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
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
34 #include "avio_internal.h"
41 #include "os_support.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
54 /* Default timeout values for read packet in seconds */
55 #define READ_PACKET_TIMEOUT_S 10
56 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
57 #define DEFAULT_REORDERING_DELAY 100000
59 #define OFFSET(x) offsetof(RTSPState, x)
60 #define DEC AV_OPT_FLAG_DECODING_PARAM
61 #define ENC AV_OPT_FLAG_ENCODING_PARAM
63 #define RTSP_FLAG_OPTS(name, longname) \
64 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
65 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
67 #define RTSP_MEDIATYPE_OPTS(name, longname) \
68 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
69 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
70 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
72 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
74 #define COMMON_OPTS() \
75 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
76 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
77 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
90 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
91 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
92 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
93 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
94 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
95 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
96 #if FF_API_OLD_RTSP_OPTIONS
97 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen) (deprecated, use listen_timeout)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC|AV_OPT_FLAG_DEPRECATED },
98 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
100 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
103 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
104 #if FF_API_OLD_RTSP_OPTIONS
105 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC|AV_OPT_FLAG_DEPRECATED },
110 static const AVOption sdp_options[] = {
111 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
112 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
113 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
114 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
115 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
120 static const AVOption rtp_options[] = {
121 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
122 { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
123 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
129 static AVDictionary *map_to_opts(RTSPState *rt)
131 AVDictionary *opts = NULL;
134 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
135 av_dict_set(&opts, "buffer_size", buf, 0);
136 snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
137 av_dict_set(&opts, "pkt_size", buf, 0);
142 static void get_word_until_chars(char *buf, int buf_size,
143 const char *sep, const char **pp)
149 p += strspn(p, SPACE_CHARS);
151 while (!strchr(sep, *p) && *p != '\0') {
152 if ((q - buf) < buf_size - 1)
161 static void get_word_sep(char *buf, int buf_size, const char *sep,
164 if (**pp == '/') (*pp)++;
165 get_word_until_chars(buf, buf_size, sep, pp);
168 static void get_word(char *buf, int buf_size, const char **pp)
170 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
173 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
175 * Used for seeking in the rtp stream.
177 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
181 p += strspn(p, SPACE_CHARS);
182 if (!av_stristart(p, "npt=", &p))
185 *start = AV_NOPTS_VALUE;
186 *end = AV_NOPTS_VALUE;
188 get_word_sep(buf, sizeof(buf), "-", &p);
189 if (av_parse_time(start, buf, 1) < 0)
193 get_word_sep(buf, sizeof(buf), "-", &p);
194 if (av_parse_time(end, buf, 1) < 0)
195 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
199 static int get_sockaddr(AVFormatContext *s,
200 const char *buf, struct sockaddr_storage *sock)
202 struct addrinfo hints = { 0 }, *ai = NULL;
205 hints.ai_flags = AI_NUMERICHOST;
206 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
207 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
212 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
218 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
219 RTSPStream *rtsp_st, AVStream *st)
221 AVCodecParameters *par = st ? st->codecpar : NULL;
225 par->codec_id = handler->codec_id;
226 rtsp_st->dynamic_handler = handler;
228 st->need_parsing = handler->need_parsing;
229 if (handler->priv_data_size) {
230 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
231 if (!rtsp_st->dynamic_protocol_context)
232 rtsp_st->dynamic_handler = NULL;
236 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
239 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
240 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
241 rtsp_st->dynamic_protocol_context);
243 if (rtsp_st->dynamic_protocol_context) {
244 if (rtsp_st->dynamic_handler->close)
245 rtsp_st->dynamic_handler->close(
246 rtsp_st->dynamic_protocol_context);
247 av_free(rtsp_st->dynamic_protocol_context);
249 rtsp_st->dynamic_protocol_context = NULL;
250 rtsp_st->dynamic_handler = NULL;
255 static int init_satip_stream(AVFormatContext *s)
257 RTSPState *rt = s->priv_data;
258 RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
260 return AVERROR(ENOMEM);
261 dynarray_add(&rt->rtsp_streams,
262 &rt->nb_rtsp_streams, rtsp_st);
264 rtsp_st->sdp_payload_type = 33; // MP2T
265 av_strlcpy(rtsp_st->control_url,
266 rt->control_uri, sizeof(rtsp_st->control_url));
268 rtsp_st->stream_index = -1;
269 init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
270 finalize_rtp_handler_init(s, rtsp_st, NULL);
274 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
275 static int sdp_parse_rtpmap(AVFormatContext *s,
276 AVStream *st, RTSPStream *rtsp_st,
277 int payload_type, const char *p)
279 AVCodecParameters *par = st->codecpar;
282 const AVCodecDescriptor *desc;
285 /* See if we can handle this kind of payload.
286 * The space should normally not be there but some Real streams or
287 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
288 * have a trailing space. */
289 get_word_sep(buf, sizeof(buf), "/ ", &p);
290 if (payload_type < RTP_PT_PRIVATE) {
291 /* We are in a standard case
292 * (from http://www.iana.org/assignments/rtp-parameters). */
293 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
296 if (par->codec_id == AV_CODEC_ID_NONE) {
297 const RTPDynamicProtocolHandler *handler =
298 ff_rtp_handler_find_by_name(buf, par->codec_type);
299 init_rtp_handler(handler, rtsp_st, st);
300 /* If no dynamic handler was found, check with the list of standard
301 * allocated types, if such a stream for some reason happens to
302 * use a private payload type. This isn't handled in rtpdec.c, since
303 * the format name from the rtpmap line never is passed into rtpdec. */
304 if (!rtsp_st->dynamic_handler)
305 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
308 desc = avcodec_descriptor_get(par->codec_id);
309 if (desc && desc->name)
314 get_word_sep(buf, sizeof(buf), "/", &p);
316 switch (par->codec_type) {
317 case AVMEDIA_TYPE_AUDIO:
318 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
319 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
320 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
322 par->sample_rate = i;
323 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
324 get_word_sep(buf, sizeof(buf), "/", &p);
329 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
331 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
334 case AVMEDIA_TYPE_VIDEO:
335 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
337 avpriv_set_pts_info(st, 32, 1, i);
342 finalize_rtp_handler_init(s, rtsp_st, st);
346 /* parse the attribute line from the fmtp a line of an sdp response. This
347 * is broken out as a function because it is used in rtp_h264.c, which is
349 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
350 char *value, int value_size)
352 *p += strspn(*p, SPACE_CHARS);
354 get_word_sep(attr, attr_size, "=", p);
357 get_word_sep(value, value_size, ";", p);
365 typedef struct SDPParseState {
367 struct sockaddr_storage default_ip;
369 int skip_media; ///< set if an unknown m= line occurs
370 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
371 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
372 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
373 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
376 char delayed_fmtp[2048];
379 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
380 struct RTSPSource ***dest, int *dest_count)
382 RTSPSource *rtsp_src, *rtsp_src2;
384 for (i = 0; i < count; i++) {
386 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
389 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
390 dynarray_add(dest, dest_count, rtsp_src2);
394 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
395 int payload_type, const char *line)
399 for (i = 0; i < rt->nb_rtsp_streams; i++) {
400 RTSPStream *rtsp_st = rt->rtsp_streams[i];
401 if (rtsp_st->sdp_payload_type == payload_type &&
402 rtsp_st->dynamic_handler &&
403 rtsp_st->dynamic_handler->parse_sdp_a_line) {
404 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
405 rtsp_st->dynamic_protocol_context, line);
410 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
411 int letter, const char *buf)
413 RTSPState *rt = s->priv_data;
414 char buf1[64], st_type[64];
416 enum AVMediaType codec_type;
420 RTSPSource *rtsp_src;
421 struct sockaddr_storage sdp_ip;
424 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
427 if (s1->skip_media && letter != 'm')
431 get_word(buf1, sizeof(buf1), &p);
432 if (strcmp(buf1, "IN") != 0)
434 get_word(buf1, sizeof(buf1), &p);
435 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
437 get_word_sep(buf1, sizeof(buf1), "/", &p);
438 if (get_sockaddr(s, buf1, &sdp_ip))
443 get_word_sep(buf1, sizeof(buf1), "/", &p);
446 if (s->nb_streams == 0) {
447 s1->default_ip = sdp_ip;
448 s1->default_ttl = ttl;
450 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
451 rtsp_st->sdp_ip = sdp_ip;
452 rtsp_st->sdp_ttl = ttl;
456 av_dict_set(&s->metadata, "title", p, 0);
459 if (s->nb_streams == 0) {
460 av_dict_set(&s->metadata, "comment", p, 0);
469 codec_type = AVMEDIA_TYPE_UNKNOWN;
470 get_word(st_type, sizeof(st_type), &p);
471 if (!strcmp(st_type, "audio")) {
472 codec_type = AVMEDIA_TYPE_AUDIO;
473 } else if (!strcmp(st_type, "video")) {
474 codec_type = AVMEDIA_TYPE_VIDEO;
475 } else if (!strcmp(st_type, "application")) {
476 codec_type = AVMEDIA_TYPE_DATA;
477 } else if (!strcmp(st_type, "text")) {
478 codec_type = AVMEDIA_TYPE_SUBTITLE;
480 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
481 !(rt->media_type_mask & (1 << codec_type)) ||
482 rt->nb_rtsp_streams >= s->max_streams
487 rtsp_st = av_mallocz(sizeof(RTSPStream));
490 rtsp_st->stream_index = -1;
491 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
493 rtsp_st->sdp_ip = s1->default_ip;
494 rtsp_st->sdp_ttl = s1->default_ttl;
496 copy_default_source_addrs(s1->default_include_source_addrs,
497 s1->nb_default_include_source_addrs,
498 &rtsp_st->include_source_addrs,
499 &rtsp_st->nb_include_source_addrs);
500 copy_default_source_addrs(s1->default_exclude_source_addrs,
501 s1->nb_default_exclude_source_addrs,
502 &rtsp_st->exclude_source_addrs,
503 &rtsp_st->nb_exclude_source_addrs);
505 get_word(buf1, sizeof(buf1), &p); /* port */
506 rtsp_st->sdp_port = atoi(buf1);
508 get_word(buf1, sizeof(buf1), &p); /* protocol */
509 if (!strcmp(buf1, "udp"))
510 rt->transport = RTSP_TRANSPORT_RAW;
511 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
512 rtsp_st->feedback = 1;
514 /* XXX: handle list of formats */
515 get_word(buf1, sizeof(buf1), &p); /* format list */
516 rtsp_st->sdp_payload_type = atoi(buf1);
518 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
519 /* no corresponding stream */
520 if (rt->transport == RTSP_TRANSPORT_RAW) {
521 if (CONFIG_RTPDEC && !rt->ts)
522 rt->ts = avpriv_mpegts_parse_open(s);
524 const RTPDynamicProtocolHandler *handler;
525 handler = ff_rtp_handler_find_by_id(
526 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
527 init_rtp_handler(handler, rtsp_st, NULL);
528 finalize_rtp_handler_init(s, rtsp_st, NULL);
530 } else if (rt->server_type == RTSP_SERVER_WMS &&
531 codec_type == AVMEDIA_TYPE_DATA) {
532 /* RTX stream, a stream that carries all the other actual
533 * audio/video streams. Don't expose this to the callers. */
535 st = avformat_new_stream(s, NULL);
538 st->id = rt->nb_rtsp_streams - 1;
539 rtsp_st->stream_index = st->index;
540 st->codecpar->codec_type = codec_type;
541 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
542 const RTPDynamicProtocolHandler *handler;
543 /* if standard payload type, we can find the codec right now */
544 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
545 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
546 st->codecpar->sample_rate > 0)
547 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
548 /* Even static payload types may need a custom depacketizer */
549 handler = ff_rtp_handler_find_by_id(
550 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
551 init_rtp_handler(handler, rtsp_st, st);
552 finalize_rtp_handler_init(s, rtsp_st, st);
554 if (rt->default_lang[0])
555 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
557 /* put a default control url */
558 av_strlcpy(rtsp_st->control_url, rt->control_uri,
559 sizeof(rtsp_st->control_url));
562 if (av_strstart(p, "control:", &p)) {
563 if (s->nb_streams == 0) {
564 if (!strncmp(p, "rtsp://", 7))
565 av_strlcpy(rt->control_uri, p,
566 sizeof(rt->control_uri));
569 /* get the control url */
570 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
572 /* XXX: may need to add full url resolution */
573 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
575 if (proto[0] == '\0') {
576 /* relative control URL */
577 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
578 av_strlcat(rtsp_st->control_url, "/",
579 sizeof(rtsp_st->control_url));
580 av_strlcat(rtsp_st->control_url, p,
581 sizeof(rtsp_st->control_url));
583 av_strlcpy(rtsp_st->control_url, p,
584 sizeof(rtsp_st->control_url));
586 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
587 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
588 get_word(buf1, sizeof(buf1), &p);
589 payload_type = atoi(buf1);
590 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
591 if (rtsp_st->stream_index >= 0) {
592 st = s->streams[rtsp_st->stream_index];
593 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
597 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
599 } else if (av_strstart(p, "fmtp:", &p) ||
600 av_strstart(p, "framesize:", &p)) {
601 // let dynamic protocol handlers have a stab at the line.
602 get_word(buf1, sizeof(buf1), &p);
603 payload_type = atoi(buf1);
604 if (s1->seen_rtpmap) {
605 parse_fmtp(s, rt, payload_type, buf);
608 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
610 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
611 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
612 get_word(buf1, sizeof(buf1), &p);
613 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
614 } else if (av_strstart(p, "range:", &p)) {
617 // this is so that seeking on a streamed file can work.
618 rtsp_parse_range_npt(p, &start, &end);
619 s->start_time = start;
620 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
621 s->duration = (end == AV_NOPTS_VALUE) ?
622 AV_NOPTS_VALUE : end - start;
623 } else if (av_strstart(p, "lang:", &p)) {
624 if (s->nb_streams > 0) {
625 get_word(buf1, sizeof(buf1), &p);
626 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
627 if (rtsp_st->stream_index >= 0) {
628 st = s->streams[rtsp_st->stream_index];
629 av_dict_set(&st->metadata, "language", buf1, 0);
632 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
633 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
635 rt->transport = RTSP_TRANSPORT_RDT;
636 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
638 st = s->streams[s->nb_streams - 1];
639 st->codecpar->sample_rate = atoi(p);
640 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
642 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
643 get_word(buf1, sizeof(buf1), &p); // ignore tag
644 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
645 p += strspn(p, SPACE_CHARS);
646 if (av_strstart(p, "inline:", &p))
647 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
648 } else if (av_strstart(p, "source-filter:", &p)) {
650 get_word(buf1, sizeof(buf1), &p);
651 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
653 exclude = !strcmp(buf1, "excl");
655 get_word(buf1, sizeof(buf1), &p);
656 if (strcmp(buf1, "IN") != 0)
658 get_word(buf1, sizeof(buf1), &p);
659 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
661 // not checking that the destination address actually matches or is wildcard
662 get_word(buf1, sizeof(buf1), &p);
665 rtsp_src = av_mallocz(sizeof(*rtsp_src));
668 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
670 if (s->nb_streams == 0) {
671 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
673 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
674 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
677 if (s->nb_streams == 0) {
678 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
680 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
681 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
686 if (rt->server_type == RTSP_SERVER_WMS)
687 ff_wms_parse_sdp_a_line(s, p);
688 if (s->nb_streams > 0) {
689 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
691 if (rt->server_type == RTSP_SERVER_REAL)
692 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
694 if (rtsp_st->dynamic_handler &&
695 rtsp_st->dynamic_handler->parse_sdp_a_line)
696 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
697 rtsp_st->stream_index,
698 rtsp_st->dynamic_protocol_context, buf);
705 int ff_sdp_parse(AVFormatContext *s, const char *content)
709 char buf[SDP_MAX_SIZE], *q;
710 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
714 p += strspn(p, SPACE_CHARS);
722 /* get the content */
724 while (*p != '\n' && *p != '\r' && *p != '\0') {
725 if ((q - buf) < sizeof(buf) - 1)
730 sdp_parse_line(s, s1, letter, buf);
732 while (*p != '\n' && *p != '\0')
738 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
739 av_freep(&s1->default_include_source_addrs[i]);
740 av_freep(&s1->default_include_source_addrs);
741 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
742 av_freep(&s1->default_exclude_source_addrs[i]);
743 av_freep(&s1->default_exclude_source_addrs);
747 #endif /* CONFIG_RTPDEC */
749 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
751 RTSPState *rt = s->priv_data;
754 for (i = 0; i < rt->nb_rtsp_streams; i++) {
755 RTSPStream *rtsp_st = rt->rtsp_streams[i];
758 if (rtsp_st->transport_priv) {
760 AVFormatContext *rtpctx = rtsp_st->transport_priv;
761 av_write_trailer(rtpctx);
762 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
763 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
764 ff_rtsp_tcp_write_packet(s, rtsp_st);
765 ffio_free_dyn_buf(&rtpctx->pb);
767 avio_closep(&rtpctx->pb);
769 avformat_free_context(rtpctx);
770 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
771 ff_rdt_parse_close(rtsp_st->transport_priv);
772 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
773 ff_rtp_parse_close(rtsp_st->transport_priv);
775 rtsp_st->transport_priv = NULL;
776 ffurl_closep(&rtsp_st->rtp_handle);
780 /* close and free RTSP streams */
781 void ff_rtsp_close_streams(AVFormatContext *s)
783 RTSPState *rt = s->priv_data;
787 ff_rtsp_undo_setup(s, 0);
788 for (i = 0; i < rt->nb_rtsp_streams; i++) {
789 rtsp_st = rt->rtsp_streams[i];
791 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
792 if (rtsp_st->dynamic_handler->close)
793 rtsp_st->dynamic_handler->close(
794 rtsp_st->dynamic_protocol_context);
795 av_free(rtsp_st->dynamic_protocol_context);
797 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
798 av_freep(&rtsp_st->include_source_addrs[j]);
799 av_freep(&rtsp_st->include_source_addrs);
800 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
801 av_freep(&rtsp_st->exclude_source_addrs[j]);
802 av_freep(&rtsp_st->exclude_source_addrs);
807 av_freep(&rt->rtsp_streams);
809 avformat_close_input(&rt->asf_ctx);
811 if (CONFIG_RTPDEC && rt->ts)
812 avpriv_mpegts_parse_close(rt->ts);
814 av_freep(&rt->recvbuf);
817 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
819 RTSPState *rt = s->priv_data;
821 int reordering_queue_size = rt->reordering_queue_size;
822 if (reordering_queue_size < 0) {
823 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
824 reordering_queue_size = 0;
826 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
829 /* open the RTP context */
830 if (rtsp_st->stream_index >= 0)
831 st = s->streams[rtsp_st->stream_index];
833 s->ctx_flags |= AVFMTCTX_NOHEADER;
835 if (CONFIG_RTSP_MUXER && s->oformat && st) {
836 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
837 s, st, rtsp_st->rtp_handle,
838 RTSP_TCP_MAX_PACKET_SIZE,
839 rtsp_st->stream_index);
840 /* Ownership of rtp_handle is passed to the rtp mux context */
841 rtsp_st->rtp_handle = NULL;
844 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
845 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
846 return 0; // Don't need to open any parser here
847 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
848 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
849 rtsp_st->dynamic_protocol_context,
850 rtsp_st->dynamic_handler);
851 else if (CONFIG_RTPDEC)
852 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
853 rtsp_st->sdp_payload_type,
854 reordering_queue_size);
856 if (!rtsp_st->transport_priv) {
857 return AVERROR(ENOMEM);
858 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
860 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
861 rtpctx->ssrc = rtsp_st->ssrc;
862 if (rtsp_st->dynamic_handler) {
863 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
864 rtsp_st->dynamic_protocol_context,
865 rtsp_st->dynamic_handler);
867 if (rtsp_st->crypto_suite[0])
868 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
869 rtsp_st->crypto_suite,
870 rtsp_st->crypto_params);
876 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
877 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
884 q += strspn(q, SPACE_CHARS);
885 v = strtol(q, &p, 10);
889 v = strtol(p, &p, 10);
898 /* XXX: only one transport specification is parsed */
899 static void rtsp_parse_transport(AVFormatContext *s,
900 RTSPMessageHeader *reply, const char *p)
902 char transport_protocol[16];
904 char lower_transport[16];
906 RTSPTransportField *th;
909 reply->nb_transports = 0;
912 p += strspn(p, SPACE_CHARS);
916 th = &reply->transports[reply->nb_transports];
918 get_word_sep(transport_protocol, sizeof(transport_protocol),
920 if (!av_strcasecmp (transport_protocol, "rtp")) {
921 get_word_sep(profile, sizeof(profile), "/;,", &p);
922 lower_transport[0] = '\0';
923 /* rtp/avp/<protocol> */
925 get_word_sep(lower_transport, sizeof(lower_transport),
928 th->transport = RTSP_TRANSPORT_RTP;
929 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
930 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
931 /* x-pn-tng/<protocol> */
932 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
934 th->transport = RTSP_TRANSPORT_RDT;
935 } else if (!av_strcasecmp(transport_protocol, "raw")) {
936 get_word_sep(profile, sizeof(profile), "/;,", &p);
937 lower_transport[0] = '\0';
938 /* raw/raw/<protocol> */
940 get_word_sep(lower_transport, sizeof(lower_transport),
943 th->transport = RTSP_TRANSPORT_RAW;
945 if (!av_strcasecmp(lower_transport, "TCP"))
946 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
948 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
952 /* get each parameter */
953 while (*p != '\0' && *p != ',') {
954 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
955 if (!strcmp(parameter, "port")) {
958 rtsp_parse_range(&th->port_min, &th->port_max, &p);
960 } else if (!strcmp(parameter, "client_port")) {
963 rtsp_parse_range(&th->client_port_min,
964 &th->client_port_max, &p);
966 } else if (!strcmp(parameter, "server_port")) {
969 rtsp_parse_range(&th->server_port_min,
970 &th->server_port_max, &p);
972 } else if (!strcmp(parameter, "interleaved")) {
975 rtsp_parse_range(&th->interleaved_min,
976 &th->interleaved_max, &p);
978 } else if (!strcmp(parameter, "multicast")) {
979 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
980 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
981 } else if (!strcmp(parameter, "ttl")) {
985 th->ttl = strtol(p, &end, 10);
988 } else if (!strcmp(parameter, "destination")) {
991 get_word_sep(buf, sizeof(buf), ";,", &p);
992 get_sockaddr(s, buf, &th->destination);
994 } else if (!strcmp(parameter, "source")) {
997 get_word_sep(buf, sizeof(buf), ";,", &p);
998 av_strlcpy(th->source, buf, sizeof(th->source));
1000 } else if (!strcmp(parameter, "mode")) {
1003 get_word_sep(buf, sizeof(buf), ";, ", &p);
1004 if (!strcmp(buf, "record") ||
1005 !strcmp(buf, "receive"))
1006 th->mode_record = 1;
1010 while (*p != ';' && *p != '\0' && *p != ',')
1018 reply->nb_transports++;
1019 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1024 static void handle_rtp_info(RTSPState *rt, const char *url,
1025 uint32_t seq, uint32_t rtptime)
1028 if (!rtptime || !url[0])
1030 if (rt->transport != RTSP_TRANSPORT_RTP)
1032 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1033 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1034 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1037 if (!strcmp(rtsp_st->control_url, url)) {
1038 rtpctx->base_timestamp = rtptime;
1044 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1047 char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
1048 uint32_t seq = 0, rtptime = 0;
1051 p += strspn(p, SPACE_CHARS);
1054 get_word_sep(key, sizeof(key), "=", &p);
1058 get_word_sep(value, sizeof(value), ";, ", &p);
1060 if (!strcmp(key, "url"))
1061 av_strlcpy(url, value, sizeof(url));
1062 else if (!strcmp(key, "seq"))
1063 seq = strtoul(value, NULL, 10);
1064 else if (!strcmp(key, "rtptime"))
1065 rtptime = strtoul(value, NULL, 10);
1067 handle_rtp_info(rt, url, seq, rtptime);
1076 handle_rtp_info(rt, url, seq, rtptime);
1079 void ff_rtsp_parse_line(AVFormatContext *s,
1080 RTSPMessageHeader *reply, const char *buf,
1081 RTSPState *rt, const char *method)
1085 /* NOTE: we do case independent match for broken servers */
1087 if (av_stristart(p, "Session:", &p)) {
1089 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1090 if (av_stristart(p, ";timeout=", &p) &&
1091 (t = strtol(p, NULL, 10)) > 0) {
1094 } else if (av_stristart(p, "Content-Length:", &p)) {
1095 reply->content_length = strtol(p, NULL, 10);
1096 } else if (av_stristart(p, "Transport:", &p)) {
1097 rtsp_parse_transport(s, reply, p);
1098 } else if (av_stristart(p, "CSeq:", &p)) {
1099 reply->seq = strtol(p, NULL, 10);
1100 } else if (av_stristart(p, "Range:", &p)) {
1101 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1102 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1103 p += strspn(p, SPACE_CHARS);
1104 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1105 } else if (av_stristart(p, "Server:", &p)) {
1106 p += strspn(p, SPACE_CHARS);
1107 av_strlcpy(reply->server, p, sizeof(reply->server));
1108 } else if (av_stristart(p, "Notice:", &p) ||
1109 av_stristart(p, "X-Notice:", &p)) {
1110 reply->notice = strtol(p, NULL, 10);
1111 } else if (av_stristart(p, "Location:", &p)) {
1112 p += strspn(p, SPACE_CHARS);
1113 av_strlcpy(reply->location, p , sizeof(reply->location));
1114 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1115 p += strspn(p, SPACE_CHARS);
1116 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1117 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1118 p += strspn(p, SPACE_CHARS);
1119 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1120 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1121 p += strspn(p, SPACE_CHARS);
1122 if (method && !strcmp(method, "DESCRIBE"))
1123 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1124 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1125 p += strspn(p, SPACE_CHARS);
1126 if (method && !strcmp(method, "PLAY"))
1127 rtsp_parse_rtp_info(rt, p);
1128 } else if (av_stristart(p, "Public:", &p) && rt) {
1129 if (strstr(p, "GET_PARAMETER") &&
1130 method && !strcmp(method, "OPTIONS"))
1131 rt->get_parameter_supported = 1;
1132 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1133 p += strspn(p, SPACE_CHARS);
1134 rt->accept_dynamic_rate = atoi(p);
1135 } else if (av_stristart(p, "Content-Type:", &p)) {
1136 p += strspn(p, SPACE_CHARS);
1137 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1138 } else if (av_stristart(p, "com.ses.streamID:", &p)) {
1139 p += strspn(p, SPACE_CHARS);
1140 av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
1144 /* skip a RTP/TCP interleaved packet */
1145 void ff_rtsp_skip_packet(AVFormatContext *s)
1147 RTSPState *rt = s->priv_data;
1149 uint8_t buf[MAX_URL_SIZE];
1151 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1154 len = AV_RB16(buf + 1);
1156 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1161 if (len1 > sizeof(buf))
1163 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1170 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1171 unsigned char **content_ptr,
1172 int return_on_interleaved_data, const char *method)
1174 RTSPState *rt = s->priv_data;
1175 char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
1178 int ret, content_length, line_count = 0, request = 0;
1179 unsigned char *content = NULL;
1185 memset(reply, 0, sizeof(*reply));
1187 /* parse reply (XXX: use buffers) */
1188 rt->last_reply[0] = '\0';
1192 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1193 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1198 if (ch == '$' && q == buf) {
1199 if (return_on_interleaved_data) {
1202 ff_rtsp_skip_packet(s);
1203 } else if (ch != '\r') {
1204 if ((q - buf) < sizeof(buf) - 1)
1210 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1212 /* test if last line */
1216 if (line_count == 0) {
1217 /* get reply code */
1218 get_word(buf1, sizeof(buf1), &p);
1219 if (!strncmp(buf1, "RTSP/", 5)) {
1220 get_word(buf1, sizeof(buf1), &p);
1221 reply->status_code = atoi(buf1);
1222 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1224 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1225 get_word(buf1, sizeof(buf1), &p); // object
1229 ff_rtsp_parse_line(s, reply, p, rt, method);
1230 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1231 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1236 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1237 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1239 content_length = reply->content_length;
1240 if (content_length > 0) {
1241 /* leave some room for a trailing '\0' (useful for simple parsing) */
1242 content = av_malloc(content_length + 1);
1244 return AVERROR(ENOMEM);
1245 if (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length)
1246 return AVERROR(EIO);
1247 content[content_length] = '\0';
1250 *content_ptr = content;
1255 char buf[MAX_URL_SIZE];
1256 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1257 const char* ptr = buf;
1259 if (!strcmp(reply->reason, "OPTIONS")) {
1260 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1262 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1263 if (reply->session_id[0])
1264 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1267 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1269 av_strlcat(buf, "\r\n", sizeof(buf));
1271 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1272 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1275 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1277 rt->last_cmd_time = av_gettime_relative();
1278 /* Even if the request from the server had data, it is not the data
1279 * that the caller wants or expects. The memory could also be leaked
1280 * if the actual following reply has content data. */
1282 av_freep(content_ptr);
1283 /* If method is set, this is called from ff_rtsp_send_cmd,
1284 * where a reply to exactly this request is awaited. For
1285 * callers from within packet receiving, we just want to
1286 * return to the caller and go back to receiving packets. */
1292 if (rt->seq != reply->seq) {
1293 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1294 rt->seq, reply->seq);
1298 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1299 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1300 reply->notice == 2306 /* Continuous Feed Terminated */) {
1301 rt->state = RTSP_STATE_IDLE;
1302 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1303 return AVERROR(EIO); /* data or server error */
1304 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1305 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1306 return AVERROR(EPERM);
1312 * Send a command to the RTSP server without waiting for the reply.
1314 * @param s RTSP (de)muxer context
1315 * @param method the method for the request
1316 * @param url the target url for the request
1317 * @param headers extra header lines to include in the request
1318 * @param send_content if non-null, the data to send as request body content
1319 * @param send_content_length the length of the send_content data, or 0 if
1320 * send_content is null
1322 * @return zero if success, nonzero otherwise
1324 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1325 const char *method, const char *url,
1326 const char *headers,
1327 const unsigned char *send_content,
1328 int send_content_length)
1330 RTSPState *rt = s->priv_data;
1331 char buf[MAX_URL_SIZE], *out_buf;
1332 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1334 if (!rt->rtsp_hd_out)
1335 return AVERROR(ENOTCONN);
1337 /* Add in RTSP headers */
1340 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1342 av_strlcat(buf, headers, sizeof(buf));
1343 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1344 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1345 if (rt->session_id[0] != '\0' && (!headers ||
1346 !strstr(headers, "\nIf-Match:"))) {
1347 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1350 char *str = ff_http_auth_create_response(&rt->auth_state,
1351 rt->auth, url, method);
1353 av_strlcat(buf, str, sizeof(buf));
1356 if (send_content_length > 0 && send_content)
1357 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1358 av_strlcat(buf, "\r\n", sizeof(buf));
1360 /* base64 encode rtsp if tunneling */
1361 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1362 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1363 out_buf = base64buf;
1366 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1368 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1369 if (send_content_length > 0 && send_content) {
1370 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1371 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1372 return AVERROR_PATCHWELCOME;
1374 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1376 rt->last_cmd_time = av_gettime_relative();
1381 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1382 const char *url, const char *headers)
1384 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1387 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1388 const char *headers, RTSPMessageHeader *reply,
1389 unsigned char **content_ptr)
1391 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1392 content_ptr, NULL, 0);
1395 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1396 const char *method, const char *url,
1398 RTSPMessageHeader *reply,
1399 unsigned char **content_ptr,
1400 const unsigned char *send_content,
1401 int send_content_length)
1403 RTSPState *rt = s->priv_data;
1404 HTTPAuthType cur_auth_type;
1405 int ret, attempts = 0;
1408 cur_auth_type = rt->auth_state.auth_type;
1409 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1411 send_content_length)))
1414 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1418 if (reply->status_code == 401 &&
1419 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1420 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1423 if (reply->status_code > 400){
1424 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1428 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1434 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1435 int lower_transport, const char *real_challenge)
1437 RTSPState *rt = s->priv_data;
1438 int rtx = 0, j, i, err, interleave = 0, port_off;
1439 RTSPStream *rtsp_st;
1440 RTSPMessageHeader reply1, *reply = &reply1;
1441 char cmd[MAX_URL_SIZE];
1442 const char *trans_pref;
1444 if (rt->transport == RTSP_TRANSPORT_RDT)
1445 trans_pref = "x-pn-tng";
1446 else if (rt->transport == RTSP_TRANSPORT_RAW)
1447 trans_pref = "RAW/RAW";
1449 trans_pref = "RTP/AVP";
1451 /* default timeout: 1 minute */
1454 /* Choose a random starting offset within the first half of the
1455 * port range, to allow for a number of ports to try even if the offset
1456 * happens to be at the end of the random range. */
1457 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1458 /* even random offset */
1459 port_off -= port_off & 0x01;
1461 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1462 char transport[MAX_URL_SIZE];
1465 * WMS serves all UDP data over a single connection, the RTX, which
1466 * isn't necessarily the first in the SDP but has to be the first
1467 * to be set up, else the second/third SETUP will fail with a 461.
1469 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1470 rt->server_type == RTSP_SERVER_WMS) {
1473 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1474 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1476 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1480 if (rtx == rt->nb_rtsp_streams)
1481 return -1; /* no RTX found */
1482 rtsp_st = rt->rtsp_streams[rtx];
1484 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1486 rtsp_st = rt->rtsp_streams[i];
1489 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1492 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1493 port = reply->transports[0].client_port_min;
1497 /* first try in specified port range */
1498 while (j <= rt->rtp_port_max) {
1499 AVDictionary *opts = map_to_opts(rt);
1501 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1502 "?localport=%d", j);
1503 /* we will use two ports per rtp stream (rtp and rtcp) */
1505 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1506 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1508 av_dict_free(&opts);
1513 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1518 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1520 av_strlcpy(transport, trans_pref, sizeof(transport));
1521 av_strlcat(transport,
1522 rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
1524 if (rt->server_type != RTSP_SERVER_REAL)
1525 av_strlcat(transport, "unicast;", sizeof(transport));
1526 av_strlcatf(transport, sizeof(transport),
1527 "client_port=%d", port);
1528 if (rt->transport == RTSP_TRANSPORT_RTP &&
1529 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1530 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1534 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1535 /* For WMS streams, the application streams are only used for
1536 * UDP. When trying to set it up for TCP streams, the server
1537 * will return an error. Therefore, we skip those streams. */
1538 if (rt->server_type == RTSP_SERVER_WMS &&
1539 (rtsp_st->stream_index < 0 ||
1540 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1543 snprintf(transport, sizeof(transport) - 1,
1544 "%s/TCP;", trans_pref);
1545 if (rt->transport != RTSP_TRANSPORT_RDT)
1546 av_strlcat(transport, "unicast;", sizeof(transport));
1547 av_strlcatf(transport, sizeof(transport),
1548 "interleaved=%d-%d",
1549 interleave, interleave + 1);
1553 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1554 snprintf(transport, sizeof(transport) - 1,
1555 "%s/UDP;multicast", trans_pref);
1558 av_strlcat(transport, ";mode=record", sizeof(transport));
1559 } else if (rt->server_type == RTSP_SERVER_REAL ||
1560 rt->server_type == RTSP_SERVER_WMS)
1561 av_strlcat(transport, ";mode=play", sizeof(transport));
1562 snprintf(cmd, sizeof(cmd),
1563 "Transport: %s\r\n",
1565 if (rt->accept_dynamic_rate)
1566 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1567 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1568 char real_res[41], real_csum[9];
1569 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1571 av_strlcatf(cmd, sizeof(cmd),
1573 "RealChallenge2: %s, sd=%s\r\n",
1574 rt->session_id, real_res, real_csum);
1576 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1577 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1580 } else if (reply->status_code != RTSP_STATUS_OK ||
1581 reply->nb_transports != 1) {
1582 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1586 if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
1587 char proto[128], host[128], path[512], auth[128];
1589 av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
1590 &port, path, sizeof(path), rt->control_uri);
1591 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
1592 port, "/stream=%s", reply->stream_id);
1595 /* XXX: same protocol for all streams is required */
1597 if (reply->transports[0].lower_transport != rt->lower_transport ||
1598 reply->transports[0].transport != rt->transport) {
1599 err = AVERROR_INVALIDDATA;
1603 rt->lower_transport = reply->transports[0].lower_transport;
1604 rt->transport = reply->transports[0].transport;
1607 /* Fail if the server responded with another lower transport mode
1608 * than what we requested. */
1609 if (reply->transports[0].lower_transport != lower_transport) {
1610 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1611 err = AVERROR_INVALIDDATA;
1615 switch(reply->transports[0].lower_transport) {
1616 case RTSP_LOWER_TRANSPORT_TCP:
1617 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1618 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1621 case RTSP_LOWER_TRANSPORT_UDP: {
1622 char url[MAX_URL_SIZE], options[30] = "";
1623 const char *peer = host;
1625 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1626 av_strlcpy(options, "?connect=1", sizeof(options));
1627 /* Use source address if specified */
1628 if (reply->transports[0].source[0])
1629 peer = reply->transports[0].source;
1630 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1631 reply->transports[0].server_port_min, "%s", options);
1632 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1633 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1634 err = AVERROR_INVALIDDATA;
1639 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1640 char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
1641 struct sockaddr_storage addr;
1643 AVDictionary *opts = map_to_opts(rt);
1645 if (reply->transports[0].destination.ss_family) {
1646 addr = reply->transports[0].destination;
1647 port = reply->transports[0].port_min;
1648 ttl = reply->transports[0].ttl;
1650 addr = rtsp_st->sdp_ip;
1651 port = rtsp_st->sdp_port;
1652 ttl = rtsp_st->sdp_ttl;
1655 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1656 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1657 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1658 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1659 port, "%s", optbuf);
1660 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1661 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1662 av_dict_free(&opts);
1665 err = AVERROR_INVALIDDATA;
1672 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1676 if (rt->nb_rtsp_streams && reply->timeout > 0)
1677 rt->timeout = reply->timeout;
1679 if (rt->server_type == RTSP_SERVER_REAL)
1680 rt->need_subscription = 1;
1685 ff_rtsp_undo_setup(s, 0);
1689 void ff_rtsp_close_connections(AVFormatContext *s)
1691 RTSPState *rt = s->priv_data;
1692 if (rt->rtsp_hd_out != rt->rtsp_hd)
1693 ffurl_closep(&rt->rtsp_hd_out);
1694 rt->rtsp_hd_out = NULL;
1695 ffurl_closep(&rt->rtsp_hd);
1698 int ff_rtsp_connect(AVFormatContext *s)
1700 RTSPState *rt = s->priv_data;
1701 char proto[128], host[1024], path[1024];
1702 char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
1703 const char *lower_rtsp_proto = "tcp";
1704 int port, err, tcp_fd;
1705 RTSPMessageHeader reply1, *reply = &reply1;
1706 int lower_transport_mask = 0;
1707 int default_port = RTSP_DEFAULT_PORT;
1708 int https_tunnel = 0;
1709 char real_challenge[64] = "";
1710 struct sockaddr_storage peer;
1711 socklen_t peer_len = sizeof(peer);
1713 if (rt->rtp_port_max < rt->rtp_port_min) {
1714 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1715 "than min port %d\n", rt->rtp_port_max,
1717 return AVERROR(EINVAL);
1720 if (!ff_network_init())
1721 return AVERROR(EIO);
1723 if (s->max_delay < 0) /* Not set by the caller */
1724 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1726 rt->control_transport = RTSP_MODE_PLAIN;
1727 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1728 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1729 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1730 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1731 rt->control_transport = RTSP_MODE_TUNNEL;
1733 /* Only pass through valid flags from here */
1734 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1737 memset(&reply1, 0, sizeof(reply1));
1738 /* extract hostname and port */
1739 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1740 host, sizeof(host), &port, path, sizeof(path), s->url);
1742 if (!strcmp(proto, "rtsps")) {
1743 lower_rtsp_proto = "tls";
1744 default_port = RTSPS_DEFAULT_PORT;
1745 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1746 } else if (!strcmp(proto, "satip")) {
1747 av_strlcpy(proto, "rtsp", sizeof(proto));
1748 rt->server_type = RTSP_SERVER_SATIP;
1752 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1755 port = default_port;
1757 lower_transport_mask = rt->lower_transport_mask;
1759 if (!lower_transport_mask)
1760 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1763 /* Only UDP or TCP - UDP multicast isn't supported. */
1764 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1765 (1 << RTSP_LOWER_TRANSPORT_TCP);
1766 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1767 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1768 "only UDP and TCP are supported for output.\n");
1769 err = AVERROR(EINVAL);
1774 /* Construct the URI used in request; this is similar to s->url,
1775 * but with authentication credentials removed and RTSP specific options
1777 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1778 host, port, "%s", path);
1780 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1781 /* set up initial handshake for tunneling */
1782 char httpname[1024];
1783 char sessioncookie[17];
1785 AVDictionary *options = NULL;
1787 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1789 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1790 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1791 av_get_random_seed(), av_get_random_seed());
1794 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1795 &s->interrupt_callback) < 0) {
1800 /* generate GET headers */
1801 snprintf(headers, sizeof(headers),
1802 "x-sessioncookie: %s\r\n"
1803 "Accept: application/x-rtsp-tunnelled\r\n"
1804 "Pragma: no-cache\r\n"
1805 "Cache-Control: no-cache\r\n",
1807 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1809 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1810 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1811 if (!rt->rtsp_hd->protocol_whitelist) {
1812 err = AVERROR(ENOMEM);
1817 /* complete the connection */
1818 if (ffurl_connect(rt->rtsp_hd, &options)) {
1819 av_dict_free(&options);
1825 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1826 &s->interrupt_callback) < 0 ) {
1831 /* generate POST headers */
1832 snprintf(headers, sizeof(headers),
1833 "x-sessioncookie: %s\r\n"
1834 "Content-Type: application/x-rtsp-tunnelled\r\n"
1835 "Pragma: no-cache\r\n"
1836 "Cache-Control: no-cache\r\n"
1837 "Content-Length: 32767\r\n"
1838 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1840 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1841 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1842 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1844 /* Initialize the authentication state for the POST session. The HTTP
1845 * protocol implementation doesn't properly handle multi-pass
1846 * authentication for POST requests, since it would require one of
1848 * - implementing Expect: 100-continue, which many HTTP servers
1849 * don't support anyway, even less the RTSP servers that do HTTP
1851 * - sending the whole POST data until getting a 401 reply specifying
1852 * what authentication method to use, then resending all that data
1853 * - waiting for potential 401 replies directly after sending the
1854 * POST header (waiting for some unspecified time)
1855 * Therefore, we copy the full auth state, which works for both basic
1856 * and digest. (For digest, we would have to synchronize the nonce
1857 * count variable between the two sessions, if we'd do more requests
1858 * with the original session, though.)
1860 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1862 /* complete the connection */
1863 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1864 av_dict_free(&options);
1868 av_dict_free(&options);
1871 /* open the tcp connection */
1872 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1874 "?timeout=%d", rt->stimeout);
1875 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1876 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1880 rt->rtsp_hd_out = rt->rtsp_hd;
1884 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1889 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1890 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1891 NULL, 0, NI_NUMERICHOST);
1894 /* request options supported by the server; this also detects server
1896 if (rt->server_type != RTSP_SERVER_SATIP)
1897 rt->server_type = RTSP_SERVER_RTP;
1900 if (rt->server_type == RTSP_SERVER_REAL)
1903 * The following entries are required for proper
1904 * streaming from a Realmedia server. They are
1905 * interdependent in some way although we currently
1906 * don't quite understand how. Values were copied
1907 * from mplayer SVN r23589.
1908 * ClientChallenge is a 16-byte ID in hex
1909 * CompanyID is a 16-byte ID in base64
1911 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1912 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1913 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1914 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1916 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1917 if (reply->status_code != RTSP_STATUS_OK) {
1918 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1922 /* detect server type if not standard-compliant RTP */
1923 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1924 rt->server_type = RTSP_SERVER_REAL;
1926 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1927 rt->server_type = RTSP_SERVER_WMS;
1928 } else if (rt->server_type == RTSP_SERVER_REAL)
1929 strcpy(real_challenge, reply->real_challenge);
1933 if (CONFIG_RTSP_DEMUXER && s->iformat) {
1934 if (rt->server_type == RTSP_SERVER_SATIP)
1935 err = init_satip_stream(s);
1937 err = ff_rtsp_setup_input_streams(s, reply);
1938 } else if (CONFIG_RTSP_MUXER)
1939 err = ff_rtsp_setup_output_streams(s, host);
1946 int lower_transport = ff_log2_tab[lower_transport_mask &
1947 ~(lower_transport_mask - 1)];
1949 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1950 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1951 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1953 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1954 rt->server_type == RTSP_SERVER_REAL ?
1955 real_challenge : NULL);
1958 lower_transport_mask &= ~(1 << lower_transport);
1959 if (lower_transport_mask == 0 && err == 1) {
1960 err = AVERROR(EPROTONOSUPPORT);
1965 rt->lower_transport_mask = lower_transport_mask;
1966 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1967 rt->state = RTSP_STATE_IDLE;
1968 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1971 ff_rtsp_close_streams(s);
1972 ff_rtsp_close_connections(s);
1973 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1974 char *new_url = av_strdup(reply->location);
1976 err = AVERROR(ENOMEM);
1979 ff_format_set_url(s, new_url);
1980 rt->session_id[0] = '\0';
1981 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1990 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1993 static int parse_rtsp_message(AVFormatContext *s)
1995 RTSPState *rt = s->priv_data;
1998 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1999 if (rt->state == RTSP_STATE_STREAMING) {
2000 return ff_rtsp_parse_streaming_commands(s);
2004 RTSPMessageHeader reply;
2005 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2008 /* XXX: parse message */
2009 if (rt->state != RTSP_STATE_STREAMING)
2016 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
2017 uint8_t *buf, int buf_size, int64_t wait_end)
2019 RTSPState *rt = s->priv_data;
2020 RTSPStream *rtsp_st;
2022 struct pollfd *p = rt->p;
2023 int *fds = NULL, fdsnum, fdsidx;
2024 int runs = rt->initial_timeout * 1000LL / POLLING_TIME;
2027 p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
2029 return AVERROR(ENOMEM);
2032 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2033 p[rt->max_p++].events = POLLIN;
2035 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2036 rtsp_st = rt->rtsp_streams[i];
2037 if (rtsp_st->rtp_handle) {
2038 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2040 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2044 av_log(s, AV_LOG_ERROR,
2045 "Number of fds %d not supported\n", fdsnum);
2046 return AVERROR_INVALIDDATA;
2048 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2049 p[rt->max_p].fd = fds[fdsidx];
2050 p[rt->max_p++].events = POLLIN;
2058 if (ff_check_interrupt(&s->interrupt_callback))
2059 return AVERROR_EXIT;
2060 if (wait_end && wait_end - av_gettime_relative() < 0)
2061 return AVERROR(EAGAIN);
2062 n = poll(p, rt->max_p, POLLING_TIME);
2064 int j = rt->rtsp_hd ? 1 : 0;
2065 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2066 rtsp_st = rt->rtsp_streams[i];
2067 if (rtsp_st->rtp_handle) {
2068 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2069 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2071 *prtsp_st = rtsp_st;
2078 #if CONFIG_RTSP_DEMUXER
2079 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2080 if ((ret = parse_rtsp_message(s)) < 0) {
2085 } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) {
2086 return AVERROR(ETIMEDOUT);
2087 } else if (n < 0 && errno != EINTR)
2088 return AVERROR(errno);
2092 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2093 const uint8_t *buf, int len)
2095 RTSPState *rt = s->priv_data;
2099 if (rt->nb_rtsp_streams == 1) {
2100 *rtsp_st = rt->rtsp_streams[0];
2103 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2104 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2106 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2107 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2110 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2111 *rtsp_st = rt->rtsp_streams[i];
2118 av_log(s, AV_LOG_WARNING,
2119 "Unable to pick stream for packet - SSRC not known for "
2121 return AVERROR(EAGAIN);
2124 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2125 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2126 *rtsp_st = rt->rtsp_streams[i];
2132 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2133 return AVERROR(EAGAIN);
2136 static int read_packet(AVFormatContext *s,
2137 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2140 RTSPState *rt = s->priv_data;
2143 switch(rt->lower_transport) {
2145 #if CONFIG_RTSP_DEMUXER
2146 case RTSP_LOWER_TRANSPORT_TCP:
2147 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2150 case RTSP_LOWER_TRANSPORT_UDP:
2151 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2152 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2153 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2154 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2156 case RTSP_LOWER_TRANSPORT_CUSTOM:
2157 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2158 wait_end && wait_end < av_gettime_relative())
2159 len = AVERROR(EAGAIN);
2161 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2162 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2163 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2164 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2174 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2176 RTSPState *rt = s->priv_data;
2178 RTSPStream *rtsp_st, *first_queue_st = NULL;
2179 int64_t wait_end = 0;
2181 if (rt->nb_byes == rt->nb_rtsp_streams)
2184 /* get next frames from the same RTP packet */
2185 if (rt->cur_transport_priv) {
2186 if (rt->transport == RTSP_TRANSPORT_RDT) {
2187 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2188 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2189 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2190 } else if (CONFIG_RTPDEC && rt->ts) {
2191 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2193 rt->recvbuf_pos += ret;
2194 ret = rt->recvbuf_pos < rt->recvbuf_len;
2199 rt->cur_transport_priv = NULL;
2201 } else if (ret == 1) {
2204 rt->cur_transport_priv = NULL;
2208 if (rt->transport == RTSP_TRANSPORT_RTP) {
2210 int64_t first_queue_time = 0;
2211 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2212 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2216 queue_time = ff_rtp_queued_packet_time(rtpctx);
2217 if (queue_time && (queue_time - first_queue_time < 0 ||
2218 !first_queue_time)) {
2219 first_queue_time = queue_time;
2220 first_queue_st = rt->rtsp_streams[i];
2223 if (first_queue_time) {
2224 wait_end = first_queue_time + s->max_delay;
2227 first_queue_st = NULL;
2231 /* read next RTP packet */
2233 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2235 return AVERROR(ENOMEM);
2238 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2239 if (len == AVERROR(EAGAIN) && first_queue_st &&
2240 rt->transport == RTSP_TRANSPORT_RTP) {
2241 av_log(s, AV_LOG_WARNING,
2242 "max delay reached. need to consume packet\n");
2243 rtsp_st = first_queue_st;
2244 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2250 if (rt->transport == RTSP_TRANSPORT_RDT) {
2251 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2252 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2253 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2254 if (rtsp_st->feedback) {
2255 AVIOContext *pb = NULL;
2256 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2258 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2261 /* Either bad packet, or a RTCP packet. Check if the
2262 * first_rtcp_ntp_time field was initialized. */
2263 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2264 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2265 /* first_rtcp_ntp_time has been initialized for this stream,
2266 * copy the same value to all other uninitialized streams,
2267 * in order to map their timestamp origin to the same ntp time
2270 AVStream *st = NULL;
2271 if (rtsp_st->stream_index >= 0)
2272 st = s->streams[rtsp_st->stream_index];
2273 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2274 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2275 AVStream *st2 = NULL;
2276 if (rt->rtsp_streams[i]->stream_index >= 0)
2277 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2278 if (rtpctx2 && st && st2 &&
2279 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2280 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2281 rtpctx2->rtcp_ts_offset = av_rescale_q(
2282 rtpctx->rtcp_ts_offset, st->time_base,
2286 // Make real NTP start time available in AVFormatContext
2287 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2288 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2290 s->start_time_realtime -=
2291 av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
2295 if (ret == -RTCP_BYE) {
2298 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2299 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2301 if (rt->nb_byes == rt->nb_rtsp_streams)
2305 } else if (CONFIG_RTPDEC && rt->ts) {
2306 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2309 rt->recvbuf_len = len;
2310 rt->recvbuf_pos = ret;
2311 rt->cur_transport_priv = rt->ts;
2318 return AVERROR_INVALIDDATA;
2324 /* more packets may follow, so we save the RTP context */
2325 rt->cur_transport_priv = rtsp_st->transport_priv;
2329 #endif /* CONFIG_RTPDEC */
2331 #if CONFIG_SDP_DEMUXER
2332 static int sdp_probe(const AVProbeData *p1)
2334 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2336 /* we look for a line beginning "c=IN IP" */
2337 while (p < p_end && *p != '\0') {
2338 if (sizeof("c=IN IP") - 1 < p_end - p &&
2339 av_strstart(p, "c=IN IP", NULL))
2340 return AVPROBE_SCORE_EXTENSION;
2342 while (p < p_end - 1 && *p != '\n') p++;
2351 static void append_source_addrs(char *buf, int size, const char *name,
2352 int count, struct RTSPSource **addrs)
2357 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2358 for (i = 1; i < count; i++)
2359 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2362 static int sdp_read_header(AVFormatContext *s)
2364 RTSPState *rt = s->priv_data;
2365 RTSPStream *rtsp_st;
2368 char url[MAX_URL_SIZE];
2370 if (!ff_network_init())
2371 return AVERROR(EIO);
2373 if (s->max_delay < 0) /* Not set by the caller */
2374 s->max_delay = DEFAULT_REORDERING_DELAY;
2375 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2376 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2378 /* read the whole sdp file */
2379 /* XXX: better loading */
2380 content = av_malloc(SDP_MAX_SIZE);
2383 return AVERROR(ENOMEM);
2385 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2389 return AVERROR_INVALIDDATA;
2391 content[size] ='\0';
2393 err = ff_sdp_parse(s, content);
2397 /* open each RTP stream */
2398 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2400 rtsp_st = rt->rtsp_streams[i];
2402 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2403 AVDictionary *opts = map_to_opts(rt);
2405 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2406 sizeof(rtsp_st->sdp_ip),
2407 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2409 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2411 av_dict_free(&opts);
2414 ff_url_join(url, sizeof(url), "rtp", NULL,
2415 namebuf, rtsp_st->sdp_port,
2416 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2417 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2418 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2419 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2421 append_source_addrs(url, sizeof(url), "sources",
2422 rtsp_st->nb_include_source_addrs,
2423 rtsp_st->include_source_addrs);
2424 append_source_addrs(url, sizeof(url), "block",
2425 rtsp_st->nb_exclude_source_addrs,
2426 rtsp_st->exclude_source_addrs);
2427 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2428 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2430 av_dict_free(&opts);
2433 err = AVERROR_INVALIDDATA;
2437 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2442 ff_rtsp_close_streams(s);
2447 static int sdp_read_close(AVFormatContext *s)
2449 ff_rtsp_close_streams(s);
2454 static const AVClass sdp_demuxer_class = {
2455 .class_name = "SDP demuxer",
2456 .item_name = av_default_item_name,
2457 .option = sdp_options,
2458 .version = LIBAVUTIL_VERSION_INT,
2461 AVInputFormat ff_sdp_demuxer = {
2463 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2464 .priv_data_size = sizeof(RTSPState),
2465 .read_probe = sdp_probe,
2466 .read_header = sdp_read_header,
2467 .read_packet = ff_rtsp_fetch_packet,
2468 .read_close = sdp_read_close,
2469 .priv_class = &sdp_demuxer_class,
2471 #endif /* CONFIG_SDP_DEMUXER */
2473 #if CONFIG_RTP_DEMUXER
2474 static int rtp_probe(const AVProbeData *p)
2476 if (av_strstart(p->filename, "rtp:", NULL))
2477 return AVPROBE_SCORE_MAX;
2481 static int rtp_read_header(AVFormatContext *s)
2483 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2484 char host[500], filters_buf[1000];
2486 URLContext* in = NULL;
2488 AVCodecParameters *par = NULL;
2489 struct sockaddr_storage addr;
2491 socklen_t addrlen = sizeof(addr);
2492 RTSPState *rt = s->priv_data;
2496 if (!ff_network_init())
2497 return AVERROR(EIO);
2499 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2500 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2505 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2506 if (ret == AVERROR(EAGAIN))
2511 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2515 if ((recvbuf[0] & 0xc0) != 0x80) {
2516 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2521 if (RTP_PT_IS_RTCP(recvbuf[1]))
2524 payload_type = recvbuf[1] & 0x7f;
2527 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2530 par = avcodec_parameters_alloc();
2532 ret = AVERROR(ENOMEM);
2536 if (ff_rtp_get_codec_info(par, payload_type)) {
2537 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2538 "without an SDP file describing it\n",
2540 ret = AVERROR_INVALIDDATA;
2543 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2544 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2545 "properly you need an SDP file "
2549 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2552 av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
2553 av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2554 addr.ss_family == AF_INET ? 4 : 6, host);
2556 p = strchr(s->url, '?');
2558 static const char filters[][2][8] = { { "sources", "incl" },
2559 { "block", "excl" } };
2562 for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2563 if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2565 while ((q = strchr(q, ',')) != NULL)
2567 av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2569 addr.ss_family == AF_INET ? 4 : 6, host,
2575 av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2576 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2577 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2578 port, payload_type);
2579 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2580 if (!av_bprint_is_complete(&sdp))
2582 avcodec_parameters_free(&par);
2584 ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2587 /* if sdp_read_header() fails then following ff_network_close() cancels out */
2588 /* ff_network_init() at the start of this function. Otherwise it cancels out */
2589 /* ff_network_init() inside sdp_read_header() */
2592 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2594 ret = sdp_read_header(s);
2596 av_bprint_finalize(&sdp, NULL);
2600 ret = AVERROR(ENOMEM);
2601 av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2602 av_bprint_finalize(&sdp, NULL);
2604 avcodec_parameters_free(&par);
2610 static const AVClass rtp_demuxer_class = {
2611 .class_name = "RTP demuxer",
2612 .item_name = av_default_item_name,
2613 .option = rtp_options,
2614 .version = LIBAVUTIL_VERSION_INT,
2617 AVInputFormat ff_rtp_demuxer = {
2619 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2620 .priv_data_size = sizeof(RTSPState),
2621 .read_probe = rtp_probe,
2622 .read_header = rtp_read_header,
2623 .read_packet = ff_rtsp_fetch_packet,
2624 .read_close = sdp_read_close,
2625 .flags = AVFMT_NOFILE,
2626 .priv_class = &rtp_demuxer_class,
2628 #endif /* CONFIG_RTP_DEMUXER */