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 */
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.
53 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
54 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
59 * Transport mode for the RTSP data. This may be plain, or
60 * tunneled, which is done over HTTP.
62 enum RTSPControlTransport {
63 RTSP_MODE_PLAIN, /**< Normal RTSP */
64 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
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 65000
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.
82 typedef 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
87 int interleaved_min, interleaved_max;
89 /** UDP multicast port range; the ports to which we should connect to
90 * receive multicast UDP data. */
91 int port_min, port_max;
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;
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;
101 /** time-to-live value (required for multicast); the amount of HOPs that
102 * packets will be allowed to make before being discarded. */
105 struct sockaddr_storage destination; /**< destination IP address */
106 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
108 /** data/packet transport protocol; e.g. RTP or RDT */
109 enum RTSPTransport transport;
111 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
112 enum RTSPLowerTransport lower_transport;
113 } RTSPTransportField;
116 * This describes the server response to each RTSP command.
118 typedef struct RTSPMessageHeader {
119 /** length of the data following this header */
122 enum RTSPStatusCode status_code; /**< response code from server */
124 /** number of items in the 'transports' variable below */
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;
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];
135 int seq; /**< sequence number */
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];
141 /** the "Location:" field. This value is used to handle redirection.
145 /** the "RealChallenge1:" field from the server */
146 char real_challenge[64];
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'. */
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. */
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. */
170 /** The "reason" is meant to specify better the meaning of the error code
176 * Content type header
178 char content_type[64];
182 * Client state, i.e. whether we are currently receiving data (PLAYING) or
183 * setup-but-not-receiving (PAUSED). State can be changed in applications
184 * by calling av_read_play/pause().
186 enum RTSPClientState {
187 RTSP_STATE_IDLE, /**< not initialized */
188 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
189 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
190 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
194 * Identify particular servers that require special handling, such as
195 * standards-incompliant "Transport:" lines in the SETUP request.
197 enum RTSPServerType {
198 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
199 RTSP_SERVER_REAL, /**< Realmedia-style server */
200 RTSP_SERVER_WMS, /**< Windows Media server */
205 * Private data for the RTSP demuxer.
207 * @todo Use AVIOContext instead of URLContext
209 typedef struct RTSPState {
210 const AVClass *class; /**< Class for private options. */
211 URLContext *rtsp_hd; /* RTSP TCP connection handle */
213 /** number of items in the 'rtsp_streams' variable */
216 struct RTSPStream **rtsp_streams; /**< streams in this session */
218 /** indicator of whether we are currently receiving data from the
219 * server. Basically this isn't more than a simple cache of the
220 * last PLAY/PAUSE command sent to the server, to make sure we don't
221 * send 2x the same unexpectedly or commands in the wrong state. */
222 enum RTSPClientState state;
224 /** the seek value requested when calling av_seek_frame(). This value
225 * is subsequently used as part of the "Range" parameter when emitting
226 * the RTSP PLAY command. If we are currently playing, this command is
227 * called instantly. If we are currently paused, this command is called
228 * whenever we resume playback. Either way, the value is only used once,
229 * see rtsp_read_play() and rtsp_read_seek(). */
230 int64_t seek_timestamp;
232 int seq; /**< RTSP command sequence number */
234 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
235 * identifier that the client should re-transmit in each RTSP command */
236 char session_id[512];
238 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
239 * the server will go without traffic on the RTSP/TCP line before it
240 * closes the connection. */
243 /** timestamp of the last RTSP command that we sent to the RTSP server.
244 * This is used to calculate when to send dummy commands to keep the
245 * connection alive, in conjunction with timeout. */
246 int64_t last_cmd_time;
248 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
249 enum RTSPTransport transport;
251 /** the negotiated network layer transport protocol; e.g. TCP or UDP
253 enum RTSPLowerTransport lower_transport;
255 /** brand of server that we're talking to; e.g. WMS, REAL or other.
256 * Detected based on the value of RTSPMessageHeader->server or the presence
257 * of RTSPMessageHeader->real_challenge */
258 enum RTSPServerType server_type;
260 /** the "RealChallenge1:" field from the server */
261 char real_challenge[64];
263 /** plaintext authorization line (username:password) */
266 /** authentication state */
267 HTTPAuthState auth_state;
269 /** The last reply of the server to a RTSP command */
270 char last_reply[2048]; /* XXX: allocate ? */
272 /** RTSPStream->transport_priv of the last stream that we read a
274 void *cur_transport_priv;
276 /** The following are used for Real stream selection */
278 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
279 int need_subscription;
281 /** stream setup during the last frame read. This is used to detect if
282 * we need to subscribe or unsubscribe to any new streams. */
283 enum AVDiscard *real_setup_cache;
285 /** current stream setup. This is a temporary buffer used to compare
286 * current setup to previous frame setup. */
287 enum AVDiscard *real_setup;
289 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
290 * this is used to send the same "Unsubscribe:" if stream setup changed,
291 * before sending a new "Subscribe:" command. */
292 char last_subscription[1024];
295 /** The following are used for RTP/ASF streams */
297 /** ASF demuxer context for the embedded ASF stream from WMS servers */
298 AVFormatContext *asf_ctx;
300 /** cache for position of the asf demuxer, since we load a new
301 * data packet in the bytecontext for each incoming RTSP packet. */
305 /** some MS RTSP streams contain a URL in the SDP that we need to use
306 * for all subsequent RTSP requests, rather than the input URI; in
307 * other cases, this is a copy of AVFormatContext->filename. */
308 char control_uri[1024];
310 /** Additional output handle, used when input and output are done
311 * separately, eg for HTTP tunneling. */
312 URLContext *rtsp_hd_out;
314 /** RTSP transport mode, such as plain or tunneled. */
315 enum RTSPControlTransport control_transport;
317 /* Number of RTCP BYE packets the RTSP session has received.
318 * An EOF is propagated back if nb_byes == nb_streams.
319 * This is reset after a seek. */
322 /** Reusable buffer for receiving packets */
326 * A mask with all requested transport methods
328 int lower_transport_mask;
331 * The number of returned packets
336 * Polling array for udp
341 * Whether the server supports the GET_PARAMETER method.
343 int get_parameter_supported;
346 * Do not begin to play the stream immediately.
351 * Option flags for the chained RTP muxer.
355 /** Whether the server accepts the x-Dynamic-Rate header */
356 int accept_dynamic_rate;
359 * Various option flags for the RTSP muxer/demuxer.
364 * Mask of all requested media types
369 * Minimum and maximum local UDP ports.
371 int rtp_port_min, rtp_port_max;
374 #define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
375 receive packets only from the right
376 source address and port. */
379 * Describe a single stream, as identified by a single m= line block in the
380 * SDP content. In the case of RDT, one RTSPStream can represent multiple
381 * AVStreams. In this case, each AVStream in this set has similar content
382 * (but different codec/bitrate).
384 typedef struct RTSPStream {
385 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
386 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
388 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
391 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
392 * for the selected transport. Only used for TCP. */
393 int interleaved_min, interleaved_max;
395 char control_url[1024]; /**< url for this stream (from SDP) */
397 /** The following are used only in SDP, not RTSP */
399 int sdp_port; /**< port (from SDP content) */
400 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
401 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
402 int sdp_payload_type; /**< payload type */
405 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
407 /** handler structure */
408 RTPDynamicProtocolHandler *dynamic_handler;
410 /** private data associated with the dynamic protocol */
411 PayloadContext *dynamic_protocol_context;
415 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
416 RTSPState *rt, const char *method);
419 * Send a command to the RTSP server without waiting for the reply.
421 * @see rtsp_send_cmd_with_content_async
423 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
424 const char *url, const char *headers);
427 * Send a command to the RTSP server and wait for the reply.
429 * @param s RTSP (de)muxer context
430 * @param method the method for the request
431 * @param url the target url for the request
432 * @param headers extra header lines to include in the request
433 * @param reply pointer where the RTSP message header will be stored
434 * @param content_ptr pointer where the RTSP message body, if any, will
435 * be stored (length is in reply)
436 * @param send_content if non-null, the data to send as request body content
437 * @param send_content_length the length of the send_content data, or 0 if
438 * send_content is null
440 * @return zero if success, nonzero otherwise
442 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
443 const char *method, const char *url,
445 RTSPMessageHeader *reply,
446 unsigned char **content_ptr,
447 const unsigned char *send_content,
448 int send_content_length);
451 * Send a command to the RTSP server and wait for the reply.
453 * @see rtsp_send_cmd_with_content
455 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
456 const char *url, const char *headers,
457 RTSPMessageHeader *reply, unsigned char **content_ptr);
460 * Read a RTSP message from the server, or prepare to read data
461 * packets if we're reading data interleaved over the TCP/RTSP
462 * connection as well.
464 * @param s RTSP (de)muxer context
465 * @param reply pointer where the RTSP message header will be stored
466 * @param content_ptr pointer where the RTSP message body, if any, will
467 * be stored (length is in reply)
468 * @param return_on_interleaved_data whether the function may return if we
469 * encounter a data marker ('$'), which precedes data
470 * packets over interleaved TCP/RTSP connections. If this
471 * is set, this function will return 1 after encountering
472 * a '$'. If it is not set, the function will skip any
473 * data packets (if they are encountered), until a reply
474 * has been fully parsed. If no more data is available
475 * without parsing a reply, it will return an error.
476 * @param method the RTSP method this is a reply to. This affects how
477 * some response headers are acted upon. May be NULL.
479 * @return 1 if a data packets is ready to be received, -1 on error,
482 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
483 unsigned char **content_ptr,
484 int return_on_interleaved_data, const char *method);
487 * Skip a RTP/TCP interleaved packet.
489 void ff_rtsp_skip_packet(AVFormatContext *s);
492 * Connect to the RTSP server and set up the individual media streams.
493 * This can be used for both muxers and demuxers.
495 * @param s RTSP (de)muxer context
497 * @return 0 on success, < 0 on error. Cleans up all allocations done
498 * within the function on error.
500 int ff_rtsp_connect(AVFormatContext *s);
503 * Close and free all streams within the RTSP (de)muxer
505 * @param s RTSP (de)muxer context
507 void ff_rtsp_close_streams(AVFormatContext *s);
510 * Close all connection handles within the RTSP (de)muxer
512 * @param s RTSP (de)muxer context
514 void ff_rtsp_close_connections(AVFormatContext *s);
517 * Get the description of the stream and set up the RTSPStream child
520 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
523 * Announce the stream to the server and set up the RTSPStream child
524 * objects for each media stream.
526 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
529 * Parse an SDP description of streams by populating an RTSPState struct
530 * within the AVFormatContext; also allocate the RTP streams and the
531 * pollfd array used for UDP streams.
533 int ff_sdp_parse(AVFormatContext *s, const char *content);
536 * Receive one RTP packet from an TCP interleaved RTSP stream.
538 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
539 uint8_t *buf, int buf_size);
542 * Receive one packet from the RTSPStreams set up in the AVFormatContext
543 * (which should contain a RTSPState struct as priv_data).
545 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
548 * Do the SETUP requests for each stream for the chosen
549 * lower transport mode.
550 * @return 0 on success, <0 on error, 1 if protocol is unavailable
552 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
553 int lower_transport, const char *real_challenge);
556 * Undo the effect of ff_rtsp_make_setup_request, close the
557 * transport_priv and rtp_handle fields.
559 void ff_rtsp_undo_setup(AVFormatContext *s);
561 extern const AVOption ff_rtsp_options[];
563 #endif /* AVFORMAT_RTSP_H */