转自:https://www.cnblogs.com/lidabo/p/6553212.html
RTSP简介
RTSP(Real Time Streaming Protocol)是由Real Network和Netscape共同提出的如何有效地在IP网络上传输流媒体数据的应用层协议。RTSP对流媒体提供了诸如暂停,快进等控制,而它本身并不传输数据,RTSP的作用相当于流媒体服务器的远程控制。服务器端可以自行选择使用TCP或UDP来传送串流内容,它的语法和运作跟HTTP 1.1类似,但并不特别强调时间同步,所以比较能容忍网络延迟。而且允许同时多个串流需求控制(Multicast),除了可以降低服务器端的网络用量,还可以支持多方视频会议(Video onference)。 因为与HTTP1.1的运作方式相似,所以代理服务器《Proxy》的快取功能《Cache》也同样适用于RTSP,并因RTSP具有重新导向功能,可视实际负载情况来转换提供服务的服务器,以避免过大的负载集中于同一服务器而造成延迟。
rtsp和http的区别和联系
(1)联系:两者都用纯文本来发送消息,且rtsp协议的语法也和HTTP类似。Rtsp一开始这样设计,也是为了能够兼容使用以前写的HTTP协议分析代码 。
(2)区别:rtsp是有状态的,不同的是RTSP的命令需要知道现在正处于一个什么状态,也就是说rtsp的命令总是按照顺序来发送,某个命令总在另外一个命令之前要发送。Rtsp不管处于什么状态都不会去断掉连接。,而http则不保存状态,协议在发送一个命令以后,连接就会断开,且命令之间是没有依赖性的。rtsp协议使用554端口,http使用80端口。
rtsp和sip的区别和联系
SIP(Session Initiation Protocol),是基于IP的一个应用层控制协议。由于SIP是基于纯文本的信令协议,可以管理不同接入网络上的会话等。会话可以是终端设备之间任何类型的通信,如视频会话、既时信息处理或协作会话。该协议不会定义或限制可使用的业务,传输、服务质量、计费、安全性等问题都由基本核心网络和其它协议处理。
(1)联系:sip和rtsp都是应用层的控制协议,负责一次通信过程的建立和控制和结束,不负责中间的传输部分。他们都是基于纯文本的信令协议,穿墙性能良好。支持tcp、udp,支持多方通信。他们都需要服务器支持,都支持会话中重定向。sip和rtsp 都使用sdp协议来传送媒体参数,使用rtp(rtcp)协议来传输媒体流。
(2)区别:rtsp是专门为流媒体制定的协议,在多个媒体流的时间同步方面比sip强大。rtsp还提供网络负载均衡的功能,减轻服务器压力和网络带宽要求。sip一般用来创建一次音频、视频通话(双向),而rtsp一般用来做视频点播、视频监控等(单向)。当然,从原理上讲,rtsp也可以做双向的视频通话。
RTSP和RTP(rtcp)的关系
rtsp负责建立和控制会话,rtp负责多媒体的传输,rtcp配合rtp做控制和流量统计,他们是合作的关系。
RTSP的消息
RTSP的消息有两大类,一是请求消息(request),一是回应消息(response),两种消息的格式不同。
请求消息格式:
方法 URI RTSP版本 CR LF
消息头 CR LF CR LF
消息体 CR LF
方法 URI RTSP版本 CR LF
消息头 CR LF CR LF
消息体 CR LF
其中方法包括OPTIONS、SETUP、PLAY、TEARDOWN等待,URI是接收方(服务端)的地址,例如:rtsp://192.168.22.136:5000/v0,每行后面的CR LF表示回车换行,需要接收端有相应的解析,最后一个消息头需要有两个CR LF。
回应消息格式:
RTSP版本 状态码 解释 CR LF
消息头 CR LF CR LF
消息体 CR LF
其中RTSP版本一般都是RTSP/1.0,状态码是一个数值,200表示成功,解释是与状态码对应的文本解释。
状态码由三位数组成,表示方法执行的结果,定义如下:
1XX:保留,将来使用;
2XX:成功,操作被接收、理解、接受(received,understand,accepted);
3XX:重定向,要完成操作必须进行进一步操作;
4XX:客户端出错,请求有语法错误或无法实现;
5XX:服务器出错,服务器无法实现合法的请求。
RTSP的方法
rtsp中定义的方法有:OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, SCALE, GET_PARAMETER ,SET_PARAMETER
1.OPTION
目的是得到服务器提供的可用方法:
OPTIONS rtsp://192.168.20.136:5000/xxx666 RTSP/1.0
CSeq: 1 //每个消息都有序号来标记,第一个包通常是option请求消息
User-Agent: VLC media player (LIVE555 Streaming Media v2005.11.10)
服务器的回应信息包括提供的一些方法,例如:
RTSP/1.0 200 OK
Server: UServer 0.9.7_rc1
Cseq: 1 //每个回应消息的cseq数值和请求消息的cseq相对应
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, SCALE, GET_PARAMETER //服务器提供的可用的方法
2.DESCRIBE
C向S发起DESCRIBE请求,为了得到会话描述信息(SDP):
DESCRIBE rtsp://192.168.20.136:5000/xxx666 RTSP/1.0
CSeq: 2
token:
Accept: application/sdp
User-Agent: VLC media player (LIVE555 Streaming Media v2005.11.10)
服务器回应一些对此会话的描述信息(sdp):
RTSP/1.0 200 OK
Server: UServer 0.9.7_rc1
Cseq: 2
x-prev-url: rtsp://192.168.20.136:5000
x-next-url: rtsp://192.168.20.136:5000
x-Accept-Retransmit: our-retransmit
x-Accept-Dynamic-Rate: 1
Cache-Control: must-revalidate
Last-Modified: Fri, 10 Nov 2006 12:34:38 GMT
Date: Fri, 10 Nov 2006 12:34:38 GMT
Expires: Fri, 10 Nov 2006 12:34:38 GMT
Content-Base: rtsp://192.168.20.136:5000/xxx666/
Content-Length: 344
Content-Type: application/sdp
v=0 //以下都是sdp信息
o=OnewaveUServerNG 1451516402 1025358037 IN IP4 192.168.20.136
s=/xxx666
u=http:///
e=admin@
c=IN IP4 0.0.0.0
t=0 0
a=isma-compliance:1,1.0,1
a=range:npt=0-
m=video 0 RTP/AVP 96 //m表示媒体描述,下面是对会话中视频通道的媒体描述
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=245;config=000001B0F5000001B509000001000000012000C888B0E0E0FA62D089028307
a=control:trackID=0//trackID=0表示视频流用的是通道0
3.SETUP
客户端提醒服务器建立会话,并确定传输模式:
SETUP rtsp://192.168.20.136:5000/xxx666/trackID=0 RTSP/1.0
CSeq: 3
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
User-Agent: VLC media player (LIVE555 Streaming Media v2005.11.10)
//uri中带有trackID=0,表示对该通道进行设置。Transport参数设置了传输模式,包的结构。接下来的数据包头部第二个字节位置就是interleaved,它的值是每个通道都不同的,trackID=0的interleaved值有两个0或1,0表示rtp包,1表示rtcp包,接受端根据interleaved的值来区别是哪种数据包。
服务器回应信息:
RTSP/1.0 200 OK
Server: UServer 0.9.7_rc1
Cseq: 3
Session: 6310936469860791894 //服务器回应的会话标识符
Cache-Control: no-cache
Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=6B8B4567
4.PLAY
客户端发送播放请求:
PLAY rtsp://192.168.20.136:5000/xxx666 RTSP/1.0
CSeq: 4
Session: 6310936469860791894
Range: npt=0.000- //设置播放时间的范围
User-Agent: VLC media player (LIVE555 Streaming Media v2005.11.10)
服务器回应信息:
RTSP/1.0 200 OK
Server: UServer 0.9.7_rc1
Cseq: 4
Session: 6310936469860791894
Range: npt=0.000000-
RTP-Info: url=trackID=0;seq=17040;rtptime=1467265309
//seq和rtptime都是rtp包中的信息
5.TEARDOWN
客户端发起关闭请求:
TEARDOWN rtsp://192.168.20.136:5000/xxx666 RTSP/1.0
CSeq: 5
Session: 6310936469860791894
User-Agent: VLC media player (LIVE555 Streaming Media v2005.11.10)
服务器回应:
RTSP/1.0 200 OK
Server: UServer 0.9.7_rc1
Cseq: 5
Session: 6310936469860791894
Connection: Close
以上方法都是交互过程中最为常用的,其它还有一些重要的方法如get/set_parameter,pause,redirect等等
ps:
sdp的格式
v=<version>
o=<username> <session id> <version> <network type> <address type> <address>
s=<session name>
i=<session description>
u=<URI>
e=<email address>
p=<phone number>
c=<network type> <address type> <connection address>
b=<modifier>:<bandwidth-value>
t=<start time> <stop time>
r=<repeat interval> <active duration> <list of offsets from start-time>
z=<adjustment time> <offset> <adjustment time> <offset> ....
k=<method>
k=<method>:<encryption key>
a=<attribute>
a=<attribute>:<value>
m=<media> <port> <transport> <fmt list>
v = (协议版本)
o = (所有者/创建者和会话标识符)
s = (会话名称)
i = * (会话信息)
u = * (URI 描述)
e = * (Email 地址)
p = * (电话号码)
c = * (连接信息)
b = * (带宽信息)
z = * (时间区域调整)
k = * (加密密钥)
a = * (0 个或多个会话属性行)
时间描述:
t = (会话活动时间)
r = * (0或多次重复次数)
媒体描述:
m = (媒体名称和传输地址)
i = * (媒体标题)
c = * (连接信息 — 如果包含在会话层则该字段可选)
b = * (带宽信息)
k = * (加密密钥)
a = * (0 个或多个媒体属性行)
RTSP客户端的JAVA实现
3.1 接口IEvent.java
接口IEvent.java的代码如下:
- package com.amigo.rtsp;
- import java.io.IOException;
- import java.nio.channels.SelectionKey;
- /** *//**
- * IEvent.java 网络事件处理器,当Selector可以进行操作时,调用这个接口中的方法.
- * 2007-3-22 下午03:35:51
- * @author sycheng
- * @version 1.0
- */
- public interface IEvent {
- /** *//**
- * 当channel得到connect事件时调用这个方法.
- * @param key
- * @throws IOException
- */
- void connect(SelectionKey key) throws IOException;
- /** *//**
- * 当channel可读时调用这个方法.
- * @param key
- * @throws IOException
- */
- void read(SelectionKey key) throws IOException;
- /** *//**
- * 当channel可写时调用这个方法.
- * @throws IOException
- */
- void write() throws IOException;
- /** *//**
- * 当channel发生错误时调用.
- * @param e
- */
- void error(Exception e);
- }
3.2 RTSP的测试类:RTSPClient.java
RTSP的测试类RTSPClient.java类的代码如下所示:
- package com.amigo.rtsp;
- import java.io.IOException;
- import java.net.InetSocketAddress;
- import java.nio.ByteBuffer;
- import java.nio.channels.SelectionKey;
- import java.nio.channels.Selector;
- import java.nio.channels.SocketChannel;
- import java.util.Iterator;
- import java.util.concurrent.atomic.AtomicBoolean;
- public class RTSPClient extends Thread implements IEvent {
- private static final String VERSION = " RTSP/1.0/r/n";
- private static final String RTSP_OK = "RTSP/1.0 200 OK";
- /** *//** 远程地址 */
- private final InetSocketAddress remoteAddress;
- /** *//** * 本地地址 */
- private final InetSocketAddress localAddress;
- /** *//** * 连接通道 */
- private SocketChannel socketChannel;
- /** *//** 发送缓冲区 */
- private final ByteBuffer sendBuf;
- /** *//** 接收缓冲区 */
- private final ByteBuffer receiveBuf;
- private static final int BUFFER_SIZE = 8192;
- /** *//** 端口选择器 */
- private Selector selector;
- private String address;
- private Status sysStatus;
- private String sessionid;
- /** *//** 线程是否结束的标志 */
- private AtomicBoolean shutdown;
- private int seq=1;
- private boolean isSended;
- private String trackInfo;
- private enum Status {
- init, options, describe, setup, play, pause, teardown
- }
- public RTSPClient(InetSocketAddress remoteAddress,
- InetSocketAddress localAddress, String address) {
- this.remoteAddress = remoteAddress;
- this.localAddress = localAddress;
- this.address = address;
- // 初始化缓冲区
- sendBuf = ByteBuffer.allocateDirect(BUFFER_SIZE);
- receiveBuf = ByteBuffer.allocateDirect(BUFFER_SIZE);
- if (selector == null) {
- // 创建新的Selector
- try {
- selector = Selector.open();
- } catch (final IOException e) {
- e.printStackTrace();
- }
- }
- startup();
- sysStatus = Status.init;
- shutdown=new AtomicBoolean(false);
- isSended=false;
- }
- public void startup() {
- try {
- // 打开通道
- socketChannel = SocketChannel.open();
- // 绑定到本地端口
- socketChannel.socket().setSoTimeout(30000);
- socketChannel.configureBlocking(false);
- socketChannel.socket().bind(localAddress);
- if (socketChannel.connect(remoteAddress)) {
- System.out.println("开始建立连接:" + remoteAddress);
- }
- socketChannel.register(selector, SelectionKey.OP_CONNECT
- | SelectionKey.OP_READ | SelectionKey.OP_WRITE, this);
- System.out.println("端口打开成功");
- } catch (final IOException e1) {
- e1.printStackTrace();
- }
- }
- public void send(byte[] out) {
- if (out == null || out.length < 1) {
- return;
- }
- synchronized (sendBuf) {
- sendBuf.clear();
- sendBuf.put(out);
- sendBuf.flip();
- }
- // 发送出去
- try {
- write();
- isSended=true;
- } catch (final IOException e) {
- e.printStackTrace();
- }
- }
- public void write() throws IOException {
- if (isConnected()) {
- try {
- socketChannel.write(sendBuf);
- } catch (final IOException e) {
- }
- } else {
- System.out.println("通道为空或者没有连接上");
- }
- }
- public byte[] recieve() {
- if (isConnected()) {
- try {
- int len = 0;
- int readBytes = 0;
- synchronized (receiveBuf) {
- receiveBuf.clear();
- try {
- while ((len = socketChannel.read(receiveBuf)) > 0) {
- readBytes += len;
- }
- } finally {
- receiveBuf.flip();
- }
- if (readBytes > 0) {
- final byte[] tmp = new byte[readBytes];
- receiveBuf.get(tmp);
- return tmp;
- } else {
- System.out.println("接收到数据为空,重新启动连接");
- return null;
- }
- }
- } catch (final IOException e) {
- System.out.println("接收消息错误:");
- }
- } else {
- System.out.println("端口没有连接");
- }
- return null;
- }
- public boolean isConnected() {
- return socketChannel != null && socketChannel.isConnected();
- }
- private void select() {
- int n = 0;
- try {
- if (selector == null) {
- return;
- }
- n = selector.select(1000);
- } catch (final Exception e) {
- e.printStackTrace();
- }
- // 如果select返回大于0,处理事件
- if (n > 0) {
- for (final Iterator<SelectionKey> i = selector.selectedKeys()
- .iterator(); i.hasNext();) {
- // 得到下一个Key
- final SelectionKey sk = i.next();
- i.remove();
- // 检查其是否还有效
- if (!sk.isValid()) {
- continue;
- }
- // 处理事件
- final IEvent handler = (IEvent) sk.attachment();
- try {
- if (sk.isConnectable()) {
- handler.connect(sk);
- } else if (sk.isReadable()) {
- handler.read(sk);
- } else {
- // System.err.println("Ooops");
- }
- } catch (final Exception e) {
- handler.error(e);
- sk.cancel();
- }
- }
- }
- }
- public void shutdown() {
- if (isConnected()) {
- try {
- socketChannel.close();
- System.out.println("端口关闭成功");
- } catch (final IOException e) {
- System.out.println("端口关闭错误:");
- } finally {
- socketChannel = null;
- }
- } else {
- System.out.println("通道为空或者没有连接");
- }
- }
- @Override
- public void run() {
- // 启动主循环流程
- while (!shutdown.get()) {
- try {
- if (isConnected()&&(!isSended)) {
- switch (sysStatus) {
- case init:
- doOption();
- break;
- case options:
- doDescribe();
- break;
- case describe:
- doSetup();
- break;
- case setup:
- if(sessionid==null&&sessionid.length()>0){
- System.out.println("setup还没有正常返回");
- }else{
- doPlay();
- }
- break;
- case play:
- doPause();
- break;
- case pause:
- doTeardown();
- break;
- default:
- break;
- }
- }
- // do select
- select();
- try {
- Thread.sleep(1000);
- } catch (final Exception e) {
- }
- } catch (final Exception e) {
- e.printStackTrace();
- }
- }
- shutdown();
- }
- public void connect(SelectionKey key) throws IOException {
- if (isConnected()) {
- return;
- }
- // 完成SocketChannel的连接
- socketChannel.finishConnect();
- while (!socketChannel.isConnected()) {
- try {
- Thread.sleep(300);
- } catch (final InterruptedException e) {
- e.printStackTrace();
- }
- socketChannel.finishConnect();
- }
- }
- public void error(Exception e) {
- e.printStackTrace();
- }
- public void read(SelectionKey key) throws IOException {
- // 接收消息
- final byte[] msg = recieve();
- if (msg != null) {
- handle(msg);
- } else {
- key.cancel();
- }
- }
- private void handle(byte[] msg) {
- String tmp = new String(msg);
- System.out.println("返回内容:");
- System.out.println(tmp);
- if (tmp.startsWith(RTSP_OK)) {
- switch (sysStatus) {
- case init:
- sysStatus = Status.options;
- break;
- case options:
- sysStatus = Status.describe;
- trackInfo=tmp.substring(tmp.indexOf("trackID"));
- break;
- case describe:
- sessionid = tmp.substring(tmp.indexOf("Session: ") + 9, tmp
- .indexOf("Date:"));
- if(sessionid!=null&&sessionid.length()>0){
- sysStatus = Status.setup;
- }
- break;
- case setup:
- sysStatus = Status.play;
- break;
- case play:
- sysStatus = Status.pause;
- break;
- case pause:
- sysStatus = Status.teardown;
- shutdown.set(true);
- break;
- case teardown:
- sysStatus = Status.init;
- break;
- default:
- break;
- }
- isSended=false;
- } else {
- System.out.println("返回错误:" + tmp);
- }
- }
- private void doTeardown() {
- StringBuilder sb = new StringBuilder();
- sb.append("TEARDOWN ");
- sb.append(this.address);
- sb.append("/");
- sb.append(VERSION);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("User-Agent: RealMedia Player HelixDNAClient/10.0.0.11279 (win32)/r/n");
- sb.append("Session: ");
- sb.append(sessionid);
- sb.append("/r/n");
- send(sb.toString().getBytes());
- System.out.println(sb.toString());
- }
- private void doPlay() {
- StringBuilder sb = new StringBuilder();
- sb.append("PLAY ");
- sb.append(this.address);
- sb.append(VERSION);
- sb.append("Session: ");
- sb.append(sessionid);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("/r/n");
- System.out.println(sb.toString());
- send(sb.toString().getBytes());
- }
- private void doSetup() {
- StringBuilder sb = new StringBuilder();
- sb.append("SETUP ");
- sb.append(this.address);
- sb.append("/");
- sb.append(trackInfo);
- sb.append(VERSION);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("Transport: RTP/AVP;UNICAST;client_port=16264-16265;mode=play/r/n");
- sb.append("/r/n");
- System.out.println(sb.toString());
- send(sb.toString().getBytes());
- }
- private void doOption() {
- StringBuilder sb = new StringBuilder();
- sb.append("OPTIONS ");
- sb.append(this.address.substring(0, address.lastIndexOf("/")));
- sb.append(VERSION);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("/r/n");
- System.out.println(sb.toString());
- send(sb.toString().getBytes());
- }
- private void doDescribe() {
- StringBuilder sb = new StringBuilder();
- sb.append("DESCRIBE ");
- sb.append(this.address);
- sb.append(VERSION);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("/r/n");
- System.out.println(sb.toString());
- send(sb.toString().getBytes());
- }
- private void doPause() {
- StringBuilder sb = new StringBuilder();
- sb.append("PAUSE ");
- sb.append(this.address);
- sb.append("/");
- sb.append(VERSION);
- sb.append("Cseq: ");
- sb.append(seq++);
- sb.append("/r/n");
- sb.append("Session: ");
- sb.append(sessionid);
- sb.append("/r/n");
- send(sb.toString().getBytes());
- System.out.println(sb.toString());
- }
- public static void main(String[] args) {
- try {
- // RTSPClient(InetSocketAddress remoteAddress,
- // InetSocketAddress localAddress, String address)
- RTSPClient client = new RTSPClient(
- new InetSocketAddress("218.207.101.236", 554),
- new InetSocketAddress("192.168.2.28", 0),
- "rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp");
- client.start();
- } catch (Exception e) {
- e.printStackTrace();
- }
- }
- }
其中:rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp为我在网上找到的一个rtsp的sdp地址,读者可自行更换,RTSP的默认端口为554.
3.3 运行结果
运行RTSPClient.java,运行结果如下所示:
- 端口打开成功
- OPTIONS rtsp://218.207.101.236:554/mobile/3/67A451E937422331 RTSP/1.0
- Cseq: 1
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 1
- Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS, ANNOUNCE, RECORD
- DESCRIBE rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp RTSP/1.0
- Cseq: 2
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 2
- Content-length: 421
- Date: Mon, 03 Aug 2009 08:50:36 GMT
- Expires: Mon, 03 Aug 2009 08:50:36 GMT
- Content-Type: application/sdp
- x-Accept-Retransmit: our-retransmit
- x-Accept-Dynamic-Rate: 1
- Content-Base: rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/
- v=0
- o=MediaBox 127992 137813 IN IP4 0.0.0.0
- s=RTSP Session
- i=Starv Box Live Cast
- c=IN IP4 218.207.101.236
- t=0 0
- a=range:npt=now-
- a=control:*
- m=video 0 RTP/AVP 96
- b=AS:20
- a=rtpmap:96 MP4V-ES/1000
- a=fmtp:96 profile-level-id=8; config=000001b008000001b5090000010000000120008440fa282c2090a31f; decode_buf=12586
- a=range:npt=now-
- a=framerate:5
- a=framesize:96 176-144
- a=cliprect:0,0,144,176
- a=control:trackID=1
- SETUP rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/trackID=1
- RTSP/1.0
- Cseq: 3
- Transport: RTP/AVP;UNICAST;client_port=16264-16265;mode=play
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 3
- Session: 15470472221769
- Date: Mon, 03 Aug 2009 08:50:36 GMT
- Expires: Mon, 03 Aug 2009 08:50:36 GMT
- Transport: RTP/AVP;UNICAST;mode=play;client_port=16264-16265;server_port=20080-20081
- PLAY rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp RTSP/1.0
- Session: 15470472221769
- Cseq: 4
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 4
- Session: 15470472221769
- RTP-Info: url=rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/trackID=1;seq=0;rtptime=0
- PAUSE rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/ RTSP/1.0
- Cseq: 5
- Session: 15470472221769
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 5
- Session: 15470472221769
- TEARDOWN rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/ RTSP/1.0
- Cseq: 6
- User-Agent: RealMedia Player HelixDNAClient/10.0.0.11279 (win32)
- Session: 15470472221769
- 返回内容:
- RTSP/1.0 200 OK
- Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
- Cseq: 6
- Session: 15470472221769
- Connection: Close
- 端口关闭成功
对照运行结果,读者可以熟悉RTSP的常用命令.