support publishing, reading, proxying with SRT

add rtmp.Writer, rtmp.Reader

update readme

update dependencies

fix a/v sync

add authentication

add missing line

return an error in case the random number generator fails

README.md

@@ -20,9 +20,11 @@ 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|
-|[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|
+|[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), G726, 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), G726, 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)|
 |[RTMP cameras and servers](#rtmp-cameras-and-servers)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
 |[HLS cameras and servers](#hls-cameras-and-servers)|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
@@ -33,8 +35,9 @@ 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|
-|[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|
+|[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), G726, 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)
@@ -243,7 +249,7 @@ 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
@@ -259,7 +265,7 @@ 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 \
@@ -286,7 +292,7 @@ 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`
@@ -535,6 +541,14 @@ 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:
@@ -697,7 +711,7 @@ gst-launch-1.0 rtspsrc tls-validation-flags=0 location=rtsps://ip:8322/...
 
 #### VLC
 
-VLC can read a stream from the server in multiple ways (RTSP, RTMP, HLS). The recommended one consists in reading with [RTSP](#rtsp):
+VLC can read a stream from the server in multiple ways (RTSP, RTMP, HLS, SRT). The recommended one consists in reading with [RTSP](#rtsp):
 
 ```sh
 vlc --network-caching=50 rtsp://localhost:8554/mystream
@@ -764,6 +778,10 @@ 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:
@@ -1409,31 +1427,38 @@ 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)

apidocs/openapi.yaml

@@ -356,14 +356,15 @@ components:
           enum:
           - hlsMuxer
           - hlsSource
+          - redirect
           - rpiCameraSource
           - rtmpConn
           - rtmpSource
           - rtmpsSession
           - rtspSession
           - rtspSource
           - rtspsSession
-          - redirect
+          - srtSource
           - udpSource
           - webRTCSession
         id:

go.mod

@@ -6,10 +6,11 @@ require (
 	code.cloudfoundry.org/bytefmt v0.0.0
 	github.com/abema/go-mp4 v0.11.0
 	github.com/alecthomas/kong v0.8.0
-	github.com/asticode/go-astits v1.11.0
-	github.com/bluenviron/gohlslib v0.3.0
-	github.com/bluenviron/gortsplib/v3 v3.9.0
-	github.com/bluenviron/mediacommon v0.7.0
+	github.com/asticode/go-astits v1.11.1-0.20230727094110-0df190a2dd87
+	github.com/bluenviron/gohlslib v0.3.1-0.20230730162911-eb9f86511072
+	github.com/bluenviron/gortsplib/v3 v3.9.1-0.20230730204034-8b8b52e689d9
+	github.com/bluenviron/mediacommon v0.7.1-0.20230730144331-10b74a4f6eda
+	github.com/datarhei/gosrt v0.5.2
 	github.com/fsnotify/fsnotify v1.6.0
 	github.com/gin-gonic/gin v1.9.1
 	github.com/google/uuid v1.3.0
@@ -33,6 +34,7 @@ require (
 require (
 	github.com/aler9/writerseeker v0.0.0-20220601075008-6f0e685b9c82 // indirect
 	github.com/asticode/go-astikit v0.30.0 // indirect
+	github.com/benburkert/openpgp v0.0.0-20160410205803-c2471f86866c // indirect
 	github.com/bytedance/sonic v1.9.1 // indirect
 	github.com/chenzhuoyu/base64x v0.0.0-20221115062448-fe3a3abad311 // indirect
 	github.com/davecgh/go-spew v1.1.1 // indirect

go.sum

@@ -8,14 +8,16 @@ github.com/aler9/writerseeker v0.0.0-20220601075008-6f0e685b9c82 h1:9WgSzBLo3a9T
 github.com/aler9/writerseeker v0.0.0-20220601075008-6f0e685b9c82/go.mod h1:qsMrZCbeBf/mCLOeF16KDkPu4gktn/pOWyaq1aYQE7U=
 github.com/asticode/go-astikit v0.30.0 h1:DkBkRQRIxYcknlaU7W7ksNfn4gMFsB0tqMJflxkRsZA=
 github.com/asticode/go-astikit v0.30.0/go.mod h1:h4ly7idim1tNhaVkdVBeXQZEE3L0xblP7fCWbgwipF0=
-github.com/asticode/go-astits v1.11.0 h1:GTHUXht0ZXAJXsVbsLIcyfHr1Bchi4QQwMARw2ZWAng=
-github.com/asticode/go-astits v1.11.0/go.mod h1:QSHmknZ51pf6KJdHKZHJTLlMegIrhega3LPWz3ND/iI=
-github.com/bluenviron/gohlslib v0.3.0 h1:ze8cCKszGC2LAWp0B+qIXZIlCZocB7a3BKeBo9E8Sr0=
-github.com/bluenviron/gohlslib v0.3.0/go.mod h1:aO69Vu0mMUxWrLmgS6g/S3Y3sfAhyg2SXaMEL7yNlWc=
-github.com/bluenviron/gortsplib/v3 v3.9.0 h1:aAHV6MhsDtgBF6yKaNBBCdvtSpLB8ne4kyUfLQlN7nM=
-github.com/bluenviron/gortsplib/v3 v3.9.0/go.mod h1:5h3Zu7jkzwDknYrf+89q2saab//oioKgM9mgvBEX3pg=
-github.com/bluenviron/mediacommon v0.7.0 h1:dJWLLL9oDbAqfK8KuNfnDUQwNbeMAtGeRjZc9Vo95js=
-github.com/bluenviron/mediacommon v0.7.0/go.mod h1:wuLJdxcITiSPgY1MvQqrX+qPlKmNfeV9wNvXth5M98I=
+github.com/asticode/go-astits v1.11.1-0.20230727094110-0df190a2dd87 h1:SCAqalLhgKGDghGz03yYVWr8TavHluP/i7IwshKU9yA=
+github.com/asticode/go-astits v1.11.1-0.20230727094110-0df190a2dd87/go.mod h1:QSHmknZ51pf6KJdHKZHJTLlMegIrhega3LPWz3ND/iI=
+github.com/benburkert/openpgp v0.0.0-20160410205803-c2471f86866c h1:8XZeJrs4+ZYhJeJ2aZxADI2tGADS15AzIF8MQ8XAhT4=
+github.com/benburkert/openpgp v0.0.0-20160410205803-c2471f86866c/go.mod h1:x1vxHcL/9AVzuk5HOloOEPrtJY0MaalYr78afXZ+pWI=
+github.com/bluenviron/gohlslib v0.3.1-0.20230730162911-eb9f86511072 h1:pAbC7frXsTMxP7Ck3E50hl7oFeSeD2dgc2lWjmHXztQ=
+github.com/bluenviron/gohlslib v0.3.1-0.20230730162911-eb9f86511072/go.mod h1:rK4b161qErs82QqvBEl84vpi2xhdZBUT0yubXuytZ7E=
+github.com/bluenviron/gortsplib/v3 v3.9.1-0.20230730204034-8b8b52e689d9 h1:QBdUlT/taEG0b8dxguJ6GYT7r6vJFRhvwlhs1LGWYlQ=
+github.com/bluenviron/gortsplib/v3 v3.9.1-0.20230730204034-8b8b52e689d9/go.mod h1:vBmCYjtox2pXWSvU+r9ROpEbwSOhh5lVWd+t9Sj8glc=
+github.com/bluenviron/mediacommon v0.7.1-0.20230730144331-10b74a4f6eda h1:+ungCWRNDjsy/CVL1l/UjAj4vYL4+NIJQoJJWbR3Xw8=
+github.com/bluenviron/mediacommon v0.7.1-0.20230730144331-10b74a4f6eda/go.mod h1:tfk0qGPhqnOxVCrElu8ct3LKQn6Cj4Tpu3zbbJBTKj4=
 github.com/bytedance/sonic v1.5.0/go.mod h1:ED5hyg4y6t3/9Ku1R6dU/4KyJ48DZ4jPhfY1O2AihPM=
 github.com/bytedance/sonic v1.9.1 h1:6iJ6NqdoxCDr6mbY8h18oSO+cShGSMRGCEo7F2h0x8s=
 github.com/bytedance/sonic v1.9.1/go.mod h1:i736AoUSYt75HyZLoJW9ERYxcy6eaN6h4BZXU064P/U=
@@ -25,6 +27,8 @@ github.com/chenzhuoyu/base64x v0.0.0-20221115062448-fe3a3abad311/go.mod h1:b583j
 github.com/cloudfoundry/bytefmt v0.0.0-20211005130812-5bb3c17173e5 h1:xB7KkA98BcUdzVcwyZxb5R0FGIHxNPHgZOzkjPEY5gM=
 github.com/cloudfoundry/bytefmt v0.0.0-20211005130812-5bb3c17173e5/go.mod h1:v4VVB6oBMz/c9fRY6vZrwr5xKRWOH5NPDjQZlPk0Gbs=
 github.com/creack/pty v1.1.7/go.mod h1:lj5s0c3V2DBrqTV7llrYr5NG6My20zk30Fl46Y7DoTY=
+github.com/datarhei/gosrt v0.5.2 h1:eagqZwEIiGPNJW0rLep3gwceObyaZ17+iKRc+l4VEpc=
+github.com/datarhei/gosrt v0.5.2/go.mod h1:0308GQhAu5hxe2KYdbss901aKceSSKXnwCr8Vs++eiw=
 github.com/davecgh/go-spew v1.1.0/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
 github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
 github.com/davecgh/go-spew v1.1.1/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=

internal/conf/conf.go

@@ -159,6 +159,10 @@ type Conf struct {
 	WebRTCICEUDPMuxAddress  string            `json:"webrtcICEUDPMuxAddress"`
 	WebRTCICETCPMuxAddress  string            `json:"webrtcICETCPMuxAddress"`
 
+	// SRT
+	SRT        bool   `json:"srt"`
+	SRTAddress string `json:"srtAddress"`
+
 	// paths
 	Paths map[string]*PathConf `json:"paths"`
 }
@@ -336,6 +340,10 @@ func (conf *Conf) UnmarshalJSON(b []byte) error {
 	conf.WebRTCAllowOrigin = "*"
 	conf.WebRTCICEServers2 = []WebRTCICEServer{{URL: "stun:stun.l.google.com:19302"}}
 
+	// SRT
+	conf.SRT = true
+	conf.SRTAddress = ":8890"
+
 	type alias Conf
 	d := json.NewDecoder(bytes.NewReader(b))
 	d.DisallowUnknownFields()

internal/conf/path.go

@@ -210,6 +210,16 @@ func (pconf *PathConf) check(conf *Conf, name string) error {
 			return fmt.Errorf("'%s' is not a valid IP", host)
 		}
 
+	case strings.HasPrefix(pconf.Source, "srt://"):
+		if pconf.Regexp != nil {
+			return fmt.Errorf("a path with a regular expression (or path 'all') cannot have a SRT source. use another path")
+		}
+
+		_, err := gourl.Parse(pconf.Source)
+		if err != nil {
+			return fmt.Errorf("'%s' is not a valid HLS URL", pconf.Source)
+		}
+
 	case pconf.Source == "redirect":
 		if pconf.SourceRedirect == "" {
 			return fmt.Errorf("source redirect must be filled")
@@ -337,6 +347,7 @@ func (pconf PathConf) HasStaticSource() bool {
 		strings.HasPrefix(pconf.Source, "http://") ||
 		strings.HasPrefix(pconf.Source, "https://") ||
 		strings.HasPrefix(pconf.Source, "udp://") ||
+		strings.HasPrefix(pconf.Source, "srt://") ||
 		pconf.Source == "rpiCamera"
 }
 

internal/core/api_test.go

@@ -578,7 +578,7 @@ func TestAPIProtocolList(t *testing.T) {
 				err = conn.InitializeClient(u, true)
 				require.NoError(t, err)
 
-				err = conn.WriteTracks(testFormatH264, nil)
+				_, err = rtmp.NewWriter(conn, testFormatH264, nil)
 				require.NoError(t, err)
 
 				time.Sleep(500 * time.Millisecond)
@@ -833,7 +833,7 @@ func TestAPIProtocolGet(t *testing.T) {
 				err = conn.InitializeClient(u, true)
 				require.NoError(t, err)
 
-				err = conn.WriteTracks(testFormatH264, nil)
+				_, err = rtmp.NewWriter(conn, testFormatH264, nil)
 				require.NoError(t, err)
 
 				time.Sleep(500 * time.Millisecond)
@@ -1155,7 +1155,7 @@ func TestAPIProtocolKick(t *testing.T) {
 				err = conn.InitializeClient(u, true)
 				require.NoError(t, err)
 
-				err = conn.WriteTracks(testFormatH264, nil)
+				_, err = rtmp.NewWriter(conn, testFormatH264, nil)
 				require.NoError(t, err)
 
 			case "webrtc":

internal/core/authentication.go

@@ -50,6 +50,7 @@ const (
 	authProtocolRTMP   authProtocol = "rtmp"
 	authProtocolHLS    authProtocol = "hls"
 	authProtocolWebRTC authProtocol = "webrtc"
+	authProtocolSRT    authProtocol = "srt"
 )
 
 type authCredentials struct {

internal/core/core.go

@@ -44,6 +44,7 @@ type Core struct {
 	rtmpsServer     *rtmpServer
 	hlsManager      *hlsManager
 	webRTCManager   *webRTCManager
+	srtServer       *srtServer
 	api             *api
 	confWatcher     *confwatcher.ConfWatcher
 
@@ -432,6 +433,21 @@ func (p *Core) createResources(initial bool) error {
 		}
 	}
 
+	if p.conf.SRT {
+		p.srtServer, err = newSRTServer(
+			p.conf.SRTAddress,
+			p.conf.ReadTimeout,
+			p.conf.WriteTimeout,
+			p.conf.ReadBufferCount,
+			p.conf.UDPMaxPayloadSize,
+			p.pathManager,
+			p,
+		)
+		if err != nil {
+			return err
+		}
+	}
+
 	if p.conf.API {
 		if p.api == nil {
 			p.api, err = newAPI(
@@ -595,6 +611,14 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		newConf.WebRTCICEUDPMuxAddress != p.conf.WebRTCICEUDPMuxAddress ||
 		newConf.WebRTCICETCPMuxAddress != p.conf.WebRTCICETCPMuxAddress
 
+	closeSRTServer := newConf == nil ||
+		newConf.SRTAddress != p.conf.SRTAddress ||
+		newConf.ReadTimeout != p.conf.ReadTimeout ||
+		newConf.WriteTimeout != p.conf.WriteTimeout ||
+		newConf.ReadBufferCount != p.conf.ReadBufferCount ||
+		newConf.UDPMaxPayloadSize != p.conf.UDPMaxPayloadSize ||
+		closePathManager
+
 	closeAPI := newConf == nil ||
 		newConf.API != p.conf.API ||
 		newConf.APIAddress != p.conf.APIAddress ||
@@ -620,6 +644,11 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		}
 	}
 
+	if closeSRTServer && p.srtServer != nil {
+		p.srtServer.close()
+		p.srtServer = nil
+	}
+
 	if closeWebRTCManager && p.webRTCManager != nil {
 		p.webRTCManager.close()
 		p.webRTCManager = nil