Merge pull request #20 from ziminghua/aler9

解决发布超时后，重连RTSP，无法播放的BUG
2025-09-27 20:12:14 +08:00 · 2022-02-22 21:17:24 +08:00 · 2022-02-22 21:15:35 +08:00 · 2022-01-16 13:12:11 +08:00 · 2022-01-15 16:50:40 +08:00 · 2022-01-11 17:00:40 +08:00
27 changed files with 910 additions and 12829 deletions
--- a/README.md
+++ b/README.md
@@ -1,26 +1,63 @@
-# Monibuca 的RTSP 插件
+# RTSP插件

-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
+## 插件地址

-## 插件名称
+github.com/Monibuca/plugin-rtsp

-RTSP
+## 插件引入
+```go
+import (
+    _ "github.com/Monibuca/plugin-rtsp"
+)
+```
+
+## 默认插件配置

-## 配置
 ```toml
 [RTSP]
-ListenAddr  = ":554"
-BufferLength  = 2048
-AutoPull     = false
-RemoteAddr   = "rtsp://localhost/${streamPath}"
+# 端口接收推流
+ListenAddr = ":554"
+Reconnect = true
+[RTSP.AutoPullList]
+"live/rtsp1" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=1&subtype=1"
+"live/rtsp2" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=2&subtype=1"
 ```
- ListenAddr 是监听端口，可以将rtsp流推到Monibuca中
- BufferLength是指解析拉取的rtp包的缓冲大小
- AutoPull是指当有用户订阅一个新流的时候自动向远程拉流转发
- RemoteAddr 指远程拉流地址，其中${streamPath}是占位符，实际使用流路径替换。

+- `ListenAddr`是监听的地址
+- `Reconnect` 是否自动重连
+- `RTSP.AutoPullList` 可以配置多项，用于自动拉流，key是streamPath，value是远程rtsp地址

-## 使用方法(拉流转发)
+### 特殊功能
+
+当自动拉流列表中当的streamPath为sub/xxx 这种形式的话，在gb28181的分屏显示时会优先采用rtsp流，已实现分屏观看子码流效果
+## 插件功能
+
+### 接收RTSP协议的推流
+
+例如通过ffmpeg向m7s进行推流
+
+```bash
+ffmpeg -i **** rtsp://localhost/live/test
+```
+
+会在m7s内部形成一个名为live/test的流
+
+### 从远程拉取rtsp到m7s中
+
+可调用接口
+`/api/rtsp/pull?target=[RTSP地址]&streamPath=[流标识]`
+
+## 使用编程方式拉流
 ```go
-new(RTSP).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
-```
+new(RTSPClient).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
+```
+
+### 罗列所有的rtsp协议的流
+
+可调用接口
+`/api/rtsp/list`
+
+### 从m7s中拉取rtsp协议流
+
+直接通过协议rtsp://xxx.xxx.xxx.xxx/live/user1 即可播放
+> h265 编码拉流尚未实现，敬请期待
--- a/client.go
+++ b/client.go
@@ -1,556 +1,214 @@
 package rtsp

 import (
-	"bufio"
-	"bytes"
-	"crypto/md5"
-	"encoding/binary"
-	"fmt"
-	. "github.com/Monibuca/engine/v2"
-	"github.com/pixelbender/go-sdp/sdp"
-	"io"
-	"net"
-	"net/url"
-	"regexp"
-	"strconv"
-	"strings"
+	"errors"
 	"time"
+	"unsafe"
+
+	. "github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/Monibuca/utils/v3/codec"
+	"github.com/aler9/gortsplib"
+	"github.com/aler9/gortsplib/pkg/aac"
+	"github.com/aler9/gortsplib/pkg/base"
+	"github.com/pion/rtp"
+	"github.com/pion/rtp/codecs"
 )

+type RTSPClient struct {
+	RTSPublisher
+	Transport         gortsplib.Transport
+	*gortsplib.Client `json:"-"`
+}
+
 // PullStream 从外部拉流
-func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (result bool) {
-	if result = rtsp.Publisher.Publish(streamPath); result {
-		rtsp.Stream.Type = "RTSP"
-		rtsp.RTSPInfo.StreamInfo = &rtsp.Stream.StreamInfo
-		rtsp.TransType = TRANS_TYPE_TCP
-		rtsp.vRTPChannel = 0
-		rtsp.vRTPControlChannel = 1
-		rtsp.aRTPChannel = 2
-		rtsp.aRTPControlChannel = 3
+func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error) {
+	rtsp.Stream = &Stream{
+		StreamPath: streamPath,
+		Type:       "RTSP Pull",
+		ExtraProp:  rtsp,
+	}
+	if result := rtsp.Publish(); result {
 		rtsp.URL = rtspUrl
-		if err := rtsp.requestStream();err != nil {
-			rtsp.Close()
-			return false
-		}
-		go rtsp.startStream()
-		collection.Store(streamPath, rtsp)
-		// 	go rtsp.run()
-	}
-	return
-}
-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
-}
-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 {
-					// not support yet
-					// TODO..
-				}
-			}
-			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 {
-				// not support yet
-				// TODO..
-				return "", fmt.Errorf("not support Basic auth yet")
-			}
-		}
-	}
-	return "", nil
-}
-func (client *RTSP) requestStream() (err error) {
-	timeout := time.Duration(5) * time.Second
-	l, err := url.Parse(client.URL)
-	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
-		}
-	}
-	_sdp, err := sdp.ParseString(resp.Body)
-	if err != nil {
-		return err
-	}
-	client.Sdp = _sdp
-	client.SDPRaw = resp.Body
-	client.SDPMap = ParseSDP(client.SDPRaw)
-	session := ""
-	for _, media := range _sdp.Media {
-		switch media.Type {
-		case "video":
-			client.VControl = media.Attributes.Get("control")
-			client.VCodec = media.Format[0].Name
-			client.SPS = client.SDPMap["video"].SpropParameterSets[0]
-			client.PPS = client.SDPMap["video"].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, "/")
-			}
-			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)
-			resp, err = client.RequestWithPath("SETUP", _url, headers, true)
-			if err != nil {
-				return err
-			}
-			session, _ = resp.Header["Session"].(string)
-		case "audio":
-			client.AControl = media.Attributes.Get("control")
-			client.ACodec = media.Format[0].Name
-			client.AudioSpecificConfig = client.SDPMap["audio"].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}
-				}
-				err = client.UDPServer.SetupAudio()
-				if 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
-			}
-			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)
-			resp, err = client.RequestWithPath("SETUP", _url, headers, true)
-			if err != nil {
-				return err
-			}
-			session, _ = resp.Header["Session"].(string)
-		}
-	}
-	headers = make(map[string]string)
-	if session != "" {
-		headers["Session"] = session
-	}
-	resp, err = client.Request("PLAY", headers)
-	if err != nil {
-		return err
-	}
-	return nil
-}
-
-func (client *RTSP) startStream() {
-	//startTime := time.Now()
-	//loggerTime := time.Now().Add(-10 * time.Second)
-	defer client.Stop()
-	for {
-		//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...
-		//		}
-		//	}
-		//}
-		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
-			}
-			rtpBuf := content
-			var pack *RTPPack
-			switch channel {
-			case client.aRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBuf,
-				}
-			case client.aRTPControlChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIOCONTROL,
-					Buffer: rtpBuf,
-				}
-			case client.vRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBuf,
-				}
-			case client.vRTPControlChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEOCONTROL,
-					Buffer: rtpBuf,
-				}
-			default:
-				Printf("unknow rtp pack type, channel:%v", channel)
-				continue
-			}
-			if pack == nil {
-				Printf("session tcp got nil rtp pack")
-				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)
-			client.handleRTP(pack)
-
-		default: // rtsp
-			builder := bytes.Buffer{}
-			builder.WriteByte(b)
-			contentLen := 0
-			for {
-				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)
+		if config.Reconnect {
+			go func() {
+				for rtsp.pullStream(); rtsp.Err() == nil; rtsp.pullStream() {
+					Printf("reconnecting:%s in 5 seconds", rtspUrl)
+					if rtsp.Transport == gortsplib.TransportTCP {
+						rtsp.Transport = gortsplib.TransportUDP
+					} else {
+						rtsp.Transport = gortsplib.TransportTCP
 					}
-					Printf("<<<[IN]\n%s", builder.String())
-					break
+					time.Sleep(time.Second * 5)
 				}
-				s := string(line)
-				builder.Write(line)
-				if !prefix {
-					builder.WriteString("\r\n")
+				if rtsp.IsTimeout {
+					rtsp.processFunc = nil
+					go rtsp.PullStream(streamPath, rtspUrl)
 				}
-
-				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
-					}
-				}
-			}
+			}()
+		} else {
+			go rtsp.pullStream()
 		}
-	}
-}
-
-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 {
-		return err
-	}
-	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 {
 		return
 	}
-	client.connRW.Flush()
-
-	if !needResp {
-		return nil, nil
-	}
-	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
-			}
-		}
-
-	}
-	return
+	return errors.New("publish badname")
+}
+func (rtsp *RTSPClient) PushStream(streamPath string, rtspUrl string) (err error) {
+	if s := FindStream(streamPath); s != nil {
+		var tracks gortsplib.Tracks
+		var sub RTSPSubscriber
+		sub.Type = "RTSP push out"
+		sub.vt = s.WaitVideoTrack("h264", "h265")
+		sub.at = s.WaitAudioTrack("aac", "pcma", "pcmu")
+		ssrc := uintptr(unsafe.Pointer(&sub))
+		var trackIds = 0
+		if sub.vt != nil {
+			trackId := trackIds
+			var vtrack *gortsplib.Track
+			var vpacketer rtp.Packetizer
+			switch sub.vt.CodecID {
+			case codec.CodecID_H264:
+				if vtrack, err = gortsplib.NewTrackH264(96, &gortsplib.TrackConfigH264{
+					SPS: sub.vt.ExtraData.NALUs[0],
+					PPS: sub.vt.ExtraData.NALUs[1],
+				}); err == nil {
+					vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &codecs.H264Payloader{}, rtp.NewFixedSequencer(1), 90000)
+				} else {
+					return err
+				}
+			case codec.CodecID_H265:
+				vtrack = NewH265Track(96, sub.vt.ExtraData.NALUs)
+				vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &H265Payloader{}, rtp.NewFixedSequencer(1), 90000)
+			}
+			var st uint32
+			onVideo := func(ts uint32, pack *VideoPack) {
+				for _, nalu := range pack.NALUs {
+					for _, pack := range vpacketer.Packetize(nalu, (ts-st)*90) {
+						rtp, _ := pack.Marshal()
+						rtsp.WritePacketRTP(trackId, rtp)
+					}
+				}
+				st = ts
+			}
+			sub.OnVideo = func(ts uint32, pack *VideoPack) {
+				if st = ts; st != 0 {
+					sub.OnVideo = onVideo
+				}
+				onVideo(ts, pack)
+			}
+			tracks = append(tracks, vtrack)
+			trackIds++
+		}
+		if sub.at != nil {
+			var st uint32
+			trackId := trackIds
+			switch sub.at.CodecID {
+			case codec.CodecID_PCMA, codec.CodecID_PCMU:
+				atrack := NewG711Track(97, map[byte]string{7: "pcma", 8: "pcmu"}[sub.at.CodecID])
+				apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &codecs.G711Payloader{}, rtp.NewFixedSequencer(1), 8000)
+				sub.OnAudio = func(ts uint32, pack *AudioPack) {
+					for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*8) {
+						buf, _ := pack.Marshal()
+						rtsp.WritePacketRTP(trackId, buf)
+					}
+					st = ts
+				}
+				tracks = append(tracks, atrack)
+			case codec.CodecID_AAC:
+				var mpegConf aac.MPEG4AudioConfig
+				mpegConf.Decode(sub.at.ExtraData[2:])
+				conf := &gortsplib.TrackConfigAAC{
+					Type:              int(mpegConf.Type),
+					SampleRate:        mpegConf.SampleRate,
+					ChannelCount:      mpegConf.ChannelCount,
+					AOTSpecificConfig: mpegConf.AOTSpecificConfig,
+				}
+				if atrack, err := gortsplib.NewTrackAAC(97, conf); err == nil {
+					apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &AACPayloader{}, rtp.NewFixedSequencer(1), uint32(mpegConf.SampleRate))
+					sub.OnAudio = func(ts uint32, pack *AudioPack) {
+						for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*uint32(mpegConf.SampleRate)/1000) {
+							buf, _ := pack.Marshal()
+							rtsp.WritePacketRTP(trackId, buf)
+						}
+						st = ts
+					}
+					tracks = append(tracks, atrack)
+				}
+			}
+		}
+		return rtsp.StartPublishing(rtspUrl, tracks)
+	}
+	return errors.New("stream not exist")
+}
+func (client *RTSPClient) pullStream() {
+	if client.Err() != nil {
+		return
+	}
+	client.Client = &gortsplib.Client{
+		OnPacketRTP: func(trackID int, payload []byte) {
+			// Println("OnPacketRTP", trackID, len(payload))
+			if f := client.processFunc[trackID]; f != nil {
+				var clone []byte
+				f(append(clone, payload...))
+			}
+		},
+		ReadBufferSize: config.ReadBufferSize,
+		Transport:      &client.Transport,
+	}
+	// parse URL
+	u, err := base.ParseURL(client.URL)
+	if err != nil {
+		Printf("ParseURL:%s error:%v", client.URL, err)
+		return
+	}
+	// connect to the server
+	if err = client.Start(u.Scheme, u.Host); err != nil {
+		Printf("connect:%s error:%v", client.URL, err)
+		return
+	}
+	client.OnClose = func() {
+		client.Client.Close()
+	}
+	//client.close should be after connected!
+	defer client.Client.Close()
+	var res *base.Response
+	if res, err = client.Options(u); err != nil {
+		Printf("option:%s error:%v", client.URL, err)
+		return
+	}
+	Println(res)
+	// find published tracks
+	tracks, baseURL, res, err := client.Describe(u)
+	if err != nil {
+		Printf("Describe:%s error:%v", client.URL, err)
+		return
+	}
+	Println(res)
+	if client.processFunc == nil {
+		client.setTracks(tracks)
+	}
+	for _, track := range tracks {
+		if res, err = client.Setup(true, track, baseURL, 0, 0); err != nil {
+			Printf("Setup:%s error:%v", baseURL.String(), err)
+			return
+		}
+		Println(res)
+	}
+	// start reading tracks
+	if res, err = client.Play(nil); err != nil {
+		Printf("Play:%s error:%v", baseURL.String(), err)
+		return
+	}
+	Println(res)
+	// wait until a fatal error
+	var fatalChan = make(chan error)
+	go func() {
+		fatalChan <- client.Wait()
+	}()
+	select {
+	case err := <-fatalChan:
+		Printf("Wait:%s error:%v", baseURL.String(), err)
+	case <-client.Done():
+		Printf("client:%s done", client.URL)
+	}
 }
--- a/go.mod
+++ b/go.mod
@@ -1,11 +1,11 @@
-module github.com/Monibuca/plugin-rtsp
+module github.com/Monibuca/plugin-rtsp/v3

-go 1.13
+go 1.16

 require (
-	github.com/Monibuca/engine/v2 v2.0.0
-	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/teris-io/shortid v0.0.0-20171029131806-771a37caa5cf
+	github.com/Monibuca/engine/v3 v3.4.1
+	github.com/Monibuca/utils/v3 v3.0.5
+	github.com/aler9/gortsplib v0.0.0-20211212220644-6f374e396529
+	github.com/pion/rtp v1.7.4
+	github.com/pion/sdp/v3 v3.0.4
 )
--- a/go.sum
+++ b/go.sum
@@ -1,82 +1,101 @@
-github.com/BurntSushi/toml v0.3.1 h1:WXkYYl6Yr3qBf1K79EBnL4mak0OimBfB0XUf9Vl28OQ=
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=
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--- a/main.go
+++ b/main.go
@@ -1,313 +1,113 @@
 package rtsp

 import (
-	"bufio"
-	"bytes"
+	"encoding/json"
 	"fmt"
 	"log"
-	"net"
 	"net/http"
-	"strings"
-	"sync"
 	"time"

-	. "github.com/Monibuca/engine/v2"
-	. "github.com/Monibuca/engine/v2/avformat"
-	"github.com/Monibuca/engine/v2/util"
-	"github.com/pixelbender/go-sdp/sdp"
-	"github.com/teris-io/shortid"
+	. "github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/aler9/gortsplib"
 )

-var collection = sync.Map{}
 var config = struct {
-	ListenAddr   string
-	BufferLength int
-	AutoPull     bool
-	RemoteAddr   string
-	Timeout      int
-}{":554", 2048, true, "rtsp://localhost/${streamPath}", 0}
+	ListenAddr     string
+	UDPAddr        string
+	RTCPAddr       string
+	Timeout        int
+	Reconnect      bool
+	AutoPullList   map[string]string
+	AutoPushList   map[string]string
+	ReadBufferSize int
+}{":554", ":8000", ":8001", 0, false, nil, nil, 2048}
+
+var pconfig = PluginConfig{
+	Name:   "RTSP",
+	Config: &config,
+}

 func init() {
-	InstallPlugin(&PluginConfig{
-		Name:   "RTSP",
-		Type:   PLUGIN_PUBLISHER | PLUGIN_HOOK,
-		Config: &config,
-		Run:    runPlugin,
-		HotConfig: map[string]func(interface{}){
-			"AutoPull": func(value interface{}) {
-				config.AutoPull = value.(bool)
-			},
-		},
-	})
+	pconfig.Install(runPlugin)
+}
+
+func getRtspList() (info []*Stream) {
+	for _, s := range Streams.ToList() {
+		switch s.ExtraProp.(type) {
+		case *RTSPublisher:
+			info = append(info, s)
+		case *RTSPClient:
+			info = append(info, s)
+		}
+	}
+	return
 }
 func runPlugin() {
-	OnSubscribeHooks.AddHook(func(s *Subscriber) {
-		if config.AutoPull && s.Publisher == nil {
-			new(RTSP).PullStream(s.StreamPath, strings.Replace(config.RemoteAddr, "${streamPath}", s.StreamPath, -1))
+	http.HandleFunc("/api/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
+		CORS(w, r)
+		if r.URL.Query().Get("json") != "" {
+			if jsonData, err := json.Marshal(getRtspList()); err == nil {
+				w.Write(jsonData)
+			} else {
+				w.WriteHeader(500)
+			}
+			return
 		}
-	})
-	http.HandleFunc("/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
-		sse := util.NewSSE(w, r.Context())
+		sse := NewSSE(w, r.Context())
 		var err error
 		for tick := time.NewTicker(time.Second); err == nil; <-tick.C {
-			var info []*RTSPInfo
-			collection.Range(func(key, value interface{}) bool {
-				rtsp := value.(*RTSP)
-				pinfo := &rtsp.RTSPInfo
-				//pinfo.BufferRate = len(rtsp.OutGoing) * 100 / config.BufferLength
-				info = append(info, pinfo)
-				return true
-			})
-			err = sse.WriteJSON(info)
+			err = sse.WriteJSON(getRtspList())
 		}
 	})
-	http.HandleFunc("/rtsp/pull", func(w http.ResponseWriter, r *http.Request) {
+	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")
-		var err error
-		if err == nil {
-			new(RTSP).PullStream(streamPath, targetURL)
+		save := r.URL.Query().Get("save")
+		if err := (&RTSPClient{Transport: gortsplib.TransportTCP}).PullStream(streamPath, targetURL); err == nil {
+			if save == "1" {
+				if config.AutoPullList == nil {
+					config.AutoPullList = make(map[string]string)
+				}
+				config.AutoPullList[streamPath] = targetURL
+				if err := pconfig.Save(); err != nil {
+					Println(err)
+				}
+			}
 			w.Write([]byte(`{"code":0}`))
 		} else {
 			w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
 		}
 	})
+	for streamPath, url := range config.AutoPullList {
+		if err := (&RTSPClient{Transport: gortsplib.TransportTCP}).PullStream(streamPath, url); err != nil {
+			Println(err)
+		}
+	}
+
+	go AddHook(HOOK_PUBLISH, func(s *Stream) {
+		for streamPath, url := range config.AutoPushList {
+			if s.StreamPath == streamPath {
+				(&RTSPClient{}).PushStream(streamPath, url)
+			}
+		}
+	})
 	if config.ListenAddr != "" {
-		log.Fatal(ListenRtsp(config.ListenAddr))
+		go log.Fatal(ListenRtsp(config.ListenAddr))
 	}
 }

 func ListenRtsp(addr string) error {
 	defer log.Println("rtsp server start!")
-	listener, err := net.Listen("tcp", addr)
-	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 {
-	Publisher
-	RTSPInfo
-	RTSPClientInfo
-	ID        string
-	Conn      *RichConn
-	connRW    *bufio.ReadWriter
-	connWLock sync.RWMutex
-	Type      SessionType
-	TransType TransType
-	SDPRaw    string
-	SDPMap    map[string]*SDPInfo
-	nonce     string
-	closeOld  bool
-	AControl  string
-	VControl  string
-	ACodec    string
-	VCodec    string
-	avcsent   bool
-	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
-}
-type RTSPClientInfo struct {
-	Agent     string
-	Session   string
-	Sdp       *sdp.Session
-	authLine  string
-	Seq       int
-}
-type RTSPInfo struct {
-	URL        string
-	SyncCount  int64
-	Header     *string
-	BufferRate int
-	InBytes    int
-	OutBytes   int
-
-	StreamInfo *StreamInfo
-}
-
-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))
-	} else {
-		var t time.Time
-		conn.Conn.SetReadDeadline(t)
-	}
-	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)
-}
-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)
-			}
-		}
+	s := &gortsplib.Server{
+		Handler:           &RTSPServer{},
+		RTSPAddress:       addr,
+		UDPRTPAddress:     config.UDPAddr,
+		UDPRTCPAddress:    config.RTCPAddr,
+		MulticastIPRange:  "224.1.0.0/16",
+		MulticastRTPPort:  8002,
+		MulticastRTCPPort: 8003,
 	}
+	return s.StartAndWait()
 }
--- a/payloader.go
+++ b/payloader.go
@@ -0,0 +1,24 @@
+package rtsp
+
+// AACayloader payloads AAC packets
+type AACPayloader struct{}
+
+// Payload fragments an AAC packet across one or more byte arrays
+func (p *AACPayloader) Payload(mtu uint16, payload []byte) [][]byte {
+	var out [][]byte
+	o := make([]byte, len(payload)+4)
+	//AU_HEADER_LENGTH,因为单位是bit, 除以8就是auHeader的字节长度；又因为单个auheader字节长度2字节，所以再除以2就是auheader的个数。
+	o[0] = 0x00 //高位
+	o[1] = 0x10 //低位
+	//AU_HEADER
+	o[2] = (byte)((len(payload) & 0x1fe0) >> 5) //高位
+	o[3] = (byte)((len(payload) & 0x1f) << 3)   //低位
+	copy(o[4:], payload)
+	return append(out, o)
+}
+
+type H265Payloader struct{}
+
+func (p *H265Payloader) Payload(mtu uint16, payload []byte) [][]byte {
+	return nil
+}
--- a/publisher.go
+++ b/publisher.go
@@ -0,0 +1,121 @@
+package rtsp
+
+import (
+	"encoding/base64"
+	"encoding/hex"
+	"strconv"
+	"strings"
+
+	. "github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/aler9/gortsplib"
+)
+
+type RTSPublisher struct {
+	*Stream     `json:"-"`
+	stream      *gortsplib.ServerStream
+	processFunc []func([]byte)
+}
+
+func (p *RTSPublisher) setTracks(tracks gortsplib.Tracks) {
+	if p.processFunc != nil {
+		p.processFunc = p.processFunc[:len(tracks)]
+		return
+	} else {
+		p.processFunc = make([]func([]byte), len(tracks))
+	}
+	for i, track := range tracks {
+		v, ok := track.Media.Attribute("rtpmap")
+		if !ok {
+			continue
+		}
+
+		fmtp := make(map[string]string)
+		if v, ok := track.Media.Attribute("fmtp"); ok {
+			if tmp := strings.SplitN(v, " ", 2); len(tmp) == 2 {
+				for _, kv := range strings.Split(tmp[1], ";") {
+					kv = strings.Trim(kv, " ")
+
+					if len(kv) == 0 {
+						continue
+					}
+					tmp := strings.SplitN(kv, "=", 2)
+					if len(tmp) == 2 {
+						fmtp[strings.TrimSpace(tmp[0])] = tmp[1]
+					}
+				}
+			}
+		}
+
+		v = strings.TrimSpace(v)
+		vals := strings.Split(v, " ")
+		if len(vals) != 2 {
+			continue
+		}
+		timeScale := 0
+		keyval := strings.Split(vals[1], "/")
+		if i, err := strconv.Atoi(keyval[1]); err == nil {
+			timeScale = i
+		}
+		if len(keyval) >= 2 {
+			Printf("track %d is %s", i, keyval[0])
+			switch strings.ToLower(keyval[0]) {
+			case "h264":
+				vt := p.NewRTPVideo(7)
+				if conf, err := track.ExtractConfigH264(); err == nil {
+					vt.PushNalu(0, 0, conf.SPS, conf.PPS)
+				}
+				p.processFunc[i] = vt.Push
+			case "h265", "hevc":
+				vt := p.NewRTPVideo(12)
+				if v, ok := fmtp["sprop-vps"]; ok {
+					vps, _ := base64.StdEncoding.DecodeString(v)
+					vt.PushNalu(0, 0, vps)
+				}
+				if v, ok := fmtp["sprop-sps"]; ok {
+					sps, _ := base64.StdEncoding.DecodeString(v)
+					vt.PushNalu(0, 0, sps)
+				}
+				if v, ok := fmtp["sprop-pps"]; ok {
+					pps, _ := base64.StdEncoding.DecodeString(v)
+					vt.PushNalu(0, 0, pps)
+				}
+				p.processFunc[i] = vt.Push
+			case "pcma":
+				at := p.NewRTPAudio(7)
+				at.SoundRate = timeScale
+				at.SoundSize = 16
+				if len(keyval) >= 3 {
+					x, _ := strconv.Atoi(keyval[2])
+					at.Channels = byte(x)
+				} else {
+					at.Channels = 1
+				}
+				at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
+				p.processFunc[i] = at.Push
+			case "pcmu":
+				at := p.NewRTPAudio(8)
+				at.SoundRate = timeScale
+				at.SoundSize = 16
+				if len(keyval) >= 3 {
+					x, _ := strconv.Atoi(keyval[2])
+					at.Channels = byte(x)
+				} else {
+					at.Channels = 1
+				}
+				at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
+				p.processFunc[i] = at.Push
+			case "mpeg4-generic":
+				at := p.NewRTPAudio(0)
+				if config, ok := fmtp["config"]; ok {
+					asc, _ := hex.DecodeString(config)
+					at.SetASC(asc)
+				} else {
+					Println("aac no config")
+				}
+				at.SoundSize = 16
+				p.processFunc[i] = at.Push
+			}
+		}
+	}
+}
--- 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
@@ -1,105 +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
-	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 {
-					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])
-						}
-					}
-				}
-
-			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 {
-							key := keyval[0]
-							switch key {
-							case "MPEG4-GENERIC":
-								info.Codec = "aac"
-							case "H264":
-								info.Codec = "h264"
-							case "H265":
-								info.Codec = "h265"
-							}
-							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-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 +1,264 @@
+package rtsp
+
+import (
+	"fmt"
+	"sync"
+	"unsafe"
+
+	"github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/Monibuca/utils/v3/codec"
+	"github.com/aler9/gortsplib"
+	"github.com/aler9/gortsplib/pkg/aac"
+	"github.com/aler9/gortsplib/pkg/base"
+	"github.com/pion/rtp"
+	"github.com/pion/rtp/codecs"
+)
+
+// 接收RTSP推流：OnConnOpen->OnAnnounce->OnSetup->OnSessionOpen
+// 接收RTSP拉流：OnConnOpen->OnDescribe->OnSetup->OnSessionOpen
+type RTSPServer struct {
+	sync.Map
+}
+type RTSPSubscriber struct {
+	stream *gortsplib.ServerStream
+	engine.Subscriber
+	vt *engine.VideoTrack
+	at *engine.AudioTrack
+}
+
+// called after a connection is opened.
+func (sh *RTSPServer) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	Printf("rtsp conn opened")
+}
+
+// called after a connection is closed.
+func (sh *RTSPServer) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
+	Printf("rtsp conn closed (%v)", ctx.Error)
+	if p, ok := sh.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPublisher:
+			v.Close()
+		case *RTSPSubscriber:
+			v.Close()
+		}
+		sh.Delete(ctx.Conn)
+	}
+}
+
+// called after a session is opened.
+func (sh *RTSPServer) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	Printf("rtsp session opened")
+}
+
+// called after a session is closed.
+func (sh *RTSPServer) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
+	Printf("rtsp session closed")
+	if v, ok := sh.LoadAndDelete(ctx.Session); ok {
+		switch v := v.(type) {
+		case *RTSPublisher:
+			v.Close()
+		case *RTSPSubscriber:
+			v.Close()
+		}
+	}
+}
+
+// called after receiving a DESCRIBE request.
+func (sh *RTSPServer) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
+	Printf("describe request")
+	var err error
+	if s := engine.FindStream(ctx.Path); s != nil {
+		var tracks gortsplib.Tracks
+		var stream *gortsplib.ServerStream
+		var sub RTSPSubscriber
+		sub.Type = "RTSP pull"
+		sub.vt = s.WaitVideoTrack("h264", "h265")
+		sub.at = s.WaitAudioTrack("aac", "pcma", "pcmu")
+		ssrc := uintptr(unsafe.Pointer(&stream))
+		var trackIds = 0
+		if sub.vt != nil {
+			trackId := trackIds
+			var vtrack *gortsplib.Track
+			var vpacketer rtp.Packetizer
+			switch sub.vt.CodecID {
+			case codec.CodecID_H264:
+				if vtrack, err = gortsplib.NewTrackH264(96, &gortsplib.TrackConfigH264{
+					SPS: sub.vt.ExtraData.NALUs[0],
+					PPS: sub.vt.ExtraData.NALUs[1],
+				}); err == nil {
+					vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &codecs.H264Payloader{}, rtp.NewFixedSequencer(1), 90000)
+				} else {
+					return nil, nil, err
+				}
+			case codec.CodecID_H265:
+				vtrack = NewH265Track(96, sub.vt.ExtraData.NALUs)
+				vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &H265Payloader{}, rtp.NewFixedSequencer(1), 90000)
+			}
+			var st uint32
+			onVideo := func(ts uint32, pack *engine.VideoPack) {
+				for i, nalu := range pack.NALUs {
+					var samples uint32
+					if i == len(pack.NALUs)-1 {
+						samples = (ts - st) * 90
+					} else {
+						samples = 0
+					}
+					packs := vpacketer.Packetize(nalu, samples)
+					for j, rtpack := range packs {
+						rtpack.Marker = i == len(pack.NALUs)-1 && j == len(packs)-1
+						rtp, _ := rtpack.Marshal()
+						stream.WritePacketRTP(trackId, rtp)
+					}
+				}
+				st = ts
+			}
+			sub.OnVideo = func(ts uint32, pack *engine.VideoPack) {
+				if st = ts; st != 0 {
+					sub.OnVideo = onVideo
+				}
+				onVideo(ts, pack)
+			}
+			tracks = append(tracks, vtrack)
+			trackIds++
+		}
+		if sub.at != nil {
+			var st uint32
+			trackId := trackIds
+			switch sub.at.CodecID {
+			case codec.CodecID_PCMA, codec.CodecID_PCMU:
+				atrack := NewG711Track(97, map[byte]string{7: "pcma", 8: "pcmu"}[sub.at.CodecID])
+				apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &codecs.G711Payloader{}, rtp.NewFixedSequencer(1), 8000)
+				sub.OnAudio = func(ts uint32, pack *engine.AudioPack) {
+					for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*8) {
+						buf, _ := pack.Marshal()
+						stream.WritePacketRTP(trackId, buf)
+					}
+					st = ts
+				}
+				tracks = append(tracks, atrack)
+			case codec.CodecID_AAC:
+				var mpegConf aac.MPEG4AudioConfig
+				mpegConf.Decode(sub.at.ExtraData[2:])
+				conf := &gortsplib.TrackConfigAAC{
+					Type:              int(mpegConf.Type),
+					SampleRate:        mpegConf.SampleRate,
+					ChannelCount:      mpegConf.ChannelCount,
+					AOTSpecificConfig: mpegConf.AOTSpecificConfig,
+				}
+				if atrack, err := gortsplib.NewTrackAAC(97, conf); err == nil {
+					apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &AACPayloader{}, rtp.NewFixedSequencer(1), uint32(mpegConf.SampleRate))
+					sub.OnAudio = func(ts uint32, pack *engine.AudioPack) {
+						for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*uint32(mpegConf.SampleRate)/1000) {
+							buf, _ := pack.Marshal()
+							stream.WritePacketRTP(trackId, buf)
+						}
+						st = ts
+					}
+					tracks = append(tracks, atrack)
+				}
+			}
+		}
+		stream = gortsplib.NewServerStream(tracks)
+		sub.stream = stream
+		sh.Store(ctx.Conn, &sub)
+		return &base.Response{
+			StatusCode: base.StatusOK,
+		}, stream, nil
+		// if stream, ok := s.ExtraProp.(*gortsplib.ServerStream); ok {
+		// 	return &base.Response{
+		// 		StatusCode: base.StatusOK,
+		// 	}, stream, nil
+		// }
+	}
+	return &base.Response{
+		StatusCode: base.StatusNotFound,
+	}, nil, nil
+}
+
+// called after receiving an ANNOUNCE request.
+func (sh *RTSPServer) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+	Printf("announce request")
+	p := &RTSPublisher{
+		Stream: &engine.Stream{
+			StreamPath: ctx.Path,
+			Type:       "RTSP push",
+		},
+	}
+	p.ExtraProp = p
+	p.URL = ctx.Req.URL.String()
+	if p.Publish() {
+		p.setTracks(ctx.Tracks)
+		p.stream = gortsplib.NewServerStream(ctx.Tracks)
+		sh.Store(ctx.Conn, p)
+		sh.Store(ctx.Session, p)
+	} else {
+		return &base.Response{
+			StatusCode: base.StatusBadRequest,
+		}, fmt.Errorf("streamPath is already exist")
+	}
+	return &base.Response{
+		StatusCode: base.StatusOK,
+	}, nil
+}
+
+// called after receiving a SETUP request.
+func (sh *RTSPServer) OnSetup(ctx *gortsplib.ServerHandlerOnSetupCtx) (*base.Response, *gortsplib.ServerStream, error) {
+	Printf("setup request")
+	if p, ok := sh.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPublisher:
+			return &base.Response{
+				StatusCode: base.StatusOK,
+			}, v.stream, nil
+		case *RTSPSubscriber:
+			return &base.Response{
+				StatusCode: base.StatusOK,
+			}, v.stream, nil
+		}
+	}
+	return &base.Response{
+		StatusCode: base.StatusNotFound,
+	}, nil, nil
+}
+
+// called after receiving a PLAY request.
+func (sh *RTSPServer) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
+	Printf("play request")
+	if p, ok := sh.Load(ctx.Conn); ok {
+		if sub := p.(*RTSPSubscriber); sub.Subscribe(ctx.Path) == nil {
+			go func() {
+				sub.Play(sub.at, sub.vt)
+				ctx.Conn.Close()
+			}()
+			return &base.Response{
+				StatusCode: base.StatusOK,
+			}, nil
+		}
+	}
+	return &base.Response{
+		StatusCode: base.StatusNotFound,
+	}, nil
+}
+
+// called after receiving a RECORD request.
+func (sh *RTSPServer) OnRecord(ctx *gortsplib.ServerHandlerOnRecordCtx) (*base.Response, error) {
+	Printf("record request")
+	return &base.Response{
+		StatusCode: base.StatusOK,
+	}, nil
+}
+
+// called after receiving a frame.
+func (sh *RTSPServer) OnPacketRTP(ctx *gortsplib.ServerHandlerOnPacketRTPCtx) {
+	if p, ok := sh.Load(ctx.Session); ok {
+		rtsp := p.(*RTSPublisher)
+		if rtsp.Err() != nil {
+			ctx.Session.Close()
+			return
+		}
+		if f := rtsp.processFunc[ctx.TrackID]; f != nil {
+			f(ctx.Payload)
+		}
+	}
+}
--- a/session.go
+++ b/session.go
@@ -1,638 +0,0 @@
-package rtsp
-
-import (
-	"bytes"
-	"crypto/md5"
-	"encoding/binary"
-	"fmt"
-	"io"
-	"net/url"
-	"regexp"
-	"strconv"
-	"strings"
-	"time"
-
-	. "github.com/Monibuca/engine/v2"
-	"github.com/teris-io/shortid"
-)
-
-type RTPPack struct {
-	Type   RTPType
-	Buffer []byte
-}
-
-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
-)
-
-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 (
-	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
-	}
-	if session.Running() {
-		collection.Delete(session.StreamPath)
-		session.Cancel()
-	}
-}
-
-// 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
-			}
-			var pack *RTPPack
-			switch channel {
-			case session.aRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBytes,
-				}
-				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,
-				}
-			case session.vRTPChannel:
-				pack = &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBytes,
-				}
-				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,
-				}
-			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)
-		} 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)
-		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) {
-			session.Stream.Type = "RTSP"
-			session.RTSPInfo.StreamInfo = &session.Stream.StreamInfo
-			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
-		//}
-		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
-			}
-			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 aControlUrl.Port() == "" {
-				aControlUrl.Host = fmt.Sprintf("%s:554", aControlUrl.Host)
-			}
-			aPath = aControlUrl.String()
-		} else {
-			aPath = session.AControl
-		}
-
-		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.Buffer)
-		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.Buffer)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 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.Buffer)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 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.Buffer)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 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.Buffer)))
-		session.connRW.Write(bufLen)
-		session.connRW.Write(pack.Buffer)
-		session.connRW.Flush()
-		session.connWLock.Unlock()
-		session.OutBytes += len(pack.Buffer) + 4
-	default:
-		err = fmt.Errorf("session tcp send rtp got unkown pack type[%v]", pack.Type)
-	}
-	return
-}
--- a/track.go
+++ b/track.go
@@ -0,0 +1,87 @@
+package rtsp
+
+import (
+	"encoding/base64"
+	"fmt"
+	"strconv"
+
+	"github.com/aler9/gortsplib"
+	psdp "github.com/pion/sdp/v3"
+)
+
+// func NewTrackAAC(payloadType uint8, conf *gortsplib.TrackConfigAAC) (*gortsplib.Track, error) {
+// 	mpegConf, err := aac.MPEG4AudioConfig{
+// 		Type:              aac.MPEG4AudioType(conf.Type),
+// 		SampleRate:        conf.SampleRate,
+// 		ChannelCount:      conf.ChannelCount,
+// 		AOTSpecificConfig: conf.AOTSpecificConfig,
+// 	}.Encode()
+// 	if err != nil {
+// 		return nil, err
+// 	}
+
+// 	typ := strconv.FormatInt(int64(payloadType), 10)
+
+// 	return &gortsplib.Track{
+// 		Media: &psdp.MediaDescription{
+// 			MediaName: psdp.MediaName{
+// 				Media:   "audio",
+// 				Protos:  []string{"RTP", "AVP"},
+// 				Formats: []string{typ},
+// 			},
+// 			Attributes: []psdp.Attribute{
+// 				{
+// 					Key: "rtpmap",
+// 					Value: typ + " mpeg4-generic/" + strconv.FormatInt(int64(conf.SampleRate), 10) +
+// 						"/" + strconv.FormatInt(int64(conf.ChannelCount), 10),
+// 				},
+// 				{
+// 					Key: "fmtp",
+// 					Value: typ + " profile-level-id=1; " +
+// 						"mode=AAC-hbr; " +
+// 						"sizelength=6; " +
+// 						"indexlength=2; " +
+// 						"indexdeltalength=2; " +
+// 						"config=" + hex.EncodeToString(mpegConf),
+// 				},
+// 			},
+// 		},
+// 	}, nil
+// }
+func NewG711Track(payloadType uint8, law string) *gortsplib.Track {
+	return &gortsplib.Track{
+		Media: &psdp.MediaDescription{
+			MediaName: psdp.MediaName{
+				Media:   "audio",
+				Protos:  []string{"RTP", "AVP"},
+				Formats: []string{strconv.FormatInt(int64(payloadType), 10)}},
+			Attributes: []psdp.Attribute{
+				{
+					Key:   "rtpmap",
+					Value: fmt.Sprintf("%d %s/8000/1", payloadType, law),
+				},
+			},
+		},
+	}
+}
+func NewH265Track(payloadType uint8, sprop [][]byte) *gortsplib.Track {
+	return &gortsplib.Track{
+		Media: &psdp.MediaDescription{
+			MediaName: psdp.MediaName{
+				Media:   "video",
+				Protos:  []string{"RTP", "AVP"},
+				Formats: []string{fmt.Sprintf("%d", payloadType)},
+			},
+			Attributes: []psdp.Attribute{
+				{
+					Key:   "rtpmap",
+					Value: fmt.Sprintf("%d H265/90000", payloadType),
+				},
+				{
+					Key:   "fmtp",
+					Value: fmt.Sprintf("%d packetization-mode=1;sprop-vps=%s;sprop-sps=%s;sprop-pps=%s;", payloadType, base64.StdEncoding.EncodeToString(sprop[0]), base64.StdEncoding.EncodeToString(sprop[1]), base64.StdEncoding.EncodeToString(sprop[2])),
+				},
+			},
+		},
+	}
+}
--- a/udp-client.go
+++ b/udp-client.go
@@ -1,160 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"net"
-	"strings"
-	. "github.com/Monibuca/engine/v2"
-)
-
-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.Buffer);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
@@ -1,242 +0,0 @@
-package rtsp
-
-import (
-	"fmt"
-	"net"
-	"strconv"
-	"strings"
-	"sync"
-	"time"
-
-	. "github.com/Monibuca/engine/v2"
-)
-
-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) HandleRTP(pack *RTPPack) {
-	s.Lock()
-	defer s.Unlock()
-	if s.Session != nil {
-		s.Session.handleRTP(pack)
-	}
-}
-
-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()
-				}
-				rtpBytes := make([]byte, n)
-				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
-				pack := &RTPPack{
-					Type:   RTP_TYPE_AUDIO,
-					Buffer: rtpBytes,
-				}
-				s.HandleRTP(pack)
-			} 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)
-				rtpBytes := make([]byte, n)
-				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
-				pack := &RTPPack{
-					Type:   RTP_TYPE_AUDIOCONTROL,
-					Buffer: rtpBytes,
-				}
-				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()
-				}
-				rtpBytes := make([]byte, n)
-				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
-				pack := &RTPPack{
-					Type:   RTP_TYPE_VIDEO,
-					Buffer: rtpBytes,
-				}
-				s.HandleRTP(pack)
-			} 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)
-				rtpBytes := make([]byte, n)
-				s.AddInputBytes(n)
-				copy(rtpBytes, bufUDP)
-				pack := &RTPPack{
-					Type:   RTP_TYPE_VIDEOCONTROL,
-					Buffer: rtpBytes,
-				}
-				s.HandleRTP(pack)
-			} else {
-				Println("udp server read video control pack error", err)
-				continue
-			}
-		}
-	}()
-	return
-}
--- a/ui/dist/demo.html
+++ b/ui/dist/demo.html
@@ -1,19 +0,0 @@
-<meta charset="utf-8">
-<title>plugin-rtsp demo</title>
-<script src="https://unpkg.com/vue"></script>
-<script src="./plugin-rtsp.umd.js"></script>
-
-<link rel="stylesheet" href="./plugin-rtsp.css">
-
-
-<div id="app">
-  <demo></demo>
-</div>
-
-<script>
-new Vue({
-  components: {
-    demo: plugin-rtsp
-  }
-}).$mount('#app')
-</script>
--- a/ui/dist/plugin-rtsp.common.js
+++ b/ui/dist/plugin-rtsp.common.js
@@ -1,415 +0,0 @@
-module.exports =
-/******/ (function(modules) { // webpackBootstrap
-/******/ 	// The module cache
-/******/ 	var installedModules = {};
-/******/
-/******/ 	// The require function
-/******/ 	function __webpack_require__(moduleId) {
-/******/
-/******/ 		// Check if module is in cache
-/******/ 		if(installedModules[moduleId]) {
-/******/ 			return installedModules[moduleId].exports;
-/******/ 		}
-/******/ 		// Create a new module (and put it into the cache)
-/******/ 		var module = installedModules[moduleId] = {
-/******/ 			i: moduleId,
-/******/ 			l: false,
-/******/ 			exports: {}
-/******/ 		};
-/******/
-/******/ 		// Execute the module function
-/******/ 		modules[moduleId].call(module.exports, module, module.exports, __webpack_require__);
-/******/
-/******/ 		// Flag the module as loaded
-/******/ 		module.l = true;
-/******/
-/******/ 		// Return the exports of the module
-/******/ 		return module.exports;
-/******/ 	}
-/******/
-/******/
-/******/ 	// expose the modules object (__webpack_modules__)
-/******/ 	__webpack_require__.m = modules;
-/******/
-/******/ 	// expose the module cache
-/******/ 	__webpack_require__.c = installedModules;
-/******/
-/******/ 	// define getter function for harmony exports
-/******/ 	__webpack_require__.d = function(exports, name, getter) {
-/******/ 		if(!__webpack_require__.o(exports, name)) {
-/******/ 			Object.defineProperty(exports, name, { enumerable: true, get: getter });
-/******/ 		}
-/******/ 	};
-/******/
-/******/ 	// define __esModule on exports
-/******/ 	__webpack_require__.r = function(exports) {
-/******/ 		if(typeof Symbol !== 'undefined' && Symbol.toStringTag) {
-/******/ 			Object.defineProperty(exports, Symbol.toStringTag, { value: 'Module' });
-/******/ 		}
-/******/ 		Object.defineProperty(exports, '__esModule', { value: true });
-/******/ 	};
-/******/
-/******/ 	// create a fake namespace object
-/******/ 	// mode & 1: value is a module id, require it
-/******/ 	// mode & 2: merge all properties of value into the ns
-/******/ 	// mode & 4: return value when already ns object
-/******/ 	// mode & 8|1: behave like require
-/******/ 	__webpack_require__.t = function(value, mode) {
-/******/ 		if(mode & 1) value = __webpack_require__(value);
-/******/ 		if(mode & 8) return value;
-/******/ 		if((mode & 4) && typeof value === 'object' && value && value.__esModule) return value;
-/******/ 		var ns = Object.create(null);
-/******/ 		__webpack_require__.r(ns);
-/******/ 		Object.defineProperty(ns, 'default', { enumerable: true, value: value });
-/******/ 		if(mode & 2 && typeof value != 'string') for(var key in value) __webpack_require__.d(ns, key, function(key) { return value[key]; }.bind(null, key));
-/******/ 		return ns;
-/******/ 	};
-/******/
-/******/ 	// getDefaultExport function for compatibility with non-harmony modules
-/******/ 	__webpack_require__.n = function(module) {
-/******/ 		var getter = module && module.__esModule ?
-/******/ 			function getDefault() { return module['default']; } :
-/******/ 			function getModuleExports() { return module; };
-/******/ 		__webpack_require__.d(getter, 'a', getter);
-/******/ 		return getter;
-/******/ 	};
-/******/
-/******/ 	// Object.prototype.hasOwnProperty.call
-/******/ 	__webpack_require__.o = function(object, property) { return Object.prototype.hasOwnProperty.call(object, property); };
-/******/
-/******/ 	// __webpack_public_path__
-/******/ 	__webpack_require__.p = "";
-/******/
-/******/
-/******/ 	// Load entry module and return exports
-/******/ 	return __webpack_require__(__webpack_require__.s = "fb15");
-/******/ })
-/************************************************************************/
-/******/ ({
-
-/***/ "034f":
-/***/ (function(module, __webpack_exports__, __webpack_require__) {
-
-"use strict";
-/* harmony import */ var _node_modules_mini_css_extract_plugin_dist_loader_js_ref_6_oneOf_1_0_node_modules_css_loader_dist_cjs_js_ref_6_oneOf_1_1_node_modules_vue_loader_lib_loaders_stylePostLoader_js_node_modules_postcss_loader_src_index_js_ref_6_oneOf_1_2_node_modules_cache_loader_dist_cjs_js_ref_0_0_node_modules_vue_loader_lib_index_js_vue_loader_options_App_vue_vue_type_style_index_0_lang_css___WEBPACK_IMPORTED_MODULE_0__ = __webpack_require__("85ec");
-/* harmony import */ var _node_modules_mini_css_extract_plugin_dist_loader_js_ref_6_oneOf_1_0_node_modules_css_loader_dist_cjs_js_ref_6_oneOf_1_1_node_modules_vue_loader_lib_loaders_stylePostLoader_js_node_modules_postcss_loader_src_index_js_ref_6_oneOf_1_2_node_modules_cache_loader_dist_cjs_js_ref_0_0_node_modules_vue_loader_lib_index_js_vue_loader_options_App_vue_vue_type_style_index_0_lang_css___WEBPACK_IMPORTED_MODULE_0___default = /*#__PURE__*/__webpack_require__.n(_node_modules_mini_css_extract_plugin_dist_loader_js_ref_6_oneOf_1_0_node_modules_css_loader_dist_cjs_js_ref_6_oneOf_1_1_node_modules_vue_loader_lib_loaders_stylePostLoader_js_node_modules_postcss_loader_src_index_js_ref_6_oneOf_1_2_node_modules_cache_loader_dist_cjs_js_ref_0_0_node_modules_vue_loader_lib_index_js_vue_loader_options_App_vue_vue_type_style_index_0_lang_css___WEBPACK_IMPORTED_MODULE_0__);
-/* unused harmony reexport * */
- /* unused harmony default export */ var _unused_webpack_default_export = (_node_modules_mini_css_extract_plugin_dist_loader_js_ref_6_oneOf_1_0_node_modules_css_loader_dist_cjs_js_ref_6_oneOf_1_1_node_modules_vue_loader_lib_loaders_stylePostLoader_js_node_modules_postcss_loader_src_index_js_ref_6_oneOf_1_2_node_modules_cache_loader_dist_cjs_js_ref_0_0_node_modules_vue_loader_lib_index_js_vue_loader_options_App_vue_vue_type_style_index_0_lang_css___WEBPACK_IMPORTED_MODULE_0___default.a); 
-
-/***/ }),
-
-/***/ "85ec":
-/***/ (function(module, exports, __webpack_require__) {
-
-// extracted by mini-css-extract-plugin
-
-/***/ }),
-
-/***/ "f6fd":
-/***/ (function(module, exports) {
-
-// document.currentScript polyfill by Adam Miller
-
-// MIT license
-
-(function(document){
-  var currentScript = "currentScript",
-      scripts = document.getElementsByTagName('script'); // Live NodeList collection
-
-  // If browser needs currentScript polyfill, add get currentScript() to the document object
-  if (!(currentScript in document)) {
-    Object.defineProperty(document, currentScript, {
-      get: function(){
-
-        // IE 6-10 supports script readyState
-        // IE 10+ support stack trace
-        try { throw new Error(); }
-        catch (err) {
-
-          // Find the second match for the "at" string to get file src url from stack.
-          // Specifically works with the format of stack traces in IE.
-          var i, res = ((/.*at [^\(]*\((.*):.+:.+\)$/ig).exec(err.stack) || [false])[1];
-
-          // For all scripts on the page, if src matches or if ready state is interactive, return the script tag
-          for(i in scripts){
-            if(scripts[i].src == res || scripts[i].readyState == "interactive"){
-              return scripts[i];
-            }
-          }
-
-          // If no match, return null
-          return null;
-        }
-      }
-    });
-  }
-})(document);
-
-
-/***/ }),
-
-/***/ "fb15":
-/***/ (function(module, __webpack_exports__, __webpack_require__) {
-
-"use strict";
-// ESM COMPAT FLAG
-__webpack_require__.r(__webpack_exports__);
-
-// CONCATENATED MODULE: ./node_modules/@vue/cli-service/lib/commands/build/setPublicPath.js
-// This file is imported into lib/wc client bundles.
-
-if (typeof window !== 'undefined') {
-  if (true) {
-    __webpack_require__("f6fd")
-  }
-
-  var i
-  if ((i = window.document.currentScript) && (i = i.src.match(/(.+\/)[^/]+\.js(\?.*)?$/))) {
-    __webpack_require__.p = i[1] // eslint-disable-line
-  }
-}
-
-// Indicate to webpack that this file can be concatenated
-/* harmony default export */ var setPublicPath = (null);
-
-// CONCATENATED MODULE: ./node_modules/cache-loader/dist/cjs.js?{"cacheDirectory":"node_modules/.cache/vue-loader","cacheIdentifier":"29918b3a-vue-loader-template"}!./node_modules/vue-loader/lib/loaders/templateLoader.js??vue-loader-options!./node_modules/cache-loader/dist/cjs.js??ref--0-0!./node_modules/vue-loader/lib??vue-loader-options!./src/App.vue?vue&type=template&id=3ee6bce0&
-var render = function () {var _vm=this;var _h=_vm.$createElement;var _c=_vm._self._c||_h;return _c('div',[_c('mu-data-table',{attrs:{"data":_vm.Streams,"columns":_vm.columns},scopedSlots:_vm._u([{key:"default",fn:function(ref){
-var item = ref.row;
-return [_c('td',[_vm._v(_vm._s(item.StreamInfo.StreamPath))]),_c('td',[_c('StartTime',{attrs:{"value":item.StreamInfo.StartTime}})],1),_c('td',[_c('Progress',{attrs:{"stroke-width":20,"percent":Math.ceil(item.BufferRate),"text-inside":""}})],1),_c('td',[_vm._v(_vm._s(item.SyncCount))]),_c('td',[_c('mu-button',{attrs:{"flat":""},on:{"click":function($event){return _vm.showHeader(item)}}},[_vm._v("头信息")])],1)]}}])}),_c('mu-dialog',{attrs:{"title":"拉流转发","width":"360","open":_vm.openPull},on:{"update:open":function($event){_vm.openPull=$event}}},[_c('mu-text-field',{attrs:{"label":"rtsp url","label-float":"","help-text":"Please enter URL of rtsp..."},model:{value:(_vm.remoteAddr),callback:function ($$v) {_vm.remoteAddr=$$v},expression:"remoteAddr"}}),_c('mu-text-field',{attrs:{"label":"streamPath","label-float":"","help-text":"Please enter streamPath to publish."},model:{value:(_vm.streamPath),callback:function ($$v) {_vm.streamPath=$$v},expression:"streamPath"}}),_c('mu-button',{attrs:{"slot":"actions","flat":"","color":"primary"},on:{"click":_vm.addPull},slot:"actions"},[_vm._v("确定")])],1)],1)}
-var staticRenderFns = []
-
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=template&id=3ee6bce0&
-
-// CONCATENATED MODULE: ./node_modules/cache-loader/dist/cjs.js??ref--0-0!./node_modules/vue-loader/lib??vue-loader-options!./src/App.vue?vue&type=script&lang=js&
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-
-let listES = null;
-/* harmony default export */ var Appvue_type_script_lang_js_ = ({
-    data() {
-        return {
-            currentStream: null,
-            Streams: null,
-            remoteAddr: "",
-            streamPath: "",
-            openPull: false,
-            columns: [
-                "StreamPath",
-                "开始时间",
-                "缓冲",
-                "同步数",
-                "操作"
-            ].map(title => ({ title }))
-        };
-    },
-
-    methods: {
-        fetchlist() {
-            listES = new EventSource(this.apiHost + "/rtsp/list");
-            listES.onmessage = evt => {
-                if (!evt.data) return;
-                this.Streams = JSON.parse(evt.data) || [];
-                this.Streams.sort((a, b) =>
-                    a.StreamInfo.StreamPath > b.StreamInfo.StreamPath ? 1 : -1
-                );
-            };
-        },
-        showHeader(item) {
-            this.$Modal.info({
-                title: "RTSP Header",
-                width: "1000px",
-                scrollable: true,
-                content: item.Header
-            });
-        },
-        addPull() {
-            this.openPull = false;
-            this.ajax
-                .getJSON(this.apiHost + "/rtsp/pull", {
-                    target: this.remoteAddr,
-                    streamPath: this.streamPath
-                })
-                .then(x => {
-                    if (x.code == 0) {
-                        this.$toast.success("已启动拉流");
-                    } else {
-                        this.$toast.error(x.msg);
-                    }
-                });
-        }
-    },
-    mounted() {
-        this.fetchlist();
-        let _this = this;
-        this.$parent.titleOps = [
-            {
-                template: '<m-button @click="onClick">拉流转发</m-button>',
-                methods: {
-                    onClick() {
-                        _this.openPull = true;
-                    }
-                }
-            }
-        ];
-    },
-    destroyed() {
-        listES.close();
-    }
-});
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=script&lang=js&
- /* harmony default export */ var src_Appvue_type_script_lang_js_ = (Appvue_type_script_lang_js_); 
-// EXTERNAL MODULE: ./src/App.vue?vue&type=style&index=0&lang=css&
-var Appvue_type_style_index_0_lang_css_ = __webpack_require__("034f");
-
-// CONCATENATED MODULE: ./node_modules/vue-loader/lib/runtime/componentNormalizer.js
-/* globals __VUE_SSR_CONTEXT__ */
-
-// IMPORTANT: Do NOT use ES2015 features in this file (except for modules).
-// This module is a runtime utility for cleaner component module output and will
-// be included in the final webpack user bundle.
-
-function normalizeComponent (
-  scriptExports,
-  render,
-  staticRenderFns,
-  functionalTemplate,
-  injectStyles,
-  scopeId,
-  moduleIdentifier, /* server only */
-  shadowMode /* vue-cli only */
-) {
-  // Vue.extend constructor export interop
-  var options = typeof scriptExports === 'function'
-    ? scriptExports.options
-    : scriptExports
-
-  // render functions
-  if (render) {
-    options.render = render
-    options.staticRenderFns = staticRenderFns
-    options._compiled = true
-  }
-
-  // functional template
-  if (functionalTemplate) {
-    options.functional = true
-  }
-
-  // scopedId
-  if (scopeId) {
-    options._scopeId = 'data-v-' + scopeId
-  }
-
-  var hook
-  if (moduleIdentifier) { // server build
-    hook = function (context) {
-      // 2.3 injection
-      context =
-        context || // cached call
-        (this.$vnode && this.$vnode.ssrContext) || // stateful
-        (this.parent && this.parent.$vnode && this.parent.$vnode.ssrContext) // functional
-      // 2.2 with runInNewContext: true
-      if (!context && typeof __VUE_SSR_CONTEXT__ !== 'undefined') {
-        context = __VUE_SSR_CONTEXT__
-      }
-      // inject component styles
-      if (injectStyles) {
-        injectStyles.call(this, context)
-      }
-      // register component module identifier for async chunk inferrence
-      if (context && context._registeredComponents) {
-        context._registeredComponents.add(moduleIdentifier)
-      }
-    }
-    // used by ssr in case component is cached and beforeCreate
-    // never gets called
-    options._ssrRegister = hook
-  } else if (injectStyles) {
-    hook = shadowMode
-      ? function () { injectStyles.call(this, this.$root.$options.shadowRoot) }
-      : injectStyles
-  }
-
-  if (hook) {
-    if (options.functional) {
-      // for template-only hot-reload because in that case the render fn doesn't
-      // go through the normalizer
-      options._injectStyles = hook
-      // register for functional component in vue file
-      var originalRender = options.render
-      options.render = function renderWithStyleInjection (h, context) {
-        hook.call(context)
-        return originalRender(h, context)
-      }
-    } else {
-      // inject component registration as beforeCreate hook
-      var existing = options.beforeCreate
-      options.beforeCreate = existing
-        ? [].concat(existing, hook)
-        : [hook]
-    }
-  }
-
-  return {
-    exports: scriptExports,
-    options: options
-  }
-}
-
-// CONCATENATED MODULE: ./src/App.vue
-
-
-
-
-
-
-/* normalize component */
-
-var component = normalizeComponent(
-  src_Appvue_type_script_lang_js_,
-  render,
-  staticRenderFns,
-  false,
-  null,
-  null,
-  null
-  
-)
-
-/* harmony default export */ var App = (component.exports);
-// CONCATENATED MODULE: ./node_modules/@vue/cli-service/lib/commands/build/entry-lib.js
-
-
-/* harmony default export */ var entry_lib = __webpack_exports__["default"] = (App);
-
-
-
-/***/ })
-
-/******/ })["default"];
-//# sourceMappingURL=plugin-rtsp.common.js.map
--- a/ui/dist/plugin-rtsp.common.js.map
+++ b/ui/dist/plugin-rtsp.common.js.map
--- a/ui/dist/plugin-rtsp.css
+++ b/ui/dist/plugin-rtsp.css
@@ -1 +0,0 @@
-.empty{color:#eb5e46;width:100%;min-height:500px;display:flex;justify-content:center;align-items:center}.layout{padding-bottom:30px;display:flex;flex-wrap:wrap}.ts-info{width:300px}.hls-info{width:350px;display:flex;flex-direction:column}
--- a/ui/dist/plugin-rtsp.umd.js
+++ b/ui/dist/plugin-rtsp.umd.js
@@ -1,425 +0,0 @@
-(function webpackUniversalModuleDefinition(root, factory) {
-	if(typeof exports === 'object' && typeof module === 'object')
-		module.exports = factory();
-	else if(typeof define === 'function' && define.amd)
-		define([], factory);
-	else if(typeof exports === 'object')
-		exports["plugin-rtsp"] = factory();
-	else
-		root["plugin-rtsp"] = factory();
-})((typeof self !== 'undefined' ? self : this), function() {
-return /******/ (function(modules) { // webpackBootstrap
-/******/ 	// The module cache
-/******/ 	var installedModules = {};
-/******/
-/******/ 	// The require function
-/******/ 	function __webpack_require__(moduleId) {
-/******/
-/******/ 		// Check if module is in cache
-/******/ 		if(installedModules[moduleId]) {
-/******/ 			return installedModules[moduleId].exports;
-/******/ 		}
-/******/ 		// Create a new module (and put it into the cache)
-/******/ 		var module = installedModules[moduleId] = {
-/******/ 			i: moduleId,
-/******/ 			l: false,
-/******/ 			exports: {}
-/******/ 		};
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-/***/ "f6fd":
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-(function(document){
-  var currentScript = "currentScript",
-      scripts = document.getElementsByTagName('script'); // Live NodeList collection
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-/***/ }),
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-/***/ "fb15":
-/***/ (function(module, __webpack_exports__, __webpack_require__) {
-
-"use strict";
-// ESM COMPAT FLAG
-__webpack_require__.r(__webpack_exports__);
-
-// CONCATENATED MODULE: ./node_modules/@vue/cli-service/lib/commands/build/setPublicPath.js
-// This file is imported into lib/wc client bundles.
-
-if (typeof window !== 'undefined') {
-  if (true) {
-    __webpack_require__("f6fd")
-  }
-
-  var i
-  if ((i = window.document.currentScript) && (i = i.src.match(/(.+\/)[^/]+\.js(\?.*)?$/))) {
-    __webpack_require__.p = i[1] // eslint-disable-line
-  }
-}
-
-// Indicate to webpack that this file can be concatenated
-/* harmony default export */ var setPublicPath = (null);
-
-// CONCATENATED MODULE: ./node_modules/cache-loader/dist/cjs.js?{"cacheDirectory":"node_modules/.cache/vue-loader","cacheIdentifier":"29918b3a-vue-loader-template"}!./node_modules/vue-loader/lib/loaders/templateLoader.js??vue-loader-options!./node_modules/cache-loader/dist/cjs.js??ref--0-0!./node_modules/vue-loader/lib??vue-loader-options!./src/App.vue?vue&type=template&id=3ee6bce0&
-var render = function () {var _vm=this;var _h=_vm.$createElement;var _c=_vm._self._c||_h;return _c('div',[_c('mu-data-table',{attrs:{"data":_vm.Streams,"columns":_vm.columns},scopedSlots:_vm._u([{key:"default",fn:function(ref){
-var item = ref.row;
-return [_c('td',[_vm._v(_vm._s(item.StreamInfo.StreamPath))]),_c('td',[_c('StartTime',{attrs:{"value":item.StreamInfo.StartTime}})],1),_c('td',[_c('Progress',{attrs:{"stroke-width":20,"percent":Math.ceil(item.BufferRate),"text-inside":""}})],1),_c('td',[_vm._v(_vm._s(item.SyncCount))]),_c('td',[_c('mu-button',{attrs:{"flat":""},on:{"click":function($event){return _vm.showHeader(item)}}},[_vm._v("头信息")])],1)]}}])}),_c('mu-dialog',{attrs:{"title":"拉流转发","width":"360","open":_vm.openPull},on:{"update:open":function($event){_vm.openPull=$event}}},[_c('mu-text-field',{attrs:{"label":"rtsp url","label-float":"","help-text":"Please enter URL of rtsp..."},model:{value:(_vm.remoteAddr),callback:function ($$v) {_vm.remoteAddr=$$v},expression:"remoteAddr"}}),_c('mu-text-field',{attrs:{"label":"streamPath","label-float":"","help-text":"Please enter streamPath to publish."},model:{value:(_vm.streamPath),callback:function ($$v) {_vm.streamPath=$$v},expression:"streamPath"}}),_c('mu-button',{attrs:{"slot":"actions","flat":"","color":"primary"},on:{"click":_vm.addPull},slot:"actions"},[_vm._v("确定")])],1)],1)}
-var staticRenderFns = []
-
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=template&id=3ee6bce0&
-
-// CONCATENATED MODULE: ./node_modules/cache-loader/dist/cjs.js??ref--0-0!./node_modules/vue-loader/lib??vue-loader-options!./src/App.vue?vue&type=script&lang=js&
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-
-let listES = null;
-/* harmony default export */ var Appvue_type_script_lang_js_ = ({
-    data() {
-        return {
-            currentStream: null,
-            Streams: null,
-            remoteAddr: "",
-            streamPath: "",
-            openPull: false,
-            columns: [
-                "StreamPath",
-                "开始时间",
-                "缓冲",
-                "同步数",
-                "操作"
-            ].map(title => ({ title }))
-        };
-    },
-
-    methods: {
-        fetchlist() {
-            listES = new EventSource(this.apiHost + "/rtsp/list");
-            listES.onmessage = evt => {
-                if (!evt.data) return;
-                this.Streams = JSON.parse(evt.data) || [];
-                this.Streams.sort((a, b) =>
-                    a.StreamInfo.StreamPath > b.StreamInfo.StreamPath ? 1 : -1
-                );
-            };
-        },
-        showHeader(item) {
-            this.$Modal.info({
-                title: "RTSP Header",
-                width: "1000px",
-                scrollable: true,
-                content: item.Header
-            });
-        },
-        addPull() {
-            this.openPull = false;
-            this.ajax
-                .getJSON(this.apiHost + "/rtsp/pull", {
-                    target: this.remoteAddr,
-                    streamPath: this.streamPath
-                })
-                .then(x => {
-                    if (x.code == 0) {
-                        this.$toast.success("已启动拉流");
-                    } else {
-                        this.$toast.error(x.msg);
-                    }
-                });
-        }
-    },
-    mounted() {
-        this.fetchlist();
-        let _this = this;
-        this.$parent.titleOps = [
-            {
-                template: '<m-button @click="onClick">拉流转发</m-button>',
-                methods: {
-                    onClick() {
-                        _this.openPull = true;
-                    }
-                }
-            }
-        ];
-    },
-    destroyed() {
-        listES.close();
-    }
-});
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=script&lang=js&
- /* harmony default export */ var src_Appvue_type_script_lang_js_ = (Appvue_type_script_lang_js_); 
-// EXTERNAL MODULE: ./src/App.vue?vue&type=style&index=0&lang=css&
-var Appvue_type_style_index_0_lang_css_ = __webpack_require__("034f");
-
-// CONCATENATED MODULE: ./node_modules/vue-loader/lib/runtime/componentNormalizer.js
-/* globals __VUE_SSR_CONTEXT__ */
-
-// IMPORTANT: Do NOT use ES2015 features in this file (except for modules).
-// This module is a runtime utility for cleaner component module output and will
-// be included in the final webpack user bundle.
-
-function normalizeComponent (
-  scriptExports,
-  render,
-  staticRenderFns,
-  functionalTemplate,
-  injectStyles,
-  scopeId,
-  moduleIdentifier, /* server only */
-  shadowMode /* vue-cli only */
-) {
-  // Vue.extend constructor export interop
-  var options = typeof scriptExports === 'function'
-    ? scriptExports.options
-    : scriptExports
-
-  // render functions
-  if (render) {
-    options.render = render
-    options.staticRenderFns = staticRenderFns
-    options._compiled = true
-  }
-
-  // functional template
-  if (functionalTemplate) {
-    options.functional = true
-  }
-
-  // scopedId
-  if (scopeId) {
-    options._scopeId = 'data-v-' + scopeId
-  }
-
-  var hook
-  if (moduleIdentifier) { // server build
-    hook = function (context) {
-      // 2.3 injection
-      context =
-        context || // cached call
-        (this.$vnode && this.$vnode.ssrContext) || // stateful
-        (this.parent && this.parent.$vnode && this.parent.$vnode.ssrContext) // functional
-      // 2.2 with runInNewContext: true
-      if (!context && typeof __VUE_SSR_CONTEXT__ !== 'undefined') {
-        context = __VUE_SSR_CONTEXT__
-      }
-      // inject component styles
-      if (injectStyles) {
-        injectStyles.call(this, context)
-      }
-      // register component module identifier for async chunk inferrence
-      if (context && context._registeredComponents) {
-        context._registeredComponents.add(moduleIdentifier)
-      }
-    }
-    // used by ssr in case component is cached and beforeCreate
-    // never gets called
-    options._ssrRegister = hook
-  } else if (injectStyles) {
-    hook = shadowMode
-      ? function () { injectStyles.call(this, this.$root.$options.shadowRoot) }
-      : injectStyles
-  }
-
-  if (hook) {
-    if (options.functional) {
-      // for template-only hot-reload because in that case the render fn doesn't
-      // go through the normalizer
-      options._injectStyles = hook
-      // register for functional component in vue file
-      var originalRender = options.render
-      options.render = function renderWithStyleInjection (h, context) {
-        hook.call(context)
-        return originalRender(h, context)
-      }
-    } else {
-      // inject component registration as beforeCreate hook
-      var existing = options.beforeCreate
-      options.beforeCreate = existing
-        ? [].concat(existing, hook)
-        : [hook]
-    }
-  }
-
-  return {
-    exports: scriptExports,
-    options: options
-  }
-}
-
-// CONCATENATED MODULE: ./src/App.vue
-
-
-
-
-
-
-/* normalize component */
-
-var component = normalizeComponent(
-  src_Appvue_type_script_lang_js_,
-  render,
-  staticRenderFns,
-  false,
-  null,
-  null,
-  null
-  
-)
-
-/* harmony default export */ var App = (component.exports);
-// CONCATENATED MODULE: ./node_modules/@vue/cli-service/lib/commands/build/entry-lib.js
-
-
-/* harmony default export */ var entry_lib = __webpack_exports__["default"] = (App);
-
-
-
-/***/ })
-
-/******/ })["default"];
-});
-//# sourceMappingURL=plugin-rtsp.umd.js.map
--- a/ui/dist/plugin-rtsp.umd.js.map
+++ b/ui/dist/plugin-rtsp.umd.js.map
--- a/ui/dist/plugin-rtsp.umd.min.js
+++ b/ui/dist/plugin-rtsp.umd.min.js
@@ -1,2 +0,0 @@
-(function(t,e){"object"===typeof exports&&"object"===typeof module?module.exports=e():"function"===typeof define&&define.amd?define([],e):"object"===typeof exports?exports["plugin-rtsp"]=e():t["plugin-rtsp"]=e()})("undefined"!==typeof self?self:this,(function(){return function(t){var e={};function r(n){if(e[n])return e[n].exports;var o=e[n]={i:n,l:!1,exports:{}};return t[n].call(o.exports,o,o.exports,r),o.l=!0,o.exports}return r.m=t,r.c=e,r.d=function(t,e,n){r.o(t,e)||Object.defineProperty(t,e,{enumerable:!0,get:n})},r.r=function(t){"undefined"!==typeof Symbol&&Symbol.toStringTag&&Object.defineProperty(t,Symbol.toStringTag,{value:"Module"}),Object.defineProperty(t,"__esModule",{value:!0})},r.t=function(t,e){if(1&e&&(t=r(t)),8&e)return t;if(4&e&&"object"===typeof t&&t&&t.__esModule)return t;var n=Object.create(null);if(r.r(n),Object.defineProperty(n,"default",{enumerable:!0,value:t}),2&e&&"string"!=typeof t)for(var o in t)r.d(n,o,function(e){return t[e]}.bind(null,o));return n},r.n=function(t){var e=t&&t.__esModule?function(){return t["default"]}:function(){return t};return r.d(e,"a",e),e},r.o=function(t,e){return Object.prototype.hasOwnProperty.call(t,e)},r.p="",r(r.s="fb15")}({"034f":function(t,e,r){"use strict";var n=r("85ec"),o=r.n(n);o.a},"85ec":function(t,e,r){},f6fd:function(t,e){(function(t){var e="currentScript",r=t.getElementsByTagName("script");e in t||Object.defineProperty(t,e,{get:function(){try{throw new Error}catch(n){var t,e=(/.*at [^\(]*\((.*):.+:.+\)$/gi.exec(n.stack)||[!1])[1];for(t in r)if(r[t].src==e||"interactive"==r[t].readyState)return r[t];return null}}})})(document)},fb15:function(t,e,r){"use strict";var n;(r.r(e),"undefined"!==typeof window)&&(r("f6fd"),(n=window.document.currentScript)&&(n=n.src.match(/(.+\/)[^/]+\.js(\?.*)?$/))&&(r.p=n[1]));var o=function(){var t=this,e=t.$createElement,r=t._self._c||e;return r("div",[r("mu-data-table",{attrs:{data:t.Streams,columns:t.columns},scopedSlots:t._u([{key:"default",fn:function(e){var n=e.row;return[r("td",[t._v(t._s(n.StreamInfo.StreamPath))]),r("td",[r("StartTime",{attrs:{value:n.StreamInfo.StartTime}})],1),r("td",[r("Progress",{attrs:{"stroke-width":20,percent:Math.ceil(n.BufferRate),"text-inside":""}})],1),r("td",[t._v(t._s(n.SyncCount))]),r("td",[r("mu-button",{attrs:{flat:""},on:{click:function(e){return t.showHeader(n)}}},[t._v("头信息")])],1)]}}])}),r("mu-dialog",{attrs:{title:"拉流转发",width:"360",open:t.openPull},on:{"update:open":function(e){t.openPull=e}}},[r("mu-text-field",{attrs:{label:"rtsp url","label-float":"","help-text":"Please enter URL of rtsp..."},model:{value:t.remoteAddr,callback:function(e){t.remoteAddr=e},expression:"remoteAddr"}}),r("mu-text-field",{attrs:{label:"streamPath","label-float":"","help-text":"Please enter streamPath to publish."},model:{value:t.streamPath,callback:function(e){t.streamPath=e},expression:"streamPath"}}),r("mu-button",{attrs:{slot:"actions",flat:"",color:"primary"},on:{click:t.addPull},slot:"actions"},[t._v("确定")])],1)],1)},a=[];let s=null;var i={data(){return{currentStream:null,Streams:null,remoteAddr:"",streamPath:"",openPull:!1,columns:["StreamPath","开始时间","缓冲","同步数","操作"].map(t=>({title:t}))}},methods:{fetchlist(){s=new EventSource(this.apiHost+"/rtsp/list"),s.onmessage=t=>{t.data&&(this.Streams=JSON.parse(t.data)||[],this.Streams.sort((t,e)=>t.StreamInfo.StreamPath>e.StreamInfo.StreamPath?1:-1))}},showHeader(t){this.$Modal.info({title:"RTSP Header",width:"1000px",scrollable:!0,content:t.Header})},addPull(){this.openPull=!1,this.ajax.getJSON(this.apiHost+"/rtsp/pull",{target:this.remoteAddr,streamPath:this.streamPath}).then(t=>{0==t.code?this.$toast.success("已启动拉流"):this.$toast.error(t.msg)})}},mounted(){this.fetchlist();let t=this;this.$parent.titleOps=[{template:'<m-button @click="onClick">拉流转发</m-button>',methods:{onClick(){t.openPull=!0}}}]},destroyed(){s.close()}},l=i;r("034f");function u(t,e,r,n,o,a,s,i){var l,u="function"===typeof t?t.options:t;if(e&&(u.render=e,u.staticRenderFns=r,u._compiled=!0),n&&(u.functional=!0),a&&(u._scopeId="data-v-"+a),s?(l=function(t){t=t||this.$vnode&&this.$vnode.ssrContext||this.parent&&this.parent.$vnode&&this.parent.$vnode.ssrContext,t||"undefined"===typeof __VUE_SSR_CONTEXT__||(t=__VUE_SSR_CONTEXT__),o&&o.call(this,t),t&&t._registeredComponents&&t._registeredComponents.add(s)},u._ssrRegister=l):o&&(l=i?function(){o.call(this,this.$root.$options.shadowRoot)}:o),l)if(u.functional){u._injectStyles=l;var c=u.render;u.render=function(t,e){return l.call(e),c(t,e)}}else{var d=u.beforeCreate;u.beforeCreate=d?[].concat(d,l):[l]}return{exports:t,options:u}}var c=u(l,o,a,!1,null,null,null),d=c.exports;e["default"]=d}})["default"]}));
-//# sourceMappingURL=plugin-rtsp.umd.min.js.map
--- a/ui/dist/plugin-rtsp.umd.min.js.map
+++ b/ui/dist/plugin-rtsp.umd.min.js.map
--- a/ui/package-lock.json
+++ b/ui/package-lock.json
--- a/ui/package.json
+++ b/ui/package.json
@@ -1,15 +0,0 @@
-{
-  "name": "dashboard",
-  "version": "1.0.0",
-  "description": "dashboard of rtsp plugin for monibuca",
-  "main": "index.js",
-  "scripts": {
-    "build": "vue-cli-service build --target lib --name plugin-rtsp"
-  },
-  "author": "dexter",
-  "license": "ISC",
-  "devDependencies": {
-    "@vue/cli-service": "^4.2.3",
-    "vue-template-compiler": "^2.6.11"
-  }
-}
--- a/ui/src/App.vue
+++ b/ui/src/App.vue
@@ -1,125 +0,0 @@
-<template>
-    <div>
-        <mu-data-table :data="Streams" :columns="columns">
-            <template #default="{row:item}">
-                <td>{{item.StreamInfo.StreamPath}}</td>
-                <td>
-                    <StartTime :value="item.StreamInfo.StartTime"></StartTime>
-                </td>
-                <td><Progress :stroke-width="20" :percent="Math.ceil(item.BufferRate)" text-inside /></td>
-                <td>{{item.SyncCount}}</td>
-                <td>
-                    <mu-button flat @click="showHeader(item)">头信息</mu-button>
-                </td>
-            </template>
-        </mu-data-table>
-        <mu-dialog title="拉流转发" width="360" :open.sync="openPull">
-            <mu-text-field v-model="remoteAddr" label="rtsp url" label-float help-text="Please enter URL of rtsp...">
-            </mu-text-field>
-            <mu-text-field v-model="streamPath" label="streamPath" label-float
-                help-text="Please enter streamPath to publish."></mu-text-field>
-            <mu-button slot="actions" flat color="primary" @click="addPull">确定</mu-button>
-        </mu-dialog>
-    </div>
-</template>
-
-<script>
-let listES = null;
-export default {
-    data() {
-        return {
-            currentStream: null,
-            Streams: null,
-            remoteAddr: "",
-            streamPath: "",
-            openPull: false,
-            columns: [
-                "StreamPath",
-                "开始时间",
-                "缓冲",
-                "同步数",
-                "操作"
-            ].map(title => ({ title }))
-        };
-    },
-
-    methods: {
-        fetchlist() {
-            listES = new EventSource(this.apiHost + "/rtsp/list");
-            listES.onmessage = evt => {
-                if (!evt.data) return;
-                this.Streams = JSON.parse(evt.data) || [];
-                this.Streams.sort((a, b) =>
-                    a.StreamInfo.StreamPath > b.StreamInfo.StreamPath ? 1 : -1
-                );
-            };
-        },
-        showHeader(item) {
-            this.$Modal.info({
-                title: "RTSP Header",
-                width: "1000px",
-                scrollable: true,
-                content: item.Header
-            });
-        },
-        addPull() {
-            this.openPull = false;
-            this.ajax
-                .getJSON(this.apiHost + "/rtsp/pull", {
-                    target: this.remoteAddr,
-                    streamPath: this.streamPath
-                })
-                .then(x => {
-                    if (x.code == 0) {
-                        this.$toast.success("已启动拉流");
-                    } else {
-                        this.$toast.error(x.msg);
-                    }
-                });
-        }
-    },
-    mounted() {
-        this.fetchlist();
-        let _this = this;
-        this.$parent.titleOps = [
-            {
-                template: '<m-button @click="onClick">拉流转发</m-button>',
-                methods: {
-                    onClick() {
-                        _this.openPull = true;
-                    }
-                }
-            }
-        ];
-    },
-    destroyed() {
-        listES.close();
-    }
-};
-</script>
-
-<style>
-.empty {
-    color: #eb5e46;
-    width: 100%;
-    min-height: 500px;
-    display: flex;
-    justify-content: center;
-    align-items: center;
-}
-
-.layout {
-    padding-bottom: 30px;
-    display: flex;
-    flex-wrap: wrap;
-}
-.ts-info {
-    width: 300px;
-}
-
-.hls-info {
-    width: 350px;
-    display: flex;
-    flex-direction: column;
-}
-</style>
Author	SHA1	Message	Date
dexter	e6d0489d9c	Merge pull request #20 from ziminghua/aler9 解决发布超时后，重连RTSP，无法播放的BUG	2022-02-22 21:17:24 +08:00
訾明华	68d0d9aa08	解决发布超时后，重连RTSP，无法播放的BUG 清空rtsp.processFunc数据，避免接收到流时，指向旧的track，导致新stream无法播放	2022-02-22 21:15:35 +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
langhuihui	fd8ebcd87c	内存复制	2021-07-09 22:50:33 +08:00
langhuihui	cc731a25f0	适配3.1	2021-07-09 08:18:19 +08:00
dexter	ba9f39853f	Merge pull request #9 from dwdcth/v3 修复sps pps vps为空	2021-06-24 17:29:20 +08:00
banshan	0c8bd62e81	修复sps pps vps为空	2021-06-24 16:35:57 +08:00
李宇翔	dfe462a7d1	修复对Codec判断	2021-06-23 17:54:53 +08:00
李宇翔	cc7b899922	优化track设定	2021-06-23 15:57:44 +08:00
langhuihui	b0c3cdb21a	适配引擎版本升级	2021-06-15 08:06:40 +08:00
李宇翔	d08230bf0c	v3去除UI	2021-03-30 18:52:30 +08:00
dexter	8a7fdedc0f	Merge pull request #8 from dwdcth/patch-1 embed	2021-03-20 12:53:07 +08:00
banshan	b7d59b0198	embed	2021-03-19 14:22:34 +08:00
dwdcth	dc65348ccb	Update go.mod	2021-03-19 14:13:53 +08:00
langhuihui	7fa6d0dcce	修改API	2021-02-26 22:03:50 +08:00
langhuihui	0689154012	适配3.0	2021-02-23 13:23:39 +08:00
langhuihui	2e39eabcba	适配3.0	2021-02-17 21:54:37 +08:00
langhuihui	a9cb4cd853	增加cors头	2020-11-16 21:23:16 +08:00
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
langhuihui	655170cb24	修复中止拉流的操作	2020-05-20 09:55:39 +08:00
李宇翔	55bd2ce785	增加中止操作	2020-05-19 19:25:44 +08:00
				`@@ -1 +0,0 @@`
				`.empty{color:#eb5e46;width:100%;min-height:500px;display:flex;justify-content:center;align-items:center}.layout{padding-bottom:30px;display:flex;flex-wrap:wrap}.ts-info{width:300px}.hls-info{width:350px;display:flex;flex-direction:column}`