74733361f920b279bd21f28fbc0f1a0baec2708a
[libav.git] / libavformat / rtsp.h
1 /*
2 * RTSP definitions
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav 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 * Libav 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 Libav; 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 */
37 enum RTSPLowerTransport {
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 */
41 RTSP_LOWER_TRANSPORT_NB,
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 */
56 enum RTSPTransport {
57 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
58 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
59 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
60 RTSP_TRANSPORT_NB
61 };
62
63 /**
64 * Transport mode for the RTSP data. This may be plain, or
65 * tunneled, which is done over HTTP.
66 */
67 enum RTSPControlTransport {
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 10000
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. */
93 int interleaved_min, interleaved_max;
94
95 /** UDP multicast port range; the ports to which we should connect to
96 * receive multicast UDP data. */
97 int port_min, port_max;
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. */
101 int client_port_min, client_port_max;
102
103 /** UDP unicast server port range; the ports to which we should connect
104 * to receive unicast UDP RTP/RTCP data. */
105 int server_port_min, server_port_max;
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 */
112 int mode_record;
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 */
118 enum RTSPTransport transport;
119
120 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
121 enum RTSPLowerTransport lower_transport;
122 } RTSPTransportField;
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 */
129 int content_length;
130
131 enum RTSPStatusCode status_code; /**< response code from server */
132
133 /** number of items in the 'transports' variable below */
134 int nb_transports;
135
136 /** Time range of the streams that the server will stream. In
137 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
138 int64_t range_start, range_end;
139
140 /** describes the complete "Transport:" line of the server in response
141 * to a SETUP RTSP command by the client */
142 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
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];
188 } RTSPMessageHeader;
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 */
195 enum RTSPClientState {
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 */
206 enum RTSPServerType {
207 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
208 RTSP_SERVER_REAL, /**< Realmedia-style server */
209 RTSP_SERVER_WMS, /**< Windows Media server */
210 RTSP_SERVER_NB
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 */
223 int nb_rtsp_streams;
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. */
231 enum RTSPClientState 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 */
258 enum RTSPTransport transport;
259
260 /** the negotiated network layer transport protocol; e.g. TCP or UDP
261 * uni-/multicast */
262 enum RTSPLowerTransport lower_transport;
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 */
267 enum RTSPServerType server_type;
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 */
276 HTTPAuthState auth_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 */
283 void *cur_transport_priv;
284
285 /** The following are used for Real stream selection */
286 //@{
287 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
288 int need_subscription;
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. */
292 enum AVDiscard *real_setup_cache;
293
294 /** current stream setup. This is a temporary buffer used to compare
295 * current setup to previous frame setup. */
296 enum AVDiscard *real_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 */
307 AVFormatContext *asf_ctx;
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;
322 int recvbuf_pos;
323 int recvbuf_len;
324 //@}
325
326 /** Additional output handle, used when input and output are done
327 * separately, eg for HTTP tunneling. */
328 URLContext *rtsp_hd_out;
329
330 /** RTSP transport mode, such as plain or tunneled. */
331 enum RTSPControlTransport control_transport;
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 */
339 uint8_t* recvbuf;
340
341 /**
342 * A mask with all requested transport methods
343 */
344 int lower_transport_mask;
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 */
359 int get_parameter_supported;
360
361 /**
362 * Do not begin to play the stream immediately.
363 */
364 int initial_pause;
365
366 /**
367 * Option flags for the chained RTP muxer.
368 */
369 int rtp_muxer_flags;
370
371 /** Whether the server accepts the x-Dynamic-Rate header */
372 int accept_dynamic_rate;
373
374 /**
375 * Various option flags for the RTSP muxer/demuxer.
376 */
377 int rtsp_flags;
378
379 /**
380 * Mask of all requested media types
381 */
382 int media_type_mask;
383
384 /**
385 * Minimum and maximum local UDP ports.
386 */
387 int rtp_port_min, rtp_port_max;
388
389 /**
390 * Timeout to wait for incoming connections.
391 */
392 int initial_timeout;
393
394 /**
395 * Size of RTP packet reordering queue.
396 */
397 int reordering_queue_size;
398
399 char default_lang[4];
400 int buffer_size;
401 } RTSPState;
402
403 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
404 receive packets only from the right
405 source address and port. */
406 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
407 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
408 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
409 address of received packets. */
410
411 typedef struct RTSPSource {
412 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
413 } RTSPSource;
414
415 /**
416 * Describe a single stream, as identified by a single m= line block in the
417 * SDP content. In the case of RDT, one RTSPStream can represent multiple
418 * AVStreams. In this case, each AVStream in this set has similar content
419 * (but different codec/bitrate).
420 */
421 typedef struct RTSPStream {
422 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
423 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
424
425 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
426 int stream_index;
427
428 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
429 * for the selected transport. Only used for TCP. */
430 int interleaved_min, interleaved_max;
431
432 char control_url[1024]; /**< url for this stream (from SDP) */
433
434 /** The following are used only in SDP, not RTSP */
435 //@{
436 int sdp_port; /**< port (from SDP content) */
437 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
438 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
439 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
440 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
441 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
442 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
443 int sdp_payload_type; /**< payload type */
444 //@}
445
446 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
447 //@{
448 /** handler structure */
449 RTPDynamicProtocolHandler *dynamic_handler;
450
451 /** private data associated with the dynamic protocol */
452 PayloadContext *dynamic_protocol_context;
453 //@}
454
455 /** Enable sending RTCP feedback messages according to RFC 4585 */
456 int feedback;
457
458 char crypto_suite[40];
459 char crypto_params[100];
460 } RTSPStream;
461
462 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
463 RTSPState *rt, const char *method);
464
465 /**
466 * Send a command to the RTSP server without waiting for the reply.
467 *
468 * @see rtsp_send_cmd_with_content_async
469 */
470 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
471 const char *url, const char *headers);
472
473 /**
474 * Send a command to the RTSP server and wait for the reply.
475 *
476 * @param s RTSP (de)muxer context
477 * @param method the method for the request
478 * @param url the target url for the request
479 * @param headers extra header lines to include in the request
480 * @param reply pointer where the RTSP message header will be stored
481 * @param content_ptr pointer where the RTSP message body, if any, will
482 * be stored (length is in reply)
483 * @param send_content if non-null, the data to send as request body content
484 * @param send_content_length the length of the send_content data, or 0 if
485 * send_content is null
486 *
487 * @return zero if success, nonzero otherwise
488 */
489 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
490 const char *method, const char *url,
491 const char *headers,
492 RTSPMessageHeader *reply,
493 unsigned char **content_ptr,
494 const unsigned char *send_content,
495 int send_content_length);
496
497 /**
498 * Send a command to the RTSP server and wait for the reply.
499 *
500 * @see rtsp_send_cmd_with_content
501 */
502 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
503 const char *url, const char *headers,
504 RTSPMessageHeader *reply, unsigned char **content_ptr);
505
506 /**
507 * Read a RTSP message from the server, or prepare to read data
508 * packets if we're reading data interleaved over the TCP/RTSP
509 * connection as well.
510 *
511 * @param s RTSP (de)muxer context
512 * @param reply pointer where the RTSP message header will be stored
513 * @param content_ptr pointer where the RTSP message body, if any, will
514 * be stored (length is in reply)
515 * @param return_on_interleaved_data whether the function may return if we
516 * encounter a data marker ('$'), which precedes data
517 * packets over interleaved TCP/RTSP connections. If this
518 * is set, this function will return 1 after encountering
519 * a '$'. If it is not set, the function will skip any
520 * data packets (if they are encountered), until a reply
521 * has been fully parsed. If no more data is available
522 * without parsing a reply, it will return an error.
523 * @param method the RTSP method this is a reply to. This affects how
524 * some response headers are acted upon. May be NULL.
525 *
526 * @return 1 if a data packets is ready to be received, -1 on error,
527 * and 0 on success.
528 */
529 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
530 unsigned char **content_ptr,
531 int return_on_interleaved_data, const char *method);
532
533 /**
534 * Skip a RTP/TCP interleaved packet.
535 */
536 void ff_rtsp_skip_packet(AVFormatContext *s);
537
538 /**
539 * Connect to the RTSP server and set up the individual media streams.
540 * This can be used for both muxers and demuxers.
541 *
542 * @param s RTSP (de)muxer context
543 *
544 * @return 0 on success, < 0 on error. Cleans up all allocations done
545 * within the function on error.
546 */
547 int ff_rtsp_connect(AVFormatContext *s);
548
549 /**
550 * Close and free all streams within the RTSP (de)muxer
551 *
552 * @param s RTSP (de)muxer context
553 */
554 void ff_rtsp_close_streams(AVFormatContext *s);
555
556 /**
557 * Close all connection handles within the RTSP (de)muxer
558 *
559 * @param s RTSP (de)muxer context
560 */
561 void ff_rtsp_close_connections(AVFormatContext *s);
562
563 /**
564 * Get the description of the stream and set up the RTSPStream child
565 * objects.
566 */
567 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
568
569 /**
570 * Announce the stream to the server and set up the RTSPStream child
571 * objects for each media stream.
572 */
573 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
574
575 /**
576 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
577 * listen mode.
578 */
579 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
580
581 /**
582 * Parse an SDP description of streams by populating an RTSPState struct
583 * within the AVFormatContext; also allocate the RTP streams and the
584 * pollfd array used for UDP streams.
585 */
586 int ff_sdp_parse(AVFormatContext *s, const char *content);
587
588 /**
589 * Receive one RTP packet from an TCP interleaved RTSP stream.
590 */
591 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
592 uint8_t *buf, int buf_size);
593
594 /**
595 * Send buffered packets over TCP.
596 */
597 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
598
599 /**
600 * Receive one packet from the RTSPStreams set up in the AVFormatContext
601 * (which should contain a RTSPState struct as priv_data).
602 */
603 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
604
605 /**
606 * Do the SETUP requests for each stream for the chosen
607 * lower transport mode.
608 * @return 0 on success, <0 on error, 1 if protocol is unavailable
609 */
610 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
611 int lower_transport, const char *real_challenge);
612
613 /**
614 * Undo the effect of ff_rtsp_make_setup_request, close the
615 * transport_priv and rtp_handle fields.
616 */
617 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
618
619 /**
620 * Open RTSP transport context.
621 */
622 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
623
624 extern const AVOption ff_rtsp_options[];
625
626 #endif /* AVFORMAT_RTSP_H */