composition, layout, and real-time processing ā€” enabling broadcast-grade output from multi-peer sessions.

Frame compositing
Multi-view layouts
Real-time overlays
Stream mixing
Recording pipelines

Frames are combined into layouts using GPU acceleration and MFOverlay rendering.

Server-Side Frame Receive
// Receive frame from peer current_channel.ChannelReceive( 0, -1, id, out frame, ""); // Combine into GPU-accelerated layout // using MFOverlay rendering
Custom Web Clients

Custom Web Guest & Preview Applications

WebRTC 2.0 includes React-based web clients embedded in the signaling server. Developers can fully customize the interface and deployment workflow.

Customization
What Developers Can Modify
  • UI workflows
  • Authentication logic
  • Interaction patterns
  • Branded interfaces
Build Process
Development Commands
  • yarn install
  • yarn start
  • yarn build
Deployment
Deployment Steps
  • Copy build output to /webguest2/webguest2/public
  • Restart Docker container
Use Cases

Where WebRTC 2.0 Is Used

šŸŽ¬ Remote production systems
šŸ“ŗ Interactive broadcasting platforms
šŸ“· Multi-camera contribution workflows
šŸ›ļø Government and judicial communication systems
šŸ­ Industrial monitoring and teleoperation
šŸ¢ Enterprise real-time collaboration tools
Why Not Only Browser WebRTC

Browser WebRTC vs. WebRTC 2.0 Infrastructure

Capability Browser WebRTC WebRTC 2.0 Infrastructure
Topology Peer-to-peer only Managed channel architecture
Audio routing Limited audio routing Mix-minus per participant
Signaling No signaling control Full signaling infrastructure
Server processing No server processing GPU media processing
Workflow scale Simple conferencing Broadcast-grade workflows
SDK Integration

Developer Access

WebRTC 2.0 is part of the Medialooks Video SDK ecosystem, available across multiple development languages with full signaling and channel management tooling.

Supported Languages
.NET C++ Delphi
What's Included
  • Signaling examples
  • Duplex communication samples
  • Channel management API
  • TURN deployment tools
  • Web client customization
Final CTA ā€” Get Started

Explore WebRTC 2.0
Media Infrastructure

Evaluate WebRTC 2.0 architecture, signaling model, and SDK capabilities for building production-grade real-time video systems.

Explore Video SDK Contact Engineering Team