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/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "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" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { 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" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
77 #define COMMON_OPTS() \
78 { "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 }, \
79 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
82 const AVOption ff_rtsp_options[] = {
83 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
84 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
91 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
92 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
93 { "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" },
94 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
95 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
96 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
97 { "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 },
98 #if FF_API_OLD_RTSP_OPTIONS
99 { "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 },
100 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
102 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
105 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
106 #if FF_API_OLD_RTSP_OPTIONS
107 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
112 static const AVOption sdp_options[] = {
113 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
114 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
115 { "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" },
116 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
121 static const AVOption rtp_options[] = {
122 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
128 static AVDictionary *map_to_opts(RTSPState *rt)
130 AVDictionary *opts = NULL;
133 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
134 av_dict_set(&opts, "buffer_size", buf, 0);
139 static void get_word_until_chars(char *buf, int buf_size,
140 const char *sep, const char **pp)
146 p += strspn(p, SPACE_CHARS);
148 while (!strchr(sep, *p) && *p != '\0') {
149 if ((q - buf) < buf_size - 1)
158 static void get_word_sep(char *buf, int buf_size, const char *sep,
161 if (**pp == '/') (*pp)++;
162 get_word_until_chars(buf, buf_size, sep, pp);
165 static void get_word(char *buf, int buf_size, const char **pp)
167 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
170 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
172 * Used for seeking in the rtp stream.
174 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
178 p += strspn(p, SPACE_CHARS);
179 if (!av_stristart(p, "npt=", &p))
182 *start = AV_NOPTS_VALUE;
183 *end = AV_NOPTS_VALUE;
185 get_word_sep(buf, sizeof(buf), "-", &p);
186 if (av_parse_time(start, buf, 1) < 0)
190 get_word_sep(buf, sizeof(buf), "-", &p);
191 if (av_parse_time(end, buf, 1) < 0)
192 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
196 static int get_sockaddr(AVFormatContext *s,
197 const char *buf, struct sockaddr_storage *sock)
199 struct addrinfo hints = { 0 }, *ai = NULL;
202 hints.ai_flags = AI_NUMERICHOST;
203 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
204 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
209 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
215 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
216 RTSPStream *rtsp_st, AVStream *st)
218 AVCodecParameters *par = st ? st->codecpar : NULL;
222 par->codec_id = handler->codec_id;
223 rtsp_st->dynamic_handler = handler;
225 st->need_parsing = handler->need_parsing;
226 if (handler->priv_data_size) {
227 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
228 if (!rtsp_st->dynamic_protocol_context)
229 rtsp_st->dynamic_handler = NULL;
233 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
236 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
237 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
238 rtsp_st->dynamic_protocol_context);
240 if (rtsp_st->dynamic_protocol_context) {
241 if (rtsp_st->dynamic_handler->close)
242 rtsp_st->dynamic_handler->close(
243 rtsp_st->dynamic_protocol_context);
244 av_free(rtsp_st->dynamic_protocol_context);
246 rtsp_st->dynamic_protocol_context = NULL;
247 rtsp_st->dynamic_handler = NULL;
252 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
253 static int sdp_parse_rtpmap(AVFormatContext *s,
254 AVStream *st, RTSPStream *rtsp_st,
255 int payload_type, const char *p)
257 AVCodecParameters *par = st->codecpar;
260 const AVCodecDescriptor *desc;
263 /* See if we can handle this kind of payload.
264 * The space should normally not be there but some Real streams or
265 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
266 * have a trailing space. */
267 get_word_sep(buf, sizeof(buf), "/ ", &p);
268 if (payload_type < RTP_PT_PRIVATE) {
269 /* We are in a standard case
270 * (from http://www.iana.org/assignments/rtp-parameters). */
271 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
274 if (par->codec_id == AV_CODEC_ID_NONE) {
275 const RTPDynamicProtocolHandler *handler =
276 ff_rtp_handler_find_by_name(buf, par->codec_type);
277 init_rtp_handler(handler, rtsp_st, st);
278 /* If no dynamic handler was found, check with the list of standard
279 * allocated types, if such a stream for some reason happens to
280 * use a private payload type. This isn't handled in rtpdec.c, since
281 * the format name from the rtpmap line never is passed into rtpdec. */
282 if (!rtsp_st->dynamic_handler)
283 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
286 desc = avcodec_descriptor_get(par->codec_id);
287 if (desc && desc->name)
292 get_word_sep(buf, sizeof(buf), "/", &p);
294 switch (par->codec_type) {
295 case AVMEDIA_TYPE_AUDIO:
296 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
297 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
298 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
300 par->sample_rate = i;
301 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
302 get_word_sep(buf, sizeof(buf), "/", &p);
307 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
309 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
312 case AVMEDIA_TYPE_VIDEO:
313 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
315 avpriv_set_pts_info(st, 32, 1, i);
320 finalize_rtp_handler_init(s, rtsp_st, st);
324 /* parse the attribute line from the fmtp a line of an sdp response. This
325 * is broken out as a function because it is used in rtp_h264.c, which is
327 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
328 char *value, int value_size)
330 *p += strspn(*p, SPACE_CHARS);
332 get_word_sep(attr, attr_size, "=", p);
335 get_word_sep(value, value_size, ";", p);
343 typedef struct SDPParseState {
345 struct sockaddr_storage default_ip;
347 int skip_media; ///< set if an unknown m= line occurs
348 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
349 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
350 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
351 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
354 char delayed_fmtp[2048];
357 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
358 struct RTSPSource ***dest, int *dest_count)
360 RTSPSource *rtsp_src, *rtsp_src2;
362 for (i = 0; i < count; i++) {
364 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
367 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
368 dynarray_add(dest, dest_count, rtsp_src2);
372 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
373 int payload_type, const char *line)
377 for (i = 0; i < rt->nb_rtsp_streams; i++) {
378 RTSPStream *rtsp_st = rt->rtsp_streams[i];
379 if (rtsp_st->sdp_payload_type == payload_type &&
380 rtsp_st->dynamic_handler &&
381 rtsp_st->dynamic_handler->parse_sdp_a_line) {
382 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
383 rtsp_st->dynamic_protocol_context, line);
388 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
389 int letter, const char *buf)
391 RTSPState *rt = s->priv_data;
392 char buf1[64], st_type[64];
394 enum AVMediaType codec_type;
398 RTSPSource *rtsp_src;
399 struct sockaddr_storage sdp_ip;
402 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
405 if (s1->skip_media && letter != 'm')
409 get_word(buf1, sizeof(buf1), &p);
410 if (strcmp(buf1, "IN") != 0)
412 get_word(buf1, sizeof(buf1), &p);
413 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
415 get_word_sep(buf1, sizeof(buf1), "/", &p);
416 if (get_sockaddr(s, buf1, &sdp_ip))
421 get_word_sep(buf1, sizeof(buf1), "/", &p);
424 if (s->nb_streams == 0) {
425 s1->default_ip = sdp_ip;
426 s1->default_ttl = ttl;
428 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
429 rtsp_st->sdp_ip = sdp_ip;
430 rtsp_st->sdp_ttl = ttl;
434 av_dict_set(&s->metadata, "title", p, 0);
437 if (s->nb_streams == 0) {
438 av_dict_set(&s->metadata, "comment", p, 0);
447 codec_type = AVMEDIA_TYPE_UNKNOWN;
448 get_word(st_type, sizeof(st_type), &p);
449 if (!strcmp(st_type, "audio")) {
450 codec_type = AVMEDIA_TYPE_AUDIO;
451 } else if (!strcmp(st_type, "video")) {
452 codec_type = AVMEDIA_TYPE_VIDEO;
453 } else if (!strcmp(st_type, "application")) {
454 codec_type = AVMEDIA_TYPE_DATA;
455 } else if (!strcmp(st_type, "text")) {
456 codec_type = AVMEDIA_TYPE_SUBTITLE;
458 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
459 !(rt->media_type_mask & (1 << codec_type)) ||
460 rt->nb_rtsp_streams >= s->max_streams
465 rtsp_st = av_mallocz(sizeof(RTSPStream));
468 rtsp_st->stream_index = -1;
469 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
471 rtsp_st->sdp_ip = s1->default_ip;
472 rtsp_st->sdp_ttl = s1->default_ttl;
474 copy_default_source_addrs(s1->default_include_source_addrs,
475 s1->nb_default_include_source_addrs,
476 &rtsp_st->include_source_addrs,
477 &rtsp_st->nb_include_source_addrs);
478 copy_default_source_addrs(s1->default_exclude_source_addrs,
479 s1->nb_default_exclude_source_addrs,
480 &rtsp_st->exclude_source_addrs,
481 &rtsp_st->nb_exclude_source_addrs);
483 get_word(buf1, sizeof(buf1), &p); /* port */
484 rtsp_st->sdp_port = atoi(buf1);
486 get_word(buf1, sizeof(buf1), &p); /* protocol */
487 if (!strcmp(buf1, "udp"))
488 rt->transport = RTSP_TRANSPORT_RAW;
489 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
490 rtsp_st->feedback = 1;
492 /* XXX: handle list of formats */
493 get_word(buf1, sizeof(buf1), &p); /* format list */
494 rtsp_st->sdp_payload_type = atoi(buf1);
496 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
497 /* no corresponding stream */
498 if (rt->transport == RTSP_TRANSPORT_RAW) {
499 if (CONFIG_RTPDEC && !rt->ts)
500 rt->ts = avpriv_mpegts_parse_open(s);
502 const RTPDynamicProtocolHandler *handler;
503 handler = ff_rtp_handler_find_by_id(
504 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
505 init_rtp_handler(handler, rtsp_st, NULL);
506 finalize_rtp_handler_init(s, rtsp_st, NULL);
508 } else if (rt->server_type == RTSP_SERVER_WMS &&
509 codec_type == AVMEDIA_TYPE_DATA) {
510 /* RTX stream, a stream that carries all the other actual
511 * audio/video streams. Don't expose this to the callers. */
513 st = avformat_new_stream(s, NULL);
516 st->id = rt->nb_rtsp_streams - 1;
517 rtsp_st->stream_index = st->index;
518 st->codecpar->codec_type = codec_type;
519 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
520 const RTPDynamicProtocolHandler *handler;
521 /* if standard payload type, we can find the codec right now */
522 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
523 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
524 st->codecpar->sample_rate > 0)
525 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
526 /* Even static payload types may need a custom depacketizer */
527 handler = ff_rtp_handler_find_by_id(
528 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
529 init_rtp_handler(handler, rtsp_st, st);
530 finalize_rtp_handler_init(s, rtsp_st, st);
532 if (rt->default_lang[0])
533 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
535 /* put a default control url */
536 av_strlcpy(rtsp_st->control_url, rt->control_uri,
537 sizeof(rtsp_st->control_url));
540 if (av_strstart(p, "control:", &p)) {
541 if (s->nb_streams == 0) {
542 if (!strncmp(p, "rtsp://", 7))
543 av_strlcpy(rt->control_uri, p,
544 sizeof(rt->control_uri));
547 /* get the control url */
548 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
550 /* XXX: may need to add full url resolution */
551 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
553 if (proto[0] == '\0') {
554 /* relative control URL */
555 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
556 av_strlcat(rtsp_st->control_url, "/",
557 sizeof(rtsp_st->control_url));
558 av_strlcat(rtsp_st->control_url, p,
559 sizeof(rtsp_st->control_url));
561 av_strlcpy(rtsp_st->control_url, p,
562 sizeof(rtsp_st->control_url));
564 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
565 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
566 get_word(buf1, sizeof(buf1), &p);
567 payload_type = atoi(buf1);
568 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
569 if (rtsp_st->stream_index >= 0) {
570 st = s->streams[rtsp_st->stream_index];
571 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
575 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
577 } else if (av_strstart(p, "fmtp:", &p) ||
578 av_strstart(p, "framesize:", &p)) {
579 // let dynamic protocol handlers have a stab at the line.
580 get_word(buf1, sizeof(buf1), &p);
581 payload_type = atoi(buf1);
582 if (s1->seen_rtpmap) {
583 parse_fmtp(s, rt, payload_type, buf);
586 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
588 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
589 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
590 get_word(buf1, sizeof(buf1), &p);
591 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
592 } else if (av_strstart(p, "range:", &p)) {
595 // this is so that seeking on a streamed file can work.
596 rtsp_parse_range_npt(p, &start, &end);
597 s->start_time = start;
598 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
599 s->duration = (end == AV_NOPTS_VALUE) ?
600 AV_NOPTS_VALUE : end - start;
601 } else if (av_strstart(p, "lang:", &p)) {
602 if (s->nb_streams > 0) {
603 get_word(buf1, sizeof(buf1), &p);
604 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
605 if (rtsp_st->stream_index >= 0) {
606 st = s->streams[rtsp_st->stream_index];
607 av_dict_set(&st->metadata, "language", buf1, 0);
610 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
611 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
613 rt->transport = RTSP_TRANSPORT_RDT;
614 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
616 st = s->streams[s->nb_streams - 1];
617 st->codecpar->sample_rate = atoi(p);
618 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
620 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
621 get_word(buf1, sizeof(buf1), &p); // ignore tag
622 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
623 p += strspn(p, SPACE_CHARS);
624 if (av_strstart(p, "inline:", &p))
625 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
626 } else if (av_strstart(p, "source-filter:", &p)) {
628 get_word(buf1, sizeof(buf1), &p);
629 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
631 exclude = !strcmp(buf1, "excl");
633 get_word(buf1, sizeof(buf1), &p);
634 if (strcmp(buf1, "IN") != 0)
636 get_word(buf1, sizeof(buf1), &p);
637 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
639 // not checking that the destination address actually matches or is wildcard
640 get_word(buf1, sizeof(buf1), &p);
643 rtsp_src = av_mallocz(sizeof(*rtsp_src));
646 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
648 if (s->nb_streams == 0) {
649 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
651 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
652 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
655 if (s->nb_streams == 0) {
656 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
658 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
659 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
664 if (rt->server_type == RTSP_SERVER_WMS)
665 ff_wms_parse_sdp_a_line(s, p);
666 if (s->nb_streams > 0) {
667 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
669 if (rt->server_type == RTSP_SERVER_REAL)
670 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
672 if (rtsp_st->dynamic_handler &&
673 rtsp_st->dynamic_handler->parse_sdp_a_line)
674 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
675 rtsp_st->stream_index,
676 rtsp_st->dynamic_protocol_context, buf);
683 int ff_sdp_parse(AVFormatContext *s, const char *content)
687 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
688 * contain long SDP lines containing complete ASF Headers (several
689 * kB) or arrays of MDPR (RM stream descriptor) headers plus
690 * "rulebooks" describing their properties. Therefore, the SDP line
693 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
694 * in rtpdec_xiph.c. */
696 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
700 p += strspn(p, SPACE_CHARS);
708 /* get the content */
710 while (*p != '\n' && *p != '\r' && *p != '\0') {
711 if ((q - buf) < sizeof(buf) - 1)
716 sdp_parse_line(s, s1, letter, buf);
718 while (*p != '\n' && *p != '\0')
724 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
725 av_freep(&s1->default_include_source_addrs[i]);
726 av_freep(&s1->default_include_source_addrs);
727 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
728 av_freep(&s1->default_exclude_source_addrs[i]);
729 av_freep(&s1->default_exclude_source_addrs);
733 #endif /* CONFIG_RTPDEC */
735 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
737 RTSPState *rt = s->priv_data;
740 for (i = 0; i < rt->nb_rtsp_streams; i++) {
741 RTSPStream *rtsp_st = rt->rtsp_streams[i];
744 if (rtsp_st->transport_priv) {
746 AVFormatContext *rtpctx = rtsp_st->transport_priv;
747 av_write_trailer(rtpctx);
748 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
749 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
750 ff_rtsp_tcp_write_packet(s, rtsp_st);
751 ffio_free_dyn_buf(&rtpctx->pb);
753 avio_closep(&rtpctx->pb);
755 avformat_free_context(rtpctx);
756 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
757 ff_rdt_parse_close(rtsp_st->transport_priv);
758 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
759 ff_rtp_parse_close(rtsp_st->transport_priv);
761 rtsp_st->transport_priv = NULL;
762 if (rtsp_st->rtp_handle)
763 ffurl_close(rtsp_st->rtp_handle);
764 rtsp_st->rtp_handle = NULL;
768 /* close and free RTSP streams */
769 void ff_rtsp_close_streams(AVFormatContext *s)
771 RTSPState *rt = s->priv_data;
775 ff_rtsp_undo_setup(s, 0);
776 for (i = 0; i < rt->nb_rtsp_streams; i++) {
777 rtsp_st = rt->rtsp_streams[i];
779 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
780 if (rtsp_st->dynamic_handler->close)
781 rtsp_st->dynamic_handler->close(
782 rtsp_st->dynamic_protocol_context);
783 av_free(rtsp_st->dynamic_protocol_context);
785 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
786 av_freep(&rtsp_st->include_source_addrs[j]);
787 av_freep(&rtsp_st->include_source_addrs);
788 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
789 av_freep(&rtsp_st->exclude_source_addrs[j]);
790 av_freep(&rtsp_st->exclude_source_addrs);
795 av_freep(&rt->rtsp_streams);
797 avformat_close_input(&rt->asf_ctx);
799 if (CONFIG_RTPDEC && rt->ts)
800 avpriv_mpegts_parse_close(rt->ts);
802 av_freep(&rt->recvbuf);
805 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
807 RTSPState *rt = s->priv_data;
809 int reordering_queue_size = rt->reordering_queue_size;
810 if (reordering_queue_size < 0) {
811 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
812 reordering_queue_size = 0;
814 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
817 /* open the RTP context */
818 if (rtsp_st->stream_index >= 0)
819 st = s->streams[rtsp_st->stream_index];
821 s->ctx_flags |= AVFMTCTX_NOHEADER;
823 if (CONFIG_RTSP_MUXER && s->oformat && st) {
824 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
825 s, st, rtsp_st->rtp_handle,
826 RTSP_TCP_MAX_PACKET_SIZE,
827 rtsp_st->stream_index);
828 /* Ownership of rtp_handle is passed to the rtp mux context */
829 rtsp_st->rtp_handle = NULL;
832 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
833 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
834 return 0; // Don't need to open any parser here
835 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
836 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
837 rtsp_st->dynamic_protocol_context,
838 rtsp_st->dynamic_handler);
839 else if (CONFIG_RTPDEC)
840 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
841 rtsp_st->sdp_payload_type,
842 reordering_queue_size);
844 if (!rtsp_st->transport_priv) {
845 return AVERROR(ENOMEM);
846 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
848 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
849 rtpctx->ssrc = rtsp_st->ssrc;
850 if (rtsp_st->dynamic_handler) {
851 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
852 rtsp_st->dynamic_protocol_context,
853 rtsp_st->dynamic_handler);
855 if (rtsp_st->crypto_suite[0])
856 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
857 rtsp_st->crypto_suite,
858 rtsp_st->crypto_params);
864 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
865 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
872 q += strspn(q, SPACE_CHARS);
873 v = strtol(q, &p, 10);
877 v = strtol(p, &p, 10);
886 /* XXX: only one transport specification is parsed */
887 static void rtsp_parse_transport(AVFormatContext *s,
888 RTSPMessageHeader *reply, const char *p)
890 char transport_protocol[16];
892 char lower_transport[16];
894 RTSPTransportField *th;
897 reply->nb_transports = 0;
900 p += strspn(p, SPACE_CHARS);
904 th = &reply->transports[reply->nb_transports];
906 get_word_sep(transport_protocol, sizeof(transport_protocol),
908 if (!av_strcasecmp (transport_protocol, "rtp")) {
909 get_word_sep(profile, sizeof(profile), "/;,", &p);
910 lower_transport[0] = '\0';
911 /* rtp/avp/<protocol> */
913 get_word_sep(lower_transport, sizeof(lower_transport),
916 th->transport = RTSP_TRANSPORT_RTP;
917 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
918 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
919 /* x-pn-tng/<protocol> */
920 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
922 th->transport = RTSP_TRANSPORT_RDT;
923 } else if (!av_strcasecmp(transport_protocol, "raw")) {
924 get_word_sep(profile, sizeof(profile), "/;,", &p);
925 lower_transport[0] = '\0';
926 /* raw/raw/<protocol> */
928 get_word_sep(lower_transport, sizeof(lower_transport),
931 th->transport = RTSP_TRANSPORT_RAW;
933 if (!av_strcasecmp(lower_transport, "TCP"))
934 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
936 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
940 /* get each parameter */
941 while (*p != '\0' && *p != ',') {
942 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
943 if (!strcmp(parameter, "port")) {
946 rtsp_parse_range(&th->port_min, &th->port_max, &p);
948 } else if (!strcmp(parameter, "client_port")) {
951 rtsp_parse_range(&th->client_port_min,
952 &th->client_port_max, &p);
954 } else if (!strcmp(parameter, "server_port")) {
957 rtsp_parse_range(&th->server_port_min,
958 &th->server_port_max, &p);
960 } else if (!strcmp(parameter, "interleaved")) {
963 rtsp_parse_range(&th->interleaved_min,
964 &th->interleaved_max, &p);
966 } else if (!strcmp(parameter, "multicast")) {
967 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
968 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
969 } else if (!strcmp(parameter, "ttl")) {
973 th->ttl = strtol(p, &end, 10);
976 } else if (!strcmp(parameter, "destination")) {
979 get_word_sep(buf, sizeof(buf), ";,", &p);
980 get_sockaddr(s, buf, &th->destination);
982 } else if (!strcmp(parameter, "source")) {
985 get_word_sep(buf, sizeof(buf), ";,", &p);
986 av_strlcpy(th->source, buf, sizeof(th->source));
988 } else if (!strcmp(parameter, "mode")) {
991 get_word_sep(buf, sizeof(buf), ";, ", &p);
992 if (!strcmp(buf, "record") ||
993 !strcmp(buf, "receive"))
998 while (*p != ';' && *p != '\0' && *p != ',')
1006 reply->nb_transports++;
1007 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1012 static void handle_rtp_info(RTSPState *rt, const char *url,
1013 uint32_t seq, uint32_t rtptime)
1016 if (!rtptime || !url[0])
1018 if (rt->transport != RTSP_TRANSPORT_RTP)
1020 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1021 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1022 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1025 if (!strcmp(rtsp_st->control_url, url)) {
1026 rtpctx->base_timestamp = rtptime;
1032 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1035 char key[20], value[1024], url[1024] = "";
1036 uint32_t seq = 0, rtptime = 0;
1039 p += strspn(p, SPACE_CHARS);
1042 get_word_sep(key, sizeof(key), "=", &p);
1046 get_word_sep(value, sizeof(value), ";, ", &p);
1048 if (!strcmp(key, "url"))
1049 av_strlcpy(url, value, sizeof(url));
1050 else if (!strcmp(key, "seq"))
1051 seq = strtoul(value, NULL, 10);
1052 else if (!strcmp(key, "rtptime"))
1053 rtptime = strtoul(value, NULL, 10);
1055 handle_rtp_info(rt, url, seq, rtptime);
1064 handle_rtp_info(rt, url, seq, rtptime);
1067 void ff_rtsp_parse_line(AVFormatContext *s,
1068 RTSPMessageHeader *reply, const char *buf,
1069 RTSPState *rt, const char *method)
1073 /* NOTE: we do case independent match for broken servers */
1075 if (av_stristart(p, "Session:", &p)) {
1077 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1078 if (av_stristart(p, ";timeout=", &p) &&
1079 (t = strtol(p, NULL, 10)) > 0) {
1082 } else if (av_stristart(p, "Content-Length:", &p)) {
1083 reply->content_length = strtol(p, NULL, 10);
1084 } else if (av_stristart(p, "Transport:", &p)) {
1085 rtsp_parse_transport(s, reply, p);
1086 } else if (av_stristart(p, "CSeq:", &p)) {
1087 reply->seq = strtol(p, NULL, 10);
1088 } else if (av_stristart(p, "Range:", &p)) {
1089 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1090 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1091 p += strspn(p, SPACE_CHARS);
1092 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1093 } else if (av_stristart(p, "Server:", &p)) {
1094 p += strspn(p, SPACE_CHARS);
1095 av_strlcpy(reply->server, p, sizeof(reply->server));
1096 } else if (av_stristart(p, "Notice:", &p) ||
1097 av_stristart(p, "X-Notice:", &p)) {
1098 reply->notice = strtol(p, NULL, 10);
1099 } else if (av_stristart(p, "Location:", &p)) {
1100 p += strspn(p, SPACE_CHARS);
1101 av_strlcpy(reply->location, p , sizeof(reply->location));
1102 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1103 p += strspn(p, SPACE_CHARS);
1104 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1105 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1106 p += strspn(p, SPACE_CHARS);
1107 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1108 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1109 p += strspn(p, SPACE_CHARS);
1110 if (method && !strcmp(method, "DESCRIBE"))
1111 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1112 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1113 p += strspn(p, SPACE_CHARS);
1114 if (method && !strcmp(method, "PLAY"))
1115 rtsp_parse_rtp_info(rt, p);
1116 } else if (av_stristart(p, "Public:", &p) && rt) {
1117 if (strstr(p, "GET_PARAMETER") &&
1118 method && !strcmp(method, "OPTIONS"))
1119 rt->get_parameter_supported = 1;
1120 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1121 p += strspn(p, SPACE_CHARS);
1122 rt->accept_dynamic_rate = atoi(p);
1123 } else if (av_stristart(p, "Content-Type:", &p)) {
1124 p += strspn(p, SPACE_CHARS);
1125 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1129 /* skip a RTP/TCP interleaved packet */
1130 void ff_rtsp_skip_packet(AVFormatContext *s)
1132 RTSPState *rt = s->priv_data;
1136 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1139 len = AV_RB16(buf + 1);
1141 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1146 if (len1 > sizeof(buf))
1148 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1155 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1156 unsigned char **content_ptr,
1157 int return_on_interleaved_data, const char *method)
1159 RTSPState *rt = s->priv_data;
1160 char buf[4096], buf1[1024], *q;
1163 int ret, content_length, line_count = 0, request = 0;
1164 unsigned char *content = NULL;
1170 memset(reply, 0, sizeof(*reply));
1172 /* parse reply (XXX: use buffers) */
1173 rt->last_reply[0] = '\0';
1177 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1178 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1183 if (ch == '$' && q == buf) {
1184 if (return_on_interleaved_data) {
1187 ff_rtsp_skip_packet(s);
1188 } else if (ch != '\r') {
1189 if ((q - buf) < sizeof(buf) - 1)
1195 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1197 /* test if last line */
1201 if (line_count == 0) {
1202 /* get reply code */
1203 get_word(buf1, sizeof(buf1), &p);
1204 if (!strncmp(buf1, "RTSP/", 5)) {
1205 get_word(buf1, sizeof(buf1), &p);
1206 reply->status_code = atoi(buf1);
1207 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1209 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1210 get_word(buf1, sizeof(buf1), &p); // object
1214 ff_rtsp_parse_line(s, reply, p, rt, method);
1215 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1216 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1221 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1222 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1224 content_length = reply->content_length;
1225 if (content_length > 0) {
1226 /* leave some room for a trailing '\0' (useful for simple parsing) */
1227 content = av_malloc(content_length + 1);
1229 return AVERROR(ENOMEM);
1230 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1231 content[content_length] = '\0';
1234 *content_ptr = content;
1240 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1241 const char* ptr = buf;
1243 if (!strcmp(reply->reason, "OPTIONS")) {
1244 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1246 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1247 if (reply->session_id[0])
1248 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1251 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1253 av_strlcat(buf, "\r\n", sizeof(buf));
1255 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1256 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1259 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1261 rt->last_cmd_time = av_gettime_relative();
1262 /* Even if the request from the server had data, it is not the data
1263 * that the caller wants or expects. The memory could also be leaked
1264 * if the actual following reply has content data. */
1266 av_freep(content_ptr);
1267 /* If method is set, this is called from ff_rtsp_send_cmd,
1268 * where a reply to exactly this request is awaited. For
1269 * callers from within packet receiving, we just want to
1270 * return to the caller and go back to receiving packets. */
1276 if (rt->seq != reply->seq) {
1277 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1278 rt->seq, reply->seq);
1282 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1283 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1284 reply->notice == 2306 /* Continuous Feed Terminated */) {
1285 rt->state = RTSP_STATE_IDLE;
1286 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1287 return AVERROR(EIO); /* data or server error */
1288 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1289 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1290 return AVERROR(EPERM);
1296 * Send a command to the RTSP server without waiting for the reply.
1298 * @param s RTSP (de)muxer context
1299 * @param method the method for the request
1300 * @param url the target url for the request
1301 * @param headers extra header lines to include in the request
1302 * @param send_content if non-null, the data to send as request body content
1303 * @param send_content_length the length of the send_content data, or 0 if
1304 * send_content is null
1306 * @return zero if success, nonzero otherwise
1308 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1309 const char *method, const char *url,
1310 const char *headers,
1311 const unsigned char *send_content,
1312 int send_content_length)
1314 RTSPState *rt = s->priv_data;
1315 char buf[4096], *out_buf;
1316 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1318 /* Add in RTSP headers */
1321 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1323 av_strlcat(buf, headers, sizeof(buf));
1324 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1325 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1326 if (rt->session_id[0] != '\0' && (!headers ||
1327 !strstr(headers, "\nIf-Match:"))) {
1328 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1331 char *str = ff_http_auth_create_response(&rt->auth_state,
1332 rt->auth, url, method);
1334 av_strlcat(buf, str, sizeof(buf));
1337 if (send_content_length > 0 && send_content)
1338 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1339 av_strlcat(buf, "\r\n", sizeof(buf));
1341 /* base64 encode rtsp if tunneling */
1342 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1343 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1344 out_buf = base64buf;
1347 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1349 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1350 if (send_content_length > 0 && send_content) {
1351 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1352 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1353 return AVERROR_PATCHWELCOME;
1355 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1357 rt->last_cmd_time = av_gettime_relative();
1362 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1363 const char *url, const char *headers)
1365 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1368 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1369 const char *headers, RTSPMessageHeader *reply,
1370 unsigned char **content_ptr)
1372 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1373 content_ptr, NULL, 0);
1376 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1377 const char *method, const char *url,
1379 RTSPMessageHeader *reply,
1380 unsigned char **content_ptr,
1381 const unsigned char *send_content,
1382 int send_content_length)
1384 RTSPState *rt = s->priv_data;
1385 HTTPAuthType cur_auth_type;
1386 int ret, attempts = 0;
1389 cur_auth_type = rt->auth_state.auth_type;
1390 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1392 send_content_length)))
1395 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1399 if (reply->status_code == 401 &&
1400 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1401 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1404 if (reply->status_code > 400){
1405 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1409 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1415 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1416 int lower_transport, const char *real_challenge)
1418 RTSPState *rt = s->priv_data;
1419 int rtx = 0, j, i, err, interleave = 0, port_off;
1420 RTSPStream *rtsp_st;
1421 RTSPMessageHeader reply1, *reply = &reply1;
1423 const char *trans_pref;
1425 if (rt->transport == RTSP_TRANSPORT_RDT)
1426 trans_pref = "x-pn-tng";
1427 else if (rt->transport == RTSP_TRANSPORT_RAW)
1428 trans_pref = "RAW/RAW";
1430 trans_pref = "RTP/AVP";
1432 /* default timeout: 1 minute */
1435 /* Choose a random starting offset within the first half of the
1436 * port range, to allow for a number of ports to try even if the offset
1437 * happens to be at the end of the random range. */
1438 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1439 /* even random offset */
1440 port_off -= port_off & 0x01;
1442 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1443 char transport[2048];
1446 * WMS serves all UDP data over a single connection, the RTX, which
1447 * isn't necessarily the first in the SDP but has to be the first
1448 * to be set up, else the second/third SETUP will fail with a 461.
1450 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1451 rt->server_type == RTSP_SERVER_WMS) {
1454 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1455 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1457 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1461 if (rtx == rt->nb_rtsp_streams)
1462 return -1; /* no RTX found */
1463 rtsp_st = rt->rtsp_streams[rtx];
1465 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1467 rtsp_st = rt->rtsp_streams[i];
1470 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1473 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1474 port = reply->transports[0].client_port_min;
1478 /* first try in specified port range */
1479 while (j <= rt->rtp_port_max) {
1480 AVDictionary *opts = map_to_opts(rt);
1482 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1483 "?localport=%d", j);
1484 /* we will use two ports per rtp stream (rtp and rtcp) */
1486 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1487 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1489 av_dict_free(&opts);
1494 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1499 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1501 snprintf(transport, sizeof(transport) - 1,
1502 "%s/UDP;", trans_pref);
1503 if (rt->server_type != RTSP_SERVER_REAL)
1504 av_strlcat(transport, "unicast;", sizeof(transport));
1505 av_strlcatf(transport, sizeof(transport),
1506 "client_port=%d", port);
1507 if (rt->transport == RTSP_TRANSPORT_RTP &&
1508 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1509 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1513 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1514 /* For WMS streams, the application streams are only used for
1515 * UDP. When trying to set it up for TCP streams, the server
1516 * will return an error. Therefore, we skip those streams. */
1517 if (rt->server_type == RTSP_SERVER_WMS &&
1518 (rtsp_st->stream_index < 0 ||
1519 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1522 snprintf(transport, sizeof(transport) - 1,
1523 "%s/TCP;", trans_pref);
1524 if (rt->transport != RTSP_TRANSPORT_RDT)
1525 av_strlcat(transport, "unicast;", sizeof(transport));
1526 av_strlcatf(transport, sizeof(transport),
1527 "interleaved=%d-%d",
1528 interleave, interleave + 1);
1532 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1533 snprintf(transport, sizeof(transport) - 1,
1534 "%s/UDP;multicast", trans_pref);
1537 av_strlcat(transport, ";mode=record", sizeof(transport));
1538 } else if (rt->server_type == RTSP_SERVER_REAL ||
1539 rt->server_type == RTSP_SERVER_WMS)
1540 av_strlcat(transport, ";mode=play", sizeof(transport));
1541 snprintf(cmd, sizeof(cmd),
1542 "Transport: %s\r\n",
1544 if (rt->accept_dynamic_rate)
1545 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1546 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1547 char real_res[41], real_csum[9];
1548 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1550 av_strlcatf(cmd, sizeof(cmd),
1552 "RealChallenge2: %s, sd=%s\r\n",
1553 rt->session_id, real_res, real_csum);
1555 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1556 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1559 } else if (reply->status_code != RTSP_STATUS_OK ||
1560 reply->nb_transports != 1) {
1561 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1565 /* XXX: same protocol for all streams is required */
1567 if (reply->transports[0].lower_transport != rt->lower_transport ||
1568 reply->transports[0].transport != rt->transport) {
1569 err = AVERROR_INVALIDDATA;
1573 rt->lower_transport = reply->transports[0].lower_transport;
1574 rt->transport = reply->transports[0].transport;
1577 /* Fail if the server responded with another lower transport mode
1578 * than what we requested. */
1579 if (reply->transports[0].lower_transport != lower_transport) {
1580 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1581 err = AVERROR_INVALIDDATA;
1585 switch(reply->transports[0].lower_transport) {
1586 case RTSP_LOWER_TRANSPORT_TCP:
1587 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1588 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1591 case RTSP_LOWER_TRANSPORT_UDP: {
1592 char url[1024], options[30] = "";
1593 const char *peer = host;
1595 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1596 av_strlcpy(options, "?connect=1", sizeof(options));
1597 /* Use source address if specified */
1598 if (reply->transports[0].source[0])
1599 peer = reply->transports[0].source;
1600 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1601 reply->transports[0].server_port_min, "%s", options);
1602 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1603 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1604 err = AVERROR_INVALIDDATA;
1609 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1610 char url[1024], namebuf[50], optbuf[20] = "";
1611 struct sockaddr_storage addr;
1614 if (reply->transports[0].destination.ss_family) {
1615 addr = reply->transports[0].destination;
1616 port = reply->transports[0].port_min;
1617 ttl = reply->transports[0].ttl;
1619 addr = rtsp_st->sdp_ip;
1620 port = rtsp_st->sdp_port;
1621 ttl = rtsp_st->sdp_ttl;
1624 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1625 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1626 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1627 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1628 port, "%s", optbuf);
1629 if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1630 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL) < 0) {
1631 err = AVERROR_INVALIDDATA;
1638 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1642 if (rt->nb_rtsp_streams && reply->timeout > 0)
1643 rt->timeout = reply->timeout;
1645 if (rt->server_type == RTSP_SERVER_REAL)
1646 rt->need_subscription = 1;
1651 ff_rtsp_undo_setup(s, 0);
1655 void ff_rtsp_close_connections(AVFormatContext *s)
1657 RTSPState *rt = s->priv_data;
1658 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1659 ffurl_close(rt->rtsp_hd);
1660 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1663 int ff_rtsp_connect(AVFormatContext *s)
1665 RTSPState *rt = s->priv_data;
1666 char proto[128], host[1024], path[1024];
1667 char tcpname[1024], cmd[2048], auth[128];
1668 const char *lower_rtsp_proto = "tcp";
1669 int port, err, tcp_fd;
1670 RTSPMessageHeader reply1, *reply = &reply1;
1671 int lower_transport_mask = 0;
1672 int default_port = RTSP_DEFAULT_PORT;
1673 int https_tunnel = 0;
1674 char real_challenge[64] = "";
1675 struct sockaddr_storage peer;
1676 socklen_t peer_len = sizeof(peer);
1678 if (rt->rtp_port_max < rt->rtp_port_min) {
1679 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1680 "than min port %d\n", rt->rtp_port_max,
1682 return AVERROR(EINVAL);
1685 if (!ff_network_init())
1686 return AVERROR(EIO);
1688 if (s->max_delay < 0) /* Not set by the caller */
1689 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1691 rt->control_transport = RTSP_MODE_PLAIN;
1692 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1693 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1694 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1695 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1696 rt->control_transport = RTSP_MODE_TUNNEL;
1698 /* Only pass through valid flags from here */
1699 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1702 memset(&reply1, 0, sizeof(reply1));
1703 /* extract hostname and port */
1704 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1705 host, sizeof(host), &port, path, sizeof(path), s->url);
1707 if (!strcmp(proto, "rtsps")) {
1708 lower_rtsp_proto = "tls";
1709 default_port = RTSPS_DEFAULT_PORT;
1710 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1714 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1717 port = default_port;
1719 lower_transport_mask = rt->lower_transport_mask;
1721 if (!lower_transport_mask)
1722 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1725 /* Only UDP or TCP - UDP multicast isn't supported. */
1726 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1727 (1 << RTSP_LOWER_TRANSPORT_TCP);
1728 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1729 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1730 "only UDP and TCP are supported for output.\n");
1731 err = AVERROR(EINVAL);
1736 /* Construct the URI used in request; this is similar to s->url,
1737 * but with authentication credentials removed and RTSP specific options
1739 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1740 host, port, "%s", path);
1742 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1743 /* set up initial handshake for tunneling */
1744 char httpname[1024];
1745 char sessioncookie[17];
1748 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1749 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1750 av_get_random_seed(), av_get_random_seed());
1753 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1754 &s->interrupt_callback) < 0) {
1759 /* generate GET headers */
1760 snprintf(headers, sizeof(headers),
1761 "x-sessioncookie: %s\r\n"
1762 "Accept: application/x-rtsp-tunnelled\r\n"
1763 "Pragma: no-cache\r\n"
1764 "Cache-Control: no-cache\r\n",
1766 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1768 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1769 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1770 if (!rt->rtsp_hd->protocol_whitelist) {
1771 err = AVERROR(ENOMEM);
1776 /* complete the connection */
1777 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1783 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1784 &s->interrupt_callback) < 0 ) {
1789 /* generate POST headers */
1790 snprintf(headers, sizeof(headers),
1791 "x-sessioncookie: %s\r\n"
1792 "Content-Type: application/x-rtsp-tunnelled\r\n"
1793 "Pragma: no-cache\r\n"
1794 "Cache-Control: no-cache\r\n"
1795 "Content-Length: 32767\r\n"
1796 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1798 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1799 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1800 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1802 /* Initialize the authentication state for the POST session. The HTTP
1803 * protocol implementation doesn't properly handle multi-pass
1804 * authentication for POST requests, since it would require one of
1806 * - implementing Expect: 100-continue, which many HTTP servers
1807 * don't support anyway, even less the RTSP servers that do HTTP
1809 * - sending the whole POST data until getting a 401 reply specifying
1810 * what authentication method to use, then resending all that data
1811 * - waiting for potential 401 replies directly after sending the
1812 * POST header (waiting for some unspecified time)
1813 * Therefore, we copy the full auth state, which works for both basic
1814 * and digest. (For digest, we would have to synchronize the nonce
1815 * count variable between the two sessions, if we'd do more requests
1816 * with the original session, though.)
1818 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1820 /* complete the connection */
1821 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1827 /* open the tcp connection */
1828 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1830 "?timeout=%d", rt->stimeout);
1831 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1832 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1836 rt->rtsp_hd_out = rt->rtsp_hd;
1840 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1845 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1846 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1847 NULL, 0, NI_NUMERICHOST);
1850 /* request options supported by the server; this also detects server
1852 for (rt->server_type = RTSP_SERVER_RTP;;) {
1854 if (rt->server_type == RTSP_SERVER_REAL)
1857 * The following entries are required for proper
1858 * streaming from a Realmedia server. They are
1859 * interdependent in some way although we currently
1860 * don't quite understand how. Values were copied
1861 * from mplayer SVN r23589.
1862 * ClientChallenge is a 16-byte ID in hex
1863 * CompanyID is a 16-byte ID in base64
1865 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1866 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1867 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1868 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1870 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1871 if (reply->status_code != RTSP_STATUS_OK) {
1872 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1876 /* detect server type if not standard-compliant RTP */
1877 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1878 rt->server_type = RTSP_SERVER_REAL;
1880 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1881 rt->server_type = RTSP_SERVER_WMS;
1882 } else if (rt->server_type == RTSP_SERVER_REAL)
1883 strcpy(real_challenge, reply->real_challenge);
1887 if (CONFIG_RTSP_DEMUXER && s->iformat)
1888 err = ff_rtsp_setup_input_streams(s, reply);
1889 else if (CONFIG_RTSP_MUXER)
1890 err = ff_rtsp_setup_output_streams(s, host);
1897 int lower_transport = ff_log2_tab[lower_transport_mask &
1898 ~(lower_transport_mask - 1)];
1900 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1901 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1902 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1904 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1905 rt->server_type == RTSP_SERVER_REAL ?
1906 real_challenge : NULL);
1909 lower_transport_mask &= ~(1 << lower_transport);
1910 if (lower_transport_mask == 0 && err == 1) {
1911 err = AVERROR(EPROTONOSUPPORT);
1916 rt->lower_transport_mask = lower_transport_mask;
1917 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1918 rt->state = RTSP_STATE_IDLE;
1919 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1922 ff_rtsp_close_streams(s);
1923 ff_rtsp_close_connections(s);
1924 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1925 char *new_url = av_strdup(reply->location);
1927 err = AVERROR(ENOMEM);
1930 ff_format_set_url(s, new_url);
1931 rt->session_id[0] = '\0';
1932 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1941 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1944 static int parse_rtsp_message(AVFormatContext *s)
1946 RTSPState *rt = s->priv_data;
1949 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1950 if (rt->state == RTSP_STATE_STREAMING) {
1951 if (!ff_rtsp_parse_streaming_commands(s))
1954 av_log(s, AV_LOG_WARNING,
1955 "Unable to answer to TEARDOWN\n");
1959 RTSPMessageHeader reply;
1960 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1963 /* XXX: parse message */
1964 if (rt->state != RTSP_STATE_STREAMING)
1971 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1972 uint8_t *buf, int buf_size, int64_t wait_end)
1974 RTSPState *rt = s->priv_data;
1975 RTSPStream *rtsp_st;
1976 int n, i, ret, timeout_cnt = 0;
1977 struct pollfd *p = rt->p;
1978 int *fds = NULL, fdsnum, fdsidx;
1981 p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1983 return AVERROR(ENOMEM);
1986 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1987 p[rt->max_p++].events = POLLIN;
1989 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1990 rtsp_st = rt->rtsp_streams[i];
1991 if (rtsp_st->rtp_handle) {
1992 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1994 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1998 av_log(s, AV_LOG_ERROR,
1999 "Number of fds %d not supported\n", fdsnum);
2000 return AVERROR_INVALIDDATA;
2002 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2003 p[rt->max_p].fd = fds[fdsidx];
2004 p[rt->max_p++].events = POLLIN;
2012 if (ff_check_interrupt(&s->interrupt_callback))
2013 return AVERROR_EXIT;
2014 if (wait_end && wait_end - av_gettime_relative() < 0)
2015 return AVERROR(EAGAIN);
2016 n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2018 int j = rt->rtsp_hd ? 1 : 0;
2020 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2021 rtsp_st = rt->rtsp_streams[i];
2022 if (rtsp_st->rtp_handle) {
2023 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2024 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2026 *prtsp_st = rtsp_st;
2033 #if CONFIG_RTSP_DEMUXER
2034 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2035 if ((ret = parse_rtsp_message(s)) < 0) {
2040 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2041 return AVERROR(ETIMEDOUT);
2042 } else if (n < 0 && errno != EINTR)
2043 return AVERROR(errno);
2047 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2048 const uint8_t *buf, int len)
2050 RTSPState *rt = s->priv_data;
2054 if (rt->nb_rtsp_streams == 1) {
2055 *rtsp_st = rt->rtsp_streams[0];
2058 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2059 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2061 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2062 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2065 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2066 *rtsp_st = rt->rtsp_streams[i];
2073 av_log(s, AV_LOG_WARNING,
2074 "Unable to pick stream for packet - SSRC not known for "
2076 return AVERROR(EAGAIN);
2079 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2080 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2081 *rtsp_st = rt->rtsp_streams[i];
2087 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2088 return AVERROR(EAGAIN);
2091 static int read_packet(AVFormatContext *s,
2092 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2095 RTSPState *rt = s->priv_data;
2098 switch(rt->lower_transport) {
2100 #if CONFIG_RTSP_DEMUXER
2101 case RTSP_LOWER_TRANSPORT_TCP:
2102 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2105 case RTSP_LOWER_TRANSPORT_UDP:
2106 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2107 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2108 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2109 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2111 case RTSP_LOWER_TRANSPORT_CUSTOM:
2112 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2113 wait_end && wait_end < av_gettime_relative())
2114 len = AVERROR(EAGAIN);
2116 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2117 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2118 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2119 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2129 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2131 RTSPState *rt = s->priv_data;
2133 RTSPStream *rtsp_st, *first_queue_st = NULL;
2134 int64_t wait_end = 0;
2136 if (rt->nb_byes == rt->nb_rtsp_streams)
2139 /* get next frames from the same RTP packet */
2140 if (rt->cur_transport_priv) {
2141 if (rt->transport == RTSP_TRANSPORT_RDT) {
2142 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2143 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2144 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2145 } else if (CONFIG_RTPDEC && rt->ts) {
2146 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2148 rt->recvbuf_pos += ret;
2149 ret = rt->recvbuf_pos < rt->recvbuf_len;
2154 rt->cur_transport_priv = NULL;
2156 } else if (ret == 1) {
2159 rt->cur_transport_priv = NULL;
2163 if (rt->transport == RTSP_TRANSPORT_RTP) {
2165 int64_t first_queue_time = 0;
2166 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2167 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2171 queue_time = ff_rtp_queued_packet_time(rtpctx);
2172 if (queue_time && (queue_time - first_queue_time < 0 ||
2173 !first_queue_time)) {
2174 first_queue_time = queue_time;
2175 first_queue_st = rt->rtsp_streams[i];
2178 if (first_queue_time) {
2179 wait_end = first_queue_time + s->max_delay;
2182 first_queue_st = NULL;
2186 /* read next RTP packet */
2188 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2190 return AVERROR(ENOMEM);
2193 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2194 if (len == AVERROR(EAGAIN) && first_queue_st &&
2195 rt->transport == RTSP_TRANSPORT_RTP) {
2196 av_log(s, AV_LOG_WARNING,
2197 "max delay reached. need to consume packet\n");
2198 rtsp_st = first_queue_st;
2199 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2205 if (rt->transport == RTSP_TRANSPORT_RDT) {
2206 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2207 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2208 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2209 if (rtsp_st->feedback) {
2210 AVIOContext *pb = NULL;
2211 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2213 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2216 /* Either bad packet, or a RTCP packet. Check if the
2217 * first_rtcp_ntp_time field was initialized. */
2218 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2219 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2220 /* first_rtcp_ntp_time has been initialized for this stream,
2221 * copy the same value to all other uninitialized streams,
2222 * in order to map their timestamp origin to the same ntp time
2225 AVStream *st = NULL;
2226 if (rtsp_st->stream_index >= 0)
2227 st = s->streams[rtsp_st->stream_index];
2228 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2229 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2230 AVStream *st2 = NULL;
2231 if (rt->rtsp_streams[i]->stream_index >= 0)
2232 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2233 if (rtpctx2 && st && st2 &&
2234 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2235 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2236 rtpctx2->rtcp_ts_offset = av_rescale_q(
2237 rtpctx->rtcp_ts_offset, st->time_base,
2241 // Make real NTP start time available in AVFormatContext
2242 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2243 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2245 s->start_time_realtime -=
2246 av_rescale (rtpctx->rtcp_ts_offset,
2247 (uint64_t) rtpctx->st->time_base.num * 1000000,
2248 rtpctx->st->time_base.den);
2252 if (ret == -RTCP_BYE) {
2255 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2256 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2258 if (rt->nb_byes == rt->nb_rtsp_streams)
2262 } else if (CONFIG_RTPDEC && rt->ts) {
2263 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2266 rt->recvbuf_len = len;
2267 rt->recvbuf_pos = ret;
2268 rt->cur_transport_priv = rt->ts;
2275 return AVERROR_INVALIDDATA;
2281 /* more packets may follow, so we save the RTP context */
2282 rt->cur_transport_priv = rtsp_st->transport_priv;
2286 #endif /* CONFIG_RTPDEC */
2288 #if CONFIG_SDP_DEMUXER
2289 static int sdp_probe(const AVProbeData *p1)
2291 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2293 /* we look for a line beginning "c=IN IP" */
2294 while (p < p_end && *p != '\0') {
2295 if (sizeof("c=IN IP") - 1 < p_end - p &&
2296 av_strstart(p, "c=IN IP", NULL))
2297 return AVPROBE_SCORE_EXTENSION;
2299 while (p < p_end - 1 && *p != '\n') p++;
2308 static void append_source_addrs(char *buf, int size, const char *name,
2309 int count, struct RTSPSource **addrs)
2314 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2315 for (i = 1; i < count; i++)
2316 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2319 static int sdp_read_header(AVFormatContext *s)
2321 RTSPState *rt = s->priv_data;
2322 RTSPStream *rtsp_st;
2327 if (!ff_network_init())
2328 return AVERROR(EIO);
2330 if (s->max_delay < 0) /* Not set by the caller */
2331 s->max_delay = DEFAULT_REORDERING_DELAY;
2332 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2333 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2335 /* read the whole sdp file */
2336 /* XXX: better loading */
2337 content = av_malloc(SDP_MAX_SIZE);
2339 return AVERROR(ENOMEM);
2340 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2343 return AVERROR_INVALIDDATA;
2345 content[size] ='\0';
2347 err = ff_sdp_parse(s, content);
2351 /* open each RTP stream */
2352 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2354 rtsp_st = rt->rtsp_streams[i];
2356 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2357 AVDictionary *opts = map_to_opts(rt);
2359 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2360 sizeof(rtsp_st->sdp_ip),
2361 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2363 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2365 av_dict_free(&opts);
2368 ff_url_join(url, sizeof(url), "rtp", NULL,
2369 namebuf, rtsp_st->sdp_port,
2370 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2371 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2372 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2373 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2375 append_source_addrs(url, sizeof(url), "sources",
2376 rtsp_st->nb_include_source_addrs,
2377 rtsp_st->include_source_addrs);
2378 append_source_addrs(url, sizeof(url), "block",
2379 rtsp_st->nb_exclude_source_addrs,
2380 rtsp_st->exclude_source_addrs);
2381 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2382 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2384 av_dict_free(&opts);
2387 err = AVERROR_INVALIDDATA;
2391 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2396 ff_rtsp_close_streams(s);
2401 static int sdp_read_close(AVFormatContext *s)
2403 ff_rtsp_close_streams(s);
2408 static const AVClass sdp_demuxer_class = {
2409 .class_name = "SDP demuxer",
2410 .item_name = av_default_item_name,
2411 .option = sdp_options,
2412 .version = LIBAVUTIL_VERSION_INT,
2415 AVInputFormat ff_sdp_demuxer = {
2417 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2418 .priv_data_size = sizeof(RTSPState),
2419 .read_probe = sdp_probe,
2420 .read_header = sdp_read_header,
2421 .read_packet = ff_rtsp_fetch_packet,
2422 .read_close = sdp_read_close,
2423 .priv_class = &sdp_demuxer_class,
2425 #endif /* CONFIG_SDP_DEMUXER */
2427 #if CONFIG_RTP_DEMUXER
2428 static int rtp_probe(const AVProbeData *p)
2430 if (av_strstart(p->filename, "rtp:", NULL))
2431 return AVPROBE_SCORE_MAX;
2435 static int rtp_read_header(AVFormatContext *s)
2437 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2438 char host[500], sdp[500];
2440 URLContext* in = NULL;
2442 AVCodecParameters *par = NULL;
2443 struct sockaddr_storage addr;
2445 socklen_t addrlen = sizeof(addr);
2446 RTSPState *rt = s->priv_data;
2448 if (!ff_network_init())
2449 return AVERROR(EIO);
2451 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2452 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2457 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2458 if (ret == AVERROR(EAGAIN))
2463 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2467 if ((recvbuf[0] & 0xc0) != 0x80) {
2468 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2473 if (RTP_PT_IS_RTCP(recvbuf[1]))
2476 payload_type = recvbuf[1] & 0x7f;
2479 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2483 par = avcodec_parameters_alloc();
2485 ret = AVERROR(ENOMEM);
2489 if (ff_rtp_get_codec_info(par, payload_type)) {
2490 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2491 "without an SDP file describing it\n",
2495 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2496 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2497 "properly you need an SDP file "
2501 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2504 snprintf(sdp, sizeof(sdp),
2505 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2506 addr.ss_family == AF_INET ? 4 : 6, host,
2507 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2508 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2509 port, payload_type);
2510 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2511 avcodec_parameters_free(&par);
2513 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2516 /* sdp_read_header initializes this again */
2519 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2521 ret = sdp_read_header(s);
2526 avcodec_parameters_free(&par);
2533 static const AVClass rtp_demuxer_class = {
2534 .class_name = "RTP demuxer",
2535 .item_name = av_default_item_name,
2536 .option = rtp_options,
2537 .version = LIBAVUTIL_VERSION_INT,
2540 AVInputFormat ff_rtp_demuxer = {
2542 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2543 .priv_data_size = sizeof(RTSPState),
2544 .read_probe = rtp_probe,
2545 .read_header = rtp_read_header,
2546 .read_packet = ff_rtsp_fetch_packet,
2547 .read_close = sdp_read_close,
2548 .flags = AVFMT_NOFILE,
2549 .priv_class = &rtp_demuxer_class,
2551 #endif /* CONFIG_RTP_DEMUXER */