support publishing, reading, proxying with SRT

README.md

@@ -20,7 +20,9 @@ Live streams can be published to the server with:
 
 |protocol|variants|video codecs|audio codecs|
 |--------|--------|------------|------------|
-|[WebRTC](#webrtc)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711|
+|[SRT clients](#srt-clients)||H265, H264|Opus, MPEG-4 Audio (AAC)|
+|[SRT servers](#srt-servers)||H265, H264|Opus, MPEG-4 Audio (AAC)|
+|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711|
 |[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
 |[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
 |[RTMP clients](#rtmp-clients)|RTMP, RTMPS, Enhanced RTMP|AV1, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
@@ -33,7 +35,8 @@ And can be read from the server with:
 
 |protocol|variants|video codecs|audio codecs|
 |--------|--------|------------|------------|
-|[WebRTC](#webrtc-1)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711|
+|[SRT](#srt)||H265, H264|Opus, MPEG-4 Audio (AAC)|
+|[WebRTC](#webrtc)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711|
 |[RTSP](#rtsp)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
 |[RTMP](#rtmp)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
 |[HLS](#hls)|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
@@ -76,7 +79,9 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
     * [Generic webcam](#generic-webcam)
     * [Raspberry Pi Cameras](#raspberry-pi-cameras)
   * [By protocol](#by-protocol)
-    * [WebRTC](#webrtc)
+    * [SRT clients](#srt-clients)
+    * [SRT servers](#srt-servers)
+    * [WebRTC clients](#webrtc-clients)
     * [RTSP clients](#rtsp-clients)
     * [RTSP cameras and servers](#rtsp-cameras-and-servers)
     * [RTMP clients](#rtmp-clients)
@@ -90,7 +95,8 @@ _rtsp-simple-server_ has been rebranded as _MediaMTX_. The reason is pretty obvi
     * [VLC](#vlc)
     * [Web browsers](#web-browsers-1)
   * [By protocol](#by-protocol-1)
-    * [WebRTC](#webrtc-1)
+    * [SRT](#srt)
+    * [WebRTC](#webrtc)
     * [RTSP](#rtsp)
     * [RTMP](#rtmp)
     * [HLS](#hls)
@@ -242,7 +248,7 @@ makepkg -si
 
 #### FFmpeg
 
-FFmpeg can publish a stream to the server in multiple ways (RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
+FFmpeg can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
 
 ```
 ffmpeg -re -stream_loop -1 -i file.ts -c copy -f rtsp rtsp://localhost:8554/mystream
@@ -258,7 +264,7 @@ The resulting stream will be available in path `/mystream`.
 
 #### GStreamer
 
-GStreamer can publish a stream to the server in multiple ways (RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
+GStreamer can publish a stream to the server in multiple ways (SRT client, SRT server, RTSP client, RTMP client, UDP/MPEG-TS, WebRTC with WHIP). The recommended one consists in publishing as a [RTSP client](#rtsp-clients):
 
 ```sh
 gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream \
@@ -285,7 +291,7 @@ The resulting stream will be available in path `/mystream`.
 
 #### OBS Studio
 
-OBS Studio can publish to the server as a [RTMP client](#rtmp-clients). In `Settings -> Stream` (or in the Auto-configuration Wizard), use the following parameters:
+OBS Studio can publish to the server in multiple ways (SRT client, RTMP client, WebRTC client). The recommended one consists in publishing as a [RTMP client](#rtmp-clients). In `Settings -> Stream` (or in the Auto-configuration Wizard), use the following parameters:
 
 * Service: `Custom...`
 * Server: `rtmp://localhost`
@@ -534,6 +540,14 @@ The resulting stream will be available in path `/cam_with_audio`.
 
 ### By protocol
 
+#### SRT clients
+
+TODO
+
+#### SRT servers
+
+TODO
+
 #### WebRTC
 
 WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can publish a stream with WebRTC and a web browser by visiting:
@@ -755,6 +769,10 @@ This web page can be embedded into another web page by using an iframe:
 
 ### By protocol
 
+#### SRT
+
+TODO
+
 #### WebRTC
 
 WebRTC is an API that makes use of a set of protocols and methods to connect two clients together and allow them to exchange real-time media or data streams. You can read a stream with WebRTC and a web browser by visiting:
@@ -1401,31 +1419,38 @@ The command will produce tarballs in folder `binaries/`.
 ## Standards
 
 * RTSP
+
   * [RTSP / RTP / RTCP standards](https://github.com/bluenviron/gortsplib#standards)
 
 * HLS
+
   * [HLS standards](https://github.com/bluenviron/gohlslib#standards)
 
 * RTMP
+
   * [RTMP](https://rtmp.veriskope.com/pdf/rtmp_specification_1.0.pdf)
   * [Enhanced RTMP](https://raw.githubusercontent.com/veovera/enhanced-rtmp/main/enhanced-rtmp-v1.pdf)
 
 * WebRTC
+
   * [WebRTC: Real-Time Communication in Browsers](https://www.w3.org/TR/webrtc/)
   * [WebRTC HTTP Ingestion Protocol (WHIP)](https://datatracker.ietf.org/doc/draft-ietf-wish-whip/)
   * [WebRTC HTTP Egress Protocol (WHEP)](https://datatracker.ietf.org/doc/draft-murillo-whep/)
 
 * Video and audio codecs
+
   * [Codec standards](https://github.com/bluenviron/mediacommon#standards)
 
 * Other
+
   * [Golang project layout](https://github.com/golang-standards/project-layout)
 
 ## Related projects
 
 * [gortsplib (RTSP library used internally)](https://github.com/bluenviron/gortsplib)
 * [gohlslib (HLS library used internally)](https://github.com/bluenviron/gohlslib)
 * [mediacommon (codecs and formats library used internally)](https://github.com/bluenviron/mediacommon)
+* [datarhei/gosrt (SRT library used internally)](https://github.com/datarhei/gosrt)
 * [pion/webrtc (WebRTC library used internally)](https://github.com/pion/webrtc)
 * [pion/sdp (SDP library used internally)](https://github.com/pion/sdp)
 * [pion/rtp (RTP library used internally)](https://github.com/pion/rtp)

internal/conf/conf.go

@@ -151,8 +151,8 @@ type Conf struct {
 	WebRTCICETCPMuxAddress  string            `json:"webrtcICETCPMuxAddress"`
 
 	// SRT
-	SRTDisable bool `json:"srtDisable"`
-	SRTAddress string `json:"srtAddress`
+	SRTDisable bool   `json:"srtDisable"`
+	SRTAddress string `json:"srtAddress"`
 
 	// paths
 	Paths map[string]*PathConf `json:"paths"`

internal/core/authentication.go

@@ -41,6 +41,7 @@ const (
 	authProtocolRTMP   authProtocol = "rtmp"
 	authProtocolHLS    authProtocol = "hls"
 	authProtocolWebRTC authProtocol = "webrtc"
+	authProtocolSRT    authProtocol = "srt"
 )
 
 func externalAuth(
@@ -95,7 +96,7 @@ func externalAuth(
 	return nil
 }
 
-type authCredentials struct {
+type authParameters struct {
 	query       string
 	ip          net.IP
 	user        string
@@ -113,28 +114,28 @@ func authenticate(
 	pathName string,
 	pathConf *conf.PathConf,
 	publish bool,
-	credentials authCredentials,
+	params authParameters,
 ) error {
 	var rtspAuth headers.Authorization
-	if credentials.rtspRequest != nil {
-		err := rtspAuth.Unmarshal(credentials.rtspRequest.Header["Authorization"])
+	if params.rtspRequest != nil {
+		err := rtspAuth.Unmarshal(params.rtspRequest.Header["Authorization"])
 		if err == nil && rtspAuth.Method == headers.AuthBasic {
-			credentials.user = rtspAuth.BasicUser
-			credentials.pass = rtspAuth.BasicPass
+			params.user = rtspAuth.BasicUser
+			params.pass = rtspAuth.BasicPass
 		}
 	}
 
 	if externalAuthenticationURL != "" {
 		err := externalAuth(
 			externalAuthenticationURL,
-			credentials.ip.String(),
-			credentials.user,
-			credentials.pass,
+			params.ip.String(),
+			params.user,
+			params.pass,
 			pathName,
-			credentials.proto,
-			credentials.id,
+			params.proto,
+			params.id,
 			publish,
-			credentials.query,
+			params.query,
 		)
 		if err != nil {
 			return fmt.Errorf("external authentication failed: %s", err)
@@ -156,26 +157,26 @@ func authenticate(
 	}
 
 	if pathIPs != nil {
-		if !ipEqualOrInRange(credentials.ip, pathIPs) {
-			return fmt.Errorf("IP '%s' not allowed", credentials.ip)
+		if !ipEqualOrInRange(params.ip, pathIPs) {
+			return fmt.Errorf("IP '%s' not allowed", params.ip)
 		}
 	}
 
 	if pathUser != "" {
-		if credentials.rtspRequest != nil && rtspAuth.Method == headers.AuthDigest {
+		if params.rtspRequest != nil && rtspAuth.Method == headers.AuthDigest {
 			err := auth.Validate(
-				credentials.rtspRequest,
+				params.rtspRequest,
 				pathUser,
 				pathPass,
-				credentials.rtspBaseURL,
+				params.rtspBaseURL,
 				rtspAuthMethods,
 				"IPCAM",
-				credentials.rtspNonce)
+				params.rtspNonce)
 			if err != nil {
 				return err
 			}
-		} else if !checkCredential(pathUser, credentials.user) ||
-			!checkCredential(pathPass, credentials.pass) {
+		} else if !checkCredential(pathUser, params.user) ||
+			!checkCredential(pathPass, params.pass) {
 			return fmt.Errorf("invalid credentials")
 		}
 	}

internal/core/core.go

@@ -45,7 +45,7 @@ type Core struct {
 	rtmpsServer     *rtmpServer
 	hlsManager      *hlsManager
 	webRTCManager   *webRTCManager
-	srtServer *srtServer
+	srtServer       *srtServer
 	api             *api
 	confWatcher     *confwatcher.ConfWatcher
 
@@ -437,6 +437,9 @@ func (p *Core) createResources(initial bool) error {
 	if !p.conf.SRTDisable {
 		p.srtServer, err = newSRTServer(
 			p.conf.SRTAddress,
+			p.conf.ReadTimeout,
+			p.pathManager,
+			p,
 		)
 		if err != nil {
 			return err

internal/core/hls_http_server.go

@@ -145,7 +145,7 @@ func (s *hlsHTTPServer) onRequest(ctx *gin.Context) {
 	res := s.pathManager.getPathConf(pathGetPathConfReq{
 		name:    dir,
 		publish: false,
-		credentials: authCredentials{
+		authParameters: authParameters{
 			query: ctx.Request.URL.RawQuery,
 			ip:    net.ParseIP(ctx.ClientIP()),
 			user:  user,