RTMP Server

Overview

This project is an RTMP server inspired by LiveKit Ingress Service. It receives an RTMP stream from the user in a room, transcodes the audio from AAC to Opus (making it WebRTC compatible) and the video to H264, and then pushes it to WebRTC tracks connected to clients. The server acts as a peer, maintaining a P2P connection with each client.

Features

• RTMP to WebRTC: Receives RTMP streams and delivers them to WebRTC clients.
• Audio Transcoding: Transcodes AAC audio to Opus for WebRTC compatibility.
• Video Transcoding: Ensures video is encoded in H264 for WebRTC delivery.
• Webhook Notifications: Uses webhooks to notify the publishing state of the stream to different rooms.
• WebSocket Signaling: Establishes WebRTC connections using WebSockets for offer/answer exchange.
• Concurrency for Performance: Utilizes Go's concurrency patterns and channels to enhance streaming performance and reduce latency.

Core Libraries and Packages

• Pion WebRTC: Used for handling WebRTC connections.
• Yuptopp RTMP: Used for handling RTMP streams.
• fdkaac: For AAC decoding.
• gopkg.in/hraban/opus.v2: For Opus encoding.
• go-chi/chi: Lightweight, idiomatic, and composable router for building Go HTTP services.
• logrus: For logging.

Installation

Clone the repository:

```sh
git clone https://github.com/Harshitk-cp/rtmp_server.git
cd rtmp_server

How It Works

RTMP Server

The RTMP server listens for incoming RTMP streams. When a stream is published:

• Audio Processing: Decodes AAC audio and encodes it into Opus format using fdkaac and opus.
• Video Processing: Ensures the video stream is in H264 format.
• WebRTC Integration: Sends processed audio and video to WebRTC tracks connected to clients.

WebRTC Connection

The WebRTC connection is established via WebSockets:

• WebSocket Handler: Manages WebRTC signaling (offer/answer exchange) using WebSockets.
• Peer Connection: Each client establishes a peer connection with the server.
• Track Delivery: Delivers audio and video tracks to clients via WebRTC.

Webhooks

Webhooks lietens to the audio and video channel and notify the state of streams to it's subscribers:

• Notifications: Sent when streams start or stop using a webhook manager.

Performance Enhancements

• Concurrency: Utilizes Go's concurrency patterns to efficiently handle multiple streams.
• Channels: Uses channels for buffering video and audio data, ensuring smooth delivery to WebRTC tracks.
• Optimized Transcoding: Efficiently transcodes audio and video to minimize latency.