void MultiFramedRTPSource::networkReadHandler1() {
  BufferedPacket* bPacket = fPacketReadInProgress;
  if (bPacket == NULL) {
    // Normal case: Get a free BufferedPacket descriptor to hold the new network packet:
    bPacket = fReorderingBuffer->getFreePacket(this);
  }

  // Read the network packet, and perform sanity checks on the RTP header:
  Boolean readSuccess = False;
  do {
    Boolean packetReadWasIncomplete = fPacketReadInProgress != NULL;
    if (!bPacket->fillInData(fRTPInterface, packetReadWasIncomplete)) break;
    if (packetReadWasIncomplete) {
      // We need additional read(s) before we can process the incoming packet:
      fPacketReadInProgress = bPacket;
      return;
    } else {
      fPacketReadInProgress = NULL;
    }
#ifdef TEST_LOSS
    setPacketReorderingThresholdTime(0);
       // don't wait for 'lost' packets to arrive out-of-order later
    if ((our_random()%10) == 0) break; // simulate 10% packet loss
#endif

    // Check for the 12-byte RTP header:
    if (bPacket->dataSize() < 12) break;
    unsigned rtpHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
    Boolean rtpMarkerBit = (rtpHdr&0x00800000) >> 23;
    unsigned rtpTimestamp = ntohl(*(u_int32_t*)(bPacket->data()));ADVANCE(4);
    unsigned rtpSSRC = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
    // rcr it looks like the java version has 10 extra bytes after SSRC
    // Check the RTP version number (it should be 2):
    if ((rtpHdr&0xC0000000) != 0x80000000) break;

    // Skip over any CSRC identifiers in the header:
    unsigned cc = (rtpHdr>>24)&0xF;
    if (bPacket->dataSize() < cc) break;
    ADVANCE(cc*4);

    //rcr  the above does not seem to be skipping any bytes

    // Check for (& ignore) any RTP header extension
    if (rtpHdr&0x10000000) {
      if (bPacket->dataSize() < 4) break;
      unsigned extHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
      unsigned remExtSize = 4*(extHdr&0xFFFF);
      if (bPacket->dataSize() < remExtSize) break;
      ADVANCE(remExtSize);
    }
    //rcr the above does not seem to b skipping

    // Discard any padding bytes:
    if (rtpHdr&0x20000000) {
      if (bPacket->dataSize() == 0) break;
      unsigned numPaddingBytes
	= (unsigned)(bPacket->data())[bPacket->dataSize()-1];
      if (bPacket->dataSize() < numPaddingBytes) break;
      bPacket->removePadding(numPaddingBytes);
    }

    //rcr the above does not seem skipping bytes

    // Check the Payload Type.
    if ((unsigned char)((rtpHdr&0x007F0000)>>16)
	!= rtpPayloadFormat()) {
      break;
    }

    // The rest of the packet is the usable data.  Record and save it:
    fLastReceivedSSRC = rtpSSRC;
    unsigned short rtpSeqNo = (unsigned short)(rtpHdr&0xFFFF);
    Boolean usableInJitterCalculation
      = packetIsUsableInJitterCalculation((bPacket->data()),
						  bPacket->dataSize());
    struct timeval presentationTime; // computed by:
    Boolean hasBeenSyncedUsingRTCP; // computed by:
    receptionStatsDB()
      .noteIncomingPacket(rtpSSRC, rtpSeqNo, rtpTimestamp,
			  timestampFrequency(),
			  usableInJitterCalculation, presentationTime,
			  hasBeenSyncedUsingRTCP, bPacket->dataSize());
    //rcr this dataSize is the same as the java getPacketData().length
    // envir() << "M-frmRTPsrc.readHndlr pck sz " << bPacket->dataSize();

    // Fill in the rest of the packet descriptor, and store it:
    struct timeval timeNow;
    gettimeofday(&timeNow, NULL);
    bPacket->assignMiscParams(rtpSeqNo, rtpTimestamp, presentationTime,
			      hasBeenSyncedUsingRTCP, rtpMarkerBit,
			      timeNow);
    if (!fReorderingBuffer->storePacket(bPacket)) break;

    readSuccess = True;
  } while (0);
  if (!readSuccess) fReorderingBuffer->freePacket(bPacket);

  doGetNextFrame1();
  // If we didn't get proper data this time, we'll get another chance
}

cpp code to parse packet headers and find the actual audio frames in the packet

RTP packet handler

C++