Video RTP transmittion problem

Hey guys (and girls),
I'm writing two programs that either send or receive a RTP video stream captured with a webcam. Using JMStudio I can capture and transmit the video and happily receive it with my client program, but when I try to capture and transmit the video using my server program the client can't realize because it's not got any data.
Here is my transmitter code, my server class creates a new instance of the camTransmitter class.
import java.io.*;
import java.awt.*;
import java.util.Vector;
import javax.swing.*;
import java.awt.event.*;
import javax.media.*;
import javax.media.format.*;
import javax.media.protocol.*;
import javax.media.control.*;
import javax.media.ControllerListener;
import javax.media.rtp.*;
import javax.media.rtp.rtcp.*;
import javax.media.rtp.event.*;
import javax.media.rtp.SendStreamListener.*;
import java.net.InetAddress;
public class camTransmitter implements ControllerListener
     Processor p = null;
     CaptureDeviceInfo di = null;
     boolean configured, realized = false;
     public String ipAddress = "192.168.123.150";
     public int port = 22222;
     public RTPManager rtpMgrs[];
     public DataSource vDS;
     public SessionAddress localAddr, destAddr;
     public InetAddress ipAddr;
     public SendStream sendStream;
    public camTransmitter()
     // First, we'll need a DataSource that captures live video
     Format vFormat = new VideoFormat(VideoFormat.RGB);
     Vector devices = CaptureDeviceManager.getDeviceList(vFormat);
     if (devices.size() > 0)
          di = (CaptureDeviceInfo) devices.elementAt(0);
     else
          // exit if we could not find the relevant capture device.
          System.out.println("no devises");          
          System.exit(-1);
     MediaLocator camLoctation = di.getLocator();
     // Create a processor for this capture device & exit if we
     // cannot create it
     try
          vDS = Manager.createDataSource(camLoctation);
          p = Manager.createProcessor(vDS);
          p.addControllerListener(this);
     catch (IOException e) {
          System.exit(-1);
     }catch (NoProcessorException e) {
          System.exit(-1);
     }catch(NoDataSourceException e) {
          System.exit(-1);
     // at this point, we have succesfully created the processor.
     // Realize it and block until it is configured.
     p.configure();
     while(configured != true){}
     // block until it has been configured
     p.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW_RTP));
     TrackControl track[] = p.getTrackControls();
     boolean encodingOk = false;
     // Go through the tracks and try to program one of them to
     // output ULAW_RTP data.
     for (int i = 0; i < track.length; i++)
          if (!encodingOk && track[i] instanceof FormatControl)
               if (((FormatControl)track).setFormat(vFormat) == null)
                    track[i].setEnabled(false);
               else
                    encodingOk = true;
          else
               // we could not set this track to gsm, so disable it
               track[i].setEnabled(false);
     // Realize it and block until it is realized.
     p.realize();
     while(realized != true){}
     // block until realized.
     // get the output datasource of the processor and exit
     // if we fail
     DataSource ds = null;
     try
          ds = p.getDataOutput();
     catch (NotRealizedError e)
          System.exit(-1);
     PushBufferDataSource pbds = (PushBufferDataSource)ds;
     PushBufferStream pbss[] = pbds.getStreams();
     rtpMgrs = new RTPManager[pbss.length];
     for (int i = 0; i < pbss.length; i++) {
     try {
          rtpMgrs[i] = RTPManager.newInstance();
          ipAddr = InetAddress.getByName(ipAddress);
          localAddr = new SessionAddress(InetAddress.getLocalHost(),port);
          destAddr = new SessionAddress(ipAddr, port);
          System.out.println(localAddr);
          rtpMgrs[i].initialize(destAddr);
          rtpMgrs[i].addTarget(destAddr);
          System.out.println( "Created RTP session: " + ipAddress + " " + port);
          sendStream = rtpMgrs[i].createSendStream(ds, i);          
          sendStream.start();
     } catch (Exception e) {
          System.err.println(e.getMessage());
     //System.out.println("RTP Stream created and running");
public void controllerUpdate(ControllerEvent e)
     if(e instanceof ConfigureCompleteEvent)
          System.out.println("Configure Complete");
          configured = true;
     if(e instanceof RealizeCompleteEvent)
          System.out.println("Ralizing complete");
          realized = true;
When I run this the processor realizes and the RTP seems to stream, but I have no idea why I can't receive the video from this.
Pls help if anyone can Thanks

Dear andreyvk ,
I've read your post
http://forum.java.sun.com/thread.jspa?threadID=785134&tstart=165
about how to use a single RTP session for both media reception and trasmission (I'm referring to you RTPSocketAdapter modified version), but, at the moment, I'receive a BIND error.
I think that your post is an EXCELLENT solution. I'modified AVReceive3 and AVTransmit3 in order to accept all parameters (Local IP & Port, Remote IP & Port).
Can you please give me a simple scenario so I can understand what the mistake?
I'use AVTransmit3 and AVReceive3 from different prompts and if I run these 2 classes contemporarely both in 2 different PC (172.17.32.27 and 172.17.32.30) I can transmit the media (vfw://0 for example) using AVTransmit3 but I'can't receive nothing if I run also AVReceive3 in the same PC?
What's the problem? Furthermore, If I run first AVReceive3 from a MSDOS Prompt and subsequently I run AVTransmit3 from another prompt I see a BIND error (port already in use).
How can I use your modified RTPSocketAdapter in order to send and receive a single media from the same port (e.g. 7500).
I've used this scenario PC1: IP 172.17.32.30 Local Port 5000 and PC2:IP 172.17.32.27 LocalPort 10000
So in the PC1 I run:
AVTransmit3 vfw://0 <Local IP 172.17.32.30> <5000> <Remote IP 172.17.32.27> <10000>
AVReceive3 <Local IP 172.17.32.30/5000> <Remote IP 172.17.32.27/10000>
and in PC2:
AVTransmit3 vfw://0 <Local IP 172.17.32.27> <10000> <Remote IP 172.17.32.30> <5000>
AVReceive3 <Local IP 172.17.32.27/10000> <Remote IP 172.17.32.30/5000>
I'd like to use the same port 5000 (in PC1) and 10000 (in PC2) in order to transmit and receive rtp packets. How can i do that without receive a Bind Error? How can I receive packets (and playing these media if audio &/or video) from the same port used to send stream over the network?
How can I obtain a RTP Symmetric Transmission/Reception solution?
Please give me an hint. If you can't post this is my email: [email protected]

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    a=rtpmap:0 PCMU/8000
    a=rtpID:3253226420
    55.976046 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:16402 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK1c92401a6e250ccf
    Max-Forwards: 70
    To: "u0" <sip:user@rip:16402>
    From: "0" <sip:user@lip:16402>;tag=1537014430
    Call-ID: 4860ab94-b4d4-11dc-92ee-be9358c74012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 399
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3253226420
    61.478080 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50431 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK1cc0b6ce6e567e1e
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50431>
    From: "0" <sip:user@lip:16402>;tag=1593623839
    Call-ID: 4c8d2436-b4d4-11dc-92ee-eb3fab0f4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 399
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3253226420
    61.978667 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50431 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK1cc0b6ce6e567e1e
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50431>
    From: "0" <sip:user@lip:16402>;tag=1593623839
    Call-ID: 4c8d2436-b4d4-11dc-92ee-eb3fab0f4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 399
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3253226420
    62.979288 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50431 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK1cc0b6ce6e567e1e
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50431>
    From: "0" <sip:user@lip:16402>;tag=1593623839
    Call-ID: 4c8d2436-b4d4-11dc-92ee-eb3fab0f4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 399
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3253226420
    221.875355 @Video Conference/VCInitiateConference.m:1582 type=2 (00000000/0)
    [Connection Data for call id: 5 returns 1
    226.522875 @Video Conference/VCInitiateConference.m:1597 type=2 (00000000/0)
    [Prepare Connection With Remote Data - remote VCConnectionData: 1, local VCConnectionData: 1
    226.527478 @Video Conference/VCInitiateConference.m:1701 type=2 (00000000/0)
    [Initiate Conference To User: u0 with Remote VCConnectionData: 1 with Local Connection Data: 1 conferenceSettings: 1]
    237.541363 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:16402 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK7c8b8e053141eae8
    Max-Forwards: 70
    To: "u0" <sip:user@rip:16402>
    From: "0" <sip:user@lip:16402>;tag=538595231
    Call-ID: b57cadae-b4d4-11dc-92ee-d6e5d2954012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    238.041987 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:16402 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK7c8b8e053141eae8
    Max-Forwards: 70
    To: "u0" <sip:user@rip:16402>
    From: "0" <sip:user@lip:16402>;tag=538595231
    Call-ID: b57cadae-b4d4-11dc-92ee-d6e5d2954012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    239.042442 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:16402 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK7c8b8e053141eae8
    Max-Forwards: 70
    To: "u0" <sip:user@rip:16402>
    From: "0" <sip:user@lip:16402>;tag=538595231
    Call-ID: b57cadae-b4d4-11dc-92ee-d6e5d2954012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    244.543484 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50468 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK47f51d1236ccf090
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50468>
    From: "0" <sip:user@lip:16402>;tag=289044260
    Call-ID: b9a91160-b4d4-11dc-92ee-a8a976dd4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    245.044025 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50468 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK47f51d1236ccf090
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50468>
    From: "0" <sip:user@lip:16402>;tag=289044260
    Call-ID: b9a91160-b4d4-11dc-92ee-a8a976dd4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    246.044704 @SIP/Transport.c:2353 type=1 (00000000/0)
    [INVITE sip:user@rip:50468 SIP/2.0
    Via: SIP/2.0/UDP lip:16402;branch=z9hG4bK47f51d1236ccf090
    Max-Forwards: 70
    To: "u0" <sip:user@rip:50468>
    From: "0" <sip:user@lip:16402>;tag=289044260
    Call-ID: b9a91160-b4d4-11dc-92ee-a8a976dd4012@lip
    CSeq: 1 INVITE
    Contact: <sip:user@lip:16402>;isfocus
    User-Agent: Viceroy 1.3
    Content-Type: application/sdp
    Content-Length: 734
    v=0
    o=howardberry 0 0 IN IP4 lip
    s=0
    c=IN IP4 lip
    b=AS:2147483647
    t=0 0
    a=hwi:17412:2:2330
    a=iChatEncryption:NO
    a=bandwidthDetection:YES
    m=audio 16402 RTP/AVP 110 121 12 3 0
    a=rtcp:16402
    a=rtpmap:121 speex/16000
    a=rtpmap:122 speex/8000
    a=rtpmap:113 X-AAC_LD/44100
    a=rtpmap:110 X-AAC_LD/22050
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:3830094153
    m=video 16402 RTP/AVP 123 126 34
    a=rtcp:16402
    a=rtpmap:123 H264/90000
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:30
    a=RTCP:AUDIO 16402 VIDEO 16402
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480:20
    a=fmtp:123 imagesize 0 rules 20:640:480:640:480:20
    a=rtpID:196956011
    Video Conference User Report:
    0.000000 @:0 type=5 (00000000/16402)
    [Local SIP port]
    0.000017 @:0 type=5 (00000000/16402)
    [Local SIP port]
    0.000032 @:0 type=5 (00000000/16402)
    [Local SIP port]
    0.000064 @:0 type=5 (00000000/16402)
    [Local SIP port]
    0.000080 @:0 type=5 (00000000/16402)
    [Local SIP port]
    252.557643 @Video Conference/VideoConferenceMultiController.m:1476 type=5 (00000000/0)
    [IP And Port Data With Caller IP And Port Data: Obtained 400 bytes of local IP and port data (10 entries). Remote data was 0 bytes (0 entries).
    []

    Hi,
    Technically it should not matter if you are using AIM or @Mac names.
    However you have solved that bit.
    The Screen Sharing part uses three ports.
    Two randomly picked ports for the Screen Sharing and the same A/V port for the Audio component. As the Airport is normally set to Share An IP and does NAT on the fly so to speak this also should not be a problem
    The ADSLNAtion site is a bit behind on it's Airport page
    http://www.adslnation.com/support/knowledgebase/ht007.php as it shows the older Firmware pics.
    I found the new page
    http://www.adslnation.com/knowledgebase/xmodem+airport-11n.php
    This says that the DNS server info in the Airport needs to be removed.
    It is not clear if this is a variation for the X-Modem M3
    Very few of the posts here http://forum.adslnation.com/viewforum.php?f=3 are after Leopard came out and nothing appears to be about iChat.
    So it would somewhat depend if the Airport is Sharing and IP or is Static to the computer and Port Mapping used.
    1:56 PM Saturday; December 29, 2007

  • H263 RTP and JPEG RTP buffering problem

    Hi
    I have a problem with buffering received video stream. I use a buffercontrol of rtpmanager and for audio stream eveything works fine but for video buffercontrol is null, sa i can't set it (the same problem is with player or porcessor). Audio stream has all parameters (bitrate, samplerate etc) but video has only encoding and size parameters.
    I tried to change format (by processor) of received stream to supported format (for example jpeg) and set all parameters (size, framerate, maxdatalength and encoding) and use output datasource but still player can't get buffercontrol. When i use a porcessor as a player i can see in media properties that it is set like i want, but buffercontrol is null.
    I think that h263_rtp and jpeg_rtp video format don't support frameate (i tried set it but nothing happens) and it's played as it comes.
    I need to set a buffer because the framerate of incoming stream is really low (it's p2p application) and smothnes of video is bad.
    Please helpme where i can find any suggestion to solve this problem. Maybe i have to use custom transport layer or something like that.

    cziz13 wrote:
    I don't know what you want to say. I said exactly what I wanted to say.
    i got buffercontrol on realized player (it cals rendering and it's in solution AudioBufferControl.java). I got the buffer even on video file, but only when i play it from my hd. when i try to do it on received video stream the buffer is not working.Good for you. But that wasn't my point.
    Whenever you request a "Player" object, you don't get a "Player" object, you get some specific "Player" subclass that, depending on what it's designed to do, will implement certain interfaces.
    The BufferControl object you're getting on your Player class that will play from the hard drive is more of a prefetch cache than a buffer (in the sense that I think of a buffer, at least). It's using a pull datasource, so it can control getting data from its datasource.
    However, the player object that's playing a RTP stream is relying upon a pull buffer data source. It has absolutely no control over the data, and it's told by the DataSource when data is ready to be played...so it just loads when it's told to do by the DataSource.
    And the datasource is just handed data from the RTP streams inside the RTPManager based on the internal RTP buffering that is handled by the RTPManager...
    Any more suggestions?My original "suggestion" still stands. Get your BufferControl object on the RTPManager that you're using to receive the stream...

  • RTP transmition problems!!!

    I write a application that send video stream to the other PC. The RTP transimition code is like the AVTransmit sample. It works well.but when I want to close or shut down my RTP transmition, there is a big problem that how can I close it absolutely without qiut the application. The close method is like this:
    public void stopTrans()
    //synchronized (this)
    if (processor != null)
    processor.stop();
    processor.close();
    processor = null;
    for (int i = 0; i < rtpMgrs.length; i++)
    rtpMgrs.removeTargets("Session ended.");
    rtpMgrs[i].dispose();
    rtpMgrs[i] = null;
    rtpMgrs = null;
    when I press the send button to start a new transmition again, there is a exception said:
    Can not create RTP session: javax.media.rtp.InvalidSessionAddressException: Can't open local data port: 3000
    port 3000 is the port number I used last time. So if I want to create transmition again using the same port, it failed. Why , how can I release the port form a RTP transmition? Please help me. Thank you very much.

    I have exactly the same problem with Rick.T, that is I cant use the same port 2nd time.... I get the message "can't open local data port xxxxx"
    For every unicast session I use 4 ports per client, 2 to transmit (video-audio) 2 to receive (video-audio). So, it doesnt make sense to use every time new ports...
    I have tried to close/stop/deallocate the Processor, to stop/disconnect DataSource, to stop/close SendStream and of course to removeTargets/dispose the RTPManager.... But I still face the same problem
    Its the very final stage of my project and I dont know what else to try....
    I have searched for other similar topics here but there is no answer to that problem.
    Is there anyone who has the solution to release the ports??
    thanx in advance

  • Rtp transmitting problem

    i am passing with a problem.......
    i read in JMF api that are three methods to transmit voice onthe network
    1-e a MediaLocator that has the parameters of the RTP session to construct an RTP DataSink by calling Manager.createDataSink.
    2-e a session manager to create send streams for the content and control the transmission.
    3-through sockets..
    i got the code for ist method of datasinking....
    but i dont know that how to get this string....
    that is url
    // hand this datasource to manager for creating an RTP
    // datasink our RTP datasimnk will multicast the audio
    try {
    String url= "rtp://224.144.251.104:49150/audio/1";
    MediaLocator m = new MediaLocator(url);
    DataSink d = Manager.createDataSink(ds, m);
    d.open();
    d.start();
    } catch (Exception e) {
    System.exit(-1);
    }

    I've faced a similar problem, what I did was to use TWO session managers, one for transmitter and the other for receiver. I then used this objects in a container.
    Both peer use the same port, and know each other's address.
    For receiving data from the peer I do:
    RTPManager mymgr = RTPManager.newInstance();
    // add as listener to create players for each incoming stream
    mymgr.addReceiveStreamListener(this);
    SessionAddress localaddr =new SessionAddress( InetAddress.getLocalHost(), port);
    SessionAddress sessaddr = new SessionAddress( peerAddress, port);
    mymgr.initialize(localaddr);
    mymgr.addTarget( sessaddr);For transmitting data to the peer I do:
    RTPManager mymgr = RTPManager.newInstance();
    // add as listener to create players for each incoming stream
    mymgr.addReceiveStreamListener(this);
    SessionAddress localaddr =new SessionAddress( InetAddress.getLocalHost(), SessionAddress.ANY_PORT);
    SessionAddress sessaddr = new SessionAddress( peerAddress, port);
    mymgr.initialize(localaddr);
    mymgr.addTarget( sessaddr);
    // the processor is capturing audio from the mic
    processor.start();
    SendStream sendStream = mymgr.createSendStream(processor.getDataOutput(), i);
    sendStream.start();I probably forgot something when doing copy paste from my code... but I think this might help you
    Good luck
    Marco

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