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⭐ 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>
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@@ -2,12 +2,13 @@ package common
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import (
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"fmt"
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"log/slog"
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"strconv"
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"github.com/libp2p/go-reuseport"
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"github.com/pion/ice/v4"
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"github.com/pion/interceptor"
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"github.com/pion/webrtc/v4"
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"log/slog"
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"strconv"
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)
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var globalWebRTCAPI *webrtc.API
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@@ -24,17 +25,9 @@ func InitWebRTCAPI() error {
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// Media engine
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mediaEngine := &webrtc.MediaEngine{}
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// Register additional header extensions to reduce latency
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// Playout Delay
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if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
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URI: ExtensionPlayoutDelay,
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}, webrtc.RTPCodecTypeVideo); err != nil {
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return err
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}
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if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
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URI: ExtensionPlayoutDelay,
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}, webrtc.RTPCodecTypeAudio); err != nil {
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return err
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// Register our extensions
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if err := RegisterExtensions(mediaEngine); err != nil {
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return fmt.Errorf("failed to register extensions: %w", err)
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}
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// Default codecs cover most of our needs
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@@ -75,9 +68,10 @@ func InitWebRTCAPI() error {
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// New in v4, reduces CPU usage and latency when enabled
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settingEngine.EnableSCTPZeroChecksum(true)
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nat11IPs := GetFlags().NAT11IPs
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if len(nat11IPs) > 0 {
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settingEngine.SetNAT1To1IPs(nat11IPs, webrtc.ICECandidateTypeHost)
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nat11IP := GetFlags().NAT11IP
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if len(nat11IP) > 0 {
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settingEngine.SetNAT1To1IPs([]string{nat11IP}, webrtc.ICECandidateTypeSrflx)
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slog.Info("Using NAT 1:1 IP for WebRTC", "nat11_ip", nat11IP)
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}
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muxPort := GetFlags().UDPMuxPort
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@@ -85,7 +79,7 @@ func InitWebRTCAPI() error {
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// Use reuseport to allow multiple listeners on the same port
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pktListener, err := reuseport.ListenPacket("udp", ":"+strconv.Itoa(muxPort))
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if err != nil {
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return fmt.Errorf("failed to create UDP listener: %w", err)
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return fmt.Errorf("failed to create WebRTC muxed UDP listener: %w", err)
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}
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mux := ice.NewMultiUDPMuxDefault(ice.NewUDPMuxDefault(ice.UDPMuxParams{
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@@ -95,10 +89,13 @@ func InitWebRTCAPI() error {
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settingEngine.SetICEUDPMux(mux)
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}
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// Set the UDP port range used by WebRTC
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err = settingEngine.SetEphemeralUDPPortRange(uint16(flags.WebRTCUDPStart), uint16(flags.WebRTCUDPEnd))
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if err != nil {
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return err
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if flags.WebRTCUDPStart > 0 && flags.WebRTCUDPEnd > 0 && flags.WebRTCUDPStart < flags.WebRTCUDPEnd {
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// Set the UDP port range used by WebRTC
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err = settingEngine.SetEphemeralUDPPortRange(uint16(flags.WebRTCUDPStart), uint16(flags.WebRTCUDPEnd))
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if err != nil {
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return err
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}
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slog.Info("Using WebRTC UDP Port Range", "start", flags.WebRTCUDPStart, "end", flags.WebRTCUDPEnd)
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}
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settingEngine.SetIncludeLoopbackCandidate(true) // Just in case
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@@ -109,11 +106,6 @@ func InitWebRTCAPI() error {
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return nil
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}
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// GetWebRTCAPI returns the global WebRTC API
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func GetWebRTCAPI() *webrtc.API {
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return globalWebRTCAPI
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}
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// CreatePeerConnection sets up a new peer connection
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func CreatePeerConnection(onClose func()) (*webrtc.PeerConnection, error) {
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pc, err := globalWebRTCAPI.NewPeerConnection(globalWebRTCConfig)
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