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rtsp.h
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1 /*
2  * RTSP definitions
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 #ifndef AVFORMAT_RTSP_H
22 #define AVFORMAT_RTSP_H
23 
24 #include <stdint.h>
25 #include "avformat.h"
26 #include "rtspcodes.h"
27 #include "rtpdec.h"
28 #include "network.h"
29 #include "httpauth.h"
30 
31 #include "libavutil/log.h"
32 #include "libavutil/opt.h"
33 
34 /**
35  * Network layer over which RTP/etc packet data will be transported.
36  */
38  RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
39  RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
40  RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
42  RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
43  transport mode as such,
44  only for use via AVOptions */
45  RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
46  option for lower_transport_mask,
47  but set in the SDP demuxer based
48  on a flag. */
49 };
50 
51 /**
52  * Packet profile of the data that we will be receiving. Real servers
53  * commonly send RDT (although they can sometimes send RTP as well),
54  * whereas most others will send RTP.
55  */
57  RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
58  RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
59  RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
61 };
62 
63 /**
64  * Transport mode for the RTSP data. This may be plain, or
65  * tunneled, which is done over HTTP.
66  */
68  RTSP_MODE_PLAIN, /**< Normal RTSP */
69  RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
70 };
71 
72 #define RTSP_DEFAULT_PORT 554
73 #define RTSP_MAX_TRANSPORTS 8
74 #define RTSP_TCP_MAX_PACKET_SIZE 1472
75 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
76 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
77 #define RTSP_RTP_PORT_MIN 5000
78 #define RTSP_RTP_PORT_MAX 65000
79 
80 /**
81  * This describes a single item in the "Transport:" line of one stream as
82  * negotiated by the SETUP RTSP command. Multiple transports are comma-
83  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
84  * client_port=1000-1001;server_port=1800-1801") and described in separate
85  * RTSPTransportFields.
86  */
87 typedef struct RTSPTransportField {
88  /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
89  * with a '$', stream length and stream ID. If the stream ID is within
90  * the range of this interleaved_min-max, then the packet belongs to
91  * this stream. */
93 
94  /** UDP multicast port range; the ports to which we should connect to
95  * receive multicast UDP data. */
97 
98  /** UDP client ports; these should be the local ports of the UDP RTP
99  * (and RTCP) sockets over which we receive RTP/RTCP data. */
101 
102  /** UDP unicast server port range; the ports to which we should connect
103  * to receive unicast UDP RTP/RTCP data. */
105 
106  /** time-to-live value (required for multicast); the amount of HOPs that
107  * packets will be allowed to make before being discarded. */
108  int ttl;
109 
110  /** transport set to record data */
112 
113  struct sockaddr_storage destination; /**< destination IP address */
114  char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
115 
116  /** data/packet transport protocol; e.g. RTP or RDT */
118 
119  /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
122 
123 /**
124  * This describes the server response to each RTSP command.
125  */
126 typedef struct RTSPMessageHeader {
127  /** length of the data following this header */
129 
130  enum RTSPStatusCode status_code; /**< response code from server */
131 
132  /** number of items in the 'transports' variable below */
134 
135  /** Time range of the streams that the server will stream. In
136  * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
138 
139  /** describes the complete "Transport:" line of the server in response
140  * to a SETUP RTSP command by the client */
142 
143  int seq; /**< sequence number */
144 
145  /** the "Session:" field. This value is initially set by the server and
146  * should be re-transmitted by the client in every RTSP command. */
147  char session_id[512];
148 
149  /** the "Location:" field. This value is used to handle redirection.
150  */
151  char location[4096];
152 
153  /** the "RealChallenge1:" field from the server */
154  char real_challenge[64];
155 
156  /** the "Server: field, which can be used to identify some special-case
157  * servers that are not 100% standards-compliant. We use this to identify
158  * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
159  * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
160  * use something like "Helix [..] Server Version v.e.r.sion (platform)
161  * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
162  * where platform is the output of $uname -msr | sed 's/ /-/g'. */
163  char server[64];
164 
165  /** The "timeout" comes as part of the server response to the "SETUP"
166  * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
167  * time, in seconds, that the server will go without traffic over the
168  * RTSP/TCP connection before it closes the connection. To prevent
169  * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
170  * than this value. */
171  int timeout;
172 
173  /** The "Notice" or "X-Notice" field value. See
174  * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
175  * for a complete list of supported values. */
176  int notice;
177 
178  /** The "reason" is meant to specify better the meaning of the error code
179  * returned
180  */
181  char reason[256];
182 
183  /**
184  * Content type header
185  */
186  char content_type[64];
188 
189 /**
190  * Client state, i.e. whether we are currently receiving data (PLAYING) or
191  * setup-but-not-receiving (PAUSED). State can be changed in applications
192  * by calling av_read_play/pause().
193  */
195  RTSP_STATE_IDLE, /**< not initialized */
196  RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
197  RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
198  RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
199 };
200 
201 /**
202  * Identify particular servers that require special handling, such as
203  * standards-incompliant "Transport:" lines in the SETUP request.
204  */
206  RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
207  RTSP_SERVER_REAL, /**< Realmedia-style server */
208  RTSP_SERVER_WMS, /**< Windows Media server */
210 };
211 
212 /**
213  * Private data for the RTSP demuxer.
214  *
215  * @todo Use AVIOContext instead of URLContext
216  */
217 typedef struct RTSPState {
218  const AVClass *class; /**< Class for private options. */
219  URLContext *rtsp_hd; /* RTSP TCP connection handle */
220 
221  /** number of items in the 'rtsp_streams' variable */
223 
224  struct RTSPStream **rtsp_streams; /**< streams in this session */
225 
226  /** indicator of whether we are currently receiving data from the
227  * server. Basically this isn't more than a simple cache of the
228  * last PLAY/PAUSE command sent to the server, to make sure we don't
229  * send 2x the same unexpectedly or commands in the wrong state. */
231 
232  /** the seek value requested when calling av_seek_frame(). This value
233  * is subsequently used as part of the "Range" parameter when emitting
234  * the RTSP PLAY command. If we are currently playing, this command is
235  * called instantly. If we are currently paused, this command is called
236  * whenever we resume playback. Either way, the value is only used once,
237  * see rtsp_read_play() and rtsp_read_seek(). */
238  int64_t seek_timestamp;
239 
240  int seq; /**< RTSP command sequence number */
241 
242  /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
243  * identifier that the client should re-transmit in each RTSP command */
244  char session_id[512];
245 
246  /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
247  * the server will go without traffic on the RTSP/TCP line before it
248  * closes the connection. */
249  int timeout;
250 
251  /** timestamp of the last RTSP command that we sent to the RTSP server.
252  * This is used to calculate when to send dummy commands to keep the
253  * connection alive, in conjunction with timeout. */
254  int64_t last_cmd_time;
255 
256  /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
258 
259  /** the negotiated network layer transport protocol; e.g. TCP or UDP
260  * uni-/multicast */
262 
263  /** brand of server that we're talking to; e.g. WMS, REAL or other.
264  * Detected based on the value of RTSPMessageHeader->server or the presence
265  * of RTSPMessageHeader->real_challenge */
267 
268  /** the "RealChallenge1:" field from the server */
269  char real_challenge[64];
270 
271  /** plaintext authorization line (username:password) */
272  char auth[128];
273 
274  /** authentication state */
276 
277  /** The last reply of the server to a RTSP command */
278  char last_reply[2048]; /* XXX: allocate ? */
279 
280  /** RTSPStream->transport_priv of the last stream that we read a
281  * packet from */
283 
284  /** The following are used for Real stream selection */
285  //@{
286  /** whether we need to send a "SET_PARAMETER Subscribe:" command */
288 
289  /** stream setup during the last frame read. This is used to detect if
290  * we need to subscribe or unsubscribe to any new streams. */
292 
293  /** current stream setup. This is a temporary buffer used to compare
294  * current setup to previous frame setup. */
296 
297  /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
298  * this is used to send the same "Unsubscribe:" if stream setup changed,
299  * before sending a new "Subscribe:" command. */
300  char last_subscription[1024];
301  //@}
302 
303  /** The following are used for RTP/ASF streams */
304  //@{
305  /** ASF demuxer context for the embedded ASF stream from WMS servers */
307 
308  /** cache for position of the asf demuxer, since we load a new
309  * data packet in the bytecontext for each incoming RTSP packet. */
310  uint64_t asf_pb_pos;
311  //@}
312 
313  /** some MS RTSP streams contain a URL in the SDP that we need to use
314  * for all subsequent RTSP requests, rather than the input URI; in
315  * other cases, this is a copy of AVFormatContext->filename. */
316  char control_uri[1024];
317 
318  /** The following are used for parsing raw mpegts in udp */
319  //@{
320  struct MpegTSContext *ts;
323  //@}
324 
325  /** Additional output handle, used when input and output are done
326  * separately, eg for HTTP tunneling. */
328 
329  /** RTSP transport mode, such as plain or tunneled. */
331 
332  /* Number of RTCP BYE packets the RTSP session has received.
333  * An EOF is propagated back if nb_byes == nb_streams.
334  * This is reset after a seek. */
335  int nb_byes;
336 
337  /** Reusable buffer for receiving packets */
339 
340  /**
341  * A mask with all requested transport methods
342  */
344 
345  /**
346  * The number of returned packets
347  */
348  uint64_t packets;
349 
350  /**
351  * Polling array for udp
352  */
353  struct pollfd *p;
354 
355  /**
356  * Whether the server supports the GET_PARAMETER method.
357  */
359 
360  /**
361  * Do not begin to play the stream immediately.
362  */
364 
365  /**
366  * Option flags for the chained RTP muxer.
367  */
369 
370  /** Whether the server accepts the x-Dynamic-Rate header */
372 
373  /**
374  * Various option flags for the RTSP muxer/demuxer.
375  */
377 
378  /**
379  * Mask of all requested media types
380  */
382 
383  /**
384  * Minimum and maximum local UDP ports.
385  */
387 
388  /**
389  * Timeout to wait for incoming connections.
390  */
392 
393  /**
394  * timeout of socket i/o operations.
395  */
396  int stimeout;
397 
398  /**
399  * Size of RTP packet reordering queue.
400  */
402 
403  /**
404  * User-Agent string
405  */
406  char *user_agent;
407 } RTSPState;
408 
409 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
410  receive packets only from the right
411  source address and port. */
412 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
413 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
414 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
415  address of received packets. */
416 
417 typedef struct RTSPSource {
418  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
419 } RTSPSource;
420 
421 /**
422  * Describe a single stream, as identified by a single m= line block in the
423  * SDP content. In the case of RDT, one RTSPStream can represent multiple
424  * AVStreams. In this case, each AVStream in this set has similar content
425  * (but different codec/bitrate).
426  */
427 typedef struct RTSPStream {
428  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
429  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
430 
431  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
433 
434  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
435  * for the selected transport. Only used for TCP. */
437 
438  char control_url[1024]; /**< url for this stream (from SDP) */
439 
440  /** The following are used only in SDP, not RTSP */
441  //@{
442  int sdp_port; /**< port (from SDP content) */
443  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
444  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
445  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
446  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
447  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
448  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
449  int sdp_payload_type; /**< payload type */
450  //@}
451 
452  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
453  //@{
454  /** handler structure */
456 
457  /** private data associated with the dynamic protocol */
459  //@}
460 
461  /** Enable sending RTCP feedback messages according to RFC 4585 */
462  int feedback;
463 
464  char crypto_suite[40];
465  char crypto_params[100];
466 } RTSPStream;
467 
468 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
469  RTSPState *rt, const char *method);
470 
471 /**
472  * Send a command to the RTSP server without waiting for the reply.
473  *
474  * @see rtsp_send_cmd_with_content_async
475  */
476 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
477  const char *url, const char *headers);
478 
479 /**
480  * Send a command to the RTSP server and wait for the reply.
481  *
482  * @param s RTSP (de)muxer context
483  * @param method the method for the request
484  * @param url the target url for the request
485  * @param headers extra header lines to include in the request
486  * @param reply pointer where the RTSP message header will be stored
487  * @param content_ptr pointer where the RTSP message body, if any, will
488  * be stored (length is in reply)
489  * @param send_content if non-null, the data to send as request body content
490  * @param send_content_length the length of the send_content data, or 0 if
491  * send_content is null
492  *
493  * @return zero if success, nonzero otherwise
494  */
496  const char *method, const char *url,
497  const char *headers,
498  RTSPMessageHeader *reply,
499  unsigned char **content_ptr,
500  const unsigned char *send_content,
501  int send_content_length);
502 
503 /**
504  * Send a command to the RTSP server and wait for the reply.
505  *
506  * @see rtsp_send_cmd_with_content
507  */
508 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
509  const char *url, const char *headers,
510  RTSPMessageHeader *reply, unsigned char **content_ptr);
511 
512 /**
513  * Read a RTSP message from the server, or prepare to read data
514  * packets if we're reading data interleaved over the TCP/RTSP
515  * connection as well.
516  *
517  * @param s RTSP (de)muxer context
518  * @param reply pointer where the RTSP message header will be stored
519  * @param content_ptr pointer where the RTSP message body, if any, will
520  * be stored (length is in reply)
521  * @param return_on_interleaved_data whether the function may return if we
522  * encounter a data marker ('$'), which precedes data
523  * packets over interleaved TCP/RTSP connections. If this
524  * is set, this function will return 1 after encountering
525  * a '$'. If it is not set, the function will skip any
526  * data packets (if they are encountered), until a reply
527  * has been fully parsed. If no more data is available
528  * without parsing a reply, it will return an error.
529  * @param method the RTSP method this is a reply to. This affects how
530  * some response headers are acted upon. May be NULL.
531  *
532  * @return 1 if a data packets is ready to be received, -1 on error,
533  * and 0 on success.
534  */
536  unsigned char **content_ptr,
537  int return_on_interleaved_data, const char *method);
538 
539 /**
540  * Skip a RTP/TCP interleaved packet.
541  */
543 
544 /**
545  * Connect to the RTSP server and set up the individual media streams.
546  * This can be used for both muxers and demuxers.
547  *
548  * @param s RTSP (de)muxer context
549  *
550  * @return 0 on success, < 0 on error. Cleans up all allocations done
551  * within the function on error.
552  */
554 
555 /**
556  * Close and free all streams within the RTSP (de)muxer
557  *
558  * @param s RTSP (de)muxer context
559  */
561 
562 /**
563  * Close all connection handles within the RTSP (de)muxer
564  *
565  * @param s RTSP (de)muxer context
566  */
568 
569 /**
570  * Get the description of the stream and set up the RTSPStream child
571  * objects.
572  */
574 
575 /**
576  * Announce the stream to the server and set up the RTSPStream child
577  * objects for each media stream.
578  */
580 
581 /**
582  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
583  * listen mode.
584  */
586 
587 /**
588  * Parse an SDP description of streams by populating an RTSPState struct
589  * within the AVFormatContext; also allocate the RTP streams and the
590  * pollfd array used for UDP streams.
591  */
592 int ff_sdp_parse(AVFormatContext *s, const char *content);
593 
594 /**
595  * Receive one RTP packet from an TCP interleaved RTSP stream.
596  */
598  uint8_t *buf, int buf_size);
599 
600 /**
601  * Send buffered packets over TCP.
602  */
604 
605 /**
606  * Receive one packet from the RTSPStreams set up in the AVFormatContext
607  * (which should contain a RTSPState struct as priv_data).
608  */
610 
611 /**
612  * Do the SETUP requests for each stream for the chosen
613  * lower transport mode.
614  * @return 0 on success, <0 on error, 1 if protocol is unavailable
615  */
616 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
617  int lower_transport, const char *real_challenge);
618 
619 /**
620  * Undo the effect of ff_rtsp_make_setup_request, close the
621  * transport_priv and rtp_handle fields.
622  */
623 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
624 
625 /**
626  * Open RTSP transport context.
627  */
629 
630 extern const AVOption ff_rtsp_options[];
631 
632 #endif /* AVFORMAT_RTSP_H */