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 }, \
80 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
83 const AVOption ff_rtsp_options[] = {
84 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
85 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
86 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
87 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
89 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
90 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
91 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
92 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
93 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
94 { "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" },
95 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
96 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
97 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
98 { "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 },
99 #if FF_API_OLD_RTSP_OPTIONS
100 { "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 },
101 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
103 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
106 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
107 #if FF_API_OLD_RTSP_OPTIONS
108 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
113 static const AVOption sdp_options[] = {
114 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
115 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
116 { "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" },
117 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
122 static const AVOption rtp_options[] = {
123 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
129 static AVDictionary *map_to_opts(RTSPState *rt)
131 AVDictionary *opts = NULL;
134 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
135 av_dict_set(&opts, "buffer_size", buf, 0);
136 snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
137 av_dict_set(&opts, "pkt_size", buf, 0);
142 static void get_word_until_chars(char *buf, int buf_size,
143 const char *sep, const char **pp)
149 p += strspn(p, SPACE_CHARS);
151 while (!strchr(sep, *p) && *p != '\0') {
152 if ((q - buf) < buf_size - 1)
161 static void get_word_sep(char *buf, int buf_size, const char *sep,
164 if (**pp == '/') (*pp)++;
165 get_word_until_chars(buf, buf_size, sep, pp);
168 static void get_word(char *buf, int buf_size, const char **pp)
170 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
173 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
175 * Used for seeking in the rtp stream.
177 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
181 p += strspn(p, SPACE_CHARS);
182 if (!av_stristart(p, "npt=", &p))
185 *start = AV_NOPTS_VALUE;
186 *end = AV_NOPTS_VALUE;
188 get_word_sep(buf, sizeof(buf), "-", &p);
189 if (av_parse_time(start, buf, 1) < 0)
193 get_word_sep(buf, sizeof(buf), "-", &p);
194 if (av_parse_time(end, buf, 1) < 0)
195 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
199 static int get_sockaddr(AVFormatContext *s,
200 const char *buf, struct sockaddr_storage *sock)
202 struct addrinfo hints = { 0 }, *ai = NULL;
205 hints.ai_flags = AI_NUMERICHOST;
206 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
207 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
212 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
218 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
219 RTSPStream *rtsp_st, AVStream *st)
221 AVCodecParameters *par = st ? st->codecpar : NULL;
225 par->codec_id = handler->codec_id;
226 rtsp_st->dynamic_handler = handler;
228 st->need_parsing = handler->need_parsing;
229 if (handler->priv_data_size) {
230 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
231 if (!rtsp_st->dynamic_protocol_context)
232 rtsp_st->dynamic_handler = NULL;
236 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
239 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
240 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
241 rtsp_st->dynamic_protocol_context);
243 if (rtsp_st->dynamic_protocol_context) {
244 if (rtsp_st->dynamic_handler->close)
245 rtsp_st->dynamic_handler->close(
246 rtsp_st->dynamic_protocol_context);
247 av_free(rtsp_st->dynamic_protocol_context);
249 rtsp_st->dynamic_protocol_context = NULL;
250 rtsp_st->dynamic_handler = NULL;
255 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
256 static int sdp_parse_rtpmap(AVFormatContext *s,
257 AVStream *st, RTSPStream *rtsp_st,
258 int payload_type, const char *p)
260 AVCodecParameters *par = st->codecpar;
263 const AVCodecDescriptor *desc;
266 /* See if we can handle this kind of payload.
267 * The space should normally not be there but some Real streams or
268 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
269 * have a trailing space. */
270 get_word_sep(buf, sizeof(buf), "/ ", &p);
271 if (payload_type < RTP_PT_PRIVATE) {
272 /* We are in a standard case
273 * (from http://www.iana.org/assignments/rtp-parameters). */
274 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
277 if (par->codec_id == AV_CODEC_ID_NONE) {
278 const RTPDynamicProtocolHandler *handler =
279 ff_rtp_handler_find_by_name(buf, par->codec_type);
280 init_rtp_handler(handler, rtsp_st, st);
281 /* If no dynamic handler was found, check with the list of standard
282 * allocated types, if such a stream for some reason happens to
283 * use a private payload type. This isn't handled in rtpdec.c, since
284 * the format name from the rtpmap line never is passed into rtpdec. */
285 if (!rtsp_st->dynamic_handler)
286 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
289 desc = avcodec_descriptor_get(par->codec_id);
290 if (desc && desc->name)
295 get_word_sep(buf, sizeof(buf), "/", &p);
297 switch (par->codec_type) {
298 case AVMEDIA_TYPE_AUDIO:
299 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
300 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
301 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
303 par->sample_rate = i;
304 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
305 get_word_sep(buf, sizeof(buf), "/", &p);
310 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
312 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
315 case AVMEDIA_TYPE_VIDEO:
316 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
318 avpriv_set_pts_info(st, 32, 1, i);
323 finalize_rtp_handler_init(s, rtsp_st, st);
327 /* parse the attribute line from the fmtp a line of an sdp response. This
328 * is broken out as a function because it is used in rtp_h264.c, which is
330 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
331 char *value, int value_size)
333 *p += strspn(*p, SPACE_CHARS);
335 get_word_sep(attr, attr_size, "=", p);
338 get_word_sep(value, value_size, ";", p);
346 typedef struct SDPParseState {
348 struct sockaddr_storage default_ip;
350 int skip_media; ///< set if an unknown m= line occurs
351 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
352 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
353 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
354 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
357 char delayed_fmtp[2048];
360 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
361 struct RTSPSource ***dest, int *dest_count)
363 RTSPSource *rtsp_src, *rtsp_src2;
365 for (i = 0; i < count; i++) {
367 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
370 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
371 dynarray_add(dest, dest_count, rtsp_src2);
375 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
376 int payload_type, const char *line)
380 for (i = 0; i < rt->nb_rtsp_streams; i++) {
381 RTSPStream *rtsp_st = rt->rtsp_streams[i];
382 if (rtsp_st->sdp_payload_type == payload_type &&
383 rtsp_st->dynamic_handler &&
384 rtsp_st->dynamic_handler->parse_sdp_a_line) {
385 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
386 rtsp_st->dynamic_protocol_context, line);
391 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
392 int letter, const char *buf)
394 RTSPState *rt = s->priv_data;
395 char buf1[64], st_type[64];
397 enum AVMediaType codec_type;
401 RTSPSource *rtsp_src;
402 struct sockaddr_storage sdp_ip;
405 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
408 if (s1->skip_media && letter != 'm')
412 get_word(buf1, sizeof(buf1), &p);
413 if (strcmp(buf1, "IN") != 0)
415 get_word(buf1, sizeof(buf1), &p);
416 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
418 get_word_sep(buf1, sizeof(buf1), "/", &p);
419 if (get_sockaddr(s, buf1, &sdp_ip))
424 get_word_sep(buf1, sizeof(buf1), "/", &p);
427 if (s->nb_streams == 0) {
428 s1->default_ip = sdp_ip;
429 s1->default_ttl = ttl;
431 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
432 rtsp_st->sdp_ip = sdp_ip;
433 rtsp_st->sdp_ttl = ttl;
437 av_dict_set(&s->metadata, "title", p, 0);
440 if (s->nb_streams == 0) {
441 av_dict_set(&s->metadata, "comment", p, 0);
450 codec_type = AVMEDIA_TYPE_UNKNOWN;
451 get_word(st_type, sizeof(st_type), &p);
452 if (!strcmp(st_type, "audio")) {
453 codec_type = AVMEDIA_TYPE_AUDIO;
454 } else if (!strcmp(st_type, "video")) {
455 codec_type = AVMEDIA_TYPE_VIDEO;
456 } else if (!strcmp(st_type, "application")) {
457 codec_type = AVMEDIA_TYPE_DATA;
458 } else if (!strcmp(st_type, "text")) {
459 codec_type = AVMEDIA_TYPE_SUBTITLE;
461 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
462 !(rt->media_type_mask & (1 << codec_type)) ||
463 rt->nb_rtsp_streams >= s->max_streams
468 rtsp_st = av_mallocz(sizeof(RTSPStream));
471 rtsp_st->stream_index = -1;
472 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
474 rtsp_st->sdp_ip = s1->default_ip;
475 rtsp_st->sdp_ttl = s1->default_ttl;
477 copy_default_source_addrs(s1->default_include_source_addrs,
478 s1->nb_default_include_source_addrs,
479 &rtsp_st->include_source_addrs,
480 &rtsp_st->nb_include_source_addrs);
481 copy_default_source_addrs(s1->default_exclude_source_addrs,
482 s1->nb_default_exclude_source_addrs,
483 &rtsp_st->exclude_source_addrs,
484 &rtsp_st->nb_exclude_source_addrs);
486 get_word(buf1, sizeof(buf1), &p); /* port */
487 rtsp_st->sdp_port = atoi(buf1);
489 get_word(buf1, sizeof(buf1), &p); /* protocol */
490 if (!strcmp(buf1, "udp"))
491 rt->transport = RTSP_TRANSPORT_RAW;
492 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
493 rtsp_st->feedback = 1;
495 /* XXX: handle list of formats */
496 get_word(buf1, sizeof(buf1), &p); /* format list */
497 rtsp_st->sdp_payload_type = atoi(buf1);
499 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
500 /* no corresponding stream */
501 if (rt->transport == RTSP_TRANSPORT_RAW) {
502 if (CONFIG_RTPDEC && !rt->ts)
503 rt->ts = avpriv_mpegts_parse_open(s);
505 const RTPDynamicProtocolHandler *handler;
506 handler = ff_rtp_handler_find_by_id(
507 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
508 init_rtp_handler(handler, rtsp_st, NULL);
509 finalize_rtp_handler_init(s, rtsp_st, NULL);
511 } else if (rt->server_type == RTSP_SERVER_WMS &&
512 codec_type == AVMEDIA_TYPE_DATA) {
513 /* RTX stream, a stream that carries all the other actual
514 * audio/video streams. Don't expose this to the callers. */
516 st = avformat_new_stream(s, NULL);
519 st->id = rt->nb_rtsp_streams - 1;
520 rtsp_st->stream_index = st->index;
521 st->codecpar->codec_type = codec_type;
522 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
523 const RTPDynamicProtocolHandler *handler;
524 /* if standard payload type, we can find the codec right now */
525 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
526 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
527 st->codecpar->sample_rate > 0)
528 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
529 /* Even static payload types may need a custom depacketizer */
530 handler = ff_rtp_handler_find_by_id(
531 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
532 init_rtp_handler(handler, rtsp_st, st);
533 finalize_rtp_handler_init(s, rtsp_st, st);
535 if (rt->default_lang[0])
536 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
538 /* put a default control url */
539 av_strlcpy(rtsp_st->control_url, rt->control_uri,
540 sizeof(rtsp_st->control_url));
543 if (av_strstart(p, "control:", &p)) {
544 if (s->nb_streams == 0) {
545 if (!strncmp(p, "rtsp://", 7))
546 av_strlcpy(rt->control_uri, p,
547 sizeof(rt->control_uri));
550 /* get the control url */
551 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
553 /* XXX: may need to add full url resolution */
554 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
556 if (proto[0] == '\0') {
557 /* relative control URL */
558 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
559 av_strlcat(rtsp_st->control_url, "/",
560 sizeof(rtsp_st->control_url));
561 av_strlcat(rtsp_st->control_url, p,
562 sizeof(rtsp_st->control_url));
564 av_strlcpy(rtsp_st->control_url, p,
565 sizeof(rtsp_st->control_url));
567 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
568 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
569 get_word(buf1, sizeof(buf1), &p);
570 payload_type = atoi(buf1);
571 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
572 if (rtsp_st->stream_index >= 0) {
573 st = s->streams[rtsp_st->stream_index];
574 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
578 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
580 } else if (av_strstart(p, "fmtp:", &p) ||
581 av_strstart(p, "framesize:", &p)) {
582 // let dynamic protocol handlers have a stab at the line.
583 get_word(buf1, sizeof(buf1), &p);
584 payload_type = atoi(buf1);
585 if (s1->seen_rtpmap) {
586 parse_fmtp(s, rt, payload_type, buf);
589 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
591 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
592 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593 get_word(buf1, sizeof(buf1), &p);
594 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
595 } else if (av_strstart(p, "range:", &p)) {
598 // this is so that seeking on a streamed file can work.
599 rtsp_parse_range_npt(p, &start, &end);
600 s->start_time = start;
601 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
602 s->duration = (end == AV_NOPTS_VALUE) ?
603 AV_NOPTS_VALUE : end - start;
604 } else if (av_strstart(p, "lang:", &p)) {
605 if (s->nb_streams > 0) {
606 get_word(buf1, sizeof(buf1), &p);
607 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
608 if (rtsp_st->stream_index >= 0) {
609 st = s->streams[rtsp_st->stream_index];
610 av_dict_set(&st->metadata, "language", buf1, 0);
613 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
614 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
616 rt->transport = RTSP_TRANSPORT_RDT;
617 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
619 st = s->streams[s->nb_streams - 1];
620 st->codecpar->sample_rate = atoi(p);
621 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
623 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
624 get_word(buf1, sizeof(buf1), &p); // ignore tag
625 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
626 p += strspn(p, SPACE_CHARS);
627 if (av_strstart(p, "inline:", &p))
628 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
629 } else if (av_strstart(p, "source-filter:", &p)) {
631 get_word(buf1, sizeof(buf1), &p);
632 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
634 exclude = !strcmp(buf1, "excl");
636 get_word(buf1, sizeof(buf1), &p);
637 if (strcmp(buf1, "IN") != 0)
639 get_word(buf1, sizeof(buf1), &p);
640 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
642 // not checking that the destination address actually matches or is wildcard
643 get_word(buf1, sizeof(buf1), &p);
646 rtsp_src = av_mallocz(sizeof(*rtsp_src));
649 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
651 if (s->nb_streams == 0) {
652 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
654 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
655 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
658 if (s->nb_streams == 0) {
659 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
661 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
662 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
667 if (rt->server_type == RTSP_SERVER_WMS)
668 ff_wms_parse_sdp_a_line(s, p);
669 if (s->nb_streams > 0) {
670 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
672 if (rt->server_type == RTSP_SERVER_REAL)
673 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
675 if (rtsp_st->dynamic_handler &&
676 rtsp_st->dynamic_handler->parse_sdp_a_line)
677 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
678 rtsp_st->stream_index,
679 rtsp_st->dynamic_protocol_context, buf);
686 int ff_sdp_parse(AVFormatContext *s, const char *content)
690 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
691 * contain long SDP lines containing complete ASF Headers (several
692 * kB) or arrays of MDPR (RM stream descriptor) headers plus
693 * "rulebooks" describing their properties. Therefore, the SDP line
696 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
697 * in rtpdec_xiph.c. */
699 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
703 p += strspn(p, SPACE_CHARS);
711 /* get the content */
713 while (*p != '\n' && *p != '\r' && *p != '\0') {
714 if ((q - buf) < sizeof(buf) - 1)
719 sdp_parse_line(s, s1, letter, buf);
721 while (*p != '\n' && *p != '\0')
727 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
728 av_freep(&s1->default_include_source_addrs[i]);
729 av_freep(&s1->default_include_source_addrs);
730 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
731 av_freep(&s1->default_exclude_source_addrs[i]);
732 av_freep(&s1->default_exclude_source_addrs);
736 #endif /* CONFIG_RTPDEC */
738 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
740 RTSPState *rt = s->priv_data;
743 for (i = 0; i < rt->nb_rtsp_streams; i++) {
744 RTSPStream *rtsp_st = rt->rtsp_streams[i];
747 if (rtsp_st->transport_priv) {
749 AVFormatContext *rtpctx = rtsp_st->transport_priv;
750 av_write_trailer(rtpctx);
751 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
752 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
753 ff_rtsp_tcp_write_packet(s, rtsp_st);
754 ffio_free_dyn_buf(&rtpctx->pb);
756 avio_closep(&rtpctx->pb);
758 avformat_free_context(rtpctx);
759 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
760 ff_rdt_parse_close(rtsp_st->transport_priv);
761 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
762 ff_rtp_parse_close(rtsp_st->transport_priv);
764 rtsp_st->transport_priv = NULL;
765 if (rtsp_st->rtp_handle)
766 ffurl_close(rtsp_st->rtp_handle);
767 rtsp_st->rtp_handle = NULL;
771 /* close and free RTSP streams */
772 void ff_rtsp_close_streams(AVFormatContext *s)
774 RTSPState *rt = s->priv_data;
778 ff_rtsp_undo_setup(s, 0);
779 for (i = 0; i < rt->nb_rtsp_streams; i++) {
780 rtsp_st = rt->rtsp_streams[i];
782 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
783 if (rtsp_st->dynamic_handler->close)
784 rtsp_st->dynamic_handler->close(
785 rtsp_st->dynamic_protocol_context);
786 av_free(rtsp_st->dynamic_protocol_context);
788 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
789 av_freep(&rtsp_st->include_source_addrs[j]);
790 av_freep(&rtsp_st->include_source_addrs);
791 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
792 av_freep(&rtsp_st->exclude_source_addrs[j]);
793 av_freep(&rtsp_st->exclude_source_addrs);
798 av_freep(&rt->rtsp_streams);
800 avformat_close_input(&rt->asf_ctx);
802 if (CONFIG_RTPDEC && rt->ts)
803 avpriv_mpegts_parse_close(rt->ts);
805 av_freep(&rt->recvbuf);
808 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
810 RTSPState *rt = s->priv_data;
812 int reordering_queue_size = rt->reordering_queue_size;
813 if (reordering_queue_size < 0) {
814 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
815 reordering_queue_size = 0;
817 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
820 /* open the RTP context */
821 if (rtsp_st->stream_index >= 0)
822 st = s->streams[rtsp_st->stream_index];
824 s->ctx_flags |= AVFMTCTX_NOHEADER;
826 if (CONFIG_RTSP_MUXER && s->oformat && st) {
827 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
828 s, st, rtsp_st->rtp_handle,
829 RTSP_TCP_MAX_PACKET_SIZE,
830 rtsp_st->stream_index);
831 /* Ownership of rtp_handle is passed to the rtp mux context */
832 rtsp_st->rtp_handle = NULL;
835 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
836 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
837 return 0; // Don't need to open any parser here
838 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
839 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
840 rtsp_st->dynamic_protocol_context,
841 rtsp_st->dynamic_handler);
842 else if (CONFIG_RTPDEC)
843 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
844 rtsp_st->sdp_payload_type,
845 reordering_queue_size);
847 if (!rtsp_st->transport_priv) {
848 return AVERROR(ENOMEM);
849 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
851 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
852 rtpctx->ssrc = rtsp_st->ssrc;
853 if (rtsp_st->dynamic_handler) {
854 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
855 rtsp_st->dynamic_protocol_context,
856 rtsp_st->dynamic_handler);
858 if (rtsp_st->crypto_suite[0])
859 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
860 rtsp_st->crypto_suite,
861 rtsp_st->crypto_params);
867 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
868 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
875 q += strspn(q, SPACE_CHARS);
876 v = strtol(q, &p, 10);
880 v = strtol(p, &p, 10);
889 /* XXX: only one transport specification is parsed */
890 static void rtsp_parse_transport(AVFormatContext *s,
891 RTSPMessageHeader *reply, const char *p)
893 char transport_protocol[16];
895 char lower_transport[16];
897 RTSPTransportField *th;
900 reply->nb_transports = 0;
903 p += strspn(p, SPACE_CHARS);
907 th = &reply->transports[reply->nb_transports];
909 get_word_sep(transport_protocol, sizeof(transport_protocol),
911 if (!av_strcasecmp (transport_protocol, "rtp")) {
912 get_word_sep(profile, sizeof(profile), "/;,", &p);
913 lower_transport[0] = '\0';
914 /* rtp/avp/<protocol> */
916 get_word_sep(lower_transport, sizeof(lower_transport),
919 th->transport = RTSP_TRANSPORT_RTP;
920 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
921 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
922 /* x-pn-tng/<protocol> */
923 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
925 th->transport = RTSP_TRANSPORT_RDT;
926 } else if (!av_strcasecmp(transport_protocol, "raw")) {
927 get_word_sep(profile, sizeof(profile), "/;,", &p);
928 lower_transport[0] = '\0';
929 /* raw/raw/<protocol> */
931 get_word_sep(lower_transport, sizeof(lower_transport),
934 th->transport = RTSP_TRANSPORT_RAW;
936 if (!av_strcasecmp(lower_transport, "TCP"))
937 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
939 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
943 /* get each parameter */
944 while (*p != '\0' && *p != ',') {
945 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
946 if (!strcmp(parameter, "port")) {
949 rtsp_parse_range(&th->port_min, &th->port_max, &p);
951 } else if (!strcmp(parameter, "client_port")) {
954 rtsp_parse_range(&th->client_port_min,
955 &th->client_port_max, &p);
957 } else if (!strcmp(parameter, "server_port")) {
960 rtsp_parse_range(&th->server_port_min,
961 &th->server_port_max, &p);
963 } else if (!strcmp(parameter, "interleaved")) {
966 rtsp_parse_range(&th->interleaved_min,
967 &th->interleaved_max, &p);
969 } else if (!strcmp(parameter, "multicast")) {
970 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
971 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
972 } else if (!strcmp(parameter, "ttl")) {
976 th->ttl = strtol(p, &end, 10);
979 } else if (!strcmp(parameter, "destination")) {
982 get_word_sep(buf, sizeof(buf), ";,", &p);
983 get_sockaddr(s, buf, &th->destination);
985 } else if (!strcmp(parameter, "source")) {
988 get_word_sep(buf, sizeof(buf), ";,", &p);
989 av_strlcpy(th->source, buf, sizeof(th->source));
991 } else if (!strcmp(parameter, "mode")) {
994 get_word_sep(buf, sizeof(buf), ";, ", &p);
995 if (!strcmp(buf, "record") ||
996 !strcmp(buf, "receive"))
1001 while (*p != ';' && *p != '\0' && *p != ',')
1009 reply->nb_transports++;
1010 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1015 static void handle_rtp_info(RTSPState *rt, const char *url,
1016 uint32_t seq, uint32_t rtptime)
1019 if (!rtptime || !url[0])
1021 if (rt->transport != RTSP_TRANSPORT_RTP)
1023 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1024 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1025 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1028 if (!strcmp(rtsp_st->control_url, url)) {
1029 rtpctx->base_timestamp = rtptime;
1035 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1038 char key[20], value[1024], url[1024] = "";
1039 uint32_t seq = 0, rtptime = 0;
1042 p += strspn(p, SPACE_CHARS);
1045 get_word_sep(key, sizeof(key), "=", &p);
1049 get_word_sep(value, sizeof(value), ";, ", &p);
1051 if (!strcmp(key, "url"))
1052 av_strlcpy(url, value, sizeof(url));
1053 else if (!strcmp(key, "seq"))
1054 seq = strtoul(value, NULL, 10);
1055 else if (!strcmp(key, "rtptime"))
1056 rtptime = strtoul(value, NULL, 10);
1058 handle_rtp_info(rt, url, seq, rtptime);
1067 handle_rtp_info(rt, url, seq, rtptime);
1070 void ff_rtsp_parse_line(AVFormatContext *s,
1071 RTSPMessageHeader *reply, const char *buf,
1072 RTSPState *rt, const char *method)
1076 /* NOTE: we do case independent match for broken servers */
1078 if (av_stristart(p, "Session:", &p)) {
1080 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1081 if (av_stristart(p, ";timeout=", &p) &&
1082 (t = strtol(p, NULL, 10)) > 0) {
1085 } else if (av_stristart(p, "Content-Length:", &p)) {
1086 reply->content_length = strtol(p, NULL, 10);
1087 } else if (av_stristart(p, "Transport:", &p)) {
1088 rtsp_parse_transport(s, reply, p);
1089 } else if (av_stristart(p, "CSeq:", &p)) {
1090 reply->seq = strtol(p, NULL, 10);
1091 } else if (av_stristart(p, "Range:", &p)) {
1092 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1093 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1094 p += strspn(p, SPACE_CHARS);
1095 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1096 } else if (av_stristart(p, "Server:", &p)) {
1097 p += strspn(p, SPACE_CHARS);
1098 av_strlcpy(reply->server, p, sizeof(reply->server));
1099 } else if (av_stristart(p, "Notice:", &p) ||
1100 av_stristart(p, "X-Notice:", &p)) {
1101 reply->notice = strtol(p, NULL, 10);
1102 } else if (av_stristart(p, "Location:", &p)) {
1103 p += strspn(p, SPACE_CHARS);
1104 av_strlcpy(reply->location, p , sizeof(reply->location));
1105 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1106 p += strspn(p, SPACE_CHARS);
1107 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1108 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1109 p += strspn(p, SPACE_CHARS);
1110 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1111 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1112 p += strspn(p, SPACE_CHARS);
1113 if (method && !strcmp(method, "DESCRIBE"))
1114 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1115 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1116 p += strspn(p, SPACE_CHARS);
1117 if (method && !strcmp(method, "PLAY"))
1118 rtsp_parse_rtp_info(rt, p);
1119 } else if (av_stristart(p, "Public:", &p) && rt) {
1120 if (strstr(p, "GET_PARAMETER") &&
1121 method && !strcmp(method, "OPTIONS"))
1122 rt->get_parameter_supported = 1;
1123 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1124 p += strspn(p, SPACE_CHARS);
1125 rt->accept_dynamic_rate = atoi(p);
1126 } else if (av_stristart(p, "Content-Type:", &p)) {
1127 p += strspn(p, SPACE_CHARS);
1128 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1132 /* skip a RTP/TCP interleaved packet */
1133 void ff_rtsp_skip_packet(AVFormatContext *s)
1135 RTSPState *rt = s->priv_data;
1139 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1142 len = AV_RB16(buf + 1);
1144 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1149 if (len1 > sizeof(buf))
1151 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1158 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1159 unsigned char **content_ptr,
1160 int return_on_interleaved_data, const char *method)
1162 RTSPState *rt = s->priv_data;
1163 char buf[4096], buf1[1024], *q;
1166 int ret, content_length, line_count = 0, request = 0;
1167 unsigned char *content = NULL;
1173 memset(reply, 0, sizeof(*reply));
1175 /* parse reply (XXX: use buffers) */
1176 rt->last_reply[0] = '\0';
1180 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1181 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1186 if (ch == '$' && q == buf) {
1187 if (return_on_interleaved_data) {
1190 ff_rtsp_skip_packet(s);
1191 } else if (ch != '\r') {
1192 if ((q - buf) < sizeof(buf) - 1)
1198 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1200 /* test if last line */
1204 if (line_count == 0) {
1205 /* get reply code */
1206 get_word(buf1, sizeof(buf1), &p);
1207 if (!strncmp(buf1, "RTSP/", 5)) {
1208 get_word(buf1, sizeof(buf1), &p);
1209 reply->status_code = atoi(buf1);
1210 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1212 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1213 get_word(buf1, sizeof(buf1), &p); // object
1217 ff_rtsp_parse_line(s, reply, p, rt, method);
1218 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1219 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1224 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1225 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1227 content_length = reply->content_length;
1228 if (content_length > 0) {
1229 /* leave some room for a trailing '\0' (useful for simple parsing) */
1230 content = av_malloc(content_length + 1);
1232 return AVERROR(ENOMEM);
1233 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1234 content[content_length] = '\0';
1237 *content_ptr = content;
1243 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1244 const char* ptr = buf;
1246 if (!strcmp(reply->reason, "OPTIONS")) {
1247 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1249 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1250 if (reply->session_id[0])
1251 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1254 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1256 av_strlcat(buf, "\r\n", sizeof(buf));
1258 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1259 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1262 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1264 rt->last_cmd_time = av_gettime_relative();
1265 /* Even if the request from the server had data, it is not the data
1266 * that the caller wants or expects. The memory could also be leaked
1267 * if the actual following reply has content data. */
1269 av_freep(content_ptr);
1270 /* If method is set, this is called from ff_rtsp_send_cmd,
1271 * where a reply to exactly this request is awaited. For
1272 * callers from within packet receiving, we just want to
1273 * return to the caller and go back to receiving packets. */
1279 if (rt->seq != reply->seq) {
1280 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1281 rt->seq, reply->seq);
1285 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1286 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1287 reply->notice == 2306 /* Continuous Feed Terminated */) {
1288 rt->state = RTSP_STATE_IDLE;
1289 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1290 return AVERROR(EIO); /* data or server error */
1291 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1292 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1293 return AVERROR(EPERM);
1299 * Send a command to the RTSP server without waiting for the reply.
1301 * @param s RTSP (de)muxer context
1302 * @param method the method for the request
1303 * @param url the target url for the request
1304 * @param headers extra header lines to include in the request
1305 * @param send_content if non-null, the data to send as request body content
1306 * @param send_content_length the length of the send_content data, or 0 if
1307 * send_content is null
1309 * @return zero if success, nonzero otherwise
1311 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1312 const char *method, const char *url,
1313 const char *headers,
1314 const unsigned char *send_content,
1315 int send_content_length)
1317 RTSPState *rt = s->priv_data;
1318 char buf[4096], *out_buf;
1319 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1321 if (!rt->rtsp_hd_out)
1322 return AVERROR(ENOTCONN);
1324 /* Add in RTSP headers */
1327 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1329 av_strlcat(buf, headers, sizeof(buf));
1330 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1331 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1332 if (rt->session_id[0] != '\0' && (!headers ||
1333 !strstr(headers, "\nIf-Match:"))) {
1334 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1337 char *str = ff_http_auth_create_response(&rt->auth_state,
1338 rt->auth, url, method);
1340 av_strlcat(buf, str, sizeof(buf));
1343 if (send_content_length > 0 && send_content)
1344 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1345 av_strlcat(buf, "\r\n", sizeof(buf));
1347 /* base64 encode rtsp if tunneling */
1348 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1349 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1350 out_buf = base64buf;
1353 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1355 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1356 if (send_content_length > 0 && send_content) {
1357 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1358 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1359 return AVERROR_PATCHWELCOME;
1361 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1363 rt->last_cmd_time = av_gettime_relative();
1368 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1369 const char *url, const char *headers)
1371 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1374 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1375 const char *headers, RTSPMessageHeader *reply,
1376 unsigned char **content_ptr)
1378 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1379 content_ptr, NULL, 0);
1382 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1383 const char *method, const char *url,
1385 RTSPMessageHeader *reply,
1386 unsigned char **content_ptr,
1387 const unsigned char *send_content,
1388 int send_content_length)
1390 RTSPState *rt = s->priv_data;
1391 HTTPAuthType cur_auth_type;
1392 int ret, attempts = 0;
1395 cur_auth_type = rt->auth_state.auth_type;
1396 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1398 send_content_length)))
1401 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1405 if (reply->status_code == 401 &&
1406 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1407 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1410 if (reply->status_code > 400){
1411 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1415 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1421 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1422 int lower_transport, const char *real_challenge)
1424 RTSPState *rt = s->priv_data;
1425 int rtx = 0, j, i, err, interleave = 0, port_off;
1426 RTSPStream *rtsp_st;
1427 RTSPMessageHeader reply1, *reply = &reply1;
1429 const char *trans_pref;
1431 if (rt->transport == RTSP_TRANSPORT_RDT)
1432 trans_pref = "x-pn-tng";
1433 else if (rt->transport == RTSP_TRANSPORT_RAW)
1434 trans_pref = "RAW/RAW";
1436 trans_pref = "RTP/AVP";
1438 /* default timeout: 1 minute */
1441 /* Choose a random starting offset within the first half of the
1442 * port range, to allow for a number of ports to try even if the offset
1443 * happens to be at the end of the random range. */
1444 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1445 /* even random offset */
1446 port_off -= port_off & 0x01;
1448 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1449 char transport[2048];
1452 * WMS serves all UDP data over a single connection, the RTX, which
1453 * isn't necessarily the first in the SDP but has to be the first
1454 * to be set up, else the second/third SETUP will fail with a 461.
1456 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1457 rt->server_type == RTSP_SERVER_WMS) {
1460 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1461 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1463 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1467 if (rtx == rt->nb_rtsp_streams)
1468 return -1; /* no RTX found */
1469 rtsp_st = rt->rtsp_streams[rtx];
1471 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1473 rtsp_st = rt->rtsp_streams[i];
1476 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1479 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1480 port = reply->transports[0].client_port_min;
1484 /* first try in specified port range */
1485 while (j <= rt->rtp_port_max) {
1486 AVDictionary *opts = map_to_opts(rt);
1488 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1489 "?localport=%d", j);
1490 /* we will use two ports per rtp stream (rtp and rtcp) */
1492 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1493 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1495 av_dict_free(&opts);
1500 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1505 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1507 snprintf(transport, sizeof(transport) - 1,
1508 "%s/UDP;", trans_pref);
1509 if (rt->server_type != RTSP_SERVER_REAL)
1510 av_strlcat(transport, "unicast;", sizeof(transport));
1511 av_strlcatf(transport, sizeof(transport),
1512 "client_port=%d", port);
1513 if (rt->transport == RTSP_TRANSPORT_RTP &&
1514 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1515 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1519 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1520 /* For WMS streams, the application streams are only used for
1521 * UDP. When trying to set it up for TCP streams, the server
1522 * will return an error. Therefore, we skip those streams. */
1523 if (rt->server_type == RTSP_SERVER_WMS &&
1524 (rtsp_st->stream_index < 0 ||
1525 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1528 snprintf(transport, sizeof(transport) - 1,
1529 "%s/TCP;", trans_pref);
1530 if (rt->transport != RTSP_TRANSPORT_RDT)
1531 av_strlcat(transport, "unicast;", sizeof(transport));
1532 av_strlcatf(transport, sizeof(transport),
1533 "interleaved=%d-%d",
1534 interleave, interleave + 1);
1538 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1539 snprintf(transport, sizeof(transport) - 1,
1540 "%s/UDP;multicast", trans_pref);
1543 av_strlcat(transport, ";mode=record", sizeof(transport));
1544 } else if (rt->server_type == RTSP_SERVER_REAL ||
1545 rt->server_type == RTSP_SERVER_WMS)
1546 av_strlcat(transport, ";mode=play", sizeof(transport));
1547 snprintf(cmd, sizeof(cmd),
1548 "Transport: %s\r\n",
1550 if (rt->accept_dynamic_rate)
1551 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1552 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1553 char real_res[41], real_csum[9];
1554 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1556 av_strlcatf(cmd, sizeof(cmd),
1558 "RealChallenge2: %s, sd=%s\r\n",
1559 rt->session_id, real_res, real_csum);
1561 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1562 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1565 } else if (reply->status_code != RTSP_STATUS_OK ||
1566 reply->nb_transports != 1) {
1567 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1571 /* XXX: same protocol for all streams is required */
1573 if (reply->transports[0].lower_transport != rt->lower_transport ||
1574 reply->transports[0].transport != rt->transport) {
1575 err = AVERROR_INVALIDDATA;
1579 rt->lower_transport = reply->transports[0].lower_transport;
1580 rt->transport = reply->transports[0].transport;
1583 /* Fail if the server responded with another lower transport mode
1584 * than what we requested. */
1585 if (reply->transports[0].lower_transport != lower_transport) {
1586 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1587 err = AVERROR_INVALIDDATA;
1591 switch(reply->transports[0].lower_transport) {
1592 case RTSP_LOWER_TRANSPORT_TCP:
1593 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1594 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1597 case RTSP_LOWER_TRANSPORT_UDP: {
1598 char url[1024], options[30] = "";
1599 const char *peer = host;
1601 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1602 av_strlcpy(options, "?connect=1", sizeof(options));
1603 /* Use source address if specified */
1604 if (reply->transports[0].source[0])
1605 peer = reply->transports[0].source;
1606 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1607 reply->transports[0].server_port_min, "%s", options);
1608 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1609 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1610 err = AVERROR_INVALIDDATA;
1615 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1616 char url[1024], namebuf[50], optbuf[20] = "";
1617 struct sockaddr_storage addr;
1620 if (reply->transports[0].destination.ss_family) {
1621 addr = reply->transports[0].destination;
1622 port = reply->transports[0].port_min;
1623 ttl = reply->transports[0].ttl;
1625 addr = rtsp_st->sdp_ip;
1626 port = rtsp_st->sdp_port;
1627 ttl = rtsp_st->sdp_ttl;
1630 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1631 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1632 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1633 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1634 port, "%s", optbuf);
1635 if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1636 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL) < 0) {
1637 err = AVERROR_INVALIDDATA;
1644 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1648 if (rt->nb_rtsp_streams && reply->timeout > 0)
1649 rt->timeout = reply->timeout;
1651 if (rt->server_type == RTSP_SERVER_REAL)
1652 rt->need_subscription = 1;
1657 ff_rtsp_undo_setup(s, 0);
1661 void ff_rtsp_close_connections(AVFormatContext *s)
1663 RTSPState *rt = s->priv_data;
1664 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1665 ffurl_close(rt->rtsp_hd);
1666 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1669 int ff_rtsp_connect(AVFormatContext *s)
1671 RTSPState *rt = s->priv_data;
1672 char proto[128], host[1024], path[1024];
1673 char tcpname[1024], cmd[2048], auth[128];
1674 const char *lower_rtsp_proto = "tcp";
1675 int port, err, tcp_fd;
1676 RTSPMessageHeader reply1, *reply = &reply1;
1677 int lower_transport_mask = 0;
1678 int default_port = RTSP_DEFAULT_PORT;
1679 int https_tunnel = 0;
1680 char real_challenge[64] = "";
1681 struct sockaddr_storage peer;
1682 socklen_t peer_len = sizeof(peer);
1684 if (rt->rtp_port_max < rt->rtp_port_min) {
1685 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1686 "than min port %d\n", rt->rtp_port_max,
1688 return AVERROR(EINVAL);
1691 if (!ff_network_init())
1692 return AVERROR(EIO);
1694 if (s->max_delay < 0) /* Not set by the caller */
1695 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1697 rt->control_transport = RTSP_MODE_PLAIN;
1698 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1699 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1700 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1701 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1702 rt->control_transport = RTSP_MODE_TUNNEL;
1704 /* Only pass through valid flags from here */
1705 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1708 memset(&reply1, 0, sizeof(reply1));
1709 /* extract hostname and port */
1710 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1711 host, sizeof(host), &port, path, sizeof(path), s->url);
1713 if (!strcmp(proto, "rtsps")) {
1714 lower_rtsp_proto = "tls";
1715 default_port = RTSPS_DEFAULT_PORT;
1716 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1720 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1723 port = default_port;
1725 lower_transport_mask = rt->lower_transport_mask;
1727 if (!lower_transport_mask)
1728 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1731 /* Only UDP or TCP - UDP multicast isn't supported. */
1732 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1733 (1 << RTSP_LOWER_TRANSPORT_TCP);
1734 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1735 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1736 "only UDP and TCP are supported for output.\n");
1737 err = AVERROR(EINVAL);
1742 /* Construct the URI used in request; this is similar to s->url,
1743 * but with authentication credentials removed and RTSP specific options
1745 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1746 host, port, "%s", path);
1748 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1749 /* set up initial handshake for tunneling */
1750 char httpname[1024];
1751 char sessioncookie[17];
1753 AVDictionary *options = NULL;
1755 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1757 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1758 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1759 av_get_random_seed(), av_get_random_seed());
1762 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1763 &s->interrupt_callback) < 0) {
1768 /* generate GET headers */
1769 snprintf(headers, sizeof(headers),
1770 "x-sessioncookie: %s\r\n"
1771 "Accept: application/x-rtsp-tunnelled\r\n"
1772 "Pragma: no-cache\r\n"
1773 "Cache-Control: no-cache\r\n",
1775 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1777 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1778 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1779 if (!rt->rtsp_hd->protocol_whitelist) {
1780 err = AVERROR(ENOMEM);
1785 /* complete the connection */
1786 if (ffurl_connect(rt->rtsp_hd, &options)) {
1787 av_dict_free(&options);
1793 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1794 &s->interrupt_callback) < 0 ) {
1799 /* generate POST headers */
1800 snprintf(headers, sizeof(headers),
1801 "x-sessioncookie: %s\r\n"
1802 "Content-Type: application/x-rtsp-tunnelled\r\n"
1803 "Pragma: no-cache\r\n"
1804 "Cache-Control: no-cache\r\n"
1805 "Content-Length: 32767\r\n"
1806 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1808 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1809 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1810 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1812 /* Initialize the authentication state for the POST session. The HTTP
1813 * protocol implementation doesn't properly handle multi-pass
1814 * authentication for POST requests, since it would require one of
1816 * - implementing Expect: 100-continue, which many HTTP servers
1817 * don't support anyway, even less the RTSP servers that do HTTP
1819 * - sending the whole POST data until getting a 401 reply specifying
1820 * what authentication method to use, then resending all that data
1821 * - waiting for potential 401 replies directly after sending the
1822 * POST header (waiting for some unspecified time)
1823 * Therefore, we copy the full auth state, which works for both basic
1824 * and digest. (For digest, we would have to synchronize the nonce
1825 * count variable between the two sessions, if we'd do more requests
1826 * with the original session, though.)
1828 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1830 /* complete the connection */
1831 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1832 av_dict_free(&options);
1836 av_dict_free(&options);
1839 /* open the tcp connection */
1840 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1842 "?timeout=%d", rt->stimeout);
1843 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1844 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1848 rt->rtsp_hd_out = rt->rtsp_hd;
1852 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1857 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1858 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1859 NULL, 0, NI_NUMERICHOST);
1862 /* request options supported by the server; this also detects server
1864 for (rt->server_type = RTSP_SERVER_RTP;;) {
1866 if (rt->server_type == RTSP_SERVER_REAL)
1869 * The following entries are required for proper
1870 * streaming from a Realmedia server. They are
1871 * interdependent in some way although we currently
1872 * don't quite understand how. Values were copied
1873 * from mplayer SVN r23589.
1874 * ClientChallenge is a 16-byte ID in hex
1875 * CompanyID is a 16-byte ID in base64
1877 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1878 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1879 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1880 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1882 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1883 if (reply->status_code != RTSP_STATUS_OK) {
1884 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1888 /* detect server type if not standard-compliant RTP */
1889 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1890 rt->server_type = RTSP_SERVER_REAL;
1892 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1893 rt->server_type = RTSP_SERVER_WMS;
1894 } else if (rt->server_type == RTSP_SERVER_REAL)
1895 strcpy(real_challenge, reply->real_challenge);
1899 if (CONFIG_RTSP_DEMUXER && s->iformat)
1900 err = ff_rtsp_setup_input_streams(s, reply);
1901 else if (CONFIG_RTSP_MUXER)
1902 err = ff_rtsp_setup_output_streams(s, host);
1909 int lower_transport = ff_log2_tab[lower_transport_mask &
1910 ~(lower_transport_mask - 1)];
1912 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1913 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1914 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1916 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1917 rt->server_type == RTSP_SERVER_REAL ?
1918 real_challenge : NULL);
1921 lower_transport_mask &= ~(1 << lower_transport);
1922 if (lower_transport_mask == 0 && err == 1) {
1923 err = AVERROR(EPROTONOSUPPORT);
1928 rt->lower_transport_mask = lower_transport_mask;
1929 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1930 rt->state = RTSP_STATE_IDLE;
1931 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1934 ff_rtsp_close_streams(s);
1935 ff_rtsp_close_connections(s);
1936 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1937 char *new_url = av_strdup(reply->location);
1939 err = AVERROR(ENOMEM);
1942 ff_format_set_url(s, new_url);
1943 rt->session_id[0] = '\0';
1944 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1953 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1956 static int parse_rtsp_message(AVFormatContext *s)
1958 RTSPState *rt = s->priv_data;
1961 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1962 if (rt->state == RTSP_STATE_STREAMING) {
1963 if (!ff_rtsp_parse_streaming_commands(s))
1966 av_log(s, AV_LOG_WARNING,
1967 "Unable to answer to TEARDOWN\n");
1971 RTSPMessageHeader reply;
1972 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1975 /* XXX: parse message */
1976 if (rt->state != RTSP_STATE_STREAMING)
1983 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1984 uint8_t *buf, int buf_size, int64_t wait_end)
1986 RTSPState *rt = s->priv_data;
1987 RTSPStream *rtsp_st;
1988 int n, i, ret, timeout_cnt = 0;
1989 struct pollfd *p = rt->p;
1990 int *fds = NULL, fdsnum, fdsidx;
1993 p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
1995 return AVERROR(ENOMEM);
1998 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1999 p[rt->max_p++].events = POLLIN;
2001 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2002 rtsp_st = rt->rtsp_streams[i];
2003 if (rtsp_st->rtp_handle) {
2004 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2006 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2010 av_log(s, AV_LOG_ERROR,
2011 "Number of fds %d not supported\n", fdsnum);
2012 return AVERROR_INVALIDDATA;
2014 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2015 p[rt->max_p].fd = fds[fdsidx];
2016 p[rt->max_p++].events = POLLIN;
2024 if (ff_check_interrupt(&s->interrupt_callback))
2025 return AVERROR_EXIT;
2026 if (wait_end && wait_end - av_gettime_relative() < 0)
2027 return AVERROR(EAGAIN);
2028 n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2030 int j = rt->rtsp_hd ? 1 : 0;
2032 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2033 rtsp_st = rt->rtsp_streams[i];
2034 if (rtsp_st->rtp_handle) {
2035 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2036 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2038 *prtsp_st = rtsp_st;
2045 #if CONFIG_RTSP_DEMUXER
2046 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2047 if ((ret = parse_rtsp_message(s)) < 0) {
2052 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2053 return AVERROR(ETIMEDOUT);
2054 } else if (n < 0 && errno != EINTR)
2055 return AVERROR(errno);
2059 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2060 const uint8_t *buf, int len)
2062 RTSPState *rt = s->priv_data;
2066 if (rt->nb_rtsp_streams == 1) {
2067 *rtsp_st = rt->rtsp_streams[0];
2070 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2071 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2073 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2077 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2078 *rtsp_st = rt->rtsp_streams[i];
2085 av_log(s, AV_LOG_WARNING,
2086 "Unable to pick stream for packet - SSRC not known for "
2088 return AVERROR(EAGAIN);
2091 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2092 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2093 *rtsp_st = rt->rtsp_streams[i];
2099 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2100 return AVERROR(EAGAIN);
2103 static int read_packet(AVFormatContext *s,
2104 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2107 RTSPState *rt = s->priv_data;
2110 switch(rt->lower_transport) {
2112 #if CONFIG_RTSP_DEMUXER
2113 case RTSP_LOWER_TRANSPORT_TCP:
2114 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2117 case RTSP_LOWER_TRANSPORT_UDP:
2118 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2119 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2120 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2121 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2123 case RTSP_LOWER_TRANSPORT_CUSTOM:
2124 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2125 wait_end && wait_end < av_gettime_relative())
2126 len = AVERROR(EAGAIN);
2128 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2129 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2130 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2131 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2141 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2143 RTSPState *rt = s->priv_data;
2145 RTSPStream *rtsp_st, *first_queue_st = NULL;
2146 int64_t wait_end = 0;
2148 if (rt->nb_byes == rt->nb_rtsp_streams)
2151 /* get next frames from the same RTP packet */
2152 if (rt->cur_transport_priv) {
2153 if (rt->transport == RTSP_TRANSPORT_RDT) {
2154 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2155 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2156 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2157 } else if (CONFIG_RTPDEC && rt->ts) {
2158 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2160 rt->recvbuf_pos += ret;
2161 ret = rt->recvbuf_pos < rt->recvbuf_len;
2166 rt->cur_transport_priv = NULL;
2168 } else if (ret == 1) {
2171 rt->cur_transport_priv = NULL;
2175 if (rt->transport == RTSP_TRANSPORT_RTP) {
2177 int64_t first_queue_time = 0;
2178 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2179 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2183 queue_time = ff_rtp_queued_packet_time(rtpctx);
2184 if (queue_time && (queue_time - first_queue_time < 0 ||
2185 !first_queue_time)) {
2186 first_queue_time = queue_time;
2187 first_queue_st = rt->rtsp_streams[i];
2190 if (first_queue_time) {
2191 wait_end = first_queue_time + s->max_delay;
2194 first_queue_st = NULL;
2198 /* read next RTP packet */
2200 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2202 return AVERROR(ENOMEM);
2205 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2206 if (len == AVERROR(EAGAIN) && first_queue_st &&
2207 rt->transport == RTSP_TRANSPORT_RTP) {
2208 av_log(s, AV_LOG_WARNING,
2209 "max delay reached. need to consume packet\n");
2210 rtsp_st = first_queue_st;
2211 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2217 if (rt->transport == RTSP_TRANSPORT_RDT) {
2218 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2219 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2220 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2221 if (rtsp_st->feedback) {
2222 AVIOContext *pb = NULL;
2223 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2225 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2228 /* Either bad packet, or a RTCP packet. Check if the
2229 * first_rtcp_ntp_time field was initialized. */
2230 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2231 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2232 /* first_rtcp_ntp_time has been initialized for this stream,
2233 * copy the same value to all other uninitialized streams,
2234 * in order to map their timestamp origin to the same ntp time
2237 AVStream *st = NULL;
2238 if (rtsp_st->stream_index >= 0)
2239 st = s->streams[rtsp_st->stream_index];
2240 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2241 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2242 AVStream *st2 = NULL;
2243 if (rt->rtsp_streams[i]->stream_index >= 0)
2244 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2245 if (rtpctx2 && st && st2 &&
2246 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2247 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2248 rtpctx2->rtcp_ts_offset = av_rescale_q(
2249 rtpctx->rtcp_ts_offset, st->time_base,
2253 // Make real NTP start time available in AVFormatContext
2254 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2255 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2257 s->start_time_realtime -=
2258 av_rescale (rtpctx->rtcp_ts_offset,
2259 (uint64_t) rtpctx->st->time_base.num * 1000000,
2260 rtpctx->st->time_base.den);
2264 if (ret == -RTCP_BYE) {
2267 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2268 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2270 if (rt->nb_byes == rt->nb_rtsp_streams)
2274 } else if (CONFIG_RTPDEC && rt->ts) {
2275 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2278 rt->recvbuf_len = len;
2279 rt->recvbuf_pos = ret;
2280 rt->cur_transport_priv = rt->ts;
2287 return AVERROR_INVALIDDATA;
2293 /* more packets may follow, so we save the RTP context */
2294 rt->cur_transport_priv = rtsp_st->transport_priv;
2298 #endif /* CONFIG_RTPDEC */
2300 #if CONFIG_SDP_DEMUXER
2301 static int sdp_probe(const AVProbeData *p1)
2303 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2305 /* we look for a line beginning "c=IN IP" */
2306 while (p < p_end && *p != '\0') {
2307 if (sizeof("c=IN IP") - 1 < p_end - p &&
2308 av_strstart(p, "c=IN IP", NULL))
2309 return AVPROBE_SCORE_EXTENSION;
2311 while (p < p_end - 1 && *p != '\n') p++;
2320 static void append_source_addrs(char *buf, int size, const char *name,
2321 int count, struct RTSPSource **addrs)
2326 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2327 for (i = 1; i < count; i++)
2328 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2331 static int sdp_read_header(AVFormatContext *s)
2333 RTSPState *rt = s->priv_data;
2334 RTSPStream *rtsp_st;
2339 if (!ff_network_init())
2340 return AVERROR(EIO);
2342 if (s->max_delay < 0) /* Not set by the caller */
2343 s->max_delay = DEFAULT_REORDERING_DELAY;
2344 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2345 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2347 /* read the whole sdp file */
2348 /* XXX: better loading */
2349 content = av_malloc(SDP_MAX_SIZE);
2351 return AVERROR(ENOMEM);
2352 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2355 return AVERROR_INVALIDDATA;
2357 content[size] ='\0';
2359 err = ff_sdp_parse(s, content);
2363 /* open each RTP stream */
2364 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2366 rtsp_st = rt->rtsp_streams[i];
2368 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2369 AVDictionary *opts = map_to_opts(rt);
2371 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2372 sizeof(rtsp_st->sdp_ip),
2373 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2375 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2377 av_dict_free(&opts);
2380 ff_url_join(url, sizeof(url), "rtp", NULL,
2381 namebuf, rtsp_st->sdp_port,
2382 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2383 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2384 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2385 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2387 append_source_addrs(url, sizeof(url), "sources",
2388 rtsp_st->nb_include_source_addrs,
2389 rtsp_st->include_source_addrs);
2390 append_source_addrs(url, sizeof(url), "block",
2391 rtsp_st->nb_exclude_source_addrs,
2392 rtsp_st->exclude_source_addrs);
2393 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2394 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2396 av_dict_free(&opts);
2399 err = AVERROR_INVALIDDATA;
2403 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2408 ff_rtsp_close_streams(s);
2413 static int sdp_read_close(AVFormatContext *s)
2415 ff_rtsp_close_streams(s);
2420 static const AVClass sdp_demuxer_class = {
2421 .class_name = "SDP demuxer",
2422 .item_name = av_default_item_name,
2423 .option = sdp_options,
2424 .version = LIBAVUTIL_VERSION_INT,
2427 AVInputFormat ff_sdp_demuxer = {
2429 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2430 .priv_data_size = sizeof(RTSPState),
2431 .read_probe = sdp_probe,
2432 .read_header = sdp_read_header,
2433 .read_packet = ff_rtsp_fetch_packet,
2434 .read_close = sdp_read_close,
2435 .priv_class = &sdp_demuxer_class,
2437 #endif /* CONFIG_SDP_DEMUXER */
2439 #if CONFIG_RTP_DEMUXER
2440 static int rtp_probe(const AVProbeData *p)
2442 if (av_strstart(p->filename, "rtp:", NULL))
2443 return AVPROBE_SCORE_MAX;
2447 static int rtp_read_header(AVFormatContext *s)
2449 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2450 char host[500], sdp[500];
2452 URLContext* in = NULL;
2454 AVCodecParameters *par = NULL;
2455 struct sockaddr_storage addr;
2457 socklen_t addrlen = sizeof(addr);
2458 RTSPState *rt = s->priv_data;
2460 if (!ff_network_init())
2461 return AVERROR(EIO);
2463 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2464 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2469 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2470 if (ret == AVERROR(EAGAIN))
2475 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2479 if ((recvbuf[0] & 0xc0) != 0x80) {
2480 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2485 if (RTP_PT_IS_RTCP(recvbuf[1]))
2488 payload_type = recvbuf[1] & 0x7f;
2491 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2495 par = avcodec_parameters_alloc();
2497 ret = AVERROR(ENOMEM);
2501 if (ff_rtp_get_codec_info(par, payload_type)) {
2502 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2503 "without an SDP file describing it\n",
2507 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2508 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2509 "properly you need an SDP file "
2513 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2516 snprintf(sdp, sizeof(sdp),
2517 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2518 addr.ss_family == AF_INET ? 4 : 6, host,
2519 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2520 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2521 port, payload_type);
2522 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2523 avcodec_parameters_free(&par);
2525 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2528 /* sdp_read_header initializes this again */
2531 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2533 ret = sdp_read_header(s);
2538 avcodec_parameters_free(&par);
2545 static const AVClass rtp_demuxer_class = {
2546 .class_name = "RTP demuxer",
2547 .item_name = av_default_item_name,
2548 .option = rtp_options,
2549 .version = LIBAVUTIL_VERSION_INT,
2552 AVInputFormat ff_rtp_demuxer = {
2554 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2555 .priv_data_size = sizeof(RTSPState),
2556 .read_probe = rtp_probe,
2557 .read_header = rtp_read_header,
2558 .read_packet = ff_rtsp_fetch_packet,
2559 .read_close = sdp_read_close,
2560 .flags = AVFMT_NOFILE,
2561 .priv_class = &rtp_demuxer_class,
2563 #endif /* CONFIG_RTP_DEMUXER */