C#实现RTP数据包传输

闲暇时折腾IP网络视频监控系统，需要支持视频帧数据包在网络内的传输。
未采用H.264或MPEG4等编码压缩方式，直接使用Bitmap图片。
由于对帧的准确到达要求不好，所以采用UDP传输。如果发生网络丢包现象则直接将帧丢弃。
为了记录数据包的传输顺序和帧的时间戳，所以研究了下RFC3550协议，采用RTP包封装视频帧。
并未全面深究，所以未使用SSRC和CSRC，因为不确切了解其用意。不过目前的实现情况已经足够了。

  /// <summary>
  /// RTP(RFC3550)协议数据包
  /// </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版本标识，当前规范定义值为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
    /// 如果设定padding，在报文的末端就会包含一个或者多个padding 字节，这不属于payload。
    /// 最后一个字节的padding 有一个计数器，标识需要忽略多少个padding 字节(包括自己)。
    /// 一些加密算法可能需要固定块长度的padding，或者是为了在更低层数据单元中携带一些RTP 报文。
    /// 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 
    /// 如果设定了extension 位，定长头字段后面会有一个头扩展。
    /// 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 标识了定长头字段中包含的CSRC identifier 的数量。
    /// 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 是由一个profile 定义的。用来允许标识在像报文流中界定帧界等的事件。
    /// 一个profile 可能定义了附加的标识位或者通过修改payload type 域中的位数量来指定没有标识位.
    /// 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
    /// 这个字段定一个RTPpayload 的格式和在应用中定义解释。
    /// profile 可能指定一个从payload type 码字到payload format 的默认静态映射。
    /// 也可以通过non-RTP 方法来定义附加的payload type 码字(见第3 章)。
    /// 在 RFC 3551[1]中定义了一系列的默认音视频映射。
    /// 一个RTP 源有可能在会话中改变payload type，但是这个域在复用独立的媒体时是不同的。(见5.2 节)。
    /// 接收者必须忽略它不识别的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
    /// 每发送一个RTP 数据报文序列号值加一，接收者也可用来检测丢失的包或者重建报文序列。
    /// 初始的值是随机的，这样就使得known-plaintext 攻击更加困难， 即使源并没有加密(见9。1)，
    /// 因为要通过的translator 会做这些事情。关于选择随机数方面的技术见[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 数据报文中的第一个字段的采样时刻的时间瞬时值。
    /// 采样时间值必须是从恒定的和线性的时间中得到以便于同步和jitter 计算(见第6.4.1 节)。
    /// 必须保证同步和测量保温jitter 到来所需要的时间精度(一帧一个tick 一般情况下是不够的)。
    /// 时钟频率是与payload 所携带的数据格式有关的，在profile 中静态的定义或是在定义格式的payload format 中，
    /// 或通过non-RTP 方法所定义的payload format 中动态的定义。如果RTP 报文周期的生成，就采用虚拟的(nominal) 
    /// 采样时钟而不是从系统时钟读数。例如，在固定比特率的音频中，timestamp 时钟会在每个采样周期时加一。
    /// 如果音频应用中从输入设备中读入160 个采样周期的块，the timestamp 就会每一块增加160，
    /// 而不管块是否传输了或是丢弃了。
    /// 对于序列号来说，timestamp 初始值是随机的。只要它们是同时(逻辑上)同时生成的，
    /// 这些连续的的 RTP 报文就会有相同的timestamp，
    /// 例如，同属一个视频帧。正像在MPEG 中内插视频帧一样，
    /// 连续的但不是按顺序发送的RTP 报文可能含有相同的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 域识别同步源。为了防止在一个会话中有相同的同步源有相同的SSRC identifier， 
    /// 这个identifier 必须随机选取。
    /// 生成随机 identifier 的算法见目录A.6 。虽然选择相同的identifier 概率很小，
    /// 但是所有的RTP implementation 必须检测和解决冲突。
    /// 第8 章描述了冲突的概率和解决机制和RTP 级的检测机制，根据唯一的 SSRCidentifier 前向循环。
    /// 如果有源改变了它的源传输地址，
    /// 就必须为它选择一个新的SSRCidentifier 来避免被识别为循环过的源(见第8.2 节)。
    /// 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>
    /// 每一个RTP包中都有前12个字节定长的头字段
    /// The first twelve octets are present in every RTP packet
    /// </summary>
    public const int HeaderSize = 12;
    /// <summary>
    /// RTP消息头
    /// </summary>
    private byte[] _header;
    /// <summary>
    /// RTP消息头
    /// </summary>
    public byte[] Header { get { return _header; } }

    /// <summary>
    /// RTP有效载荷长度
    /// </summary>
    private int _payloadSize;
    /// <summary>
    /// RTP有效载荷长度
    /// </summary>
    public int PayloadSize { get { return _payloadSize; } }

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

    /// <summary>
    /// RTP消息总长度，包括Header和Payload
    /// </summary>
    public int Length { get { return HeaderSize + PayloadSize; } }

    /// <summary>
    /// RTP(RFC3550)协议数据包
    /// </summary>
    /// <param name="playloadType">数据报文有效载荷类型</param>
    /// <param name="sequenceNumber">数据报文序列号值</param>
    /// <param name="timestamp">数据报文采样时刻</param>
    /// <param name="data">数据</param>
    /// <param name="dataSize">数据长度</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)协议数据包
    /// </summary>
    /// <param name="playloadType">数据报文有效载荷类型</param>
    /// <param name="sequenceNumber">数据报文序列号值</param>
    /// <param name="timestamp">数据报文采样时刻</param>
    /// <param name="frame">图片</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)协议数据包
    /// </summary>
    /// <param name="packet">数据包</param>
    /// <param name="packetSize">数据包长度</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>
    /// 将消息转换成byte数组
    /// </summary>
    /// <returns>消息byte数组</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>
    /// 将消息体转换成图片
    /// </summary>
    /// <returns>图片</returns>
    public Bitmap ToBitmap()
    {
      return new Bitmap(new MemoryStream(_payload));
    }

    /// <summary>
    /// 将消息体转换成图片
    /// </summary>
    /// <returns>图片</returns>
    public Image ToImage()
    {
      return Image.FromStream(new MemoryStream(_payload));
    }

    /// <summary>
    /// 将图片转换成消息
    /// </summary>
    /// <param name="playloadType">数据报文有效载荷类型</param>
    /// <param name="sequenceNumber">数据报文序列号值</param>
    /// <param name="timestamp">数据报文采样时刻</param>
    /// <param name="frame">图片帧</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);
    }
  }