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 */ 37 enum 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_HTTPS, /**< HTTPS tunneled */ 46 RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public 47 option for lower_transport_mask, 48 but set in the SDP demuxer based 49 on a flag. */ 50 }; 51 52 /** 53 * Packet profile of the data that we will be receiving. Real servers 54 * commonly send RDT (although they can sometimes send RTP as well), 55 * whereas most others will send RTP. 56 */ 57 enum RTSPTransport { 58 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */ 59 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */ 60 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */ 61 RTSP_TRANSPORT_NB 62 }; 63 64 /** 65 * Transport mode for the RTSP data. This may be plain, or 66 * tunneled, which is done over HTTP. 67 */ 68 enum RTSPControlTransport { 69 RTSP_MODE_PLAIN, /**< Normal RTSP */ 70 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */ 71 }; 72 73 #define RTSP_DEFAULT_PORT 554 74 #define RTSPS_DEFAULT_PORT 322 75 #define RTSP_MAX_TRANSPORTS 8 76 #define RTSP_TCP_MAX_PACKET_SIZE 1472 77 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1 78 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100 79 #define RTSP_RTP_PORT_MIN 5000 80 #define RTSP_RTP_PORT_MAX 65000 81 82 /** 83 * This describes a single item in the "Transport:" line of one stream as 84 * negotiated by the SETUP RTSP command. Multiple transports are comma- 85 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp; 86 * client_port=1000-1001;server_port=1800-1801") and described in separate 87 * RTSPTransportFields. 88 */ 89 typedef struct RTSPTransportField { 90 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts 91 * with a '$', stream length and stream ID. If the stream ID is within 92 * the range of this interleaved_min-max, then the packet belongs to 93 * this stream. */ 94 int interleaved_min, interleaved_max; 95 96 /** UDP multicast port range; the ports to which we should connect to 97 * receive multicast UDP data. */ 98 int port_min, port_max; 99 100 /** UDP client ports; these should be the local ports of the UDP RTP 101 * (and RTCP) sockets over which we receive RTP/RTCP data. */ 102 int client_port_min, client_port_max; 103 104 /** UDP unicast server port range; the ports to which we should connect 105 * to receive unicast UDP RTP/RTCP data. */ 106 int server_port_min, server_port_max; 107 108 /** time-to-live value (required for multicast); the amount of HOPs that 109 * packets will be allowed to make before being discarded. */ 110 int ttl; 111 112 /** transport set to record data */ 113 int mode_record; 114 115 struct sockaddr_storage destination; /**< destination IP address */ 116 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */ 117 118 /** data/packet transport protocol; e.g. RTP or RDT */ 119 enum RTSPTransport transport; 120 121 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */ 122 enum RTSPLowerTransport lower_transport; 123 } RTSPTransportField; 124 125 /** 126 * This describes the server response to each RTSP command. 127 */ 128 typedef struct RTSPMessageHeader { 129 /** length of the data following this header */ 130 int content_length; 131 132 enum RTSPStatusCode status_code; /**< response code from server */ 133 134 /** number of items in the 'transports' variable below */ 135 int nb_transports; 136 137 /** Time range of the streams that the server will stream. In 138 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */ 139 int64_t range_start, range_end; 140 141 /** describes the complete "Transport:" line of the server in response 142 * to a SETUP RTSP command by the client */ 143 RTSPTransportField transports[RTSP_MAX_TRANSPORTS]; 144 145 int seq; /**< sequence number */ 146 147 /** the "Session:" field. This value is initially set by the server and 148 * should be re-transmitted by the client in every RTSP command. */ 149 char session_id[512]; 150 151 /** the "Location:" field. This value is used to handle redirection. 152 */ 153 char location[4096]; 154 155 /** the "RealChallenge1:" field from the server */ 156 char real_challenge[64]; 157 158 /** the "Server: field, which can be used to identify some special-case 159 * servers that are not 100% standards-compliant. We use this to identify 160 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where 161 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers 162 * use something like "Helix [..] Server Version v.e.r.sion (platform) 163 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)", 164 * where platform is the output of $uname -msr | sed 's/ /-/g'. */ 165 char server[64]; 166 167 /** The "timeout" comes as part of the server response to the "SETUP" 168 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the 169 * time, in seconds, that the server will go without traffic over the 170 * RTSP/TCP connection before it closes the connection. To prevent 171 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller 172 * than this value. */ 173 int timeout; 174 175 /** The "Notice" or "X-Notice" field value. See 176 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00 177 * for a complete list of supported values. */ 178 int notice; 179 180 /** The "reason" is meant to specify better the meaning of the error code 181 * returned 182 */ 183 char reason[256]; 184 185 /** 186 * Content type header 187 */ 188 char content_type[64]; 189 } RTSPMessageHeader; 190 191 /** 192 * Client state, i.e. whether we are currently receiving data (PLAYING) or 193 * setup-but-not-receiving (PAUSED). State can be changed in applications 194 * by calling av_read_play/pause(). 195 */ 196 enum RTSPClientState { 197 RTSP_STATE_IDLE, /**< not initialized */ 198 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ 199 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ 200 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ 201 }; 202 203 /** 204 * Identify particular servers that require special handling, such as 205 * standards-incompliant "Transport:" lines in the SETUP request. 206 */ 207 enum RTSPServerType { 208 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */ 209 RTSP_SERVER_REAL, /**< Realmedia-style server */ 210 RTSP_SERVER_WMS, /**< Windows Media server */ 211 RTSP_SERVER_NB 212 }; 213 214 /** 215 * Private data for the RTSP demuxer. 216 * 217 * @todo Use AVIOContext instead of URLContext 218 */ 219 typedef struct RTSPState { 220 const AVClass *class; /**< Class for private options. */ 221 URLContext *rtsp_hd; /* RTSP TCP connection handle */ 222 223 /** number of items in the 'rtsp_streams' variable */ 224 int nb_rtsp_streams; 225 226 struct RTSPStream **rtsp_streams; /**< streams in this session */ 227 228 /** indicator of whether we are currently receiving data from the 229 * server. Basically this isn't more than a simple cache of the 230 * last PLAY/PAUSE command sent to the server, to make sure we don't 231 * send 2x the same unexpectedly or commands in the wrong state. */ 232 enum RTSPClientState state; 233 234 /** the seek value requested when calling av_seek_frame(). This value 235 * is subsequently used as part of the "Range" parameter when emitting 236 * the RTSP PLAY command. If we are currently playing, this command is 237 * called instantly. If we are currently paused, this command is called 238 * whenever we resume playback. Either way, the value is only used once, 239 * see rtsp_read_play() and rtsp_read_seek(). */ 240 int64_t seek_timestamp; 241 242 int seq; /**< RTSP command sequence number */ 243 244 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session 245 * identifier that the client should re-transmit in each RTSP command */ 246 char session_id[512]; 247 248 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that 249 * the server will go without traffic on the RTSP/TCP line before it 250 * closes the connection. */ 251 int timeout; 252 253 /** timestamp of the last RTSP command that we sent to the RTSP server. 254 * This is used to calculate when to send dummy commands to keep the 255 * connection alive, in conjunction with timeout. */ 256 int64_t last_cmd_time; 257 258 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */ 259 enum RTSPTransport transport; 260 261 /** the negotiated network layer transport protocol; e.g. TCP or UDP 262 * uni-/multicast */ 263 enum RTSPLowerTransport lower_transport; 264 265 /** brand of server that we're talking to; e.g. WMS, REAL or other. 266 * Detected based on the value of RTSPMessageHeader->server or the presence 267 * of RTSPMessageHeader->real_challenge */ 268 enum RTSPServerType server_type; 269 270 /** the "RealChallenge1:" field from the server */ 271 char real_challenge[64]; 272 273 /** plaintext authorization line (username:password) */ 274 char auth[128]; 275 276 /** authentication state */ 277 HTTPAuthState auth_state; 278 279 /** The last reply of the server to a RTSP command */ 280 char last_reply[2048]; /* XXX: allocate ? */ 281 282 /** RTSPStream->transport_priv of the last stream that we read a 283 * packet from */ 284 void *cur_transport_priv; 285 286 /** The following are used for Real stream selection */ 287 //@{ 288 /** whether we need to send a "SET_PARAMETER Subscribe:" command */ 289 int need_subscription; 290 291 /** stream setup during the last frame read. This is used to detect if 292 * we need to subscribe or unsubscribe to any new streams. */ 293 enum AVDiscard *real_setup_cache; 294 295 /** current stream setup. This is a temporary buffer used to compare 296 * current setup to previous frame setup. */ 297 enum AVDiscard *real_setup; 298 299 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command. 300 * this is used to send the same "Unsubscribe:" if stream setup changed, 301 * before sending a new "Subscribe:" command. */ 302 char last_subscription[1024]; 303 //@} 304 305 /** The following are used for RTP/ASF streams */ 306 //@{ 307 /** ASF demuxer context for the embedded ASF stream from WMS servers */ 308 AVFormatContext *asf_ctx; 309 310 /** cache for position of the asf demuxer, since we load a new 311 * data packet in the bytecontext for each incoming RTSP packet. */ 312 uint64_t asf_pb_pos; 313 //@} 314 315 /** some MS RTSP streams contain a URL in the SDP that we need to use 316 * for all subsequent RTSP requests, rather than the input URI; in 317 * other cases, this is a copy of AVFormatContext->filename. */ 318 char control_uri[1024]; 319 320 /** The following are used for parsing raw mpegts in udp */ 321 //@{ 322 struct MpegTSContext *ts; 323 int recvbuf_pos; 324 int recvbuf_len; 325 //@} 326 327 /** Additional output handle, used when input and output are done 328 * separately, eg for HTTP tunneling. */ 329 URLContext *rtsp_hd_out; 330 331 /** RTSP transport mode, such as plain or tunneled. */ 332 enum RTSPControlTransport control_transport; 333 334 /* Number of RTCP BYE packets the RTSP session has received. 335 * An EOF is propagated back if nb_byes == nb_streams. 336 * This is reset after a seek. */ 337 int nb_byes; 338 339 /** Reusable buffer for receiving packets */ 340 uint8_t* recvbuf; 341 342 /** 343 * A mask with all requested transport methods 344 */ 345 int lower_transport_mask; 346 347 /** 348 * The number of returned packets 349 */ 350 uint64_t packets; 351 352 /** 353 * Polling array for udp 354 */ 355 struct pollfd *p; 356 int max_p; 357 358 /** 359 * Whether the server supports the GET_PARAMETER method. 360 */ 361 int get_parameter_supported; 362 363 /** 364 * Do not begin to play the stream immediately. 365 */ 366 int initial_pause; 367 368 /** 369 * Option flags for the chained RTP muxer. 370 */ 371 int rtp_muxer_flags; 372 373 /** Whether the server accepts the x-Dynamic-Rate header */ 374 int accept_dynamic_rate; 375 376 /** 377 * Various option flags for the RTSP muxer/demuxer. 378 */ 379 int rtsp_flags; 380 381 /** 382 * Mask of all requested media types 383 */ 384 int media_type_mask; 385 386 /** 387 * Minimum and maximum local UDP ports. 388 */ 389 int rtp_port_min, rtp_port_max; 390 391 /** 392 * Timeout to wait for incoming connections. 393 */ 394 int initial_timeout; 395 396 /** 397 * timeout of socket i/o operations. 398 */ 399 int stimeout; 400 401 /** 402 * Size of RTP packet reordering queue. 403 */ 404 int reordering_queue_size; 405 406 /** 407 * User-Agent string 408 */ 409 char *user_agent; 410 411 char default_lang[4]; 412 int buffer_size; 413 int pkt_size; 414 } RTSPState; 415 416 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets - 417 receive packets only from the right 418 source address and port. */ 419 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */ 420 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */ 421 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source 422 address of received packets. */ 423 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */ 424 425 typedef struct RTSPSource { 426 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */ 427 } RTSPSource; 428 429 /** 430 * Describe a single stream, as identified by a single m= line block in the 431 * SDP content. In the case of RDT, one RTSPStream can represent multiple 432 * AVStreams. In this case, each AVStream in this set has similar content 433 * (but different codec/bitrate). 434 */ 435 typedef struct RTSPStream { 436 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */ 437 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */ 438 439 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */ 440 int stream_index; 441 442 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max 443 * for the selected transport. Only used for TCP. */ 444 int interleaved_min, interleaved_max; 445 446 char control_url[1024]; /**< url for this stream (from SDP) */ 447 448 /** The following are used only in SDP, not RTSP */ 449 //@{ 450 int sdp_port; /**< port (from SDP content) */ 451 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */ 452 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */ 453 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */ 454 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */ 455 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */ 456 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */ 457 int sdp_payload_type; /**< payload type */ 458 //@} 459 460 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */ 461 //@{ 462 /** handler structure */ 463 const RTPDynamicProtocolHandler *dynamic_handler; 464 465 /** private data associated with the dynamic protocol */ 466 PayloadContext *dynamic_protocol_context; 467 //@} 468 469 /** Enable sending RTCP feedback messages according to RFC 4585 */ 470 int feedback; 471 472 /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */ 473 uint32_t ssrc; 474 475 char crypto_suite[40]; 476 char crypto_params[100]; 477 } RTSPStream; 478 479 void ff_rtsp_parse_line(AVFormatContext *s, 480 RTSPMessageHeader *reply, const char *buf, 481 RTSPState *rt, const char *method); 482 483 /** 484 * Send a command to the RTSP server without waiting for the reply. 485 * 486 * @see rtsp_send_cmd_with_content_async 487 */ 488 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, 489 const char *url, const char *headers); 490 491 /** 492 * Send a command to the RTSP server and wait for the reply. 493 * 494 * @param s RTSP (de)muxer context 495 * @param method the method for the request 496 * @param url the target url for the request 497 * @param headers extra header lines to include in the request 498 * @param reply pointer where the RTSP message header will be stored 499 * @param content_ptr pointer where the RTSP message body, if any, will 500 * be stored (length is in reply) 501 * @param send_content if non-null, the data to send as request body content 502 * @param send_content_length the length of the send_content data, or 0 if 503 * send_content is null 504 * 505 * @return zero if success, nonzero otherwise 506 */ 507 int ff_rtsp_send_cmd_with_content(AVFormatContext *s, 508 const char *method, const char *url, 509 const char *headers, 510 RTSPMessageHeader *reply, 511 unsigned char **content_ptr, 512 const unsigned char *send_content, 513 int send_content_length); 514 515 /** 516 * Send a command to the RTSP server and wait for the reply. 517 * 518 * @see rtsp_send_cmd_with_content 519 */ 520 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, 521 const char *url, const char *headers, 522 RTSPMessageHeader *reply, unsigned char **content_ptr); 523 524 /** 525 * Read a RTSP message from the server, or prepare to read data 526 * packets if we're reading data interleaved over the TCP/RTSP 527 * connection as well. 528 * 529 * @param s RTSP (de)muxer context 530 * @param reply pointer where the RTSP message header will be stored 531 * @param content_ptr pointer where the RTSP message body, if any, will 532 * be stored (length is in reply) 533 * @param return_on_interleaved_data whether the function may return if we 534 * encounter a data marker ('$'), which precedes data 535 * packets over interleaved TCP/RTSP connections. If this 536 * is set, this function will return 1 after encountering 537 * a '$'. If it is not set, the function will skip any 538 * data packets (if they are encountered), until a reply 539 * has been fully parsed. If no more data is available 540 * without parsing a reply, it will return an error. 541 * @param method the RTSP method this is a reply to. This affects how 542 * some response headers are acted upon. May be NULL. 543 * 544 * @return 1 if a data packets is ready to be received, -1 on error, 545 * and 0 on success. 546 */ 547 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, 548 unsigned char **content_ptr, 549 int return_on_interleaved_data, const char *method); 550 551 /** 552 * Skip a RTP/TCP interleaved packet. 553 */ 554 void ff_rtsp_skip_packet(AVFormatContext *s); 555 556 /** 557 * Connect to the RTSP server and set up the individual media streams. 558 * This can be used for both muxers and demuxers. 559 * 560 * @param s RTSP (de)muxer context 561 * 562 * @return 0 on success, < 0 on error. Cleans up all allocations done 563 * within the function on error. 564 */ 565 int ff_rtsp_connect(AVFormatContext *s); 566 567 /** 568 * Close and free all streams within the RTSP (de)muxer 569 * 570 * @param s RTSP (de)muxer context 571 */ 572 void ff_rtsp_close_streams(AVFormatContext *s); 573 574 /** 575 * Close all connection handles within the RTSP (de)muxer 576 * 577 * @param s RTSP (de)muxer context 578 */ 579 void ff_rtsp_close_connections(AVFormatContext *s); 580 581 /** 582 * Get the description of the stream and set up the RTSPStream child 583 * objects. 584 */ 585 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply); 586 587 /** 588 * Announce the stream to the server and set up the RTSPStream child 589 * objects for each media stream. 590 */ 591 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr); 592 593 /** 594 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in 595 * listen mode. 596 */ 597 int ff_rtsp_parse_streaming_commands(AVFormatContext *s); 598 599 /** 600 * Parse an SDP description of streams by populating an RTSPState struct 601 * within the AVFormatContext; also allocate the RTP streams and the 602 * pollfd array used for UDP streams. 603 */ 604 int ff_sdp_parse(AVFormatContext *s, const char *content); 605 606 /** 607 * Receive one RTP packet from an TCP interleaved RTSP stream. 608 */ 609 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, 610 uint8_t *buf, int buf_size); 611 612 /** 613 * Send buffered packets over TCP. 614 */ 615 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st); 616 617 /** 618 * Receive one packet from the RTSPStreams set up in the AVFormatContext 619 * (which should contain a RTSPState struct as priv_data). 620 */ 621 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt); 622 623 /** 624 * Do the SETUP requests for each stream for the chosen 625 * lower transport mode. 626 * @return 0 on success, <0 on error, 1 if protocol is unavailable 627 */ 628 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, 629 int lower_transport, const char *real_challenge); 630 631 /** 632 * Undo the effect of ff_rtsp_make_setup_request, close the 633 * transport_priv and rtp_handle fields. 634 */ 635 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets); 636 637 /** 638 * Open RTSP transport context. 639 */ 640 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st); 641 642 extern const AVOption ff_rtsp_options[]; 643 644 #endif /* AVFORMAT_RTSP_H */ 645