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