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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 
409  char default_lang[4];
411 } RTSPState;
412 
413 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
414  receive packets only from the right
415  source address and port. */
416 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
417 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
418 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
419  address of received packets. */
420 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
421 
422 typedef struct RTSPSource {
423  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
424 } RTSPSource;
426 /**
427  * Describe a single stream, as identified by a single m= line block in the
428  * SDP content. In the case of RDT, one RTSPStream can represent multiple
429  * AVStreams. In this case, each AVStream in this set has similar content
430  * (but different codec/bitrate).
431  */
432 typedef struct RTSPStream {
433  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
434  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
436  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
438 
439  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
440  * for the selected transport. Only used for TCP. */
442 
443  char control_url[1024]; /**< url for this stream (from SDP) */
445  /** The following are used only in SDP, not RTSP */
446  //@{
447  int sdp_port; /**< port (from SDP content) */
448  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
449  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
450  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
451  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
452  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
453  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
454  int sdp_payload_type; /**< payload type */
455  //@}
457  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
458  //@{
459  /** handler structure */
461 
462  /** private data associated with the dynamic protocol */
464  //@}
465 
466  /** Enable sending RTCP feedback messages according to RFC 4585 */
467  int feedback;
468 
469  char crypto_suite[40];
470  char crypto_params[100];
471 } RTSPStream;
473 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
474  RTSPState *rt, const char *method);
475 
476 /**
477  * Send a command to the RTSP server without waiting for the reply.
478  *
479  * @see rtsp_send_cmd_with_content_async
480  */
481 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
482  const char *url, const char *headers);
483 
484 /**
485  * Send a command to the RTSP server and wait for the reply.
486  *
487  * @param s RTSP (de)muxer context
488  * @param method the method for the request
489  * @param url the target url for the request
490  * @param headers extra header lines to include in the request
491  * @param reply pointer where the RTSP message header will be stored
492  * @param content_ptr pointer where the RTSP message body, if any, will
493  * be stored (length is in reply)
494  * @param send_content if non-null, the data to send as request body content
495  * @param send_content_length the length of the send_content data, or 0 if
496  * send_content is null
497  *
498  * @return zero if success, nonzero otherwise
499  */
501  const char *method, const char *url,
502  const char *headers,
503  RTSPMessageHeader *reply,
504  unsigned char **content_ptr,
505  const unsigned char *send_content,
506  int send_content_length);
507 
508 /**
509  * Send a command to the RTSP server and wait for the reply.
510  *
511  * @see rtsp_send_cmd_with_content
512  */
513 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
514  const char *url, const char *headers,
515  RTSPMessageHeader *reply, unsigned char **content_ptr);
516 
517 /**
518  * Read a RTSP message from the server, or prepare to read data
519  * packets if we're reading data interleaved over the TCP/RTSP
520  * connection as well.
521  *
522  * @param s RTSP (de)muxer context
523  * @param reply pointer where the RTSP message header will be stored
524  * @param content_ptr pointer where the RTSP message body, if any, will
525  * be stored (length is in reply)
526  * @param return_on_interleaved_data whether the function may return if we
527  * encounter a data marker ('$'), which precedes data
528  * packets over interleaved TCP/RTSP connections. If this
529  * is set, this function will return 1 after encountering
530  * a '$'. If it is not set, the function will skip any
531  * data packets (if they are encountered), until a reply
532  * has been fully parsed. If no more data is available
533  * without parsing a reply, it will return an error.
534  * @param method the RTSP method this is a reply to. This affects how
535  * some response headers are acted upon. May be NULL.
536  *
537  * @return 1 if a data packets is ready to be received, -1 on error,
538  * and 0 on success.
539  */
541  unsigned char **content_ptr,
542  int return_on_interleaved_data, const char *method);
543 
544 /**
545  * Skip a RTP/TCP interleaved packet.
546  */
548 
549 /**
550  * Connect to the RTSP server and set up the individual media streams.
551  * This can be used for both muxers and demuxers.
552  *
553  * @param s RTSP (de)muxer context
554  *
555  * @return 0 on success, < 0 on error. Cleans up all allocations done
556  * within the function on error.
557  */
559 
560 /**
561  * Close and free all streams within the RTSP (de)muxer
562  *
563  * @param s RTSP (de)muxer context
564  */
566 
567 /**
568  * Close all connection handles within the RTSP (de)muxer
569  *
570  * @param s RTSP (de)muxer context
571  */
573 
574 /**
575  * Get the description of the stream and set up the RTSPStream child
576  * objects.
577  */
579 
580 /**
581  * Announce the stream to the server and set up the RTSPStream child
582  * objects for each media stream.
583  */
585 
586 /**
587  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
588  * listen mode.
589  */
591 
592 /**
593  * Parse an SDP description of streams by populating an RTSPState struct
594  * within the AVFormatContext; also allocate the RTP streams and the
595  * pollfd array used for UDP streams.
596  */
597 int ff_sdp_parse(AVFormatContext *s, const char *content);
598 
599 /**
600  * Receive one RTP packet from an TCP interleaved RTSP stream.
601  */
603  uint8_t *buf, int buf_size);
604 
605 /**
606  * Send buffered packets over TCP.
607  */
609 
610 /**
611  * Receive one packet from the RTSPStreams set up in the AVFormatContext
612  * (which should contain a RTSPState struct as priv_data).
613  */
615 
616 /**
617  * Do the SETUP requests for each stream for the chosen
618  * lower transport mode.
619  * @return 0 on success, <0 on error, 1 if protocol is unavailable
620  */
621 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
622  int lower_transport, const char *real_challenge);
623 
624 /**
625  * Undo the effect of ff_rtsp_make_setup_request, close the
626  * transport_priv and rtp_handle fields.
627  */
628 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
629 
630 /**
631  * Open RTSP transport context.
632  */
634 
635 extern const AVOption ff_rtsp_options[];
636 
637 #endif /* AVFORMAT_RTSP_H */
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:273
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:93
char crypto_suite[40]
Definition: rtsp.h:472
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:387
const char * s
Definition: avisynth_c.h:631
Realmedia Data Transport.
Definition: rtsp.h:58
RTSPLowerTransport
Network layer over which RTP/etc packet data will be transported.
Definition: rtsp.h:37
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:785
AVOption.
Definition: opt.h:255
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:115
char content_type[64]
Content type header.
Definition: rtsp.h:187
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
Windows Media server.
Definition: rtsp.h:209
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:354
RTP/JPEG specific private data.
Definition: rdt.c:83
enum AVDiscard * real_setup
current stream setup.
Definition: rtsp.h:296
char * user_agent
User-Agent string.
Definition: rtsp.h:407
char location[4096]
the "Location:" field.
Definition: rtsp.h:152
enum AVDiscard * real_setup_cache
stream setup during the last frame read.
Definition: rtsp.h:292
int mode_record
transport set to record data
Definition: rtsp.h:112
UDP/unicast.
Definition: rtsp.h:38
int seq
sequence number
Definition: rtsp.h:144
initialized and sending/receiving data
Definition: rtsp.h:197
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:270
static AVPacket pkt
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content) ...
Definition: rtsp.h:452
RTSPClientState
Client state, i.e.
Definition: rtsp.h:195
HTTP Authentication state structure.
Definition: httpauth.h:55
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:245
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:239
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.
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:121
This describes the server response to each RTSP command.
Definition: rtsp.h:127
RTSPTransport
Packet profile of the data that we will be receiving.
Definition: rtsp.h:56
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:142
Format I/O context.
Definition: avformat.h:1272
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
Standards-compliant RTP-server.
Definition: rtsp.h:207
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:402
int recvbuf_len
Definition: rtsp.h:323
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:359
Standards-compliant RTP.
Definition: rtsp.h:57
uint8_t
char session_id[512]
the "Session:" field.
Definition: rtsp.h:148
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:74
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:109
AVOptions.
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:470
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
int initial_timeout
Timeout to wait for incoming connections.
Definition: rtsp.h:392
int rtp_muxer_flags
Option flags for the chained RTP muxer.
Definition: rtsp.h:369
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:372
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:435
Custom IO - not a public option for lower_transport_mask, but set in the SDP demuxer based on a flag...
Definition: rtsp.h:45
RTSPServerType
Identify particular servers that require special handling, such as standards-incompliant "Transport:"...
Definition: rtsp.h:206
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:464
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.
Normal RTSP.
Definition: rtsp.h:68
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:134
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:177
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:455
Private data for the RTSP demuxer.
Definition: rtsp.h:218
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:255
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:250
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:715
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:182
URLContext * rtsp_hd
Definition: rtsp.h:220
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:331
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:453
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:164
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:440
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:311
int seq
RTSP command sequence number.
Definition: rtsp.h:241
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:339
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:307
int recvbuf_pos
Definition: rtsp.h:322
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:283
int content_length
length of the data following this header
Definition: rtsp.h:129
char last_subscription[1024]
the last value of the "SET_PARAMETER Subscribe:" RTSP command.
Definition: rtsp.h:301
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:172
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:42
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:88
RTSP over HTTP (tunneling)
Definition: rtsp.h:69
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:140
char crypto_params[100]
Definition: rtsp.h:473
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:450
Raw data (over UDP)
Definition: rtsp.h:59
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:321
int sdp_payload_type
payload type
Definition: rtsp.h:457
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content) ...
Definition: rtsp.h:454
int nb_byes
Definition: rtsp.h:336
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
RTSPControlTransport
Transport mode for the RTSP data.
Definition: rtsp.h:67
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:426
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:451
int rtp_port_max
Definition: rtsp.h:387
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:382
int server_port_max
Definition: rtsp.h:105
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:258
void * buf
Definition: avisynth_c.h:553
Definition: url.h:39
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
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:377
int client_port_max
Definition: rtsp.h:101
Describe the class of an AVClass context structure.
Definition: log.h:67
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:466
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:279
not initialized
Definition: rtsp.h:196
int64_t range_end
Definition: rtsp.h:138
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:118
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:155
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:750
int buffer_size
Definition: rtsp.h:410
int interleaved_max
Definition: rtsp.h:93
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:267
RTSPStatusCode
RTSP handling.
Definition: rtspcodes.h:31
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:138
#define INET6_ADDRSTRLEN
Definition: network.h:227
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
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...
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.
Main libavformat public API header.
initialized, requesting a seek
Definition: rtsp.h:199
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:288
RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:463
char default_lang[4]
Definition: rtsp.h:409
initialized, but not receiving data
Definition: rtsp.h:198
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...
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:749
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:397
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.
TCP; interleaved in RTSP.
Definition: rtsp.h:39
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:276
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:446
uint64_t packets
The number of returned packets.
Definition: rtsp.h:349
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:593
AVDiscard
Definition: avcodec.h:665
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:456
Realmedia-style server.
Definition: rtsp.h:208
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:344
int interleaved_max
Definition: rtsp.h:444
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:114
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:444
This structure stores compressed data.
Definition: avcodec.h:1150
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:105
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:436
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data...
Definition: rtsp.h:97
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:437
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:101
int initial_pause
Do not begin to play the stream immediately.
Definition: rtsp.h:364