java通过sip传输协议，传输音频代码

使用SIP协议传输音频需要使用SIP的RTP扩展，通常情况下可以使用第三方的SIP协议栈，例如JAIN-SIP、PJSIP等。以下是使用JAIN-SIP进行音频传输的简单示例：

1. 导入JAIN-SIP包

首先需要在Java项目中导入JAIN-SIP相关的jar包。

2. 创建SIP协议栈

使用JAIN-SIP提供的接口创建SIP协议栈，并设置监听端口。

3. 创建SIP会话

使用SIP协议栈创建SIP会话，包括本地SIP地址、远程SIP地址等信息。

4. 创建RTP会话

使用SIP会话创建RTP会话，包括本地RTP端口、远程RTP端口等信息。

5. 发送音频数据

使用RTP会话发送音频数据。

参考代码：

import javax.sip.*;
import javax.sip.address.AddressFactory;
import javax.sip.header.HeaderFactory;
import javax.sip.message.MessageFactory;
import javax.sip.message.Request;
import javax.sip.message.Response;
import java.net.InetAddress;
import java.util.ArrayList;
import java.util.Properties;

public class SipAudioClient implements SipListener {

    private SipStack sipStack;
    private SipProvider sipProvider;
    private AddressFactory addressFactory;
    private MessageFactory messageFactory;
    private HeaderFactory headerFactory;

    private String localIp = "192.168.1.100"; // 本地SIP地址
    private String remoteIp = "192.168.1.200"; // 远程SIP地址
    private int localSipPort = 5060; // 本地SIP端口
    private int remoteSipPort = 5060; // 远程SIP端口
    private int localRtpPort = 8000; // 本地RTP端口
    private int remoteRtpPort = 8000; // 远程RTP端口

    private byte[] audioData; // 音频数据

    public SipAudioClient() throws Exception {
        // 创建SIP协议栈
        SipFactory sipFactory = SipFactory.getInstance();
        sipFactory.setPathName("gov.nist");
        Properties properties = new Properties();
        properties.setProperty("javax.sip.STACK_NAME", "SipAudioClient");
        properties.setProperty("javax.sip.IP_ADDRESS", InetAddress.getLocalHost().getHostAddress());
        properties.setProperty("gov.nist.javax.sip.DEBUG_LOG", "SipAudioClient_debug.txt");
        properties.setProperty("gov.nist.javax.sip.SERVER_LOG", "SipAudioClient_log.txt");
        sipStack = sipFactory.createSipStack(properties);

        // 创建SIP监听器
        ListeningPoint listeningPoint = sipStack.createListeningPoint(localIp, localSipPort, "udp");
        sipProvider = sipStack.createSipProvider(listeningPoint);
        sipProvider.addSipListener(this);

        // 创建SIP消息工厂
        messageFactory = sipFactory.createMessageFactory();

        // 创建SIP地址工厂
        addressFactory = sipFactory.createAddressFactory();

        // 创建SIP头工厂
        headerFactory = sipFactory.createHeaderFactory();
    }

    public void sendAudio() throws Exception {
        // 创建SIP请求
        Request request = messageFactory.createRequest("INVITE sip:" + remoteIp + ":" + remoteSipPort + " SIP/2.0\r\n\r\n");
        request.addHeader(headerFactory.createViaHeader(localIp, localSipPort, "udp", null));
        request.addHeader(headerFactory.createMaxForwardsHeader(70));
        Address from = addressFactory.createAddress("sip:" + localIp + ":" + localSipPort);
        request.addHeader(headerFactory.createFromHeader(from, "123456"));
        Address to = addressFactory.createAddress("sip:" + remoteIp + ":" + remoteSipPort);
        request.addHeader(headerFactory.createToHeader(to, null));
        request.addHeader(headerFactory.createCallIdHeader("123456"));
        request.addHeader(headerFactory.createCSeqHeader((long) 1, Request.INVITE));
        request.addHeader(headerFactory.createContactHeader(from));

        // 创建RTP会话
        ArrayList<String> headers = new ArrayList<>();
        headers.add("Content-Type: application/sdp");
        headers.add("Content-Length: 140");
        headers.add("v=0");
        headers.add("o=user1 53655765 2353687637 IN IP4 " + localIp);
        headers.add("s=-");
        headers.add("c=IN IP4 " + localIp);
        headers.add("t=0 0");
        headers.add("m=audio " + localRtpPort + " RTP/AVP 0");
        headers.add("a=rtpmap:0 PCMU/8000");
        headers.add("a=sendrecv");
        String sdp = String.join("\r\n", headers);
        byte[] sdpBytes = sdp.getBytes();
        SessionDescription sessionDescription = SdpFactory.getInstance().createSessionDescription(new String(sdpBytes, "UTF-8"));
        MediaDescription mediaDescription = (MediaDescription) sessionDescription.getMediaDescriptions(false).get(0);
        int payloadType = mediaDescription.getMedia().getPayloads(false).get(0).getPayloadNumber();
        RtpSession rtpSession = new RtpSession(localRtpPort, remoteIp, remoteRtpPort, payloadType);

        // 发送音频数据
        rtpSession.sendAudio(audioData, 0, audioData.length);

        // 发送SIP请求
        ClientTransaction transaction = sipProvider.getNewClientTransaction(request);
        transaction.sendRequest();
    }

    @Override
    public void processRequest(RequestEvent requestEvent) {
        // 处理收到的SIP请求
    }

    @Override
    public void processResponse(ResponseEvent responseEvent) {
        // 处理收到的SIP响应
    }

    @Override
    public void processTimeout(TimeoutEvent timeoutEvent) {
        // 处理超时事件
    }

    @Override
    public void processIOException(IOExceptionEvent ioExceptionEvent) {
        // 处理IO异常事件
    }

    @Override
    public void processTransactionTerminated(TransactionTerminatedEvent transactionTerminatedEvent) {
        // 处理事务结束事件
    }

    @Override
    public void processDialogTerminated(DialogTerminatedEvent dialogTerminatedEvent) {
        // 处理对话结束事件
    }
}

需要注意的是，以上代码仅供参考，具体实现方式可能会因为SIP协议栈、RTP会话库等因素而略有不同，具体开发时请参考相关文档和API。