Examples of Rtp

org.jnetpcap.protocol.voip.Rtp
The real-time transport protocol (RTP). RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers.

Note that RTP itself does not provide any mechanism to ensure timely delivery or provide other quality-of-service guarantees, but relies on lower-layer services to do so. It does not guarantee delivery or prevent out-of-order delivery, nor does it assume that the underlying network is reliable and delivers packets in sequence. The sequence numbers included in RTP allow the receiver to reconstruct the sender's packet sequence, but sequence numbers might also be used to determine the proper location of a packet, for example in video decoding, without necessarily decoding packets in sequence.

The RTP header has the following format:
```
 0                   1                   2                   3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|X|  CC   |M|     PT      |       sequence number         | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |                           timestamp                           | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |           synchronization source (SSRC) identifier            | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |            contributing source (CSRC) identifiers             | |                             ....                              | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 
```
While RTP is primarily designed to satisfy the needs of multi- participant multimedia conferences, it is not limited to that particular application. Storage of continuous data, interactive distributed simulation, active badge, and control and measurement applications may also find RTP applicable.
@author Mark Bednarczyk @author Sly Technologies, Inc.
rocks.xmpp.extensions.jingle.apps.rtp.model.Rtp
The implementation of the {@code } element in the {@code urn:xmpp:jingle:apps:rtp:1} namespace. @author Christian Schudt

Examples of org.jnetpcap.protocol.voip.Rtp

   * 
   * @throws IOException
   *           Signals that an I/O exception has occurred.
   */
  public void SKIP_testRtpAudioExtract() throws IOException {
    Rtp rtp = new Rtp();


    try {
      for (PcapPacket packet : super.getIterable(SIP_G711)) {
        assertNotNull(packet);
        if (packet.hasHeader(rtp)) {


          if (rtp.hasPostfix() || rtp.paddingLength() != 0) {
            System.out.println(rtp);
          }


          FileOutputStream out = getOutput(rtp.ssrc());


          out.write(rtp.getPayload());
        }
      }
    } catch (Exception e) {
      e.printStackTrace();
    }

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Examples of rocks.xmpp.extensions.jingle.apps.rtp.model.Rtp

                "           <parameter name='vbr' value='on'/>\n" +
                "           <parameter name='cng' value='on'/>\n" +
                "        </payload-type>\n" +
                "      </description>\n";


        Rtp rtp = unmarshal(xml, Rtp.class);
        Assert.assertEquals(rtp.getMedia(), "audio");
        Assert.assertEquals(rtp.getPayloadTypes().size(), 7);
        Assert.assertEquals(rtp.getPayloadTypes().get(0).getName(), "speex");
        Assert.assertEquals(rtp.getPayloadTypes().get(0).getClockRate(), 16000);
        Assert.assertEquals(rtp.getPayloadTypes().get(4).getChannels(), 2);
        Assert.assertEquals(rtp.getPayloadTypes().get(6).getParameters().size(), 2);
        Assert.assertEquals(rtp.getPayloadTypes().get(6).getParameters().get(0).getName(), "vbr");
        Assert.assertEquals(rtp.getPayloadTypes().get(6).getParameters().get(0).getValue(), "on");
        Assert.assertEquals(rtp.getPayloadTypes().get(6).getMaxPacketTime(), 0);
    }

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Examples of rocks.xmpp.extensions.jingle.apps.rtp.model.Rtp

                "           session-params='KDR=1 UNENCRYPTED_SRTCP'\n" +
                "           tag='1'/>\n" +
                "       </encryption>\n" +
                "      </description>\n";


        Rtp rtp = unmarshal(xml, Rtp.class);
        Assert.assertNotNull(rtp.getEncryption());
        Assert.assertEquals(rtp.getEncryption().getCrypto().size(), 1);
        Assert.assertEquals(rtp.getEncryption().getCrypto().get(0).getCryptoSuite(), "AES_CM_128_HMAC_SHA1_80");
        Assert.assertEquals(rtp.getEncryption().getCrypto().get(0).getKeyParameters(), "inline:WVNfX19zZW1jdGwgKCkgewkyMjA7fQp9CnVubGVz|2^20|1:32");
        Assert.assertEquals(rtp.getEncryption().getCrypto().get(0).getSessionParameters(), "KDR=1 UNENCRYPTED_SRTCP");
        Assert.assertEquals(rtp.getEncryption().getCrypto().get(0).getTag(), "1");
    }

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