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"
34 * Network layer over which RTP/etc packet data will be transported.
36 enum RTSPLowerTransport {
37 RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
38 RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
39 RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
40 RTSP_LOWER_TRANSPORT_NB
44 * Packet profile of the data that we will be receiving. Real servers
45 * commonly send RDT (although they can sometimes send RTP as well),
46 * whereas most others will send RTP.
49 RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
50 RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
55 * Transport mode for the RTSP data. This may be plain, or
56 * tunneled, which is done over HTTP.
58 enum RTSPControlTransport {
59 RTSP_MODE_PLAIN, /**< Normal RTSP */
60 RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
63 #define RTSP_DEFAULT_PORT 554
64 #define RTSP_MAX_TRANSPORTS 8
65 #define RTSP_TCP_MAX_PACKET_SIZE 1472
66 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
67 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
68 #define RTSP_RTP_PORT_MIN 5000
69 #define RTSP_RTP_PORT_MAX 10000
72 * This describes a single item in the "Transport:" line of one stream as
73 * negotiated by the SETUP RTSP command. Multiple transports are comma-
74 * separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
75 * client_port=1000-1001;server_port=1800-1801") and described in separate
76 * RTSPTransportFields.
78 typedef struct RTSPTransportField {
79 /** interleave ids, if TCP transport; each TCP/RTSP data packet starts
80 * with a '$', stream length and stream ID. If the stream ID is within
81 * the range of this interleaved_min-max, then the packet belongs to
83 int interleaved_min, interleaved_max;
85 /** UDP multicast port range; the ports to which we should connect to
86 * receive multicast UDP data. */
87 int port_min, port_max;
89 /** UDP client ports; these should be the local ports of the UDP RTP
90 * (and RTCP) sockets over which we receive RTP/RTCP data. */
91 int client_port_min, client_port_max;
93 /** UDP unicast server port range; the ports to which we should connect
94 * to receive unicast UDP RTP/RTCP data. */
95 int server_port_min, server_port_max;
97 /** time-to-live value (required for multicast); the amount of HOPs that
98 * packets will be allowed to make before being discarded. */
101 struct sockaddr_storage destination; /**< destination IP address */
102 char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
104 /** data/packet transport protocol; e.g. RTP or RDT */
105 enum RTSPTransport transport;
107 /** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
108 enum RTSPLowerTransport lower_transport;
109 } RTSPTransportField;
112 * This describes the server response to each RTSP command.
114 typedef struct RTSPMessageHeader {
115 /** length of the data following this header */
118 enum RTSPStatusCode status_code; /**< response code from server */
120 /** number of items in the 'transports' variable below */
123 /** Time range of the streams that the server will stream. In
124 * AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
125 int64_t range_start, range_end;
127 /** describes the complete "Transport:" line of the server in response
128 * to a SETUP RTSP command by the client */
129 RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
131 int seq; /**< sequence number */
133 /** the "Session:" field. This value is initially set by the server and
134 * should be re-transmitted by the client in every RTSP command. */
135 char session_id[512];
137 /** the "Location:" field. This value is used to handle redirection.
141 /** the "RealChallenge1:" field from the server */
142 char real_challenge[64];
144 /** the "Server: field, which can be used to identify some special-case
145 * servers that are not 100% standards-compliant. We use this to identify
146 * Windows Media Server, which has a value "WMServer/v.e.r.sion", where
147 * version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
148 * use something like "Helix [..] Server Version v.e.r.sion (platform)
149 * (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
150 * where platform is the output of $uname -msr | sed 's/ /-/g'. */
153 /** The "timeout" comes as part of the server response to the "SETUP"
154 * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
155 * time, in seconds, that the server will go without traffic over the
156 * RTSP/TCP connection before it closes the connection. To prevent
157 * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
158 * than this value. */
161 /** The "Notice" or "X-Notice" field value. See
162 * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
163 * for a complete list of supported values. */
166 /** The "reason" is meant to specify better the meaning of the error code
173 * Client state, i.e. whether we are currently receiving data (PLAYING) or
174 * setup-but-not-receiving (PAUSED). State can be changed in applications
175 * by calling av_read_play/pause().
177 enum RTSPClientState {
178 RTSP_STATE_IDLE, /**< not initialized */
179 RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
180 RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
181 RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
185 * Identifies particular servers that require special handling, such as
186 * standards-incompliant "Transport:" lines in the SETUP request.
188 enum RTSPServerType {
189 RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
190 RTSP_SERVER_REAL, /**< Realmedia-style server */
191 RTSP_SERVER_WMS, /**< Windows Media server */
196 * Private data for the RTSP demuxer.
198 * @todo Use AVIOContext instead of URLContext
200 typedef struct RTSPState {
201 const AVClass *class; /**< Class for private options. */
202 URLContext *rtsp_hd; /* RTSP TCP connection handle */
204 /** number of items in the 'rtsp_streams' variable */
207 struct RTSPStream **rtsp_streams; /**< streams in this session */
209 /** indicator of whether we are currently receiving data from the
210 * server. Basically this isn't more than a simple cache of the
211 * last PLAY/PAUSE command sent to the server, to make sure we don't
212 * send 2x the same unexpectedly or commands in the wrong state. */
213 enum RTSPClientState state;
215 /** the seek value requested when calling av_seek_frame(). This value
216 * is subsequently used as part of the "Range" parameter when emitting
217 * the RTSP PLAY command. If we are currently playing, this command is
218 * called instantly. If we are currently paused, this command is called
219 * whenever we resume playback. Either way, the value is only used once,
220 * see rtsp_read_play() and rtsp_read_seek(). */
221 int64_t seek_timestamp;
223 /* XXX: currently we use unbuffered input */
224 // AVIOContext rtsp_gb;
226 int seq; /**< RTSP command sequence number */
228 /** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
229 * identifier that the client should re-transmit in each RTSP command */
230 char session_id[512];
232 /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
233 * the server will go without traffic on the RTSP/TCP line before it
234 * closes the connection. */
237 /** timestamp of the last RTSP command that we sent to the RTSP server.
238 * This is used to calculate when to send dummy commands to keep the
239 * connection alive, in conjunction with timeout. */
240 int64_t last_cmd_time;
242 /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
243 enum RTSPTransport transport;
245 /** the negotiated network layer transport protocol; e.g. TCP or UDP
247 enum RTSPLowerTransport lower_transport;
249 /** brand of server that we're talking to; e.g. WMS, REAL or other.
250 * Detected based on the value of RTSPMessageHeader->server or the presence
251 * of RTSPMessageHeader->real_challenge */
252 enum RTSPServerType server_type;
254 /** the "RealChallenge1:" field from the server */
255 char real_challenge[64];
257 /** plaintext authorization line (username:password) */
260 /** authentication state */
261 HTTPAuthState auth_state;
263 /** The last reply of the server to a RTSP command */
264 char last_reply[2048]; /* XXX: allocate ? */
266 /** RTSPStream->transport_priv of the last stream that we read a
268 void *cur_transport_priv;
270 /** The following are used for Real stream selection */
272 /** whether we need to send a "SET_PARAMETER Subscribe:" command */
273 int need_subscription;
275 /** stream setup during the last frame read. This is used to detect if
276 * we need to subscribe or unsubscribe to any new streams. */
277 enum AVDiscard *real_setup_cache;
279 /** current stream setup. This is a temporary buffer used to compare
280 * current setup to previous frame setup. */
281 enum AVDiscard *real_setup;
283 /** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
284 * this is used to send the same "Unsubscribe:" if stream setup changed,
285 * before sending a new "Subscribe:" command. */
286 char last_subscription[1024];
289 /** The following are used for RTP/ASF streams */
291 /** ASF demuxer context for the embedded ASF stream from WMS servers */
292 AVFormatContext *asf_ctx;
294 /** cache for position of the asf demuxer, since we load a new
295 * data packet in the bytecontext for each incoming RTSP packet. */
299 /** some MS RTSP streams contain a URL in the SDP that we need to use
300 * for all subsequent RTSP requests, rather than the input URI; in
301 * other cases, this is a copy of AVFormatContext->filename. */
302 char control_uri[1024];
304 /** Additional output handle, used when input and output are done
305 * separately, eg for HTTP tunneling. */
306 URLContext *rtsp_hd_out;
308 /** RTSP transport mode, such as plain or tunneled. */
309 enum RTSPControlTransport control_transport;
311 /* Number of RTCP BYE packets the RTSP session has received.
312 * An EOF is propagated back if nb_byes == nb_streams.
313 * This is reset after a seek. */
316 /** Reusable buffer for receiving packets */
319 /** Filter incoming UDP packets - receive packets only from the right
320 * source address and port. */
324 * A mask with all requested transport methods
326 int lower_transport_mask;
329 * The number of returned packets
334 * Polling array for udp
339 * Whether the server supports the GET_PARAMETER method.
341 int get_parameter_supported;
344 * Do not begin to play the stream immediately.
349 * Option flags for the chained RTP muxer.
355 * Describes a single stream, as identified by a single m= line block in the
356 * SDP content. In the case of RDT, one RTSPStream can represent multiple
357 * AVStreams. In this case, each AVStream in this set has similar content
358 * (but different codec/bitrate).
360 typedef struct RTSPStream {
361 URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
362 void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
364 /** corresponding stream index, if any. -1 if none (MPEG2TS case) */
367 /** interleave IDs; copies of RTSPTransportField->interleaved_min/max
368 * for the selected transport. Only used for TCP. */
369 int interleaved_min, interleaved_max;
371 char control_url[1024]; /**< url for this stream (from SDP) */
373 /** The following are used only in SDP, not RTSP */
375 int sdp_port; /**< port (from SDP content) */
376 struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
377 int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
378 int sdp_payload_type; /**< payload type */
381 /** The following are used for dynamic protocols (rtp_*.c/rdt.c) */
383 /** handler structure */
384 RTPDynamicProtocolHandler *dynamic_handler;
386 /** private data associated with the dynamic protocol */
387 PayloadContext *dynamic_protocol_context;
391 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
392 RTSPState *rt, const char *method);
394 extern int rtsp_rtp_port_min;
395 extern int rtsp_rtp_port_max;
398 * Send a command to the RTSP server without waiting for the reply.
400 * @see rtsp_send_cmd_with_content_async
402 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
403 const char *url, const char *headers);
406 * Send a command to the RTSP server and wait for the reply.
408 * @param s RTSP (de)muxer context
409 * @param method the method for the request
410 * @param url the target url for the request
411 * @param headers extra header lines to include in the request
412 * @param reply pointer where the RTSP message header will be stored
413 * @param content_ptr pointer where the RTSP message body, if any, will
414 * be stored (length is in reply)
415 * @param send_content if non-null, the data to send as request body content
416 * @param send_content_length the length of the send_content data, or 0 if
417 * send_content is null
419 * @return zero if success, nonzero otherwise
421 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
422 const char *method, const char *url,
424 RTSPMessageHeader *reply,
425 unsigned char **content_ptr,
426 const unsigned char *send_content,
427 int send_content_length);
430 * Send a command to the RTSP server and wait for the reply.
432 * @see rtsp_send_cmd_with_content
434 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
435 const char *url, const char *headers,
436 RTSPMessageHeader *reply, unsigned char **content_ptr);
439 * Read a RTSP message from the server, or prepare to read data
440 * packets if we're reading data interleaved over the TCP/RTSP
441 * connection as well.
443 * @param s RTSP (de)muxer context
444 * @param reply pointer where the RTSP message header will be stored
445 * @param content_ptr pointer where the RTSP message body, if any, will
446 * be stored (length is in reply)
447 * @param return_on_interleaved_data whether the function may return if we
448 * encounter a data marker ('$'), which precedes data
449 * packets over interleaved TCP/RTSP connections. If this
450 * is set, this function will return 1 after encountering
451 * a '$'. If it is not set, the function will skip any
452 * data packets (if they are encountered), until a reply
453 * has been fully parsed. If no more data is available
454 * without parsing a reply, it will return an error.
455 * @param method the RTSP method this is a reply to. This affects how
456 * some response headers are acted upon. May be NULL.
458 * @return 1 if a data packets is ready to be received, -1 on error,
461 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
462 unsigned char **content_ptr,
463 int return_on_interleaved_data, const char *method);
466 * Skip a RTP/TCP interleaved packet.
468 void ff_rtsp_skip_packet(AVFormatContext *s);
471 * Connect to the RTSP server and set up the individual media streams.
472 * This can be used for both muxers and demuxers.
474 * @param s RTSP (de)muxer context
476 * @return 0 on success, < 0 on error. Cleans up all allocations done
477 * within the function on error.
479 int ff_rtsp_connect(AVFormatContext *s);
482 * Close and free all streams within the RTSP (de)muxer
484 * @param s RTSP (de)muxer context
486 void ff_rtsp_close_streams(AVFormatContext *s);
489 * Close all connection handles within the RTSP (de)muxer
491 * @param rt RTSP (de)muxer context
493 void ff_rtsp_close_connections(AVFormatContext *rt);
496 * Get the description of the stream and set up the RTSPStream child
499 int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
502 * Announce the stream to the server and set up the RTSPStream child
503 * objects for each media stream.
505 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
508 * Parse a SDP description of streams by populating an RTSPState struct
509 * within the AVFormatContext.
511 int ff_sdp_parse(AVFormatContext *s, const char *content);
514 * Receive one RTP packet from an TCP interleaved RTSP stream.
516 int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
517 uint8_t *buf, int buf_size);
520 * Receive one packet from the RTSPStreams set up in the AVFormatContext
521 * (which should contain a RTSPState struct as priv_data).
523 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
526 * Do the SETUP requests for each stream for the chosen
527 * lower transport mode.
529 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
530 int lower_transport, const char *real_challenge);
533 * Undo the effect of ff_rtsp_make_setup_request, close the
534 * transport_priv and rtp_handle fields.
536 void ff_rtsp_undo_setup(AVFormatContext *s);
538 #endif /* AVFORMAT_RTSP_H */