fix: upgrade gortsplib

chore: fit engine udpate
docs: update readme
2025-09-27 20:12:14 +08:00 · 2023-08-25 15:00:57 +08:00 · 2023-08-14 10:04:37 +08:00 · 2023-08-06 18:05:06 +08:00 · 2023-08-06 15:26:50 +08:00 · 2023-08-06 15:15:26 +08:00
17 changed files with 1025 additions and 2104 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -1,2 +0,0 @@
-.vscode
-node_modules
--- a/README.en.md
+++ b/README.en.md
@@ -0,0 +1,71 @@
+_[简体中文](https://github.com/Monibuca/plugin-rtsp) | English_
+# RTSP Plugin
+
+The RTSP plugin provides the ability to push and pull the RTSP protocol and also to push and pull the RTSP protocol to remote servers.
+
+## Plugin address
+
+https://github.com/Monibuca/plugin-rtsp
+
+## Plugin introduction
+
+```go
+import (
+    _ "m7s.live/plugin/rtsp/v4"
+)
+```
+
+## Push and Pull address form
+
+```
+rtsp://localhost/live/test
+```
+- `localhost` is the m7s server domain name or IP address, and the default port `554` can be omitted, otherwise it is required to be written.
+- `live` represents `appName`
+- `test` represents `streamName`
+- `live/test` in m7s will serve as the stream identity.
+
+For example, push stream to m7s through ffmpeg
+
+```bash
+ffmpeg -i [video source] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
+```
+
+This will create a stream named `live/test` inside m7s.
+
+If the `live/test` stream already exists in m7s, then you can use the RTSP protocol to play it.
+
+```bash
+ffplay rtsp://localhost/live/test
+```
+
+## Configuration
+
+```yaml
+rtsp:
+    publish: # Refer to the global configuration format
+    subscribe: # Refer to the global configuration format
+    pull: # Format reference document https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
+    push: # Format reference document https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
+    listenaddr: :554
+    udpaddr: :8000
+    rtcpaddr: :8001
+    readbuffercount: 2048
+    writebuffercount: 2048
+    pullprotocol: tcp # auto, tcp, udp
+```
+:::tip Configuration override
+publish and subscribe, any section not configured will use global configuration.
+:::
+
+## API
+
+### `rtsp/api/list`
+Get all RTSP streams
+
+### `rtsp/api/pull?target=[RTSP address]&streamPath=[Stream identity]&save=[0|1|2]`
+Pull the RTSP to m7s from a remote server
+- save meaning: 0, do not save; 1, save to pullonstart; 2, save to pullonsub
+- The RTSP address needs to be urlencoded to prevent special characters from affecting parsing
+### `rtsp/api/push?target=[RTSP address]&streamPath=[Stream identity]`
+Push local streams to remote servers
--- a/README.md
+++ b/README.md
@@ -1,29 +1,69 @@
-# Monibuca 的RTSP 插件
+_[English](https://github.com/Monibuca/plugin-rtsp/blob/v4/README.en.md) | 简体中文_
+# RTSP插件
+rtsp插件提供rtsp协议的推拉流能力，以及向远程服务器推拉rtsp协议的能力。
+## 插件地址

-主要功能是提供RTSP的端口监听接受RTSP推流，以及对RTSP地址进行拉流转发
+https://github.com/Monibuca/plugin-rtsp

-## 插件名称
+## 插件引入
+```go
+import (
+    _ "m7s.live/plugin/rtsp/v4"
+)
+```

-RTSP
+## 推拉地址形式
+```
+rtsp://localhost/live/test
+```
+- `localhost`是m7s的服务器域名或者IP地址，默认端口`554`可以不写，否则需要写
+- `live`代表`appName`
+- `test`代表`streamName`
+- m7s中`live/test`将作为`streamPath`为流的唯一标识
+
+
+例如通过ffmpeg向m7s进行推流
+
+```bash
+ffmpeg -i [视频源] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
+```
+
+会在m7s内部形成一个名为live/test的流
+
+
+如果m7s中已经存在live/test流的话就可以用rtsp协议进行播放
+```bash
+ffplay rtsp://localhost/live/test
+```

 ## 配置
-```toml
-[RTSP]
-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
-new(RTSP).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
-```
+```yaml
+rtsp:
+    publish: # 参考全局配置格式
+    subscribe: # 参考全局配置格式
+    pull: # 格式参考文档 https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
+    push: # 格式参考文档 https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
+    listenaddr: :554
+    udpaddr: :8000
+    rtcpaddr: :8001
+    readbuffercount: 2048 # 读取缓存队列大小
+    writebuffercount: 2048 # 写出缓存队列大小
+    pullprotocol: tcp # auto, tcp, udp
+```
+:::tip 配置覆盖
+publish
+subscribe
+两项中未配置部分将使用全局配置
+:::
+## API
+
+### `rtsp/api/list`
+获取所有rtsp流
+
+### `rtsp/api/pull?target=[RTSP地址]&streamPath=[流标识]&save=[0|1|2]`
+从远程拉取rtsp到m7s中
+- save含义：0、不保存；1、保存到pullonstart；2、保存到pullonsub
+- RTSP地址需要进行urlencode 防止其中的特殊字符影响解析
+### `rtsp/api/push?target=[RTSP地址]&streamPath=[流标识]`
+将本地的流推送到远端
--- a/client.go
+++ b/client.go
@@ -1,549 +1,151 @@
 package rtsp

 import (
-	"bufio"
-	"bytes"
-	"crypto/md5"
-	"encoding/base64"
-	"encoding/binary"
-	"errors"
-	"fmt"
-	"io"
+	"context"
 	"net"
-	"net/url"
-	"regexp"
-	"strconv"
-	"strings"
-	"time"

-	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
+	"github.com/bluenviron/gortsplib/v3"
+	"github.com/bluenviron/gortsplib/v3/pkg/url"
+	"go.uber.org/zap"
+	"m7s.live/engine/v4"
 )

-// PullStream 从外部拉流
-func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (err error) {
-	rtsp.Stream = &Stream{
-		StreamPath: streamPath,
-		Type:       "RTSP Pull",
-	}
-	if result := rtsp.Publish(); result {
-		rtsp.TransType = TRANS_TYPE_TCP
-		rtsp.vRTPChannel = 0
-		rtsp.vRTPControlChannel = 1
-		rtsp.aRTPChannel = 2
-		rtsp.aRTPControlChannel = 3
-		rtsp.URL = rtspUrl
-		rtsp.UDPServer = &UDPServer{Session: rtsp}
-		if err = rtsp.requestStream(); err != nil {
-			Println(err)
-			rtsp.Close()
-			return
-		}
-		go rtsp.startStream()
-		collection.Store(streamPath, rtsp)
-		return
-	}
-	return errors.New("publish badname")
+type RTSPClient struct {
+	*gortsplib.Client `json:"-" yaml:"-"`
+	gortsplib.Transport
+	DialContext func(ctx context.Context, network, address string) (net.Conn, error) `json:"-" yaml:"-"`
 }
-func DigestAuth(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)
-	}
-	realm := ""
-	nonce := ""
-	realmRex := regexp.MustCompile(`realm="(.*?)"`)
-	result1 := realmRex.FindStringSubmatch(authLine)
-
-	nonceRex := regexp.MustCompile(`nonce="(.*?)"`)
-	result2 := nonceRex.FindStringSubmatch(authLine)
-
-	if len(result1) == 2 {
-		realm = result1[1]
-	} else {
-		return "", fmt.Errorf("auth error : no realm found")
-	}
-	if len(result2) == 2 {
-		nonce = result2[1]
-	} else {
-		return "", fmt.Errorf("auth error : no nonce found")
-	}
-	// response= md5(md5(username:realm:password):nonce:md5(public_method:url));
-	username := l.User.Username()
-	password, _ := l.User.Password()
-	l.User = nil
-	if l.Port() == "" {
-		l.Host = fmt.Sprintf("%s:%s", l.Host, "554")
-	}
-	md5UserRealmPwd := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s:%s", username, realm, password))))
-	md5MethodURL := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s", method, l.String()))))
-
-	response := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s:%s", md5UserRealmPwd, nonce, md5MethodURL))))
-	Authorization := fmt.Sprintf("Digest username=\"%s\", realm=\"%s\", nonce=\"%s\", uri=\"%s\", response=\"%s\"", username, realm, nonce, l.String(), response)
-	return Authorization, nil
+type RTSPPuller struct {
+	RTSPPublisher
+	engine.Puller
+	RTSPClient
 }

-// 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)
+func (p *RTSPClient) Disconnect() {
+	if p.Client != nil {
+		p.Client.Close()
 	}
-	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.
-		AuthHeaders := resp.Header["WWW-Authenticate"]
-		auths, ok := AuthHeaders.([]string)
-		if ok {
-			for _, authLine := range auths {
-				if strings.IndexAny(authLine, "Digest") == 0 {
-					// 					realm="HipcamRealServer",
-					// nonce="3b27a446bfa49b0c48c3edb83139543d"
-					client.authLine = authLine
-					return DigestAuth(authLine, method, client.URL)
-				} else if strings.IndexAny(authLine, "Basic") == 0 {
-					return BasicAuth(authLine, method, client.URL)
-				}
-			}
-			return "", fmt.Errorf("auth error")
-		} else {
-			authLine, _ := AuthHeaders.(string)
-			if strings.IndexAny(authLine, "Digest") == 0 {
-				client.authLine = authLine
-				return DigestAuth(authLine, method, client.URL)
-			} else if strings.IndexAny(authLine, "Basic") == 0 {
-				return BasicAuth(authLine, method, client.URL)
-			}
-		}
+
+func (p *RTSPPuller) Connect() error {
+	p.Client = &gortsplib.Client{
+		DialContext:     p.DialContext,
+		ReadBufferCount: rtspConfig.ReadBufferCount,
+		AnyPortEnable:   true,
 	}
-	return "", nil
-}
-func (client *RTSP) requestStream() (err error) {
-	timeout := time.Duration(5) * time.Second
-	l, err := url.Parse(client.URL)
+
+	switch rtspConfig.PullProtocol {
+	case "tcp", "TCP":
+		p.Transport = gortsplib.TransportTCP
+		p.Client.Transport = &p.Transport
+	case "udp", "UDP":
+		p.Transport = gortsplib.TransportUDP
+		p.Client.Transport = &p.Transport
+	}
+	// parse URL
+	u, err := url.Parse(p.RemoteURL)
 	if err != nil {
 		return err
 	}
-	if strings.ToLower(l.Scheme) != "rtsp" {
-		err = fmt.Errorf("RTSP url is invalid")
-		return err
-	}
-	if strings.ToLower(l.Hostname()) == "" {
-		err = fmt.Errorf("RTSP url is invalid")
-		return err
-	}
-	port := l.Port()
-	if len(port) == 0 {
-		port = "554"
-	}
-	conn, err := net.DialTimeout("tcp", l.Hostname()+":"+port, timeout)
-	if err != nil {
-		// handle error
-		return err
-	}
-
-	networkBuffer := 204800
-
-	timeoutConn := RichConn{
-		conn,
-		timeout,
-	}
-	client.Conn = &timeoutConn
-	client.connRW = bufio.NewReadWriter(bufio.NewReaderSize(&timeoutConn, networkBuffer), bufio.NewWriterSize(&timeoutConn, networkBuffer))
-
-	headers := make(map[string]string)
-	//headers["Require"] = "implicit-play"
-	// An OPTIONS request returns the request types the server will accept.
-	resp, err := client.Request("OPTIONS", headers)
-	if err != nil {
-		if resp != nil {
-			Authorization, _ := client.checkAuth("OPTIONS", resp)
-			if len(Authorization) > 0 {
-				headers := make(map[string]string)
-				headers["Require"] = "implicit-play"
-				headers["Authorization"] = Authorization
-				// An OPTIONS request returns the request types the server will accept.
-				resp, err = client.Request("OPTIONS", headers)
-			}
-			if err != nil {
-				return err
-			}
-		} else {
-			return err
-		}
-	}
-
-	// A DESCRIBE request includes an RTSP URL (rtsp://...), and the type of reply data that can be handled. This reply includes the presentation description,
-	// typically in Session Description Protocol (SDP) format. Among other things, the presentation description lists the media streams controlled with the aggregate URL.
-	// In the typical case, there is one media stream each for audio and video.
-	headers = make(map[string]string)
-	headers["Accept"] = "application/sdp"
-	resp, err = client.Request("DESCRIBE", headers)
-	if err != nil {
-		if resp != nil {
-			authorization, _ := client.checkAuth("DESCRIBE", resp)
-			if len(authorization) > 0 {
-				headers := make(map[string]string)
-				headers["Authorization"] = authorization
-				headers["Accept"] = "application/sdp"
-				resp, err = client.Request("DESCRIBE", headers)
-			}
-			if err != nil {
-				return err
-			}
-		} else {
-			return err
-		}
-	}
-	client.SDPRaw = resp.Body
-	client.SDPMap = ParseSDP(client.SDPRaw)
-	client.VSdp, client.HasVideo = client.SDPMap["video"]
-	client.ASdp, client.HasAudio = client.SDPMap["audio"]
-	session := ""
-	otherChannel := 4
-	for t, sdpInfo := range client.SDPMap {
-		headers = make(map[string]string)
-		if session != "" {
-			headers["Session"] = session
-		}
-		var _url = sdpInfo.Control
-		if !strings.HasPrefix(strings.ToLower(sdpInfo.Control), "rtsp://") {
-			_url = strings.TrimRight(client.URL, "/") + "/" + strings.TrimLeft(sdpInfo.Control, "/")
-		}
-		switch t {
-		case "video":
-			client.setVideoTrack()
-			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":
-			client.setAudioTrack()
-			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
-			}
-		}
-		if resp, err = client.RequestWithPath("SETUP", _url, headers, true); err != nil {
-			return err
-		}
-		session, _ = resp.Header["Session"].(string)
-		session = strings.Split(session, ";")[0]
-	}
-	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()
-	//loggerTime := time.Now().Add(-10 * time.Second)
-	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 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)
-			return
-		}
-		switch b {
-		case 0x24: // rtp
-			header := make([]byte, 4)
-			header[0] = b
-			_, err := io.ReadFull(client.connRW, header[1:])
-			if err != nil {
-				Printf("io.ReadFull err:%v", err)
-				return
-			}
-			channel := int(header[1])
-			length := binary.BigEndian.Uint16(header[2:])
-			content := make([]byte, length)
-			_, err = io.ReadFull(client.connRW, content)
-			if err != nil {
-				Printf("io.ReadFull err:%v", err)
-				return
-			}
-			
-			switch channel {
-			case client.aRTPChannel:
-				client.RtpAudio.Push(content)
-			case client.aRTPControlChannel:
-
-			case client.vRTPChannel:
-				client.RtpVideo.Push(content)
-			case client.vRTPControlChannel:
-
-			default:
-				Printf("unknow rtp pack type, channel:%v", channel)
-				continue
-			}
-
-			//if client.debugLogEnable {
-			//	rtp := ParseRTP(pack.Buffer)
-			//	if rtp != nil {
-			//		rtpSN := uint16(rtp.SequenceNumber)
-			//		if client.lastRtpSN != 0 && client.lastRtpSN+1 != rtpSN {
-			//			Printf("%s, %d packets lost, current SN=%d, last SN=%d\n", client.String(), rtpSN-client.lastRtpSN, rtpSN, client.lastRtpSN)
-			//		}
-			//		client.lastRtpSN = rtpSN
-			//	}
-			//
-			//	elapsed := time.Now().Sub(loggerTime)
-			//	if elapsed >= 30*time.Second {
-			//		Printf("%v read rtp frame.", client)
-			//		loggerTime = time.Now()
-			//	}
-			//}
-
-			client.InBytes += int(length + 4)
-
-		default: // rtsp
-			builder := bytes.Buffer{}
-			builder.WriteByte(b)
-			contentLen := 0
-			for client.Err() == nil {
-				line, prefix, err := client.connRW.ReadLine()
-				if err != nil {
-					Printf("client.connRW.ReadLine err:%v", err)
-					return
-				}
-				if len(line) == 0 {
-					if contentLen != 0 {
-						content := make([]byte, contentLen)
-						_, err = io.ReadFull(client.connRW, content)
-						if err != nil {
-							err = fmt.Errorf("Read content err.ContentLength:%d", contentLen)
-							return
-						}
-						builder.Write(content)
-					}
-					Printf("<<<[IN]\n%s", builder.String())
-					break
-				}
-				s := string(line)
-				builder.Write(line)
-				if !prefix {
-					builder.WriteString("\r\n")
-				}
-
-				if strings.Index(s, "Content-Length:") == 0 {
-					splits := strings.Split(s, ":")
-					contentLen, err = strconv.Atoi(strings.TrimSpace(splits[1]))
-					if err != nil {
-						Printf("strconv.Atoi err:%v, str:%v", err, splits[1])
-						return
-					}
-				}
-			}
-		}
-	}
-}
-
-func (client *RTSP) Request(method string, headers map[string]string) (*Response, error) {
-	l, err := url.Parse(client.URL)
-	if err != nil {
-		return nil, fmt.Errorf("Url parse error:%v", err)
-	}
-	l.User = nil
-	return client.RequestWithPath(method, l.String(), headers, true)
-}
-
-func (client *RTSP) RequestNoResp(method string, headers map[string]string) (err error) {
-	l, err := url.Parse(client.URL)
-	if err != nil {
-		return fmt.Errorf("Url parse error:%v", err)
-	}
-	l.User = nil
-	if _, err = client.RequestWithPath(method, l.String(), headers, false); err != nil {
+	// connect to the server
+	if err = p.Client.Start(u.Scheme, u.Host); err != nil {
 		return err
 	}
+	p.SetIO(p.Client)
 	return nil
 }

-func (client *RTSP) RequestWithPath(method string, path string, headers map[string]string, needResp bool) (resp *Response, err error) {
-	headers["User-Agent"] = client.Agent
-	if len(headers["Authorization"]) == 0 {
-		if len(client.authLine) != 0 {
-			Authorization, _ := DigestAuth(client.authLine, method, client.URL)
-			if len(Authorization) > 0 {
-				headers["Authorization"] = Authorization
-			}
-		}
-	}
-	if len(client.Session) > 0 {
-		headers["Session"] = client.Session
-	}
-	client.Seq++
-	cseq := client.Seq
-	builder := bytes.Buffer{}
-	builder.WriteString(fmt.Sprintf("%s %s RTSP/1.0\r\n", method, path))
-	builder.WriteString(fmt.Sprintf("CSeq: %d\r\n", cseq))
-	for k, v := range headers {
-		builder.WriteString(fmt.Sprintf("%s: %s\r\n", k, v))
-	}
-	builder.WriteString(fmt.Sprintf("\r\n"))
-	s := builder.String()
-	Printf("[OUT]>>>\n%s", s)
-	_, err = client.connRW.WriteString(s)
-	if err != nil {
+func (p *RTSPPuller) Pull() (err error) {
+	u, _ := url.Parse(p.RemoteURL)
+	if _, err = p.Options(u); err != nil {
+		p.Error("Options", zap.Error(err))
 		return
 	}
-	client.connRW.Flush()
-
-	if !needResp {
-		return nil, nil
+	// find published tracks
+	tracks, baseURL, _, err := p.Describe(u)
+	if err != nil {
+		p.Error("Describe", zap.Error(err))
+		return
 	}
-	lineCount := 0
-	statusCode := 200
-	status := ""
-	sid := ""
-	contentLen := 0
-	respHeader := make(map[string]interface{})
-	var line []byte
-	builder.Reset()
-	for {
-		isPrefix := false
-		if line, isPrefix, err = client.connRW.ReadLine(); err != nil {
-			return
-		}
-		s := string(line)
-		builder.Write(line)
-		if !isPrefix {
-			builder.WriteString("\r\n")
-		}
-		if len(line) == 0 {
-			body := ""
-			if contentLen > 0 {
-				content := make([]byte, contentLen)
-				_, err = io.ReadFull(client.connRW, content)
-				if err != nil {
-					err = fmt.Errorf("Read content err.ContentLength:%d", contentLen)
-					return
-				}
-				body = string(content)
-				builder.Write(content)
-			}
-			resp = NewResponse(statusCode, status, strconv.Itoa(cseq), sid, body)
-			resp.Header = respHeader
-			Printf("<<<[IN]\n%s", builder.String())
-
-			if !(statusCode >= 200 && statusCode <= 300) {
-				err = fmt.Errorf("Response StatusCode is :%d", statusCode)
-				return
-			}
-			return
-		}
-		if lineCount == 0 {
-			splits := strings.Split(s, " ")
-			if len(splits) < 3 {
-				err = fmt.Errorf("StatusCode Line error:%s", s)
-				return
-			}
-			statusCode, err = strconv.Atoi(splits[1])
-			if err != nil {
-				return
-			}
-			status = splits[2]
-		}
-		lineCount++
-		splits := strings.Split(s, ":")
-		if len(splits) == 2 {
-			if val, ok := respHeader[splits[0]]; ok {
-				if slice, ok2 := val.([]string); ok2 {
-					slice = append(slice, strings.TrimSpace(splits[1]))
-					respHeader[splits[0]] = slice
-				} else {
-					str, _ := val.(string)
-					slice := []string{str, strings.TrimSpace(splits[1])}
-					respHeader[splits[0]] = slice
-				}
-			} else {
-				respHeader[splits[0]] = strings.TrimSpace(splits[1])
-			}
-		}
-		if strings.Index(s, "Session:") == 0 {
-			splits := strings.Split(s, ":")
-			sid = strings.TrimSpace(splits[1])
-		}
-		//if strings.Index(s, "CSeq:") == 0 {
-		//	splits := strings.Split(s, ":")
-		//	cseq, err = strconv.Atoi(strings.TrimSpace(splits[1]))
-		//	if err != nil {
-		//		err = fmt.Errorf("Atoi CSeq err. line:%s", s)
-		//		return
-		//	}
-		//}
-		if strings.Index(s, "Content-Length:") == 0 {
-			splits := strings.Split(s, ":")
-			contentLen, err = strconv.Atoi(strings.TrimSpace(splits[1]))
-			if err != nil {
-				return
-			}
-		}
-
+	p.tracks = tracks
+	err = p.SetTracks()
+	if err != nil {
+		p.Error("SetTracks", zap.Error(err))
+		return
 	}
+	if err = p.SetupAll(tracks, baseURL); err != nil {
+		p.Error("SetupAndPlay", zap.Error(err))
+		return
+	}
+	p.OnPacketRTPAny(p.OnPacket)
+	_, err = p.Play(nil)
+	if err != nil {
+		p.Error("Play", zap.Error(err))
+		return
+	}
+	return p.Wait()
+}
+
+type RTSPPusher struct {
+	RTSPSubscriber
+	engine.Pusher
+	RTSPClient
+}
+
+func (p *RTSPPusher) OnEvent(event any) {
+	switch v := event.(type) {
+	case engine.VideoRTP:
+		p.Client.WritePacketRTP(p.videoTrack, v.Packet)
+	case engine.AudioRTP:
+		p.Client.WritePacketRTP(p.audioTrack, v.Packet)
+	default:
+		p.RTSPSubscriber.OnEvent(event)
+	}
+}
+
+func (p *RTSPPusher) Connect() error {
+	p.Client = &gortsplib.Client{
+		DialContext:      p.DialContext,
+		WriteBufferCount: rtspConfig.WriteBufferCount,
+	}
+	// parse URL
+	u, err := url.Parse(p.RemoteURL)
+	if err != nil {
+		p.Error("url.Parse", zap.Error(err))
+		return err
+	}
+	// connect to the server
+	if err = p.Client.Start(u.Scheme, u.Host); err != nil {
+		p.Error("Client.Start", zap.Error(err))
+		return err
+	}
+	p.SetIO(p.Client)
+	_, err = p.Client.Options(u)
+	return err
+}
+func (p *RTSPPusher) Push() (err error) {
+	var u *url.URL
+	u, err = url.Parse(p.RemoteURL)
+	// startTime := time.Now()
+	// for len(p.tracks) < 2 {
+	// 	if time.Sleep(time.Second); time.Since(startTime) > time.Second*10 {
+	// 		return fmt.Errorf("timeout")
+	// 	}
+	// }
+	if _, err = p.Announce(u, p.tracks); err != nil {
+		p.Error("Announce", zap.Error(err))
+		return
+	}
+	err = p.SetupAll(p.tracks, u)
+	if err != nil {
+		p.Error("Setup", zap.Error(err))
+		return
+	}
+
+	if _, err = p.Record(); err != nil {
+		p.Error("Record", zap.Error(err))
+		return
+	}
+	p.PlayRTP()
 	return
 }
--- a/go.mod
+++ b/go.mod
@@ -1,11 +1,50 @@
-module github.com/Monibuca/plugin-rtsp/v3
+module m7s.live/plugin/rtsp/v4

-go 1.16
+go 1.18

 require (
-	github.com/Monibuca/engine/v3 v3.0.0-beta5
-	github.com/Monibuca/utils/v3 v3.0.0-beta
-	github.com/pion/rtp v1.6.5
-	github.com/teris-io/shortid v0.0.0-20201117134242-e59966efd125
-	golang.org/x/sys v0.0.0-20210611083646-a4fc73990273 // indirect
+	github.com/bluenviron/gortsplib/v3 v3.10.0
+	github.com/bluenviron/mediacommon v1.0.0
+	github.com/pion/rtp v1.8.1
+	go.uber.org/zap v1.24.0
+	m7s.live/engine/v4 v4.13.7
+)
+
+require (
+	github.com/aler9/gortsplib/v2 v2.2.2 // indirect
+	github.com/cnotch/ipchub v1.1.0 // indirect
+	github.com/denisbrodbeck/machineid v1.0.1 // indirect
+	github.com/go-ole/go-ole v1.2.6 // indirect
+	github.com/go-task/slim-sprig v0.0.0-20230315185526-52ccab3ef572 // indirect
+	github.com/golang/mock v1.6.0 // indirect
+	github.com/google/pprof v0.0.0-20210407192527-94a9f03dee38 // indirect
+	github.com/google/uuid v1.3.0 // indirect
+	github.com/logrusorgru/aurora v2.0.3+incompatible // indirect
+	github.com/lufia/plan9stats v0.0.0-20220913051719-115f729f3c8c // indirect
+	github.com/mcuadros/go-defaults v1.2.0 // indirect
+	github.com/onsi/ginkgo/v2 v2.9.5 // indirect
+	github.com/pion/randutil v0.1.0 // indirect
+	github.com/pion/rtcp v1.2.10 // indirect
+	github.com/pion/sdp/v3 v3.0.6 // indirect
+	github.com/pion/webrtc/v3 v3.2.14 // indirect
+	github.com/power-devops/perfstat v0.0.0-20220216144756-c35f1ee13d7c // indirect
+	github.com/q191201771/naza v0.30.8 // indirect
+	github.com/quic-go/qtls-go1-20 v0.3.1 // indirect
+	github.com/quic-go/quic-go v0.37.4 // indirect
+	github.com/shirou/gopsutil/v3 v3.23.7 // indirect
+	github.com/shoenig/go-m1cpu v0.1.6 // indirect
+	github.com/tklauser/go-sysconf v0.3.11 // indirect
+	github.com/tklauser/numcpus v0.6.0 // indirect
+	github.com/yapingcat/gomedia v0.0.0-20230727105416-c491e66c9d2a // indirect
+	github.com/yusufpapurcu/wmi v1.2.3 // indirect
+	go.uber.org/atomic v1.10.0 // indirect
+	go.uber.org/multierr v1.8.0 // indirect
+	golang.org/x/crypto v0.12.0 // indirect
+	golang.org/x/exp v0.0.0-20221205204356-47842c84f3db // indirect
+	golang.org/x/mod v0.10.0 // indirect
+	golang.org/x/net v0.14.0 // indirect
+	golang.org/x/sync v0.2.0 // indirect
+	golang.org/x/sys v0.11.0 // indirect
+	golang.org/x/tools v0.9.1 // indirect
+	gopkg.in/yaml.v3 v3.0.1 // indirect
 )
--- a/go.sum
+++ b/go.sum
@@ -1,32 +1,310 @@
-github.com/BurntSushi/toml v0.3.1 h1:WXkYYl6Yr3qBf1K79EBnL4mak0OimBfB0XUf9Vl28OQ=
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=
-github.com/Monibuca/engine/v3 v3.0.0-beta5 h1:b27ZQDfvf5dBMZbCSIUXItUwVIFs95fpkAV4xjN7BNE=
-github.com/Monibuca/engine/v3 v3.0.0-beta5/go.mod h1:SMgnlwih4pBA/HkTLjKXZFYkv3ukRzFjv65CARRLVIk=
-github.com/Monibuca/utils/v3 v3.0.0-beta h1:z4p/BSH5J9Ja/gwoDmj1RyN+b0q28Nmn/fqXiwq2hGY=
-github.com/Monibuca/utils/v3 v3.0.0-beta/go.mod h1:mQYP/OMox1tkWP6Qut7pBfARr1TXSRkK662dexQl6kI=
-github.com/funny/slab v0.0.0-20180511031532-b1fad5e5d478 h1:Db9StoJ6RZN3YttC0Pm0I4Y5izITRYch3RMbT59BYN0=
-github.com/funny/slab v0.0.0-20180511031532-b1fad5e5d478/go.mod h1:0j1+svBH8ABEIPdUP0AIg4qedsybnXGJBakCEw8cfoo=
-github.com/funny/utest v0.0.0-20161029064919-43870a374500 h1:Z0r1CZnoIWFB/Uiwh1BU5FYmuFe6L5NPi6XWQEmsTRg=
-github.com/funny/utest v0.0.0-20161029064919-43870a374500/go.mod h1:mUn39tBov9jKnTWV1RlOYoNzxdBFHiSzXWdY1FoNGGg=
+github.com/aler9/gortsplib/v2 v2.2.2 h1:tTw8pdKSOEjlZjjE1S4ftXPHJkYOqjNNv3hjQ0Nto9M=
+github.com/aler9/gortsplib/v2 v2.2.2/go.mod h1:k6uBVHGwsIc/0L5SLLqWwi6bSJUb4VR0HfvncyHlKQI=
+github.com/benbjohnson/clock v1.1.0 h1:Q92kusRqC1XV2MjkWETPvjJVqKetz1OzxZB7mHJLju8=
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+gopkg.in/yaml.v2 v2.2.1/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
+gopkg.in/yaml.v2 v2.2.2/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
+gopkg.in/yaml.v2 v2.2.4/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
+gopkg.in/yaml.v2 v2.3.0/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=
+gopkg.in/yaml.v2 v2.4.0/go.mod h1:RDklbk79AGWmwhnvt/jBztapEOGDOx6ZbXqjP6csGnQ=
+gopkg.in/yaml.v3 v3.0.0-20200313102051-9f266ea9e77c/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=
+gopkg.in/yaml.v3 v3.0.0-20210107192922-496545a6307b/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=
+gopkg.in/yaml.v3 v3.0.1 h1:fxVm/GzAzEWqLHuvctI91KS9hhNmmWOoWu0XTYJS7CA=
+gopkg.in/yaml.v3 v3.0.1/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=
+m7s.live/engine/v4 v4.13.7 h1:DDU3U7JQkgzNLWeTNlbeG7Y5wP0+WHZgtxjWupjmods=
+m7s.live/engine/v4 v4.13.7/go.mod h1:LoALBfV5rmsz5TJQr6cmLxM33mfUE5BKBq/sMtXOVlc=
--- a/main.go
+++ b/main.go
@@ -1,230 +1,103 @@
 package rtsp

 import (
-	"bufio"
-	"fmt"
-	"log"
-	"net"
 	"net/http"
+	"strconv"
 	"sync"
-	"time"

-	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
-	"github.com/teris-io/shortid"
+	"github.com/bluenviron/gortsplib/v3"
+	"go.uber.org/zap"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/config"
+	"m7s.live/engine/v4/util"
 )

-var collection sync.Map
-var config = struct {
-	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{
-		Name:   "RTSP",
-		Config: &config,
-		Run:    runPlugin,
-		HotConfig: map[string]func(interface{}){
-			"AutoPull": func(value interface{}) {
-				config.AutoPull = value.(bool)
-			},
-		},
-	})
+type RTSPConfig struct {
+	config.Publish
+	config.Subscribe
+	config.Pull
+	config.Push
+	ListenAddr       string `default:":554"`
+	UDPAddr          string `default:":8000"`
+	RTCPAddr         string `default:":8001"`
+	ReadBufferCount  int    `default:"2048"`
+	WriteBufferCount int    `default:"2048"`
+	PullProtocol     string `default:"tcp"` //tcp、udp、auto
+	sync.Map
 }
-func runPlugin() {

-	http.HandleFunc("/api/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
-		sse := NewSSE(w, r.Context())
-		var err error
-		for tick := time.NewTicker(time.Second); err == nil; <-tick.C {
-			var info []*RTSP
-			collection.Range(func(key, value interface{}) bool {
-				rtsp := value.(*RTSP)
-				info = append(info, rtsp)
-				return true
-			})
-			err = sse.WriteJSON(info)
+func (conf *RTSPConfig) OnEvent(event any) {
+	switch v := event.(type) {
+	case FirstConfig:
+		s := &gortsplib.Server{
+			Handler:           conf,
+			RTSPAddress:       conf.ListenAddr,
+			UDPRTPAddress:     conf.UDPAddr,
+			UDPRTCPAddress:    conf.RTCPAddr,
+			MulticastIPRange:  "224.1.0.0/16",
+			MulticastRTPPort:  8002,
+			MulticastRTCPPort: 8003,
+			ReadBufferCount:   conf.ReadBufferCount,
+			WriteBufferCount:  conf.WriteBufferCount,
 		}
-	})
-	http.HandleFunc("/api/rtsp/pull", func(w http.ResponseWriter, r *http.Request) {
-		CORS(w, r)
-		targetURL := r.URL.Query().Get("target")
-		streamPath := r.URL.Query().Get("streamPath")
-		if err := new(RTSP).PullStream(streamPath, targetURL); err == nil {
-			w.Write([]byte(`{"code":0}`))
-		} else {
-			w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
+		if err := s.Start(); err != nil {
+			RTSPPlugin.Error("server start", zap.Error(err))
+			RTSPPlugin.Disabled = true
 		}
-	})
-	if len(config.AutoPullList) > 0 {
-		for _, info := range config.AutoPullList {
-			if err := new(RTSP).PullStream(info.StreamPath, info.URL); err != nil {
-				Println(err)
+		for streamPath, url := range conf.PullOnStart {
+			if err := RTSPPlugin.Pull(streamPath, url, new(RTSPPuller), 0); err != nil {
+				RTSPPlugin.Error("pull", zap.String("streamPath", streamPath), zap.String("url", url), zap.Error(err))
+			}
+		}
+	case SEpublish:
+		if url, ok := conf.PushList[v.Target.Path]; ok {
+			if err := RTSPPlugin.Push(v.Target.Path, url, new(RTSPPusher), false); err != nil {
+				RTSPPlugin.Error("push", zap.String("streamPath", v.Target.Path), zap.String("url", url), zap.Error(err))
+			}
+		}
+	case InvitePublish: //按需拉流
+		if url, ok := conf.PullOnSub[v.Target]; ok {
+			if err := RTSPPlugin.Pull(v.Target, url, new(RTSPPuller), 0); err != nil {
+				RTSPPlugin.Error("pull", zap.String("streamPath", v.Target), zap.String("url", url), zap.Error(err))
 			}
 		}
 	}
-	if config.ListenAddr != "" {
-		go log.Fatal(ListenRtsp(config.ListenAddr))
-	}
-	// AddHook(HOOK_SUBSCRIBE, func(value interface{}) {
-	// 	s := value.(*Subscriber)
-	// 	if config.AutoPull && s.Publisher == nil {
-	// 		new(RTSP).PullStream(s.StreamPath, strings.Replace(config.RemoteAddr, "${streamPath}", s.StreamPath, -1))
-	// 	}
-	// })
 }

-func ListenRtsp(addr string) error {
-	defer log.Println("rtsp server start!")
-	listener, err := net.Listen("tcp", addr)
+var rtspConfig = &RTSPConfig{}
+var RTSPPlugin = InstallPlugin(rtspConfig)
+
+func filterStreams() (ss []*Stream) {
+	Streams.Range(func(key string, s *Stream) {
+		switch s.Publisher.(type) {
+		case *RTSPPublisher, *RTSPPuller:
+			ss = append(ss, s)
+		}
+	})
+	return
+}
+
+func (*RTSPConfig) API_list(w http.ResponseWriter, r *http.Request) {
+	util.ReturnFetchValue(filterStreams, w, r)
+}
+
+func (*RTSPConfig) API_Pull(rw http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+	save, _ := strconv.Atoi(query.Get("save"))
+	err := RTSPPlugin.Pull(query.Get("streamPath"), query.Get("target"), new(RTSPPuller), save)
 	if err != nil {
-		return err
-	}
-	var tempDelay time.Duration
-	networkBuffer := 204800
-	timeoutMillis := config.Timeout
-	for {
-		conn, err := listener.Accept()
-		conn.(*net.TCPConn).SetNoDelay(false)
-		if err != nil {
-			if ne, ok := err.(net.Error); ok && ne.Temporary() {
-				if tempDelay == 0 {
-					tempDelay = 5 * time.Millisecond
-				} else {
-					tempDelay *= 2
-				}
-				if max := 1 * time.Second; tempDelay > max {
-					tempDelay = max
-				}
-				fmt.Printf("rtsp: Accept error: %v; retrying in %v", err, tempDelay)
-				time.Sleep(tempDelay)
-				continue
-			}
-			return err
-		}
-
-		tempDelay = 0
-		timeoutTCPConn := &RichConn{conn, time.Duration(timeoutMillis) * time.Millisecond}
-		go (&RTSP{
-			ID:                 shortid.MustGenerate(),
-			Conn:               timeoutTCPConn,
-			connRW:             bufio.NewReadWriter(bufio.NewReaderSize(timeoutTCPConn, networkBuffer), bufio.NewWriterSize(timeoutTCPConn, networkBuffer)),
-			Timeout:            config.Timeout,
-			vRTPChannel:        -1,
-			vRTPControlChannel: -1,
-			aRTPChannel:        -1,
-			aRTPControlChannel: -1,
-		}).AcceptPush()
-	}
-	return nil
-}
-
-type RTSP struct {
-	*Stream
-	URL      string
-	SDPRaw   string
-	InBytes  int
-	OutBytes int
-	RTSPClientInfo
-	ID        string
-	Conn      *RichConn `json:"-"`
-	connRW    *bufio.ReadWriter
-	connWLock sync.RWMutex
-	Type      SessionType
-	TransType TransType
-
-	SDPMap  map[string]*SDPInfo
-	nonce   string
-	ASdp    *SDPInfo
-	VSdp    *SDPInfo
-	Timeout int
-	//tcp channels
-	aRTPChannel        int
-	aRTPControlChannel int
-	vRTPChannel        int
-	vRTPControlChannel int
-	UDPServer          *UDPServer          `json:"-"`
-	UDPClient          *UDPClient          `json:"-"`
-	Auth               func(string) string `json:"-"`
-	HasVideo           bool
-	HasAudio           bool
-	RtpAudio           *RTPAudio
-	RtpVideo           *RTPVideo
-}
-
-func (rtsp *RTSP) setVideoTrack() {
-	if rtsp.VSdp.Codec == "H264" {
-		rtsp.RtpVideo = rtsp.NewRTPVideo(7)
-		if len(rtsp.VSdp.SpropParameterSets) > 1 {
-			rtsp.RtpVideo.PushNalu(0, 0, rtsp.VSdp.SpropParameterSets...)
-		}
-	} else if rtsp.VSdp.Codec == "H265" {
-		rtsp.RtpVideo = rtsp.NewRTPVideo(12)
-		if len(rtsp.VSdp.VPS) > 0 {
-			rtsp.RtpVideo.PushNalu(0, 0, rtsp.VSdp.VPS, rtsp.VSdp.SPS, rtsp.VSdp.PPS)
-		}
-	}
-}
-func (rtsp *RTSP) setAudioTrack() {
-	var at *RTPAudio
-	if len(rtsp.ASdp.Control) > 0 {
-		at = rtsp.NewRTPAudio(0)
-		at.SetASC(rtsp.ASdp.Config)
+		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
 	} else {
-		switch rtsp.ASdp.Codec {
-		case "AAC":
-			at = rtsp.NewRTPAudio(10)
-		case "PCMA":
-			at = rtsp.NewRTPAudio(7)
-			at.SoundRate = rtsp.ASdp.TimeScale
-			at.SoundSize = 16
-		case "PCMU":
-			at = rtsp.NewRTPAudio(8)
-			at.SoundRate = rtsp.ASdp.TimeScale
-			at.SoundSize = 16
-		default:
-			Printf("rtsp audio codec not support:%s", rtsp.ASdp.Codec)
-			return
-		}
+		util.ReturnOK(rw, r)
 	}
-	rtsp.RtpAudio = at
 }

-type RTSPClientInfo struct {
-	Agent    string
-	Session  string
-	authLine string
-	Seq      int
-}
-type RichConn struct {
-	net.Conn
-	timeout time.Duration
-}
-
-func (conn *RichConn) Read(b []byte) (n int, err error) {
-	if conn.timeout > 0 {
-		conn.Conn.SetReadDeadline(time.Now().Add(conn.timeout))
+func (*RTSPConfig) API_Push(rw http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+	err := RTSPPlugin.Push(query.Get("streamPath"), query.Get("target"), new(RTSPPusher), query.Has("save"))
+	if err != nil {
+		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
 	} else {
-		var t time.Time
-		conn.Conn.SetReadDeadline(t)
+		util.ReturnOK(rw, r)
 	}
-	return conn.Conn.Read(b)
-}
-
-func (conn *RichConn) Write(b []byte) (n int, err error) {
-	if conn.timeout > 0 {
-		conn.Conn.SetWriteDeadline(time.Now().Add(conn.timeout))
-	} else {
-		var t time.Time
-		conn.Conn.SetWriteDeadline(t)
-	}
-	return conn.Conn.Write(b)
 }
--- a/publisher.go
+++ b/publisher.go
@@ -0,0 +1,119 @@
+package rtsp
+
+import (
+	"strings"
+
+	"github.com/bluenviron/gortsplib/v3/pkg/formats"
+	"github.com/bluenviron/gortsplib/v3/pkg/media"
+	"github.com/bluenviron/mediacommon/pkg/codecs/mpeg4audio"
+	"github.com/pion/rtp"
+	"go.uber.org/zap"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/common"
+	. "m7s.live/engine/v4/track"
+)
+
+type RTSPPublisher struct {
+	Publisher
+	Tracks map[*media.Media]common.AVTrack `json:"-" yaml:"-"`
+	RTSPIO
+}
+
+func (p *RTSPPublisher) SetTracks() error {
+	p.Tracks = make(map[*media.Media]common.AVTrack, len(p.tracks))
+	defer func() {
+		for _, track := range p.Tracks {
+			p.Info("set track", zap.String("name", track.GetName()))
+		}
+	}()
+	for _, track := range p.tracks {
+		for _, forma := range track.Formats {
+			switch f := forma.(type) {
+			case *formats.H264:
+				vt := p.VideoTrack
+				if vt == nil {
+					vt = NewH264(p.Stream, f.PayloadType())
+					p.VideoTrack = vt
+				}
+				p.Tracks[track] = p.VideoTrack
+				if len(f.SPS) > 0 {
+					vt.WriteSliceBytes(f.SPS)
+				}
+				if len(f.PPS) > 0 {
+					vt.WriteSliceBytes(f.PPS)
+				}
+			case *formats.H265:
+				vt := p.VideoTrack
+				if vt == nil {
+					vt = NewH265(p.Stream, f.PayloadType())
+					p.VideoTrack = vt
+				}
+				p.Tracks[track] = p.VideoTrack
+				if len(f.VPS) > 0 {
+					vt.WriteSliceBytes(f.VPS)
+				}
+				if len(f.SPS) > 0 {
+					vt.WriteSliceBytes(f.SPS)
+				}
+				if len(f.PPS) > 0 {
+					vt.WriteSliceBytes(f.PPS)
+				}
+			case *formats.MPEG4Audio:
+				at := p.AudioTrack
+				if at == nil {
+					at := NewAAC(p.Stream, f.PayloadType(), uint32(f.Config.SampleRate))
+					at.IndexDeltaLength = f.IndexDeltaLength
+					at.IndexLength = f.IndexLength
+					at.SizeLength = f.SizeLength
+					if f.Config.Type == mpeg4audio.ObjectTypeAACLC {
+						at.Mode = 1
+					}
+					at.Channels = uint8(f.Config.ChannelCount)
+					asc, _ := f.Config.Marshal()
+					// 复用AVCC写入逻辑，解析出AAC的配置信息
+					at.WriteSequenceHead(append([]byte{0xAF, 0x00}, asc...))
+					p.AudioTrack = at
+				}
+				p.Tracks[track] = p.AudioTrack
+			case *formats.G711:
+				at := p.AudioTrack
+				if at == nil {
+					at := NewG711(p.Stream, !f.MULaw, f.PayloadType(), uint32(f.ClockRate()))
+					p.AudioTrack = at
+				}
+				p.Tracks[track] = p.AudioTrack
+			default:
+				rtpMap := strings.ToLower(forma.RTPMap())
+				if strings.Contains(rtpMap, "pcm") {
+					isMulaw := false
+					if strings.Contains(rtpMap, "pcmu") {
+						isMulaw = true
+					}
+					at := p.AudioTrack
+					if at == nil {
+						at := NewG711(p.Stream, !isMulaw, f.PayloadType(), uint32(f.ClockRate()))
+						p.AudioTrack = at
+					}
+					p.Tracks[track] = p.AudioTrack
+				} else {
+					p.Warn("unknown format", zap.Any("format", f.String()))
+				}
+			}
+		}
+	}
+	if p.VideoTrack == nil {
+		p.Config.PubVideo = false
+		p.Info("no video track")
+	}
+	if p.AudioTrack == nil {
+		p.Config.PubAudio = false
+		p.Info("no audio track")
+	}
+	return nil
+}
+
+func (p *RTSPPublisher) OnPacket(m *media.Media, f formats.Format, pack *rtp.Packet) {
+	if t, ok := p.Tracks[m]; ok {
+		t.WriteRTPPack(pack)
+	}
+}
--- a/request.go
+++ b/request.go
@@ -1,100 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"log"
-	"regexp"
-	"strconv"
-	"strings"
-)
-
-const (
-	RTSP_VERSION = "RTSP/1.0"
-)
-
-const (
-	// Client to server for presentation and stream objects; recommended
-	DESCRIBE = "DESCRIBE"
-	// Bidirectional for client and stream objects; optional
-	ANNOUNCE = "ANNOUNCE"
-	// Bidirectional for client and stream objects; optional
-	GET_PARAMETER = "GET_PARAMETER"
-	// Bidirectional for client and stream objects; required for Client to server, optional for server to client
-	OPTIONS = "OPTIONS"
-	// Client to server for presentation and stream objects; recommended
-	PAUSE = "PAUSE"
-	// Client to server for presentation and stream objects; required
-	PLAY = "PLAY"
-	// Client to server for presentation and stream objects; optional
-	RECORD = "RECORD"
-	// Server to client for presentation and stream objects; optional
-	REDIRECT = "REDIRECT"
-	// Client to server for stream objects; required
-	SETUP = "SETUP"
-	// Bidirectional for presentation and stream objects; optional
-	SET_PARAMETER = "SET_PARAMETER"
-	// Client to server for presentation and stream objects; required
-	TEARDOWN = "TEARDOWN"
-	DATA     = "DATA"
-)
-
-type Request struct {
-	Method  string
-	URL     string
-	Version string
-	Header  map[string]string
-	Content string
-	Body    string
-}
-
-func NewRequest(content string) *Request {
-	lines := strings.Split(strings.TrimSpace(content), "\r\n")
-	if len(lines) == 0 {
-		return nil
-	}
-	items := regexp.MustCompile("\\s+").Split(strings.TrimSpace(lines[0]), -1)
-	if len(items) < 3 {
-		return nil
-	}
-	if !strings.HasPrefix(items[2], "RTSP") {
-		log.Printf("invalid rtsp request, line[0] %s", lines[0])
-		return nil
-	}
-	header := make(map[string]string)
-	for i := 1; i < len(lines); i++ {
-		line := strings.TrimSpace(lines[i])
-		headerItems := regexp.MustCompile(":\\s+").Split(line, 2)
-		if len(headerItems) < 2 {
-			continue
-		}
-		header[headerItems[0]] = headerItems[1]
-	}
-	return &Request{
-		Method:  items[0],
-		URL:     items[1],
-		Version: items[2],
-		Header:  header,
-		Content: content,
-		Body:    "",
-	}
-}
-
-func (r *Request) String() string {
-	str := fmt.Sprintf("%s %s %s\r\n", r.Method, r.URL, r.Version)
-	for key, value := range r.Header {
-		str += fmt.Sprintf("%s: %s\r\n", key, value)
-	}
-	str += "\r\n"
-	str += r.Body
-	return str
-}
-
-func (r *Request) GetContentLength() int {
-	v, err := strconv.ParseInt(r.Header["Content-Length"], 10, 64)
-	if err != nil {
-		return 0
-	} else {
-		return int(v)
-	}
-}
-
--- a/response.go
+++ b/response.go
@@ -1,51 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"strconv"
-)
-
-type Response struct {
-	Version    string
-	StatusCode int
-	Status     string
-	Header     map[string]interface{}
-	Body       string
-}
-
-func NewResponse(statusCode int, status, cSeq, sid, body string) *Response {
-	res := &Response{
-		Version:    RTSP_VERSION,
-		StatusCode: statusCode,
-		Status:     status,
-		Header:     map[string]interface{}{"CSeq": cSeq, "Session": sid},
-		Body:       body,
-	}
-	len := len(body)
-	if len > 0 {
-		res.Header["Content-Length"] = strconv.Itoa(len)
-	} else {
-		delete(res.Header, "Content-Length")
-	}
-	return res
-}
-
-func (r *Response) String() string {
-	str := fmt.Sprintf("%s %d %s\r\n", r.Version, r.StatusCode, r.Status)
-	for key, value := range r.Header {
-		str += fmt.Sprintf("%s: %s\r\n", key, value)
-	}
-	str += "\r\n"
-	str += r.Body
-	return str
-}
-
-func (r *Response) SetBody(body string) {
-	len := len(body)
-	r.Body = body
-	if len > 0 {
-		r.Header["Content-Length"] = strconv.Itoa(len)
-	} else {
-		delete(r.Header, "Content-Length")
-	}
-}
--- a/rtp-parser.go
+++ b/rtp-parser.go
@@ -1,68 +0,0 @@
-package rtsp
-
-import (
-	"encoding/binary"
-)
-
-const (
-	RTP_FIXED_HEADER_LENGTH = 12
-)
-
-type RTPInfo struct {
-	Version        int
-	Padding        bool
-	Extension      bool
-	CSRCCnt        int
-	Marker         bool
-	PayloadType    int
-	SequenceNumber int
-	Timestamp      int
-	SSRC           int
-	Payload        []byte
-	PayloadOffset  int
-}
-
-func ParseRTP(rtpBytes []byte) *RTPInfo {
-	if len(rtpBytes) < RTP_FIXED_HEADER_LENGTH {
-		return nil
-	}
-	firstByte := rtpBytes[0]
-	secondByte := rtpBytes[1]
-	info := &RTPInfo{
-		Version:   int(firstByte >> 6),
-		Padding:   (firstByte>>5)&1 == 1,
-		Extension: (firstByte>>4)&1 == 1,
-		CSRCCnt:   int(firstByte & 0x0f),
-
-		Marker:         secondByte>>7 == 1,
-		PayloadType:    int(secondByte & 0x7f),
-		SequenceNumber: int(binary.BigEndian.Uint16(rtpBytes[2:])),
-		Timestamp:      int(binary.BigEndian.Uint32(rtpBytes[4:])),
-		SSRC:           int(binary.BigEndian.Uint32(rtpBytes[8:])),
-	}
-	offset := RTP_FIXED_HEADER_LENGTH
-	end := len(rtpBytes)
-	if end-offset >= 4*info.CSRCCnt {
-		offset += 4 * info.CSRCCnt
-	}
-	if info.Extension && end-offset >= 4 {
-		extLen := 4 * int(binary.BigEndian.Uint16(rtpBytes[offset+2:]))
-		offset += 4
-		if end-offset >= extLen {
-			offset += extLen
-		}
-	}
-	if info.Padding && end-offset > 0 {
-		paddingLen := int(rtpBytes[end-1])
-		if end-offset >= paddingLen {
-			end -= paddingLen
-		}
-	}
-	info.Payload = rtpBytes[offset:end]
-	info.PayloadOffset = offset
-	if end-offset < 1 {
-		return nil
-	}
-
-	return info
-}
--- a/sdp-parser.go
+++ b/sdp-parser.go
@@ -1,109 +0,0 @@
-package rtsp
-
-import (
-	"encoding/base64"
-	"encoding/hex"
-	"strconv"
-	"strings"
-)
-
-type SDPInfo struct {
-	AVType             string
-	Codec              string
-	TimeScale          int
-	Control            string
-	Rtpmap             int
-	Config             []byte
-	SpropParameterSets [][]byte
-	VPS                []byte
-	PPS                []byte
-	SPS                []byte
-	PayloadType        int
-	SizeLength         int
-	IndexLength        int
-}
-
-func ParseSDP(sdpRaw string) map[string]*SDPInfo {
-	sdpMap := make(map[string]*SDPInfo)
-	var info *SDPInfo
-	for _, line := range strings.Split(sdpRaw, "\n") {
-		line = strings.TrimSpace(line)
-		typeval := strings.SplitN(line, "=", 2)
-		if len(typeval) == 2 {
-			fields := strings.SplitN(typeval[1], " ", 2)
-			switch typeval[0] {
-			case "m":
-				if len(fields) > 0 {
-					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 {
-						keyval := strings.SplitN(field, ":", 2)
-						if len(keyval) >= 2 {
-							key := keyval[0]
-							val := keyval[1]
-							switch key {
-							case "control":
-								info.Control = val
-							case "rtpmap":
-								info.Rtpmap, _ = strconv.Atoi(val)
-							}
-						}
-						keyval = strings.Split(field, "/")
-						if len(keyval) >= 2 {
-							switch keyval[0] {
-							case "H264", "H265", "PCMA", "PCMU":
-								info.Codec = keyval[0]
-							case "HEVC":
-								info.Codec = "H265"
-							case "MPEG4-GENERIC":
-								info.Codec = "AAC"
-							}
-							if i, err := strconv.Atoi(keyval[1]); err == nil {
-								info.TimeScale = i
-							}
-						}
-						keyval = strings.Split(field, ";")
-						if len(keyval) > 1 {
-							for _, field := range keyval {
-								keyval := strings.SplitN(field, "=", 2)
-								if len(keyval) == 2 {
-									key := strings.TrimSpace(keyval[0])
-									val := keyval[1]
-									switch key {
-									case "config":
-										info.Config, _ = hex.DecodeString(val)
-									case "sizelength":
-										info.SizeLength, _ = strconv.Atoi(val)
-									case "indexlength":
-										info.IndexLength, _ = strconv.Atoi(val)
-									case "sprop-vps":
-										info.VPS, _ = base64.StdEncoding.DecodeString(val)
-									case "sprop-sps":
-										info.SPS, _ = base64.StdEncoding.DecodeString(val)
-									case "sprop-pps":
-										info.PPS, _ = base64.StdEncoding.DecodeString(val)
-									case "sprop-parameter-sets":
-										fields := strings.Split(val, ",")
-										for _, field := range fields {
-											val, _ := base64.StdEncoding.DecodeString(field)
-											info.SpropParameterSets = append(info.SpropParameterSets, val)
-										}
-									}
-								}
-							}
-						}
-					}
-				}
-
-			}
-		}
-	}
-	return sdpMap
-}
--- a/server.go
+++ b/server.go
@@ -0,0 +1,120 @@
+package rtsp
+
+import (
+	"github.com/bluenviron/gortsplib/v3"
+	"github.com/bluenviron/gortsplib/v3/pkg/base"
+	"github.com/bluenviron/gortsplib/v3/pkg/media"
+	"go.uber.org/zap"
+	. "m7s.live/engine/v4"
+)
+
+type RTSPIO struct {
+	tracks     media.Medias
+	stream     *gortsplib.ServerStream
+	audioTrack *media.Media
+	videoTrack *media.Media
+}
+
+func (conf *RTSPConfig) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	RTSPPlugin.Debug("conn opened")
+}
+
+func (conf *RTSPConfig) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
+	RTSPPlugin.Debug("conn closed")
+	if p, ok := conf.LoadAndDelete(ctx.Conn); ok {
+		p.(IIO).Stop(zap.String("conn", "closed"))
+	}
+}
+
+func (conf *RTSPConfig) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	RTSPPlugin.Debug("session opened")
+}
+
+func (conf *RTSPConfig) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
+	RTSPPlugin.Debug("session closed")
+	// if p, ok := conf.LoadAndDelete(ctx.Session); ok {
+	// 	p.(IIO).Stop()
+	// }
+}
+
+// called after receiving a DESCRIBE request.
+func (conf *RTSPConfig) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
+	RTSPPlugin.Debug("describe request")
+	var suber RTSPSubscriber
+	suber.RemoteAddr = ctx.Conn.NetConn().RemoteAddr().String()
+	suber.SetIO(ctx.Conn.NetConn())
+	streamPath := ctx.Path
+	if ctx.Query != "" {
+		streamPath = streamPath + "?" + ctx.Query
+	}
+	if err := RTSPPlugin.Subscribe(streamPath, &suber); err == nil {
+		RTSPPlugin.Debug("describe replay ok")
+		conf.Store(ctx.Conn, &suber)
+		return &base.Response{
+			StatusCode: base.StatusOK,
+		}, suber.stream, nil
+	} else {
+		return &base.Response{
+			StatusCode: base.StatusNotFound,
+		}, suber.stream, nil
+	}
+}
+
+func (conf *RTSPConfig) OnSetup(ctx *gortsplib.ServerHandlerOnSetupCtx) (*base.Response, *gortsplib.ServerStream, error) {
+	var resp base.Response
+	resp.StatusCode = base.StatusOK
+	if p, ok := conf.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPSubscriber:
+			return &resp, v.stream, nil
+		case *RTSPPublisher:
+			return &resp, v.stream, nil
+		}
+	}
+	resp.StatusCode = base.StatusNotFound
+	return &resp, nil, nil
+}
+
+func (conf *RTSPConfig) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
+	var resp base.Response
+	resp.StatusCode = base.StatusNotFound
+	if p, ok := conf.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPSubscriber:
+			resp.StatusCode = base.StatusOK
+			go func() {
+				v.PlayRTP()
+				ctx.Session.Close()
+			}()
+		}
+	}
+	return &resp, nil
+}
+func (conf *RTSPConfig) OnRecord(ctx *gortsplib.ServerHandlerOnRecordCtx) (*base.Response, error) {
+	if p, ok := conf.Load(ctx.Session); ok {
+		ctx.Session.OnPacketRTPAny(p.(*RTSPPublisher).OnPacket)
+	}
+	return &base.Response{
+		StatusCode: base.StatusOK,
+	}, nil
+}
+func (conf *RTSPConfig) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+	p := &RTSPPublisher{}
+	p.SetIO(ctx.Conn.NetConn())
+	if err := RTSPPlugin.Publish(ctx.Path, p); err == nil {
+		p.tracks = ctx.Medias
+		p.stream = gortsplib.NewServerStream(ctx.Medias)
+		if err = p.SetTracks(); err != nil {
+			return nil, err
+		}
+		conf.Store(ctx.Conn, p)
+		conf.Store(ctx.Session, p)
+	} else {
+		return &base.Response{
+			StatusCode: base.StatusBadRequest,
+		}, nil
+	}
+	return &base.Response{
+		StatusCode: base.StatusOK,
+	}, nil
+}
--- a/session.go
+++ b/session.go
@@ -1,611 +0,0 @@
-package rtsp
-
-import (
-	"bytes"
-	"crypto/md5"
-	"encoding/binary"
-	"fmt"
-	"io"
-	"net/url"
-	"regexp"
-	"strconv"
-	"strings"
-	"time"
-
-	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
-	"github.com/pion/rtp"
-	"github.com/teris-io/shortid"
-)
-
-type RTPPack struct {
-	Type RTPType
-	rtp.Packet
-}
-type SessionType int
-
-const (
-	SESSION_TYPE_PUSHER SessionType = iota
-	SESSEION_TYPE_PLAYER
-)
-
-func (st SessionType) String() string {
-	switch st {
-	case SESSION_TYPE_PUSHER:
-		return "pusher"
-	case SESSEION_TYPE_PLAYER:
-		return "player"
-	}
-	return "unknow"
-}
-
-type RTPType int
-
-const (
-	RTP_TYPE_AUDIO RTPType = iota
-	RTP_TYPE_VIDEO
-	RTP_TYPE_AUDIOCONTROL
-	RTP_TYPE_VIDEOCONTROL
-)
-
-type TransType int
-
-const (
-	TRANS_TYPE_TCP TransType = iota
-	TRANS_TYPE_UDP
-)
-
-func (tt TransType) String() string {
-	switch tt {
-	case TRANS_TYPE_TCP:
-		return "TCP"
-	case TRANS_TYPE_UDP:
-		return "UDP"
-	}
-	return "unknow"
-}
-
-const UDP_BUF_SIZE = 1048576
-
-func (session *RTSP) SessionString() string {
-	return fmt.Sprintf("session[%v][%v][%s][%s][%s]", session.Type, session.TransType, session.StreamPath, session.ID, session.Conn.RemoteAddr().String())
-}
-
-func (session *RTSP) Stop() {
-	if session.Conn != nil {
-		session.connRW.Flush()
-		session.Conn.Close()
-		session.Conn = nil
-	}
-	if session.UDPClient != nil {
-		session.UDPClient.Stop()
-		session.UDPClient = nil
-	}
-	if session.UDPServer != nil {
-		session.UDPServer.Stop()
-		session.UDPServer = nil
-	}
-	session.Close()
-	if session.Stream != nil {
-		collection.Delete(session.StreamPath)
-	}
-}
-
-// AcceptPush 接受推流
-func (session *RTSP) AcceptPush() {
-	defer session.Stop()
-	buf2 := make([]byte, 2)
-	timer := time.Unix(0, 0)
-	for {
-		buf1, err := session.connRW.ReadByte()
-		if err != nil {
-			Println(err)
-			return
-		}
-		if buf1 == 0x24 { //rtp data
-			if buf1, err = session.connRW.ReadByte(); err != nil {
-				Println(err)
-				return
-			}
-			if _, err := io.ReadFull(session.connRW, buf2); err != nil {
-				Println(err)
-				return
-			}
-			channel := int(buf1)
-			rtpLen := int(binary.BigEndian.Uint16(buf2))
-			rtpBytes := make([]byte, rtpLen)
-			if _, err := io.ReadFull(session.connRW, rtpBytes); err != nil {
-				Println(err)
-				return
-			}
-
-			// t := pack.Timestamp / 90
-			switch channel {
-			case session.aRTPChannel:
-				// pack.Type = RTP_TYPE_AUDIO
-				elapsed := time.Since(timer)
-				if elapsed >= 30*time.Second {
-					Println("Recv an audio RTP package")
-					timer = time.Now()
-				}
-				session.RtpAudio.Push(rtpBytes)
-			case session.aRTPControlChannel:
-				// pack.Type = RTP_TYPE_AUDIOCONTROL
-			case session.vRTPChannel:
-				// pack.Type = RTP_TYPE_VIDEO
-				elapsed := time.Since(timer)
-				if elapsed >= 30*time.Second {
-					Println("Recv an video RTP package")
-					timer = time.Now()
-				}
-				session.RtpVideo.Push(rtpBytes)
-			case session.vRTPControlChannel:
-				// pack.Type = RTP_TYPE_VIDEOCONTROL
-			default:
-				//	Printf("unknow rtp pack type, %v", pack.Type)
-				continue
-			}
-			session.InBytes += rtpLen + 4
-		} else { // rtsp cmd
-			reqBuf := bytes.NewBuffer(nil)
-			reqBuf.WriteByte(buf1)
-			for {
-				if line, isPrefix, err := session.connRW.ReadLine(); err != nil {
-					Println(err)
-					return
-				} else {
-					reqBuf.Write(line)
-					if !isPrefix {
-						reqBuf.WriteString("\r\n")
-					}
-					if len(line) == 0 {
-						req := NewRequest(reqBuf.String())
-						if req == nil {
-							break
-						}
-						session.InBytes += reqBuf.Len()
-						contentLen := req.GetContentLength()
-						session.InBytes += contentLen
-						if contentLen > 0 {
-							bodyBuf := make([]byte, contentLen)
-							if n, err := io.ReadFull(session.connRW, bodyBuf); err != nil {
-								Println(err)
-								return
-							} else if n != contentLen {
-								Printf("read rtsp request body failed, expect size[%d], got size[%d]", contentLen, n)
-								return
-							}
-							req.Body = string(bodyBuf)
-						}
-						session.handleRequest(req)
-						break
-					}
-				}
-			}
-		}
-	}
-}
-
-func (session *RTSP) CheckAuth(authLine string, method string) error {
-	realmRex := regexp.MustCompile(`realm="(.*?)"`)
-	nonceRex := regexp.MustCompile(`nonce="(.*?)"`)
-	usernameRex := regexp.MustCompile(`username="(.*?)"`)
-	responseRex := regexp.MustCompile(`response="(.*?)"`)
-	uriRex := regexp.MustCompile(`uri="(.*?)"`)
-
-	realm := ""
-	nonce := ""
-	username := ""
-	response := ""
-	uri := ""
-	result1 := realmRex.FindStringSubmatch(authLine)
-	if len(result1) == 2 {
-		realm = result1[1]
-	} else {
-		return fmt.Errorf("CheckAuth error : no realm found")
-	}
-	result1 = nonceRex.FindStringSubmatch(authLine)
-	if len(result1) == 2 {
-		nonce = result1[1]
-	} else {
-		return fmt.Errorf("CheckAuth error : no nonce found")
-	}
-	if session.nonce != nonce {
-		return fmt.Errorf("CheckAuth error : sessionNonce not same as nonce")
-	}
-
-	result1 = usernameRex.FindStringSubmatch(authLine)
-	if len(result1) == 2 {
-		username = result1[1]
-	} else {
-		return fmt.Errorf("CheckAuth error : username not found")
-	}
-
-	result1 = responseRex.FindStringSubmatch(authLine)
-	if len(result1) == 2 {
-		response = result1[1]
-	} else {
-		return fmt.Errorf("CheckAuth error : response not found")
-	}
-
-	result1 = uriRex.FindStringSubmatch(authLine)
-	if len(result1) == 2 {
-		uri = result1[1]
-	} else {
-		return fmt.Errorf("CheckAuth error : uri not found")
-	}
-	// var user models.User
-	// err := db.SQLite.Where("Username = ?", username).First(&user).Error
-	// if err != nil {
-	// 	return fmt.Errorf("CheckAuth error : user not exists")
-	// }
-	md5UserRealmPwd := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s:%s", username, realm, session.Auth(username)))))
-	md5MethodURL := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s", method, uri))))
-	myResponse := fmt.Sprintf("%x", md5.Sum([]byte(fmt.Sprintf("%s:%s:%s", md5UserRealmPwd, nonce, md5MethodURL))))
-	if myResponse != response {
-		return fmt.Errorf("CheckAuth error : response not equal")
-	}
-	return nil
-}
-
-func (session *RTSP) handleRequest(req *Request) {
-	//if session.Timeout > 0 {
-	//	session.Conn.SetDeadline(time.Now().Add(time.Duration(session.Timeout) * time.Second))
-	//}
-	Printf("<<<\n%s", req)
-	res := NewResponse(200, "OK", req.Header["CSeq"], session.ID, "")
-	defer func() {
-		if p := recover(); p != nil {
-			Printf("handleRequest err ocurs:%v", p)
-			res.StatusCode = 500
-			res.Status = fmt.Sprintf("Inner Server Error, %v", p)
-		}
-		Printf(">>>\n%s", res)
-		outBytes := []byte(res.String())
-		session.connWLock.Lock()
-		session.connRW.Write(outBytes)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(outBytes)
-		switch req.Method {
-		case "PLAY", "RECORD":
-			switch session.Type {
-			case SESSEION_TYPE_PLAYER:
-				// if session.Pusher.HasPlayer(session.Player) {
-				// 	session.Player.Pause(false)
-				// } else {
-				// 	session.Pusher.AddPlayer(session.Player)
-				// }
-			}
-		case "TEARDOWN":
-			{
-				session.Stop()
-				return
-			}
-		}
-		if res.StatusCode != 200 && res.StatusCode != 401 {
-			Printf("Response request error[%d]. stop session.", res.StatusCode)
-			session.Stop()
-		}
-	}()
-	if req.Method != "OPTIONS" {
-		if session.Auth != nil {
-			authLine := req.Header["Authorization"]
-			authFailed := true
-			if authLine != "" {
-				err := session.CheckAuth(authLine, req.Method)
-				if err == nil {
-					authFailed = false
-				} else {
-					Printf("%v", err)
-				}
-			}
-			if authFailed {
-				res.StatusCode = 401
-				res.Status = "Unauthorized"
-				nonce := fmt.Sprintf("%x", md5.Sum([]byte(shortid.MustGenerate())))
-				session.nonce = nonce
-				res.Header["WWW-Authenticate"] = fmt.Sprintf(`Digest realm="EasyDarwin", nonce="%s", algorithm="MD5"`, nonce)
-				return
-			}
-		}
-	}
-	switch req.Method {
-	case "OPTIONS":
-		res.Header["Public"] = "DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS, ANNOUNCE, RECORD"
-	case "ANNOUNCE":
-		session.Type = SESSION_TYPE_PUSHER
-		session.URL = req.URL
-
-		url, err := url.Parse(req.URL)
-		if err != nil {
-			res.StatusCode = 500
-			res.Status = "Invalid URL"
-			return
-		}
-		streamPath := strings.TrimPrefix(url.Path, "/")
-
-		session.SDPRaw = req.Body
-		session.SDPMap = ParseSDP(req.Body)
-		stream := &Stream{
-			StreamPath: streamPath,
-			Type:       "RTSP",
-		}
-		session.Stream = stream
-		if session.Publish() {
-			if session.ASdp, session.HasAudio = session.SDPMap["audio"]; session.HasAudio {
-				session.setAudioTrack()
-				Printf("audio codec[%s]\n", session.ASdp.Codec)
-			}
-			if session.VSdp, session.HasVideo = session.SDPMap["video"]; session.HasVideo {
-				session.setVideoTrack()
-				Printf("video codec[%s]\n", session.VSdp.Codec)
-			}
-			session.Stream.Type = "RTSP"
-			collection.Store(streamPath, session)
-		}
-	case "DESCRIBE":
-		session.Type = SESSEION_TYPE_PLAYER
-		session.URL = req.URL
-		url, err := url.Parse(req.URL)
-		if err != nil {
-			res.StatusCode = 500
-			res.Status = "Invalid URL"
-			return
-		}
-		streamPath := url.Path
-		stream := FindStream(streamPath)
-		if stream == nil {
-			return
-		}
-		//
-		//res.SetBody(session.SDPRaw)
-	case "SETUP":
-		ts := req.Header["Transport"]
-		// control字段可能是`stream=1`字样，也可能是rtsp://...字样。即control可能是url的path，也可能是整个url
-		// 例1：
-		// a=control:streamid=1
-		// 例2：
-		// a=control:rtsp://192.168.1.64/trackID=1
-		// 例3：
-		// a=control:?ctype=video
-		setupUrl, err := url.Parse(req.URL)
-		if err != nil {
-			res.StatusCode = 500
-			res.Status = "Invalid URL"
-			return
-		}
-		if setupUrl.Port() == "" {
-			setupUrl.Host = fmt.Sprintf("%s:554", setupUrl.Host)
-		}
-		setupPath := setupUrl.String()
-
-		// error status. SETUP without ANNOUNCE or DESCRIBE.
-		//if session.Pusher == nil {
-		//	res.StatusCode = 500
-		//	res.Status = "Error Status"
-		//	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 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
-			}
-		}
-
-		mtcp := regexp.MustCompile("interleaved=(\\d+)(-(\\d+))?")
-		mudp := regexp.MustCompile("client_port=(\\d+)(-(\\d+))?")
-
-		if tcpMatchs := mtcp.FindStringSubmatch(ts); tcpMatchs != nil {
-			session.TransType = TRANS_TYPE_TCP
-			if setupPath == aPath || aPath != "" && strings.LastIndex(setupPath, aPath) == len(setupPath)-len(aPath) {
-				session.aRTPChannel, _ = strconv.Atoi(tcpMatchs[1])
-				session.aRTPControlChannel, _ = strconv.Atoi(tcpMatchs[3])
-			} else if setupPath == vPath || vPath != "" && strings.LastIndex(setupPath, vPath) == len(setupPath)-len(vPath) {
-				session.vRTPChannel, _ = strconv.Atoi(tcpMatchs[1])
-				session.vRTPControlChannel, _ = strconv.Atoi(tcpMatchs[3])
-			} else {
-				res.StatusCode = 500
-				res.Status = fmt.Sprintf("SETUP [TCP] got UnKown control:%s", setupPath)
-				Printf("SETUP [TCP] got UnKown control:%s", setupPath)
-			}
-			Printf("Parse SETUP req.TRANSPORT:TCP.Session.Type:%d,control:%s, AControl:%s,VControl:%s", session.Type, setupPath, aPath, vPath)
-		} else if udpMatchs := mudp.FindStringSubmatch(ts); udpMatchs != nil {
-			session.TransType = TRANS_TYPE_UDP
-			// no need for tcp timeout.
-			session.Conn.timeout = 0
-			if session.Type == SESSEION_TYPE_PLAYER && session.UDPClient == nil {
-				session.UDPClient = &UDPClient{}
-			}
-			if session.Type == SESSION_TYPE_PUSHER && session.UDPServer == nil {
-				session.UDPServer = &UDPServer{
-					Session: session,
-				}
-			}
-			Printf("Parse SETUP req.TRANSPORT:UDP.Session.Type:%d,control:%s, AControl:%s,VControl:%s", session.Type, setupPath, aPath, vPath)
-			if setupPath == aPath || aPath != "" && strings.LastIndex(setupPath, aPath) == len(setupPath)-len(aPath) {
-				if session.Type == SESSEION_TYPE_PLAYER {
-					session.UDPClient.APort, _ = strconv.Atoi(udpMatchs[1])
-					session.UDPClient.AControlPort, _ = strconv.Atoi(udpMatchs[3])
-					if err := session.UDPClient.SetupAudio(); err != nil {
-						res.StatusCode = 500
-						res.Status = fmt.Sprintf("udp client setup audio error, %v", err)
-						return
-					}
-				}
-				if session.Type == SESSION_TYPE_PUSHER {
-					if err := session.UDPServer.SetupAudio(); err != nil {
-						res.StatusCode = 500
-						res.Status = fmt.Sprintf("udp server setup audio error, %v", err)
-						return
-					}
-					tss := strings.Split(ts, ";")
-					idx := -1
-					for i, val := range tss {
-						if val == udpMatchs[0] {
-							idx = i
-						}
-					}
-					tail := append([]string{}, tss[idx+1:]...)
-					tss = append(tss[:idx+1], fmt.Sprintf("server_port=%d-%d", session.UDPServer.APort, session.UDPServer.AControlPort))
-					tss = append(tss, tail...)
-					ts = strings.Join(tss, ";")
-				}
-			} else if setupPath == vPath || vPath != "" && strings.LastIndex(setupPath, vPath) == len(setupPath)-len(vPath) {
-				if session.Type == SESSEION_TYPE_PLAYER {
-					session.UDPClient.VPort, _ = strconv.Atoi(udpMatchs[1])
-					session.UDPClient.VControlPort, _ = strconv.Atoi(udpMatchs[3])
-					if err := session.UDPClient.SetupVideo(); err != nil {
-						res.StatusCode = 500
-						res.Status = fmt.Sprintf("udp client setup video error, %v", err)
-						return
-					}
-				}
-
-				if session.Type == SESSION_TYPE_PUSHER {
-					if err := session.UDPServer.SetupVideo(); err != nil {
-						res.StatusCode = 500
-						res.Status = fmt.Sprintf("udp server setup video error, %v", err)
-						return
-					}
-					tss := strings.Split(ts, ";")
-					idx := -1
-					for i, val := range tss {
-						if val == udpMatchs[0] {
-							idx = i
-						}
-					}
-					tail := append([]string{}, tss[idx+1:]...)
-					tss = append(tss[:idx+1], fmt.Sprintf("server_port=%d-%d", session.UDPServer.VPort, session.UDPServer.VControlPort))
-					tss = append(tss, tail...)
-					ts = strings.Join(tss, ";")
-				}
-			} else {
-				Printf("SETUP [UDP] got UnKown control:%s", setupPath)
-			}
-		}
-		res.Header["Transport"] = ts
-	case "PLAY":
-		// error status. PLAY without ANNOUNCE or DESCRIBE.
-		// if session.Pusher == nil {
-		// 	res.StatusCode = 500
-		// 	res.Status = "Error Status"
-		// 	return
-		// }
-		res.Header["Range"] = req.Header["Range"]
-	case "RECORD":
-		// error status. RECORD without ANNOUNCE or DESCRIBE.
-		// if session.Pusher == nil {
-		// 	res.StatusCode = 500
-		// 	res.Status = "Error Status"
-		// 	return
-		// }
-	case "PAUSE":
-		// if session.Player == nil {
-		// 	res.StatusCode = 500
-		// 	res.Status = "Error Status"
-		// 	return
-		// }
-		// session.Player.Pause(true)
-	}
-}
-
-func (session *RTSP) SendRTP(pack *RTPPack) (err error) {
-	if pack == nil {
-		err = fmt.Errorf("player send rtp got nil pack")
-		return
-	}
-	if session.TransType == TRANS_TYPE_UDP {
-		if session.UDPClient == nil {
-			err = fmt.Errorf("player use udp transport but udp client not found")
-			return
-		}
-		err = session.UDPClient.SendRTP(pack)
-		session.OutBytes += len(pack.Raw)
-		return
-	}
-	switch pack.Type {
-	case RTP_TYPE_AUDIO:
-		bufChannel := make([]byte, 2)
-		bufChannel[0] = 0x24
-		bufChannel[1] = byte(session.aRTPChannel)
-		session.connWLock.Lock()
-		session.connRW.Write(bufChannel)
-		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Raw)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Raw) + 4
-	case RTP_TYPE_AUDIOCONTROL:
-		bufChannel := make([]byte, 2)
-		bufChannel[0] = 0x24
-		bufChannel[1] = byte(session.aRTPControlChannel)
-		session.connWLock.Lock()
-		session.connRW.Write(bufChannel)
-		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Raw)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Raw) + 4
-	case RTP_TYPE_VIDEO:
-		bufChannel := make([]byte, 2)
-		bufChannel[0] = 0x24
-		bufChannel[1] = byte(session.vRTPChannel)
-		session.connWLock.Lock()
-		session.connRW.Write(bufChannel)
-		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Raw)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Raw) + 4
-	case RTP_TYPE_VIDEOCONTROL:
-		bufChannel := make([]byte, 2)
-		bufChannel[0] = 0x24
-		bufChannel[1] = byte(session.vRTPControlChannel)
-		session.connWLock.Lock()
-		session.connRW.Write(bufChannel)
-		bufLen := make([]byte, 2)
-		binary.BigEndian.PutUint16(bufLen, uint16(len(pack.Raw)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Raw)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Raw) + 4
-	default:
-		err = fmt.Errorf("session tcp send rtp got unkown pack type[%v]", pack.Type)
-	}
-	return
-}
--- a/subscriber.go
+++ b/subscriber.go
@@ -0,0 +1,100 @@
+package rtsp
+
+import (
+	"github.com/bluenviron/gortsplib/v3"
+	"github.com/bluenviron/mediacommon/pkg/codecs/mpeg4audio"
+	"github.com/bluenviron/gortsplib/v3/pkg/formats"
+	"github.com/bluenviron/gortsplib/v3/pkg/media"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/codec"
+	"m7s.live/engine/v4/track"
+)
+
+type RTSPSubscriber struct {
+	Subscriber
+	RTSPIO
+}
+
+func (s *RTSPSubscriber) OnEvent(event any) {
+	switch v := event.(type) {
+	case *track.Video:
+		if s.Video != nil {
+			return
+		}
+		switch v.CodecID {
+		case codec.CodecID_H264:
+			video := &media.Media{
+				Type: media.TypeVideo,
+				Formats: []formats.Format{&formats.H264{
+					PacketizationMode: 1,
+					PayloadTyp:        v.PayloadType,
+					SPS:               v.ParamaterSets[0],
+					PPS:               v.ParamaterSets[1],
+				}},
+			}
+			s.videoTrack = video
+			s.tracks = append(s.tracks, video)
+		case codec.CodecID_H265:
+			video := &media.Media{
+				Type: media.TypeVideo,
+				Formats: []formats.Format{&formats.H265{
+					PayloadTyp: v.PayloadType,
+					VPS:        v.ParamaterSets[0],
+					SPS:        v.ParamaterSets[1],
+					PPS:        v.ParamaterSets[2],
+				}},
+			}
+			s.videoTrack = video
+			s.tracks = append(s.tracks, video)
+		}
+		s.AddTrack(v)
+	case *track.Audio:
+		if s.Audio != nil {
+			return
+		}
+		switch v.CodecID {
+		case codec.CodecID_AAC:
+			audio := &media.Media{
+				Type: media.TypeAudio,
+				Formats: []formats.Format{&formats.MPEG4Audio{
+					PayloadTyp: v.PayloadType,
+					Config: &mpeg4audio.Config{
+						Type:         mpeg4audio.ObjectTypeAACLC,
+						SampleRate:   int(v.SampleRate),
+						ChannelCount: int(v.Channels),
+					},
+					SizeLength:       v.SizeLength,
+					IndexLength:      v.IndexLength,
+					IndexDeltaLength: v.IndexDeltaLength,
+				}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		case codec.CodecID_PCMA:
+			audio := &media.Media{
+				Type:    media.TypeAudio,
+				Formats: []formats.Format{&formats.G711{}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		case codec.CodecID_PCMU:
+			audio := &media.Media{
+				Type: media.TypeAudio,
+				Formats: []formats.Format{&formats.G711{
+					MULaw: true,
+				}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		}
+		s.AddTrack(v)
+	case ISubscriber:
+		s.stream = gortsplib.NewServerStream(s.tracks)
+	case VideoRTP:
+		s.stream.WritePacketRTP(s.videoTrack, v.Packet)
+	case AudioRTP:
+		s.stream.WritePacketRTP(s.audioTrack, v.Packet)
+	default:
+		s.Subscriber.OnEvent(event)
+	}
+}
--- a/udp-client.go
+++ b/udp-client.go
@@ -1,161 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"net"
-	"strings"
-
-	. "github.com/Monibuca/utils/v3"
-)
-
-type UDPClient struct {
-	APort        int
-	AConn        *net.UDPConn
-	AControlPort int
-	AControlConn *net.UDPConn
-	VPort        int
-	VConn        *net.UDPConn
-	VControlPort int
-	VControlConn *net.UDPConn
-
-	Stoped bool
-}
-
-func (s *UDPClient) Stop() {
-	if s.Stoped {
-		return
-	}
-	s.Stoped = true
-	if s.AConn != nil {
-		s.AConn.Close()
-		s.AConn = nil
-	}
-	if s.AControlConn != nil {
-		s.AControlConn.Close()
-		s.AControlConn = nil
-	}
-	if s.VConn != nil {
-		s.VConn.Close()
-		s.VConn = nil
-	}
-	if s.VControlConn != nil {
-		s.VControlConn.Close()
-		s.VControlConn = nil
-	}
-}
-
-func (c *UDPClient) SetupAudio() (err error) {
-	defer func() {
-		if err != nil {
-			Println(err)
-			c.Stop()
-		}
-	}()
-	host := c.AConn.RemoteAddr().String()
-	host = host[:strings.LastIndex(host, ":")]
-	addr, err := net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", host, c.APort))
-	if err != nil {
-		return
-	}
-	c.AConn, err = net.DialUDP("udp", nil, addr)
-	if err != nil {
-		return
-	}
-	networkBuffer := 1048576
-	if err := c.AConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp client audio conn set read buffer error, %v", err)
-	}
-	if err := c.AConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp client audio conn set write buffer error, %v", err)
-	}
-
-	addr, err = net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", host, c.AControlPort))
-	if err != nil {
-		return
-	}
-	c.AControlConn, err = net.DialUDP("udp", nil, addr)
-	if err != nil {
-		return
-	}
-	if err := c.AControlConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp client audio control conn set read buffer error, %v", err)
-	}
-	if err := c.AControlConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp client audio control conn set write buffer error, %v", err)
-	}
-	return
-}
-
-func (c *UDPClient) SetupVideo() (err error) {
-	defer func() {
-		if err != nil {
-			Println(err)
-			c.Stop()
-		}
-	}()
-	host := c.VConn.RemoteAddr().String()
-	host = host[:strings.LastIndex(host, ":")]
-	addr, err := net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", host, c.VPort))
-	if err != nil {
-		return
-	}
-	c.VConn, err = net.DialUDP("udp", nil, addr)
-	if err != nil {
-		return
-	}
-	networkBuffer := 1048576
-	if err := c.VConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp client video conn set read buffer error, %v", err)
-	}
-	if err := c.VConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp client video conn set write buffer error, %v", err)
-	}
-
-	addr, err = net.ResolveUDPAddr("udp", fmt.Sprintf("%s:%d", host, c.VControlPort))
-	if err != nil {
-		return
-	}
-	c.VControlConn, err = net.DialUDP("udp", nil, addr)
-	if err != nil {
-		return
-	}
-	if err := c.VControlConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp client video control conn set read buffer error, %v", err)
-	}
-	if err := c.VControlConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp client video control conn set write buffer error, %v", err)
-	}
-	return
-}
-
-func (c *UDPClient) SendRTP(pack *RTPPack) (err error) {
-	if pack == nil {
-		err = fmt.Errorf("udp client send rtp got nil pack")
-		return
-	}
-	var conn *net.UDPConn
-	switch pack.Type {
-	case RTP_TYPE_AUDIO:
-		conn = c.AConn
-	case RTP_TYPE_AUDIOCONTROL:
-		conn = c.AControlConn
-	case RTP_TYPE_VIDEO:
-		conn = c.VConn
-	case RTP_TYPE_VIDEOCONTROL:
-		conn = c.VControlConn
-	default:
-		err = fmt.Errorf("udp client send rtp got unkown pack type[%v]", pack.Type)
-		return
-	}
-	if conn == nil {
-		err = fmt.Errorf("udp client send rtp pack type[%v] failed, conn not found", pack.Type)
-		return
-	}
-
-	if _, err = conn.Write(pack.Raw); err != nil {
-		err = fmt.Errorf("udp client write bytes error, %v", err)
-		return
-	}
-	// Printf("udp client write [%d/%d]", n, pack.Buffer.Len())
-	return
-}
--- a/udp-server.go
+++ b/udp-server.go
@@ -1,219 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"net"
-	"strconv"
-	"strings"
-	"sync"
-
-	. "github.com/Monibuca/utils/v3"
-)
-
-type UDPServer struct {
-	Session *RTSP
-	UDPClient
-	sync.Mutex
-}
-
-func (s *UDPServer) AddInputBytes(bytes int) {
-	if s.Session != nil {
-		s.Session.InBytes += bytes
-		return
-	}
-	panic(fmt.Errorf("session and RTSPClient both nil"))
-}
-
-func (s *UDPServer) Stop() {
-	if s.Stoped {
-		return
-	}
-	s.Stoped = true
-	if s.AConn != nil {
-		s.AConn.Close()
-		s.AConn = nil
-	}
-	if s.AControlConn != nil {
-		s.AControlConn.Close()
-		s.AControlConn = nil
-	}
-	if s.VConn != nil {
-		s.VConn.Close()
-		s.VConn = nil
-	}
-	if s.VControlConn != nil {
-		s.VControlConn.Close()
-		s.VControlConn = nil
-	}
-}
-
-func (s *UDPServer) SetupAudio() (err error) {
-	addr, err := net.ResolveUDPAddr("udp", ":0")
-	if err != nil {
-		return
-	}
-	s.AConn, err = net.ListenUDP("udp", addr)
-	if err != nil {
-		return
-	}
-	networkBuffer := 1048576
-	if err := s.AConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp server audio conn set read buffer error, %v", err)
-	}
-	if err := s.AConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp server audio conn set write buffer error, %v", err)
-	}
-	la := s.AConn.LocalAddr().String()
-	strPort := la[strings.LastIndex(la, ":")+1:]
-	s.APort, err = strconv.Atoi(strPort)
-	if err != nil {
-		return
-	}
-	go func() {
-		bufUDP := make([]byte, UDP_BUF_SIZE)
-		Printf("udp server start listen audio port[%d]", s.APort)
-		defer Printf("udp server stop listen audio port[%d]", s.APort)
-		// timer := time.Unix(0, 0)
-		for !s.Stoped {
-			if n, _, err := s.AConn.ReadFromUDP(bufUDP); err == nil {
-				// elapsed := time.Now().Sub(timer)
-				// if elapsed >= 30*time.Second {
-				// 	Printf("Package recv from AConn.len:%d\n", n)
-				// 	timer = time.Now()
-				// }
-				s.AddInputBytes(n)
-				var bytes []byte
-				s.Session.RtpAudio.Push(append(bytes, bufUDP[:n]...))
-			} else {
-				Println("udp server read audio pack error", err)
-				continue
-			}
-		}
-	}()
-	addr, err = net.ResolveUDPAddr("udp", ":0")
-	if err != nil {
-		return
-	}
-	s.AControlConn, err = net.ListenUDP("udp", addr)
-	if err != nil {
-		return
-	}
-	if err := s.AControlConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp server audio control conn set read buffer error, %v", err)
-	}
-	if err := s.AControlConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp server audio control conn set write buffer error, %v", err)
-	}
-	la = s.AControlConn.LocalAddr().String()
-	strPort = la[strings.LastIndex(la, ":")+1:]
-	s.AControlPort, err = strconv.Atoi(strPort)
-	if err != nil {
-		return
-	}
-	go func() {
-		bufUDP := make([]byte, UDP_BUF_SIZE)
-		Printf("udp server start listen audio control port[%d]", s.AControlPort)
-		defer Printf("udp server stop listen audio control port[%d]", s.AControlPort)
-		for !s.Stoped {
-			if n, _, err := s.AControlConn.ReadFromUDP(bufUDP); err == nil {
-				//Printf("Package recv from AControlConn.len:%d\n", n)
-				s.AddInputBytes(n)
-				// pack := RTPPack{
-				// 	Type: RTP_TYPE_AUDIOCONTROL,
-				// }
-				// pack.Unmarshal(bufUDP[:n])
-				// s.HandleRTP(pack)
-			} else {
-				Println("udp server read audio control pack error", err)
-				continue
-			}
-		}
-	}()
-	return
-}
-
-func (s *UDPServer) SetupVideo() (err error) {
-	addr, err := net.ResolveUDPAddr("udp", ":0")
-	if err != nil {
-		return
-	}
-	s.VConn, err = net.ListenUDP("udp", addr)
-	if err != nil {
-		return
-	}
-	networkBuffer := 1048576
-	if err := s.VConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp server video conn set read buffer error, %v", err)
-	}
-	if err := s.VConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp server video conn set write buffer error, %v", err)
-	}
-	la := s.VConn.LocalAddr().String()
-	strPort := la[strings.LastIndex(la, ":")+1:]
-	s.VPort, err = strconv.Atoi(strPort)
-	if err != nil {
-		return
-	}
-	go func() {
-		bufUDP := make([]byte, UDP_BUF_SIZE)
-		Printf("udp server start listen video port[%d]", s.VPort)
-		defer Printf("udp server stop listen video port[%d]", s.VPort)
-		// timer := time.Unix(0, 0)
-		for !s.Stoped {
-			if n, _, err := s.VConn.ReadFromUDP(bufUDP); err == nil {
-				// elapsed := time.Now().Sub(timer)
-				// if elapsed >= 30*time.Second {
-				// 	Printf("Package recv from VConn.len:%d\n", n)
-				// 	timer = time.Now()
-				// }
-				s.AddInputBytes(n)
-				var bytes []byte
-				s.Session.RtpVideo.Push(append(bytes, bufUDP[:n]...))
-			} else {
-				Println("udp server read video pack error", err)
-				continue
-			}
-		}
-	}()
-
-	addr, err = net.ResolveUDPAddr("udp", ":0")
-	if err != nil {
-		return
-	}
-	s.VControlConn, err = net.ListenUDP("udp", addr)
-	if err != nil {
-		return
-	}
-	if err := s.VControlConn.SetReadBuffer(networkBuffer); err != nil {
-		Printf("udp server video control conn set read buffer error, %v", err)
-	}
-	if err := s.VControlConn.SetWriteBuffer(networkBuffer); err != nil {
-		Printf("udp server video control conn set write buffer error, %v", err)
-	}
-	la = s.VControlConn.LocalAddr().String()
-	strPort = la[strings.LastIndex(la, ":")+1:]
-	s.VControlPort, err = strconv.Atoi(strPort)
-	if err != nil {
-		return
-	}
-	go func() {
-		bufUDP := make([]byte, UDP_BUF_SIZE)
-		Printf("udp server start listen video control port[%d]", s.VControlPort)
-		defer Printf("udp server stop listen video control port[%d]", s.VControlPort)
-		for !s.Stoped {
-			if n, _, err := s.VControlConn.ReadFromUDP(bufUDP); err == nil {
-				//Printf("Package recv from VControlConn.len:%d\n", n)
-				s.AddInputBytes(n)
-				// pack := RTPPack{
-				// 	Type: RTP_TYPE_VIDEOCONTROL,
-				// }
-				// pack.Unmarshal(bufUDP[:n])
-				// s.HandleRTP(pack)
-			} else {
-				Println("udp server read video control pack error", err)
-				continue
-			}
-		}
-	}()
-	return
-}
Author	SHA1	Message	Date
langhuihui	9936b2ca20	fix: upgrade gortsplib	2023-08-25 15:00:57 +08:00
langhuihui	943ba46f70	chore: fit engine udpate	2023-08-14 10:04:37 +08:00
langhuihui	035c3647eb	docs: update readme	2023-08-06 18:05:06 +08:00
langhuihui	17874da782	docs: udpate readme	2023-08-06 15:26:50 +08:00
langhuihui	24b63ca07c	feat: add stop reason, pull protocol default is tcp now	2023-08-06 15:15:26 +08:00
dexter	eb91dd1cee	Merge pull request #42 from Monibuca/rtsp-pcma Update publisher.go	2023-07-24 13:45:37 +08:00
banshan	e5d0d7d5d6	Update publisher.go ffmpeg rtsp pcma 推流	2023-07-24 13:43:44 +08:00
langhuihui	311b8732a8	fit engine update	2023-06-16 22:42:43 +08:00
langhuihui	351ff9fa0c	refactor: remove some stupid code	2023-05-25 14:15:09 +08:00
langhuihui	9defe4eec3	fit engine update	2023-05-23 20:58:59 +08:00
langhuihui	8e84bb637a	适配引擎修改	2023-04-24 14:13:46 +08:00
langhuihui	dc62df1861	更新readme	2023-04-23 13:00:11 +08:00
langhuihui	1ad45bfbc1	gortsplib升级到v3.2.1 增加默认缓存大小，解决高码率推拉流问题增加英文文档	2023-04-23 11:26:02 +08:00
langhuihui	a4773b4044	使用内置的协议自动选择机制	2023-04-15 08:48:44 +08:00
dexter	225bb482d5	适配引擎修改	2023-03-28 19:48:59 +08:00
dexter	ea5ea6f48a	SizeLength、IndexLength、IndexDeltaLength	2023-03-03 22:06:38 +08:00
dexter	beda2dd9e0	适配引擎修改	2023-02-26 10:38:53 +08:00
dexter	344ab6f463	移除clockrate	2023-02-20 00:19:39 +08:00
dexter	d5e73d11db	Merge pull request #32 from Monibuca/dependabot/go_modules/golang.org/x/net-0.7.0 Bump golang.org/x/net from 0.2.0 to 0.7.0	2023-02-19 10:22:56 +08:00
dependabot[bot]	12350b8af0	Bump golang.org/x/net from 0.2.0 to 0.7.0 Bumps [golang.org/x/net](https://github.com/golang/net) from 0.2.0 to 0.7.0. - [Release notes](https://github.com/golang/net/releases) - [Commits](https://github.com/golang/net/compare/v0.2.0...v0.7.0) --- updated-dependencies: - dependency-name: golang.org/x/net dependency-type: indirect ... Signed-off-by: dependabot[bot] <support@github.com>	2023-02-18 03:22:09 +00:00
dexter	3b671606b6	添加时钟频率配置	2023-02-08 14:19:36 +08:00
dexter	70321ccd1f	更新依赖库aler9 到2.1.0	2023-02-03 21:36:33 +08:00
dexter	5ddbd7a852	优化代码	2023-02-02 12:08:16 +08:00
dexter	f32f44db40	适配引擎升级	2023-02-01 11:00:36 +08:00
dexter	b44efd9749	推拉在异常时对连接关闭	2023-01-18 21:20:06 +08:00
dexter	7f4bfe4c78	使用rtsp拉流时将参数传入订阅者	2023-01-02 21:12:45 +08:00
dexter	187335d311	适配engine v4.9.5	2022-12-31 22:18:50 +08:00
dexter	1f3f4bfb0e	适配引擎4.9.3	2022-12-19 09:06:17 +08:00
dexter	3cbf22576b	🐛 FIX: 关闭流后断开rtsp拉流或者推流连接	2022-11-23 09:22:23 +08:00
dexter	1206a4636f	👌 IMPROVE: 升级gortsplib版本	2022-11-17 23:48:13 +08:00
dexter	6063f42181	🐛 FIX: enable not enabled	2022-10-10 21:51:08 +08:00
dexter	2e6679b3ea	👌 IMPROVE: 添加初始化监听失败日志	2022-10-03 17:57:49 +08:00
dexter	022857c3d7	👌 IMPROVE: 增加一些错误信息输出	2022-09-06 18:56:19 +08:00
dexter	0501a84da6	👌 IMPROVE: 导出RTSP插件	2022-09-04 12:43:42 +08:00
dexter	7d08e06922	兼容sdp中没有包含sps和pps的情况	2022-08-21 14:49:54 +08:00
dexter	754a28e506	更新依赖版本	2022-07-29 20:43:39 +08:00
dexter	9b058153d2	增加强制指定拉流协议的配置	2022-07-23 11:59:26 +08:00
dexter	ae37279dd1	aler9未处理回调中带有err的情况	2022-07-03 01:14:23 +08:00
dexter	93d6eedff2	适配engine 4.4.0	2022-06-25 20:11:32 +08:00
dexter	2c1d908d7e	跟随引擎4.3.0版本	2022-06-19 23:44:28 +08:00
dexter	ced54a94a4	添加序列化屏蔽字段	2022-06-19 15:30:06 +08:00
dexter	f9bc450d01	跟随aler9升级	2022-06-05 21:09:37 +08:00
dexter	44fb77d121	去掉g711代码	2022-05-27 13:42:06 +08:00
dexter	1b6981cccb	使用官方创建的g711	2022-05-27 12:41:00 +08:00
dexter	2cd295d32c	payloadType推拉对应	2022-05-26 20:35:27 +08:00
dexter	57ec489fef	engine已实现等待track功能	2022-05-26 11:20:11 +08:00
dexter	a2100768b4	rtp降为v1	2022-05-15 16:32:45 +08:00
dexter	76956b16d1	对推流可能无法解析的情况进行报错	2022-05-10 21:58:35 +08:00
dexter	91e1726920	暂时修复向远程推流	2022-05-10 15:30:14 +08:00
dexter	4a0da71ee9	修改推流	2022-05-09 22:03:55 +08:00
langhuihui	701d55469d	update readme	2022-05-04 21:38:20 +08:00
langhuihui	e273d0010e	将pcma的payloadType设置为8	2022-05-03 22:28:21 +08:00
langhuihui	96834e26a1	加上空格	2022-05-03 22:24:07 +08:00
dexter	7d0451c204	优化代码	2022-04-24 09:55:37 +08:00
langhuihui	c7c8858d36	添加API接口	2022-04-05 21:19:08 +08:00
dexter	731521f771	升级playblock	2022-03-27 11:33:34 +08:00
dexter	17334dd106	发布功能完善	2022-03-20 00:16:16 +08:00
dexter	2d9838bdfa	加上client逻辑	2022-03-19 01:15:19 +08:00
dexter	bd5a146ea6	v4 update	2022-03-18 01:24:32 +08:00
dexter	e411d30e91	Merge pull request #17 from jianglieshan/aler9 fix:修改rtsp插件作为服务端出流时，ssrc为0的bug	2022-01-16 13:12:11 +08:00
jianglieshan	709a4cee7b	fix:修改rtsp插件作为服务端出流时，ssrc为0的bug	2022-01-15 16:50:40 +08:00
dexter	a90f52769d	Merge pull request #16 from ziminghua/aler9 增加RTSPClient关闭的事件订阅，同步关闭客户端连接	2022-01-11 17:00:40 +08:00
訾明华	3764a26bbd	增加RTSPClient关闭的事件订阅，同步关闭客户端连接增加RTSPClient关闭的事件订阅，同步关闭客户端连接	2022-01-11 16:52:14 +08:00
dexter	2533ab2604	Merge pull request #15 from ziminghua/aler9 多slice的情况下，同步同一帧的时间戳	2022-01-11 11:40:38 +08:00
訾明华	db07f0d588	多slice的情况下，同步同一帧的时间戳 `vpacketer.Packetize`再打包的过程中会把当前的timestamp+samples作为下一次打包的时间戳，如果多slice会连续传递samples导致同一帧的时间戳不一致	2022-01-11 11:34:29 +08:00
dexter	f110513d70	增加配置项ReadBufferSize	2021-12-29 22:59:45 +08:00
dexter	8901f4c117	修复bug	2021-12-29 22:16:11 +08:00
dexter	2f7c2de352	增加读取缓存大小，设置Mark标志位	2021-12-29 20:18:12 +08:00
dexter	af053bb5e6	对处理回调判空	2021-12-27 20:42:06 +08:00
dexter	bed7ba8a87	修复一个低级错误	2021-12-22 16:32:06 +08:00
dexter	0cbc4beb0f	Merge pull request #13 from lhong1001/rtsp-syld modified by syld 2021-12-20	2021-12-20 18:24:49 +08:00
root	edbfc07275	modified by syld 2021-12-20	2021-12-20 17:43:08 +08:00
dexter	329f93022e	修复流终止时仍然在拉流的bug	2021-12-14 14:21:52 +08:00
dexter	4895f2ec42	修复获取rtsp列表信息	2021-12-13 10:17:54 +08:00
dexter	9eb117811d	改名	2021-11-28 23:08:13 +08:00
dexter	00ecd3469f	每次重连切换连接方式	2021-11-23 12:40:21 +08:00
dexter	4107d31c79	默认拉tcp	2021-11-19 21:24:30 +08:00
dexter	5094fd0db7	加入转推功能	2021-11-18 19:29:31 +08:00
dexter	ef106e42f8	跟随升级gotsplib	2021-11-18 19:05:06 +08:00
dexter	0ac9513920	更新readme	2021-11-16 19:14:23 +08:00
dexter	a900613c70	初步改造完成	2021-11-16 19:06:24 +08:00
dexter	ac8aa96350	format	2021-10-06 09:22:59 +08:00
dexter	f267b1ca52	rtp依赖1.6.5不能用1.7版本	2021-08-08 08:07:22 +08:00
dexter	229370c083	更改类型适配pion的rtp类型升级	2021-08-08 07:40:52 +08:00
dexter	bb1e8ba1d8	适配3.3	2021-08-07 22:00:28 +08:00
dexter	8cf3e0c0fc	增加对publisher的非空判断	2021-08-04 15:27:55 +08:00
dexter	1ecb45d904	修改readme	2021-08-03 15:33:55 +08:00
dexter	3ea5bb7f27	更新readme	2021-08-02 09:21:38 +08:00
langhuihui	9aec4ec4be	防止json循环引用	2021-07-24 11:38:24 +08:00
langhuihui	da2fc9d462	更新重连逻辑	2021-07-24 09:38:22 +08:00
李宇翔	f68a3ee14b	实现rtsp拉流播放	2021-07-19 20:07:01 +08:00
langhuihui	a2f5cb87b1	修复音频初始化问题	2021-07-12 23:24:04 +08:00
langhuihui	5cdbc220de	修改rtsp自动拉流配置结构	2021-07-11 21:43:15 +08:00
langhuihui	f0a00f3db9	更新readme	2021-07-10 17:52:50 +08:00