Adds ability to set STUN/TURN, and WebRTC Min/Max Ports

README.md

@@ -61,15 +61,20 @@ $ docker run --name rtsp-to-web \
 ### Server settings
 
 ```text
-debug         - enable debug output
-log_level     - log level (trace, debug, info, warning, error, fatal, or panic)
-
-http_demo     - serve static files
-http_debug    - debug http api server
-http_login    - http auth login
-http_password - http auth password
-http_port     - http server port
-http_dir      - path to serve static files from
+debug           - enable debug output
+log_level       - log level (trace, debug, info, warning, error, fatal, or panic)
+
+http_demo       - serve static files
+http_debug      - debug http api server
+http_login      - http auth login
+http_password   - http auth password
+http_port       - http server port
+http_dir        - path to serve static files from
+ice_servers     - array of servers to use for STUN/TURN
+ice_username    - username to use for STUN/TURN
+ice_credential  - credential to use for STUN/TURN
+webrtc_port_min - minimum WebRTC port to use (UDP)
+webrtc_port_max - maximum WebRTC port to use (UDP)
 
 https
 https_auto_tls
@@ -78,24 +83,24 @@ https_cert
 https_key
 https_port
 
-rtsp_port     - rtsp server port
+rtsp_port       - rtsp server port
 ```
 
 ### Stream settings
 
 ```text
-name          - stream name
+name            - stream name
 ```
 
 ### Channel settings
 
 ```text
-name          - channel name
-url           - channel rtsp url
-on_demand     - stream mode static (run any time) or ondemand (run only has viewers)
-debug         - enable debug output (RTSP client)
-audio         - enable audio
-status        - default stream status
+name            - channel name
+url             - channel rtsp url
+on_demand       - stream mode static (run any time) or ondemand (run only has viewers)
+debug           - enable debug output (RTSP client)
+audio           - enable audio
+status          - default stream status
 ```
 
 #### Authorization play video
@@ -138,6 +143,7 @@ file.php need response json
     "http_login": "demo",
     "http_password": "demo",
     "http_port": ":8083",
+    "ice_servers": ["stun:stun.l.google.com:19302"],
     "rtsp_port": ":5541"
   },
   "streams": {

apiHTTPWebRTC.go

@@ -33,7 +33,7 @@ func HTTPAPIServerStreamWebRTC(c *gin.Context) {
 		}).Errorln(err.Error())
 		return
 	}
-	muxerWebRTC := webrtc.NewMuxer(webrtc.Options{})
+	muxerWebRTC := webrtc.NewMuxer(webrtc.Options{ICEServers: Storage.ServerICEServers(), ICEUsername: Storage.ServerICEUsername(), ICECredential: Storage.ServerICECredential(), PortMin: Storage.ServerWebRTCPortMin(), PortMax: Storage.ServerWebRTCPortMax()})
 	answer, err := muxerWebRTC.WriteHeader(codecs, c.PostForm("data"))
 	if err != nil {
 		c.IndentedJSON(400, Message{Status: 0, Payload: err.Error()})

config.json

@@ -13,6 +13,7 @@
     "https_cert": "server.crt",
     "https_key": "server.key",
     "https_port": ":443",
+    "ice_servers": ["stun:stun.l.google.com:19302"],
     "log_level": "debug",
     "rtsp_port": ":5541",
     "token": {

storageServer.go

@@ -112,6 +112,27 @@ func (obj *StorageST) ServerHTTPSKey() string {
 	return obj.Server.HTTPSKey
 }
 
+// ServerICEServers read ICE servers
+func (obj *StorageST) ServerICEServers() []string {
+	obj.mutex.Lock()
+	defer obj.mutex.Unlock()
+	return obj.Server.ICEServers
+}
+
+// ServerICEServers read ICE username
+func (obj *StorageST) ServerICEUsername() string {
+	obj.mutex.Lock()
+	defer obj.mutex.Unlock()
+	return obj.Server.ICEUsername
+}
+
+// ServerICEServers read ICE credential
+func (obj *StorageST) ServerICECredential() string {
+	obj.mutex.Lock()
+	defer obj.mutex.Unlock()
+	return obj.Server.ICECredential
+}
+
 //ServerTokenEnable read HTTPS Key options
 func (obj *StorageST) ServerTokenEnable() bool {
 	obj.mutex.RLock()
@@ -125,3 +146,17 @@ func (obj *StorageST) ServerTokenBackend() string {
 	defer obj.mutex.RUnlock()
 	return obj.Server.Token.Backend
 }
+
+// ServerWebRTCPortMin read WebRTC Port Min
+func (obj *StorageST) ServerWebRTCPortMin() uint16 {
+	obj.mutex.Lock()
+	defer obj.mutex.Unlock()
+	return obj.Server.WebRTCPortMin
+}
+
+// ServerWebRTCPortMax read WebRTC Port Max
+func (obj *StorageST) ServerWebRTCPortMax() uint16 {
+	obj.mutex.Lock()
+	defer obj.mutex.Unlock()
+	return obj.Server.WebRTCPortMax
+}

storageStruct.go

@@ -67,7 +67,12 @@ type ServerST struct {
 	HTTPSKey           string       `json:"https_key" groups:"api,config"`
 	HTTPSAutoTLSEnable bool         `json:"https_auto_tls" groups:"api,config"`
 	HTTPSAutoTLSName   string       `json:"https_auto_tls_name" groups:"api,config"`
+	ICEServers         []string     `json:"ice_servers" groups:"api,config"`
+	ICEUsername        string       `json:"ice_username" groups:"api,config"`
+	ICECredential      string       `json:"ice_credential" groups:"api,config"`
 	Token              Token        `json:"token,omitempty" groups:"api,config"`
+	WebRTCPortMin      uint16       `json:"webrtc_port_min" groups:"api,config"`
+	WebRTCPortMax      uint16       `json:"webrtc_port_max" groups:"api,config"`
 }
 
 //Token auth