I am new to SIP and RTP both. I have successfully managed to create a SIP call but I still dont have voice for the session.
I understand that I have to create a RTP stream and send packets. But I am unable to decide where to start from. I found JMf libraries (jar) but I am unable to understand how to use them. Also I want to play the Audio to the person I call during the transaction.
Do I have to start the RTP session inside the SIP INVITE or Do i have to create the RTP Session after the Call is answered Separately. I am not able to find answers to my Question.
Also I would like to know how do I create a RTP Session and I am doing simple Java Programming, I found a tutorial with JMf but with installation. I want to knoe if its possible with Simple Java Programming. I have the jmf-2.1.1e.jar file. I would like to know how to use it.
public SoundSenderDemo(boolean isLocal, int RTPsocket) {
DatagramSocket rtpSocket = null;
DatagramSocket rtcpSocket = null;
int socket = RTPsocket;
try {
rtpSocket = new DatagramSocket(socket);
rtcpSocket = new DatagramSocket(socket+1);
} catch (Exception e) {
System.out.println("RTPSession failed to obtain port");
}
rtpSession = new RTPSession(rtpSocket, rtcpSocket);
rtpSession.RTPSessionRegister(this,null, null);
Participant p = new Participant("sip:username@password",socket,(socket + 1));
// rtpSession.addParticipant(p);
System.out.println("CNAME: " + rtpSession.CNAME());
System.out.println("RTPSession: " + rtpSession.toString());
System.out.println("Participant: " + rtpSession.getParticipants());
System.out.println("unicast Receivers: " + rtpSession.getUnicastReceivers());
this.local = isLocal;
}
public void run() {
if(RTPSession.rtpDebugLevel > 1) {
System.out.println("-> Run()");
}
File soundFile = new File(filename);
if (!soundFile.exists()) {
System.err.println("Wave file not found: " + filename);
return;
}
AudioInputStream audioInputStream = null;
try {
audioInputStream = AudioSystem.getAudioInputStream(soundFile);
} catch (UnsupportedAudioFileException e1) {
e1.printStackTrace();
return;
} catch (IOException e1) {
e1.printStackTrace();
return;
}
//AudioFormat format = audioInputStream.getFormat();
AudioFormat.Encoding encoding = new AudioFormat.Encoding("PCM_SIGNED");
AudioFormat format = new AudioFormat(encoding,((float) 8000.0), 16, 1, 2, ((float) 8000.0) ,false);
System.out.println(format.toString());
if(! this.local) {
// To time the output correctly, we also play at the input:
auline = null;
DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
try {
auline = (SourceDataLine) AudioSystem.getLine(info);
auline.open(format);
} catch (LineUnavailableException e) {
e.printStackTrace();
return;
} catch (Exception e) {
e.printStackTrace();
return;
}
if (auline.isControlSupported(FloatControl.Type.PAN)) {
FloatControl pan = (FloatControl) auline
.getControl(FloatControl.Type.PAN);
if (this.curPosition == Position.RIGHT)
pan.setValue(1.0f);
else if (this.curPosition == Position.LEFT)
pan.setValue(-1.0f);
}
auline.start();
}
int nBytesRead = 0;
byte[] abData = new byte[EXTERNAL_BUFFER_SIZE];
long start = System.currentTimeMillis();
try {
while (nBytesRead != -1 && pktCount < 200) {
nBytesRead = audioInputStream.read(abData, 0, abData.length);
if (nBytesRead >= 0) {
rtpSession.sendData(abData);
//if(!this.local) {
auline.write(abData, 0, abData.length);
//dataCount += abData.length;
//if(pktCount % 10 == 0) {
// System.out.println("pktCount:" + pktCount + " dataCount:" + dataCount);
//
// long test = 0;
// for(int i=0; i<abData.length; i++) {
// test += abData[i];
// }
// System.out.println(Long.toString(test));
//}
pktCount++;
//if(pktCount == 100) {
// System.out.println("Time!!!!!!!!! " + Long.toString(System.currentTimeMillis()));
//}
//System.out.println("yep");
}
if(pktCount == 100) {
Enumeration<Participant> iter = this.rtpSession.getParticipants();
//System.out.println("iter " + iter.hasMoreElements());
Participant p = null;
while(iter.hasMoreElements()) {
p = iter.nextElement();
String name = "name";
byte[] nameBytes = name.getBytes();
String data= "abcd";
byte[] dataBytes = data.getBytes();
int ret = rtpSession.sendRTCPAppPacket(p.getSSRC(), 0, nameBytes, dataBytes);
System.out.println("!!!!!!!!!!!! ADDED APPLICATION SPECIFIC " + ret);
continue;
}
if(p == null)
System.out.println("No participant with SSRC available :(");
}
}
} catch (IOException e) {
e.printStackTrace();
return;
}
System.out.println("Time: " + (System.currentTimeMillis() - start)/1000 + " s");
try { Thread.sleep(200);} catch(Exception e) {}
this.rtpSession.endSession();
try { Thread.sleep(2000);} catch(Exception e) {}
if(RTPSession.rtpDebugLevel > 1) {
System.out.println("<- Run()");
}
}
While sending ACK
dialog.sendAck(ackRequest);
// System.out.println(ackRequest.toString());
logger.debug(ackRequest.toString());
aDemo = new SoundSenderDemo(false, RTPsocket);
RTPstart();
public void RTPstart(){
// Start RTP Session
String file = "C:/universAAL/workspaces/SIPfinaltest withRTP/SIPfinaltest/JSIP/garfield_converted.wav";
// SoundSenderDemo aDemo = new SoundSenderDemo(false);
aDemo.filename = args[0];
aDemo.run();
System.out.println("pktCount: " + aDemo.pktCount);
}
Also in the invite I have set :
String sdpData = "v=0\n" +
"o=user1 795808818 480847547 IN IP4 "+localIP+"\n" +
"s=-\n" +
"c=IN IP4 "+localIP+"\n" +
"t=0 0\n" +
"m=audio 8000 RTP/AVP 0 8 101\n" +
"a=rtpmap:0 PCMU/8000\n" +
"a=rtpmap:8 PCMA/8000\n" +
"a=rtpmap:101 telephone-event/8000\n" +
"a=sendrecv";
byte[] contents = sdpData.getBytes();
This is the response:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.134.149:5060;branch=z9hG4bK-333831-44ef6fc075d847c6420a0f95b2022345;received=10.99.134.149;rport=5060
From: <sip:[email protected]>;tag=-1209613008
To: <sip:[email protected]>;tag=as12f64e9a
Call-ID: [email protected]
CSeq: 5 INVITE
Server: Asterisk PBX 10.5.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 532626251 532626252 IN IP4 10.99.64.2
s=Asterisk PBX 10.5.1
c=IN IP4 10.99.64.2
t=0 0
m=audio 7758 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
