3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
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.
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.
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
21 #ifndef AVFORMAT_RTSP_H
22 #define AVFORMAT_RTSP_H
26 #include "rtspcodes.h"
31 #include "libavutil/log.h"
32 #include "libavutil/opt.h"
35 * Network layer over which RTP/etc packet data will be transported.
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
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.
58 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
59 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
60 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
65 * Transport mode for the RTSP data. This may be plain, or
66 * tunneled, which is done over HTTP.
68 enum RTSPControlTransport {
69 RTSP_MODE_PLAIN, /**< Normal RTSP */
70 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
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
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.
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
94 int interleaved_min, interleaved_max;
96 /** UDP multicast port range; the ports to which we should connect to
97 * receive multicast UDP data. */
98 int port_min, port_max;
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;
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;
108 /** time-to-live value (required for multicast); the amount of HOPs that
109 * packets will be allowed to make before being discarded. */
112 /** transport set to record data */
115 struct sockaddr_storage destination; /**< destination IP address */
116 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
118 /** data/packet transport protocol; e.g. RTP or RDT */
119 enum RTSPTransport transport;
121 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
122 enum RTSPLowerTransport lower_transport;
123 } RTSPTransportField;
126 * This describes the server response to each RTSP command.
128 typedef struct RTSPMessageHeader {
129 /** length of the data following this header */
132 enum RTSPStatusCode status_code; /**< response code from server */
134 /** number of items in the 'transports' variable below */
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;
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];
145 int seq; /**< sequence number */
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];
151 /** the "Location:" field. This value is used to handle redirection.
155 /** the "RealChallenge1:" field from the server */
156 char real_challenge[64];
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'. */
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. */
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. */
180 /** The "reason" is meant to specify better the meaning of the error code
186 * Content type header
188 char content_type[64];
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().
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 */
204 * Identify particular servers that require special handling, such as
205 * standards-incompliant "Transport:" lines in the SETUP request.
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 */
215 * Private data for the RTSP demuxer.
217 * @todo Use AVIOContext instead of URLContext
219 typedef struct RTSPState {
220 const AVClass *class; /**< Class for private options. */
221 URLContext *rtsp_hd; /* RTSP TCP connection handle */
223 /** number of items in the 'rtsp_streams' variable */
226 struct RTSPStream **rtsp_streams; /**< streams in this session */
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;
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;
242 int seq; /**< RTSP command sequence number */
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];
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. */
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;
258 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
259 enum RTSPTransport transport;
261 /** the negotiated network layer transport protocol; e.g. TCP or UDP
263 enum RTSPLowerTransport lower_transport;
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;
270 /** the "RealChallenge1:" field from the server */
271 char real_challenge[64];
273 /** plaintext authorization line (username:password) */
276 /** authentication state */
277 HTTPAuthState auth_state;
279 /** The last reply of the server to a RTSP command */
280 char last_reply[2048]; /* XXX: allocate ? */
282 /** RTSPStream->transport_priv of the last stream that we read a
284 void *cur_transport_priv;
286 /** The following are used for Real stream selection */
288 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
289 int need_subscription;
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;
295 /** current stream setup. This is a temporary buffer used to compare
296 * current setup to previous frame setup. */
297 enum AVDiscard *real_setup;
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];
305 /** The following are used for RTP/ASF streams */
307 /** ASF demuxer context for the embedded ASF stream from WMS servers */
308 AVFormatContext *asf_ctx;
310 /** cache for position of the asf demuxer, since we load a new
311 * data packet in the bytecontext for each incoming RTSP packet. */
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];
320 /** The following are used for parsing raw mpegts in udp */
322 struct MpegTSContext *ts;
327 /** Additional output handle, used when input and output are done
328 * separately, eg for HTTP tunneling. */
329 URLContext *rtsp_hd_out;
331 /** RTSP transport mode, such as plain or tunneled. */
332 enum RTSPControlTransport control_transport;
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. */
339 /** Reusable buffer for receiving packets */
343 * A mask with all requested transport methods
345 int lower_transport_mask;
348 * The number of returned packets
353 * Polling array for udp
359 * Whether the server supports the GET_PARAMETER method.
361 int get_parameter_supported;
364 * Do not begin to play the stream immediately.
369 * Option flags for the chained RTP muxer.
373 /** Whether the server accepts the x-Dynamic-Rate header */
374 int accept_dynamic_rate;
377 * Various option flags for the RTSP muxer/demuxer.
382 * Mask of all requested media types
387 * Minimum and maximum local UDP ports.
389 int rtp_port_min, rtp_port_max;
392 * Timeout to wait for incoming connections.
397 * timeout of socket i/o operations.
402 * Size of RTP packet reordering queue.
404 int reordering_queue_size;
411 char default_lang[4];
415 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
416 receive packets only from the right
417 source address and port. */
418 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
419 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
420 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
421 address of received packets. */
422 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
424 typedef struct RTSPSource {
425 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
429 * Describe a single stream, as identified by a single m= line block in the
430 * SDP content. In the case of RDT, one RTSPStream can represent multiple
431 * AVStreams. In this case, each AVStream in this set has similar content
432 * (but different codec/bitrate).
434 typedef struct RTSPStream {
435 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
436 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
438 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
441 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
442 * for the selected transport. Only used for TCP. */
443 int interleaved_min, interleaved_max;
445 char control_url[1024]; /**< url for this stream (from SDP) */
447 /** The following are used only in SDP, not RTSP */
449 int sdp_port; /**< port (from SDP content) */
450 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
451 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
452 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
453 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
454 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
455 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
456 int sdp_payload_type; /**< payload type */
459 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
461 /** handler structure */
462 const RTPDynamicProtocolHandler *dynamic_handler;
464 /** private data associated with the dynamic protocol */
465 PayloadContext *dynamic_protocol_context;
468 /** Enable sending RTCP feedback messages according to RFC 4585 */
471 /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
474 char crypto_suite[40];
475 char crypto_params[100];
478 void ff_rtsp_parse_line(AVFormatContext *s,
479 RTSPMessageHeader *reply, const char *buf,
480 RTSPState *rt, const char *method);
483 * Send a command to the RTSP server without waiting for the reply.
485 * @see rtsp_send_cmd_with_content_async
487 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
488 const char *url, const char *headers);
491 * Send a command to the RTSP server and wait for the reply.
493 * @param s RTSP (de)muxer context
494 * @param method the method for the request
495 * @param url the target url for the request
496 * @param headers extra header lines to include in the request
497 * @param reply pointer where the RTSP message header will be stored
498 * @param content_ptr pointer where the RTSP message body, if any, will
499 * be stored (length is in reply)
500 * @param send_content if non-null, the data to send as request body content
501 * @param send_content_length the length of the send_content data, or 0 if
502 * send_content is null
504 * @return zero if success, nonzero otherwise
506 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
507 const char *method, const char *url,
509 RTSPMessageHeader *reply,
510 unsigned char **content_ptr,
511 const unsigned char *send_content,
512 int send_content_length);
515 * Send a command to the RTSP server and wait for the reply.
517 * @see rtsp_send_cmd_with_content
519 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
520 const char *url, const char *headers,
521 RTSPMessageHeader *reply, unsigned char **content_ptr);
524 * Read a RTSP message from the server, or prepare to read data
525 * packets if we're reading data interleaved over the TCP/RTSP
526 * connection as well.
528 * @param s RTSP (de)muxer context
529 * @param reply pointer where the RTSP message header will be stored
530 * @param content_ptr pointer where the RTSP message body, if any, will
531 * be stored (length is in reply)
532 * @param return_on_interleaved_data whether the function may return if we
533 * encounter a data marker ('$'), which precedes data
534 * packets over interleaved TCP/RTSP connections. If this
535 * is set, this function will return 1 after encountering
536 * a '$'. If it is not set, the function will skip any
537 * data packets (if they are encountered), until a reply
538 * has been fully parsed. If no more data is available
539 * without parsing a reply, it will return an error.
540 * @param method the RTSP method this is a reply to. This affects how
541 * some response headers are acted upon. May be NULL.
543 * @return 1 if a data packets is ready to be received, -1 on error,
546 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
547 unsigned char **content_ptr,
548 int return_on_interleaved_data, const char *method);
551 * Skip a RTP/TCP interleaved packet.
553 void ff_rtsp_skip_packet(AVFormatContext *s);
556 * Connect to the RTSP server and set up the individual media streams.
557 * This can be used for both muxers and demuxers.
559 * @param s RTSP (de)muxer context
561 * @return 0 on success, < 0 on error. Cleans up all allocations done
562 * within the function on error.
564 int ff_rtsp_connect(AVFormatContext *s);
567 * Close and free all streams within the RTSP (de)muxer
569 * @param s RTSP (de)muxer context
571 void ff_rtsp_close_streams(AVFormatContext *s);
574 * Close all connection handles within the RTSP (de)muxer
576 * @param s RTSP (de)muxer context
578 void ff_rtsp_close_connections(AVFormatContext *s);
581 * Get the description of the stream and set up the RTSPStream child
584 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
587 * Announce the stream to the server and set up the RTSPStream child
588 * objects for each media stream.
590 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
593 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
596 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
599 * Parse an SDP description of streams by populating an RTSPState struct
600 * within the AVFormatContext; also allocate the RTP streams and the
601 * pollfd array used for UDP streams.
603 int ff_sdp_parse(AVFormatContext *s, const char *content);
606 * Receive one RTP packet from an TCP interleaved RTSP stream.
608 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
609 uint8_t *buf, int buf_size);
612 * Send buffered packets over TCP.
614 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
617 * Receive one packet from the RTSPStreams set up in the AVFormatContext
618 * (which should contain a RTSPState struct as priv_data).
620 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
623 * Do the SETUP requests for each stream for the chosen
624 * lower transport mode.
625 * @return 0 on success, <0 on error, 1 if protocol is unavailable
627 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
628 int lower_transport, const char *real_challenge);
631 * Undo the effect of ff_rtsp_make_setup_request, close the
632 * transport_priv and rtp_handle fields.
634 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
637 * Open RTSP transport context.
639 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
641 extern const AVOption ff_rtsp_options[];
643 #endif /* AVFORMAT_RTSP_H */