websocket rtp

WebSocket是一种在Web浏览器和服务器之间进行双向通信的协议。它允许服务器主动向客户端推送数据，并且客户端也可以向服务器发送数据。与传统的HTTP请求-响应模式不同，WebSocket创建了一个持久的连接，通过这个连接可以实时地传输数据。

RTP（Real-time Transport Protocol）是一种用于实时传输音频和视频数据的协议。它常用于实时通信，如语音通话、视频会议和实时流媒体。RTP协议通过UDP传输数据，可以提供实时性和可靠性。

WebSocket和RTP是两个不同的协议，用于不同的目的。WebSocket用于实现双向通信，适合于实时聊天、实时更新等场景。而RTP主要用于音视频传输，适合实时流媒体和通信应用。在某些情况下，可以结合使用WebSocket和RTP来实现实时音视频通信，但它们是独立的协议。

相关问题

sip和websocket

SIP (Session Initiation Protocol) 和 WebSocket 都是用于实现实时通信的协议，但它们在使用场景和功能上有一些不同。

SIP 是一种应用层协议，用于建立、修改和终止多媒体会话，例如语音通话或视频通话。SIP 可以在 IP 网络上传输信令信息，用于会话控制、呼叫管理和媒体协商。它使用文本格式的消息，基于请求-应答模型。SIP 可以与其他协议（如 RTP）结合使用，以传输媒体数据。

WebSocket 也是一种应用层协议，用于在客户端和服务器之间进行双向通信。WebSocket 的主要目的是提供一种更高效的替代方法，以取代传统的 HTTP 轮询和长轮询技术。WebSocket 建立在 TCP 之上，允许客户端和服务器之间进行实时数据交换。它通过一个持久的连接来实现全双工通信，并且可以在任何时候发送数据。

虽然 SIP 和 WebSocket 都可以用于实现实时通信，但它们在协议设计、使用场景和功能上有所不同。SIP 更适用于多媒体会话的建立和管理，而 WebSocket 更适用于实时数据交换和即时通信。

使用springboot开启rtp端口，接收rtp流，并将PCMA/PCMU的音频数据转化为PCM格式，不是16比特的音频转换为16比特。给出调用案例和详细的注释

首先，需要添加以下依赖：

<dependency>
    <groupId>org.springframework.boot</groupId>
    <artifactId>spring-boot-starter-websocket</artifactId>
</dependency>
<dependency>
    <groupId>io.github.srt</groupId>
    <artifactId>jlibrtp</artifactId>
    <version>2.3.1</version>
</dependency>
<dependency>
    <groupId>javax.media</groupId>
    <artifactId>jmf</artifactId>
    <version>2.1.1e</version>
</dependency>

其中，`jlibrtp` 是一个 Java 实现的 RTP 协议库，`jmf` 是 Java Media Framework，用于音频数据的转换。

接下来，创建一个 WebSocket 的控制器类：

import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;
import javax.sound.sampled.DataLine.Info;
import javax.websocket.server.PathParam;
import javax.websocket.server.ServerEndpoint;
import javax.websocket.OnMessage;
import javax.websocket.Session;
import javax.websocket.server.OnOpen;
import javax.websocket.server.OnClose;
import java.util.concurrent.Executors;
import java.util.concurrent.ExecutorService;
import java.util.Arrays;
import java.io.IOException;
import java.net.SocketException;
import org.springframework.stereotype.Component;
import io.github.srt.jlibrtp.*;
import javax.media.*;
import javax.media.format.*;
import java.awt.*;
import java.net.*;
import java.util.Vector;

@Component
@ServerEndpoint(value = "/websocket/{port}")
public class WebSocketController {
    private Session session;
    private int port;
    private ReceiverThread receiverThread;
    private ExecutorService executor = Executors.newSingleThreadExecutor();

    @OnOpen
    public void onOpen(Session session, @PathParam("port") String port) {
        this.session = session;
        this.port = Integer.parseInt(port);
        executor.submit(() -> {
            try {
                startRTP();
            } catch (Exception e) {
                e.printStackTrace();
            }
        });
    }

    @OnClose
    public void onClose(Session session) {
        try {
            receiverThread.stop();
        } catch (Exception e) {
            e.printStackTrace();
        }
    }

    @OnMessage
    public void onMessage(String message, Session session) {
        // do nothing
    }

    private void startRTP() throws Exception {
        // create a new RTP session
        SessionAddress localAddress = new SessionAddress(InetAddress.getLocalHost(), port);
        RTPSession rtpSession = new RTPSession(localAddress, null);

        // set the session parameters
        rtpSession.getSessionParameters().setAcceptOwnPackets(true);
        rtpSession.getSessionParameters().setSessionBandwidth(64000.0);
        rtpSession.getSessionParameters().setProbationType(RTPSession.SIMPLE_PROBATION);

        // create a new receiver thread
        receiverThread = new ReceiverThread(rtpSession);
        receiverThread.start();
    }

    private class ReceiverThread extends Thread implements ReceiveStreamListener {
        private RTPManager rtpManager;

        public ReceiverThread(RTPSession rtpSession) throws Exception {
            // create a new RTP manager
            rtpManager = RTPManager.newInstance();

            // initialize the RTP manager
            rtpManager.addReceiveStreamListener(this);
            rtpManager.initialize(rtpSession);
            rtpManager.addTarget(new SessionAddress(InetAddress.getLocalHost(), port));
        }

        @Override
        public void run() {
            try {
                while (true) {
                    Thread.sleep(1000);
                }
            } catch (InterruptedException e) {
                e.printStackTrace();
            }
        }

        @Override
        public void update(ReceiveStreamEvent event) {
            if (event instanceof NewReceiveStreamEvent) {
                // create a new player for the stream
                try {
                    ReceiveStream stream = event.getReceiveStream();
                    DataSource dataSource = stream.getDataSource();
                    if (dataSource instanceof PushBufferDataSource) {
                        Player player = Manager.createPlayer(dataSource);
                        player.start();
                        processAudioData(player.getGainControl(), player.getFormat());
                    }
                } catch (Exception e) {
                    e.printStackTrace();
                }
            }
        }

        private void processAudioData(GainControl gainControl, Format format) throws Exception {
            // check if the format is supported
            if (!(format instanceof AudioFormat)) {
                throw new UnsupportedFormatException(format);
            }
            AudioFormat audioFormat = (AudioFormat) format;

            // create a new audio data line
            Info info = new Info(SourceDataLine.class, audioFormat);
            SourceDataLine line = (SourceDataLine) AudioSystem.getLine(info);
            line.open(audioFormat);
            line.start();

            // create a new target data line
            info = new Info(TargetDataLine.class, audioFormat);
            TargetDataLine targetLine = (TargetDataLine) AudioSystem.getLine(info);
            targetLine.open(audioFormat);
            targetLine.start();

            byte[] buffer = new byte[1024];
            while (true) {
                // read audio data from the target data line
                int bytesRead = targetLine.read(buffer, 0, buffer.length);

                // encode the audio data as PCMA/PCMU
                byte[] encodedBuffer = encodeAudioData(buffer, audioFormat);

                // write the encoded audio data to the RTP stream
                rtpManager.sendData(encodedBuffer, 0, encodedBuffer.length);

                // decode the encoded audio data back to PCM
                byte[] decodedBuffer = decodeAudioData(encodedBuffer, audioFormat);

                // write the decoded audio data to the audio data line
                line.write(decodedBuffer, 0, decodedBuffer.length);

                // adjust the gain if necessary
                if (gainControl != null) {
                    float gain = (float) Math.pow(10.0, gainControl.getDB() / 20.0);
                    for (int i = 0; i < decodedBuffer.length; i += 2) {
                        short sample = (short) ((decodedBuffer[i] & 0xff) | (decodedBuffer[i + 1] << 8));
                        sample = (short) (sample * gain);
                        decodedBuffer[i] = (byte) (sample & 0xff);
                        decodedBuffer[i + 1] = (byte) ((sample >> 8) & 0xff);
                    }
                }
            }
        }

        private byte[] encodeAudioData(byte[] buffer, AudioFormat audioFormat) {
            if (audioFormat.getSampleSizeInBits() == 16) {
                return buffer;
            }

            byte[] encodedBuffer = new byte[buffer.length / 2];
            for (int i = 0, j = 0; i < buffer.length; i += 2, j++) {
                int sample = (buffer[i] & 0xff) | (buffer[i + 1] << 8);
                if (audioFormat.getEncoding().equals(AudioFormat.ULAW)) {
                    sample = ULawEncoder.encode(sample);
                } else if (audioFormat.getEncoding().equals(AudioFormat.ALAW)) {
                    sample = ALawEncoder.encode(sample);
                } else {
                    throw new UnsupportedFormatException(audioFormat);
                }
                encodedBuffer[j] = (byte) sample;
            }
            return encodedBuffer;
        }

        private byte[] decodeAudioData(byte[] buffer, AudioFormat audioFormat) {
            if (audioFormat.getSampleSizeInBits() == 16) {
                return buffer;
            }

            byte[] decodedBuffer = new byte[buffer.length * 2];
            for (int i = 0, j = 0; i < buffer.length; i++, j += 2) {
                int sample = buffer[i] & 0xff;
                if (audioFormat.getEncoding().equals(AudioFormat.ULAW)) {
                    sample = ULawDecoder.decode(sample);
                } else if (audioFormat.getEncoding().equals(AudioFormat.ALAW)) {
                    sample = ALawDecoder.decode(sample);
                } else {
                    throw new UnsupportedFormatException(audioFormat);
                }
                decodedBuffer[j] = (byte) (sample & 0xff);
                decodedBuffer[j + 1] = (byte) ((sample >> 8) & 0xff);
            }
            return decodedBuffer;
        }
    }
}

在上述代码中，我们使用 `jlibrtp` 创建了一个 RTP 会话，并创建了一个 RTP 接收线程。当有新的 RTP 流到达时，会创建一个新的音频播放器，并将音频数据从 RTP 流中读取出来，编码为 PCMA/PCMU，然后发送回 RTP 流中。

接下来，我们需要编写一个测试用例来启动 WebSocket 服务器：

import org.springframework.boot.SpringApplication;
import org.springframework.boot.autoconfigure.SpringBootApplication;
import org.springframework.context.annotation.ComponentScan;

@SpringBootApplication
@ComponentScan("com.example.websocket")
public class Application {
    public static void main(String[] args) {
        SpringApplication.run(Application.class, args);
    }
}

最后，在客户端中，我们可以使用以下 JavaScript 代码来连接 WebSocket 服务器并接收音频流：

var audioContext = new (window.AudioContext || window.webkitAudioContext)();
var inputBuffer = new Float32Array(1024);
var outputBuffer = new Float32Array(1024);
var inputBufferOffset = 0;
var outputBufferOffset = 0;
var websocket = new WebSocket("ws://localhost:8080/websocket/1234");
websocket.binaryType = 'arraybuffer';
websocket.onmessage = function(event) {
    var audioData = new Int16Array(event.data);
    for (var i = 0; i < audioData.length; i++) {
        inputBuffer[inputBufferOffset++] = audioData[i] / 32768.0;
        if (inputBufferOffset == inputBuffer.length) {
            processAudio();
            inputBufferOffset = 0;
        }
    }
};

function processAudio() {
    // process the audio data
    // ...

    // write the output data to the output buffer
    for (var i = 0; i < outputBuffer.length; i++) {
        audioData[outputBufferOffset++] = outputBuffer[i] * 32768.0;
    }
}

function playAudio() {
    var source = audioContext.createBufferSource();
    var buffer = audioContext.createBuffer(1, audioData.length, audioContext.sampleRate);
    buffer.getChannelData(0).set(audioData);
    source.buffer = buffer;
    source.connect(audioContext.destination);
    source.start();
}

在上述代码中，我们使用 `WebSocket` 对象与 WebSocket 服务器建立连接，并将音频数据从 WebSocket 服务器接收到后，解码为 PCM 格式并存储到 `audioData` 数组中。然后，我们使用 `AudioContext` 对象创建一个音频源，并将 `audioData` 数组中的音频数据写入到音频源中，最后播放音频。