RTSP实例解析

转载自：    http://www.cnblogs.com/qq78292959/archive/2010/08/12/2077039.html

 

 

 

核心提示：rtsp简介(ZT) Real Time Streaming Protocol或者RTSP（实时流媒体协议），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提供一 种可扩展的框架，使能够提供能控制的，按需传输实时数据，比如音频和视频文件
rtsp简介(ZT)
Real Time Streaming Protocol或者RTSP（实时流媒体协议），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提供一
种可扩展的框架，使能够提供能控制的，按需传输实时数据，比如音频和视频文件。源
数据可以包括现场数据的反馈和存贮的文件。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作用相当于流媒体服务器的远程控制。传输数据可以
通过传输层的tcp，udp协议，rtsp也提供了基于rtp传输机制的一些有效的方法。
一. 参考资料
1. 《RTSP简单命令》：http://blog.csdn.net/feidragon319/archive/2007/08/14/1742357.aspx
2. http://bbs.21eic.com/dispbbs.asp?boardid=15&Id=22948
3. 《RTSP客户端的Java实现》：http://hi.baidu.com/ssyuan/blog/item/566df6defac1dc5094ee37eb.html
二. RTSP的常用命令与解释
其中C是客户端，S是服务端。
2.1 OPTIONS
C->S: OPTION request //询问S有哪些方法可用
S->C: OPTION response //S回应信息中包括提供的所有可用方法
使用举例：
客户端到服务端：
OPTIONS rtsp://218.207.101.236:554/mobile/3/67A451E937422331 RTSP/1.0
Cseq: 1
服务端对OPTIONS的回应：
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
2.2 DESCRIBE
C->S: DESCRIBE request //要求得到S提供的媒体初始化描述信息
S->C: DESCRIBE response //S回应媒体初始化描述信息，主要是sdp
使用举例：
客户端到服务端：2.4 PLAY
C->S: PLAY request //C请求播放
S->C: PLAY response //S回应该请求的信息
客户端到服务端的请求举例:
DESCRIBE rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp RTSP/1.0
Cseq: 2
服务端对OPTIONS的回应：
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 08:21:33 GMT
Expires: Mon, 03 Aug 2009 08:21:33 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
2.3 SETUP
C->S: SETUP request //设置会话的属性，以及传输模式，提醒S建立会话
S->C: SETUP response //S建立会话，返回会话标识符，以及会话相关信息
客户端到服务端的请求举例：
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
服务端对客户端的回应举例：
RTSP/1.0 200 OK
Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
Cseq: 3
Session: 26633092229589
Date: Mon, 03 Aug 2009 08:21:33 GMT
Expires: Mon, 03 Aug 2009 08:21:33 GMT
Transport: RTP/AVP;UNICAST;mode=play;client_port=16264-16265;server_port=20026-20027

PLAY rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp RTSP/1.0
Session: 26633092229589
Cseq: 4
服务端对客户端的回应举例：
RTSP/1.0 200 OK
Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
Cseq: 4
Session: 26633092229589
RTP-Info: url=rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/trackID=1;seq=0;rtptime=0
2.5 PAUSE
C->S: PAUSE request //C请求暂停播放
S->C: PAUSE response //S回应该请求的信息
客户端到服务端的请求举例:
PAUSE rtsp://218.207.101.236:554/mobile/3/67A451E937422331/8jH5QPU5GWS07Ugn.sdp/ RTSP/1.0
Cseq: 5
Session: 26633092229589
服务端对客户端的回应举例：
RTSP/1.0 200 OK
Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
Cseq: 5
Session: 26633092229589
2.6 TEARDOWN
C->S: TEARDOWN request //C请求关闭会话
S->C: TEARDOWN response //S回应该请求
客户端到服务端的请求举例:
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: 26633092229589
服务端对客户端的回应举例：
RTSP/1.0 200 OK
Server: PVSS/1.4.8 (Build/20090111; Platform/Win32; Release/StarValley; )
Cseq: 6
Session: 26633092229589
Connection: Close
三. 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