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