Files
netris-nestri/packages/relay/internal/common/common.go
Kristian Ollikainen 6e82eff9e2 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>
2025-06-06 16:48:49 +03:00

132 lines
3.9 KiB
Go

package common
import (
"fmt"
"log/slog"
"strconv"
"github.com/libp2p/go-reuseport"
"github.com/pion/ice/v4"
"github.com/pion/interceptor"
"github.com/pion/webrtc/v4"
)
var globalWebRTCAPI *webrtc.API
var globalWebRTCConfig = webrtc.Configuration{
ICETransportPolicy: webrtc.ICETransportPolicyAll,
BundlePolicy: webrtc.BundlePolicyBalanced,
SDPSemantics: webrtc.SDPSemanticsUnifiedPlan,
}
func InitWebRTCAPI() error {
var err error
flags := GetFlags()
// Media engine
mediaEngine := &webrtc.MediaEngine{}
// Register our extensions
if err := RegisterExtensions(mediaEngine); err != nil {
return fmt.Errorf("failed to register extensions: %w", err)
}
// Default codecs cover most of our needs
err = mediaEngine.RegisterDefaultCodecs()
if err != nil {
return err
}
// Add H.265 for special cases
videoRTCPFeedback := []webrtc.RTCPFeedback{{"goog-remb", ""}, {"ccm", "fir"}, {"nack", ""}, {"nack", "pli"}}
for _, codec := range []webrtc.RTPCodecParameters{
{
RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH265, ClockRate: 90000, RTCPFeedback: videoRTCPFeedback},
PayloadType: 48,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeRTX, ClockRate: 90000, SDPFmtpLine: "apt=48"},
PayloadType: 49,
},
} {
if err = mediaEngine.RegisterCodec(codec, webrtc.RTPCodecTypeVideo); err != nil {
return err
}
}
// Interceptor registry
interceptorRegistry := &interceptor.Registry{}
// Use default set
err = webrtc.RegisterDefaultInterceptors(mediaEngine, interceptorRegistry)
if err != nil {
return err
}
// Setting engine
settingEngine := webrtc.SettingEngine{}
// New in v4, reduces CPU usage and latency when enabled
settingEngine.EnableSCTPZeroChecksum(true)
nat11IP := GetFlags().NAT11IP
if len(nat11IP) > 0 {
settingEngine.SetNAT1To1IPs([]string{nat11IP}, webrtc.ICECandidateTypeSrflx)
slog.Info("Using NAT 1:1 IP for WebRTC", "nat11_ip", nat11IP)
}
muxPort := GetFlags().UDPMuxPort
if muxPort > 0 {
// Use reuseport to allow multiple listeners on the same port
pktListener, err := reuseport.ListenPacket("udp", ":"+strconv.Itoa(muxPort))
if err != nil {
return fmt.Errorf("failed to create WebRTC muxed UDP listener: %w", err)
}
mux := ice.NewMultiUDPMuxDefault(ice.NewUDPMuxDefault(ice.UDPMuxParams{
UDPConn: pktListener,
}))
slog.Info("Using UDP Mux for WebRTC", "port", muxPort)
settingEngine.SetICEUDPMux(mux)
}
if flags.WebRTCUDPStart > 0 && flags.WebRTCUDPEnd > 0 && flags.WebRTCUDPStart < flags.WebRTCUDPEnd {
// Set the UDP port range used by WebRTC
err = settingEngine.SetEphemeralUDPPortRange(uint16(flags.WebRTCUDPStart), uint16(flags.WebRTCUDPEnd))
if err != nil {
return err
}
slog.Info("Using WebRTC UDP Port Range", "start", flags.WebRTCUDPStart, "end", flags.WebRTCUDPEnd)
}
settingEngine.SetIncludeLoopbackCandidate(true) // Just in case
// Create a new API object with our customized settings
globalWebRTCAPI = webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine), webrtc.WithSettingEngine(settingEngine), webrtc.WithInterceptorRegistry(interceptorRegistry))
return nil
}
// CreatePeerConnection sets up a new peer connection
func CreatePeerConnection(onClose func()) (*webrtc.PeerConnection, error) {
pc, err := globalWebRTCAPI.NewPeerConnection(globalWebRTCConfig)
if err != nil {
return nil, err
}
// Log connection state changes and handle failed/disconnected connections
pc.OnConnectionStateChange(func(connectionState webrtc.PeerConnectionState) {
// Close PeerConnection in cases
if connectionState == webrtc.PeerConnectionStateFailed ||
connectionState == webrtc.PeerConnectionStateDisconnected ||
connectionState == webrtc.PeerConnectionStateClosed {
err = pc.Close()
if err != nil {
slog.Error("Failed to close PeerConnection", "err", err)
}
onClose()
}
})
return pc, nil
}