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 { "satip_raw", "export raw MPEG-TS stream instead of demuxing", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_SATIP_RAW}, 0, 0, DEC, "rtsp_flags" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "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 },
97 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
99 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103 static const AVOption sdp_options[] = {
104 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106 { "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" },
107 { "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 },
108 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
113 static const AVOption rtp_options[] = {
114 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
115 { "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 },
116 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
122 static AVDictionary *map_to_opts(RTSPState *rt)
124 AVDictionary *opts = NULL;
127 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
128 av_dict_set(&opts, "buffer_size", buf, 0);
129 snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
130 av_dict_set(&opts, "pkt_size", buf, 0);
135 static void get_word_until_chars(char *buf, int buf_size,
136 const char *sep, const char **pp)
142 p += strspn(p, SPACE_CHARS);
144 while (!strchr(sep, *p) && *p != '\0') {
145 if ((q - buf) < buf_size - 1)
154 static void get_word_sep(char *buf, int buf_size, const char *sep,
157 if (**pp == '/') (*pp)++;
158 get_word_until_chars(buf, buf_size, sep, pp);
161 static void get_word(char *buf, int buf_size, const char **pp)
163 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
166 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
168 * Used for seeking in the rtp stream.
170 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
174 p += strspn(p, SPACE_CHARS);
175 if (!av_stristart(p, "npt=", &p))
178 *start = AV_NOPTS_VALUE;
179 *end = AV_NOPTS_VALUE;
181 get_word_sep(buf, sizeof(buf), "-", &p);
182 if (av_parse_time(start, buf, 1) < 0)
186 get_word_sep(buf, sizeof(buf), "-", &p);
187 if (av_parse_time(end, buf, 1) < 0)
188 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
192 static int get_sockaddr(AVFormatContext *s,
193 const char *buf, struct sockaddr_storage *sock)
195 struct addrinfo hints = { 0 }, *ai = NULL;
198 hints.ai_flags = AI_NUMERICHOST;
199 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
200 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
205 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
211 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
212 RTSPStream *rtsp_st, AVStream *st)
214 AVCodecParameters *par = st ? st->codecpar : NULL;
218 par->codec_id = handler->codec_id;
219 rtsp_st->dynamic_handler = handler;
221 st->need_parsing = handler->need_parsing;
222 if (handler->priv_data_size) {
223 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
224 if (!rtsp_st->dynamic_protocol_context)
225 rtsp_st->dynamic_handler = NULL;
229 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
232 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
233 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
234 rtsp_st->dynamic_protocol_context);
236 if (rtsp_st->dynamic_protocol_context) {
237 if (rtsp_st->dynamic_handler->close)
238 rtsp_st->dynamic_handler->close(
239 rtsp_st->dynamic_protocol_context);
240 av_free(rtsp_st->dynamic_protocol_context);
242 rtsp_st->dynamic_protocol_context = NULL;
243 rtsp_st->dynamic_handler = NULL;
248 static int init_satip_stream(AVFormatContext *s)
250 RTSPState *rt = s->priv_data;
251 RTSPStream *rtsp_st = av_mallocz(sizeof(RTSPStream));
253 return AVERROR(ENOMEM);
254 dynarray_add(&rt->rtsp_streams,
255 &rt->nb_rtsp_streams, rtsp_st);
257 rtsp_st->sdp_payload_type = 33; // MP2T
258 av_strlcpy(rtsp_st->control_url,
259 rt->control_uri, sizeof(rtsp_st->control_url));
261 if (rt->rtsp_flags & RTSP_FLAG_SATIP_RAW) {
262 AVStream *st = avformat_new_stream(s, NULL);
264 return AVERROR(ENOMEM);
265 st->id = rt->nb_rtsp_streams - 1;
266 rtsp_st->stream_index = st->index;
267 st->codecpar->codec_type = AVMEDIA_TYPE_DATA;
268 st->codecpar->codec_id = AV_CODEC_ID_MPEG2TS;
270 rtsp_st->stream_index = -1;
271 init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL);
272 finalize_rtp_handler_init(s, rtsp_st, NULL);
277 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
278 static int sdp_parse_rtpmap(AVFormatContext *s,
279 AVStream *st, RTSPStream *rtsp_st,
280 int payload_type, const char *p)
282 AVCodecParameters *par = st->codecpar;
285 const AVCodecDescriptor *desc;
288 /* See if we can handle this kind of payload.
289 * The space should normally not be there but some Real streams or
290 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
291 * have a trailing space. */
292 get_word_sep(buf, sizeof(buf), "/ ", &p);
293 if (payload_type < RTP_PT_PRIVATE) {
294 /* We are in a standard case
295 * (from http://www.iana.org/assignments/rtp-parameters). */
296 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
299 if (par->codec_id == AV_CODEC_ID_NONE) {
300 const RTPDynamicProtocolHandler *handler =
301 ff_rtp_handler_find_by_name(buf, par->codec_type);
302 init_rtp_handler(handler, rtsp_st, st);
303 /* If no dynamic handler was found, check with the list of standard
304 * allocated types, if such a stream for some reason happens to
305 * use a private payload type. This isn't handled in rtpdec.c, since
306 * the format name from the rtpmap line never is passed into rtpdec. */
307 if (!rtsp_st->dynamic_handler)
308 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
311 desc = avcodec_descriptor_get(par->codec_id);
312 if (desc && desc->name)
317 get_word_sep(buf, sizeof(buf), "/", &p);
319 switch (par->codec_type) {
320 case AVMEDIA_TYPE_AUDIO:
321 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
322 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
323 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
325 par->sample_rate = i;
326 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
327 get_word_sep(buf, sizeof(buf), "/", &p);
332 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
334 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
337 case AVMEDIA_TYPE_VIDEO:
338 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
340 avpriv_set_pts_info(st, 32, 1, i);
345 finalize_rtp_handler_init(s, rtsp_st, st);
349 /* parse the attribute line from the fmtp a line of an sdp response. This
350 * is broken out as a function because it is used in rtp_h264.c, which is
352 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
353 char *value, int value_size)
355 *p += strspn(*p, SPACE_CHARS);
357 get_word_sep(attr, attr_size, "=", p);
360 get_word_sep(value, value_size, ";", p);
368 typedef struct SDPParseState {
370 struct sockaddr_storage default_ip;
372 int skip_media; ///< set if an unknown m= line occurs
373 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
374 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
375 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
376 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
379 char delayed_fmtp[2048];
382 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
383 struct RTSPSource ***dest, int *dest_count)
385 RTSPSource *rtsp_src, *rtsp_src2;
387 for (i = 0; i < count; i++) {
389 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
392 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
393 dynarray_add(dest, dest_count, rtsp_src2);
397 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
398 int payload_type, const char *line)
402 for (i = 0; i < rt->nb_rtsp_streams; i++) {
403 RTSPStream *rtsp_st = rt->rtsp_streams[i];
404 if (rtsp_st->sdp_payload_type == payload_type &&
405 rtsp_st->dynamic_handler &&
406 rtsp_st->dynamic_handler->parse_sdp_a_line) {
407 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
408 rtsp_st->dynamic_protocol_context, line);
413 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
414 int letter, const char *buf)
416 RTSPState *rt = s->priv_data;
417 char buf1[64], st_type[64];
419 enum AVMediaType codec_type;
423 RTSPSource *rtsp_src;
424 struct sockaddr_storage sdp_ip;
427 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
430 if (s1->skip_media && letter != 'm')
434 get_word(buf1, sizeof(buf1), &p);
435 if (strcmp(buf1, "IN") != 0)
437 get_word(buf1, sizeof(buf1), &p);
438 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
440 get_word_sep(buf1, sizeof(buf1), "/", &p);
441 if (get_sockaddr(s, buf1, &sdp_ip))
446 get_word_sep(buf1, sizeof(buf1), "/", &p);
449 if (s->nb_streams == 0) {
450 s1->default_ip = sdp_ip;
451 s1->default_ttl = ttl;
453 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
454 rtsp_st->sdp_ip = sdp_ip;
455 rtsp_st->sdp_ttl = ttl;
459 av_dict_set(&s->metadata, "title", p, 0);
462 if (s->nb_streams == 0) {
463 av_dict_set(&s->metadata, "comment", p, 0);
472 codec_type = AVMEDIA_TYPE_UNKNOWN;
473 get_word(st_type, sizeof(st_type), &p);
474 if (!strcmp(st_type, "audio")) {
475 codec_type = AVMEDIA_TYPE_AUDIO;
476 } else if (!strcmp(st_type, "video")) {
477 codec_type = AVMEDIA_TYPE_VIDEO;
478 } else if (!strcmp(st_type, "application")) {
479 codec_type = AVMEDIA_TYPE_DATA;
480 } else if (!strcmp(st_type, "text")) {
481 codec_type = AVMEDIA_TYPE_SUBTITLE;
483 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
484 !(rt->media_type_mask & (1 << codec_type)) ||
485 rt->nb_rtsp_streams >= s->max_streams
490 rtsp_st = av_mallocz(sizeof(RTSPStream));
493 rtsp_st->stream_index = -1;
494 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
496 rtsp_st->sdp_ip = s1->default_ip;
497 rtsp_st->sdp_ttl = s1->default_ttl;
499 copy_default_source_addrs(s1->default_include_source_addrs,
500 s1->nb_default_include_source_addrs,
501 &rtsp_st->include_source_addrs,
502 &rtsp_st->nb_include_source_addrs);
503 copy_default_source_addrs(s1->default_exclude_source_addrs,
504 s1->nb_default_exclude_source_addrs,
505 &rtsp_st->exclude_source_addrs,
506 &rtsp_st->nb_exclude_source_addrs);
508 get_word(buf1, sizeof(buf1), &p); /* port */
509 rtsp_st->sdp_port = atoi(buf1);
511 get_word(buf1, sizeof(buf1), &p); /* protocol */
512 if (!strcmp(buf1, "udp"))
513 rt->transport = RTSP_TRANSPORT_RAW;
514 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
515 rtsp_st->feedback = 1;
517 /* XXX: handle list of formats */
518 get_word(buf1, sizeof(buf1), &p); /* format list */
519 rtsp_st->sdp_payload_type = atoi(buf1);
521 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
522 /* no corresponding stream */
523 if (rt->transport == RTSP_TRANSPORT_RAW) {
524 if (CONFIG_RTPDEC && !rt->ts)
525 rt->ts = avpriv_mpegts_parse_open(s);
527 const RTPDynamicProtocolHandler *handler;
528 handler = ff_rtp_handler_find_by_id(
529 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
530 init_rtp_handler(handler, rtsp_st, NULL);
531 finalize_rtp_handler_init(s, rtsp_st, NULL);
533 } else if (rt->server_type == RTSP_SERVER_WMS &&
534 codec_type == AVMEDIA_TYPE_DATA) {
535 /* RTX stream, a stream that carries all the other actual
536 * audio/video streams. Don't expose this to the callers. */
538 st = avformat_new_stream(s, NULL);
541 st->id = rt->nb_rtsp_streams - 1;
542 rtsp_st->stream_index = st->index;
543 st->codecpar->codec_type = codec_type;
544 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
545 const RTPDynamicProtocolHandler *handler;
546 /* if standard payload type, we can find the codec right now */
547 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
548 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
549 st->codecpar->sample_rate > 0)
550 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
551 /* Even static payload types may need a custom depacketizer */
552 handler = ff_rtp_handler_find_by_id(
553 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
554 init_rtp_handler(handler, rtsp_st, st);
555 finalize_rtp_handler_init(s, rtsp_st, st);
557 if (rt->default_lang[0])
558 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
560 /* put a default control url */
561 av_strlcpy(rtsp_st->control_url, rt->control_uri,
562 sizeof(rtsp_st->control_url));
565 if (av_strstart(p, "control:", &p)) {
566 if (rt->nb_rtsp_streams == 0) {
567 if (!strncmp(p, "rtsp://", 7))
568 av_strlcpy(rt->control_uri, p,
569 sizeof(rt->control_uri));
572 /* get the control url */
573 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
575 /* XXX: may need to add full url resolution */
576 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
578 if (proto[0] == '\0') {
579 /* relative control URL */
580 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
581 av_strlcat(rtsp_st->control_url, "/",
582 sizeof(rtsp_st->control_url));
583 av_strlcat(rtsp_st->control_url, p,
584 sizeof(rtsp_st->control_url));
586 av_strlcpy(rtsp_st->control_url, p,
587 sizeof(rtsp_st->control_url));
589 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
590 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
591 get_word(buf1, sizeof(buf1), &p);
592 payload_type = atoi(buf1);
593 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
594 if (rtsp_st->stream_index >= 0) {
595 st = s->streams[rtsp_st->stream_index];
596 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
600 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
602 } else if (av_strstart(p, "fmtp:", &p) ||
603 av_strstart(p, "framesize:", &p)) {
604 // let dynamic protocol handlers have a stab at the line.
605 get_word(buf1, sizeof(buf1), &p);
606 payload_type = atoi(buf1);
607 if (s1->seen_rtpmap) {
608 parse_fmtp(s, rt, payload_type, buf);
611 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
613 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
614 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
615 get_word(buf1, sizeof(buf1), &p);
616 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
617 } else if (av_strstart(p, "range:", &p)) {
620 // this is so that seeking on a streamed file can work.
621 rtsp_parse_range_npt(p, &start, &end);
622 s->start_time = start;
623 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
624 s->duration = (end == AV_NOPTS_VALUE) ?
625 AV_NOPTS_VALUE : end - start;
626 } else if (av_strstart(p, "lang:", &p)) {
627 if (s->nb_streams > 0) {
628 get_word(buf1, sizeof(buf1), &p);
629 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
630 if (rtsp_st->stream_index >= 0) {
631 st = s->streams[rtsp_st->stream_index];
632 av_dict_set(&st->metadata, "language", buf1, 0);
635 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
636 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
638 rt->transport = RTSP_TRANSPORT_RDT;
639 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
641 st = s->streams[s->nb_streams - 1];
642 st->codecpar->sample_rate = atoi(p);
643 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
645 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
646 get_word(buf1, sizeof(buf1), &p); // ignore tag
647 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
648 p += strspn(p, SPACE_CHARS);
649 if (av_strstart(p, "inline:", &p))
650 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
651 } else if (av_strstart(p, "source-filter:", &p)) {
653 get_word(buf1, sizeof(buf1), &p);
654 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
656 exclude = !strcmp(buf1, "excl");
658 get_word(buf1, sizeof(buf1), &p);
659 if (strcmp(buf1, "IN") != 0)
661 get_word(buf1, sizeof(buf1), &p);
662 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
664 // not checking that the destination address actually matches or is wildcard
665 get_word(buf1, sizeof(buf1), &p);
668 rtsp_src = av_mallocz(sizeof(*rtsp_src));
671 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
673 if (s->nb_streams == 0) {
674 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
676 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
677 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
680 if (s->nb_streams == 0) {
681 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
683 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
684 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
689 if (rt->server_type == RTSP_SERVER_WMS)
690 ff_wms_parse_sdp_a_line(s, p);
691 if (s->nb_streams > 0) {
692 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
694 if (rt->server_type == RTSP_SERVER_REAL)
695 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
697 if (rtsp_st->dynamic_handler &&
698 rtsp_st->dynamic_handler->parse_sdp_a_line)
699 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
700 rtsp_st->stream_index,
701 rtsp_st->dynamic_protocol_context, buf);
708 int ff_sdp_parse(AVFormatContext *s, const char *content)
712 char buf[SDP_MAX_SIZE], *q;
713 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
717 p += strspn(p, SPACE_CHARS);
725 /* get the content */
727 while (*p != '\n' && *p != '\r' && *p != '\0') {
728 if ((q - buf) < sizeof(buf) - 1)
733 sdp_parse_line(s, s1, letter, buf);
735 while (*p != '\n' && *p != '\0')
741 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
742 av_freep(&s1->default_include_source_addrs[i]);
743 av_freep(&s1->default_include_source_addrs);
744 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
745 av_freep(&s1->default_exclude_source_addrs[i]);
746 av_freep(&s1->default_exclude_source_addrs);
750 #endif /* CONFIG_RTPDEC */
752 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
754 RTSPState *rt = s->priv_data;
757 for (i = 0; i < rt->nb_rtsp_streams; i++) {
758 RTSPStream *rtsp_st = rt->rtsp_streams[i];
761 if (rtsp_st->transport_priv) {
763 AVFormatContext *rtpctx = rtsp_st->transport_priv;
764 av_write_trailer(rtpctx);
765 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
766 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
767 ff_rtsp_tcp_write_packet(s, rtsp_st);
768 ffio_free_dyn_buf(&rtpctx->pb);
770 avio_closep(&rtpctx->pb);
772 avformat_free_context(rtpctx);
773 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
774 ff_rdt_parse_close(rtsp_st->transport_priv);
775 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
776 ff_rtp_parse_close(rtsp_st->transport_priv);
778 rtsp_st->transport_priv = NULL;
779 ffurl_closep(&rtsp_st->rtp_handle);
783 /* close and free RTSP streams */
784 void ff_rtsp_close_streams(AVFormatContext *s)
786 RTSPState *rt = s->priv_data;
790 ff_rtsp_undo_setup(s, 0);
791 for (i = 0; i < rt->nb_rtsp_streams; i++) {
792 rtsp_st = rt->rtsp_streams[i];
794 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
795 if (rtsp_st->dynamic_handler->close)
796 rtsp_st->dynamic_handler->close(
797 rtsp_st->dynamic_protocol_context);
798 av_free(rtsp_st->dynamic_protocol_context);
800 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
801 av_freep(&rtsp_st->include_source_addrs[j]);
802 av_freep(&rtsp_st->include_source_addrs);
803 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
804 av_freep(&rtsp_st->exclude_source_addrs[j]);
805 av_freep(&rtsp_st->exclude_source_addrs);
810 av_freep(&rt->rtsp_streams);
812 avformat_close_input(&rt->asf_ctx);
814 if (CONFIG_RTPDEC && rt->ts)
815 avpriv_mpegts_parse_close(rt->ts);
817 av_freep(&rt->recvbuf);
820 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
822 RTSPState *rt = s->priv_data;
824 int reordering_queue_size = rt->reordering_queue_size;
825 if (reordering_queue_size < 0) {
826 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
827 reordering_queue_size = 0;
829 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
832 /* open the RTP context */
833 if (rtsp_st->stream_index >= 0)
834 st = s->streams[rtsp_st->stream_index];
836 s->ctx_flags |= AVFMTCTX_NOHEADER;
838 if (CONFIG_RTSP_MUXER && s->oformat && st) {
839 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
840 s, st, rtsp_st->rtp_handle,
841 RTSP_TCP_MAX_PACKET_SIZE,
842 rtsp_st->stream_index);
843 /* Ownership of rtp_handle is passed to the rtp mux context */
844 rtsp_st->rtp_handle = NULL;
847 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
848 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
849 return 0; // Don't need to open any parser here
850 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
851 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
852 rtsp_st->dynamic_protocol_context,
853 rtsp_st->dynamic_handler);
854 else if (CONFIG_RTPDEC)
855 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
856 rtsp_st->sdp_payload_type,
857 reordering_queue_size);
859 if (!rtsp_st->transport_priv) {
860 return AVERROR(ENOMEM);
861 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
863 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
864 rtpctx->ssrc = rtsp_st->ssrc;
865 if (rtsp_st->dynamic_handler) {
866 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
867 rtsp_st->dynamic_protocol_context,
868 rtsp_st->dynamic_handler);
870 if (rtsp_st->crypto_suite[0])
871 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
872 rtsp_st->crypto_suite,
873 rtsp_st->crypto_params);
879 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
880 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
887 q += strspn(q, SPACE_CHARS);
888 v = strtol(q, &p, 10);
892 v = strtol(p, &p, 10);
901 /* XXX: only one transport specification is parsed */
902 static void rtsp_parse_transport(AVFormatContext *s,
903 RTSPMessageHeader *reply, const char *p)
905 char transport_protocol[16];
907 char lower_transport[16];
909 RTSPTransportField *th;
912 reply->nb_transports = 0;
915 p += strspn(p, SPACE_CHARS);
919 th = &reply->transports[reply->nb_transports];
921 get_word_sep(transport_protocol, sizeof(transport_protocol),
923 if (!av_strcasecmp (transport_protocol, "rtp")) {
924 get_word_sep(profile, sizeof(profile), "/;,", &p);
925 lower_transport[0] = '\0';
926 /* rtp/avp/<protocol> */
928 get_word_sep(lower_transport, sizeof(lower_transport),
931 th->transport = RTSP_TRANSPORT_RTP;
932 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
933 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
934 /* x-pn-tng/<protocol> */
935 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
937 th->transport = RTSP_TRANSPORT_RDT;
938 } else if (!av_strcasecmp(transport_protocol, "raw")) {
939 get_word_sep(profile, sizeof(profile), "/;,", &p);
940 lower_transport[0] = '\0';
941 /* raw/raw/<protocol> */
943 get_word_sep(lower_transport, sizeof(lower_transport),
946 th->transport = RTSP_TRANSPORT_RAW;
948 if (!av_strcasecmp(lower_transport, "TCP"))
949 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
951 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
955 /* get each parameter */
956 while (*p != '\0' && *p != ',') {
957 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
958 if (!strcmp(parameter, "port")) {
961 rtsp_parse_range(&th->port_min, &th->port_max, &p);
963 } else if (!strcmp(parameter, "client_port")) {
966 rtsp_parse_range(&th->client_port_min,
967 &th->client_port_max, &p);
969 } else if (!strcmp(parameter, "server_port")) {
972 rtsp_parse_range(&th->server_port_min,
973 &th->server_port_max, &p);
975 } else if (!strcmp(parameter, "interleaved")) {
978 rtsp_parse_range(&th->interleaved_min,
979 &th->interleaved_max, &p);
981 } else if (!strcmp(parameter, "multicast")) {
982 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
983 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
984 } else if (!strcmp(parameter, "ttl")) {
988 th->ttl = strtol(p, &end, 10);
991 } else if (!strcmp(parameter, "destination")) {
994 get_word_sep(buf, sizeof(buf), ";,", &p);
995 get_sockaddr(s, buf, &th->destination);
997 } else if (!strcmp(parameter, "source")) {
1000 get_word_sep(buf, sizeof(buf), ";,", &p);
1001 av_strlcpy(th->source, buf, sizeof(th->source));
1003 } else if (!strcmp(parameter, "mode")) {
1006 get_word_sep(buf, sizeof(buf), ";, ", &p);
1007 if (!strcmp(buf, "record") ||
1008 !strcmp(buf, "receive"))
1009 th->mode_record = 1;
1013 while (*p != ';' && *p != '\0' && *p != ',')
1021 reply->nb_transports++;
1022 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1027 static void handle_rtp_info(RTSPState *rt, const char *url,
1028 uint32_t seq, uint32_t rtptime)
1031 if (!rtptime || !url[0])
1033 if (rt->transport != RTSP_TRANSPORT_RTP)
1035 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1036 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1037 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1040 if (!strcmp(rtsp_st->control_url, url)) {
1041 rtpctx->base_timestamp = rtptime;
1047 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1050 char key[20], value[MAX_URL_SIZE], url[MAX_URL_SIZE] = "";
1051 uint32_t seq = 0, rtptime = 0;
1054 p += strspn(p, SPACE_CHARS);
1057 get_word_sep(key, sizeof(key), "=", &p);
1061 get_word_sep(value, sizeof(value), ";, ", &p);
1063 if (!strcmp(key, "url"))
1064 av_strlcpy(url, value, sizeof(url));
1065 else if (!strcmp(key, "seq"))
1066 seq = strtoul(value, NULL, 10);
1067 else if (!strcmp(key, "rtptime"))
1068 rtptime = strtoul(value, NULL, 10);
1070 handle_rtp_info(rt, url, seq, rtptime);
1079 handle_rtp_info(rt, url, seq, rtptime);
1082 void ff_rtsp_parse_line(AVFormatContext *s,
1083 RTSPMessageHeader *reply, const char *buf,
1084 RTSPState *rt, const char *method)
1088 /* NOTE: we do case independent match for broken servers */
1090 if (av_stristart(p, "Session:", &p)) {
1092 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1093 if (av_stristart(p, ";timeout=", &p) &&
1094 (t = strtol(p, NULL, 10)) > 0) {
1097 } else if (av_stristart(p, "Content-Length:", &p)) {
1098 reply->content_length = strtol(p, NULL, 10);
1099 } else if (av_stristart(p, "Transport:", &p)) {
1100 rtsp_parse_transport(s, reply, p);
1101 } else if (av_stristart(p, "CSeq:", &p)) {
1102 reply->seq = strtol(p, NULL, 10);
1103 } else if (av_stristart(p, "Range:", &p)) {
1104 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1105 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1106 p += strspn(p, SPACE_CHARS);
1107 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1108 } else if (av_stristart(p, "Server:", &p)) {
1109 p += strspn(p, SPACE_CHARS);
1110 av_strlcpy(reply->server, p, sizeof(reply->server));
1111 } else if (av_stristart(p, "Notice:", &p) ||
1112 av_stristart(p, "X-Notice:", &p)) {
1113 reply->notice = strtol(p, NULL, 10);
1114 } else if (av_stristart(p, "Location:", &p)) {
1115 p += strspn(p, SPACE_CHARS);
1116 av_strlcpy(reply->location, p , sizeof(reply->location));
1117 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1118 p += strspn(p, SPACE_CHARS);
1119 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1120 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1121 p += strspn(p, SPACE_CHARS);
1122 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1123 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1124 p += strspn(p, SPACE_CHARS);
1125 if (method && !strcmp(method, "DESCRIBE"))
1126 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1127 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1128 p += strspn(p, SPACE_CHARS);
1129 if (method && !strcmp(method, "PLAY"))
1130 rtsp_parse_rtp_info(rt, p);
1131 } else if (av_stristart(p, "Public:", &p) && rt) {
1132 if (strstr(p, "GET_PARAMETER") &&
1133 method && !strcmp(method, "OPTIONS"))
1134 rt->get_parameter_supported = 1;
1135 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1136 p += strspn(p, SPACE_CHARS);
1137 rt->accept_dynamic_rate = atoi(p);
1138 } else if (av_stristart(p, "Content-Type:", &p)) {
1139 p += strspn(p, SPACE_CHARS);
1140 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1141 } else if (av_stristart(p, "com.ses.streamID:", &p)) {
1142 p += strspn(p, SPACE_CHARS);
1143 av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id));
1147 /* skip a RTP/TCP interleaved packet */
1148 void ff_rtsp_skip_packet(AVFormatContext *s)
1150 RTSPState *rt = s->priv_data;
1152 uint8_t buf[MAX_URL_SIZE];
1154 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1157 len = AV_RB16(buf + 1);
1159 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1164 if (len1 > sizeof(buf))
1166 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1173 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1174 unsigned char **content_ptr,
1175 int return_on_interleaved_data, const char *method)
1177 RTSPState *rt = s->priv_data;
1178 char buf[MAX_URL_SIZE], buf1[MAX_URL_SIZE], *q;
1181 int ret, content_length, line_count = 0, request = 0;
1182 unsigned char *content = NULL;
1188 memset(reply, 0, sizeof(*reply));
1190 /* parse reply (XXX: use buffers) */
1191 rt->last_reply[0] = '\0';
1195 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1196 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1201 if (ch == '$' && q == buf) {
1202 if (return_on_interleaved_data) {
1205 ff_rtsp_skip_packet(s);
1206 } else if (ch != '\r') {
1207 if ((q - buf) < sizeof(buf) - 1)
1213 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1215 /* test if last line */
1219 if (line_count == 0) {
1220 /* get reply code */
1221 get_word(buf1, sizeof(buf1), &p);
1222 if (!strncmp(buf1, "RTSP/", 5)) {
1223 get_word(buf1, sizeof(buf1), &p);
1224 reply->status_code = atoi(buf1);
1225 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1227 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1228 get_word(buf1, sizeof(buf1), &p); // object
1232 ff_rtsp_parse_line(s, reply, p, rt, method);
1233 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1234 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1239 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1240 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1242 content_length = reply->content_length;
1243 if (content_length > 0) {
1244 /* leave some room for a trailing '\0' (useful for simple parsing) */
1245 content = av_malloc(content_length + 1);
1247 return AVERROR(ENOMEM);
1248 if (ffurl_read_complete(rt->rtsp_hd, content, content_length) != content_length)
1249 return AVERROR(EIO);
1250 content[content_length] = '\0';
1253 *content_ptr = content;
1258 char buf[MAX_URL_SIZE];
1259 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1260 const char* ptr = buf;
1262 if (!strcmp(reply->reason, "OPTIONS")) {
1263 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1265 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1266 if (reply->session_id[0])
1267 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1270 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1272 av_strlcat(buf, "\r\n", sizeof(buf));
1274 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1275 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1278 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1280 rt->last_cmd_time = av_gettime_relative();
1281 /* Even if the request from the server had data, it is not the data
1282 * that the caller wants or expects. The memory could also be leaked
1283 * if the actual following reply has content data. */
1285 av_freep(content_ptr);
1286 /* If method is set, this is called from ff_rtsp_send_cmd,
1287 * where a reply to exactly this request is awaited. For
1288 * callers from within packet receiving, we just want to
1289 * return to the caller and go back to receiving packets. */
1295 if (rt->seq != reply->seq) {
1296 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1297 rt->seq, reply->seq);
1301 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1302 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1303 reply->notice == 2306 /* Continuous Feed Terminated */) {
1304 rt->state = RTSP_STATE_IDLE;
1305 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1306 return AVERROR(EIO); /* data or server error */
1307 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1308 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1309 return AVERROR(EPERM);
1315 * Send a command to the RTSP server without waiting for the reply.
1317 * @param s RTSP (de)muxer context
1318 * @param method the method for the request
1319 * @param url the target url for the request
1320 * @param headers extra header lines to include in the request
1321 * @param send_content if non-null, the data to send as request body content
1322 * @param send_content_length the length of the send_content data, or 0 if
1323 * send_content is null
1325 * @return zero if success, nonzero otherwise
1327 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1328 const char *method, const char *url,
1329 const char *headers,
1330 const unsigned char *send_content,
1331 int send_content_length)
1333 RTSPState *rt = s->priv_data;
1334 char buf[MAX_URL_SIZE], *out_buf;
1335 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1337 if (!rt->rtsp_hd_out)
1338 return AVERROR(ENOTCONN);
1340 /* Add in RTSP headers */
1343 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1345 av_strlcat(buf, headers, sizeof(buf));
1346 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1347 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1348 if (rt->session_id[0] != '\0' && (!headers ||
1349 !strstr(headers, "\nIf-Match:"))) {
1350 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1353 char *str = ff_http_auth_create_response(&rt->auth_state,
1354 rt->auth, url, method);
1356 av_strlcat(buf, str, sizeof(buf));
1359 if (send_content_length > 0 && send_content)
1360 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1361 av_strlcat(buf, "\r\n", sizeof(buf));
1363 /* base64 encode rtsp if tunneling */
1364 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1365 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1366 out_buf = base64buf;
1369 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1371 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1372 if (send_content_length > 0 && send_content) {
1373 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1374 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1375 return AVERROR_PATCHWELCOME;
1377 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1379 rt->last_cmd_time = av_gettime_relative();
1384 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1385 const char *url, const char *headers)
1387 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1390 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1391 const char *headers, RTSPMessageHeader *reply,
1392 unsigned char **content_ptr)
1394 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1395 content_ptr, NULL, 0);
1398 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1399 const char *method, const char *url,
1401 RTSPMessageHeader *reply,
1402 unsigned char **content_ptr,
1403 const unsigned char *send_content,
1404 int send_content_length)
1406 RTSPState *rt = s->priv_data;
1407 HTTPAuthType cur_auth_type;
1408 int ret, attempts = 0;
1411 cur_auth_type = rt->auth_state.auth_type;
1412 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1414 send_content_length)))
1417 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1421 if (reply->status_code == 401 &&
1422 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1423 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1426 if (reply->status_code > 400){
1427 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1431 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1437 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1438 int lower_transport, const char *real_challenge)
1440 RTSPState *rt = s->priv_data;
1441 int rtx = 0, j, i, err, interleave = 0, port_off;
1442 RTSPStream *rtsp_st;
1443 RTSPMessageHeader reply1, *reply = &reply1;
1444 char cmd[MAX_URL_SIZE];
1445 const char *trans_pref;
1447 if (rt->transport == RTSP_TRANSPORT_RDT)
1448 trans_pref = "x-pn-tng";
1449 else if (rt->transport == RTSP_TRANSPORT_RAW)
1450 trans_pref = "RAW/RAW";
1452 trans_pref = "RTP/AVP";
1454 /* default timeout: 1 minute */
1457 /* Choose a random starting offset within the first half of the
1458 * port range, to allow for a number of ports to try even if the offset
1459 * happens to be at the end of the random range. */
1460 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1461 /* even random offset */
1462 port_off -= port_off & 0x01;
1464 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1465 char transport[MAX_URL_SIZE];
1468 * WMS serves all UDP data over a single connection, the RTX, which
1469 * isn't necessarily the first in the SDP but has to be the first
1470 * to be set up, else the second/third SETUP will fail with a 461.
1472 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1473 rt->server_type == RTSP_SERVER_WMS) {
1476 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1477 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1479 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1483 if (rtx == rt->nb_rtsp_streams)
1484 return -1; /* no RTX found */
1485 rtsp_st = rt->rtsp_streams[rtx];
1487 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1489 rtsp_st = rt->rtsp_streams[i];
1492 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1495 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1496 port = reply->transports[0].client_port_min;
1500 /* first try in specified port range */
1501 while (j <= rt->rtp_port_max) {
1502 AVDictionary *opts = map_to_opts(rt);
1504 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1505 "?localport=%d", j);
1506 /* we will use two ports per rtp stream (rtp and rtcp) */
1508 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1509 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1511 av_dict_free(&opts);
1516 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1521 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1523 av_strlcpy(transport, trans_pref, sizeof(transport));
1524 av_strlcat(transport,
1525 rt->server_type == RTSP_SERVER_SATIP ? ";" : "/UDP;",
1527 if (rt->server_type != RTSP_SERVER_REAL)
1528 av_strlcat(transport, "unicast;", sizeof(transport));
1529 av_strlcatf(transport, sizeof(transport),
1530 "client_port=%d", port);
1531 if (rt->transport == RTSP_TRANSPORT_RTP &&
1532 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1533 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1537 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1538 /* For WMS streams, the application streams are only used for
1539 * UDP. When trying to set it up for TCP streams, the server
1540 * will return an error. Therefore, we skip those streams. */
1541 if (rt->server_type == RTSP_SERVER_WMS &&
1542 (rtsp_st->stream_index < 0 ||
1543 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1546 snprintf(transport, sizeof(transport) - 1,
1547 "%s/TCP;", trans_pref);
1548 if (rt->transport != RTSP_TRANSPORT_RDT)
1549 av_strlcat(transport, "unicast;", sizeof(transport));
1550 av_strlcatf(transport, sizeof(transport),
1551 "interleaved=%d-%d",
1552 interleave, interleave + 1);
1556 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1557 snprintf(transport, sizeof(transport) - 1,
1558 "%s/UDP;multicast", trans_pref);
1561 av_strlcat(transport, ";mode=record", sizeof(transport));
1562 } else if (rt->server_type == RTSP_SERVER_REAL ||
1563 rt->server_type == RTSP_SERVER_WMS)
1564 av_strlcat(transport, ";mode=play", sizeof(transport));
1565 snprintf(cmd, sizeof(cmd),
1566 "Transport: %s\r\n",
1568 if (rt->accept_dynamic_rate)
1569 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1570 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1571 char real_res[41], real_csum[9];
1572 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1574 av_strlcatf(cmd, sizeof(cmd),
1576 "RealChallenge2: %s, sd=%s\r\n",
1577 rt->session_id, real_res, real_csum);
1579 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1580 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1583 } else if (reply->status_code != RTSP_STATUS_OK ||
1584 reply->nb_transports != 1) {
1585 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1589 if (rt->server_type == RTSP_SERVER_SATIP && reply->stream_id[0]) {
1590 char proto[128], host[128], path[512], auth[128];
1592 av_url_split(proto, sizeof(proto), auth, sizeof(auth), host, sizeof(host),
1593 &port, path, sizeof(path), rt->control_uri);
1594 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL, host,
1595 port, "/stream=%s", reply->stream_id);
1598 /* XXX: same protocol for all streams is required */
1600 if (reply->transports[0].lower_transport != rt->lower_transport ||
1601 reply->transports[0].transport != rt->transport) {
1602 err = AVERROR_INVALIDDATA;
1606 rt->lower_transport = reply->transports[0].lower_transport;
1607 rt->transport = reply->transports[0].transport;
1610 /* Fail if the server responded with another lower transport mode
1611 * than what we requested. */
1612 if (reply->transports[0].lower_transport != lower_transport) {
1613 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1614 err = AVERROR_INVALIDDATA;
1618 switch(reply->transports[0].lower_transport) {
1619 case RTSP_LOWER_TRANSPORT_TCP:
1620 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1621 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1624 case RTSP_LOWER_TRANSPORT_UDP: {
1625 char url[MAX_URL_SIZE], options[30] = "";
1626 const char *peer = host;
1628 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1629 av_strlcpy(options, "?connect=1", sizeof(options));
1630 /* Use source address if specified */
1631 if (reply->transports[0].source[0])
1632 peer = reply->transports[0].source;
1633 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1634 reply->transports[0].server_port_min, "%s", options);
1635 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1636 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1637 err = AVERROR_INVALIDDATA;
1642 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1643 char url[MAX_URL_SIZE], namebuf[50], optbuf[20] = "";
1644 struct sockaddr_storage addr;
1646 AVDictionary *opts = map_to_opts(rt);
1648 if (reply->transports[0].destination.ss_family) {
1649 addr = reply->transports[0].destination;
1650 port = reply->transports[0].port_min;
1651 ttl = reply->transports[0].ttl;
1653 addr = rtsp_st->sdp_ip;
1654 port = rtsp_st->sdp_port;
1655 ttl = rtsp_st->sdp_ttl;
1658 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1659 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1660 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1661 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1662 port, "%s", optbuf);
1663 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1664 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1665 av_dict_free(&opts);
1668 err = AVERROR_INVALIDDATA;
1675 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1679 if (rt->nb_rtsp_streams && reply->timeout > 0)
1680 rt->timeout = reply->timeout;
1682 if (rt->server_type == RTSP_SERVER_REAL)
1683 rt->need_subscription = 1;
1688 ff_rtsp_undo_setup(s, 0);
1692 void ff_rtsp_close_connections(AVFormatContext *s)
1694 RTSPState *rt = s->priv_data;
1695 if (rt->rtsp_hd_out != rt->rtsp_hd)
1696 ffurl_closep(&rt->rtsp_hd_out);
1697 rt->rtsp_hd_out = NULL;
1698 ffurl_closep(&rt->rtsp_hd);
1701 int ff_rtsp_connect(AVFormatContext *s)
1703 RTSPState *rt = s->priv_data;
1704 char proto[128], host[1024], path[1024];
1705 char tcpname[1024], cmd[MAX_URL_SIZE], auth[128];
1706 const char *lower_rtsp_proto = "tcp";
1707 int port, err, tcp_fd;
1708 RTSPMessageHeader reply1, *reply = &reply1;
1709 int lower_transport_mask = 0;
1710 int default_port = RTSP_DEFAULT_PORT;
1711 int https_tunnel = 0;
1712 char real_challenge[64] = "";
1713 struct sockaddr_storage peer;
1714 socklen_t peer_len = sizeof(peer);
1716 if (rt->rtp_port_max < rt->rtp_port_min) {
1717 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1718 "than min port %d\n", rt->rtp_port_max,
1720 return AVERROR(EINVAL);
1723 if (!ff_network_init())
1724 return AVERROR(EIO);
1726 if (s->max_delay < 0) /* Not set by the caller */
1727 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1729 rt->control_transport = RTSP_MODE_PLAIN;
1730 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1731 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1732 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1733 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1734 rt->control_transport = RTSP_MODE_TUNNEL;
1736 /* Only pass through valid flags from here */
1737 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1740 memset(&reply1, 0, sizeof(reply1));
1741 /* extract hostname and port */
1742 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1743 host, sizeof(host), &port, path, sizeof(path), s->url);
1745 if (!strcmp(proto, "rtsps")) {
1746 lower_rtsp_proto = "tls";
1747 default_port = RTSPS_DEFAULT_PORT;
1748 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1749 } else if (!strcmp(proto, "satip")) {
1750 av_strlcpy(proto, "rtsp", sizeof(proto));
1751 rt->server_type = RTSP_SERVER_SATIP;
1755 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1758 port = default_port;
1760 lower_transport_mask = rt->lower_transport_mask;
1762 if (!lower_transport_mask)
1763 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1766 /* Only UDP or TCP - UDP multicast isn't supported. */
1767 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1768 (1 << RTSP_LOWER_TRANSPORT_TCP);
1769 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1770 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1771 "only UDP and TCP are supported for output.\n");
1772 err = AVERROR(EINVAL);
1777 /* Construct the URI used in request; this is similar to s->url,
1778 * but with authentication credentials removed and RTSP specific options
1780 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1781 host, port, "%s", path);
1783 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1784 /* set up initial handshake for tunneling */
1785 char httpname[1024];
1786 char sessioncookie[17];
1788 AVDictionary *options = NULL;
1790 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1792 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1793 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1794 av_get_random_seed(), av_get_random_seed());
1797 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1798 &s->interrupt_callback) < 0) {
1803 /* generate GET headers */
1804 snprintf(headers, sizeof(headers),
1805 "x-sessioncookie: %s\r\n"
1806 "Accept: application/x-rtsp-tunnelled\r\n"
1807 "Pragma: no-cache\r\n"
1808 "Cache-Control: no-cache\r\n",
1810 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1812 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1813 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1814 if (!rt->rtsp_hd->protocol_whitelist) {
1815 err = AVERROR(ENOMEM);
1820 /* complete the connection */
1821 if (ffurl_connect(rt->rtsp_hd, &options)) {
1822 av_dict_free(&options);
1828 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1829 &s->interrupt_callback) < 0 ) {
1834 /* generate POST headers */
1835 snprintf(headers, sizeof(headers),
1836 "x-sessioncookie: %s\r\n"
1837 "Content-Type: application/x-rtsp-tunnelled\r\n"
1838 "Pragma: no-cache\r\n"
1839 "Cache-Control: no-cache\r\n"
1840 "Content-Length: 32767\r\n"
1841 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1843 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1844 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1845 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1847 /* Initialize the authentication state for the POST session. The HTTP
1848 * protocol implementation doesn't properly handle multi-pass
1849 * authentication for POST requests, since it would require one of
1851 * - implementing Expect: 100-continue, which many HTTP servers
1852 * don't support anyway, even less the RTSP servers that do HTTP
1854 * - sending the whole POST data until getting a 401 reply specifying
1855 * what authentication method to use, then resending all that data
1856 * - waiting for potential 401 replies directly after sending the
1857 * POST header (waiting for some unspecified time)
1858 * Therefore, we copy the full auth state, which works for both basic
1859 * and digest. (For digest, we would have to synchronize the nonce
1860 * count variable between the two sessions, if we'd do more requests
1861 * with the original session, though.)
1863 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1865 /* complete the connection */
1866 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1867 av_dict_free(&options);
1871 av_dict_free(&options);
1874 /* open the tcp connection */
1875 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1877 "?timeout=%d", rt->stimeout);
1878 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1879 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1883 rt->rtsp_hd_out = rt->rtsp_hd;
1887 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1892 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1893 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1894 NULL, 0, NI_NUMERICHOST);
1897 /* request options supported by the server; this also detects server
1899 if (rt->server_type != RTSP_SERVER_SATIP)
1900 rt->server_type = RTSP_SERVER_RTP;
1903 if (rt->server_type == RTSP_SERVER_REAL)
1906 * The following entries are required for proper
1907 * streaming from a Realmedia server. They are
1908 * interdependent in some way although we currently
1909 * don't quite understand how. Values were copied
1910 * from mplayer SVN r23589.
1911 * ClientChallenge is a 16-byte ID in hex
1912 * CompanyID is a 16-byte ID in base64
1914 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1915 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1916 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1917 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1919 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1920 if (reply->status_code != RTSP_STATUS_OK) {
1921 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1925 /* detect server type if not standard-compliant RTP */
1926 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1927 rt->server_type = RTSP_SERVER_REAL;
1929 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1930 rt->server_type = RTSP_SERVER_WMS;
1931 } else if (rt->server_type == RTSP_SERVER_REAL)
1932 strcpy(real_challenge, reply->real_challenge);
1936 #if CONFIG_RTSP_DEMUXER
1938 if (rt->server_type == RTSP_SERVER_SATIP)
1939 err = init_satip_stream(s);
1941 err = ff_rtsp_setup_input_streams(s, reply);
1944 if (CONFIG_RTSP_MUXER)
1945 err = ff_rtsp_setup_output_streams(s, host);
1952 int lower_transport = ff_log2_tab[lower_transport_mask &
1953 ~(lower_transport_mask - 1)];
1955 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1956 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1957 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1959 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1960 rt->server_type == RTSP_SERVER_REAL ?
1961 real_challenge : NULL);
1964 lower_transport_mask &= ~(1 << lower_transport);
1965 if (lower_transport_mask == 0 && err == 1) {
1966 err = AVERROR(EPROTONOSUPPORT);
1971 rt->lower_transport_mask = lower_transport_mask;
1972 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1973 rt->state = RTSP_STATE_IDLE;
1974 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1977 ff_rtsp_close_streams(s);
1978 ff_rtsp_close_connections(s);
1979 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1980 char *new_url = av_strdup(reply->location);
1982 err = AVERROR(ENOMEM);
1985 ff_format_set_url(s, new_url);
1986 rt->session_id[0] = '\0';
1987 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1996 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1999 static int parse_rtsp_message(AVFormatContext *s)
2001 RTSPState *rt = s->priv_data;
2004 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
2005 if (rt->state == RTSP_STATE_STREAMING) {
2006 return ff_rtsp_parse_streaming_commands(s);
2010 RTSPMessageHeader reply;
2011 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
2014 /* XXX: parse message */
2015 if (rt->state != RTSP_STATE_STREAMING)
2022 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
2023 uint8_t *buf, int buf_size, int64_t wait_end)
2025 RTSPState *rt = s->priv_data;
2026 RTSPStream *rtsp_st;
2028 struct pollfd *p = rt->p;
2029 int *fds = NULL, fdsnum, fdsidx;
2030 int runs = rt->initial_timeout * 1000LL / POLLING_TIME;
2033 p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(*p));
2035 return AVERROR(ENOMEM);
2038 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2039 p[rt->max_p++].events = POLLIN;
2041 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2042 rtsp_st = rt->rtsp_streams[i];
2043 if (rtsp_st->rtp_handle) {
2044 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2046 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2050 av_log(s, AV_LOG_ERROR,
2051 "Number of fds %d not supported\n", fdsnum);
2052 return AVERROR_INVALIDDATA;
2054 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2055 p[rt->max_p].fd = fds[fdsidx];
2056 p[rt->max_p++].events = POLLIN;
2064 if (ff_check_interrupt(&s->interrupt_callback))
2065 return AVERROR_EXIT;
2066 if (wait_end && wait_end - av_gettime_relative() < 0)
2067 return AVERROR(EAGAIN);
2068 n = poll(p, rt->max_p, POLLING_TIME);
2070 int j = rt->rtsp_hd ? 1 : 0;
2071 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2072 rtsp_st = rt->rtsp_streams[i];
2073 if (rtsp_st->rtp_handle) {
2074 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2075 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2077 *prtsp_st = rtsp_st;
2084 #if CONFIG_RTSP_DEMUXER
2085 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2086 if ((ret = parse_rtsp_message(s)) < 0) {
2091 } else if (n == 0 && rt->initial_timeout > 0 && --runs <= 0) {
2092 return AVERROR(ETIMEDOUT);
2093 } else if (n < 0 && errno != EINTR)
2094 return AVERROR(errno);
2098 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2099 const uint8_t *buf, int len)
2101 RTSPState *rt = s->priv_data;
2105 if (rt->nb_rtsp_streams == 1) {
2106 *rtsp_st = rt->rtsp_streams[0];
2109 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2110 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2112 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2113 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2116 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2117 *rtsp_st = rt->rtsp_streams[i];
2124 av_log(s, AV_LOG_WARNING,
2125 "Unable to pick stream for packet - SSRC not known for "
2127 return AVERROR(EAGAIN);
2130 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2131 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2132 *rtsp_st = rt->rtsp_streams[i];
2138 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2139 return AVERROR(EAGAIN);
2142 static int read_packet(AVFormatContext *s,
2143 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2146 RTSPState *rt = s->priv_data;
2149 switch(rt->lower_transport) {
2151 #if CONFIG_RTSP_DEMUXER
2152 case RTSP_LOWER_TRANSPORT_TCP:
2153 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2156 case RTSP_LOWER_TRANSPORT_UDP:
2157 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2158 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2159 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2160 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2162 case RTSP_LOWER_TRANSPORT_CUSTOM:
2163 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2164 wait_end && wait_end < av_gettime_relative())
2165 len = AVERROR(EAGAIN);
2167 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2168 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2169 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2170 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2180 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2182 RTSPState *rt = s->priv_data;
2184 RTSPStream *rtsp_st, *first_queue_st = NULL;
2185 int64_t wait_end = 0;
2187 if (rt->nb_byes == rt->nb_rtsp_streams)
2190 /* get next frames from the same RTP packet */
2191 if (rt->cur_transport_priv) {
2192 if (rt->transport == RTSP_TRANSPORT_RDT) {
2193 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2194 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2195 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2196 } else if (CONFIG_RTPDEC && rt->ts) {
2197 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2199 rt->recvbuf_pos += ret;
2200 ret = rt->recvbuf_pos < rt->recvbuf_len;
2205 rt->cur_transport_priv = NULL;
2207 } else if (ret == 1) {
2210 rt->cur_transport_priv = NULL;
2214 if (rt->transport == RTSP_TRANSPORT_RTP) {
2216 int64_t first_queue_time = 0;
2217 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2218 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2222 queue_time = ff_rtp_queued_packet_time(rtpctx);
2223 if (queue_time && (queue_time - first_queue_time < 0 ||
2224 !first_queue_time)) {
2225 first_queue_time = queue_time;
2226 first_queue_st = rt->rtsp_streams[i];
2229 if (first_queue_time) {
2230 wait_end = first_queue_time + s->max_delay;
2233 first_queue_st = NULL;
2237 /* read next RTP packet */
2239 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2241 return AVERROR(ENOMEM);
2244 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2245 if (len == AVERROR(EAGAIN) && first_queue_st &&
2246 rt->transport == RTSP_TRANSPORT_RTP) {
2247 av_log(s, AV_LOG_WARNING,
2248 "max delay reached. need to consume packet\n");
2249 rtsp_st = first_queue_st;
2250 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2256 if (rt->transport == RTSP_TRANSPORT_RDT) {
2257 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2258 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2259 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2260 if (rtsp_st->feedback) {
2261 AVIOContext *pb = NULL;
2262 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2264 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2267 /* Either bad packet, or a RTCP packet. Check if the
2268 * first_rtcp_ntp_time field was initialized. */
2269 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2270 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2271 /* first_rtcp_ntp_time has been initialized for this stream,
2272 * copy the same value to all other uninitialized streams,
2273 * in order to map their timestamp origin to the same ntp time
2276 AVStream *st = NULL;
2277 if (rtsp_st->stream_index >= 0)
2278 st = s->streams[rtsp_st->stream_index];
2279 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2280 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2281 AVStream *st2 = NULL;
2282 if (rt->rtsp_streams[i]->stream_index >= 0)
2283 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2284 if (rtpctx2 && st && st2 &&
2285 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2286 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2287 rtpctx2->rtcp_ts_offset = av_rescale_q(
2288 rtpctx->rtcp_ts_offset, st->time_base,
2292 // Make real NTP start time available in AVFormatContext
2293 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2294 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2296 s->start_time_realtime -=
2297 av_rescale_q (rtpctx->rtcp_ts_offset, rtpctx->st->time_base, AV_TIME_BASE_Q);
2301 if (ret == -RTCP_BYE) {
2304 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2305 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2307 if (rt->nb_byes == rt->nb_rtsp_streams)
2311 } else if (CONFIG_RTPDEC && rt->ts) {
2312 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2315 rt->recvbuf_len = len;
2316 rt->recvbuf_pos = ret;
2317 rt->cur_transport_priv = rt->ts;
2324 return AVERROR_INVALIDDATA;
2330 /* more packets may follow, so we save the RTP context */
2331 rt->cur_transport_priv = rtsp_st->transport_priv;
2335 #endif /* CONFIG_RTPDEC */
2337 #if CONFIG_SDP_DEMUXER
2338 static int sdp_probe(const AVProbeData *p1)
2340 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2342 /* we look for a line beginning "c=IN IP" */
2343 while (p < p_end && *p != '\0') {
2344 if (sizeof("c=IN IP") - 1 < p_end - p &&
2345 av_strstart(p, "c=IN IP", NULL))
2346 return AVPROBE_SCORE_EXTENSION;
2348 while (p < p_end - 1 && *p != '\n') p++;
2357 static void append_source_addrs(char *buf, int size, const char *name,
2358 int count, struct RTSPSource **addrs)
2363 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2364 for (i = 1; i < count; i++)
2365 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2368 static int sdp_read_header(AVFormatContext *s)
2370 RTSPState *rt = s->priv_data;
2371 RTSPStream *rtsp_st;
2374 char url[MAX_URL_SIZE];
2376 if (!ff_network_init())
2377 return AVERROR(EIO);
2379 if (s->max_delay < 0) /* Not set by the caller */
2380 s->max_delay = DEFAULT_REORDERING_DELAY;
2381 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2382 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2384 /* read the whole sdp file */
2385 /* XXX: better loading */
2386 content = av_malloc(SDP_MAX_SIZE);
2389 return AVERROR(ENOMEM);
2391 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2395 return AVERROR_INVALIDDATA;
2397 content[size] ='\0';
2399 err = ff_sdp_parse(s, content);
2403 /* open each RTP stream */
2404 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2406 rtsp_st = rt->rtsp_streams[i];
2408 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2409 AVDictionary *opts = map_to_opts(rt);
2411 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2412 sizeof(rtsp_st->sdp_ip),
2413 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2415 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2417 av_dict_free(&opts);
2420 ff_url_join(url, sizeof(url), "rtp", NULL,
2421 namebuf, rtsp_st->sdp_port,
2422 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2423 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2424 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2425 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2427 append_source_addrs(url, sizeof(url), "sources",
2428 rtsp_st->nb_include_source_addrs,
2429 rtsp_st->include_source_addrs);
2430 append_source_addrs(url, sizeof(url), "block",
2431 rtsp_st->nb_exclude_source_addrs,
2432 rtsp_st->exclude_source_addrs);
2433 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2434 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2436 av_dict_free(&opts);
2439 err = AVERROR_INVALIDDATA;
2443 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2448 ff_rtsp_close_streams(s);
2453 static int sdp_read_close(AVFormatContext *s)
2455 ff_rtsp_close_streams(s);
2460 static const AVClass sdp_demuxer_class = {
2461 .class_name = "SDP demuxer",
2462 .item_name = av_default_item_name,
2463 .option = sdp_options,
2464 .version = LIBAVUTIL_VERSION_INT,
2467 AVInputFormat ff_sdp_demuxer = {
2469 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2470 .priv_data_size = sizeof(RTSPState),
2471 .read_probe = sdp_probe,
2472 .read_header = sdp_read_header,
2473 .read_packet = ff_rtsp_fetch_packet,
2474 .read_close = sdp_read_close,
2475 .priv_class = &sdp_demuxer_class,
2477 #endif /* CONFIG_SDP_DEMUXER */
2479 #if CONFIG_RTP_DEMUXER
2480 static int rtp_probe(const AVProbeData *p)
2482 if (av_strstart(p->filename, "rtp:", NULL))
2483 return AVPROBE_SCORE_MAX;
2487 static int rtp_read_header(AVFormatContext *s)
2489 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2490 char host[500], filters_buf[1000];
2492 URLContext* in = NULL;
2494 AVCodecParameters *par = NULL;
2495 struct sockaddr_storage addr;
2497 socklen_t addrlen = sizeof(addr);
2498 RTSPState *rt = s->priv_data;
2501 AVDictionary *opts = NULL;
2503 if (!ff_network_init())
2504 return AVERROR(EIO);
2506 opts = map_to_opts(rt);
2507 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2508 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2509 av_dict_free(&opts);
2514 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2515 if (ret == AVERROR(EAGAIN))
2520 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2524 if ((recvbuf[0] & 0xc0) != 0x80) {
2525 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2530 if (RTP_PT_IS_RTCP(recvbuf[1]))
2533 payload_type = recvbuf[1] & 0x7f;
2536 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2539 par = avcodec_parameters_alloc();
2541 ret = AVERROR(ENOMEM);
2545 if (ff_rtp_get_codec_info(par, payload_type)) {
2546 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2547 "without an SDP file describing it\n",
2549 ret = AVERROR_INVALIDDATA;
2552 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2553 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2554 "properly you need an SDP file "
2558 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2561 av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
2562 av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2563 addr.ss_family == AF_INET ? 4 : 6, host);
2565 p = strchr(s->url, '?');
2567 static const char filters[][2][8] = { { "sources", "incl" },
2568 { "block", "excl" } };
2571 for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2572 if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2574 while ((q = strchr(q, ',')) != NULL)
2576 av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2578 addr.ss_family == AF_INET ? 4 : 6, host,
2584 av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2585 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2586 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2587 port, payload_type);
2588 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2589 if (!av_bprint_is_complete(&sdp))
2591 avcodec_parameters_free(&par);
2593 ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2596 /* if sdp_read_header() fails then following ff_network_close() cancels out */
2597 /* ff_network_init() at the start of this function. Otherwise it cancels out */
2598 /* ff_network_init() inside sdp_read_header() */
2601 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2603 ret = sdp_read_header(s);
2605 av_bprint_finalize(&sdp, NULL);
2609 ret = AVERROR(ENOMEM);
2610 av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2611 av_bprint_finalize(&sdp, NULL);
2613 avcodec_parameters_free(&par);
2619 static const AVClass rtp_demuxer_class = {
2620 .class_name = "RTP demuxer",
2621 .item_name = av_default_item_name,
2622 .option = rtp_options,
2623 .version = LIBAVUTIL_VERSION_INT,
2626 AVInputFormat ff_rtp_demuxer = {
2628 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2629 .priv_data_size = sizeof(RTSPState),
2630 .read_probe = rtp_probe,
2631 .read_header = rtp_read_header,
2632 .read_packet = ff_rtsp_fetch_packet,
2633 .read_close = sdp_read_close,
2634 .flags = AVFMT_NOFILE,
2635 .priv_class = &rtp_demuxer_class,
2637 #endif /* CONFIG_RTP_DEMUXER */