C#實現RTP數據包傳輸

閑暇時折騰IP網絡視頻監控系統，需要支持視頻幀數據包在網絡內的傳輸。
未采用H.264或MPEG4等編碼壓縮方式，直接使用Bitmap圖片。
由於對幀的准確到達要求不好，所以采用UDP傳輸。如果發生網絡丟包現象則直接將幀丟棄。
為了記錄數據包的傳輸順序和幀的時間戳，所以研究了下RFC3550協議，采用RTP包封裝視頻幀。
並未全面深究，所以未使用SSRC和CSRC，因為不確切了解其用意。不過目前的實現情況已經足夠了。

  /// <summary>
  /// RTP(RFC3550)協議數據包
  /// </summary>
  /// <remarks>
  /// 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                        |
  /// | ....                                                          |
  /// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  /// </remarks>
  public class RtpPacket
  {
    /// <summary>
    /// version (V): 2 bits
    /// RTP版本標識，當前規范定義值為2.
    /// This field identifies the version of RTP. The version defined by this specification is two (2).
    /// (The value 1 is used by the first draft version of RTP and the value 0 is used by the protocol
    /// initially implemented in the \vat" audio tool.)
    /// </summary>
    public int Version { get { return 2; } }

    /// <summary>
    /// padding (P)：1 bit
    /// 如果設定padding，在報文的末端就會包含一個或者多個padding 字節，這不屬於payload。
    /// 最后一個字節的padding 有一個計數器，標識需要忽略多少個padding 字節(包括自己)。
    /// 一些加密算法可能需要固定塊長度的padding，或者是為了在更低層數據單元中攜帶一些RTP 報文。
    /// If the padding bit is set, the packet contains one or more additional padding octets at the
    /// end which are not part of the payload. The last octet of the padding contains a count of
    /// how many padding octets should be ignored, including itself. Padding may be needed by
    /// some encryption algorithms with fixed block sizes or for carrying several RTP packets in a
    /// lower-layer protocol data unit.
    /// </summary>
    public int Padding { get { return 0; } }

    /// <summary>
    /// extension (X)：1 bit 
    /// 如果設定了extension 位，定長頭字段后面會有一個頭擴展。
    /// If the extension bit is set, the fixed header must be followed by exactly one header extensio.
    /// </summary>
    public int Extension { get { return 0; } }

    /// <summary>
    /// CSRC count (CC)：4 bits 
    /// CSRC count 標識了定長頭字段中包含的CSRC identifier 的數量。
    /// The CSRC count contains the number of CSRC identifiers that follow the fixed header.
    /// </summary>
    public int CC { get { return 0; } }

    /// <summary>
    /// marker (M)：1 bit 
    /// marker 是由一個profile 定義的。用來允許標識在像報文流中界定幀界等的事件。
    /// 一個profile 可能定義了附加的標識位或者通過修改payload type 域中的位數量來指定沒有標識位.
    /// The interpretation of the marker is defined by a profile. It is intended to allow significant
    /// events such as frame boundaries to be marked in the packet stream. A profile may define
    /// additional marker bits or specify that there is no marker bit by changing the number of bits
    /// in the payload type field.
    /// </summary>
    public int Marker { get { return 0; } }

    /// <summary>
    /// payload type (PT)：7 bits
    /// 這個字段定一個RTPpayload 的格式和在應用中定義解釋。
    /// profile 可能指定一個從payload type 碼字到payload format 的默認靜態映射。
    /// 也可以通過non-RTP 方法來定義附加的payload type 碼字(見第3 章)。
    /// 在 RFC 3551[1]中定義了一系列的默認音視頻映射。
    /// 一個RTP 源有可能在會話中改變payload type，但是這個域在復用獨立的媒體時是不同的。(見5.2 節)。
    /// 接收者必須忽略它不識別的payload type。
    /// This field identifies the format of the RTP payload and determines its interpretation by the
    /// application. A profile may specify a default static mapping of payload type codes to payload
    /// formats. Additional payload type codes may be defined dynamically through non-RTP means
    /// (see Section 3). A set of default mappings for audio and video is specified in the companion
    /// RFC 3551 [1]. An RTP source may change the payload type during a session, but this field
    /// should not be used for multiplexing separate media streams (see Section 5.2).
    /// A receiver must ignore packets with payload types that it does not understand.
    /// </summary>
    public RtpPayloadType PayloadType { get; private set; }

    /// <summary>
    /// sequence number：16 bits
    /// 每發送一個RTP 數據報文序列號值加一，接收者也可用來檢測丟失的包或者重建報文序列。
    /// 初始的值是隨機的，這樣就使得known-plaintext 攻擊更加困難， 即使源並沒有加密(見9。1)，
    /// 因為要通過的translator 會做這些事情。關於選擇隨機數方面的技術見[17]。
    /// The sequence number increments by one for each RTP data packet sent, and may be used
    /// by the receiver to detect packet loss and to restore packet sequence. The initial value of the
    /// sequence number should be random (unpredictable) to make known-plaintext attacks on
    /// encryption more dificult, even if the source itself does not encrypt according to the method
    /// in Section 9.1, because the packets may flow through a translator that does. Techniques for
    /// choosing unpredictable numbers are discussed in [17].
    /// </summary>
    public int SequenceNumber { get; private set; }

    /// <summary>
    /// timestamp：32 bits
    /// timestamp 反映的是RTP 數據報文中的第一個字段的采樣時刻的時間瞬時值。
    /// 采樣時間值必須是從恆定的和線性的時間中得到以便於同步和jitter 計算(見第6.4.1 節)。
    /// 必須保證同步和測量保溫jitter 到來所需要的時間精度(一幀一個tick 一般情況下是不夠的)。
    /// 時鍾頻率是與payload 所攜帶的數據格式有關的，在profile 中靜態的定義或是在定義格式的payload format 中，
    /// 或通過non-RTP 方法所定義的payload format 中動態的定義。如果RTP 報文周期的生成，就采用虛擬的(nominal) 
    /// 采樣時鍾而不是從系統時鍾讀數。例如，在固定比特率的音頻中，timestamp 時鍾會在每個采樣周期時加一。
    /// 如果音頻應用中從輸入設備中讀入160 個采樣周期的塊，the timestamp 就會每一塊增加160，
    /// 而不管塊是否傳輸了或是丟棄了。
    /// 對於序列號來說，timestamp 初始值是隨機的。只要它們是同時(邏輯上)同時生成的，
    /// 這些連續的的 RTP 報文就會有相同的timestamp，
    /// 例如，同屬一個視頻幀。正像在MPEG 中內插視頻幀一樣，
    /// 連續的但不是按順序發送的RTP 報文可能含有相同的timestamp。
    /// The timestamp reflects the sampling instant of the first octet in the RTP data packet. The
    /// sampling instant must be derived from a clock that increments monotonically and linearly
    /// in time to allow synchronization and jitter calculations (see Section 6.4.1). The resolution
    /// of the clock must be suficient for the desired synchronization accuracy and for measuring
    /// packet arrival jitter (one tick per video frame is typically not suficient). The clock frequency
    /// is dependent on the format of data carried as payload and is specified statically in the profile
    /// or payload format specification that defines the format, or may be specified dynamically for
    /// payload formats defined through non-RTP means. If RTP packets are generated periodically,
    /// the nominal sampling instant as determined from the sampling clock is to be used, not a
    /// reading of the system clock. As an example, for fixed-rate audio the timestamp clock would
    /// likely increment by one for each sampling period. If an audio application reads blocks covering
    /// 160 sampling periods from the input device, the timestamp would be increased by 160 for
    /// each such block, regardless of whether the block is transmitted in a packet or dropped as silent.
    /// </summary>
    public long Timestamp { get; private set; }

    /// <summary>
    /// SSRC：32 bits
    /// SSRC 域識別同步源。為了防止在一個會話中有相同的同步源有相同的SSRC identifier， 
    /// 這個identifier 必須隨機選取。
    /// 生成隨機 identifier 的算法見目錄A.6 。雖然選擇相同的identifier 概率很小，
    /// 但是所有的RTP implementation 必須檢測和解決沖突。
    /// 第8 章描述了沖突的概率和解決機制和RTP 級的檢測機制，根據唯一的 SSRCidentifier 前向循環。
    /// 如果有源改變了它的源傳輸地址，
    /// 就必須為它選擇一個新的SSRCidentifier 來避免被識別為循環過的源(見第8.2 節)。
    /// The SSRC field identifies the synchronization source. This identifier should be chosen
    /// randomly, with the intent that no two synchronization sources within the same RTP session
    /// will have the same SSRC identifier. An example algorithm for generating a random identifier
    /// is presented in Appendix A.6. Although the probability of multiple sources choosing the same
    /// identifier is low, all RTP implementations must be prepared to detect and resolve collisions.
    /// Section 8 describes the probability of collision along with a mechanism for resolving collisions
    /// and detecting RTP-level forwarding loops based on the uniqueness of the SSRC identifier. If
    /// a source changes its source transport address, it must also choose a new SSRC identifier to
    /// avoid being interpreted as a looped source (see Section 8.2).
    /// </summary>
    public int SSRC { get { return 0; } }

    /// <summary>
    /// 每一個RTP包中都有前12個字節定長的頭字段
    /// The first twelve octets are present in every RTP packet
    /// </summary>
    public const int HeaderSize = 12;
    /// <summary>
    /// RTP消息頭
    /// </summary>
    private byte[] _header;
    /// <summary>
    /// RTP消息頭
    /// </summary>
    public byte[] Header { get { return _header; } }

    /// <summary>
    /// RTP有效載荷長度
    /// </summary>
    private int _payloadSize;
    /// <summary>
    /// RTP有效載荷長度
    /// </summary>
    public int PayloadSize { get { return _payloadSize; } }

    /// <summary>
    /// RTP有效載荷
    /// </summary>
    private byte[] _payload;
    /// <summary>
    /// RTP有效載荷
    /// </summary>
    public byte[] Payload { get { return _payload; } }

    /// <summary>
    /// RTP消息總長度，包括Header和Payload
    /// </summary>
    public int Length { get { return HeaderSize + PayloadSize; } }

    /// <summary>
    /// RTP(RFC3550)協議數據包
    /// </summary>
    /// <param name="playloadType">數據報文有效載荷類型</param>
    /// <param name="sequenceNumber">數據報文序列號值</param>
    /// <param name="timestamp">數據報文采樣時刻</param>
    /// <param name="data">數據</param>
    /// <param name="dataSize">數據長度</param>
    public RtpPacket(
      RtpPayloadType playloadType, 
      int sequenceNumber, 
      long timestamp, 
      byte[] data, 
      int dataSize)
    {
      // fill changing header fields
      SequenceNumber = sequenceNumber;
      Timestamp = timestamp;
      PayloadType = playloadType;

      // build the header bistream
      _header = new byte[HeaderSize];

      // fill the header array of byte with RTP header fields
      _header[0] = (byte)((Version << 6) | (Padding << 5) | (Extension << 4) | CC);
      _header[1] = (byte)((Marker << 7) | (int)PayloadType);
      _header[2] = (byte)(SequenceNumber >> 8);
      _header[3] = (byte)(SequenceNumber);
      for (int i = 0; i < 4; i++)
      {
        _header[7 - i] = (byte)(Timestamp >> (8 * i));
      }
      for (int i = 0; i < 4; i++)
      {
        _header[11 - i] = (byte)(SSRC >> (8 * i));
      }

      // fill the payload bitstream
      _payload = new byte[dataSize];
      _payloadSize = dataSize;

      // fill payload array of byte from data (given in parameter of the constructor)
      Array.Copy(data, 0, _payload, 0, dataSize);
    }

    /// <summary>
    /// RTP(RFC3550)協議數據包
    /// </summary>
    /// <param name="playloadType">數據報文有效載荷類型</param>
    /// <param name="sequenceNumber">數據報文序列號值</param>
    /// <param name="timestamp">數據報文采樣時刻</param>
    /// <param name="frame">圖片</param>
    public RtpPacket(
      RtpPayloadType playloadType, 
      int sequenceNumber, 
      long timestamp, 
      Image frame)
    {
      // fill changing header fields
      SequenceNumber = sequenceNumber;
      Timestamp = timestamp;
      PayloadType = playloadType;

      // build the header bistream
      _header = new byte[HeaderSize];

      // fill the header array of byte with RTP header fields
      _header[0] = (byte)((Version << 6) | (Padding << 5) | (Extension << 4) | CC);
      _header[1] = (byte)((Marker << 7) | (int)PayloadType);
      _header[2] = (byte)(SequenceNumber >> 8);
      _header[3] = (byte)(SequenceNumber);
      for (int i = 0; i < 4; i++)
      {
        _header[7 - i] = (byte)(Timestamp >> (8 * i));
      }
      for (int i = 0; i < 4; i++)
      {
        _header[11 - i] = (byte)(SSRC >> (8 * i));
      }

      // fill the payload bitstream
      using (MemoryStream ms = new MemoryStream())
      {
        frame.Save(ms, ImageFormat.Jpeg);
        _payload = ms.ToArray();
        _payloadSize = _payload.Length;
      }
    }

    /// <summary>
    /// RTP(RFC3550)協議數據包
    /// </summary>
    /// <param name="packet">數據包</param>
    /// <param name="packetSize">數據包長度</param>
    public RtpPacket(byte[] packet, int packetSize)
    {
      //check if total packet size is lower than the header size
      if (packetSize >= HeaderSize)
      {
        //get the header bitsream
        _header = new byte[HeaderSize];
        for (int i = 0; i < HeaderSize; i++)
        {
          _header[i] = packet[i];
        }

        //get the payload bitstream
        _payloadSize = packetSize - HeaderSize;
        _payload = new byte[_payloadSize];
        for (int i = HeaderSize; i < packetSize; i++)
        {
          _payload[i - HeaderSize] = packet[i];
        }

        //interpret the changing fields of the header
        PayloadType = (RtpPayloadType)(_header[1] & 127);
        SequenceNumber = UnsignedInt(_header[3]) + 256 * UnsignedInt(_header[2]);
        Timestamp = UnsignedInt(_header[7]) 
          + 256 * UnsignedInt(_header[6]) 
          + 65536 * UnsignedInt(_header[5]) 
          + 16777216 * UnsignedInt(_header[4]);
      }
    }

    /// <summary>
    /// 將消息轉換成byte數組
    /// </summary>
    /// <returns>消息byte數組</returns>
    public byte[] ToArray()
    {
      byte[] packet = new byte[Length];

      Array.Copy(_header, 0, packet, 0, HeaderSize);
      Array.Copy(_payload, 0, packet, HeaderSize, PayloadSize);

      return packet;
    }

    /// <summary>
    /// 將消息體轉換成圖片
    /// </summary>
    /// <returns>圖片</returns>
    public Bitmap ToBitmap()
    {
      return new Bitmap(new MemoryStream(_payload));
    }

    /// <summary>
    /// 將消息體轉換成圖片
    /// </summary>
    /// <returns>圖片</returns>
    public Image ToImage()
    {
      return Image.FromStream(new MemoryStream(_payload));
    }

    /// <summary>
    /// 將圖片轉換成消息
    /// </summary>
    /// <param name="playloadType">數據報文有效載荷類型</param>
    /// <param name="sequenceNumber">數據報文序列號值</param>
    /// <param name="timestamp">數據報文采樣時刻</param>
    /// <param name="frame">圖片幀</param>
    /// <returns>
    /// RTP消息
    /// </returns>
    public static RtpPacket FromImage(
      RtpPayloadType playloadType, 
      int sequenceNumber, 
      long timestamp, 
      Image frame)
    {
      return new RtpPacket(playloadType, sequenceNumber, timestamp, frame);
    }

    /// <summary>
    /// return the unsigned value of 8-bit integer nb
    /// </summary>
    /// <param name="nb"></param>
    /// <returns></returns>
    private static int UnsignedInt(int nb)
    {
      if (nb >= 0)
        return (nb);
      else
        return (256 + nb);
    }
  }