我正在使用Java应用程序使用JAIN SIP Java API与nuance语音服务器建立SIP会话。然后,我通过发送一些MRCP命令(例如GET_PARAMS,SET-PARAMS,Define Grammar和使用mrcp4j API进行识别)来设置系统,以进行识别。
然后,我使用JMF api与语音服务器建立了rtp和rtcp会话,以发送音频进行识别。服务器已接收到音频,但是直到接收到RTCP再见,它才能识别。
但是问题是我无法使用rtcp bye结束rtp会话,因为我无法在JMF文档中找到解决该问题的方法。
如果有人可以指导我,那将真的很有帮助。我已经附上了RTP会话的代码。
JMF api文档的链接为here
// send Audio data
// create the RTP Manager
RTPManager rtpManager = RTPManager.newInstance();
// create the local endpoint for the local interface on any local port
int port = Integer.parseInt(rtpPORT);;
SessionAddress localAddress = new SessionAddress();
InetAddress IP = InetAddress.getByName("hydhtc284704d");
localAddress.setControlHostAddress(IP);
localAddress.setControlPort(24501);
localAddress.setDataHostAddress(IP);
localAddress.setDataPort(24500);
// initialize the RTPManager
rtpManager.initialize(localAddress);
//rtpManager.initialize(rtpConnector);
// specify the remote endpoint of this unicast session
InetAddress ipAddress = InetAddress.getByName("hydhtc227033d");
SessionAddress remoteAddress = new SessionAddress(ipAddress, port, ipAddress, port + 1);
//System.out.println(remoteAddress);
// open the connection
rtpManager.addTarget(remoteAddress);
rtpManager.addSendStreamListener(new SendStreamListener() {
@Override
public void update(SendStreamEvent arg0) {
//System.out.println("Send Stream Event: " + arg0.getSource());
System.out.println("Number of bytes transmitted: " + arg0.getSendStream().getSourceTransmissionStats().getBytesTransmitted());
System.out.println("Sender Report: " + arg0.getSendStream().getSenderReport());
}
});
rtpManager.addReceiveStreamListener(new ReceiveStreamListener() {
@Override
public void update(ReceiveStreamEvent arg0) {
// TODO Auto-generated method stub
}
});
File audioFile = new File("C:\\Users\\Bhanu_Verma\\Desktop\\eclipse\\one.wav");
Processor processor= Manager.createProcessor(audioFile.toURI().toURL());
processor.configure();
// Block until the Processor has been configured
while (processor.getState() != processor.Configured) {
}
processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW_RTP));
TrackControl track[] = processor.getTrackControls();
//ContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW_RTP);
//processor.setContentDescriptor(cd);
boolean encodingOk = false;
// Go through the tracks and try to program one of them to
// output ulaw data.
for (int i = 0; i < track.length; i++) {
if (!encodingOk && track[i] instanceof FormatControl) {
if (((FormatControl)track[i]).setFormat(new AudioFormat(AudioFormat.ULAW_RTP,8000,8,1)) == null)
{
track[i].setEnabled(false);
}
else
{
encodingOk = true;
}
}
else
{
// we could not set this track to ulaw, so disable it
track[i].setEnabled(false);
}
}
// At this point, we have determined where we can send out ulaw data or not.
// realize the processor
if (encodingOk) {
processor.realize();
// block until realized.
while (processor.getState() != processor.Realized) {
}
// get the output datasource of the processor and exit if we fail
DataSource dataOutput = processor.getDataOutput();
// create a send stream for the output data source of a processor and start it
SendStream sendStream = rtpManager.createSendStream(dataOutput,0);
sendStream.start();
System.out.println("Starting processor" + "\n");
processor.start();
while(processor.getState()== processor.Started)
{
System.out.println("Sending Audio..");
}
System.out.println("Processor was started and audio was sent to server");
Wait(2000); // waiting so that audio could be given to the server
// close the connection if no longer needed.
rtpManager.removeTarget(remoteAddress, "Client disconnected.");
// call dispose at the end of the life-cycle of this RTPManager so
// it is prepared to be garbage-collected.
rtpManager.dispose();
最佳答案
嗯,没有使用JMF来发送rtcp bye的明确方法。但是,当您关闭SendStream时,JMF会在内部发送RTCP再见。
请注意,关闭和停止SendStream是不同的。关闭流将删除会话,而停止SendStream仅会停止数据传输。要发送RTCP再见,只需在完成发送媒体后停止处理器并关闭SendStream。
因此,要发送RTCP再见,只需将这两行添加到您的代码中。
processor.stop(); //processor needs to be stopped as well before closing the sendStream
sendStream.close();