Restructure projet and add webrtc settings

docs/ghostream.example.yml

@@ -21,7 +21,14 @@ auth:
 monitoring:
   listenAddress: 127.0.0.1:2112
 
-# Customize web server
+# Configure SRT server
+srt:
+  listenAddress: 127.0.0.1:9710
+
+  # Max number of active SRT connections
+  maxClients: 64
+
+# Configure web server
 web:
   listenAddress: 127.0.0.1:8080
   name: Demo
@@ -29,8 +36,8 @@ web:
   favicon: https://www.crans.org/images/favicon.ico
   widgetURL: https://example.com/
 
-srt:
+# Configure WebRTC server
-  listenAddress: 127.0.0.1:9710
+webrtc:
-
+  # UDP port range used to stream
-  # Max number of active SRT connections
+  minPortUDP: 10000
-  maxClients: 64
+  maxPortUDP: 10005

main.go

@@ -6,12 +6,11 @@ import (
 	"log"
 	"strings"
 
-	"github.com/pion/webrtc/v3"
 	"github.com/spf13/viper"
 	"gitlab.crans.org/nounous/ghostream/auth"
 	"gitlab.crans.org/nounous/ghostream/internal/monitoring"
-	"gitlab.crans.org/nounous/ghostream/stream"
 	"gitlab.crans.org/nounous/ghostream/stream/srt"
+	"gitlab.crans.org/nounous/ghostream/stream/webrtc"
 	"gitlab.crans.org/nounous/ghostream/web"
 )
 
@@ -49,11 +48,13 @@ func loadConfiguration() {
 	viper.SetDefault("Auth.LDAP.UserDn", "cn=users,dc=example,dc=com")
 	viper.SetDefault("Monitoring.ListenAddress", ":2112")
 	viper.SetDefault("Srt.ListenAddress", ":9710")
-	viper.SetDefault("Srt.MaxClients", "64")
+	viper.SetDefault("Srt.MaxClients", 64)
 	viper.SetDefault("Web.ListenAddress", ":8080")
 	viper.SetDefault("Web.Name", "Ghostream")
 	viper.SetDefault("Web.Hostname", "localhost")
 	viper.SetDefault("Web.Favicon", "/favicon.ico")
+	viper.SetDefault("WebRTC.MinPortUDP", 10000)
+	viper.SetDefault("WebRTC.MaxPortUDP", 10005)
 }
 
 func main() {
@@ -64,6 +65,7 @@ func main() {
 		Monitoring monitoring.Options
 		Srt        srt.Options
 		Web        web.Options
+		WebRTC     webrtc.Options
 	}{}
 	if err := viper.Unmarshal(&cfg); err != nil {
 		log.Fatalln("Failed to load settings", err)
@@ -81,10 +83,10 @@ func main() {
 	localSdpChan := make(chan webrtc.SessionDescription)
 
 	// Start stream, web and monitoring server
-	go srt.Serve(&cfg.Srt)
-	go stream.Serve(remoteSdpChan, localSdpChan)
-	go web.Serve(remoteSdpChan, localSdpChan, &cfg.Web)
 	go monitoring.Serve(&cfg.Monitoring)
+	go srt.Serve(&cfg.Srt)
+	go web.Serve(remoteSdpChan, localSdpChan, &cfg.Web)
+	go webrtc.Serve(remoteSdpChan, localSdpChan, &cfg.WebRTC)
 
 	// Wait for routines
 	select {}

stream/stream_test.go

@@ -1 +0,0 @@
-package stream

stream/webrtc/webrtc.go

@@ -1,4 +1,4 @@
-package stream
+package webrtc
 
 import (
 	"fmt"
@@ -14,6 +14,16 @@ import (
 	"github.com/pion/webrtc/v3/pkg/media/oggreader"
 )
 
+// Options holds web package configuration
+type Options struct {
+	MinPortUDP uint16
+	MaxPortUDP uint16
+}
+
+// SessionDescription contains SDP data
+// to initiate a WebRTC connection between one client and this app
+type SessionDescription = webrtc.SessionDescription
+
 const (
 	audioFileName = "output.ogg"
 	videoFileName = "output.ivf"
@@ -36,7 +46,7 @@ func removeTrack(tracks []*webrtc.Track, track *webrtc.Track) []*webrtc.Track {
 
 // newPeerHandler is called when server receive a new session description
 // this initiates a WebRTC connection and return server description
-func newPeerHandler(remoteSdp webrtc.SessionDescription) webrtc.SessionDescription {
+func newPeerHandler(remoteSdp webrtc.SessionDescription, cfg *Options) webrtc.SessionDescription {
 	// Create media engine using client SDP
 	mediaEngine := webrtc.MediaEngine{}
 	if err := mediaEngine.PopulateFromSDP(remoteSdp); err != nil {
@@ -46,7 +56,7 @@ func newPeerHandler(remoteSdp webrtc.SessionDescription) webrtc.SessionDescripti
 
 	// Create a new PeerConnection
 	settingsEngine := webrtc.SettingEngine{}
-	if err := settingsEngine.SetEphemeralUDPPortRange(10000, 10005); err != nil {
+	if err := settingsEngine.SetEphemeralUDPPortRange(cfg.MinPortUDP, cfg.MaxPortUDP); err != nil {
 		log.Println("Failed to set min/max UDP ports", err)
 		return webrtc.SessionDescription{}
 	}
@@ -227,7 +237,9 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
 }
 
 // Serve WebRTC media streaming server
-func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
+func Serve(remoteSdpChan, localSdpChan chan webrtc.SessionDescription, cfg *Options) {
+	log.Printf("WebRTC server using UDP from %d to %d", cfg.MinPortUDP, cfg.MaxPortUDP)
+
 	go playVideo()
 	go playAudio()
 
@@ -236,6 +248,6 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
 		// Wait for incoming session description
 		// then send the local description to browser
 		offer := <-remoteSdpChan
-		localSdpChan <- newPeerHandler(offer)
+		localSdpChan <- newPeerHandler(offer, cfg)
 	}
 }

stream/webrtc/webrtc_test.go

@@ -0,0 +1 @@
+package webrtc