⭐ feat: Migrate from WebSocket to libp2p for peer-to-peer connectivity (#286)

## Description
Whew, some stuff is still not re-implemented, but it's working!

Rabbit's gonna explode with the amount of changes I reckon 😅



<!-- This is an auto-generated comment: release notes by coderabbit.ai
-->
## Summary by CodeRabbit

- **New Features**
- Introduced a peer-to-peer relay system using libp2p with enhanced
stream forwarding, room state synchronization, and mDNS peer discovery.
- Added decentralized room and participant management, metrics
publishing, and safe, size-limited, concurrent message streaming with
robust framing and callback dispatching.
- Implemented asynchronous, callback-driven message handling over custom
libp2p streams replacing WebSocket signaling.
- **Improvements**
- Migrated signaling and stream protocols from WebSocket to libp2p,
improving reliability and scalability.
- Simplified configuration and environment variables, removing
deprecated flags and adding persistent data support.
- Enhanced logging, error handling, and connection management for better
observability and robustness.
- Refined RTP header extension registration and NAT IP handling for
improved WebRTC performance.
- **Bug Fixes**
- Improved ICE candidate buffering and SDP negotiation in WebRTC
connections.
  - Fixed NAT IP and UDP port range configuration issues.
- **Refactor**
- Modularized codebase, reorganized relay and server logic, and removed
deprecated WebSocket-based components.
- Streamlined message structures, removed obsolete enums and message
types, and simplified SafeMap concurrency.
- Replaced WebSocket signaling with libp2p stream protocols in server
and relay components.
- **Chores**
- Updated and cleaned dependencies across Go, Rust, and JavaScript
packages.
  - Added `.gitignore` for persistent data directory in relay package.
<!-- end of auto-generated comment: release notes by coderabbit.ai -->

---------

Co-authored-by: DatCaptainHorse <DatCaptainHorse@users.noreply.github.com>
Co-authored-by: Philipp Neumann <3daquawolf@gmail.com>

packages/relay/internal/common/extensions.go

@@ -1,11 +1,45 @@
 package common
 
+import "github.com/pion/webrtc/v4"
+
 const (
 	ExtensionPlayoutDelay string = "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
 )
 
-// ExtensionMap maps URIs to their IDs based on registration order
-// IMPORTANT: This must match the order in which extensions are registered in common.go!
-var ExtensionMap = map[string]uint8{
-	ExtensionPlayoutDelay: 1,
+// ExtensionMap maps audio/video extension URIs to their IDs based on registration order
+var ExtensionMap = map[webrtc.RTPCodecType]map[string]uint8{}
+
+func RegisterExtensions(mediaEngine *webrtc.MediaEngine) error {
+	// Register additional header extensions to reduce latency
+	// Playout Delay (Video)
+	if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
+		URI: ExtensionPlayoutDelay,
+	}, webrtc.RTPCodecTypeVideo); err != nil {
+		return err
+	}
+	// Playout Delay (Audio)
+	if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
+		URI: ExtensionPlayoutDelay,
+	}, webrtc.RTPCodecTypeAudio); err != nil {
+		return err
+	}
+
+	// Register the extension IDs for both audio and video
+	ExtensionMap[webrtc.RTPCodecTypeAudio] = map[string]uint8{
+		ExtensionPlayoutDelay: 1,
+	}
+	ExtensionMap[webrtc.RTPCodecTypeVideo] = map[string]uint8{
+		ExtensionPlayoutDelay: 1,
+	}
+
+	return nil
+}
+
+func GetExtension(codecType webrtc.RTPCodecType, extURI string) (uint8, bool) {
+	cType, ok := ExtensionMap[codecType]
+	if !ok {
+		return 0, false
+	}
+	extID, ok := cType[extURI]
+	return extID, ok
 }