使用C#實現RTP資料包傳輸 參照RFC3550

閑暇時折騰IP網路視頻監控系統，需要可使用視訊幀資料包在網路內的傳輸。
未採用H.264或MPEG4等編碼壓縮方式，直接使用Bitmap圖片。
由於對幀的準確到達要求不好，所以採用UDP傳輸。如果發生網路丟包現象則直接將幀丟棄。
為了記錄資料包的傳輸順序和幀的時間戳記，所以研究了下RFC3550協議，採用RTP包封裝視訊框架。
並未全面深究，所以未使用SSRC和CSRC，因為不確切瞭解其用意。不過目前的實現情況已經足夠了。

複製代碼 代碼如下:/// <summary>
/// RTP(RFC3550)協議資料包
/// </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，當前規範定義值為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 位元組，這不屬於payload。
/// 最後一個位元組的padding 有一個計數器，標識需要忽略多少個padding 位元組(包括自己)。
/// 一些密碼編譯演算法可能需要固定塊長度的padding，或者是為了在更低層資料單元中攜帶一些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，定長頭欄位後面會有一個頭擴充。
/// 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 標識了定長頭欄位中包含的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 域中的位元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
/// 這個欄位定一個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 的預設靜態映射。
/// 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]中定義了一系列的預設音視頻映射。
/// 一個RTP 源有可能在會話中改變payload type，但是這個域在複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 資料報文序號值加一，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 資料報文中的第一個欄位的採樣時刻的時間瞬時值。
/// 採樣時間值必須是從恒定的和線性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)。
/// 時鐘頻率是與payload 所攜帶的資料格式有關的，在profile 中靜態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 中動態定義。如果RTP 報文周期的產生，就採用虛擬(nominal)
/// 採樣時鐘而不是從系統時鐘讀數。例如，在固定位元速率的音頻中，timestamp 時鐘會在每個採樣周期時加一。
/// 如果音頻應用中從輸入裝置中讀入160 個採樣周期的塊，the timestamp 就會每一塊增加160，
/// 而不管塊是否傳輸了或是丟棄了。
/// 對於序號來說，timestamp 初始值是隨機的。只要它們是同時(邏輯上)同時產生的，
/// 這些連續的的 RTP 報文就會有相同的timestamp，
/// 例如，同屬一個視訊框架。正像在MPEG 中插入視訊框架一樣，
/// 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。
/// 產生隨機 identifier 的演算法見目錄A.6 。雖然選擇相同的identifier 機率很小，
/// 但是所有的RTP implementation 必須檢測和解決衝突。
/// 第8 章描述了衝突的機率和解決機制和RTP 級的檢測機制，根據唯一的 SSRCidentifier 前向迴圈。
/// 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 來避免被識別為迴圈過的源(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個位元組定長的頭欄位
/// The first twelve octets are present in every RTP packet
/// </summary>
public const int HeaderSize = 12;
/// <summary>
/// RTP訊息頭
/// </summary>
private byte[] _header;
/// <summary>
/// RTP訊息頭
/// </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訊息總長度，包括Header和Payload
/// </summary>
public int Length { get { return HeaderSize + PayloadSize; } }

/// <summary>
/// RTP(RFC3550)協議資料包
/// </summary>
/// <param name="playloadType">資料報文有效載荷類型</param>
/// <param name="sequenceNumber">資料報文序號值</param>
/// <param name="timestamp">資料報文採樣時刻</param>
/// <param name="data">資料</param>
/// <param name="dataSize">資料長度</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)協議資料包
/// </summary>
/// <param name="playloadType">資料報文有效載荷類型</param>
/// <param name="sequenceNumber">資料報文序號值</param>
/// <param name="timestamp">資料報文採樣時刻</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)協議資料包
/// </summary>
/// <param name="packet">資料包</param>
/// <param name="packetSize">資料包長度</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>
/// 將訊息轉換成byte數組
/// </summary>
/// <returns>訊息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>
/// 將訊息體轉換成圖片
/// </summary>
/// <returns>圖片</returns>
public Bitmap ToBitmap()
{
return new Bitmap(new MemoryStream(_payload));
}

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

/// <summary>
/// 將圖片轉換成訊息
/// </summary>
/// <param name="playloadType">資料報文有效載荷類型</param>
/// <param name="sequenceNumber">資料報文序號值</param>
/// <param name="timestamp">資料報文採樣時刻</param>
/// <param name="frame">圖片幀</param>
/// <returns>
/// RTP訊息
/// </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);
}
}