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。
复制代码 代码如下:
/// <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);
}
}
【使用C#实现RTP数据包传输 参照RFC3550】相关文章：
★ 使用VS2010 C#开发ActiveX控件(下),完整代码打包下载