怎加初始拉流功能

版本更新
解决rtsp推流无视频时的报错
2025-09-27 12:02:20 +08:00 · 2020-10-01 20:12:26 +08:00 · 2020-09-23 23:04:07 +08:00 · 2020-09-23 22:55:52 +08:00 · 2020-09-20 15:54:27 +08:00 · 2020-08-29 08:00:25 +08:00
9 changed files with 292 additions and 430 deletions
--- a/README.md
+++ b/README.md
@@ -13,12 +13,15 @@ ListenAddr  = ":554"
 BufferLength  = 2048
 AutoPull     = false
 RemoteAddr   = "rtsp://localhost/${streamPath}"
+[[RTSP.AutoPullList]]
+URL = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=1&subtype=1"
+StreamPath = "live/rtsp"
 ```
 - ListenAddr 是监听端口，可以将rtsp流推到Monibuca中
 - BufferLength是指解析拉取的rtp包的缓冲大小
 - AutoPull是指当有用户订阅一个新流的时候自动向远程拉流转发
 - RemoteAddr 指远程拉流地址，其中${streamPath}是占位符，实际使用流路径替换。
-
+- AutoPullList 是一个数组，如果配置了该数组，则会在程序启动时自动启动拉流，StreamPath一定要是唯一的，不能重复

 ## 使用方法(拉流转发)
 ```go
--- a/client.go
+++ b/client.go
@@ -4,6 +4,7 @@ import (
 	"bufio"
 	"bytes"
 	"crypto/md5"
+	"encoding/base64"
 	"encoding/binary"
 	"errors"
 	"fmt"
@@ -16,11 +17,12 @@ import (
 	"time"

 	. "github.com/Monibuca/engine/v2"
+	. "github.com/Monibuca/plugin-rtp"
 )

 // PullStream 从外部拉流
 func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (err error) {
-	if result := rtsp.Publisher.Publish(streamPath); result {
+	if result := rtsp.Publish(streamPath); result {
 		rtsp.Stream.Type = "RTSP"
 		rtsp.RTSPInfo.StreamInfo = &rtsp.Stream.StreamInfo
 		rtsp.TransType = TRANS_TYPE_TCP
@@ -29,6 +31,7 @@ func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (err error) {
 		rtsp.aRTPChannel = 2
 		rtsp.aRTPControlChannel = 3
 		rtsp.URL = rtspUrl
+		rtsp.UDPServer = &UDPServer{Session: rtsp}
 		if err = rtsp.requestStream(); err != nil {
 			Println(err)
 			rtsp.Close()
@@ -77,6 +80,18 @@ func DigestAuth(authLine string, method string, URL string) (string, error) {
 	Authorization := fmt.Sprintf("Digest username=\"%s\", realm=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\"", username, realm, nonce, l.String(), response)
 	return Authorization, nil
 }
+// auth Basic验证
+func BasicAuth(authLine string, method string, URL string) (string, error) {
+	l, err := url.Parse(URL)
+	if err != nil {
+		return "", fmt.Errorf("Url parse error:%v,%v", URL, err)
+	}
+	username := l.User.Username()
+	password, _ := l.User.Password()
+	userAndpass := []byte(username + ":" + password)
+	Authorization := fmt.Sprintf("Basic %s", base64.StdEncoding.EncodeToString(userAndpass))
+	return Authorization, nil
+}
 func (client *RTSP) checkAuth(method string, resp *Response) (string, error) {
 	if resp.StatusCode == 401 {
 		// need auth.
@@ -90,8 +105,7 @@ func (client *RTSP) checkAuth(method string, resp *Response) (string, error) {
 					client.authLine = authLine
 					return DigestAuth(authLine, method, client.URL)
 				} else if strings.IndexAny(authLine, "Basic") == 0 {
-					// not support yet
-					// TODO..
+					return BasicAuth(authLine, method, client.URL)
 				}
 			}
 			return "", fmt.Errorf("auth error")
@@ -101,9 +115,7 @@ func (client *RTSP) checkAuth(method string, resp *Response) (string, error) {
 				client.authLine = authLine
 				return DigestAuth(authLine, method, client.URL)
 			} else if strings.IndexAny(authLine, "Basic") == 0 {
-				// not support yet
-				// TODO..
-				return "", fmt.Errorf("not support Basic auth yet")
+				return BasicAuth(authLine, method, client.URL)
 			}
 		}
 	}
@@ -143,7 +155,7 @@ func (client *RTSP) requestStream() (err error) {
 	client.connRW = bufio.NewReadWriter(bufio.NewReaderSize(&timeoutConn, networkBuffer), bufio.NewWriterSize(&timeoutConn, networkBuffer))

 	headers := make(map[string]string)
-	headers["Require"] = "implicit-play"
+	//headers["Require"] = "implicit-play"
 	// An OPTIONS request returns the request types the server will accept.
 	resp, err := client.Request("OPTIONS", headers)
 	if err != nil {
@@ -188,73 +200,60 @@ func (client *RTSP) requestStream() (err error) {
 	}
 	client.SDPRaw = resp.Body
 	client.SDPMap = ParseSDP(client.SDPRaw)
+	client.VSdp, client.HasVideo = client.SDPMap["video"]
+	client.ASdp, client.HasAudio = client.SDPMap["audio"]
 	session := ""
-	if videoInfo, ok := client.SDPMap["video"]; ok {
-		client.VControl = videoInfo.Control
-		client.VCodec = videoInfo.Codec
-		client.SPS = videoInfo.SpropParameterSets[0]
-		client.PPS = videoInfo.SpropParameterSets[1]
-		var _url = ""
-		if strings.Index(strings.ToLower(client.VControl), "rtsp://") == 0 {
-			_url = client.VControl
-		} else {
-			_url = strings.TrimRight(client.URL, "/") + "/" + strings.TrimLeft(client.VControl, "/")
-		}
+	otherChannel := 4
+	for t, sdpInfo := range client.SDPMap {
 		headers = make(map[string]string)
-		if client.TransType == TRANS_TYPE_TCP {
-			headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.vRTPChannel, client.vRTPControlChannel)
-		} else {
-			if client.UDPServer == nil {
-				client.UDPServer = &UDPServer{Session: client}
-			}
-			//RTP/AVP;unicast;client_port=64864-64865
-			err = client.UDPServer.SetupVideo()
-			if err != nil {
-				Printf("Setup video err.%v", err)
-				return err
-			}
-			headers["Transport"] = fmt.Sprintf("RTP/AVP/UDP;unicast;client_port=%d-%d", client.UDPServer.VPort, client.UDPServer.VControlPort)
-			client.Conn.timeout = 0 //	UDP ignore timeout
-		}
 		if session != "" {
 			headers["Session"] = session
 		}
-		Printf("Parse DESCRIBE response, VIDEO VControl:%s, VCode:%s, url:%s,Session:%s,vRTPChannel:%d,vRTPControlChannel:%d", client.VControl, client.VCodec, _url, session, client.vRTPChannel, client.vRTPControlChannel)
-		if resp, err = client.RequestWithPath("SETUP", _url, headers, true); err != nil {
-			return err
+		var _url = sdpInfo.Control
+		if !strings.HasPrefix(strings.ToLower(sdpInfo.Control), "rtsp://") {
+			_url = strings.TrimRight(client.URL, "/") + "/" + strings.TrimLeft(sdpInfo.Control, "/")
 		}
-		session, _ = resp.Header["Session"].(string)
-		session = strings.Split(session, ";")[0]
-	}
-	if audioInfo, ok := client.SDPMap["audio"]; ok {
-		client.AControl = audioInfo.Control
-		client.ACodec = audioInfo.Codec
-		client.AudioSpecificConfig = audioInfo.Config
-		var _url = ""
-		if strings.Index(strings.ToLower(client.AControl), "rtsp://") == 0 {
-			_url = client.AControl
-		} else {
-			_url = strings.TrimRight(client.URL, "/") + "/" + strings.TrimLeft(client.AControl, "/")
-		}
-		headers = make(map[string]string)
-		if client.TransType == TRANS_TYPE_TCP {
-			headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.aRTPChannel, client.aRTPControlChannel)
-		} else {
-			if client.UDPServer == nil {
-				client.UDPServer = &UDPServer{Session: client}
+		switch t {
+		case "video":
+			if len(sdpInfo.SpropParameterSets) > 1 {
+				client.WriteSPS(sdpInfo.SpropParameterSets[0])
+				client.WritePPS(sdpInfo.SpropParameterSets[1])
 			}
-			err = client.UDPServer.SetupAudio()
-			if err != nil {
-				Printf("Setup audio err.%v", err)
-				return err
+			if client.TransType == TRANS_TYPE_TCP {
+				headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.vRTPChannel, client.vRTPControlChannel)
+			} else {
+				//RTP/AVP;unicast;client_port=64864-64865
+				if err = client.UDPServer.SetupVideo(); err != nil {
+					Printf("Setup video err.%v", err)
+					return err
+				}
+				headers["Transport"] = fmt.Sprintf("RTP/AVP/UDP;unicast;client_port=%d-%d", client.UDPServer.VPort, client.UDPServer.VControlPort)
+				client.Conn.timeout = 0 //	UDP ignore timeout
+			}
+		case "audio":
+			if len(sdpInfo.Config) > 0 {
+				client.WriteASC(sdpInfo.Config)
+			}else{
+				client.setAudioFormat()
+			}
+			if client.TransType == TRANS_TYPE_TCP {
+				headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.aRTPChannel, client.aRTPControlChannel)
+			} else {
+				if err = client.UDPServer.SetupAudio(); err != nil {
+					Printf("Setup audio err.%v", err)
+					return err
+				}
+				headers["Transport"] = fmt.Sprintf("RTP/AVP/UDP;unicast;client_port=%d-%d", client.UDPServer.APort, client.UDPServer.AControlPort)
+				client.Conn.timeout = 0 //	UDP ignore timeout
+			}
+		default:
+			if client.TransType == TRANS_TYPE_TCP {
+				headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", otherChannel, otherChannel+1)
+				otherChannel += 2
+			} else {
+				//TODO: UDP support
 			}
-			headers["Transport"] = fmt.Sprintf("RTP/AVP/UDP;unicast;client_port=%d-%d", client.UDPServer.APort, client.UDPServer.AControlPort)
-			client.Conn.timeout = 0 //	UDP ignore timeout
 		}
-		if session != "" {
-			headers["Session"] = session
-		}
-		Printf("Parse DESCRIBE response, AUDIO AControl:%s, ACodec:%s, url:%s,Session:%s, aRTPChannel:%d,aRTPControlChannel:%d", client.AControl, client.ACodec, _url, session, client.aRTPChannel, client.aRTPControlChannel)
 		if resp, err = client.RequestWithPath("SETUP", _url, headers, true); err != nil {
 			return err
 		}
@@ -264,27 +263,51 @@ func (client *RTSP) requestStream() (err error) {
 	headers = make(map[string]string)
 	if session != "" {
 		headers["Session"] = session
+		client.Session = session
 	}
 	resp, err = client.Request("PLAY", headers)
 	return err
 }

 func (client *RTSP) startStream() {
-	//startTime := time.Now()
+	startTime := time.Now()
 	//loggerTime := time.Now().Add(-10 * time.Second)
-	defer client.Stop()
+	defer func() {
+		if client.Err() == nil && config.Reconnect {
+			Printf("reconnecting:%s", client.URL)
+			client.RTSPClientInfo = RTSPClientInfo{}
+			if err := client.requestStream(); err != nil {
+				t := time.NewTicker(time.Second * 5)
+				for {
+					Printf("reconnecting:%s in 5 seconds", client.URL)
+					select {
+					case <-client.Done():
+						client.Stop()
+						return
+					case <-t.C:
+						if err = client.requestStream(); err == nil {
+							go client.startStream()
+							return
+						}
+					}
+				}
+			} else {
+				go client.startStream()
+			}
+		} else {
+			client.Stop()
+		}
+	}()
 	for client.Err() == nil {
-		//if client.OptionIntervalMillis > 0 {
-		//	if time.Since(startTime) > time.Duration(client.OptionIntervalMillis)*time.Millisecond {
-		//		startTime = time.Now()
-		//		headers := make(map[string]string)
-		//		headers["Require"] = "implicit-play"
-		//		// An OPTIONS request returns the request types the server will accept.
-		//		if err := client.RequestNoResp("OPTIONS", headers); err != nil {
-		//			// ignore...
-		//		}
-		//	}
-		//}
+		if time.Since(startTime) > time.Minute {
+			startTime = time.Now()
+			headers := make(map[string]string)
+			headers["Require"] = "implicit-play"
+			// An OPTIONS request returns the request types the server will accept.
+			if err := client.RequestNoResp("GET_PARAMETER", headers); err != nil {
+				// ignore...
+			}
+		}
 		b, err := client.connRW.ReadByte()
 		if err != nil {
 			Printf("client.connRW.ReadByte err:%v", err)
@@ -307,38 +330,30 @@ func (client *RTSP) startStream() {
 				Printf("io.ReadFull err:%v", err)
 				return
 			}
-			rtpBuf := content
 			var pack *RTPPack
+
 			switch channel {
 			case client.aRTPChannel:
 				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBuf,
+					Type: RTP_TYPE_AUDIO,
 				}
 			case client.aRTPControlChannel:
 				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIOCONTROL,
-					Buffer: rtpBuf,
+					Type: RTP_TYPE_AUDIOCONTROL,
 				}
 			case client.vRTPChannel:
 				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBuf,
+					Type: RTP_TYPE_VIDEO,
 				}
 			case client.vRTPControlChannel:
 				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEOCONTROL,
-					Buffer: rtpBuf,
+					Type: RTP_TYPE_VIDEOCONTROL,
 				}
 			default:
 				Printf("unknow rtp pack type, channel:%v", channel)
 				continue
 			}
-			if pack == nil {
-				Printf("session tcp got nil rtp pack")
-				continue
-			}
-
+			pack.Unmarshal(content)
 			//if client.debugLogEnable {
 			//	rtp := ParseRTP(pack.Buffer)
 			//	if rtp != nil {
@@ -357,7 +372,7 @@ func (client *RTSP) startStream() {
 			//}

 			client.InBytes += int(length + 4)
-			client.handleRTP(pack)
+			client.PushPack(pack)

 		default: // rtsp
 			builder := bytes.Buffer{}
--- a/go.mod
+++ b/go.mod
@@ -3,12 +3,12 @@ module github.com/Monibuca/plugin-rtsp
 go 1.13

 require (
-	github.com/Monibuca/engine/v2 v2.0.0
-	github.com/gobwas/httphead v0.0.0-20180130184737-2c6c146eadee // indirect
-	github.com/gobwas/pool v0.2.0 // indirect
-	github.com/gobwas/ws v1.0.3 // indirect
-	github.com/jinzhu/gorm v1.9.12 // indirect
-	github.com/pixelbender/go-sdp v1.0.0
-	github.com/reactivex/rxgo v1.0.0 // indirect
+	github.com/Monibuca/engine/v2 v2.2.2
+	github.com/Monibuca/plugin-rtp v1.0.0
+	github.com/logrusorgru/aurora v2.0.3+incompatible // indirect
+	github.com/mattn/go-colorable v0.1.7 // indirect
+	github.com/pion/rtp v1.6.0 // indirect
+	github.com/shirou/gopsutil v2.20.7+incompatible // indirect
 	github.com/teris-io/shortid v0.0.0-20171029131806-771a37caa5cf
+	golang.org/x/sys v0.0.0-20200828161417-c663848e9a16 // indirect
 )
--- a/go.sum
+++ b/go.sum
@@ -1,87 +1,68 @@
 github.com/BurntSushi/toml v0.3.1 h1:WXkYYl6Yr3qBf1K79EBnL4mak0OimBfB0XUf9Vl28OQ=
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=
-github.com/EasyDarwin/EasyDarwin v8.1.0+incompatible h1:Rr8dRbZtcJhiJvGx5Vs7IENM6RUUwGkZiIj5+WrNhm8=
-github.com/EasyDarwin/EasyDarwin v8.1.0+incompatible/go.mod h1:xnmC+Q2+wugEDpQGxivSFNYPOhmNlIQHBfl0hMeriSU=
-github.com/Monibuca/engine v1.2.1 h1:TJmC6eZA1lR1MScWgempZLiEZD4T6aY/nn/rlQ9UdK8=
-github.com/Monibuca/engine v1.2.1/go.mod h1:WbDkXENLjcPjyjCR1Mix1GA+uAlwORkv/+8aMVrDX2g=
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+github.com/Monibuca/plugin-rtp v0.0.0-20200531014802-504413c0dfcb/go.mod h1:8HxBilkF835Lepe/DLUCjaw1mRiu3MxTDsG7g9UcfZA=
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-github.com/denisenkom/go-mssqldb v0.0.0-20191124224453-732737034ffd h1:83Wprp6ROGeiHFAP8WJdI2RoxALQYgdllERc3N5N2DM=
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--- a/main.go
+++ b/main.go
@@ -2,7 +2,6 @@ package rtsp

 import (
 	"bufio"
-	"bytes"
 	"fmt"
 	"log"
 	"net"
@@ -12,18 +11,23 @@ import (
 	"time"

 	. "github.com/Monibuca/engine/v2"
-	. "github.com/Monibuca/engine/v2/avformat"
 	"github.com/Monibuca/engine/v2/util"
+	. "github.com/Monibuca/plugin-rtp"
 	"github.com/teris-io/shortid"
 )

-var collection = sync.Map{}
+var collection sync.Map
 var config = struct {
-	ListenAddr string
-	AutoPull   bool
-	RemoteAddr string
-	Timeout    int
-}{":554", false, "rtsp://localhost/${streamPath}", 0}
+	ListenAddr   string
+	AutoPull     bool
+	RemoteAddr   string
+	Timeout      int
+	Reconnect    bool
+	AutoPullList []*struct {
+		URL        string
+		StreamPath string
+	}
+}{":554", false, "rtsp://localhost/${streamPath}", 0, false, nil}

 func init() {
 	InstallPlugin(&PluginConfig{
@@ -67,6 +71,13 @@ func runPlugin() {
 			w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
 		}
 	})
+	if len(config.AutoPullList) > 0 {
+		for _, info := range config.AutoPullList {
+			if err := new(RTSP).PullStream(info.StreamPath, info.URL); err != nil {
+				Println(err)
+			}
+		}
+	}
 	if config.ListenAddr != "" {
 		log.Fatal(ListenRtsp(config.ListenAddr))
 	}
@@ -118,7 +129,7 @@ func ListenRtsp(addr string) error {
 }

 type RTSP struct {
-	Publisher
+	RTP
 	RTSPInfo
 	RTSPClientInfo
 	ID        string
@@ -131,27 +142,35 @@ type RTSP struct {
 	SDPMap   map[string]*SDPInfo
 	nonce    string
 	closeOld bool
-	AControl string
-	VControl string
-	ACodec   string
-	VCodec   string
-	avcsent  bool
+	ASdp     *SDPInfo
+	VSdp     *SDPInfo
 	aacsent  bool
 	Timeout  int
-	// stats info
-	fuBuffer []byte
 	//tcp channels
-	aRTPChannel         int
-	aRTPControlChannel  int
-	vRTPChannel         int
-	vRTPControlChannel  int
-	UDPServer           *UDPServer
-	UDPClient           *UDPClient
-	SPS                 []byte
-	PPS                 []byte
-	AudioSpecificConfig []byte
-	Auth                func(string) string
+	aRTPChannel        int
+	aRTPControlChannel int
+	vRTPChannel        int
+	vRTPControlChannel int
+	UDPServer          *UDPServer
+	UDPClient          *UDPClient
+	Auth               func(string) string
 }
+
+func (rtsp *RTSP) setAudioFormat() {
+	switch rtsp.ASdp.Codec {
+	case "aac":
+		rtsp.AudioInfo.SoundFormat = 10
+	case "pcma":
+		rtsp.AudioInfo.SoundFormat = 7
+		rtsp.AudioInfo.SoundRate = rtsp.ASdp.TimeScale
+		rtsp.AudioInfo.SoundSize = 16
+	case "pcmu":
+		rtsp.AudioInfo.SoundFormat = 8
+		rtsp.AudioInfo.SoundRate = rtsp.ASdp.TimeScale
+		rtsp.AudioInfo.SoundSize = 16
+	}
+}
+
 type RTSPClientInfo struct {
 	Agent    string
 	Session  string
@@ -159,12 +178,10 @@ type RTSPClientInfo struct {
 	Seq      int
 }
 type RTSPInfo struct {
-	URL       string
-	SyncCount int64
-	SDPRaw    string
-	InBytes   int
-	OutBytes  int
-
+	URL        string
+	SDPRaw     string
+	InBytes    int
+	OutBytes   int
 	StreamInfo *StreamInfo
 }

@@ -192,115 +209,3 @@ func (conn *RichConn) Write(b []byte) (n int, err error) {
 	}
 	return conn.Conn.Write(b)
 }
-func (rtsp *RTSP) handleNALU(nalType byte, payload []byte, ts int64) {
-	rtsp.SyncCount++
-	vl := len(payload)
-	switch nalType {
-	// case NALU_SPS:
-	// 	r := bytes.NewBuffer([]byte{})
-	// 	r.Write(RTMP_AVC_HEAD)
-	// 	util.BigEndian.PutUint16(spsHead[1:], uint16(vl))
-	// 	r.Write(spsHead)
-	// 	r.Write(payload)
-	// case NALU_PPS:
-	// 	r := bytes.NewBuffer([]byte{})
-	// 	util.BigEndian.PutUint16(ppsHead[1:], uint16(vl))
-	// 	r.Write(ppsHead)
-	// 	r.Write(payload)
-	// 	rtsp.PushVideo(0, r.Bytes())
-	// avcsent = true
-	case NALU_IDR_Picture:
-		if !rtsp.avcsent {
-			r := bytes.NewBuffer([]byte{})
-			r.Write(RTMP_AVC_HEAD)
-			spsHead := []byte{0xE1, 0, 0}
-			util.BigEndian.PutUint16(spsHead[1:], uint16(len(rtsp.SPS)))
-			r.Write(spsHead)
-			r.Write(rtsp.SPS)
-			ppsHead := []byte{0x01, 0, 0}
-			util.BigEndian.PutUint16(ppsHead[1:], uint16(len(rtsp.PPS)))
-			r.Write(ppsHead)
-			r.Write(rtsp.PPS)
-			rtsp.PushVideo(0, r.Bytes())
-			rtsp.avcsent = true
-		}
-		r := bytes.NewBuffer([]byte{})
-		iframeHead := []byte{0x17, 0x01, 0, 0, 0}
-		util.BigEndian.PutUint24(iframeHead[2:], 0)
-		r.Write(iframeHead)
-		nalLength := []byte{0, 0, 0, 0}
-		util.BigEndian.PutUint32(nalLength, uint32(vl))
-		r.Write(nalLength)
-		r.Write(payload)
-		rtsp.PushVideo(uint32(ts), r.Bytes())
-	case NALU_Non_IDR_Picture:
-		r := bytes.NewBuffer([]byte{})
-		pframeHead := []byte{0x27, 0x01, 0, 0, 0}
-		util.BigEndian.PutUint24(pframeHead[2:], 0)
-		r.Write(pframeHead)
-		nalLength := []byte{0, 0, 0, 0}
-		util.BigEndian.PutUint32(nalLength, uint32(vl))
-		r.Write(nalLength)
-		r.Write(payload)
-		rtsp.PushVideo(uint32(ts), r.Bytes())
-	}
-}
-func (rtsp *RTSP) handleRTP(pack *RTPPack) {
-	data := pack.Buffer
-	switch pack.Type {
-	case RTP_TYPE_AUDIO:
-		if !rtsp.aacsent {
-			rtsp.PushAudio(0, append([]byte{0xAF, 0x00}, rtsp.AudioSpecificConfig...))
-			rtsp.aacsent = true
-		}
-		cc := data[0] & 0xF
-		rtphdr := 12 + cc*4
-		payload := data[rtphdr:]
-		auHeaderLen := (int16(payload[0]) << 8) + int16(payload[1])
-		auHeaderLen = auHeaderLen >> 3
-		auHeaderCount := int(auHeaderLen / 2)
-		var auLenArray []int
-		for iIndex := 0; iIndex < int(auHeaderCount); iIndex++ {
-			auHeaderInfo := (int16(payload[2+2*iIndex]) << 8) + int16(payload[2+2*iIndex+1])
-			auLen := auHeaderInfo >> 3
-			auLenArray = append(auLenArray, int(auLen))
-		}
-		startOffset := 2 + 2*auHeaderCount
-		for _, auLen := range auLenArray {
-			endOffset := startOffset + auLen
-			addHead := []byte{0xAF, 0x01}
-			rtsp.PushAudio(0, append(addHead, payload[startOffset:endOffset]...))
-			startOffset = startOffset + auLen
-		}
-	case RTP_TYPE_VIDEO:
-		cc := data[0] & 0xF
-		//rtp header
-		rtphdr := 12 + cc*4
-
-		//packet time
-		ts := (int64(data[4]) << 24) + (int64(data[5]) << 16) + (int64(data[6]) << 8) + (int64(data[7]))
-
-		//packet number
-		//packno := (int64(data[6]) << 8) + int64(data[7])
-		data = data[rtphdr:]
-		nalType := data[0] & 0x1F
-
-		if nalType >= 1 && nalType <= 23 {
-			rtsp.handleNALU(nalType, data, ts)
-		} else if nalType == 28 {
-			isStart := data[1]&0x80 != 0
-			isEnd := data[1]&0x40 != 0
-			nalType := data[1] & 0x1F
-			//nri := (data[1]&0x60)>>5
-			nal := data[0]&0xE0 | data[1]&0x1F
-			if isStart {
-				rtsp.fuBuffer = []byte{0}
-			}
-			rtsp.fuBuffer = append(rtsp.fuBuffer, data[2:]...)
-			if isEnd {
-				rtsp.fuBuffer[0] = nal
-				rtsp.handleNALU(nalType, rtsp.fuBuffer, ts)
-			}
-		}
-	}
-}
--- a/sdp-parser.go
+++ b/sdp-parser.go
@@ -31,17 +31,13 @@ func ParseSDP(sdpRaw string) map[string]*SDPInfo {
 			switch typeval[0] {
 			case "m":
 				if len(fields) > 0 {
-					switch fields[0] {
-					case "audio", "video":
-						sdpMap[fields[0]] = &SDPInfo{AVType: fields[0]}
-						info = sdpMap[fields[0]]
-						mfields := strings.Split(fields[1], " ")
-						if len(mfields) >= 3 {
-							info.PayloadType, _ = strconv.Atoi(mfields[2])
-						}
+					info = &SDPInfo{AVType: fields[0]}
+					sdpMap[info.AVType] = info
+					mfields := strings.Split(fields[1], " ")
+					if len(mfields) >= 3 {
+						info.PayloadType, _ = strconv.Atoi(mfields[2])
 					}
 				}
-
 			case "a":
 				if info != nil {
 					for _, field := range fields {
@@ -60,6 +56,10 @@ func ParseSDP(sdpRaw string) map[string]*SDPInfo {
 						if len(keyval) >= 2 {
 							key := keyval[0]
 							switch key {
+							case "PCMA":
+								info.Codec = "pcma"
+							case "PCMU":
+								info.Codec = "pcmu"
 							case "MPEG4-GENERIC":
 								info.Codec = "aac"
 							case "H264":
--- a/session.go
+++ b/session.go
@@ -13,14 +13,10 @@ import (
 	"time"

 	. "github.com/Monibuca/engine/v2"
+	. "github.com/Monibuca/plugin-rtp"
 	"github.com/teris-io/shortid"
 )

-type RTPPack struct {
-	Type   RTPType
-	Buffer []byte
-}
-
 type SessionType int

 const (
@@ -38,29 +34,6 @@ func (st SessionType) String() string {
 	return "unknow"
 }

-type RTPType int
-
-const (
-	RTP_TYPE_AUDIO RTPType = iota
-	RTP_TYPE_VIDEO
-	RTP_TYPE_AUDIOCONTROL
-	RTP_TYPE_VIDEOCONTROL
-)
-
-func (rt RTPType) String() string {
-	switch rt {
-	case RTP_TYPE_AUDIO:
-		return "audio"
-	case RTP_TYPE_VIDEO:
-		return "video"
-	case RTP_TYPE_AUDIOCONTROL:
-		return "audio control"
-	case RTP_TYPE_VIDEOCONTROL:
-		return "video control"
-	}
-	return "unknow"
-}
-
 type TransType int

 const (
@@ -128,53 +101,41 @@ func (session *RTSP) AcceptPush() {
 			}
 			channel := int(buf1)
 			rtpLen := int(binary.BigEndian.Uint16(buf2))
+			pack := new(RTPPack)
 			rtpBytes := make([]byte, rtpLen)
 			if _, err := io.ReadFull(session.connRW, rtpBytes); err != nil {
 				Println(err)
 				return
 			}
-			var pack *RTPPack
+			if err = pack.Unmarshal(rtpBytes); err != nil {
+				Println(err)
+				return
+			}
 			switch channel {
 			case session.aRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBytes,
-				}
+				pack.Type = RTP_TYPE_AUDIO
 				elapsed := time.Now().Sub(timer)
 				if elapsed >= 30*time.Second {
 					Println("Recv an audio RTP package")
 					timer = time.Now()
 				}
 			case session.aRTPControlChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIOCONTROL,
-					Buffer: rtpBytes,
-				}
+				pack.Type = RTP_TYPE_AUDIOCONTROL
 			case session.vRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBytes,
-				}
+				pack.Type = RTP_TYPE_VIDEO
 				elapsed := time.Now().Sub(timer)
 				if elapsed >= 30*time.Second {
 					Println("Recv an video RTP package")
 					timer = time.Now()
 				}
 			case session.vRTPControlChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEOCONTROL,
-					Buffer: rtpBytes,
-				}
+				pack.Type = RTP_TYPE_VIDEOCONTROL
 			default:
 				Printf("unknow rtp pack type, %v", pack.Type)
 				continue
 			}
-			if pack == nil {
-				Printf("session tcp got nil rtp pack")
-				continue
-			}
 			session.InBytes += rtpLen + 4
-			session.handleRTP(pack)
+			session.PushPack(pack)
 		} else { // rtsp cmd
 			reqBuf := bytes.NewBuffer(nil)
 			reqBuf.WriteByte(buf1)
@@ -356,21 +317,22 @@ func (session *RTSP) handleRequest(req *Request) {

 		session.SDPRaw = req.Body
 		session.SDPMap = ParseSDP(req.Body)
-		sdp, ok := session.SDPMap["audio"]
-		if ok {
-			session.AControl = sdp.Control
-			session.ACodec = sdp.Codec
-			session.AudioSpecificConfig = sdp.Config
-			Printf("audio codec[%s]\n", session.ACodec)
-		}
-		if sdp, ok = session.SDPMap["video"]; ok {
-			session.VControl = sdp.Control
-			session.VCodec = sdp.Codec
-			session.SPS = sdp.SpropParameterSets[0]
-			session.PPS = sdp.SpropParameterSets[1]
-			Printf("video codec[%s]\n", session.VCodec)
-		}
-		if session.Publisher.Publish(streamPath) {
+		if session.Publish(streamPath) {
+			if session.ASdp, session.HasAudio = session.SDPMap["audio"]; session.HasAudio {
+				if len(session.ASdp.Control) > 0 {
+					session.WriteASC(session.ASdp.Config)
+				} else {
+					session.setAudioFormat()
+				}
+				Printf("audio codec[%s]\n", session.ASdp.Codec)
+			}
+			if session.VSdp, session.HasVideo = session.SDPMap["video"]; session.HasVideo {
+				if len(session.VSdp.SpropParameterSets) > 1 {
+					session.WriteSPS(session.VSdp.SpropParameterSets[0])
+					session.WritePPS(session.VSdp.SpropParameterSets[1])
+				}
+				Printf("video codec[%s]\n", session.VSdp.Codec)
+			}
 			session.Stream.Type = "RTSP"
 			session.RTSPInfo.StreamInfo = &session.Stream.StreamInfo
 			collection.Store(streamPath, session)
@@ -417,36 +379,38 @@ func (session *RTSP) handleRequest(req *Request) {
 		//	res.Status = "Error Status"
 		//	return
 		//}
-		vPath := ""
-		if strings.Index(strings.ToLower(session.VControl), "rtsp://") == 0 {
-			vControlUrl, err := url.Parse(session.VControl)
-			if err != nil {
-				res.StatusCode = 500
-				res.Status = "Invalid VControl"
-				return
+		var vPath, aPath string
+		if session.HasVideo {
+			if strings.Index(strings.ToLower(session.VSdp.Control), "rtsp://") == 0 {
+				vControlUrl, err := url.Parse(session.VSdp.Control)
+				if err != nil {
+					res.StatusCode = 500
+					res.Status = "Invalid VControl"
+					return
+				}
+				if vControlUrl.Port() == "" {
+					vControlUrl.Host = fmt.Sprintf("%s:554", vControlUrl.Host)
+				}
+				vPath = vControlUrl.String()
+			} else {
+				vPath = session.VSdp.Control
 			}
-			if vControlUrl.Port() == "" {
-				vControlUrl.Host = fmt.Sprintf("%s:554", vControlUrl.Host)
-			}
-			vPath = vControlUrl.String()
-		} else {
-			vPath = session.VControl
 		}
-
-		aPath := ""
-		if strings.Index(strings.ToLower(session.AControl), "rtsp://") == 0 {
-			aControlUrl, err := url.Parse(session.AControl)
-			if err != nil {
-				res.StatusCode = 500
-				res.Status = "Invalid AControl"
-				return
+		if session.HasAudio {
+			if strings.Index(strings.ToLower(session.ASdp.Control), "rtsp://") == 0 {
+				aControlUrl, err := url.Parse(session.ASdp.Control)
+				if err != nil {
+					res.StatusCode = 500
+					res.Status = "Invalid AControl"
+					return
+				}
+				if aControlUrl.Port() == "" {
+					aControlUrl.Host = fmt.Sprintf("%s:554", aControlUrl.Host)
+				}
+				aPath = aControlUrl.String()
+			} else {
+				aPath = session.ASdp.Control
 			}
-			if aControlUrl.Port() == "" {
-				aControlUrl.Host = fmt.Sprintf("%s:554", aControlUrl.Host)
-			}
-			aPath = aControlUrl.String()
-		} else {
-			aPath = session.AControl
 		}

 		mtcp := regexp.MustCompile("interleaved=(\\d+)(-(\\d+))?")
@@ -577,7 +541,7 @@ func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
 			return
 		}
 		err = session.UDPClient.SendRTP(pack)
-		session.OutBytes += len(pack.Buffer)
+		session.OutBytes += len(pack.Raw)
 		return
 	}
 	switch pack.Type {
@@ -588,12 +552,12 @@ func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
 		session.connWLock.Lock()
 		session.connRW.Write(bufChannel)
 		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Buffer)))
+		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
 		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
+		session.connRW.Write(pack.Raw)
 		session.connRW.Flush()
 		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 4
+		session.OutBytes += len(pack.Raw) + 4
 	case RTP_TYPE_AUDIOCONTROL:
 		bufChannel := make([]byte, 2)
 		bufChannel[0] = 0x24
@@ -601,12 +565,12 @@ func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
 		session.connWLock.Lock()
 		session.connRW.Write(bufChannel)
 		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Buffer)))
+		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
 		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
+		session.connRW.Write(pack.Raw)
 		session.connRW.Flush()
 		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 4
+		session.OutBytes += len(pack.Raw) + 4
 	case RTP_TYPE_VIDEO:
 		bufChannel := make([]byte, 2)
 		bufChannel[0] = 0x24
@@ -614,12 +578,12 @@ func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
 		session.connWLock.Lock()
 		session.connRW.Write(bufChannel)
 		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Buffer)))
+		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
 		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
+		session.connRW.Write(pack.Raw)
 		session.connRW.Flush()
 		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 4
+		session.OutBytes += len(pack.Raw) + 4
 	case RTP_TYPE_VIDEOCONTROL:
 		bufChannel := make([]byte, 2)
 		bufChannel[0] = 0x24
@@ -627,12 +591,12 @@ func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
 		session.connWLock.Lock()
 		session.connRW.Write(bufChannel)
 		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Buffer)))
+		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
 		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
+		session.connRW.Write(pack.Raw)
 		session.connRW.Flush()
 		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 4
+		session.OutBytes += len(pack.Raw) + 4
 	default:
 		err = fmt.Errorf("session tcp send rtp got unkown pack type[%v]", pack.Type)
 	}
--- a/udp-client.go
+++ b/udp-client.go
@@ -2,9 +2,10 @@ package rtsp

 import (
 	"fmt"
+	. "github.com/Monibuca/engine/v2"
+	. "github.com/Monibuca/plugin-rtp"
 	"net"
 	"strings"
-	. "github.com/Monibuca/engine/v2"
 )

 type UDPClient struct {
@@ -151,7 +152,7 @@ func (c *UDPClient) SendRTP(pack *RTPPack) (err error) {
 		return
 	}

-	if _, err = conn.Write(pack.Buffer);err != nil {
+	if _, err = conn.Write(pack.Raw); err != nil {
 		err = fmt.Errorf("udp client write bytes error, %v", err)
 		return
 	}
--- a/udp-server.go
+++ b/udp-server.go
@@ -9,10 +9,11 @@ import (
 	"time"

 	. "github.com/Monibuca/engine/v2"
+	. "github.com/Monibuca/plugin-rtp"
 )

 type UDPServer struct {
-	Session    *RTSP
+	Session *RTSP
 	UDPClient
 	sync.Mutex
 }
@@ -29,7 +30,7 @@ func (s *UDPServer) HandleRTP(pack *RTPPack) {
 	s.Lock()
 	defer s.Unlock()
 	if s.Session != nil {
-		s.Session.handleRTP(pack)
+		s.Session.PushPack(pack)
 	}
 }

@@ -90,13 +91,11 @@ func (s *UDPServer) SetupAudio() (err error) {
 					Printf("Package recv from AConn.len:%d\n", n)
 					timer = time.Now()
 				}
-				rtpBytes := make([]byte, n)
 				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
 				pack := &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBytes,
+					Type: RTP_TYPE_AUDIO,
 				}
+				pack.Unmarshal(bufUDP[:n])
 				s.HandleRTP(pack)
 			} else {
 				Println("udp server read audio pack error", err)
@@ -131,13 +130,11 @@ func (s *UDPServer) SetupAudio() (err error) {
 		for !s.Stoped {
 			if n, _, err := s.AControlConn.ReadFromUDP(bufUDP); err == nil {
 				//Printf("Package recv from AControlConn.len:%d\n", n)
-				rtpBytes := make([]byte, n)
 				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
 				pack := &RTPPack{
-					Type:   RTP_TYPE_AUDIOCONTROL,
-					Buffer: rtpBytes,
+					Type: RTP_TYPE_AUDIOCONTROL,
 				}
+				pack.Unmarshal(bufUDP[:n])
 				s.HandleRTP(pack)
 			} else {
 				Println("udp server read audio control pack error", err)
@@ -182,13 +179,11 @@ func (s *UDPServer) SetupVideo() (err error) {
 					Printf("Package recv from VConn.len:%d\n", n)
 					timer = time.Now()
 				}
-				rtpBytes := make([]byte, n)
 				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
 				pack := &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBytes,
+					Type: RTP_TYPE_VIDEO,
 				}
+				pack.Unmarshal(bufUDP[:n])
 				s.HandleRTP(pack)
 			} else {
 				Println("udp server read video pack error", err)
@@ -224,13 +219,11 @@ func (s *UDPServer) SetupVideo() (err error) {
 		for !s.Stoped {
 			if n, _, err := s.VControlConn.ReadFromUDP(bufUDP); err == nil {
 				//Printf("Package recv from VControlConn.len:%d\n", n)
-				rtpBytes := make([]byte, n)
 				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
 				pack := &RTPPack{
-					Type:   RTP_TYPE_VIDEOCONTROL,
-					Buffer: rtpBytes,
+					Type: RTP_TYPE_VIDEOCONTROL,
 				}
+				pack.Unmarshal(bufUDP[:n])
 				s.HandleRTP(pack)
 			} else {
 				Println("udp server read video control pack error", err)
Author	SHA1	Message	Date
langhuihui	7e61ba71f7	怎加初始拉流功能	2020-10-01 20:12:26 +08:00
langhuihui	d6384dcbd5	版本更新	2020-09-23 23:04:07 +08:00
langhuihui	2159a6fd9b	解决rtsp推流无视频时的报错	2020-09-23 22:55:52 +08:00
langhuihui	02f3e91085	对其他音频的支持	2020-09-20 15:54:27 +08:00
langhuihui	7f40078b50	添加对engine版本的依赖	2020-08-29 08:00:25 +08:00
dexter	bb563d64c7	Merge pull request #5 from ourfor-pp/master 修复保活请求的bug	2020-08-29 07:48:16 +08:00
mqh	f7cb146b89	修复client.Session未保存，导致保活请求未携带session的bug	2020-08-28 17:05:45 +08:00
dexter	9bb49cb9f7	Merge pull request #4 from ourfor-pp/master 修改保活请求	2020-08-28 13:53:48 +08:00
mqh	087d1aab4d	增加basic登录（大华录像机测试验证）	2020-08-28 09:56:01 +08:00
mqh	f949464328	Merge branch 'master' of https://github.com/ourfor-pp/plugin-rtsp	2020-08-28 09:33:53 +08:00
mqh	d89f1e2405	将保活请求由OPTIONS改为GET_PARAMETER（来自VLC保活参考）	2020-08-28 09:26:57 +08:00
langhuihui	1d3fbfc20b	增加对纯音频的播放的支持	2020-08-27 08:50:45 +08:00
dexter	fd64a69a12	Merge pull request #2 from zbjlala/master master	2020-08-26 22:17:37 +08:00
zbj	0e4406ad14	[fix]宇视摄像头DESCRIBE 有两个RTSP流一个video matedata 需要两个SETUP才可播放	2020-08-25 11:55:49 +08:00
zbj	22f33886a9	[fix]宇视摄像头不支持该OPTIONS操作 551	2020-08-25 11:44:54 +08:00
langhuihui	8b1892209d	增加5分钟重连机制	2020-07-12 10:40:02 +08:00
langhuihui	2e9cf9a4ca	重连机制修复	2020-07-11 21:54:59 +08:00
langhuihui	67da93d8e2	增加重连时的判断	2020-07-09 22:04:00 +08:00
unknown	cb733b368f	增加重连功能	2020-07-09 20:06:35 +08:00
langhuihui	fadeccddab	更新依赖engine的版本	2020-06-09 07:12:20 +08:00
langhuihui	93df7632a6	修正升级带来的bug	2020-06-09 07:09:25 +08:00
langhuihui	53c4788df2	改用RTP插件	2020-05-31 10:01:29 +08:00
langhuihui	f5bdd6a298	内存复用	2020-05-24 22:58:38 +08:00
李宇翔	eaddc60775	过滤开头的NonIDR	2020-05-21 18:16:56 +08:00