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