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"
32 #include "libavutil/log.h"
33 #include "libavutil/opt.h"
36 * Network layer over which RTP/etc packet data will be transported.
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
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.
59 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
60 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
61 RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
66 * Transport mode for the RTSP data. This may be plain, or
67 * tunneled, which is done over HTTP.
69 enum RTSPControlTransport {
70 RTSP_MODE_PLAIN, /**< Normal RTSP */
71 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
74 #define RTSP_DEFAULT_PORT 554
75 #define RTSPS_DEFAULT_PORT 322
76 #define RTSP_MAX_TRANSPORTS 8
77 #define RTSP_TCP_MAX_PACKET_SIZE 1472
78 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
79 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
80 #define RTSP_RTP_PORT_MIN 5000
81 #define RTSP_RTP_PORT_MAX 65000
82 #define SDP_MAX_SIZE 16384
85 * This describes a single item in the "Transport:" line of one stream as
86 * negotiated by the SETUP RTSP command. Multiple transports are comma-
87 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
88 * client_port=1000-1001;server_port=1800-1801") and described in separate
89 * RTSPTransportFields.
91 typedef struct RTSPTransportField {
92 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
93 * with a '$', stream length and stream ID. If the stream ID is within
94 * the range of this interleaved_min-max, then the packet belongs to
96 int interleaved_min, interleaved_max;
98 /** UDP multicast port range; the ports to which we should connect to
99 * receive multicast UDP data. */
100 int port_min, port_max;
102 /** UDP client ports; these should be the local ports of the UDP RTP
103 * (and RTCP) sockets over which we receive RTP/RTCP data. */
104 int client_port_min, client_port_max;
106 /** UDP unicast server port range; the ports to which we should connect
107 * to receive unicast UDP RTP/RTCP data. */
108 int server_port_min, server_port_max;
110 /** time-to-live value (required for multicast); the amount of HOPs that
111 * packets will be allowed to make before being discarded. */
114 /** transport set to record data */
117 struct sockaddr_storage destination; /**< destination IP address */
118 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
120 /** data/packet transport protocol; e.g. RTP or RDT */
121 enum RTSPTransport transport;
123 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
124 enum RTSPLowerTransport lower_transport;
125 } RTSPTransportField;
128 * This describes the server response to each RTSP command.
130 typedef struct RTSPMessageHeader {
131 /** length of the data following this header */
134 enum RTSPStatusCode status_code; /**< response code from server */
136 /** number of items in the 'transports' variable below */
139 /** Time range of the streams that the server will stream. In
140 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
141 int64_t range_start, range_end;
143 /** describes the complete "Transport:" line of the server in response
144 * to a SETUP RTSP command by the client */
145 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
147 int seq; /**< sequence number */
149 /** the "Session:" field. This value is initially set by the server and
150 * should be re-transmitted by the client in every RTSP command. */
151 char session_id[512];
153 /** the "Location:" field. This value is used to handle redirection.
157 /** the "RealChallenge1:" field from the server */
158 char real_challenge[64];
160 /** the "Server: field, which can be used to identify some special-case
161 * servers that are not 100% standards-compliant. We use this to identify
162 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
163 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
164 * use something like "Helix [..] Server Version v.e.r.sion (platform)
165 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
166 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
169 /** The "timeout" comes as part of the server response to the "SETUP"
170 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
171 * time, in seconds, that the server will go without traffic over the
172 * RTSP/TCP connection before it closes the connection. To prevent
173 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
174 * than this value. */
177 /** The "Notice" or "X-Notice" field value. See
178 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
179 * for a complete list of supported values. */
182 /** The "reason" is meant to specify better the meaning of the error code
188 * Content type header
190 char content_type[64];
193 * SAT>IP com.ses.streamID header
199 * Client state, i.e. whether we are currently receiving data (PLAYING) or
200 * setup-but-not-receiving (PAUSED). State can be changed in applications
201 * by calling av_read_play/pause().
203 enum RTSPClientState {
204 RTSP_STATE_IDLE, /**< not initialized */
205 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
206 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
207 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
211 * Identify particular servers that require special handling, such as
212 * standards-incompliant "Transport:" lines in the SETUP request.
214 enum RTSPServerType {
215 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
216 RTSP_SERVER_REAL, /**< Realmedia-style server */
217 RTSP_SERVER_WMS, /**< Windows Media server */
218 RTSP_SERVER_SATIP,/**< SAT>IP server */
223 * Private data for the RTSP demuxer.
225 * @todo Use AVIOContext instead of URLContext
227 typedef struct RTSPState {
228 const AVClass *class; /**< Class for private options. */
229 URLContext *rtsp_hd; /* RTSP TCP connection handle */
231 /** number of items in the 'rtsp_streams' variable */
234 struct RTSPStream **rtsp_streams; /**< streams in this session */
236 /** indicator of whether we are currently receiving data from the
237 * server. Basically this isn't more than a simple cache of the
238 * last PLAY/PAUSE command sent to the server, to make sure we don't
239 * send 2x the same unexpectedly or commands in the wrong state. */
240 enum RTSPClientState state;
242 /** the seek value requested when calling av_seek_frame(). This value
243 * is subsequently used as part of the "Range" parameter when emitting
244 * the RTSP PLAY command. If we are currently playing, this command is
245 * called instantly. If we are currently paused, this command is called
246 * whenever we resume playback. Either way, the value is only used once,
247 * see rtsp_read_play() and rtsp_read_seek(). */
248 int64_t seek_timestamp;
250 int seq; /**< RTSP command sequence number */
252 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
253 * identifier that the client should re-transmit in each RTSP command */
254 char session_id[512];
256 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
257 * the server will go without traffic on the RTSP/TCP line before it
258 * closes the connection. */
261 /** timestamp of the last RTSP command that we sent to the RTSP server.
262 * This is used to calculate when to send dummy commands to keep the
263 * connection alive, in conjunction with timeout. */
264 int64_t last_cmd_time;
266 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
267 enum RTSPTransport transport;
269 /** the negotiated network layer transport protocol; e.g. TCP or UDP
271 enum RTSPLowerTransport lower_transport;
273 /** brand of server that we're talking to; e.g. WMS, REAL or other.
274 * Detected based on the value of RTSPMessageHeader->server or the presence
275 * of RTSPMessageHeader->real_challenge */
276 enum RTSPServerType server_type;
278 /** the "RealChallenge1:" field from the server */
279 char real_challenge[64];
281 /** plaintext authorization line (username:password) */
284 /** authentication state */
285 HTTPAuthState auth_state;
287 /** The last reply of the server to a RTSP command */
288 char last_reply[2048]; /* XXX: allocate ? */
290 /** RTSPStream->transport_priv of the last stream that we read a
292 void *cur_transport_priv;
294 /** The following are used for Real stream selection */
296 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
297 int need_subscription;
299 /** stream setup during the last frame read. This is used to detect if
300 * we need to subscribe or unsubscribe to any new streams. */
301 enum AVDiscard *real_setup_cache;
303 /** current stream setup. This is a temporary buffer used to compare
304 * current setup to previous frame setup. */
305 enum AVDiscard *real_setup;
307 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
308 * this is used to send the same "Unsubscribe:" if stream setup changed,
309 * before sending a new "Subscribe:" command. */
310 char last_subscription[1024];
313 /** The following are used for RTP/ASF streams */
315 /** ASF demuxer context for the embedded ASF stream from WMS servers */
316 AVFormatContext *asf_ctx;
318 /** cache for position of the asf demuxer, since we load a new
319 * data packet in the bytecontext for each incoming RTSP packet. */
323 /** some MS RTSP streams contain a URL in the SDP that we need to use
324 * for all subsequent RTSP requests, rather than the input URI; in
325 * other cases, this is a copy of AVFormatContext->filename. */
326 char control_uri[MAX_URL_SIZE];
328 /** The following are used for parsing raw mpegts in udp */
330 struct MpegTSContext *ts;
335 /** Additional output handle, used when input and output are done
336 * separately, eg for HTTP tunneling. */
337 URLContext *rtsp_hd_out;
339 /** RTSP transport mode, such as plain or tunneled. */
340 enum RTSPControlTransport control_transport;
342 /* Number of RTCP BYE packets the RTSP session has received.
343 * An EOF is propagated back if nb_byes == nb_streams.
344 * This is reset after a seek. */
347 /** Reusable buffer for receiving packets */
351 * A mask with all requested transport methods
353 int lower_transport_mask;
356 * The number of returned packets
361 * Polling array for udp
367 * Whether the server supports the GET_PARAMETER method.
369 int get_parameter_supported;
372 * Do not begin to play the stream immediately.
377 * Option flags for the chained RTP muxer.
381 /** Whether the server accepts the x-Dynamic-Rate header */
382 int accept_dynamic_rate;
385 * Various option flags for the RTSP muxer/demuxer.
390 * Mask of all requested media types
395 * Minimum and maximum local UDP ports.
397 int rtp_port_min, rtp_port_max;
400 * Timeout to wait for incoming connections.
405 * timeout of socket i/o operations.
410 * Size of RTP packet reordering queue.
412 int reordering_queue_size;
419 char default_lang[4];
424 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
425 receive packets only from the right
426 source address and port. */
427 #define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
428 #define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
429 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
430 address of received packets. */
431 #define RTSP_FLAG_PREFER_TCP 0x10 /**< Try RTP via TCP first if possible. */
432 #define RTSP_FLAG_SATIP_RAW 0x20 /**< Export SAT>IP stream as raw MPEG-TS */
434 typedef struct RTSPSource {
435 char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
439 * Describe a single stream, as identified by a single m= line block in the
440 * SDP content. In the case of RDT, one RTSPStream can represent multiple
441 * AVStreams. In this case, each AVStream in this set has similar content
442 * (but different codec/bitrate).
444 typedef struct RTSPStream {
445 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
446 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
448 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
451 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
452 * for the selected transport. Only used for TCP. */
453 int interleaved_min, interleaved_max;
455 char control_url[MAX_URL_SIZE]; /**< url for this stream (from SDP) */
457 /** The following are used only in SDP, not RTSP */
459 int sdp_port; /**< port (from SDP content) */
460 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
461 int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
462 struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
463 int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
464 struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
465 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
466 int sdp_payload_type; /**< payload type */
469 /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
471 /** handler structure */
472 const RTPDynamicProtocolHandler *dynamic_handler;
474 /** private data associated with the dynamic protocol */
475 PayloadContext *dynamic_protocol_context;
478 /** Enable sending RTCP feedback messages according to RFC 4585 */
481 /** SSRC for this stream, to allow identifying RTCP packets before the first RTP packet */
484 char crypto_suite[40];
485 char crypto_params[100];
488 void ff_rtsp_parse_line(AVFormatContext *s,
489 RTSPMessageHeader *reply, const char *buf,
490 RTSPState *rt, const char *method);
493 * Send a command to the RTSP server without waiting for the reply.
495 * @see rtsp_send_cmd_with_content_async
497 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
498 const char *url, const char *headers);
501 * Send a command to the RTSP server and wait for the reply.
503 * @param s RTSP (de)muxer context
504 * @param method the method for the request
505 * @param url the target url for the request
506 * @param headers extra header lines to include in the request
507 * @param reply pointer where the RTSP message header will be stored
508 * @param content_ptr pointer where the RTSP message body, if any, will
509 * be stored (length is in reply)
510 * @param send_content if non-null, the data to send as request body content
511 * @param send_content_length the length of the send_content data, or 0 if
512 * send_content is null
514 * @return zero if success, nonzero otherwise
516 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
517 const char *method, const char *url,
519 RTSPMessageHeader *reply,
520 unsigned char **content_ptr,
521 const unsigned char *send_content,
522 int send_content_length);
525 * Send a command to the RTSP server and wait for the reply.
527 * @see rtsp_send_cmd_with_content
529 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
530 const char *url, const char *headers,
531 RTSPMessageHeader *reply, unsigned char **content_ptr);
534 * Read a RTSP message from the server, or prepare to read data
535 * packets if we're reading data interleaved over the TCP/RTSP
536 * connection as well.
538 * @param s RTSP (de)muxer context
539 * @param reply pointer where the RTSP message header will be stored
540 * @param content_ptr pointer where the RTSP message body, if any, will
541 * be stored (length is in reply)
542 * @param return_on_interleaved_data whether the function may return if we
543 * encounter a data marker ('$'), which precedes data
544 * packets over interleaved TCP/RTSP connections. If this
545 * is set, this function will return 1 after encountering
546 * a '$'. If it is not set, the function will skip any
547 * data packets (if they are encountered), until a reply
548 * has been fully parsed. If no more data is available
549 * without parsing a reply, it will return an error.
550 * @param method the RTSP method this is a reply to. This affects how
551 * some response headers are acted upon. May be NULL.
553 * @return 1 if a data packets is ready to be received, -1 on error,
556 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
557 unsigned char **content_ptr,
558 int return_on_interleaved_data, const char *method);
561 * Skip a RTP/TCP interleaved packet.
563 void ff_rtsp_skip_packet(AVFormatContext *s);
566 * Connect to the RTSP server and set up the individual media streams.
567 * This can be used for both muxers and demuxers.
569 * @param s RTSP (de)muxer context
571 * @return 0 on success, < 0 on error. Cleans up all allocations done
572 * within the function on error.
574 int ff_rtsp_connect(AVFormatContext *s);
577 * Close and free all streams within the RTSP (de)muxer
579 * @param s RTSP (de)muxer context
581 void ff_rtsp_close_streams(AVFormatContext *s);
584 * Close all connection handles within the RTSP (de)muxer
586 * @param s RTSP (de)muxer context
588 void ff_rtsp_close_connections(AVFormatContext *s);
591 * Get the description of the stream and set up the RTSPStream child
594 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
597 * Announce the stream to the server and set up the RTSPStream child
598 * objects for each media stream.
600 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
603 * Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
606 int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
609 * Parse an SDP description of streams by populating an RTSPState struct
610 * within the AVFormatContext; also allocate the RTP streams and the
611 * pollfd array used for UDP streams.
613 int ff_sdp_parse(AVFormatContext *s, const char *content);
616 * Receive one RTP packet from an TCP interleaved RTSP stream.
618 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
619 uint8_t *buf, int buf_size);
622 * Send buffered packets over TCP.
624 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st);
627 * Receive one packet from the RTSPStreams set up in the AVFormatContext
628 * (which should contain a RTSPState struct as priv_data).
630 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
633 * Do the SETUP requests for each stream for the chosen
634 * lower transport mode.
635 * @return 0 on success, <0 on error, 1 if protocol is unavailable
637 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
638 int lower_transport, const char *real_challenge);
641 * Undo the effect of ff_rtsp_make_setup_request, close the
642 * transport_priv and rtp_handle fields.
644 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets);
647 * Open RTSP transport context.
649 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
651 extern const AVOption ff_rtsp_options[];
653 #endif /* AVFORMAT_RTSP_H */