refactor: rtsp和gb28181不单独声明媒体端口, 强制udp和tcp全开

api_gb.go

@@ -72,18 +72,6 @@ func (api *ApiServer) OnGBSourceCreate(v *SourceSDP, w http.ResponseWriter, r *h
 		//udp收流
 	}
 
-	if tcp && active {
-		if !tcp {
-			err = fmt.Errorf("UDP不能主动拉流")
-		} else if !stream.AppConfig.GB28181.IsEnableTCP() {
-			err = fmt.Errorf("未开启TCP收流服务,UDP不能主动拉流")
-		}
-
-		if err != nil {
-			return
-		}
-	}
-
 	var ssrc string
 	if v.SessionName == InviteTypeDownload || v.SessionName == InviteTypePlayback {
 		ssrc = gb28181.GetVodSSRC()

config.json

@@ -7,6 +7,7 @@
   "idle_timeout": 60,
   "receive_timeout":60,
   "debug": false,
+  "media_port": [50000,60000],
 
   "http": {
     "port": 8080
@@ -26,9 +27,8 @@
 
   "rtsp": {
     "enable": true,
-    "port": [554,20000,30000],
-    "password": "123456",
-    "transport": "UDP|TCP"
+    "port": 554,
+    "password": "123456"
   },
 
   "webrtc": {
@@ -38,9 +38,7 @@
   },
 
   "gb28181": {
-    "enable": true,
-    "port": [50000,60000],
-    "transport": "UDP|TCP"
+    "enable": true
   },
 
   "jt1078": {

gb28181/source.go

@@ -276,16 +276,6 @@ func (source *BaseGBSource) SetTransport(transport transport.Transport) {
 
 // NewGBSource 创建国标推流源, 返回监听的收流端口
 func NewGBSource(id string, ssrc uint32, tcp bool, active bool) (GBSource, int, error) {
-	if tcp {
-		utils.Assert(stream.AppConfig.GB28181.IsEnableTCP())
-	} else {
-		utils.Assert(stream.AppConfig.GB28181.IsEnableUDP())
-	}
-
-	if active {
-		utils.Assert(tcp && stream.AppConfig.GB28181.IsEnableTCP())
-	}
-
 	var transportServer transport.Transport
 	var source GBSource
 	var port int

main.go

@@ -75,12 +75,10 @@ func init() {
 	// 初始化日志
 	log.InitLogger(config.Log.FileLogging, zapcore.Level(stream.AppConfig.Log.Level), stream.AppConfig.Log.Name, stream.AppConfig.Log.MaxSize, stream.AppConfig.Log.MaxBackup, stream.AppConfig.Log.MaxAge, stream.AppConfig.Log.Compress)
 
-	if stream.AppConfig.GB28181.Enable {
-		gb28181.TransportManger = transport.NewTransportManager(config.ListenIP, uint16(stream.AppConfig.GB28181.Port[0]), uint16(stream.AppConfig.GB28181.Port[1]))
-	}
-
-	if stream.AppConfig.Rtsp.Enable {
-		rtsp.TransportManger = transport.NewTransportManager(config.ListenIP, uint16(stream.AppConfig.Rtsp.Port[1]), uint16(stream.AppConfig.Rtsp.Port[2]))
+	if stream.AppConfig.GB28181.Enable || stream.AppConfig.Rtsp.Enable {
+		transportManager := transport.NewTransportManager(config.ListenIP, uint16(stream.AppConfig.MediaPort[0]), uint16(stream.AppConfig.MediaPort[1]))
+		gb28181.TransportManger = transportManager
+		rtsp.TransportManger = transportManager
 	}
 
 	// 创建dump目录
@@ -112,7 +110,7 @@ func main() {
 	}
 
 	if stream.AppConfig.Rtsp.Enable {
-		rtspAddr, err := net.ResolveTCPAddr("tcp", stream.ListenAddr(stream.AppConfig.Rtsp.Port[0]))
+		rtspAddr, err := net.ResolveTCPAddr("tcp", stream.ListenAddr(stream.AppConfig.Rtsp.Port))
 		if err != nil {
 			panic(rtspAddr)
 		}

stream/config.go

@@ -77,10 +77,8 @@ type JT1078Config struct {
 }
 
 type RtspConfig struct {
-	TransportConfig
-
 	enableConfig
-	Port     []int  `json:"port"`
+	Port     int    `json:"port"`
 	Password string `json:"password"`
 }
 
@@ -106,8 +104,6 @@ type HttpConfig struct {
 
 type GB28181Config struct {
 	enableConfig
-	TransportConfig
-	Port []int `json:"port"`
 }
 
 type WebRtcConfig struct {
@@ -207,7 +203,7 @@ func GetStreamPlayUrls(source string) []string {
 
 	if AppConfig.Rtsp.Enable {
 		// 不拼接userinfo
-		urls = append(urls, fmt.Sprintf("rtsp://%s:%d/%s", AppConfig.PublicIP, AppConfig.Rtsp.Port[0], source))
+		urls = append(urls, fmt.Sprintf("rtsp://%s:%d/%s", AppConfig.PublicIP, AppConfig.Rtsp.Port, source))
 	}
 
 	//if AppConfig.Http.Enable {
@@ -265,6 +261,7 @@ type AppConfig_ struct {
 	IdleTimeout    int64  `json:"idle_timeout"`    // 多长时间(单位秒)没有拉流. 如果开启hook通知, 根据hook响应, 决定是否关闭Source(200-不关闭/非200关闭). 否则会直接关闭Source.
 	ReceiveTimeout int64  `json:"receive_timeout"` // 多长时间(单位秒)没有收到流. 如果开启hook通知, 根据hook响应, 决定是否关闭Source(200-不关闭/非200关闭). 否则会直接关闭Source.
 	Debug          bool   `json:"debug"`           // debug模式, 开启将保存推流
+	MediaPort      []int  `json:"media_port"`      // 媒体端口范围
 
 	//缓存指定时长的包，满了之后才发送给Sink. 可以降低用户态和内核态的交互频率，大幅提升性能.
 	//合并写的大小范围，应当大于一帧的时长，不超过一组GOP的时长，在实际发送流的时候也会遵循此条例.