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 */
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    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 */
56enum RTSPTransport {
57    RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
58    RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
59    RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
60    RTSP_TRANSPORT_NB
61};
62
63/**
64 * Transport mode for the RTSP data. This may be plain, or
65 * tunneled, which is done over HTTP.
66 */
67enum RTSPControlTransport {
68    RTSP_MODE_PLAIN,   /**< Normal RTSP */
69    RTSP_MODE_TUNNEL   /**< RTSP over HTTP (tunneling) */
70};
71
72#define RTSP_DEFAULT_PORT   554
73#define RTSP_MAX_TRANSPORTS 8
74#define RTSP_TCP_MAX_PACKET_SIZE 1472
75#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
76#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
77#define RTSP_RTP_PORT_MIN 5000
78#define RTSP_RTP_PORT_MAX 65000
79
80/**
81 * This describes a single item in the "Transport:" line of one stream as
82 * negotiated by the SETUP RTSP command. Multiple transports are comma-
83 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
84 * client_port=1000-1001;server_port=1800-1801") and described in separate
85 * RTSPTransportFields.
86 */
87typedef struct RTSPTransportField {
88    /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
89     * with a '$', stream length and stream ID. If the stream ID is within
90     * the range of this interleaved_min-max, then the packet belongs to
91     * this stream. */
92    int interleaved_min, interleaved_max;
93
94    /** UDP multicast port range; the ports to which we should connect to
95     * receive multicast UDP data. */
96    int port_min, port_max;
97
98    /** UDP client ports; these should be the local ports of the UDP RTP
99     * (and RTCP) sockets over which we receive RTP/RTCP data. */
100    int client_port_min, client_port_max;
101
102    /** UDP unicast server port range; the ports to which we should connect
103     * to receive unicast UDP RTP/RTCP data. */
104    int server_port_min, server_port_max;
105
106    /** time-to-live value (required for multicast); the amount of HOPs that
107     * packets will be allowed to make before being discarded. */
108    int ttl;
109
110    /** transport set to record data */
111    int mode_record;
112
113    struct sockaddr_storage destination; /**< destination IP address */
114    char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
115
116    /** data/packet transport protocol; e.g. RTP or RDT */
117    enum RTSPTransport transport;
118
119    /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
120    enum RTSPLowerTransport lower_transport;
121} RTSPTransportField;
122
123/**
124 * This describes the server response to each RTSP command.
125 */
126typedef struct RTSPMessageHeader {
127    /** length of the data following this header */
128    int content_length;
129
130    enum RTSPStatusCode status_code; /**< response code from server */
131
132    /** number of items in the 'transports' variable below */
133    int nb_transports;
134
135    /** Time range of the streams that the server will stream. In
136     * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
137    int64_t range_start, range_end;
138
139    /** describes the complete "Transport:" line of the server in response
140     * to a SETUP RTSP command by the client */
141    RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
142
143    int seq;                         /**< sequence number */
144
145    /** the "Session:" field. This value is initially set by the server and
146     * should be re-transmitted by the client in every RTSP command. */
147    char session_id[512];
148
149    /** the "Location:" field. This value is used to handle redirection.
150     */
151    char location[4096];
152
153    /** the "RealChallenge1:" field from the server */
154    char real_challenge[64];
155
156    /** the "Server: field, which can be used to identify some special-case
157     * servers that are not 100% standards-compliant. We use this to identify
158     * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
159     * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
160     * use something like "Helix [..] Server Version v.e.r.sion (platform)
161     * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
162     * where platform is the output of $uname -msr | sed 's/ /-/g'. */
163    char server[64];
164
165    /** The "timeout" comes as part of the server response to the "SETUP"
166     * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
167     * time, in seconds, that the server will go without traffic over the
168     * RTSP/TCP connection before it closes the connection. To prevent
169     * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
170     * than this value. */
171    int timeout;
172
173    /** The "Notice" or "X-Notice" field value. See
174     * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
175     * for a complete list of supported values. */
176    int notice;
177
178    /** The "reason" is meant to specify better the meaning of the error code
179     * returned
180     */
181    char reason[256];
182
183    /**
184     * Content type header
185     */
186    char content_type[64];
187} RTSPMessageHeader;
188
189/**
190 * Client state, i.e. whether we are currently receiving data (PLAYING) or
191 * setup-but-not-receiving (PAUSED). State can be changed in applications
192 * by calling av_read_play/pause().
193 */
194enum RTSPClientState {
195    RTSP_STATE_IDLE,    /**< not initialized */
196    RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
197    RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
198    RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
199};
200
201/**
202 * Identify particular servers that require special handling, such as
203 * standards-incompliant "Transport:" lines in the SETUP request.
204 */
205enum RTSPServerType {
206    RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
207    RTSP_SERVER_REAL, /**< Realmedia-style server */
208    RTSP_SERVER_WMS,  /**< Windows Media server */
209    RTSP_SERVER_NB
210};
211
212/**
213 * Private data for the RTSP demuxer.
214 *
215 * @todo Use AVIOContext instead of URLContext
216 */
217typedef struct RTSPState {
218    const AVClass *class;             /**< Class for private options. */
219    URLContext *rtsp_hd; /* RTSP TCP connection handle */
220
221    /** number of items in the 'rtsp_streams' variable */
222    int nb_rtsp_streams;
223
224    struct RTSPStream **rtsp_streams; /**< streams in this session */
225
226    /** indicator of whether we are currently receiving data from the
227     * server. Basically this isn't more than a simple cache of the
228     * last PLAY/PAUSE command sent to the server, to make sure we don't
229     * send 2x the same unexpectedly or commands in the wrong state. */
230    enum RTSPClientState state;
231
232    /** the seek value requested when calling av_seek_frame(). This value
233     * is subsequently used as part of the "Range" parameter when emitting
234     * the RTSP PLAY command. If we are currently playing, this command is
235     * called instantly. If we are currently paused, this command is called
236     * whenever we resume playback. Either way, the value is only used once,
237     * see rtsp_read_play() and rtsp_read_seek(). */
238    int64_t seek_timestamp;
239
240    int seq;                          /**< RTSP command sequence number */
241
242    /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
243     * identifier that the client should re-transmit in each RTSP command */
244    char session_id[512];
245
246    /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
247     * the server will go without traffic on the RTSP/TCP line before it
248     * closes the connection. */
249    int timeout;
250
251    /** timestamp of the last RTSP command that we sent to the RTSP server.
252     * This is used to calculate when to send dummy commands to keep the
253     * connection alive, in conjunction with timeout. */
254    int64_t last_cmd_time;
255
256    /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
257    enum RTSPTransport transport;
258
259    /** the negotiated network layer transport protocol; e.g. TCP or UDP
260     * uni-/multicast */
261    enum RTSPLowerTransport lower_transport;
262
263    /** brand of server that we're talking to; e.g. WMS, REAL or other.
264     * Detected based on the value of RTSPMessageHeader->server or the presence
265     * of RTSPMessageHeader->real_challenge */
266    enum RTSPServerType server_type;
267
268    /** the "RealChallenge1:" field from the server */
269    char real_challenge[64];
270
271    /** plaintext authorization line (username:password) */
272    char auth[128];
273
274    /** authentication state */
275    HTTPAuthState auth_state;
276
277    /** The last reply of the server to a RTSP command */
278    char last_reply[2048]; /* XXX: allocate ? */
279
280    /** RTSPStream->transport_priv of the last stream that we read a
281     * packet from */
282    void *cur_transport_priv;
283
284    /** The following are used for Real stream selection */
285    //@{
286    /** whether we need to send a "SET_PARAMETER Subscribe:" command */
287    int need_subscription;
288
289    /** stream setup during the last frame read. This is used to detect if
290     * we need to subscribe or unsubscribe to any new streams. */
291    enum AVDiscard *real_setup_cache;
292
293    /** current stream setup. This is a temporary buffer used to compare
294     * current setup to previous frame setup. */
295    enum AVDiscard *real_setup;
296
297    /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
298     * this is used to send the same "Unsubscribe:" if stream setup changed,
299     * before sending a new "Subscribe:" command. */
300    char last_subscription[1024];
301    //@}
302
303    /** The following are used for RTP/ASF streams */
304    //@{
305    /** ASF demuxer context for the embedded ASF stream from WMS servers */
306    AVFormatContext *asf_ctx;
307
308    /** cache for position of the asf demuxer, since we load a new
309     * data packet in the bytecontext for each incoming RTSP packet. */
310    uint64_t asf_pb_pos;
311    //@}
312
313    /** some MS RTSP streams contain a URL in the SDP that we need to use
314     * for all subsequent RTSP requests, rather than the input URI; in
315     * other cases, this is a copy of AVFormatContext->filename. */
316    char control_uri[1024];
317
318    /** The following are used for parsing raw mpegts in udp */
319    //@{
320    struct MpegTSContext *ts;
321    int recvbuf_pos;
322    int recvbuf_len;
323    //@}
324
325    /** Additional output handle, used when input and output are done
326     * separately, eg for HTTP tunneling. */
327    URLContext *rtsp_hd_out;
328
329    /** RTSP transport mode, such as plain or tunneled. */
330    enum RTSPControlTransport control_transport;
331
332    /* Number of RTCP BYE packets the RTSP session has received.
333     * An EOF is propagated back if nb_byes == nb_streams.
334     * This is reset after a seek. */
335    int nb_byes;
336
337    /** Reusable buffer for receiving packets */
338    uint8_t* recvbuf;
339
340    /**
341     * A mask with all requested transport methods
342     */
343    int lower_transport_mask;
344
345    /**
346     * The number of returned packets
347     */
348    uint64_t packets;
349
350    /**
351     * Polling array for udp
352     */
353    struct pollfd *p;
354
355    /**
356     * Whether the server supports the GET_PARAMETER method.
357     */
358    int get_parameter_supported;
359
360    /**
361     * Do not begin to play the stream immediately.
362     */
363    int initial_pause;
364
365    /**
366     * Option flags for the chained RTP muxer.
367     */
368    int rtp_muxer_flags;
369
370    /** Whether the server accepts the x-Dynamic-Rate header */
371    int accept_dynamic_rate;
372
373    /**
374     * Various option flags for the RTSP muxer/demuxer.
375     */
376    int rtsp_flags;
377
378    /**
379     * Mask of all requested media types
380     */
381    int media_type_mask;
382
383    /**
384     * Minimum and maximum local UDP ports.
385     */
386    int rtp_port_min, rtp_port_max;
387
388    /**
389     * Timeout to wait for incoming connections.
390     */
391    int initial_timeout;
392
393    /**
394     * timeout of socket i/o operations.
395     */
396    int stimeout;
397
398    /**
399     * Size of RTP packet reordering queue.
400     */
401    int reordering_queue_size;
402
403    /**
404     * User-Agent string
405     */
406    char *user_agent;
407} RTSPState;
408
409#define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
410                                          receive packets only from the right
411                                          source address and port. */
412#define RTSP_FLAG_LISTEN      0x2    /**< Wait for incoming connections. */
413#define RTSP_FLAG_CUSTOM_IO   0x4    /**< Do all IO via the AVIOContext. */
414#define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
415                                          address of received packets. */
416#define RTSP_FLAG_PREFER_TCP  0x10   /**< Try RTP via TCP first if possible. */
417
418typedef struct RTSPSource {
419    char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
420} RTSPSource;
421
422/**
423 * Describe a single stream, as identified by a single m= line block in the
424 * SDP content. In the case of RDT, one RTSPStream can represent multiple
425 * AVStreams. In this case, each AVStream in this set has similar content
426 * (but different codec/bitrate).
427 */
428typedef struct RTSPStream {
429    URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
430    void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
431
432    /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
433    int stream_index;
434
435    /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
436     * for the selected transport. Only used for TCP. */
437    int interleaved_min, interleaved_max;
438
439    char control_url[1024];   /**< url for this stream (from SDP) */
440
441    /** The following are used only in SDP, not RTSP */
442    //@{
443    int sdp_port;             /**< port (from SDP content) */
444    struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
445    int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
446    struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
447    int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
448    struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
449    int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
450    int sdp_payload_type;     /**< payload type */
451    //@}
452
453    /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
454    //@{
455    /** handler structure */
456    RTPDynamicProtocolHandler *dynamic_handler;
457
458    /** private data associated with the dynamic protocol */
459    PayloadContext *dynamic_protocol_context;
460    //@}
461
462    /** Enable sending RTCP feedback messages according to RFC 4585 */
463    int feedback;
464
465    char crypto_suite[40];
466    char crypto_params[100];
467} RTSPStream;
468
469void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
470                        RTSPState *rt, const char *method);
471
472/**
473 * Send a command to the RTSP server without waiting for the reply.
474 *
475 * @see rtsp_send_cmd_with_content_async
476 */
477int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
478                           const char *url, const char *headers);
479
480/**
481 * Send a command to the RTSP server and wait for the reply.
482 *
483 * @param s RTSP (de)muxer context
484 * @param method the method for the request
485 * @param url the target url for the request
486 * @param headers extra header lines to include in the request
487 * @param reply pointer where the RTSP message header will be stored
488 * @param content_ptr pointer where the RTSP message body, if any, will
489 *                    be stored (length is in reply)
490 * @param send_content if non-null, the data to send as request body content
491 * @param send_content_length the length of the send_content data, or 0 if
492 *                            send_content is null
493 *
494 * @return zero if success, nonzero otherwise
495 */
496int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
497                                  const char *method, const char *url,
498                                  const char *headers,
499                                  RTSPMessageHeader *reply,
500                                  unsigned char **content_ptr,
501                                  const unsigned char *send_content,
502                                  int send_content_length);
503
504/**
505 * Send a command to the RTSP server and wait for the reply.
506 *
507 * @see rtsp_send_cmd_with_content
508 */
509int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
510                     const char *url, const char *headers,
511                     RTSPMessageHeader *reply, unsigned char **content_ptr);
512
513/**
514 * Read a RTSP message from the server, or prepare to read data
515 * packets if we're reading data interleaved over the TCP/RTSP
516 * connection as well.
517 *
518 * @param s RTSP (de)muxer context
519 * @param reply pointer where the RTSP message header will be stored
520 * @param content_ptr pointer where the RTSP message body, if any, will
521 *                    be stored (length is in reply)
522 * @param return_on_interleaved_data whether the function may return if we
523 *                   encounter a data marker ('$'), which precedes data
524 *                   packets over interleaved TCP/RTSP connections. If this
525 *                   is set, this function will return 1 after encountering
526 *                   a '$'. If it is not set, the function will skip any
527 *                   data packets (if they are encountered), until a reply
528 *                   has been fully parsed. If no more data is available
529 *                   without parsing a reply, it will return an error.
530 * @param method the RTSP method this is a reply to. This affects how
531 *               some response headers are acted upon. May be NULL.
532 *
533 * @return 1 if a data packets is ready to be received, -1 on error,
534 *          and 0 on success.
535 */
536int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
537                       unsigned char **content_ptr,
538                       int return_on_interleaved_data, const char *method);
539
540/**
541 * Skip a RTP/TCP interleaved packet.
542 */
543void ff_rtsp_skip_packet(AVFormatContext *s);
544
545/**
546 * Connect to the RTSP server and set up the individual media streams.
547 * This can be used for both muxers and demuxers.
548 *
549 * @param s RTSP (de)muxer context
550 *
551 * @return 0 on success, < 0 on error. Cleans up all allocations done
552 *          within the function on error.
553 */
554int ff_rtsp_connect(AVFormatContext *s);
555
556/**
557 * Close and free all streams within the RTSP (de)muxer
558 *
559 * @param s RTSP (de)muxer context
560 */
561void ff_rtsp_close_streams(AVFormatContext *s);
562
563/**
564 * Close all connection handles within the RTSP (de)muxer
565 *
566 * @param s RTSP (de)muxer context
567 */
568void ff_rtsp_close_connections(AVFormatContext *s);
569
570/**
571 * Get the description of the stream and set up the RTSPStream child
572 * objects.
573 */
574int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
575
576/**
577 * Announce the stream to the server and set up the RTSPStream child
578 * objects for each media stream.
579 */
580int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
581
582/**
583 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
584 * listen mode.
585 */
586int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
587
588/**
589 * Parse an SDP description of streams by populating an RTSPState struct
590 * within the AVFormatContext; also allocate the RTP streams and the
591 * pollfd array used for UDP streams.
592 */
593int ff_sdp_parse(AVFormatContext *s, const char *content);
594
595/**
596 * Receive one RTP packet from an TCP interleaved RTSP stream.
597 */
598int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
599                            uint8_t *buf, int buf_size);
600
601/**
602 * Send buffered packets over TCP.
603 */
604int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
605
606/**
607 * Receive one packet from the RTSPStreams set up in the AVFormatContext
608 * (which should contain a RTSPState struct as priv_data).
609 */
610int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
611
612/**
613 * Do the SETUP requests for each stream for the chosen
614 * lower transport mode.
615 * @return 0 on success, <0 on error, 1 if protocol is unavailable
616 */
617int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
618                               int lower_transport, const char *real_challenge);
619
620/**
621 * Undo the effect of ff_rtsp_make_setup_request, close the
622 * transport_priv and rtp_handle fields.
623 */
624void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
625
626/**
627 * Open RTSP transport context.
628 */
629int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
630
631extern const AVOption ff_rtsp_options[];
632
633#endif /* AVFORMAT_RTSP_H */
634