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rtsp.h

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1 /*

2 * RTSP definitions

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

24 #include <stdint.h>

25 #include "avformat.h"

26 #include "rtspcodes.h"

27 #include "rtpdec.h"

28 #include "network.h"

29 #include "httpauth.h"

31 #include "libavutil/log.h"

32 #include "libavutil/opt.h"

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_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 };

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 */

56 enum 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 };

63 /**

64 * Transport mode for the RTSP data. This may be plain, or

65 * tunneled, which is done over HTTP.

66 */

67 enum RTSPControlTransport {

68 RTSP_MODE_PLAIN, /**< Normal RTSP */

69 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */

70 };

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

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 */

87 typedef 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;

94 /** UDP multicast port range; the ports to which we should connect to

95 * receive multicast UDP data. */

96 int port_min, port_max;

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 */

126 typedef 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 */

194 enum 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 */

205 enum 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 */

217 typedef 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 } RTSPState;

403

404 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -

405 receive packets only from the right

406 source address and port. */

407 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */

408 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */

409

410 /**

411 * Describe a single stream, as identified by a single m= line block in the

412 * SDP content. In the case of RDT, one RTSPStream can represent multiple

413 * AVStreams. In this case, each AVStream in this set has similar content

414 * (but different codec/bitrate).

415 */

416 typedef struct RTSPStream {

417 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */

418 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */

419

420 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */

421 int stream_index;

422

423 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max

424 * for the selected transport. Only used for TCP. */

425 int interleaved_min, interleaved_max;

426

427 char control_url[1024]; /**< url for this stream (from SDP) */

428

429 /** The following are used only in SDP, not RTSP */

430 //@{

431 int sdp_port; /**< port (from SDP content) */

432 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */

433 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */

434 int sdp_payload_type; /**< payload type */

435 //@}

436

437 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */

438 //@{

439 /** handler structure */

440 RTPDynamicProtocolHandler *dynamic_handler;

441

442 /** private data associated with the dynamic protocol */

443 PayloadContext *dynamic_protocol_context;

444 //@}

445

446 /** Enable sending RTCP feedback messages according to RFC 4585 */

447 int feedback;

448

449 char crypto_suite[40];

450 char crypto_params[100];

451 } RTSPStream;

452

453 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,

454 RTSPState *rt, const char *method);

455

456 /**

457 * Send a command to the RTSP server without waiting for the reply.

458 *

459 * @see rtsp_send_cmd_with_content_async

460 */

461 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,

462 const char *url, const char *headers);

463

464 /**

465 * Send a command to the RTSP server and wait for the reply.

466 *

467 * @param s RTSP (de)muxer context

468 * @param method the method for the request

469 * @param url the target url for the request

470 * @param headers extra header lines to include in the request

471 * @param reply pointer where the RTSP message header will be stored

472 * @param content_ptr pointer where the RTSP message body, if any, will

473 * be stored (length is in reply)

474 * @param send_content if non-null, the data to send as request body content

475 * @param send_content_length the length of the send_content data, or 0 if

476 * send_content is null

477 *

478 * @return zero if success, nonzero otherwise

479 */

480 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,

481 const char *method, const char *url,

482 const char *headers,

483 RTSPMessageHeader *reply,

484 unsigned char **content_ptr,

485 const unsigned char *send_content,

486 int send_content_length);

487

488 /**

489 * Send a command to the RTSP server and wait for the reply.

490 *

491 * @see rtsp_send_cmd_with_content

492 */

493 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,

494 const char *url, const char *headers,

495 RTSPMessageHeader *reply, unsigned char **content_ptr);

496

497 /**

498 * Read a RTSP message from the server, or prepare to read data

499 * packets if we're reading data interleaved over the TCP/RTSP

500 * connection as well.

501 *

502 * @param s RTSP (de)muxer context

503 * @param reply pointer where the RTSP message header will be stored

504 * @param content_ptr pointer where the RTSP message body, if any, will

505 * be stored (length is in reply)

506 * @param return_on_interleaved_data whether the function may return if we

507 * encounter a data marker ('$'), which precedes data

508 * packets over interleaved TCP/RTSP connections. If this

509 * is set, this function will return 1 after encountering

510 * a '$'. If it is not set, the function will skip any

511 * data packets (if they are encountered), until a reply

512 * has been fully parsed. If no more data is available

513 * without parsing a reply, it will return an error.

514 * @param method the RTSP method this is a reply to. This affects how

515 * some response headers are acted upon. May be NULL.

516 *

517 * @return 1 if a data packets is ready to be received, -1 on error,

518 * and 0 on success.

519 */

520 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,

521 unsigned char **content_ptr,

522 int return_on_interleaved_data, const char *method);

523

524 /**

525 * Skip a RTP/TCP interleaved packet.

526 */

527 void ff_rtsp_skip_packet(AVFormatContext *s);

528

529 /**

530 * Connect to the RTSP server and set up the individual media streams.

531 * This can be used for both muxers and demuxers.

532 *

533 * @param s RTSP (de)muxer context

534 *

535 * @return 0 on success, < 0 on error. Cleans up all allocations done

536 * within the function on error.

537 */

538 int ff_rtsp_connect(AVFormatContext *s);

539

540 /**

541 * Close and free all streams within the RTSP (de)muxer

542 *

543 * @param s RTSP (de)muxer context

544 */

545 void ff_rtsp_close_streams(AVFormatContext *s);

546

547 /**

548 * Close all connection handles within the RTSP (de)muxer

549 *

550 * @param s RTSP (de)muxer context

551 */

552 void ff_rtsp_close_connections(AVFormatContext *s);

553

554 /**

555 * Get the description of the stream and set up the RTSPStream child

556 * objects.

557 */

558 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);

559

560 /**

561 * Announce the stream to the server and set up the RTSPStream child

562 * objects for each media stream.

563 */

564 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);

565

566 /**

567 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in

568 * listen mode.

569 */

570 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);

571

572 /**

573 * Parse an SDP description of streams by populating an RTSPState struct

574 * within the AVFormatContext; also allocate the RTP streams and the

575 * pollfd array used for UDP streams.

576 */

577 int ff_sdp_parse(AVFormatContext *s, const char *content);

578

579 /**

580 * Receive one RTP packet from an TCP interleaved RTSP stream.

581 */

582 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,

583 uint8_t *buf, int buf_size);

584

585 /**

586 * Receive one packet from the RTSPStreams set up in the AVFormatContext

587 * (which should contain a RTSPState struct as priv_data).

588 */

589 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);

590

591 /**

592 * Do the SETUP requests for each stream for the chosen

593 * lower transport mode.

594 * @return 0 on success, <0 on error, 1 if protocol is unavailable

595 */

596 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,

597 int lower_transport, const char *real_challenge);

598

599 /**

600 * Undo the effect of ff_rtsp_make_setup_request, close the

601 * transport_priv and rtp_handle fields.

602 */

603 void ff_rtsp_undo_setup(AVFormatContext *s);

604

605 /**

606 * Open RTSP transport context.

607 */

608 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);

609

610 extern const AVOption ff_rtsp_options[];

611

612 #endif /* AVFORMAT_RTSP_H */

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