RTPを使用して音声ストリームを送受信する方法

Question

RTPを使用したSIPコールで新しくなりました。今はSIPコールにRTPを使用して 音声ストリームを送受信しようとしています。私は 2つのエミュレータに接続し、jain sipを使ってINVITEとINVITE-ACKを送信することができます。

私は私はこのようなすべてのRTPヘッダにメディアを送信するためにRtpFunctionを使用

を送受信するRtpPacket 機能を使用して、メディアストリーミング用RTPを開始したいのAckを得た後：

byte Version; 
boolean Padding; 
boolean Extension; 
byte CC; 
boolean Marker; 
byte PayloadType; 
short SequenceNumber; 
int TimeStamp; 

私は答えを見つけることができるいくつかのアイデアと実際のリンクを与えてください。

Answer 1

*We send and receive RTP data using RTPpacket. 

    import javax.media.rtp.*; 
    import javax.media.rtp.rtcp.*; 
    import javax.media.rtp.event.*; 
    import javax.media.*; 
    import javax.media.protocol.*; 
    import java.net.InetAddress; 
    import javax.media.format.AudioFormat; 
    import com.sun.media.ui.*; 
    import java.util.Vector; 
    public class RTPSourceStream<RTPPlayerWindow> implements ReceiveStreamListener, 
    ControllerListener 
    { 
@SuppressWarnings("rawtypes") 
Vector playerlist = new Vector(); 
@SuppressWarnings("deprecation") 
SessionManager mgr = null; 
boolean terminatedbyClose = false; 

@SuppressWarnings("deprecation") 
public SessionManager createManager(String address, 
     String sport, 
     String sttl, 
     boolean listener, 
     boolean sendlistener) 
{ 
    return createManager(address, 
      new Integer(sport).intValue(), 
      new Integer(sttl).intValue(), 
      listener, 
      sendlistener); 
} 
@SuppressWarnings("deprecation") 
public SessionManager createManager(String address, 
     int port, 
     int ttl, 
     boolean listener, 
     boolean sendlistener) 
{ 
    mgr = (SessionManager)new com.sun.media.rtp.RTPSessionMgr(); 
    if (mgr == null) return null; 
    mgr.addFormat(new AudioFormat(AudioFormat.DVI_RTP,44100, 4,1),18); 
    if (listener) mgr.addReceiveStreamListener(this); 
    // if (sendlistener) new RTPSendStreamWindow(mgr); 
    // ask RTPSM to generate the local participants CNAME 
    String cname = mgr.generateCNAME(); 
    String username = null; 
    try { 
     username = System.getProperty("user.name"); 
    } catch (SecurityException e){ 
     username = "jmf-user"; 
    } 
     // create our local Session Address 
    SessionAddress localaddr = new SessionAddress(); 
    try{ 
     InetAddress destaddr = InetAddress.getByName(address); 
     SessionAddress sessaddr = new SessionAddress(destaddr, 
       port, 
       destaddr, 
       port + 1); 
     SourceDescription[] userdesclist= new SourceDescription[] 
                   { 
       new SourceDescription(SourceDescription 
         .SOURCE_DESC_EMAIL, 
         "jmf-user@sun.com", 
         1, 
         false), 
         new SourceDescription(SourceDescription 
           .SOURCE_DESC_CNAME, 
           cname, 
           1, 
           false), 
           new 
      SourceDescription(SourceDescription.SOURCE_DESC_TOOL,"JMF RTP Player v2.0", 
             1, 
             false) }; 
     mgr.initSession(localaddr, 
       userdesclist, 
       0.05, 
       0.25); 
     mgr.startSession(sessaddr,ttl,null); 
    } catch (Exception e) { 
     System.err.println(e.getMessage()); 
     return null; 
    } 
    return mgr; 
    } 
public void update(ReceiveStreamEvent event) 
    { 
    Player newplayer = null; 
    RTPPacket playerWindow = null; 
     // find the sourceRTPSM for this event 
    SessionManager source = (SessionManager)event.getSource(); 
    // create a new player if a new recvstream is detected 
    if (event instanceof NewReceiveStreamEvent) 
    { 
     String cname = "Java Media Player"; 
     ReceiveStream stream = null; 
     try 
     { 
      // get a handle over the ReceiveStream 
      stream =((NewReceiveStreamEvent)event) 
      .getReceiveStream(); 
      Participant part = stream.getParticipant(); 
      if (part != null) cname = part.getCNAME(); 
      // get a handle over the ReceiveStream datasource 
      DataSource dsource = stream.getDataSource(); 
       // create a player by passing datasource to the 
       // Media Manager 
      newplayer = Manager.createPlayer(dsource); 
      System.out.println("created player " + newplayer); 
      } catch (Exception e) { 
      System.err.println("NewReceiveStreamEvent exception " 
        + e.getMessage()); 
      return; 
     } 
     if (newplayer == null) return; 
     playerlist.addElement(newplayer); 
     newplayer.addControllerListener(this); 
     // send this player to player GUI 
     playerWindow = new RTPPacket(newplayer, cname); 
    } 
    } 
    public void controllerUpdate(ControllerEvent evt) 
    { 
    // get a handle over controller, remove it from the player 
    // list. 
    // if player list is empty, close the sesssion manager. 
    if ((evt instanceof ControllerClosedEvent) || 
      (evt instanceof ControllerErrorEvent) || 
      (evt instanceof DeallocateEvent)){ 
     Player p = (Player)evt.getSourceController(); 
     if (!terminatedbyClose){ 
      if (playerlist.contains(p)) 
       playerlist.removeElement(p); 
      if ((playerlist.size() == 0) && (mgr != null)) 
       mgr.closeSession("All players are closed"); 
     } 
    } 
    } 
     public void closeManager() 
     { 
    terminatedbyClose = true; 
     // first close all the players 
    for (int i = 0; i < playerlist.size(); i++) { 
     ((Player)playerlist.elementAt(i)).close(); 
    } 
    if (mgr != null) { 
     mgr.closeSession("RTP Session Terminated"); 
     mgr = null; 
    } 
} 
class RTPPacket extends RTPSourceStream 
{ 
    public RTPPacket(Player newplayer, String cname) { 
     // TODO Auto-generated constructor stub 
    } 
    } 
    }* 

Answer 2

これは、シグナリングプロトコルとしてJAIN SIPを使用して、簡単なSIP電話機プロジェクトはここhttps://github.com/Mobicents/restcomm-android-sdk/tree/master/Examples/JAIN%20SIP