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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 RTSPS_DEFAULT_PORT 322
74 #define RTSP_MAX_TRANSPORTS 8
75 #define RTSP_TCP_MAX_PACKET_SIZE 1472
76 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
77 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
78 #define RTSP_RTP_PORT_MIN 5000
79 #define RTSP_RTP_PORT_MAX 65000
80 
81 /**
82  * This describes a single item in the "Transport:" line of one stream as
83  * negotiated by the SETUP RTSP command. Multiple transports are comma-
84  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
85  * client_port=1000-1001;server_port=1800-1801") and described in separate
86  * RTSPTransportFields.
87  */
88 typedef struct RTSPTransportField {
89  /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
90  * with a '$', stream length and stream ID. If the stream ID is within
91  * the range of this interleaved_min-max, then the packet belongs to
92  * this stream. */
94 
95  /** UDP multicast port range; the ports to which we should connect to
96  * receive multicast UDP data. */
98 
99  /** UDP client ports; these should be the local ports of the UDP RTP
100  * (and RTCP) sockets over which we receive RTP/RTCP data. */
102 
103  /** UDP unicast server port range; the ports to which we should connect
104  * to receive unicast UDP RTP/RTCP data. */
106 
107  /** time-to-live value (required for multicast); the amount of HOPs that
108  * packets will be allowed to make before being discarded. */
109  int ttl;
110 
111  /** transport set to record data */
113 
114  struct sockaddr_storage destination; /**< destination IP address */
115  char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
116 
117  /** data/packet transport protocol; e.g. RTP or RDT */
119 
120  /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
123 
124 /**
125  * This describes the server response to each RTSP command.
126  */
127 typedef struct RTSPMessageHeader {
128  /** length of the data following this header */
130 
131  enum RTSPStatusCode status_code; /**< response code from server */
132 
133  /** number of items in the 'transports' variable below */
135 
136  /** Time range of the streams that the server will stream. In
137  * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
139 
140  /** describes the complete "Transport:" line of the server in response
141  * to a SETUP RTSP command by the client */
143 
144  int seq; /**< sequence number */
145 
146  /** the "Session:" field. This value is initially set by the server and
147  * should be re-transmitted by the client in every RTSP command. */
148  char session_id[512];
149 
150  /** the "Location:" field. This value is used to handle redirection.
151  */
152  char location[4096];
153 
154  /** the "RealChallenge1:" field from the server */
155  char real_challenge[64];
156 
157  /** the "Server: field, which can be used to identify some special-case
158  * servers that are not 100% standards-compliant. We use this to identify
159  * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
160  * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
161  * use something like "Helix [..] Server Version v.e.r.sion (platform)
162  * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
163  * where platform is the output of $uname -msr | sed 's/ /-/g'. */
164  char server[64];
165 
166  /** The "timeout" comes as part of the server response to the "SETUP"
167  * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
168  * time, in seconds, that the server will go without traffic over the
169  * RTSP/TCP connection before it closes the connection. To prevent
170  * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
171  * than this value. */
172  int timeout;
173 
174  /** The "Notice" or "X-Notice" field value. See
175  * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
176  * for a complete list of supported values. */
177  int notice;
178 
179  /** The "reason" is meant to specify better the meaning of the error code
180  * returned
181  */
182  char reason[256];
183 
184  /**
185  * Content type header
186  */
187  char content_type[64];
189 
190 /**
191  * Client state, i.e. whether we are currently receiving data (PLAYING) or
192  * setup-but-not-receiving (PAUSED). State can be changed in applications
193  * by calling av_read_play/pause().
194  */
196  RTSP_STATE_IDLE, /**< not initialized */
197  RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
198  RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
199  RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
200 };
201 
202 /**
203  * Identify particular servers that require special handling, such as
204  * standards-incompliant "Transport:" lines in the SETUP request.
205  */
207  RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
208  RTSP_SERVER_REAL, /**< Realmedia-style server */
209  RTSP_SERVER_WMS, /**< Windows Media server */
211 };
212 
213 /**
214  * Private data for the RTSP demuxer.
215  *
216  * @todo Use AVIOContext instead of URLContext
217  */
218 typedef struct RTSPState {
219  const AVClass *class; /**< Class for private options. */
220  URLContext *rtsp_hd; /* RTSP TCP connection handle */
221 
222  /** number of items in the 'rtsp_streams' variable */
224 
225  struct RTSPStream **rtsp_streams; /**< streams in this session */
226 
227  /** indicator of whether we are currently receiving data from the
228  * server. Basically this isn't more than a simple cache of the
229  * last PLAY/PAUSE command sent to the server, to make sure we don't
230  * send 2x the same unexpectedly or commands in the wrong state. */
232 
233  /** the seek value requested when calling av_seek_frame(). This value
234  * is subsequently used as part of the "Range" parameter when emitting
235  * the RTSP PLAY command. If we are currently playing, this command is
236  * called instantly. If we are currently paused, this command is called
237  * whenever we resume playback. Either way, the value is only used once,
238  * see rtsp_read_play() and rtsp_read_seek(). */
239  int64_t seek_timestamp;
240 
241  int seq; /**< RTSP command sequence number */
242 
243  /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
244  * identifier that the client should re-transmit in each RTSP command */
245  char session_id[512];
246 
247  /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
248  * the server will go without traffic on the RTSP/TCP line before it
249  * closes the connection. */
250  int timeout;
251 
252  /** timestamp of the last RTSP command that we sent to the RTSP server.
253  * This is used to calculate when to send dummy commands to keep the
254  * connection alive, in conjunction with timeout. */
255  int64_t last_cmd_time;
256 
257  /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
259 
260  /** the negotiated network layer transport protocol; e.g. TCP or UDP
261  * uni-/multicast */
263 
264  /** brand of server that we're talking to; e.g. WMS, REAL or other.
265  * Detected based on the value of RTSPMessageHeader->server or the presence
266  * of RTSPMessageHeader->real_challenge */
268 
269  /** the "RealChallenge1:" field from the server */
270  char real_challenge[64];
271 
272  /** plaintext authorization line (username:password) */
273  char auth[128];
274 
275  /** authentication state */
277 
278  /** The last reply of the server to a RTSP command */
279  char last_reply[2048]; /* XXX: allocate ? */
280 
281  /** RTSPStream->transport_priv of the last stream that we read a
282  * packet from */
284 
285  /** The following are used for Real stream selection */
286  //@{
287  /** whether we need to send a "SET_PARAMETER Subscribe:" command */
289 
290  /** stream setup during the last frame read. This is used to detect if
291  * we need to subscribe or unsubscribe to any new streams. */
293 
294  /** current stream setup. This is a temporary buffer used to compare
295  * current setup to previous frame setup. */
297 
298  /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
299  * this is used to send the same "Unsubscribe:" if stream setup changed,
300  * before sending a new "Subscribe:" command. */
301  char last_subscription[1024];
302  //@}
303 
304  /** The following are used for RTP/ASF streams */
305  //@{
306  /** ASF demuxer context for the embedded ASF stream from WMS servers */
308 
309  /** cache for position of the asf demuxer, since we load a new
310  * data packet in the bytecontext for each incoming RTSP packet. */
311  uint64_t asf_pb_pos;
312  //@}
313 
314  /** some MS RTSP streams contain a URL in the SDP that we need to use
315  * for all subsequent RTSP requests, rather than the input URI; in
316  * other cases, this is a copy of AVFormatContext->filename. */
317  char control_uri[1024];
318 
319  /** The following are used for parsing raw mpegts in udp */
320  //@{
321  struct MpegTSContext *ts;
324  //@}
325 
326  /** Additional output handle, used when input and output are done
327  * separately, eg for HTTP tunneling. */
329 
330  /** RTSP transport mode, such as plain or tunneled. */
332 
333  /* Number of RTCP BYE packets the RTSP session has received.
334  * An EOF is propagated back if nb_byes == nb_streams.
335  * This is reset after a seek. */
336  int nb_byes;
337 
338  /** Reusable buffer for receiving packets */
340 
341  /**
342  * A mask with all requested transport methods
343  */
345 
346  /**
347  * The number of returned packets
348  */
349  uint64_t packets;
350 
351  /**
352  * Polling array for udp
353  */
354  struct pollfd *p;
355 
356  /**
357  * Whether the server supports the GET_PARAMETER method.
358  */
360 
361  /**
362  * Do not begin to play the stream immediately.
363  */
365 
366  /**
367  * Option flags for the chained RTP muxer.
368  */
370 
371  /** Whether the server accepts the x-Dynamic-Rate header */
373 
374  /**
375  * Various option flags for the RTSP muxer/demuxer.
376  */
378 
379  /**
380  * Mask of all requested media types
381  */
383 
384  /**
385  * Minimum and maximum local UDP ports.
386  */
388 
389  /**
390  * Timeout to wait for incoming connections.
391  */
393 
394  /**
395  * timeout of socket i/o operations.
396  */
397  int stimeout;
398 
399  /**
400  * Size of RTP packet reordering queue.
401  */
403 
404  /**
405  * User-Agent string
406  */
407  char *user_agent;
408 } RTSPState;
409 
410 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
411  receive packets only from the right
412  source address and port. */
413 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
414 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
415 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
416  address of received packets. */
417 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
418 
419 typedef struct RTSPSource {
420  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
421 } RTSPSource;
422 
423 /**
424  * Describe a single stream, as identified by a single m= line block in the
425  * SDP content. In the case of RDT, one RTSPStream can represent multiple
426  * AVStreams. In this case, each AVStream in this set has similar content
427  * (but different codec/bitrate).
428  */
429 typedef struct RTSPStream {
430  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
431  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
432 
433  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
435 
436  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
437  * for the selected transport. Only used for TCP. */
439 
440  char control_url[1024]; /**< url for this stream (from SDP) */
441 
442  /** The following are used only in SDP, not RTSP */
443  //@{
444  int sdp_port; /**< port (from SDP content) */
445  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
446  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
447  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
448  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
449  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
450  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
451  int sdp_payload_type; /**< payload type */
452  //@}
453 
454  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
455  //@{
456  /** handler structure */
458 
459  /** private data associated with the dynamic protocol */
461  //@}
462 
463  /** Enable sending RTCP feedback messages according to RFC 4585 */
464  int feedback;
465 
466  char crypto_suite[40];
467  char crypto_params[100];
468 } RTSPStream;
469 
470 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
471  RTSPState *rt, const char *method);
472 
473 /**
474  * Send a command to the RTSP server without waiting for the reply.
475  *
476  * @see rtsp_send_cmd_with_content_async
477  */
478 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
479  const char *url, const char *headers);
480 
481 /**
482  * Send a command to the RTSP server and wait for the reply.
483  *
484  * @param s RTSP (de)muxer context
485  * @param method the method for the request
486  * @param url the target url for the request
487  * @param headers extra header lines to include in the request
488  * @param reply pointer where the RTSP message header will be stored
489  * @param content_ptr pointer where the RTSP message body, if any, will
490  * be stored (length is in reply)
491  * @param send_content if non-null, the data to send as request body content
492  * @param send_content_length the length of the send_content data, or 0 if
493  * send_content is null
494  *
495  * @return zero if success, nonzero otherwise
496  */
498  const char *method, const char *url,
499  const char *headers,
500  RTSPMessageHeader *reply,
501  unsigned char **content_ptr,
502  const unsigned char *send_content,
503  int send_content_length);
504 
505 /**
506  * Send a command to the RTSP server and wait for the reply.
507  *
508  * @see rtsp_send_cmd_with_content
509  */
510 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
511  const char *url, const char *headers,
512  RTSPMessageHeader *reply, unsigned char **content_ptr);
513 
514 /**
515  * Read a RTSP message from the server, or prepare to read data
516  * packets if we're reading data interleaved over the TCP/RTSP
517  * connection as well.
518  *
519  * @param s RTSP (de)muxer context
520  * @param reply pointer where the RTSP message header will be stored
521  * @param content_ptr pointer where the RTSP message body, if any, will
522  * be stored (length is in reply)
523  * @param return_on_interleaved_data whether the function may return if we
524  * encounter a data marker ('$'), which precedes data
525  * packets over interleaved TCP/RTSP connections. If this
526  * is set, this function will return 1 after encountering
527  * a '$'. If it is not set, the function will skip any
528  * data packets (if they are encountered), until a reply
529  * has been fully parsed. If no more data is available
530  * without parsing a reply, it will return an error.
531  * @param method the RTSP method this is a reply to. This affects how
532  * some response headers are acted upon. May be NULL.
533  *
534  * @return 1 if a data packets is ready to be received, -1 on error,
535  * and 0 on success.
536  */
538  unsigned char **content_ptr,
539  int return_on_interleaved_data, const char *method);
540 
541 /**
542  * Skip a RTP/TCP interleaved packet.
543  */
545 
546 /**
547  * Connect to the RTSP server and set up the individual media streams.
548  * This can be used for both muxers and demuxers.
549  *
550  * @param s RTSP (de)muxer context
551  *
552  * @return 0 on success, < 0 on error. Cleans up all allocations done
553  * within the function on error.
554  */
556 
557 /**
558  * Close and free all streams within the RTSP (de)muxer
559  *
560  * @param s RTSP (de)muxer context
561  */
563 
564 /**
565  * Close all connection handles within the RTSP (de)muxer
566  *
567  * @param s RTSP (de)muxer context
568  */
570 
571 /**
572  * Get the description of the stream and set up the RTSPStream child
573  * objects.
574  */
576 
577 /**
578  * Announce the stream to the server and set up the RTSPStream child
579  * objects for each media stream.
580  */
582 
583 /**
584  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
585  * listen mode.
586  */
588 
589 /**
590  * Parse an SDP description of streams by populating an RTSPState struct
591  * within the AVFormatContext; also allocate the RTP streams and the
592  * pollfd array used for UDP streams.
593  */
594 int ff_sdp_parse(AVFormatContext *s, const char *content);
595 
596 /**
597  * Receive one RTP packet from an TCP interleaved RTSP stream.
598  */
600  uint8_t *buf, int buf_size);
601 
602 /**
603  * Send buffered packets over TCP.
604  */
606 
607 /**
608  * Receive one packet from the RTSPStreams set up in the AVFormatContext
609  * (which should contain a RTSPState struct as priv_data).
610  */
612 
613 /**
614  * Do the SETUP requests for each stream for the chosen
615  * lower transport mode.
616  * @return 0 on success, <0 on error, 1 if protocol is unavailable
617  */
618 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
619  int lower_transport, const char *real_challenge);
620 
621 /**
622  * Undo the effect of ff_rtsp_make_setup_request, close the
623  * transport_priv and rtp_handle fields.
624  */
625 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
626 
627 /**
628  * Open RTSP transport context.
629  */
631 
632 extern const AVOption ff_rtsp_options[];
633 
634 #endif /* AVFORMAT_RTSP_H */