libavformat/rtsp.h

   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 #include "os_support.h"
  32
  33 #include "libavutil/log.h"
  34 #include "libavutil/opt.h"
  35
  36 /**
  37  * Network layer over which RTP/etc packet data will be transported.
  38  */
  39 enum RTSPLowerTransport {
  40     RTSP_LOWER_TRANSPORT_UDP = 0,           /**< UDP/unicast */
  41     RTSP_LOWER_TRANSPORT_TCP = 1,           /**< TCP; interleaved in RTSP */
  42     RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
  43     RTSP_LOWER_TRANSPORT_NB,
  44     RTSP_LOWER_TRANSPORT_HTTP = 8,          /**< HTTP tunneled - not a proper
  45                                                  transport mode as such,
  46                                                  only for use via AVOptions */
  47     RTSP_LOWER_TRANSPORT_HTTPS,             /**< HTTPS tunneled */
  48     RTSP_LOWER_TRANSPORT_CUSTOM = 16,       /**< Custom IO - not a public
  49                                                  option for lower_transport_mask,
  50                                                  but set in the SDP demuxer based
  51                                                  on a flag. */
  52 };
  53
  54 /**
  55  * Packet profile of the data that we will be receiving. Real servers
  56  * commonly send RDT (although they can sometimes send RTP as well),
  57  * whereas most others will send RTP.
  58  */
  59 enum RTSPTransport {
  60     RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
  61     RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
  62     RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
  63     RTSP_TRANSPORT_NB
  64 };
  65
  66 /**
  67  * Transport mode for the RTSP data. This may be plain, or
  68  * tunneled, which is done over HTTP.
  69  */
  70 enum RTSPControlTransport {
  71     RTSP_MODE_PLAIN,   /**< Normal RTSP */
  72     RTSP_MODE_TUNNEL   /**< RTSP over HTTP (tunneling) */
  73 };
  74
  75 #define RTSP_DEFAULT_PORT   554
  76 #define RTSPS_DEFAULT_PORT  322
  77 #define RTSP_MAX_TRANSPORTS 8
  78 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
  79 #define RTSP_RTP_PORT_MIN 5000
  80 #define RTSP_RTP_PORT_MAX 65000
  81 #define SDP_MAX_SIZE 16384
  82
  83 /**
  84  * This describes a single item in the "Transport:" line of one stream as
  85  * negotiated by the SETUP RTSP command. Multiple transports are comma-
  86  * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
  87  * client_port=1000-1001;server_port=1800-1801") and described in separate
  88  * RTSPTransportFields.
  89  */
  90 typedef struct RTSPTransportField {
  91     /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
  92      * with a '$', stream length and stream ID. If the stream ID is within
  93      * the range of this interleaved_min-max, then the packet belongs to
  94      * this stream. */
  95     int interleaved_min, interleaved_max;
  96
  97     /** UDP multicast port range; the ports to which we should connect to
  98      * receive multicast UDP data. */
  99     int port_min, port_max;
 100
 101     /** UDP client ports; these should be the local ports of the UDP RTP
 102      * (and RTCP) sockets over which we receive RTP/RTCP data. */
 103     int client_port_min, client_port_max;
 104
 105     /** UDP unicast server port range; the ports to which we should connect
 106      * to receive unicast UDP RTP/RTCP data. */
 107     int server_port_min, server_port_max;
 108
 109     /** time-to-live value (required for multicast); the amount of HOPs that
 110      * packets will be allowed to make before being discarded. */
 111     int ttl;
 112
 113     /** transport set to record data */
 114     int mode_record;
 115
 116     struct sockaddr_storage destination; /**< destination IP address */
 117     char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
 118
 119     /** data/packet transport protocol; e.g. RTP or RDT */
 120     enum RTSPTransport transport;
 121
 122     /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
 123     enum RTSPLowerTransport lower_transport;
 124 } RTSPTransportField;
 125
 126 /**
 127  * This describes the server response to each RTSP command.
 128  */
 129 typedef struct RTSPMessageHeader {
 130     /** length of the data following this header */
 131     int content_length;
 132
 133     enum RTSPStatusCode status_code; /**< response code from server */
 134
 135     /** number of items in the 'transports' variable below */
 136     int nb_transports;
 137
 138     /** Time range of the streams that the server will stream. In
 139      * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
 140     int64_t range_start, range_end;
 141
 142     /** describes the complete "Transport:" line of the server in response
 143      * to a SETUP RTSP command by the client */
 144     RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
 145
 146     int seq;                         /**< sequence number */
 147
 148     /** the "Session:" field. This value is initially set by the server and
 149      * should be re-transmitted by the client in every RTSP command. */
 150     char session_id[512];
 151
 152     /** the "Location:" field. This value is used to handle redirection.
 153      */
 154     char location[4096];
 155
 156     /** the "RealChallenge1:" field from the server */
 157     char real_challenge[64];
 158
 159     /** the "Server: field, which can be used to identify some special-case
 160      * servers that are not 100% standards-compliant. We use this to identify
 161      * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
 162      * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
 163      * use something like "Helix [..] Server Version v.e.r.sion (platform)
 164      * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
 165      * where platform is the output of $uname -msr | sed 's/ /-/g'. */
 166     char server[64];
 167
 168     /** The "timeout" comes as part of the server response to the "SETUP"
 169      * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
 170      * time, in seconds, that the server will go without traffic over the
 171      * RTSP/TCP connection before it closes the connection. To prevent
 172      * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
 173      * than this value. */
 174     int timeout;
 175
 176     /** The "Notice" or "X-Notice" field value. See
 177      * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
 178      * for a complete list of supported values. */
 179     int notice;
 180
 181     /** The "reason" is meant to specify better the meaning of the error code
 182      * returned
 183      */
 184     char reason[256];
 185
 186     /**
 187      * Content type header
 188      */
 189     char content_type[64];
 190
 191     /**
 192      * SAT>IP com.ses.streamID header
 193      */
 194     char stream_id[64];
 195 } RTSPMessageHeader;
 196
 197 /**
 198  * Client state, i.e. whether we are currently receiving data (PLAYING) or
 199  * setup-but-not-receiving (PAUSED). State can be changed in applications
 200  * by calling av_read_play/pause().
 201  */
 202 enum RTSPClientState {
 203     RTSP_STATE_IDLE,    /**< not initialized */
 204     RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
 205     RTSP_STATE_PAUSED,  /**< initialized, but not receiving data */
 206     RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
 207 };
 208
 209 /**
 210  * Identify particular servers that require special handling, such as
 211  * standards-incompliant "Transport:" lines in the SETUP request.
 212  */
 213 enum RTSPServerType {
 214     RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
 215     RTSP_SERVER_REAL, /**< Realmedia-style server */
 216     RTSP_SERVER_WMS,  /**< Windows Media server */
 217     RTSP_SERVER_SATIP,/**< SAT>IP server */
 218     RTSP_SERVER_NB
 219 };
 220
 221 /**
 222  * Private data for the RTSP demuxer.
 223  *
 224  * @todo Use AVIOContext instead of URLContext
 225  */
 226 typedef struct RTSPState {
 227     const AVClass *class;             /**< Class for private options. */
 228     URLContext *rtsp_hd; /* RTSP TCP connection handle */
 229
 230     /** number of items in the 'rtsp_streams' variable */
 231     int nb_rtsp_streams;
 232
 233     struct RTSPStream **rtsp_streams; /**< streams in this session */
 234
 235     /** indicator of whether we are currently receiving data from the
 236      * server. Basically this isn't more than a simple cache of the
 237      * last PLAY/PAUSE command sent to the server, to make sure we don't
 238      * send 2x the same unexpectedly or commands in the wrong state. */
 239     enum RTSPClientState state;
 240
 241     /** the seek value requested when calling av_seek_frame(). This value
 242      * is subsequently used as part of the "Range" parameter when emitting
 243      * the RTSP PLAY command. If we are currently playing, this command is
 244      * called instantly. If we are currently paused, this command is called
 245      * whenever we resume playback. Either way, the value is only used once,
 246      * see rtsp_read_play() and rtsp_read_seek(). */
 247     int64_t seek_timestamp;
 248
 249     int seq;                          /**< RTSP command sequence number */
 250
 251     /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
 252      * identifier that the client should re-transmit in each RTSP command */
 253     char session_id[512];
 254
 255     /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
 256      * the server will go without traffic on the RTSP/TCP line before it
 257      * closes the connection. */
 258     int timeout;
 259
 260     /** timestamp of the last RTSP command that we sent to the RTSP server.
 261      * This is used to calculate when to send dummy commands to keep the
 262      * connection alive, in conjunction with timeout. */
 263     int64_t last_cmd_time;
 264
 265     /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
 266     enum RTSPTransport transport;
 267
 268     /** the negotiated network layer transport protocol; e.g. TCP or UDP
 269      * uni-/multicast */
 270     enum RTSPLowerTransport lower_transport;
 271
 272     /** brand of server that we're talking to; e.g. WMS, REAL or other.
 273      * Detected based on the value of RTSPMessageHeader->server or the presence
 274      * of RTSPMessageHeader->real_challenge */
 275     enum RTSPServerType server_type;
 276
 277     /** the "RealChallenge1:" field from the server */
 278     char real_challenge[64];
 279
 280     /** plaintext authorization line (username:password) */
 281     char auth[128];
 282
 283     /** authentication state */
 284     HTTPAuthState auth_state;
 285
 286     /** The last reply of the server to a RTSP command */
 287     char last_reply[2048]; /* XXX: allocate ? */
 288
 289     /** RTSPStream->transport_priv of the last stream that we read a
 290      * packet from */
 291     void *cur_transport_priv;
 292
 293     /** The following are used for Real stream selection */
 294     //@{
 295     /** whether we need to send a "SET_PARAMETER Subscribe:" command */
 296     int need_subscription;
 297
 298     /** stream setup during the last frame read. This is used to detect if
 299      * we need to subscribe or unsubscribe to any new streams. */
 300     enum AVDiscard *real_setup_cache;
 301
 302     /** current stream setup. This is a temporary buffer used to compare
 303      * current setup to previous frame setup. */
 304     enum AVDiscard *real_setup;
 305
 306     /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
 307      * this is used to send the same "Unsubscribe:" if stream setup changed,
 308      * before sending a new "Subscribe:" command. */
 309     char last_subscription[1024];
 310     //@}
 311
 312     /** The following are used for RTP/ASF streams */
 313     //@{
 314     /** ASF demuxer context for the embedded ASF stream from WMS servers */
 315     AVFormatContext *asf_ctx;
 316
 317     /** cache for position of the asf demuxer, since we load a new
 318      * data packet in the bytecontext for each incoming RTSP packet. */
 319     uint64_t asf_pb_pos;
 320     //@}
 321
 322     /** some MS RTSP streams contain a URL in the SDP that we need to use
 323      * for all subsequent RTSP requests, rather than the input URI; in
 324      * other cases, this is a copy of AVFormatContext->filename. */
 325     char control_uri[MAX_URL_SIZE];
 326
 327     /** The following are used for parsing raw mpegts in udp */
 328     //@{
 329     struct MpegTSContext *ts;
 330     int recvbuf_pos;
 331     int recvbuf_len;
 332     //@}
 333
 334     /** Additional output handle, used when input and output are done
 335      * separately, eg for HTTP tunneling. */
 336     URLContext *rtsp_hd_out;
 337
 338     /** RTSP transport mode, such as plain or tunneled. */
 339     enum RTSPControlTransport control_transport;
 340
 341     /* Number of RTCP BYE packets the RTSP session has received.
 342      * An EOF is propagated back if nb_byes == nb_streams.
 343      * This is reset after a seek. */
 344     int nb_byes;
 345
 346     /** Reusable buffer for receiving packets */
 347     uint8_t* recvbuf;
 348
 349     /**
 350      * A mask with all requested transport methods
 351      */
 352     int lower_transport_mask;
 353
 354     /**
 355      * The number of returned packets
 356      */
 357     uint64_t packets;
 358
 359     /**
 360      * Polling array for udp
 361      */
 362     struct pollfd *p;
 363     int max_p;
 364
 365     /**
 366      * Whether the server supports the GET_PARAMETER method.
 367      */
 368     int get_parameter_supported;
 369
 370     /**
 371      * Do not begin to play the stream immediately.
 372      */
 373     int initial_pause;
 374
 375     /**
 376      * Option flags for the chained RTP muxer.
 377      */
 378     int rtp_muxer_flags;
 379
 380     /** Whether the server accepts the x-Dynamic-Rate header */
 381     int accept_dynamic_rate;
 382
 383     /**
 384      * Various option flags for the RTSP muxer/demuxer.
 385      */
 386     int rtsp_flags;
 387
 388     /**
 389      * Mask of all requested media types
 390      */
 391     int media_type_mask;
 392
 393     /**
 394      * Minimum and maximum local UDP ports.
 395      */
 396     int rtp_port_min, rtp_port_max;
 397
 398     /**
 399      * Timeout to wait for incoming connections.
 400      */
 401     int initial_timeout;
 402
 403     /**
 404      * timeout of socket i/o operations.
 405      */
 406     int64_t stimeout;
 407
 408     /**
 409      * Size of RTP packet reordering queue.
 410      */
 411     int reordering_queue_size;
 412
 413     /**
 414      * User-Agent string
 415      */
 416     char *user_agent;
 417
 418     char default_lang[4];
 419     int buffer_size;
 420     int pkt_size;
 421     char *localaddr;
 422
 423     /**
 424      * Options used for TLS based RTSP streams.
 425      */
 426     struct {
 427         char *ca_file;
 428         int verify;
 429         char *cert_file;
 430         char *key_file;
 431         char *host;
 432     } tls_opts;
 433 } RTSPState;
 434
 435 #define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
 436                                           receive packets only from the right
 437                                           source address and port. */
 438 #define RTSP_FLAG_LISTEN      0x2    /**< Wait for incoming connections. */
 439 #define RTSP_FLAG_CUSTOM_IO   0x4    /**< Do all IO via the AVIOContext. */
 440 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
 441                                           address of received packets. */
 442 #define RTSP_FLAG_PREFER_TCP  0x10   /**< Try RTP via TCP first if possible. */
 443 #define RTSP_FLAG_SATIP_RAW   0x20   /**< Export SAT>IP stream as raw MPEG-TS */
 444
 445 typedef struct RTSPSource {
 446     char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
 447 } RTSPSource;
 448
 449 /**
 450  * Describe a single stream, as identified by a single m= line block in the
 451  * SDP content. In the case of RDT, one RTSPStream can represent multiple
 452  * AVStreams. In this case, each AVStream in this set has similar content
 453  * (but different codec/bitrate).
 454  */
 455 typedef struct RTSPStream {
 456     URLContext *rtp_handle;   /**< RTP stream handle (if UDP) */
 457     void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
 458
 459     /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
 460     int stream_index;
 461
 462     /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
 463      * for the selected transport. Only used for TCP. */
 464     int interleaved_min, interleaved_max;
 465
 466     char control_url[MAX_URL_SIZE];   /**< url for this stream (from SDP) */
 467
 468     /** The following are used only in SDP, not RTSP */
 469     //@{
 470     int sdp_port;             /**< port (from SDP content) */
 471     struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
 472     int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
 473     struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
 474     int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
 475     struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
 476     int sdp_ttl;              /**< IP Time-To-Live (from SDP content) */
 477     int sdp_payload_type;     /**< payload type */
 478     //@}
 479
 480     /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
 481     //@{
 482     /** handler structure */
 483     const RTPDynamicProtocolHandler *dynamic_handler;
 484
 485     /** private data associated with the dynamic protocol */
 486     PayloadContext *dynamic_protocol_context;
 487     //@}
 488
 489     /** Enable sending RTCP feedback messages according to RFC 4585 */
 490     int feedback;
 491
 492     /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
 493     uint32_t ssrc;
 494
 495     char crypto_suite[40];
 496     char crypto_params[100];
 497 } RTSPStream;
 498
 499 void ff_rtsp_parse_line(AVFormatContext *s,
 500                         RTSPMessageHeader *reply, const char *buf,
 501                         RTSPState *rt, const char *method);
 502
 503 /**
 504  * Send a command to the RTSP server without waiting for the reply.
 505  *
 506  * @see rtsp_send_cmd_with_content_async
 507  */
 508 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
 509                            const char *url, const char *headers);
 510
 511 /**
 512  * Send a command to the RTSP server and wait for the reply.
 513  *
 514  * @param s RTSP (de)muxer context
 515  * @param method the method for the request
 516  * @param url the target url for the request
 517  * @param headers extra header lines to include in the request
 518  * @param reply pointer where the RTSP message header will be stored
 519  * @param content_ptr pointer where the RTSP message body, if any, will
 520  *                    be stored (length is in reply)
 521  * @param send_content if non-null, the data to send as request body content
 522  * @param send_content_length the length of the send_content data, or 0 if
 523  *                            send_content is null
 524  *
 525  * @return zero if success, nonzero otherwise
 526  */
 527 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
 528                                   const char *method, const char *url,
 529                                   const char *headers,
 530                                   RTSPMessageHeader *reply,
 531                                   unsigned char **content_ptr,
 532                                   const unsigned char *send_content,
 533                                   int send_content_length);
 534
 535 /**
 536  * Send a command to the RTSP server and wait for the reply.
 537  *
 538  * @see rtsp_send_cmd_with_content
 539  */
 540 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
 541                      const char *url, const char *headers,
 542                      RTSPMessageHeader *reply, unsigned char **content_ptr);
 543
 544 /**
 545  * Read a RTSP message from the server, or prepare to read data
 546  * packets if we're reading data interleaved over the TCP/RTSP
 547  * connection as well.
 548  *
 549  * @param s RTSP (de)muxer context
 550  * @param reply pointer where the RTSP message header will be stored
 551  * @param content_ptr pointer where the RTSP message body, if any, will
 552  *                    be stored (length is in reply)
 553  * @param return_on_interleaved_data whether the function may return if we
 554  *                   encounter a data marker ('$'), which precedes data
 555  *                   packets over interleaved TCP/RTSP connections. If this
 556  *                   is set, this function will return 1 after encountering
 557  *                   a '$'. If it is not set, the function will skip any
 558  *                   data packets (if they are encountered), until a reply
 559  *                   has been fully parsed. If no more data is available
 560  *                   without parsing a reply, it will return an error.
 561  * @param method the RTSP method this is a reply to. This affects how
 562  *               some response headers are acted upon. May be NULL.
 563  *
 564  * @return 1 if a data packets is ready to be received, -1 on error,
 565  *          and 0 on success.
 566  */
 567 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
 568                        unsigned char **content_ptr,
 569                        int return_on_interleaved_data, const char *method);
 570
 571 /**
 572  * Skip a RTP/TCP interleaved packet.
 573  *
 574  * @return 0 on success, < 0 on failure.
 575  */
 576 int ff_rtsp_skip_packet(AVFormatContext *s);
 577
 578 /**
 579  * Connect to the RTSP server and set up the individual media streams.
 580  * This can be used for both muxers and demuxers.
 581  *
 582  * @param s RTSP (de)muxer context
 583  *
 584  * @return 0 on success, < 0 on error. Cleans up all allocations done
 585  *          within the function on error.
 586  */
 587 int ff_rtsp_connect(AVFormatContext *s);
 588
 589 /**
 590  * Close and free all streams within the RTSP (de)muxer
 591  *
 592  * @param s RTSP (de)muxer context
 593  */
 594 void ff_rtsp_close_streams(AVFormatContext *s);
 595
 596 /**
 597  * Close all connection handles within the RTSP (de)muxer
 598  *
 599  * @param s RTSP (de)muxer context
 600  */
 601 void ff_rtsp_close_connections(AVFormatContext *s);
 602
 603 /**
 604  * Get the description of the stream and set up the RTSPStream child
 605  * objects.
 606  */
 607 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
 608
 609 /**
 610  * Announce the stream to the server and set up the RTSPStream child
 611  * objects for each media stream.
 612  */
 613 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
 614
 615 /**
 616  * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
 617  * listen mode.
 618  */
 619 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
 620
 621 /**
 622  * Parse an SDP description of streams by populating an RTSPState struct
 623  * within the AVFormatContext; also allocate the RTP streams and the
 624  * pollfd array used for UDP streams.
 625  */
 626 int ff_sdp_parse(AVFormatContext *s, const char *content);
 627
 628 /**
 629  * Receive one RTP packet from an TCP interleaved RTSP stream.
 630  */
 631 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
 632                             uint8_t *buf, int buf_size);
 633
 634 /**
 635  * Send buffered packets over TCP.
 636  */
 637 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
 638
 639 /**
 640  * Receive one packet from the RTSPStreams set up in the AVFormatContext
 641  * (which should contain a RTSPState struct as priv_data).
 642  */
 643 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
 644
 645 /**
 646  * Do the SETUP requests for each stream for the chosen
 647  * lower transport mode.
 648  * @return 0 on success, <0 on error, 1 if protocol is unavailable
 649  */
 650 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
 651                                int lower_transport, const char *real_challenge);
 652
 653 /**
 654  * Undo the effect of ff_rtsp_make_setup_request, close the
 655  * transport_priv and rtp_handle fields.
 656  */
 657 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
 658
 659 /**
 660  * Open RTSP transport context.
 661  */
 662 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
 663
 664 extern const AVOption ff_rtsp_options[];
 665
 666 #endif /* AVFORMAT_RTSP_H */