1/*
2 * RTSP definitions
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21#ifndef AVFORMAT_RTSP_H
22#define AVFORMAT_RTSP_H
23
24#include <stdint.h>
25#include "avformat.h"
26#include "rtspcodes.h"
27#include "rtpdec.h"
28#include "network.h"
29#include "httpauth.h"
30
31#include "libavutil/log.h"
32#include "libavutil/opt.h"
33
34/**
35 * Network layer over which RTP/etc packet data will be transported.
36 */
37enum RTSPLowerTransport {
38    RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
39    RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
40    RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
41    RTSP_LOWER_TRANSPORT_NB,
42    RTSP_LOWER_TRANSPORT_HTTP = 8,          /**< HTTP tunneled - not a proper
43                                                 transport mode as such,
44                                                 only for use via AVOptions */
45};
46
47/**
48 * Packet profile of the data that we will be receiving. Real servers
49 * commonly send RDT (although they can sometimes send RTP as well),
50 * whereas most others will send RTP.
51 */
52enum RTSPTransport {
53    RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
54    RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
55    RTSP_TRANSPORT_NB
56};
57
58/**
59 * Transport mode for the RTSP data. This may be plain, or
60 * tunneled, which is done over HTTP.
61 */
62enum RTSPControlTransport {
63    RTSP_MODE_PLAIN,   /**< Normal RTSP */
64    RTSP_MODE_TUNNEL   /**< RTSP over HTTP (tunneling) */
65};
66
67#define RTSP_DEFAULT_PORT   554
68#define RTSP_MAX_TRANSPORTS 8
69#define RTSP_TCP_MAX_PACKET_SIZE 1472
70#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
71#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
72#define RTSP_RTP_PORT_MIN 5000
73#define RTSP_RTP_PORT_MAX 10000
74
75/**
76 * This describes a single item in the "Transport:" line of one stream as
77 * negotiated by the SETUP RTSP command. Multiple transports are comma-
78 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
79 * client_port=1000-1001;server_port=1800-1801") and described in separate
80 * RTSPTransportFields.
81 */
82typedef struct RTSPTransportField {
83    /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
84     * with a '$', stream length and stream ID. If the stream ID is within
85     * the range of this interleaved_min-max, then the packet belongs to
86     * this stream. */
87    int interleaved_min, interleaved_max;
88
89    /** UDP multicast port range; the ports to which we should connect to
90     * receive multicast UDP data. */
91    int port_min, port_max;
92
93    /** UDP client ports; these should be the local ports of the UDP RTP
94     * (and RTCP) sockets over which we receive RTP/RTCP data. */
95    int client_port_min, client_port_max;
96
97    /** UDP unicast server port range; the ports to which we should connect
98     * to receive unicast UDP RTP/RTCP data. */
99    int server_port_min, server_port_max;
100
101    /** time-to-live value (required for multicast); the amount of HOPs that
102     * packets will be allowed to make before being discarded. */
103    int ttl;
104
105    struct sockaddr_storage destination; /**< destination IP address */
106    char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
107
108    /** data/packet transport protocol; e.g. RTP or RDT */
109    enum RTSPTransport transport;
110
111    /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
112    enum RTSPLowerTransport lower_transport;
113} RTSPTransportField;
114
115/**
116 * This describes the server response to each RTSP command.
117 */
118typedef struct RTSPMessageHeader {
119    /** length of the data following this header */
120    int content_length;
121
122    enum RTSPStatusCode status_code; /**< response code from server */
123
124    /** number of items in the 'transports' variable below */
125    int nb_transports;
126
127    /** Time range of the streams that the server will stream. In
128     * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
129    int64_t range_start, range_end;
130
131    /** describes the complete "Transport:" line of the server in response
132     * to a SETUP RTSP command by the client */
133    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
134
135    int seq;                         /**< sequence number */
136
137    /** the "Session:" field. This value is initially set by the server and
138     * should be re-transmitted by the client in every RTSP command. */
139    char session_id[512];
140
141    /** the "Location:" field. This value is used to handle redirection.
142     */
143    char location[4096];
144
145    /** the "RealChallenge1:" field from the server */
146    char real_challenge[64];
147
148    /** the "Server: field, which can be used to identify some special-case
149     * servers that are not 100% standards-compliant. We use this to identify
150     * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
151     * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
152     * use something like "Helix [..] Server Version v.e.r.sion (platform)
153     * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
154     * where platform is the output of $uname -msr | sed 's/ /-/g'. */
155    char server[64];
156
157    /** The "timeout" comes as part of the server response to the "SETUP"
158     * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
159     * time, in seconds, that the server will go without traffic over the
160     * RTSP/TCP connection before it closes the connection. To prevent
161     * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
162     * than this value. */
163    int timeout;
164
165    /** The "Notice" or "X-Notice" field value. See
166     * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
167     * for a complete list of supported values. */
168    int notice;
169
170    /** The "reason" is meant to specify better the meaning of the error code
171     * returned
172     */
173    char reason[256];
174} RTSPMessageHeader;
175
176/**
177 * Client state, i.e. whether we are currently receiving data (PLAYING) or
178 * setup-but-not-receiving (PAUSED). State can be changed in applications
179 * by calling av_read_play/pause().
180 */
181enum RTSPClientState {
182    RTSP_STATE_IDLE,    /**< not initialized */
183    RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
184    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
185    RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
186};
187
188/**
189 * Identify particular servers that require special handling, such as
190 * standards-incompliant "Transport:" lines in the SETUP request.
191 */
192enum RTSPServerType {
193    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
194    RTSP_SERVER_REAL, /**< Realmedia-style server */
195    RTSP_SERVER_WMS,  /**< Windows Media server */
196    RTSP_SERVER_NB
197};
198
199/**
200 * Private data for the RTSP demuxer.
201 *
202 * @todo Use AVIOContext instead of URLContext
203 */
204typedef struct RTSPState {
205    const AVClass *class;             /**< Class for private options. */
206    URLContext *rtsp_hd; /* RTSP TCP connection handle */
207
208    /** number of items in the 'rtsp_streams' variable */
209    int nb_rtsp_streams;
210
211    struct RTSPStream **rtsp_streams; /**< streams in this session */
212
213    /** indicator of whether we are currently receiving data from the
214     * server. Basically this isn't more than a simple cache of the
215     * last PLAY/PAUSE command sent to the server, to make sure we don't
216     * send 2x the same unexpectedly or commands in the wrong state. */
217    enum RTSPClientState state;
218
219    /** the seek value requested when calling av_seek_frame(). This value
220     * is subsequently used as part of the "Range" parameter when emitting
221     * the RTSP PLAY command. If we are currently playing, this command is
222     * called instantly. If we are currently paused, this command is called
223     * whenever we resume playback. Either way, the value is only used once,
224     * see rtsp_read_play() and rtsp_read_seek(). */
225    int64_t seek_timestamp;
226
227    int seq;                          /**< RTSP command sequence number */
228
229    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
230     * identifier that the client should re-transmit in each RTSP command */
231    char session_id[512];
232
233    /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
234     * the server will go without traffic on the RTSP/TCP line before it
235     * closes the connection. */
236    int timeout;
237
238    /** timestamp of the last RTSP command that we sent to the RTSP server.
239     * This is used to calculate when to send dummy commands to keep the
240     * connection alive, in conjunction with timeout. */
241    int64_t last_cmd_time;
242
243    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
244    enum RTSPTransport transport;
245
246    /** the negotiated network layer transport protocol; e.g. TCP or UDP
247     * uni-/multicast */
248    enum RTSPLowerTransport lower_transport;
249
250    /** brand of server that we're talking to; e.g. WMS, REAL or other.
251     * Detected based on the value of RTSPMessageHeader->server or the presence
252     * of RTSPMessageHeader->real_challenge */
253    enum RTSPServerType server_type;
254
255    /** the "RealChallenge1:" field from the server */
256    char real_challenge[64];
257
258    /** plaintext authorization line (username:password) */
259    char auth[128];
260
261    /** authentication state */
262    HTTPAuthState auth_state;
263
264    /** The last reply of the server to a RTSP command */
265    char last_reply[2048]; /* XXX: allocate ? */
266
267    /** RTSPStream->transport_priv of the last stream that we read a
268     * packet from */
269    void *cur_transport_priv;
270
271    /** The following are used for Real stream selection */
272    //@{
273    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
274    int need_subscription;
275
276    /** stream setup during the last frame read. This is used to detect if
277     * we need to subscribe or unsubscribe to any new streams. */
278    enum AVDiscard *real_setup_cache;
279
280    /** current stream setup. This is a temporary buffer used to compare
281     * current setup to previous frame setup. */
282    enum AVDiscard *real_setup;
283
284    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
285     * this is used to send the same "Unsubscribe:" if stream setup changed,
286     * before sending a new "Subscribe:" command. */
287    char last_subscription[1024];
288    //@}
289
290    /** The following are used for RTP/ASF streams */
291    //@{
292    /** ASF demuxer context for the embedded ASF stream from WMS servers */
293    AVFormatContext *asf_ctx;
294
295    /** cache for position of the asf demuxer, since we load a new
296     * data packet in the bytecontext for each incoming RTSP packet. */
297    uint64_t asf_pb_pos;
298    //@}
299
300    /** some MS RTSP streams contain a URL in the SDP that we need to use
301     * for all subsequent RTSP requests, rather than the input URI; in
302     * other cases, this is a copy of AVFormatContext->filename. */
303    char control_uri[1024];
304
305    /** Additional output handle, used when input and output are done
306     * separately, eg for HTTP tunneling. */
307    URLContext *rtsp_hd_out;
308
309    /** RTSP transport mode, such as plain or tunneled. */
310    enum RTSPControlTransport control_transport;
311
312    /* Number of RTCP BYE packets the RTSP session has received.
313     * An EOF is propagated back if nb_byes == nb_streams.
314     * This is reset after a seek. */
315    int nb_byes;
316
317    /** Reusable buffer for receiving packets */
318    uint8_t* recvbuf;
319
320    /**
321     * A mask with all requested transport methods
322     */
323    int lower_transport_mask;
324
325    /**
326     * The number of returned packets
327     */
328    uint64_t packets;
329
330    /**
331     * Polling array for udp
332     */
333    struct pollfd *p;
334
335    /**
336     * Whether the server supports the GET_PARAMETER method.
337     */
338    int get_parameter_supported;
339
340    /**
341     * Do not begin to play the stream immediately.
342     */
343    int initial_pause;
344
345    /**
346     * Option flags for the chained RTP muxer.
347     */
348    int rtp_muxer_flags;
349
350    /** Whether the server accepts the x-Dynamic-Rate header */
351    int accept_dynamic_rate;
352
353    /**
354     * Various option flags for the RTSP muxer/demuxer.
355     */
356    int rtsp_flags;
357
358    /**
359     * Mask of all requested media types
360     */
361    int media_type_mask;
362} RTSPState;
363
364#define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
365                                          receive packets only from the right
366                                          source address and port. */
367
368/**
369 * Describe a single stream, as identified by a single m= line block in the
370 * SDP content. In the case of RDT, one RTSPStream can represent multiple
371 * AVStreams. In this case, each AVStream in this set has similar content
372 * (but different codec/bitrate).
373 */
374typedef struct RTSPStream {
375    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
376    void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
377
378    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
379    int stream_index;
380
381    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
382     * for the selected transport. Only used for TCP. */
383    int interleaved_min, interleaved_max;
384
385    char control_url[1024];   /**< url for this stream (from SDP) */
386
387    /** The following are used only in SDP, not RTSP */
388    //@{
389    int sdp_port;             /**< port (from SDP content) */
390    struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
391    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
392    int sdp_payload_type;     /**< payload type */
393    //@}
394
395    /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
396    //@{
397    /** handler structure */
398    RTPDynamicProtocolHandler *dynamic_handler;
399
400    /** private data associated with the dynamic protocol */
401    PayloadContext *dynamic_protocol_context;
402    //@}
403} RTSPStream;
404
405void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
406                        RTSPState *rt, const char *method);
407
408/**
409 * Send a command to the RTSP server without waiting for the reply.
410 *
411 * @see rtsp_send_cmd_with_content_async
412 */
413int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
414                           const char *url, const char *headers);
415
416/**
417 * Send a command to the RTSP server and wait for the reply.
418 *
419 * @param s RTSP (de)muxer context
420 * @param method the method for the request
421 * @param url the target url for the request
422 * @param headers extra header lines to include in the request
423 * @param reply pointer where the RTSP message header will be stored
424 * @param content_ptr pointer where the RTSP message body, if any, will
425 *                    be stored (length is in reply)
426 * @param send_content if non-null, the data to send as request body content
427 * @param send_content_length the length of the send_content data, or 0 if
428 *                            send_content is null
429 *
430 * @return zero if success, nonzero otherwise
431 */
432int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
433                                  const char *method, const char *url,
434                                  const char *headers,
435                                  RTSPMessageHeader *reply,
436                                  unsigned char **content_ptr,
437                                  const unsigned char *send_content,
438                                  int send_content_length);
439
440/**
441 * Send a command to the RTSP server and wait for the reply.
442 *
443 * @see rtsp_send_cmd_with_content
444 */
445int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
446                     const char *url, const char *headers,
447                     RTSPMessageHeader *reply, unsigned char **content_ptr);
448
449/**
450 * Read a RTSP message from the server, or prepare to read data
451 * packets if we're reading data interleaved over the TCP/RTSP
452 * connection as well.
453 *
454 * @param s RTSP (de)muxer context
455 * @param reply pointer where the RTSP message header will be stored
456 * @param content_ptr pointer where the RTSP message body, if any, will
457 *                    be stored (length is in reply)
458 * @param return_on_interleaved_data whether the function may return if we
459 *                   encounter a data marker ('$'), which precedes data
460 *                   packets over interleaved TCP/RTSP connections. If this
461 *                   is set, this function will return 1 after encountering
462 *                   a '$'. If it is not set, the function will skip any
463 *                   data packets (if they are encountered), until a reply
464 *                   has been fully parsed. If no more data is available
465 *                   without parsing a reply, it will return an error.
466 * @param method the RTSP method this is a reply to. This affects how
467 *               some response headers are acted upon. May be NULL.
468 *
469 * @return 1 if a data packets is ready to be received, -1 on error,
470 *          and 0 on success.
471 */
472int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
473                       unsigned char **content_ptr,
474                       int return_on_interleaved_data, const char *method);
475
476/**
477 * Skip a RTP/TCP interleaved packet.
478 */
479void ff_rtsp_skip_packet(AVFormatContext *s);
480
481/**
482 * Connect to the RTSP server and set up the individual media streams.
483 * This can be used for both muxers and demuxers.
484 *
485 * @param s RTSP (de)muxer context
486 *
487 * @return 0 on success, < 0 on error. Cleans up all allocations done
488 *          within the function on error.
489 */
490int ff_rtsp_connect(AVFormatContext *s);
491
492/**
493 * Close and free all streams within the RTSP (de)muxer
494 *
495 * @param s RTSP (de)muxer context
496 */
497void ff_rtsp_close_streams(AVFormatContext *s);
498
499/**
500 * Close all connection handles within the RTSP (de)muxer
501 *
502 * @param s RTSP (de)muxer context
503 */
504void ff_rtsp_close_connections(AVFormatContext *s);
505
506/**
507 * Get the description of the stream and set up the RTSPStream child
508 * objects.
509 */
510int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
511
512/**
513 * Announce the stream to the server and set up the RTSPStream child
514 * objects for each media stream.
515 */
516int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
517
518/**
519 * Parse an SDP description of streams by populating an RTSPState struct
520 * within the AVFormatContext; also allocate the RTP streams and the
521 * pollfd array used for UDP streams.
522 */
523int ff_sdp_parse(AVFormatContext *s, const char *content);
524
525/**
526 * Receive one RTP packet from an TCP interleaved RTSP stream.
527 */
528int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
529                            uint8_t *buf, int buf_size);
530
531/**
532 * Receive one packet from the RTSPStreams set up in the AVFormatContext
533 * (which should contain a RTSPState struct as priv_data).
534 */
535int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
536
537/**
538 * Do the SETUP requests for each stream for the chosen
539 * lower transport mode.
540 * @return 0 on success, <0 on error, 1 if protocol is unavailable
541 */
542int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
543                               int lower_transport, const char *real_challenge);
544
545/**
546 * Undo the effect of ff_rtsp_make_setup_request, close the
547 * transport_priv and rtp_handle fields.
548 */
549void ff_rtsp_undo_setup(AVFormatContext *s);
550
551extern const AVOption ff_rtsp_options[];
552
553#endif /* AVFORMAT_RTSP_H */
554