FFmpeg
rtsp.h
Go to the documentation of this file.
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 #include "internal.h"
31 
32 #include "libavutil/log.h"
33 #include "libavutil/opt.h"
34 
35 /**
36  * Network layer over which RTP/etc packet data will be transported.
37  */
39  RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
40  RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
41  RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
43  RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
44  transport mode as such,
45  only for use via AVOptions */
46  RTSP_LOWER_TRANSPORT_HTTPS, /**< HTTPS tunneled */
47  RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
48  option for lower_transport_mask,
49  but set in the SDP demuxer based
50  on a flag. */
51 };
52 
53 /**
54  * Packet profile of the data that we will be receiving. Real servers
55  * commonly send RDT (although they can sometimes send RTP as well),
56  * whereas most others will send RTP.
57  */
59  RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
60  RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
61  RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
63 };
64 
65 /**
66  * Transport mode for the RTSP data. This may be plain, or
67  * tunneled, which is done over HTTP.
68  */
70  RTSP_MODE_PLAIN, /**< Normal RTSP */
71  RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
72 };
73 
74 #define RTSP_DEFAULT_PORT 554
75 #define RTSPS_DEFAULT_PORT 322
76 #define RTSP_MAX_TRANSPORTS 8
77 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
78 #define RTSP_RTP_PORT_MIN 5000
79 #define RTSP_RTP_PORT_MAX 65000
80 #define SDP_MAX_SIZE 16384
81 
82 /**
83  * This describes a single item in the "Transport:" line of one stream as
84  * negotiated by the SETUP RTSP command. Multiple transports are comma-
85  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
86  * client_port=1000-1001;server_port=1800-1801") and described in separate
87  * RTSPTransportFields.
88  */
89 typedef struct RTSPTransportField {
90  /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
91  * with a '$', stream length and stream ID. If the stream ID is within
92  * the range of this interleaved_min-max, then the packet belongs to
93  * this stream. */
95 
96  /** UDP multicast port range; the ports to which we should connect to
97  * receive multicast UDP data. */
99 
100  /** UDP client ports; these should be the local ports of the UDP RTP
101  * (and RTCP) sockets over which we receive RTP/RTCP data. */
103 
104  /** UDP unicast server port range; the ports to which we should connect
105  * to receive unicast UDP RTP/RTCP data. */
107 
108  /** time-to-live value (required for multicast); the amount of HOPs that
109  * packets will be allowed to make before being discarded. */
110  int ttl;
111 
112  /** transport set to record data */
114 
115  struct sockaddr_storage destination; /**< destination IP address */
116  char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
117 
118  /** data/packet transport protocol; e.g. RTP or RDT */
120 
121  /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
124 
125 /**
126  * This describes the server response to each RTSP command.
127  */
128 typedef struct RTSPMessageHeader {
129  /** length of the data following this header */
131 
132  enum RTSPStatusCode status_code; /**< response code from server */
133 
134  /** number of items in the 'transports' variable below */
136 
137  /** Time range of the streams that the server will stream. In
138  * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
140 
141  /** describes the complete "Transport:" line of the server in response
142  * to a SETUP RTSP command by the client */
144 
145  int seq; /**< sequence number */
146 
147  /** the "Session:" field. This value is initially set by the server and
148  * should be re-transmitted by the client in every RTSP command. */
149  char session_id[512];
150 
151  /** the "Location:" field. This value is used to handle redirection.
152  */
153  char location[4096];
154 
155  /** the "RealChallenge1:" field from the server */
156  char real_challenge[64];
157 
158  /** the "Server: field, which can be used to identify some special-case
159  * servers that are not 100% standards-compliant. We use this to identify
160  * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
161  * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
162  * use something like "Helix [..] Server Version v.e.r.sion (platform)
163  * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
164  * where platform is the output of $uname -msr | sed 's/ /-/g'. */
165  char server[64];
166 
167  /** The "timeout" comes as part of the server response to the "SETUP"
168  * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
169  * time, in seconds, that the server will go without traffic over the
170  * RTSP/TCP connection before it closes the connection. To prevent
171  * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
172  * than this value. */
173  int timeout;
174 
175  /** The "Notice" or "X-Notice" field value. See
176  * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
177  * for a complete list of supported values. */
178  int notice;
179 
180  /** The "reason" is meant to specify better the meaning of the error code
181  * returned
182  */
183  char reason[256];
184 
185  /**
186  * Content type header
187  */
188  char content_type[64];
189 
190  /**
191  * SAT>IP com.ses.streamID header
192  */
193  char stream_id[64];
195 
196 /**
197  * Client state, i.e. whether we are currently receiving data (PLAYING) or
198  * setup-but-not-receiving (PAUSED). State can be changed in applications
199  * by calling av_read_play/pause().
200  */
202  RTSP_STATE_IDLE, /**< not initialized */
203  RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
204  RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
205  RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
206 };
207 
208 /**
209  * Identify particular servers that require special handling, such as
210  * standards-incompliant "Transport:" lines in the SETUP request.
211  */
213  RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
214  RTSP_SERVER_REAL, /**< Realmedia-style server */
215  RTSP_SERVER_WMS, /**< Windows Media server */
216  RTSP_SERVER_SATIP,/**< SAT>IP server */
218 };
219 
220 /**
221  * Private data for the RTSP demuxer.
222  *
223  * @todo Use AVIOContext instead of URLContext
224  */
225 typedef struct RTSPState {
226  const AVClass *class; /**< Class for private options. */
227  URLContext *rtsp_hd; /* RTSP TCP connection handle */
228 
229  /** number of items in the 'rtsp_streams' variable */
231 
232  struct RTSPStream **rtsp_streams; /**< streams in this session */
233 
234  /** indicator of whether we are currently receiving data from the
235  * server. Basically this isn't more than a simple cache of the
236  * last PLAY/PAUSE command sent to the server, to make sure we don't
237  * send 2x the same unexpectedly or commands in the wrong state. */
239 
240  /** the seek value requested when calling av_seek_frame(). This value
241  * is subsequently used as part of the "Range" parameter when emitting
242  * the RTSP PLAY command. If we are currently playing, this command is
243  * called instantly. If we are currently paused, this command is called
244  * whenever we resume playback. Either way, the value is only used once,
245  * see rtsp_read_play() and rtsp_read_seek(). */
246  int64_t seek_timestamp;
247 
248  int seq; /**< RTSP command sequence number */
249 
250  /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
251  * identifier that the client should re-transmit in each RTSP command */
252  char session_id[512];
253 
254  /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
255  * the server will go without traffic on the RTSP/TCP line before it
256  * closes the connection. */
257  int timeout;
258 
259  /** timestamp of the last RTSP command that we sent to the RTSP server.
260  * This is used to calculate when to send dummy commands to keep the
261  * connection alive, in conjunction with timeout. */
262  int64_t last_cmd_time;
263 
264  /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
266 
267  /** the negotiated network layer transport protocol; e.g. TCP or UDP
268  * uni-/multicast */
270 
271  /** brand of server that we're talking to; e.g. WMS, REAL or other.
272  * Detected based on the value of RTSPMessageHeader->server or the presence
273  * of RTSPMessageHeader->real_challenge */
275 
276  /** the "RealChallenge1:" field from the server */
277  char real_challenge[64];
278 
279  /** plaintext authorization line (username:password) */
280  char auth[128];
281 
282  /** authentication state */
284 
285  /** The last reply of the server to a RTSP command */
286  char last_reply[2048]; /* XXX: allocate ? */
287 
288  /** RTSPStream->transport_priv of the last stream that we read a
289  * packet from */
291 
292  /** The following are used for Real stream selection */
293  //@{
294  /** whether we need to send a "SET_PARAMETER Subscribe:" command */
296 
297  /** stream setup during the last frame read. This is used to detect if
298  * we need to subscribe or unsubscribe to any new streams. */
300 
301  /** current stream setup. This is a temporary buffer used to compare
302  * current setup to previous frame setup. */
304 
305  /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
306  * this is used to send the same "Unsubscribe:" if stream setup changed,
307  * before sending a new "Subscribe:" command. */
308  char last_subscription[1024];
309  //@}
310 
311  /** The following are used for RTP/ASF streams */
312  //@{
313  /** ASF demuxer context for the embedded ASF stream from WMS servers */
315 
316  /** cache for position of the asf demuxer, since we load a new
317  * data packet in the bytecontext for each incoming RTSP packet. */
318  uint64_t asf_pb_pos;
319  //@}
320 
321  /** some MS RTSP streams contain a URL in the SDP that we need to use
322  * for all subsequent RTSP requests, rather than the input URI; in
323  * other cases, this is a copy of AVFormatContext->filename. */
325 
326  /** The following are used for parsing raw mpegts in udp */
327  //@{
328  struct MpegTSContext *ts;
331  //@}
332 
333  /** Additional output handle, used when input and output are done
334  * separately, eg for HTTP tunneling. */
336 
337  /** RTSP transport mode, such as plain or tunneled. */
339 
340  /* Number of RTCP BYE packets the RTSP session has received.
341  * An EOF is propagated back if nb_byes == nb_streams.
342  * This is reset after a seek. */
343  int nb_byes;
344 
345  /** Reusable buffer for receiving packets */
346  uint8_t* recvbuf;
347 
348  /**
349  * A mask with all requested transport methods
350  */
352 
353  /**
354  * The number of returned packets
355  */
356  uint64_t packets;
357 
358  /**
359  * Polling array for udp
360  */
361  struct pollfd *p;
362  int max_p;
363 
364  /**
365  * Whether the server supports the GET_PARAMETER method.
366  */
368 
369  /**
370  * Do not begin to play the stream immediately.
371  */
373 
374  /**
375  * Option flags for the chained RTP muxer.
376  */
378 
379  /** Whether the server accepts the x-Dynamic-Rate header */
381 
382  /**
383  * Various option flags for the RTSP muxer/demuxer.
384  */
386 
387  /**
388  * Mask of all requested media types
389  */
391 
392  /**
393  * Minimum and maximum local UDP ports.
394  */
396 
397  /**
398  * Timeout to wait for incoming connections.
399  */
401 
402  /**
403  * timeout of socket i/o operations.
404  */
405  int64_t stimeout;
406 
407  /**
408  * Size of RTP packet reordering queue.
409  */
411 
412  /**
413  * User-Agent string
414  */
415  char *user_agent;
416 
417  char default_lang[4];
419  int pkt_size;
420  char *localaddr;
421 } RTSPState;
422 
423 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
424  receive packets only from the right
425  source address and port. */
426 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
427 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
428 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
429  address of received packets. */
430 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
431 #define RTSP_FLAG_SATIP_RAW 0x20 /**< Export SAT>IP stream as raw MPEG-TS */
432 
433 typedef struct RTSPSource {
434  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
435 } RTSPSource;
436 
437 /**
438  * Describe a single stream, as identified by a single m= line block in the
439  * SDP content. In the case of RDT, one RTSPStream can represent multiple
440  * AVStreams. In this case, each AVStream in this set has similar content
441  * (but different codec/bitrate).
442  */
443 typedef struct RTSPStream {
444  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
445  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
446 
447  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
449 
450  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
451  * for the selected transport. Only used for TCP. */
453 
454  char control_url[MAX_URL_SIZE]; /**< url for this stream (from SDP) */
455 
456  /** The following are used only in SDP, not RTSP */
457  //@{
458  int sdp_port; /**< port (from SDP content) */
459  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
460  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
461  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
462  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
463  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
464  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
465  int sdp_payload_type; /**< payload type */
466  //@}
467 
468  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
469  //@{
470  /** handler structure */
472 
473  /** private data associated with the dynamic protocol */
475  //@}
476 
477  /** Enable sending RTCP feedback messages according to RFC 4585 */
478  int feedback;
479 
480  /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
481  uint32_t ssrc;
482 
483  char crypto_suite[40];
484  char crypto_params[100];
485 } RTSPStream;
486 
488  RTSPMessageHeader *reply, const char *buf,
489  RTSPState *rt, const char *method);
490 
491 /**
492  * Send a command to the RTSP server without waiting for the reply.
493  *
494  * @see rtsp_send_cmd_with_content_async
495  */
496 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
497  const char *url, const char *headers);
498 
499 /**
500  * Send a command to the RTSP server and wait for the reply.
501  *
502  * @param s RTSP (de)muxer context
503  * @param method the method for the request
504  * @param url the target url for the request
505  * @param headers extra header lines to include in the request
506  * @param reply pointer where the RTSP message header will be stored
507  * @param content_ptr pointer where the RTSP message body, if any, will
508  * be stored (length is in reply)
509  * @param send_content if non-null, the data to send as request body content
510  * @param send_content_length the length of the send_content data, or 0 if
511  * send_content is null
512  *
513  * @return zero if success, nonzero otherwise
514  */
516  const char *method, const char *url,
517  const char *headers,
518  RTSPMessageHeader *reply,
519  unsigned char **content_ptr,
520  const unsigned char *send_content,
521  int send_content_length);
522 
523 /**
524  * Send a command to the RTSP server and wait for the reply.
525  *
526  * @see rtsp_send_cmd_with_content
527  */
528 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
529  const char *url, const char *headers,
530  RTSPMessageHeader *reply, unsigned char **content_ptr);
531 
532 /**
533  * Read a RTSP message from the server, or prepare to read data
534  * packets if we're reading data interleaved over the TCP/RTSP
535  * connection as well.
536  *
537  * @param s RTSP (de)muxer context
538  * @param reply pointer where the RTSP message header will be stored
539  * @param content_ptr pointer where the RTSP message body, if any, will
540  * be stored (length is in reply)
541  * @param return_on_interleaved_data whether the function may return if we
542  * encounter a data marker ('$'), which precedes data
543  * packets over interleaved TCP/RTSP connections. If this
544  * is set, this function will return 1 after encountering
545  * a '$'. If it is not set, the function will skip any
546  * data packets (if they are encountered), until a reply
547  * has been fully parsed. If no more data is available
548  * without parsing a reply, it will return an error.
549  * @param method the RTSP method this is a reply to. This affects how
550  * some response headers are acted upon. May be NULL.
551  *
552  * @return 1 if a data packets is ready to be received, -1 on error,
553  * and 0 on success.
554  */
556  unsigned char **content_ptr,
557  int return_on_interleaved_data, const char *method);
558 
559 /**
560  * Skip a RTP/TCP interleaved packet.
561  *
562  * @return 0 on success, < 0 on failure.
563  */
565 
566 /**
567  * Connect to the RTSP server and set up the individual media streams.
568  * This can be used for both muxers and demuxers.
569  *
570  * @param s RTSP (de)muxer context
571  *
572  * @return 0 on success, < 0 on error. Cleans up all allocations done
573  * within the function on error.
574  */
576 
577 /**
578  * Close and free all streams within the RTSP (de)muxer
579  *
580  * @param s RTSP (de)muxer context
581  */
583 
584 /**
585  * Close all connection handles within the RTSP (de)muxer
586  *
587  * @param s RTSP (de)muxer context
588  */
590 
591 /**
592  * Get the description of the stream and set up the RTSPStream child
593  * objects.
594  */
596 
597 /**
598  * Announce the stream to the server and set up the RTSPStream child
599  * objects for each media stream.
600  */
602 
603 /**
604  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
605  * listen mode.
606  */
608 
609 /**
610  * Parse an SDP description of streams by populating an RTSPState struct
611  * within the AVFormatContext; also allocate the RTP streams and the
612  * pollfd array used for UDP streams.
613  */
614 int ff_sdp_parse(AVFormatContext *s, const char *content);
615 
616 /**
617  * Receive one RTP packet from an TCP interleaved RTSP stream.
618  */
620  uint8_t *buf, int buf_size);
621 
622 /**
623  * Send buffered packets over TCP.
624  */
626 
627 /**
628  * Receive one packet from the RTSPStreams set up in the AVFormatContext
629  * (which should contain a RTSPState struct as priv_data).
630  */
632 
633 /**
634  * Do the SETUP requests for each stream for the chosen
635  * lower transport mode.
636  * @return 0 on success, <0 on error, 1 if protocol is unavailable
637  */
638 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
639  int lower_transport, const char *real_challenge);
640 
641 /**
642  * Undo the effect of ff_rtsp_make_setup_request, close the
643  * transport_priv and rtp_handle fields.
644  */
645 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
646 
647 /**
648  * Open RTSP transport context.
649  */
651 
652 extern const AVOption ff_rtsp_options[];
653 
654 #endif /* AVFORMAT_RTSP_H */
RTSPState::initial_timeout
int initial_timeout
Timeout to wait for incoming connections.
Definition: rtsp.h:400
RTSP_STATE_PAUSED
@ RTSP_STATE_PAUSED
initialized, but not receiving data
Definition: rtsp.h:204
RTSPState::initial_pause
int initial_pause
Do not begin to play the stream immediately.
Definition: rtsp.h:372
ff_rtsp_read_reply
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
RTSPState::last_cmd_time
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:262
opt.h
RTSPStream::transport_priv
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:445
ff_rtsp_send_cmd_with_content
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
RTSP_TRANSPORT_NB
@ RTSP_TRANSPORT_NB
Definition: rtsp.h:62
RTSPStream::rtp_handle
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:444
RTSPTransportField::port_max
int port_max
Definition: rtsp.h:98
RTSP_SERVER_RTP
@ RTSP_SERVER_RTP
Standards-compliant RTP-server.
Definition: rtsp.h:213
RTSP_STATE_SEEKING
@ RTSP_STATE_SEEKING
initialized, requesting a seek
Definition: rtsp.h:205
RTSPMessageHeader::status_code
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:132
ff_rtsp_send_cmd
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
RTSPState::control_transport
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:338
RTSP_MODE_PLAIN
@ RTSP_MODE_PLAIN
Normal RTSP.
Definition: rtsp.h:70
RTSP_TRANSPORT_RTP
@ RTSP_TRANSPORT_RTP
Standards-compliant RTP.
Definition: rtsp.h:59
RTSPTransportField::source
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:116
RTSPMessageHeader::range_end
int64_t range_end
Definition: rtsp.h:139
RTSPState::get_parameter_supported
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:367
RTSPStream::nb_include_source_addrs
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:460
RTSPTransportField::server_port_min
int server_port_min
UDP unicast server port range; the ports to which we should connect to receive unicast UDP RTP/RTCP d...
Definition: rtsp.h:106
RTSPState::auth
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:280
RTSPStream::interleaved_min
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport.
Definition: rtsp.h:452
RTSPState::recvbuf_pos
int recvbuf_pos
Definition: rtsp.h:329
AVOption
AVOption.
Definition: opt.h:251
RTSPTransportField::lower_transport
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:122
RTSPState::rtp_port_min
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:395
RTSP_LOWER_TRANSPORT_CUSTOM
@ RTSP_LOWER_TRANSPORT_CUSTOM
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag.
Definition: rtsp.h:47
RTSPTransportField::interleaved_min
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:94
RTSPTransportField::interleaved_max
int interleaved_max
Definition: rtsp.h:94
RTSPStream
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:443
ff_rtsp_undo_setup
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields.
Definition: rtsp.c:758
ff_rtsp_close_streams
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:790
ff_rtsp_send_cmd_async
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
RTSPState::real_challenge
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:277
RTSPMessageHeader::nb_transports
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:135
RTSP_SERVER_REAL
@ RTSP_SERVER_REAL
Realmedia-style server.
Definition: rtsp.h:214
RTSPState::seek_timestamp
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:246
sockaddr_storage
Definition: network.h:111
ff_sdp_parse
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
RTSPState::pkt_size
int pkt_size
Definition: rtsp.h:419
RTSPStream::feedback
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:478
RTSPState::asf_ctx
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:314
RTSPState::nb_rtsp_streams
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:230
RTSPMessageHeader::content_length
int content_length
length of the data following this header
Definition: rtsp.h:130
RTSP_TRANSPORT_RDT
@ RTSP_TRANSPORT_RDT
Realmedia Data Transport.
Definition: rtsp.h:60
RTSP_STATE_STREAMING
@ RTSP_STATE_STREAMING
initialized and sending/receiving data
Definition: rtsp.h:203
ff_rtsp_setup_input_streams
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
Get the description of the stream and set up the RTSPStream child objects.
Definition: rtspdec.c:612
RTSPState::lower_transport_mask
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:351
RTSP_MODE_TUNNEL
@ RTSP_MODE_TUNNEL
RTSP over HTTP (tunneling)
Definition: rtsp.h:71
rtspcodes.h
RTSPStream::stream_index
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:448
RTSPTransportField::destination
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:115
RTSP_LOWER_TRANSPORT_HTTPS
@ RTSP_LOWER_TRANSPORT_HTTPS
HTTPS tunneled.
Definition: rtsp.h:46
RTSPState::rtsp_hd_out
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling.
Definition: rtsp.h:335
pkt
AVPacket * pkt
Definition: movenc.c:59
RTSPControlTransport
RTSPControlTransport
Transport mode for the RTSP data.
Definition: rtsp.h:69
RTSPState::reordering_queue_size
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:410
RTSPState::ts
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:328
ff_rtsp_open_transport_ctx
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:826
s
#define s(width, name)
Definition: cbs_vp9.c:256
RTSPState::nb_byes
int nb_byes
Definition: rtsp.h:343
RTSPState::p
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:361
RTSPMessageHeader::location
char location[4096]
the "Location:" field.
Definition: rtsp.h:153
RTSPState::control_uri
char control_uri[MAX_URL_SIZE]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests,...
Definition: rtsp.h:324
RTSPMessageHeader::transports
RTSPTransportField transports[RTSP_MAX_TRANSPORTS]
describes the complete "Transport:" line of the server in response to a SETUP RTSP command by the cli...
Definition: rtsp.h:143
RTSPMessageHeader::stream_id
char stream_id[64]
SAT>IP com.ses.streamID header.
Definition: rtsp.h:193
RTSPState::buffer_size
int buffer_size
Definition: rtsp.h:418
RTSPTransportField::ttl
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:110
ff_rtsp_fetch_packet
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
Receive one packet from the RTSPStreams set up in the AVFormatContext (which should contain a RTSPSta...
RTSPStream::dynamic_handler
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:471
RTSPMessageHeader::seq
int seq
sequence number
Definition: rtsp.h:145
RTSPState::rtp_muxer_flags
int rtp_muxer_flags
Option flags for the chained RTP muxer.
Definition: rtsp.h:377
ff_rtsp_setup_output_streams
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream.
Definition: rtspenc.c:46
AVFormatContext
Format I/O context.
Definition: avformat.h:1216
internal.h
RTSPState::session_id
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:252
RTSPMessageHeader::reason
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:183
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
RTSPState::asf_pb_pos
uint64_t asf_pb_pos
cache for position of the asf demuxer, since we load a new data packet in the bytecontext for each in...
Definition: rtsp.h:318
RTSPState::rtsp_hd
URLContext * rtsp_hd
Definition: rtsp.h:227
RTSPServerType
RTSPServerType
Identify particular servers that require special handling, such as standards-incompliant "Transport:"...
Definition: rtsp.h:212
RTSPState::default_lang
char default_lang[4]
Definition: rtsp.h:417
RTSPMessageHeader::real_challenge
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:156
RTSPState::real_setup
enum AVDiscard * real_setup
current stream setup.
Definition: rtsp.h:303
RTSP_MAX_TRANSPORTS
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:76
RTSPState::state
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:238
RTSPState::recvbuf
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:346
RTSPStream::sdp_port
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:458
RTSPStream::exclude_source_addrs
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:463
rtpdec.h
ff_rtsp_options
const AVOption ff_rtsp_options[]
Definition: rtsp.c:84
HTTPAuthState
HTTP Authentication state structure.
Definition: httpauth.h:55
RTSPSource
Definition: rtsp.h:433
RTSPStream::dynamic_protocol_context
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:474
ff_rtsp_tcp_read_packet
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size)
Receive one RTP packet from an TCP interleaved RTSP stream.
Definition: rtspdec.c:783
RTSPState::rtsp_flags
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:385
ff_rtsp_close_connections
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
RTSPState
Private data for the RTSP demuxer.
Definition: rtsp.h:225
RTSPStream::include_source_addrs
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:461
RTSPState::lower_transport
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:269
RTSPMessageHeader::range_start
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:139
RTSPState::recvbuf_len
int recvbuf_len
Definition: rtsp.h:330
RTSPState::last_reply
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:286
MpegTSContext
Definition: mpegts.c:130
RTSPTransportField::transport
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:119
RTSPState::rtsp_streams
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:232
ff_rtsp_skip_packet
int ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
RTSPState::seq
int seq
RTSP command sequence number.
Definition: rtsp.h:248
RTSPStream::crypto_params
char crypto_params[100]
Definition: rtsp.h:484
RTSPState::max_p
int max_p
Definition: rtsp.h:362
RTSPState::auth_state
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:283
ff_rtsp_parse_line
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
RTSPState::last_subscription
char last_subscription[1024]
the last value of the "SET_PARAMETER Subscribe:" RTSP command.
Definition: rtsp.h:308
RTSPStream::nb_exclude_source_addrs
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:462
RTSPState::timeout
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:257
RTSPTransportField::client_port_max
int client_port_max
Definition: rtsp.h:102
RTSP_SERVER_SATIP
@ RTSP_SERVER_SATIP
SAT>IP server.
Definition: rtsp.h:216
httpauth.h
RTSPState::media_type_mask
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:390
URLContext
Definition: url.h:37
log.h
RTSPSource::addr
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:434
RTSPMessageHeader::timeout
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:173
RTSPState::need_subscription
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:295
INET6_ADDRSTRLEN
#define INET6_ADDRSTRLEN
Definition: network.h:237
ff_rtsp_tcp_write_packet
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
ff_rtsp_parse_streaming_commands
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:483
RTSPStream::ssrc
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:481
RTSP_LOWER_TRANSPORT_TCP
@ RTSP_LOWER_TRANSPORT_TCP
TCP; interleaved in RTSP.
Definition: rtsp.h:40
RTSPStream::sdp_ttl
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:464
RTSPTransportField::client_port_min
int client_port_min
UDP client ports; these should be the local ports of the UDP RTP (and RTCP) sockets over which we rec...
Definition: rtsp.h:102
RTSPLowerTransport
RTSPLowerTransport
Network layer over which RTP/etc packet data will be transported.
Definition: rtsp.h:38
RTSPState::rtp_port_max
int rtp_port_max
Definition: rtsp.h:395
RTSP_LOWER_TRANSPORT_UDP_MULTICAST
@ RTSP_LOWER_TRANSPORT_UDP_MULTICAST
UDP/multicast.
Definition: rtsp.h:41
RTSPState::cur_transport_priv
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:290
RTSPStream::sdp_payload_type
int sdp_payload_type
payload type
Definition: rtsp.h:465
avformat.h
network.h
RTSPStream::sdp_ip
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:459
RTSP_LOWER_TRANSPORT_HTTP
@ RTSP_LOWER_TRANSPORT_HTTP
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:43
RTSPTransportField
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:89
RTSPState::transport
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:265
MAX_URL_SIZE
#define MAX_URL_SIZE
Definition: internal.h:31
RTSPStream::control_url
char control_url[MAX_URL_SIZE]
url for this stream (from SDP)
Definition: rtsp.h:454
RTSPStream::interleaved_max
int interleaved_max
Definition: rtsp.h:452
RTSPState::localaddr
char * localaddr
Definition: rtsp.h:420
RTSPStatusCode
RTSPStatusCode
RTSP handling.
Definition: rtspcodes.h:31
headers
FFmpeg currently uses a custom build this text attempts to document some of its obscure features and options Makefile the full command issued by make and its output will be shown on the screen DBG Preprocess x86 external assembler files to a dbg asm file in the object which then gets compiled Helps in developing those assembler files DESTDIR Destination directory for the install useful to prepare packages or install FFmpeg in cross environments GEN Set to ‘1’ to generate the missing or mismatched references Makefile builds all the libraries and the executables fate Run the fate test note that you must have installed it fate list List all fate regression test targets install Install headers
Definition: build_system.txt:34
RTSP_SERVER_WMS
@ RTSP_SERVER_WMS
Windows Media server.
Definition: rtsp.h:215
RTSPMessageHeader
This describes the server response to each RTSP command.
Definition: rtsp.h:128
RTSP_TRANSPORT_RAW
@ RTSP_TRANSPORT_RAW
Raw data (over UDP)
Definition: rtsp.h:61
RTSPState::stimeout
int64_t stimeout
timeout of socket i/o operations.
Definition: rtsp.h:405
RTSPState::real_setup_cache
enum AVDiscard * real_setup_cache
stream setup during the last frame read.
Definition: rtsp.h:299
RTSP_STATE_IDLE
@ RTSP_STATE_IDLE
not initialized
Definition: rtsp.h:202
RTSPClientState
RTSPClientState
Client state, i.e.
Definition: rtsp.h:201
RTSP_LOWER_TRANSPORT_NB
@ RTSP_LOWER_TRANSPORT_NB
Definition: rtsp.h:42
AVPacket
This structure stores compressed data.
Definition: packet.h:351
RTSPState::server_type
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:274
RTSPMessageHeader::content_type
char content_type[64]
Content type header.
Definition: rtsp.h:188
RTSPTransportField::server_port_max
int server_port_max
Definition: rtsp.h:106
RTSPTransport
RTSPTransport
Packet profile of the data that we will be receiving.
Definition: rtsp.h:58
RTSPState::packets
uint64_t packets
The number of returned packets.
Definition: rtsp.h:356
RTSPTransportField::mode_record
int mode_record
transport set to record data
Definition: rtsp.h:113
RTSPState::accept_dynamic_rate
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:380
RTSPMessageHeader::session_id
char session_id[512]
the "Session:" field.
Definition: rtsp.h:149
RTSP_SERVER_NB
@ RTSP_SERVER_NB
Definition: rtsp.h:217
ff_rtsp_make_setup_request
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge)
Do the SETUP requests for each stream for the chosen lower transport mode.
RTSPMessageHeader::server
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:165
RTSPStream::crypto_suite
char crypto_suite[40]
Definition: rtsp.h:483
PayloadContext
RTP/JPEG specific private data.
Definition: rdt.c:83
AVDiscard
AVDiscard
Definition: defs.h:67
ff_rtsp_connect
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
RTSPTransportField::port_min
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data.
Definition: rtsp.h:98
RTSPMessageHeader::notice
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:178
RTSP_LOWER_TRANSPORT_UDP
@ RTSP_LOWER_TRANSPORT_UDP
UDP/unicast.
Definition: rtsp.h:39
RTPDynamicProtocolHandler
Definition: rtpdec.h:116
RTSPState::user_agent
char * user_agent
User-Agent string.
Definition: rtsp.h:415