FFmpeg  4.4
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_TCP_MAX_PACKET_SIZE 1472
78 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
79 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
80 #define RTSP_RTP_PORT_MIN 5000
81 #define RTSP_RTP_PORT_MAX 65000
82 #define SDP_MAX_SIZE 16384
83 
84 /**
85  * This describes a single item in the "Transport:" line of one stream as
86  * negotiated by the SETUP RTSP command. Multiple transports are comma-
87  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
88  * client_port=1000-1001;server_port=1800-1801") and described in separate
89  * RTSPTransportFields.
90  */
91 typedef struct RTSPTransportField {
92  /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
93  * with a '$', stream length and stream ID. If the stream ID is within
94  * the range of this interleaved_min-max, then the packet belongs to
95  * this stream. */
97 
98  /** UDP multicast port range; the ports to which we should connect to
99  * receive multicast UDP data. */
101 
102  /** UDP client ports; these should be the local ports of the UDP RTP
103  * (and RTCP) sockets over which we receive RTP/RTCP data. */
105 
106  /** UDP unicast server port range; the ports to which we should connect
107  * to receive unicast UDP RTP/RTCP data. */
109 
110  /** time-to-live value (required for multicast); the amount of HOPs that
111  * packets will be allowed to make before being discarded. */
112  int ttl;
113 
114  /** transport set to record data */
116 
117  struct sockaddr_storage destination; /**< destination IP address */
118  char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
119 
120  /** data/packet transport protocol; e.g. RTP or RDT */
122 
123  /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
126 
127 /**
128  * This describes the server response to each RTSP command.
129  */
130 typedef struct RTSPMessageHeader {
131  /** length of the data following this header */
133 
134  enum RTSPStatusCode status_code; /**< response code from server */
135 
136  /** number of items in the 'transports' variable below */
138 
139  /** Time range of the streams that the server will stream. In
140  * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
142 
143  /** describes the complete "Transport:" line of the server in response
144  * to a SETUP RTSP command by the client */
146 
147  int seq; /**< sequence number */
148 
149  /** the "Session:" field. This value is initially set by the server and
150  * should be re-transmitted by the client in every RTSP command. */
151  char session_id[512];
152 
153  /** the "Location:" field. This value is used to handle redirection.
154  */
155  char location[4096];
156 
157  /** the "RealChallenge1:" field from the server */
158  char real_challenge[64];
159 
160  /** the "Server: field, which can be used to identify some special-case
161  * servers that are not 100% standards-compliant. We use this to identify
162  * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
163  * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
164  * use something like "Helix [..] Server Version v.e.r.sion (platform)
165  * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
166  * where platform is the output of $uname -msr | sed 's/ /-/g'. */
167  char server[64];
168 
169  /** The "timeout" comes as part of the server response to the "SETUP"
170  * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
171  * time, in seconds, that the server will go without traffic over the
172  * RTSP/TCP connection before it closes the connection. To prevent
173  * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
174  * than this value. */
175  int timeout;
176 
177  /** The "Notice" or "X-Notice" field value. See
178  * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
179  * for a complete list of supported values. */
180  int notice;
181 
182  /** The "reason" is meant to specify better the meaning of the error code
183  * returned
184  */
185  char reason[256];
186 
187  /**
188  * Content type header
189  */
190  char content_type[64];
191 
192  /**
193  * SAT>IP com.ses.streamID header
194  */
195  char stream_id[64];
197 
198 /**
199  * Client state, i.e. whether we are currently receiving data (PLAYING) or
200  * setup-but-not-receiving (PAUSED). State can be changed in applications
201  * by calling av_read_play/pause().
202  */
204  RTSP_STATE_IDLE, /**< not initialized */
205  RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
206  RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
207  RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
208 };
209 
210 /**
211  * Identify particular servers that require special handling, such as
212  * standards-incompliant "Transport:" lines in the SETUP request.
213  */
215  RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
216  RTSP_SERVER_REAL, /**< Realmedia-style server */
217  RTSP_SERVER_WMS, /**< Windows Media server */
218  RTSP_SERVER_SATIP,/**< SAT>IP server */
220 };
221 
222 /**
223  * Private data for the RTSP demuxer.
224  *
225  * @todo Use AVIOContext instead of URLContext
226  */
227 typedef struct RTSPState {
228  const AVClass *class; /**< Class for private options. */
229  URLContext *rtsp_hd; /* RTSP TCP connection handle */
230 
231  /** number of items in the 'rtsp_streams' variable */
233 
234  struct RTSPStream **rtsp_streams; /**< streams in this session */
235 
236  /** indicator of whether we are currently receiving data from the
237  * server. Basically this isn't more than a simple cache of the
238  * last PLAY/PAUSE command sent to the server, to make sure we don't
239  * send 2x the same unexpectedly or commands in the wrong state. */
240  enum RTSPClientState state;
241 
242  /** the seek value requested when calling av_seek_frame(). This value
243  * is subsequently used as part of the "Range" parameter when emitting
244  * the RTSP PLAY command. If we are currently playing, this command is
245  * called instantly. If we are currently paused, this command is called
246  * whenever we resume playback. Either way, the value is only used once,
247  * see rtsp_read_play() and rtsp_read_seek(). */
248  int64_t seek_timestamp;
249 
250  int seq; /**< RTSP command sequence number */
251 
252  /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
253  * identifier that the client should re-transmit in each RTSP command */
254  char session_id[512];
255 
256  /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
257  * the server will go without traffic on the RTSP/TCP line before it
258  * closes the connection. */
259  int timeout;
260 
261  /** timestamp of the last RTSP command that we sent to the RTSP server.
262  * This is used to calculate when to send dummy commands to keep the
263  * connection alive, in conjunction with timeout. */
264  int64_t last_cmd_time;
265 
266  /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
268 
269  /** the negotiated network layer transport protocol; e.g. TCP or UDP
270  * uni-/multicast */
272 
273  /** brand of server that we're talking to; e.g. WMS, REAL or other.
274  * Detected based on the value of RTSPMessageHeader->server or the presence
275  * of RTSPMessageHeader->real_challenge */
277 
278  /** the "RealChallenge1:" field from the server */
279  char real_challenge[64];
280 
281  /** plaintext authorization line (username:password) */
282  char auth[128];
283 
284  /** authentication state */
286 
287  /** The last reply of the server to a RTSP command */
288  char last_reply[2048]; /* XXX: allocate ? */
289 
290  /** RTSPStream->transport_priv of the last stream that we read a
291  * packet from */
293 
294  /** The following are used for Real stream selection */
295  //@{
296  /** whether we need to send a "SET_PARAMETER Subscribe:" command */
298 
299  /** stream setup during the last frame read. This is used to detect if
300  * we need to subscribe or unsubscribe to any new streams. */
302 
303  /** current stream setup. This is a temporary buffer used to compare
304  * current setup to previous frame setup. */
306 
307  /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
308  * this is used to send the same "Unsubscribe:" if stream setup changed,
309  * before sending a new "Subscribe:" command. */
310  char last_subscription[1024];
311  //@}
312 
313  /** The following are used for RTP/ASF streams */
314  //@{
315  /** ASF demuxer context for the embedded ASF stream from WMS servers */
317 
318  /** cache for position of the asf demuxer, since we load a new
319  * data packet in the bytecontext for each incoming RTSP packet. */
320  uint64_t asf_pb_pos;
321  //@}
322 
323  /** some MS RTSP streams contain a URL in the SDP that we need to use
324  * for all subsequent RTSP requests, rather than the input URI; in
325  * other cases, this is a copy of AVFormatContext->filename. */
327 
328  /** The following are used for parsing raw mpegts in udp */
329  //@{
330  struct MpegTSContext *ts;
333  //@}
334 
335  /** Additional output handle, used when input and output are done
336  * separately, eg for HTTP tunneling. */
338 
339  /** RTSP transport mode, such as plain or tunneled. */
341 
342  /* Number of RTCP BYE packets the RTSP session has received.
343  * An EOF is propagated back if nb_byes == nb_streams.
344  * This is reset after a seek. */
345  int nb_byes;
346 
347  /** Reusable buffer for receiving packets */
349 
350  /**
351  * A mask with all requested transport methods
352  */
354 
355  /**
356  * The number of returned packets
357  */
358  uint64_t packets;
359 
360  /**
361  * Polling array for udp
362  */
363  struct pollfd *p;
364  int max_p;
365 
366  /**
367  * Whether the server supports the GET_PARAMETER method.
368  */
370 
371  /**
372  * Do not begin to play the stream immediately.
373  */
375 
376  /**
377  * Option flags for the chained RTP muxer.
378  */
380 
381  /** Whether the server accepts the x-Dynamic-Rate header */
383 
384  /**
385  * Various option flags for the RTSP muxer/demuxer.
386  */
388 
389  /**
390  * Mask of all requested media types
391  */
393 
394  /**
395  * Minimum and maximum local UDP ports.
396  */
398 
399  /**
400  * Timeout to wait for incoming connections.
401  */
403 
404  /**
405  * timeout of socket i/o operations.
406  */
407  int stimeout;
408 
409  /**
410  * Size of RTP packet reordering queue.
411  */
413 
414  /**
415  * User-Agent string
416  */
417  char *user_agent;
418 
419  char default_lang[4];
421  int pkt_size;
422 } RTSPState;
423 
424 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
425  receive packets only from the right
426  source address and port. */
427 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
428 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
429 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
430  address of received packets. */
431 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
432 #define RTSP_FLAG_SATIP_RAW 0x20 /**< Export SAT>IP stream as raw MPEG-TS */
433 
434 typedef struct RTSPSource {
435  char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
436 } RTSPSource;
437 
438 /**
439  * Describe a single stream, as identified by a single m= line block in the
440  * SDP content. In the case of RDT, one RTSPStream can represent multiple
441  * AVStreams. In this case, each AVStream in this set has similar content
442  * (but different codec/bitrate).
443  */
444 typedef struct RTSPStream {
445  URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
446  void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
447 
448  /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
450 
451  /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
452  * for the selected transport. Only used for TCP. */
454 
455  char control_url[MAX_URL_SIZE]; /**< url for this stream (from SDP) */
456 
457  /** The following are used only in SDP, not RTSP */
458  //@{
459  int sdp_port; /**< port (from SDP content) */
460  struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
461  int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
462  struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
463  int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
464  struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
465  int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
466  int sdp_payload_type; /**< payload type */
467  //@}
468 
469  /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
470  //@{
471  /** handler structure */
473 
474  /** private data associated with the dynamic protocol */
476  //@}
477 
478  /** Enable sending RTCP feedback messages according to RFC 4585 */
479  int feedback;
480 
481  /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
482  uint32_t ssrc;
483 
484  char crypto_suite[40];
485  char crypto_params[100];
486 } RTSPStream;
487 
489  RTSPMessageHeader *reply, const char *buf,
490  RTSPState *rt, const char *method);
491 
492 /**
493  * Send a command to the RTSP server without waiting for the reply.
494  *
495  * @see rtsp_send_cmd_with_content_async
496  */
497 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
498  const char *url, const char *headers);
499 
500 /**
501  * Send a command to the RTSP server and wait for the reply.
502  *
503  * @param s RTSP (de)muxer context
504  * @param method the method for the request
505  * @param url the target url for the request
506  * @param headers extra header lines to include in the request
507  * @param reply pointer where the RTSP message header will be stored
508  * @param content_ptr pointer where the RTSP message body, if any, will
509  * be stored (length is in reply)
510  * @param send_content if non-null, the data to send as request body content
511  * @param send_content_length the length of the send_content data, or 0 if
512  * send_content is null
513  *
514  * @return zero if success, nonzero otherwise
515  */
517  const char *method, const char *url,
518  const char *headers,
519  RTSPMessageHeader *reply,
520  unsigned char **content_ptr,
521  const unsigned char *send_content,
522  int send_content_length);
523 
524 /**
525  * Send a command to the RTSP server and wait for the reply.
526  *
527  * @see rtsp_send_cmd_with_content
528  */
529 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
530  const char *url, const char *headers,
531  RTSPMessageHeader *reply, unsigned char **content_ptr);
532 
533 /**
534  * Read a RTSP message from the server, or prepare to read data
535  * packets if we're reading data interleaved over the TCP/RTSP
536  * connection as well.
537  *
538  * @param s RTSP (de)muxer context
539  * @param reply pointer where the RTSP message header will be stored
540  * @param content_ptr pointer where the RTSP message body, if any, will
541  * be stored (length is in reply)
542  * @param return_on_interleaved_data whether the function may return if we
543  * encounter a data marker ('$'), which precedes data
544  * packets over interleaved TCP/RTSP connections. If this
545  * is set, this function will return 1 after encountering
546  * a '$'. If it is not set, the function will skip any
547  * data packets (if they are encountered), until a reply
548  * has been fully parsed. If no more data is available
549  * without parsing a reply, it will return an error.
550  * @param method the RTSP method this is a reply to. This affects how
551  * some response headers are acted upon. May be NULL.
552  *
553  * @return 1 if a data packets is ready to be received, -1 on error,
554  * and 0 on success.
555  */
557  unsigned char **content_ptr,
558  int return_on_interleaved_data, const char *method);
559 
560 /**
561  * Skip a RTP/TCP interleaved packet.
562  */
564 
565 /**
566  * Connect to the RTSP server and set up the individual media streams.
567  * This can be used for both muxers and demuxers.
568  *
569  * @param s RTSP (de)muxer context
570  *
571  * @return 0 on success, < 0 on error. Cleans up all allocations done
572  * within the function on error.
573  */
575 
576 /**
577  * Close and free all streams within the RTSP (de)muxer
578  *
579  * @param s RTSP (de)muxer context
580  */
582 
583 /**
584  * Close all connection handles within the RTSP (de)muxer
585  *
586  * @param s RTSP (de)muxer context
587  */
589 
590 /**
591  * Get the description of the stream and set up the RTSPStream child
592  * objects.
593  */
595 
596 /**
597  * Announce the stream to the server and set up the RTSPStream child
598  * objects for each media stream.
599  */
601 
602 /**
603  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
604  * listen mode.
605  */
607 
608 /**
609  * Parse an SDP description of streams by populating an RTSPState struct
610  * within the AVFormatContext; also allocate the RTP streams and the
611  * pollfd array used for UDP streams.
612  */
613 int ff_sdp_parse(AVFormatContext *s, const char *content);
614 
615 /**
616  * Receive one RTP packet from an TCP interleaved RTSP stream.
617  */
619  uint8_t *buf, int buf_size);
620 
621 /**
622  * Send buffered packets over TCP.
623  */
625 
626 /**
627  * Receive one packet from the RTSPStreams set up in the AVFormatContext
628  * (which should contain a RTSPState struct as priv_data).
629  */
631 
632 /**
633  * Do the SETUP requests for each stream for the chosen
634  * lower transport mode.
635  * @return 0 on success, <0 on error, 1 if protocol is unavailable
636  */
637 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
638  int lower_transport, const char *real_challenge);
639 
640 /**
641  * Undo the effect of ff_rtsp_make_setup_request, close the
642  * transport_priv and rtp_handle fields.
643  */
644 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
645 
646 /**
647  * Open RTSP transport context.
648  */
650 
651 extern const AVOption ff_rtsp_options[];
652 
653 #endif /* AVFORMAT_RTSP_H */
uint8_t
Main libavformat public API header.
#define s(width, name)
Definition: cbs_vp9.c:257
AVDiscard
Definition: avcodec.h:227
#define MAX_URL_SIZE
Definition: internal.h:30
common internal API header
#define INET6_ADDRSTRLEN
Definition: network.h:237
AVOptions.
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
int ff_sdp_parse(AVFormatContext *s, const char *content)
Parse an SDP description of streams by populating an RTSPState struct within the AVFormatContext; als...
RTSPServerType
Identify particular servers that require special handling, such as standards-incompliant "Transport:"...
Definition: rtsp.h:214
@ RTSP_SERVER_NB
Definition: rtsp.h:219
@ RTSP_SERVER_SATIP
SAT>IP server.
Definition: rtsp.h:218
@ RTSP_SERVER_WMS
Windows Media server.
Definition: rtsp.h:217
@ RTSP_SERVER_RTP
Standards-compliant RTP-server.
Definition: rtsp.h:215
@ RTSP_SERVER_REAL
Realmedia-style server.
Definition: rtsp.h:216
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.
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:603
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...
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.
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.
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
Open RTSP transport context.
Definition: rtsp.c:828
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...
#define RTSP_MAX_TRANSPORTS
Definition: rtsp.h:76
void ff_rtsp_parse_line(AVFormatContext *s, RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method)
RTSPLowerTransport
Network layer over which RTP/etc packet data will be transported.
Definition: rtsp.h:38
@ RTSP_LOWER_TRANSPORT_TCP
TCP; interleaved in RTSP.
Definition: rtsp.h:40
@ RTSP_LOWER_TRANSPORT_HTTP
HTTP tunneled - not a proper transport mode as such, only for use via AVOptions.
Definition: rtsp.h:43
@ RTSP_LOWER_TRANSPORT_NB
Definition: rtsp.h:42
@ RTSP_LOWER_TRANSPORT_UDP_MULTICAST
UDP/multicast.
Definition: rtsp.h:41
@ 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
@ RTSP_LOWER_TRANSPORT_UDP
UDP/unicast.
Definition: rtsp.h:39
@ RTSP_LOWER_TRANSPORT_HTTPS
HTTPS tunneled.
Definition: rtsp.h:46
int ff_rtsp_parse_streaming_commands(AVFormatContext *s)
Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in listen mode.
Definition: rtspdec.c:474
RTSPControlTransport
Transport mode for the RTSP data.
Definition: rtsp.h:69
@ RTSP_MODE_PLAIN
Normal RTSP.
Definition: rtsp.h:70
@ RTSP_MODE_TUNNEL
RTSP over HTTP (tunneling)
Definition: rtsp.h:71
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
RTSPClientState
Client state, i.e.
Definition: rtsp.h:203
@ RTSP_STATE_SEEKING
initialized, requesting a seek
Definition: rtsp.h:207
@ RTSP_STATE_STREAMING
initialized and sending/receiving data
Definition: rtsp.h:205
@ RTSP_STATE_PAUSED
initialized, but not receiving data
Definition: rtsp.h:206
@ RTSP_STATE_IDLE
not initialized
Definition: rtsp.h:204
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.
const AVOption ff_rtsp_options[]
Definition: rtsp.c:80
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:141
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:792
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:774
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:760
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
RTSPTransport
Packet profile of the data that we will be receiving.
Definition: rtsp.h:58
@ RTSP_TRANSPORT_RTP
Standards-compliant RTP.
Definition: rtsp.h:59
@ RTSP_TRANSPORT_RAW
Raw data (over UDP)
Definition: rtsp.h:61
@ RTSP_TRANSPORT_NB
Definition: rtsp.h:62
@ RTSP_TRANSPORT_RDT
Realmedia Data Transport.
Definition: rtsp.h:60
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:45
RTSPStatusCode
RTSP handling.
Definition: rtspcodes.h:31
Describe the class of an AVClass context structure.
Definition: log.h:67
Format I/O context.
Definition: avformat.h:1232
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
HTTP Authentication state structure.
Definition: httpauth.h:55
RTP/JPEG specific private data.
Definition: rdt.c:83
This describes the server response to each RTSP command.
Definition: rtsp.h:130
char reason[256]
The "reason" is meant to specify better the meaning of the error code returned.
Definition: rtsp.h:185
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:158
char location[4096]
the "Location:" field.
Definition: rtsp.h:155
int notice
The "Notice" or "X-Notice" field value.
Definition: rtsp.h:180
int64_t range_start
Time range of the streams that the server will stream.
Definition: rtsp.h:141
int timeout
The "timeout" comes as part of the server response to the "SETUP" command, in the "Session: <xyz>[;ti...
Definition: rtsp.h:175
char session_id[512]
the "Session:" field.
Definition: rtsp.h:151
char server[64]
the "Server: field, which can be used to identify some special-case servers that are not 100% standar...
Definition: rtsp.h:167
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:134
int seq
sequence number
Definition: rtsp.h:147
char content_type[64]
Content type header.
Definition: rtsp.h:190
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:145
int64_t range_end
Definition: rtsp.h:141
char stream_id[64]
SAT>IP com.ses.streamID header.
Definition: rtsp.h:195
int nb_transports
number of items in the 'transports' variable below
Definition: rtsp.h:137
int content_length
length of the data following this header
Definition: rtsp.h:132
char addr[128]
Source-specific multicast include source IP address (from SDP content)
Definition: rtsp.h:435
Private data for the RTSP demuxer.
Definition: rtsp.h:227
char real_challenge[64]
the "RealChallenge1:" field from the server
Definition: rtsp.h:279
int recvbuf_len
Definition: rtsp.h:332
int rtp_muxer_flags
Option flags for the chained RTP muxer.
Definition: rtsp.h:379
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:232
enum RTSPTransport transport
the negotiated data/packet transport protocol; e.g.
Definition: rtsp.h:267
int rtp_port_max
Definition: rtsp.h:397
AVFormatContext * asf_ctx
The following are used for RTP/ASF streams.
Definition: rtsp.h:316
char last_subscription[1024]
the last value of the "SET_PARAMETER Subscribe:" RTSP command.
Definition: rtsp.h:310
int timeout
copy of RTSPMessageHeader->timeout, i.e.
Definition: rtsp.h:259
int accept_dynamic_rate
Whether the server accepts the x-Dynamic-Rate header.
Definition: rtsp.h:382
HTTPAuthState auth_state
authentication state
Definition: rtsp.h:285
int initial_timeout
Timeout to wait for incoming connections.
Definition: rtsp.h:402
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling.
Definition: rtsp.h:337
int lower_transport_mask
A mask with all requested transport methods.
Definition: rtsp.h:353
enum AVDiscard * real_setup
current stream setup.
Definition: rtsp.h:305
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:271
int max_p
Definition: rtsp.h:364
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:320
int64_t last_cmd_time
timestamp of the last RTSP command that we sent to the RTSP server.
Definition: rtsp.h:264
int need_subscription
The following are used for Real stream selection.
Definition: rtsp.h:297
char session_id[512]
copy of RTSPMessageHeader->session_id, i.e.
Definition: rtsp.h:254
int media_type_mask
Mask of all requested media types.
Definition: rtsp.h:392
struct MpegTSContext * ts
The following are used for parsing raw mpegts in udp.
Definition: rtsp.h:330
enum RTSPServerType server_type
brand of server that we're talking to; e.g.
Definition: rtsp.h:276
uint8_t * recvbuf
Reusable buffer for receiving packets.
Definition: rtsp.h:348
char * user_agent
User-Agent string.
Definition: rtsp.h:417
int rtsp_flags
Various option flags for the RTSP muxer/demuxer.
Definition: rtsp.h:387
int64_t seek_timestamp
the seek value requested when calling av_seek_frame().
Definition: rtsp.h:248
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:326
enum RTSPControlTransport control_transport
RTSP transport mode, such as plain or tunneled.
Definition: rtsp.h:340
void * cur_transport_priv
RTSPStream->transport_priv of the last stream that we read a packet from.
Definition: rtsp.h:292
int buffer_size
Definition: rtsp.h:420
int reordering_queue_size
Size of RTP packet reordering queue.
Definition: rtsp.h:412
char auth[128]
plaintext authorization line (username:password)
Definition: rtsp.h:282
URLContext * rtsp_hd
Definition: rtsp.h:229
int rtp_port_min
Minimum and maximum local UDP ports.
Definition: rtsp.h:397
uint64_t packets
The number of returned packets.
Definition: rtsp.h:358
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:240
int recvbuf_pos
Definition: rtsp.h:331
enum AVDiscard * real_setup_cache
stream setup during the last frame read.
Definition: rtsp.h:301
int initial_pause
Do not begin to play the stream immediately.
Definition: rtsp.h:374
int seq
RTSP command sequence number.
Definition: rtsp.h:250
int stimeout
timeout of socket i/o operations.
Definition: rtsp.h:407
int pkt_size
Definition: rtsp.h:421
int nb_byes
Definition: rtsp.h:345
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:234
char default_lang[4]
Definition: rtsp.h:419
struct pollfd * p
Polling array for udp.
Definition: rtsp.h:363
int get_parameter_supported
Whether the server supports the GET_PARAMETER method.
Definition: rtsp.h:369
char last_reply[2048]
The last reply of the server to a RTSP command.
Definition: rtsp.h:288
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:444
struct RTSPSource ** exclude_source_addrs
Source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:464
char crypto_suite[40]
Definition: rtsp.h:484
int sdp_ttl
IP Time-To-Live (from SDP content)
Definition: rtsp.h:465
const RTPDynamicProtocolHandler * dynamic_handler
The following are used for dynamic protocols (rtpdec_*.c/rdt.c)
Definition: rtsp.h:472
int sdp_port
The following are used only in SDP, not RTSP.
Definition: rtsp.h:459
int nb_include_source_addrs
Number of source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:461
char crypto_params[100]
Definition: rtsp.h:485
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport.
Definition: rtsp.h:453
char control_url[MAX_URL_SIZE]
url for this stream (from SDP)
Definition: rtsp.h:455
int sdp_payload_type
payload type
Definition: rtsp.h:466
URLContext * rtp_handle
RTP stream handle (if UDP)
Definition: rtsp.h:445
int interleaved_max
Definition: rtsp.h:453
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:449
struct RTSPSource ** include_source_addrs
Source-specific multicast include source IP addresses (from SDP content)
Definition: rtsp.h:462
int feedback
Enable sending RTCP feedback messages according to RFC 4585.
Definition: rtsp.h:479
PayloadContext * dynamic_protocol_context
private data associated with the dynamic protocol
Definition: rtsp.h:475
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:446
uint32_t ssrc
SSRC for this stream, to allow identifying RTCP packets before the first RTP packet.
Definition: rtsp.h:482
int nb_exclude_source_addrs
Number of source-specific multicast exclude source IP addresses (from SDP content)
Definition: rtsp.h:463
struct sockaddr_storage sdp_ip
IP address (from SDP content)
Definition: rtsp.h:460
This describes a single item in the "Transport:" line of one stream as negotiated by the SETUP RTSP c...
Definition: rtsp.h:91
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:108
int interleaved_max
Definition: rtsp.h:96
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:104
char source[INET6_ADDRSTRLEN+1]
source IP address
Definition: rtsp.h:118
enum RTSPTransport transport
data/packet transport protocol; e.g.
Definition: rtsp.h:121
int mode_record
transport set to record data
Definition: rtsp.h:115
struct sockaddr_storage destination
destination IP address
Definition: rtsp.h:117
int ttl
time-to-live value (required for multicast); the amount of HOPs that packets will be allowed to make ...
Definition: rtsp.h:112
int client_port_max
Definition: rtsp.h:104
enum RTSPLowerTransport lower_transport
network layer transport protocol; e.g.
Definition: rtsp.h:124
int port_min
UDP multicast port range; the ports to which we should connect to receive multicast UDP data.
Definition: rtsp.h:100
int interleaved_min
interleave ids, if TCP transport; each TCP/RTSP data packet starts with a '$', stream length and stre...
Definition: rtsp.h:96
int server_port_max
Definition: rtsp.h:108
Definition: url.h:38
AVPacket * pkt
Definition: movenc.c:59