C#实现RTP数据包传输参照RFC3550
C#实现RTP数据包传输参照RFC3550
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 }
标签: C# , .NET , RTP
作者： Leo_wl
出处： http://www.cnblogs.com/Leo_wl/
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