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。

  1   /// <summary>
  2   /// RTP(RFC3550)协议数据包
  3   /// </summary>
  4   /// <remarks>
  5   /// The RTP header has the following format:
  6   ///  0                   1                   2                   3
  7   ///  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
  8   /// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  9   /// |V=2|P|X| CC    |M| PT          | sequence number               |
 10   /// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 11   /// | timestamp                                                     |
 12   /// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 13   /// | synchronization source (SSRC) identifier                      |
 14   /// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 15   /// | contributing source (CSRC) identifiers                        |
 16   /// | ....                                                          |
 17   /// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 18   /// </remarks>
 19   public class RtpPacket
 20   {
 21     /// <summary>
 22     /// version (V): 2 bits
 23     /// RTP版本标识,当前规范定义值为2.
 24     /// This field identifies the version of RTP. The version defined by this specification is two (2).
 25     /// (The value 1 is used by the first draft version of RTP and the value 0 is used by the protocol
 26     /// initially implemented in the \vat" audio tool.)
 27     /// </summary>
 28     public int Version { get { return 2; } }
 29 
 30     /// <summary>
 31     /// padding (P):1 bit
 32     /// 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。
 33     /// 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)。
 34     /// 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。
 35     /// If the padding bit is set, the packet contains one or more additional padding octets at the
 36     /// end which are not part of the payload. The last octet of the padding contains a count of
 37     /// how many padding octets should be ignored, including itself. Padding may be needed by
 38     /// some encryption algorithms with fixed block sizes or for carrying several RTP packets in a
 39     /// lower-layer protocol data unit.
 40     /// </summary>
 41     public int Padding { get { return 0; } }
 42 
 43     /// <summary>
 44     /// extension (X):1 bit 
 45     /// 如果设定了extension 位,定长头字段后面会有一个头扩展。
 46     /// If the extension bit is set, the fixed header must be followed by exactly one header extensio.
 47     /// </summary>
 48     public int Extension { get { return 0; } }
 49 
 50     /// <summary>
 51     /// CSRC count (CC):4 bits 
 52     /// CSRC count 标识了定长头字段中包含的CSRC identifier 的数量。
 53     /// The CSRC count contains the number of CSRC identifiers that follow the fixed header.
 54     /// </summary>
 55     public int CC { get { return 0; } }
 56 
 57     /// <summary>
 58     /// marker (M):1 bit 
 59     /// 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。
 60     /// 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.
 61     /// The interpretation of the marker is defined by a profile. It is intended to allow significant
 62     /// events such as frame boundaries to be marked in the packet stream. A profile may define
 63     /// additional marker bits or specify that there is no marker bit by changing the number of bits
 64     /// in the payload type field.
 65     /// </summary>
 66     public int Marker { get { return 0; } }
 67 
 68     /// <summary>
 69     /// payload type (PT):7 bits
 70     /// 这个字段定一个RTPpayload 的格式和在应用中定义解释。
 71     /// profile 可能指定一个从payload type 码字到payload format 的默认静态映射。
 72     /// 也可以通过non-RTP 方法来定义附加的payload type 码字(见第3 章)。
 73     /// 在 RFC 3551[1]中定义了一系列的默认音视频映射。
 74     /// 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)。
 75     /// 接收者必须忽略它不识别的payload type。
 76     /// This field identifies the format of the RTP payload and determines its interpretation by the
 77     /// application. A profile may specify a default static mapping of payload type codes to payload
 78     /// formats. Additional payload type codes may be defined dynamically through non-RTP means
 79     /// (see Section 3). A set of default mappings for audio and video is specified in the companion
 80     /// RFC 3551 [1]. An RTP source may change the payload type during a session, but this field
 81     /// should not be used for multiplexing separate media streams (see Section 5.2).
 82     /// A receiver must ignore packets with payload types that it does not understand.
 83     /// </summary>
 84     public RtpPayloadType PayloadType { get; private set; }
 85 
 86     /// <summary>
 87     /// sequence number:16 bits
 88     /// 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。
 89     /// 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),
 90     /// 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]。
 91     /// The sequence number increments by one for each RTP data packet sent, and may be used
 92     /// by the receiver to detect packet loss and to restore packet sequence. The initial value of the
 93     /// sequence number should be random (unpredictable) to make known-plaintext attacks on
 94     /// encryption more dificult, even if the source itself does not encrypt according to the method
 95     /// in Section 9.1, because the packets may flow through a translator that does. Techniques for
 96     /// choosing unpredictable numbers are discussed in [17].
 97     /// </summary>
 98     public int SequenceNumber { get; private set; }
 99 
100     /// <summary>
101     /// timestamp:32 bits
102     /// 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。
103     /// 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)。
104     /// 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)。
105     /// 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,
106     /// 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) 
107     /// 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。
108     /// 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,
109     /// 而不管块是否传输了或是丢弃了。
110     /// 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,
111     /// 这些连续的的 RTP 报文就会有相同的timestamp,
112     /// 例如,同属一个视频帧。正像在MPEG 中内插视频帧一样,
113     /// 连续的但不是按顺序发送的RTP 报文可能含有相同的timestamp。
114     /// The timestamp reflects the sampling instant of the first octet in the RTP data packet. The
115     /// sampling instant must be derived from a clock that increments monotonically and linearly
116     /// in time to allow synchronization and jitter calculations (see Section 6.4.1). The resolution
117     /// of the clock must be suficient for the desired synchronization accuracy and for measuring
118     /// packet arrival jitter (one tick per video frame is typically not suficient). The clock frequency
119     /// is dependent on the format of data carried as payload and is specified statically in the profile
120     /// or payload format specification that defines the format, or may be specified dynamically for
121     /// payload formats defined through non-RTP means. If RTP packets are generated periodically,
122     /// the nominal sampling instant as determined from the sampling clock is to be used, not a
123     /// reading of the system clock. As an example, for fixed-rate audio the timestamp clock would
124     /// likely increment by one for each sampling period. If an audio application reads blocks covering
125     /// 160 sampling periods from the input device, the timestamp would be increased by 160 for
126     /// each such block, regardless of whether the block is transmitted in a packet or dropped as silent.
127     /// </summary>
128     public long Timestamp { get; private set; }
129 
130     /// <summary>
131     /// SSRC:32 bits
132     /// 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, 
133     /// 这个identifier 必须随机选取。
134     /// 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,
135     /// 但是所有的RTP implementation 必须检测和解决冲突。
136     /// 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。
137     /// 如果有源改变了它的源传输地址,
138     /// 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)。
139     /// The SSRC field identifies the synchronization source. This identifier should be chosen
140     /// randomly, with the intent that no two synchronization sources within the same RTP session
141     /// will have the same SSRC identifier. An example algorithm for generating a random identifier
142     /// is presented in Appendix A.6. Although the probability of multiple sources choosing the same
143     /// identifier is low, all RTP implementations must be prepared to detect and resolve collisions.
144     /// Section 8 describes the probability of collision along with a mechanism for resolving collisions
145     /// and detecting RTP-level forwarding loops based on the uniqueness of the SSRC identifier. If
146     /// a source changes its source transport address, it must also choose a new SSRC identifier to
147     /// avoid being interpreted as a looped source (see Section 8.2).
148     /// </summary>
149     public int SSRC { get { return 0; } }
150 
151     /// <summary>
152     /// 每一个RTP包中都有前12个字节定长的头字段
153     /// The first twelve octets are present in every RTP packet
154     /// </summary>
155     public const int HeaderSize = 12;
156     /// <summary>
157     /// RTP消息头
158     /// </summary>
159     private byte[] _header;
160     /// <summary>
161     /// RTP消息头
162     /// </summary>
163     public byte[] Header { get { return _header; } }
164 
165     /// <summary>
166     /// RTP有效载荷长度
167     /// </summary>
168     private int _payloadSize;
169     /// <summary>
170     /// RTP有效载荷长度
171     /// </summary>
172     public int PayloadSize { get { return _payloadSize; } }
173 
174     /// <summary>
175     /// RTP有效载荷
176     /// </summary>
177     private byte[] _payload;
178     /// <summary>
179     /// RTP有效载荷
180     /// </summary>
181     public byte[] Payload { get { return _payload; } }
182 
183     /// <summary>
184     /// RTP消息总长度,包括Header和Payload
185     /// </summary>
186     public int Length { get { return HeaderSize + PayloadSize; } }
187 
188     /// <summary>
189     /// RTP(RFC3550)协议数据包
190     /// </summary>
191     /// <param name="playloadType">数据报文有效载荷类型</param>
192     /// <param name="sequenceNumber">数据报文序列号值</param>
193     /// <param name="timestamp">数据报文采样时刻</param>
194     /// <param name="data">数据</param>
195     /// <param name="dataSize">数据长度</param>
196     public RtpPacket(
197       RtpPayloadType playloadType, 
198       int sequenceNumber, 
199       long timestamp, 
200       byte[] data, 
201       int dataSize)
202     {
203       // fill changing header fields
204       SequenceNumber = sequenceNumber;
205       Timestamp = timestamp;
206       PayloadType = playloadType;
207 
208       // build the header bistream
209       _header = new byte[HeaderSize];
210 
211       // fill the header array of byte with RTP header fields
212       _header[0] = (byte)((Version << 6) | (Padding << 5) | (Extension << 4) | CC);
213       _header[1] = (byte)((Marker << 7) | (int)PayloadType);
214       _header[2] = (byte)(SequenceNumber >> 8);
215       _header[3] = (byte)(SequenceNumber);
216       for (int i = 0; i < 4; i++)
217       {
218         _header[7 - i] = (byte)(Timestamp >> (8 * i));
219       }
220       for (int i = 0; i < 4; i++)
221       {
222         _header[11 - i] = (byte)(SSRC >> (8 * i));
223       }
224 
225       // fill the payload bitstream
226       _payload = new byte[dataSize];
227       _payloadSize = dataSize;
228 
229       // fill payload array of byte from data (given in parameter of the constructor)
230       Array.Copy(data, 0, _payload, 0, dataSize);
231     }
232 
233     /// <summary>
234     /// RTP(RFC3550)协议数据包
235     /// </summary>
236     /// <param name="playloadType">数据报文有效载荷类型</param>
237     /// <param name="sequenceNumber">数据报文序列号值</param>
238     /// <param name="timestamp">数据报文采样时刻</param>
239     /// <param name="frame">图片</param>
240     public RtpPacket(
241       RtpPayloadType playloadType, 
242       int sequenceNumber, 
243       long timestamp, 
244       Image frame)
245     {
246       // fill changing header fields
247       SequenceNumber = sequenceNumber;
248       Timestamp = timestamp;
249       PayloadType = playloadType;
250 
251       // build the header bistream
252       _header = new byte[HeaderSize];
253 
254       // fill the header array of byte with RTP header fields
255       _header[0] = (byte)((Version << 6) | (Padding << 5) | (Extension << 4) | CC);
256       _header[1] = (byte)((Marker << 7) | (int)PayloadType);
257       _header[2] = (byte)(SequenceNumber >> 8);
258       _header[3] = (byte)(SequenceNumber);
259       for (int i = 0; i < 4; i++)
260       {
261         _header[7 - i] = (byte)(Timestamp >> (8 * i));
262       }
263       for (int i = 0; i < 4; i++)
264       {
265         _header[11 - i] = (byte)(SSRC >> (8 * i));
266       }
267 
268       // fill the payload bitstream
269       using (MemoryStream ms = new MemoryStream())
270       {
271         frame.Save(ms, ImageFormat.Jpeg);
272         _payload = ms.ToArray();
273         _payloadSize = _payload.Length;
274       }
275     }
276 
277     /// <summary>
278     /// RTP(RFC3550)协议数据包
279     /// </summary>
280     /// <param name="packet">数据包</param>
281     /// <param name="packetSize">数据包长度</param>
282     public RtpPacket(byte[] packet, int packetSize)
283     {
284       //check if total packet size is lower than the header size
285       if (packetSize >= HeaderSize)
286       {
287         //get the header bitsream
288         _header = new byte[HeaderSize];
289         for (int i = 0; i < HeaderSize; i++)
290         {
291           _header[i] = packet[i];
292         }
293 
294         //get the payload bitstream
295         _payloadSize = packetSize - HeaderSize;
296         _payload = new byte[_payloadSize];
297         for (int i = HeaderSize; i < packetSize; i++)
298         {
299           _payload[i - HeaderSize] = packet[i];
300         }
301 
302         //interpret the changing fields of the header
303         PayloadType = (RtpPayloadType)(_header[1] & 127);
304         SequenceNumber = UnsignedInt(_header[3]) + 256 * UnsignedInt(_header[2]);
305         Timestamp = UnsignedInt(_header[7]) 
306           + 256 * UnsignedInt(_header[6]) 
307           + 65536 * UnsignedInt(_header[5]) 
308           + 16777216 * UnsignedInt(_header[4]);
309       }
310     }
311 
312     /// <summary>
313     /// 将消息转换成byte数组
314     /// </summary>
315     /// <returns>消息byte数组</returns>
316     public byte[] ToArray()
317     {
318       byte[] packet = new byte[Length];
319 
320       Array.Copy(_header, 0, packet, 0, HeaderSize);
321       Array.Copy(_payload, 0, packet, HeaderSize, PayloadSize);
322 
323       return packet;
324     }
325 
326     /// <summary>
327     /// 将消息体转换成图片
328     /// </summary>
329     /// <returns>图片</returns>
330     public Bitmap ToBitmap()
331     {
332       return new Bitmap(new MemoryStream(_payload));
333     }
334 
335     /// <summary>
336     /// 将消息体转换成图片
337     /// </summary>
338     /// <returns>图片</returns>
339     public Image ToImage()
340     {
341       return Image.FromStream(new MemoryStream(_payload));
342     }
343 
344     /// <summary>
345     /// 将图片转换成消息
346     /// </summary>
347     /// <param name="playloadType">数据报文有效载荷类型</param>
348     /// <param name="sequenceNumber">数据报文序列号值</param>
349     /// <param name="timestamp">数据报文采样时刻</param>
350     /// <param name="frame">图片帧</param>
351     /// <returns>
352     /// RTP消息
353     /// </returns>
354     public static RtpPacket FromImage(
355       RtpPayloadType playloadType, 
356       int sequenceNumber, 
357       long timestamp, 
358       Image frame)
359     {
360       return new RtpPacket(playloadType, sequenceNumber, timestamp, frame);
361     }
362 
363     /// <summary>
364     /// return the unsigned value of 8-bit integer nb
365     /// </summary>
366     /// <param name="nb"></param>
367     /// <returns></returns>
368     private static int UnsignedInt(int nb)
369     {
370       if (nb >= 0)
371         return (nb);
372       else
373         return (256 + nb);
374     }
375   }

 

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