⭐ feat(maitred): Update maitred - hookup to the API (#198)

## Description We are attempting to hookup maitred to the API Maitred duties will be: - [ ] Hookup to the API - [ ] Wait for signal (from the API) to start Steam - [ ] Stop signal to stop the gaming session, clean up Steam... and maybe do the backup ## Summary by CodeRabbit - **New Features** - Introduced Docker-based deployment configurations for both the main and relay applications. - Added new API endpoints enabling real-time machine messaging and enhanced IoT operations. - Expanded database schema and actor types to support improved machine tracking. - **Improvements** - Enhanced real-time communication and relay management with streamlined room handling. - Upgraded dependencies, logging, and error handling for greater stability and performance.  --------- Co-authored-by: DatCaptainHorse <DatCaptainHorse@users.noreply.github.com> Co-authored-by: Kristian Ollikainen <14197772+DatCaptainHorse@users.noreply.github.com>
2025-12-12 08:45:38 +02:00 · 2025-04-07 23:23:53 +03:00
parent 6990494b34
commit de80f3e6ab
84 changed files with 7357 additions and 1331 deletions
--- a/packages/relay/internal/common/common.go
+++ b/packages/relay/internal/common/common.go
@@ -0,0 +1,139 @@
+package common
+
+import (
+	"fmt"
+	"github.com/libp2p/go-reuseport"
+	"github.com/pion/ice/v4"
+	"github.com/pion/interceptor"
+	"github.com/pion/webrtc/v4"
+	"log/slog"
+	"strconv"
+)
+
+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 additional header extensions to reduce latency
+	// Playout Delay
+	if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
+		URI: ExtensionPlayoutDelay,
+	}, webrtc.RTPCodecTypeVideo); err != nil {
+		return err
+	}
+	if err := mediaEngine.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{
+		URI: ExtensionPlayoutDelay,
+	}, webrtc.RTPCodecTypeAudio); err != nil {
+		return 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)
+
+	nat11IPs := GetFlags().NAT11IPs
+	if len(nat11IPs) > 0 {
+		settingEngine.SetNAT1To1IPs(nat11IPs, webrtc.ICECandidateTypeHost)
+	}
+
+	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 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)
+	}
+
+	// Set the UDP port range used by WebRTC
+	err = settingEngine.SetEphemeralUDPPortRange(uint16(flags.WebRTCUDPStart), uint16(flags.WebRTCUDPEnd))
+	if err != nil {
+		return err
+	}
+
+	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
+}
+
+// GetWebRTCAPI returns the global WebRTC API
+func GetWebRTCAPI() *webrtc.API {
+	return globalWebRTCAPI
+}
+
+// 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
+}