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_INT, {.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 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92 { "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" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
99 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103 static const AVOption sdp_options[] = {
104 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
107 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
112 static const AVOption rtp_options[] = {
113 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
119 static AVDictionary *map_to_opts(RTSPState *rt)
121 AVDictionary *opts = NULL;
124 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125 av_dict_set(&opts, "buffer_size", buf, 0);
130 static void get_word_until_chars(char *buf, int buf_size,
131 const char *sep, const char **pp)
137 p += strspn(p, SPACE_CHARS);
139 while (!strchr(sep, *p) && *p != '\0') {
140 if ((q - buf) < buf_size - 1)
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
152 if (**pp == '/') (*pp)++;
153 get_word_until_chars(buf, buf_size, sep, pp);
156 static void get_word(char *buf, int buf_size, const char **pp)
158 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
163 * Used for seeking in the rtp stream.
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
169 p += strspn(p, SPACE_CHARS);
170 if (!av_stristart(p, "npt=", &p))
173 *start = AV_NOPTS_VALUE;
174 *end = AV_NOPTS_VALUE;
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 if (av_parse_time(start, buf, 1) < 0)
181 get_word_sep(buf, sizeof(buf), "-", &p);
182 av_parse_time(end, buf, 1);
186 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
188 struct addrinfo hints = { 0 }, *ai = NULL;
189 hints.ai_flags = AI_NUMERICHOST;
190 if (getaddrinfo(buf, NULL, &hints, &ai))
192 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
198 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
199 RTSPStream *rtsp_st, AVStream *st)
201 AVCodecContext *codec = st ? st->codec : NULL;
205 codec->codec_id = handler->codec_id;
206 rtsp_st->dynamic_handler = handler;
208 st->need_parsing = handler->need_parsing;
209 if (handler->priv_data_size) {
210 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
211 if (!rtsp_st->dynamic_protocol_context)
212 rtsp_st->dynamic_handler = NULL;
216 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
219 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
220 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
221 rtsp_st->dynamic_protocol_context);
223 if (rtsp_st->dynamic_protocol_context) {
224 if (rtsp_st->dynamic_handler->close)
225 rtsp_st->dynamic_handler->close(
226 rtsp_st->dynamic_protocol_context);
227 av_free(rtsp_st->dynamic_protocol_context);
229 rtsp_st->dynamic_protocol_context = NULL;
230 rtsp_st->dynamic_handler = NULL;
235 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
236 static int sdp_parse_rtpmap(AVFormatContext *s,
237 AVStream *st, RTSPStream *rtsp_st,
238 int payload_type, const char *p)
240 AVCodecContext *codec = st->codec;
246 /* See if we can handle this kind of payload.
247 * The space should normally not be there but some Real streams or
248 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
249 * have a trailing space. */
250 get_word_sep(buf, sizeof(buf), "/ ", &p);
251 if (payload_type < RTP_PT_PRIVATE) {
252 /* We are in a standard case
253 * (from http://www.iana.org/assignments/rtp-parameters). */
254 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
257 if (codec->codec_id == AV_CODEC_ID_NONE) {
258 RTPDynamicProtocolHandler *handler =
259 ff_rtp_handler_find_by_name(buf, codec->codec_type);
260 init_rtp_handler(handler, rtsp_st, st);
261 /* If no dynamic handler was found, check with the list of standard
262 * allocated types, if such a stream for some reason happens to
263 * use a private payload type. This isn't handled in rtpdec.c, since
264 * the format name from the rtpmap line never is passed into rtpdec. */
265 if (!rtsp_st->dynamic_handler)
266 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
269 c = avcodec_find_decoder(codec->codec_id);
275 get_word_sep(buf, sizeof(buf), "/", &p);
277 switch (codec->codec_type) {
278 case AVMEDIA_TYPE_AUDIO:
279 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
280 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
281 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
283 codec->sample_rate = i;
284 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
285 get_word_sep(buf, sizeof(buf), "/", &p);
290 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
292 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
295 case AVMEDIA_TYPE_VIDEO:
296 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
298 avpriv_set_pts_info(st, 32, 1, i);
303 finalize_rtp_handler_init(s, rtsp_st, st);
307 /* parse the attribute line from the fmtp a line of an sdp response. This
308 * is broken out as a function because it is used in rtp_h264.c, which is
310 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
311 char *value, int value_size)
313 *p += strspn(*p, SPACE_CHARS);
315 get_word_sep(attr, attr_size, "=", p);
318 get_word_sep(value, value_size, ";", p);
326 typedef struct SDPParseState {
328 struct sockaddr_storage default_ip;
330 int skip_media; ///< set if an unknown m= line occurs
331 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
332 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
333 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
334 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
337 char delayed_fmtp[2048];
340 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
341 struct RTSPSource ***dest, int *dest_count)
343 RTSPSource *rtsp_src, *rtsp_src2;
345 for (i = 0; i < count; i++) {
347 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
350 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
351 dynarray_add(dest, dest_count, rtsp_src2);
355 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
356 int payload_type, const char *line)
360 for (i = 0; i < rt->nb_rtsp_streams; i++) {
361 RTSPStream *rtsp_st = rt->rtsp_streams[i];
362 if (rtsp_st->sdp_payload_type == payload_type &&
363 rtsp_st->dynamic_handler &&
364 rtsp_st->dynamic_handler->parse_sdp_a_line) {
365 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
366 rtsp_st->dynamic_protocol_context, line);
371 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
372 int letter, const char *buf)
374 RTSPState *rt = s->priv_data;
375 char buf1[64], st_type[64];
377 enum AVMediaType codec_type;
381 RTSPSource *rtsp_src;
382 struct sockaddr_storage sdp_ip;
385 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
388 if (s1->skip_media && letter != 'm')
392 get_word(buf1, sizeof(buf1), &p);
393 if (strcmp(buf1, "IN") != 0)
395 get_word(buf1, sizeof(buf1), &p);
396 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
398 get_word_sep(buf1, sizeof(buf1), "/", &p);
399 if (get_sockaddr(buf1, &sdp_ip))
404 get_word_sep(buf1, sizeof(buf1), "/", &p);
407 if (s->nb_streams == 0) {
408 s1->default_ip = sdp_ip;
409 s1->default_ttl = ttl;
411 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
412 rtsp_st->sdp_ip = sdp_ip;
413 rtsp_st->sdp_ttl = ttl;
417 av_dict_set(&s->metadata, "title", p, 0);
420 if (s->nb_streams == 0) {
421 av_dict_set(&s->metadata, "comment", p, 0);
430 codec_type = AVMEDIA_TYPE_UNKNOWN;
431 get_word(st_type, sizeof(st_type), &p);
432 if (!strcmp(st_type, "audio")) {
433 codec_type = AVMEDIA_TYPE_AUDIO;
434 } else if (!strcmp(st_type, "video")) {
435 codec_type = AVMEDIA_TYPE_VIDEO;
436 } else if (!strcmp(st_type, "application")) {
437 codec_type = AVMEDIA_TYPE_DATA;
438 } else if (!strcmp(st_type, "text")) {
439 codec_type = AVMEDIA_TYPE_SUBTITLE;
441 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
445 rtsp_st = av_mallocz(sizeof(RTSPStream));
448 rtsp_st->stream_index = -1;
449 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
451 rtsp_st->sdp_ip = s1->default_ip;
452 rtsp_st->sdp_ttl = s1->default_ttl;
454 copy_default_source_addrs(s1->default_include_source_addrs,
455 s1->nb_default_include_source_addrs,
456 &rtsp_st->include_source_addrs,
457 &rtsp_st->nb_include_source_addrs);
458 copy_default_source_addrs(s1->default_exclude_source_addrs,
459 s1->nb_default_exclude_source_addrs,
460 &rtsp_st->exclude_source_addrs,
461 &rtsp_st->nb_exclude_source_addrs);
463 get_word(buf1, sizeof(buf1), &p); /* port */
464 rtsp_st->sdp_port = atoi(buf1);
466 get_word(buf1, sizeof(buf1), &p); /* protocol */
467 if (!strcmp(buf1, "udp"))
468 rt->transport = RTSP_TRANSPORT_RAW;
469 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
470 rtsp_st->feedback = 1;
472 /* XXX: handle list of formats */
473 get_word(buf1, sizeof(buf1), &p); /* format list */
474 rtsp_st->sdp_payload_type = atoi(buf1);
476 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
477 /* no corresponding stream */
478 if (rt->transport == RTSP_TRANSPORT_RAW) {
479 if (CONFIG_RTPDEC && !rt->ts)
480 rt->ts = avpriv_mpegts_parse_open(s);
482 RTPDynamicProtocolHandler *handler;
483 handler = ff_rtp_handler_find_by_id(
484 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
485 init_rtp_handler(handler, rtsp_st, NULL);
486 finalize_rtp_handler_init(s, rtsp_st, NULL);
488 } else if (rt->server_type == RTSP_SERVER_WMS &&
489 codec_type == AVMEDIA_TYPE_DATA) {
490 /* RTX stream, a stream that carries all the other actual
491 * audio/video streams. Don't expose this to the callers. */
493 st = avformat_new_stream(s, NULL);
496 st->id = rt->nb_rtsp_streams - 1;
497 rtsp_st->stream_index = st->index;
498 st->codec->codec_type = codec_type;
499 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
500 RTPDynamicProtocolHandler *handler;
501 /* if standard payload type, we can find the codec right now */
502 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
503 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
504 st->codec->sample_rate > 0)
505 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
506 /* Even static payload types may need a custom depacketizer */
507 handler = ff_rtp_handler_find_by_id(
508 rtsp_st->sdp_payload_type, st->codec->codec_type);
509 init_rtp_handler(handler, rtsp_st, st);
510 finalize_rtp_handler_init(s, rtsp_st, st);
512 if (rt->default_lang[0])
513 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
515 /* put a default control url */
516 av_strlcpy(rtsp_st->control_url, rt->control_uri,
517 sizeof(rtsp_st->control_url));
520 if (av_strstart(p, "control:", &p)) {
521 if (s->nb_streams == 0) {
522 if (!strncmp(p, "rtsp://", 7))
523 av_strlcpy(rt->control_uri, p,
524 sizeof(rt->control_uri));
527 /* get the control url */
528 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
530 /* XXX: may need to add full url resolution */
531 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
533 if (proto[0] == '\0') {
534 /* relative control URL */
535 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
536 av_strlcat(rtsp_st->control_url, "/",
537 sizeof(rtsp_st->control_url));
538 av_strlcat(rtsp_st->control_url, p,
539 sizeof(rtsp_st->control_url));
541 av_strlcpy(rtsp_st->control_url, p,
542 sizeof(rtsp_st->control_url));
544 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
545 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
546 get_word(buf1, sizeof(buf1), &p);
547 payload_type = atoi(buf1);
548 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
549 if (rtsp_st->stream_index >= 0) {
550 st = s->streams[rtsp_st->stream_index];
551 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
555 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
557 } else if (av_strstart(p, "fmtp:", &p) ||
558 av_strstart(p, "framesize:", &p)) {
559 // let dynamic protocol handlers have a stab at the line.
560 get_word(buf1, sizeof(buf1), &p);
561 payload_type = atoi(buf1);
562 if (s1->seen_rtpmap) {
563 parse_fmtp(s, rt, payload_type, buf);
566 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
568 } else if (av_strstart(p, "range:", &p)) {
571 // this is so that seeking on a streamed file can work.
572 rtsp_parse_range_npt(p, &start, &end);
573 s->start_time = start;
574 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
575 s->duration = (end == AV_NOPTS_VALUE) ?
576 AV_NOPTS_VALUE : end - start;
577 } else if (av_strstart(p, "lang:", &p)) {
578 if (s->nb_streams > 0) {
579 get_word(buf1, sizeof(buf1), &p);
580 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
581 if (rtsp_st->stream_index >= 0) {
582 st = s->streams[rtsp_st->stream_index];
583 av_dict_set(&st->metadata, "language", buf1, 0);
586 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
587 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
589 rt->transport = RTSP_TRANSPORT_RDT;
590 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
592 st = s->streams[s->nb_streams - 1];
593 st->codec->sample_rate = atoi(p);
594 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
596 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
597 get_word(buf1, sizeof(buf1), &p); // ignore tag
598 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
599 p += strspn(p, SPACE_CHARS);
600 if (av_strstart(p, "inline:", &p))
601 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
602 } else if (av_strstart(p, "source-filter:", &p)) {
604 get_word(buf1, sizeof(buf1), &p);
605 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
607 exclude = !strcmp(buf1, "excl");
609 get_word(buf1, sizeof(buf1), &p);
610 if (strcmp(buf1, "IN") != 0)
612 get_word(buf1, sizeof(buf1), &p);
613 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
615 // not checking that the destination address actually matches or is wildcard
616 get_word(buf1, sizeof(buf1), &p);
619 rtsp_src = av_mallocz(sizeof(*rtsp_src));
622 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
624 if (s->nb_streams == 0) {
625 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
627 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
628 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
631 if (s->nb_streams == 0) {
632 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
634 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
635 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
640 if (rt->server_type == RTSP_SERVER_WMS)
641 ff_wms_parse_sdp_a_line(s, p);
642 if (s->nb_streams > 0) {
643 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
645 if (rt->server_type == RTSP_SERVER_REAL)
646 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
648 if (rtsp_st->dynamic_handler &&
649 rtsp_st->dynamic_handler->parse_sdp_a_line)
650 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
651 rtsp_st->stream_index,
652 rtsp_st->dynamic_protocol_context, buf);
659 int ff_sdp_parse(AVFormatContext *s, const char *content)
661 RTSPState *rt = s->priv_data;
664 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
665 * contain long SDP lines containing complete ASF Headers (several
666 * kB) or arrays of MDPR (RM stream descriptor) headers plus
667 * "rulebooks" describing their properties. Therefore, the SDP line
670 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
671 * in rtpdec_xiph.c. */
673 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
677 p += strspn(p, SPACE_CHARS);
685 /* get the content */
687 while (*p != '\n' && *p != '\r' && *p != '\0') {
688 if ((q - buf) < sizeof(buf) - 1)
693 sdp_parse_line(s, s1, letter, buf);
695 while (*p != '\n' && *p != '\0')
701 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
702 av_freep(&s1->default_include_source_addrs[i]);
703 av_freep(&s1->default_include_source_addrs);
704 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
705 av_freep(&s1->default_exclude_source_addrs[i]);
706 av_freep(&s1->default_exclude_source_addrs);
708 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
709 if (!rt->p) return AVERROR(ENOMEM);
712 #endif /* CONFIG_RTPDEC */
714 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
716 RTSPState *rt = s->priv_data;
719 for (i = 0; i < rt->nb_rtsp_streams; i++) {
720 RTSPStream *rtsp_st = rt->rtsp_streams[i];
723 if (rtsp_st->transport_priv) {
725 AVFormatContext *rtpctx = rtsp_st->transport_priv;
726 av_write_trailer(rtpctx);
727 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
728 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
729 ff_rtsp_tcp_write_packet(s, rtsp_st);
730 ffio_free_dyn_buf(&rtpctx->pb);
732 avio_closep(&rtpctx->pb);
734 avformat_free_context(rtpctx);
735 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
736 ff_rdt_parse_close(rtsp_st->transport_priv);
737 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
738 ff_rtp_parse_close(rtsp_st->transport_priv);
740 rtsp_st->transport_priv = NULL;
741 if (rtsp_st->rtp_handle)
742 ffurl_close(rtsp_st->rtp_handle);
743 rtsp_st->rtp_handle = NULL;
747 /* close and free RTSP streams */
748 void ff_rtsp_close_streams(AVFormatContext *s)
750 RTSPState *rt = s->priv_data;
754 ff_rtsp_undo_setup(s, 0);
755 for (i = 0; i < rt->nb_rtsp_streams; i++) {
756 rtsp_st = rt->rtsp_streams[i];
758 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
759 if (rtsp_st->dynamic_handler->close)
760 rtsp_st->dynamic_handler->close(
761 rtsp_st->dynamic_protocol_context);
762 av_free(rtsp_st->dynamic_protocol_context);
764 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
765 av_freep(&rtsp_st->include_source_addrs[j]);
766 av_freep(&rtsp_st->include_source_addrs);
767 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
768 av_freep(&rtsp_st->exclude_source_addrs[j]);
769 av_freep(&rtsp_st->exclude_source_addrs);
774 av_freep(&rt->rtsp_streams);
776 avformat_close_input(&rt->asf_ctx);
778 if (CONFIG_RTPDEC && rt->ts)
779 avpriv_mpegts_parse_close(rt->ts);
781 av_freep(&rt->recvbuf);
784 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
786 RTSPState *rt = s->priv_data;
788 int reordering_queue_size = rt->reordering_queue_size;
789 if (reordering_queue_size < 0) {
790 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
791 reordering_queue_size = 0;
793 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
796 /* open the RTP context */
797 if (rtsp_st->stream_index >= 0)
798 st = s->streams[rtsp_st->stream_index];
800 s->ctx_flags |= AVFMTCTX_NOHEADER;
802 if (CONFIG_RTSP_MUXER && s->oformat && st) {
803 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
804 s, st, rtsp_st->rtp_handle,
805 RTSP_TCP_MAX_PACKET_SIZE,
806 rtsp_st->stream_index);
807 /* Ownership of rtp_handle is passed to the rtp mux context */
808 rtsp_st->rtp_handle = NULL;
811 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
812 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
813 return 0; // Don't need to open any parser here
814 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
815 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
816 rtsp_st->dynamic_protocol_context,
817 rtsp_st->dynamic_handler);
818 else if (CONFIG_RTPDEC)
819 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
820 rtsp_st->sdp_payload_type,
821 reordering_queue_size);
823 if (!rtsp_st->transport_priv) {
824 return AVERROR(ENOMEM);
825 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
826 if (rtsp_st->dynamic_handler) {
827 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
828 rtsp_st->dynamic_protocol_context,
829 rtsp_st->dynamic_handler);
831 if (rtsp_st->crypto_suite[0])
832 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
833 rtsp_st->crypto_suite,
834 rtsp_st->crypto_params);
840 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
841 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
848 q += strspn(q, SPACE_CHARS);
849 v = strtol(q, &p, 10);
853 v = strtol(p, &p, 10);
862 /* XXX: only one transport specification is parsed */
863 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
865 char transport_protocol[16];
867 char lower_transport[16];
869 RTSPTransportField *th;
872 reply->nb_transports = 0;
875 p += strspn(p, SPACE_CHARS);
879 th = &reply->transports[reply->nb_transports];
881 get_word_sep(transport_protocol, sizeof(transport_protocol),
883 if (!av_strcasecmp (transport_protocol, "rtp")) {
884 get_word_sep(profile, sizeof(profile), "/;,", &p);
885 lower_transport[0] = '\0';
886 /* rtp/avp/<protocol> */
888 get_word_sep(lower_transport, sizeof(lower_transport),
891 th->transport = RTSP_TRANSPORT_RTP;
892 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
893 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
894 /* x-pn-tng/<protocol> */
895 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
897 th->transport = RTSP_TRANSPORT_RDT;
898 } else if (!av_strcasecmp(transport_protocol, "raw")) {
899 get_word_sep(profile, sizeof(profile), "/;,", &p);
900 lower_transport[0] = '\0';
901 /* raw/raw/<protocol> */
903 get_word_sep(lower_transport, sizeof(lower_transport),
906 th->transport = RTSP_TRANSPORT_RAW;
908 if (!av_strcasecmp(lower_transport, "TCP"))
909 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
911 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
915 /* get each parameter */
916 while (*p != '\0' && *p != ',') {
917 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
918 if (!strcmp(parameter, "port")) {
921 rtsp_parse_range(&th->port_min, &th->port_max, &p);
923 } else if (!strcmp(parameter, "client_port")) {
926 rtsp_parse_range(&th->client_port_min,
927 &th->client_port_max, &p);
929 } else if (!strcmp(parameter, "server_port")) {
932 rtsp_parse_range(&th->server_port_min,
933 &th->server_port_max, &p);
935 } else if (!strcmp(parameter, "interleaved")) {
938 rtsp_parse_range(&th->interleaved_min,
939 &th->interleaved_max, &p);
941 } else if (!strcmp(parameter, "multicast")) {
942 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
943 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
944 } else if (!strcmp(parameter, "ttl")) {
948 th->ttl = strtol(p, &end, 10);
951 } else if (!strcmp(parameter, "destination")) {
954 get_word_sep(buf, sizeof(buf), ";,", &p);
955 get_sockaddr(buf, &th->destination);
957 } else if (!strcmp(parameter, "source")) {
960 get_word_sep(buf, sizeof(buf), ";,", &p);
961 av_strlcpy(th->source, buf, sizeof(th->source));
963 } else if (!strcmp(parameter, "mode")) {
966 get_word_sep(buf, sizeof(buf), ";, ", &p);
967 if (!strcmp(buf, "record") ||
968 !strcmp(buf, "receive"))
973 while (*p != ';' && *p != '\0' && *p != ',')
981 reply->nb_transports++;
982 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
987 static void handle_rtp_info(RTSPState *rt, const char *url,
988 uint32_t seq, uint32_t rtptime)
991 if (!rtptime || !url[0])
993 if (rt->transport != RTSP_TRANSPORT_RTP)
995 for (i = 0; i < rt->nb_rtsp_streams; i++) {
996 RTSPStream *rtsp_st = rt->rtsp_streams[i];
997 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1000 if (!strcmp(rtsp_st->control_url, url)) {
1001 rtpctx->base_timestamp = rtptime;
1007 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1010 char key[20], value[1024], url[1024] = "";
1011 uint32_t seq = 0, rtptime = 0;
1014 p += strspn(p, SPACE_CHARS);
1017 get_word_sep(key, sizeof(key), "=", &p);
1021 get_word_sep(value, sizeof(value), ";, ", &p);
1023 if (!strcmp(key, "url"))
1024 av_strlcpy(url, value, sizeof(url));
1025 else if (!strcmp(key, "seq"))
1026 seq = strtoul(value, NULL, 10);
1027 else if (!strcmp(key, "rtptime"))
1028 rtptime = strtoul(value, NULL, 10);
1030 handle_rtp_info(rt, url, seq, rtptime);
1039 handle_rtp_info(rt, url, seq, rtptime);
1042 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1043 RTSPState *rt, const char *method)
1047 /* NOTE: we do case independent match for broken servers */
1049 if (av_stristart(p, "Session:", &p)) {
1051 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1052 if (av_stristart(p, ";timeout=", &p) &&
1053 (t = strtol(p, NULL, 10)) > 0) {
1056 } else if (av_stristart(p, "Content-Length:", &p)) {
1057 reply->content_length = strtol(p, NULL, 10);
1058 } else if (av_stristart(p, "Transport:", &p)) {
1059 rtsp_parse_transport(reply, p);
1060 } else if (av_stristart(p, "CSeq:", &p)) {
1061 reply->seq = strtol(p, NULL, 10);
1062 } else if (av_stristart(p, "Range:", &p)) {
1063 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1064 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1065 p += strspn(p, SPACE_CHARS);
1066 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1067 } else if (av_stristart(p, "Server:", &p)) {
1068 p += strspn(p, SPACE_CHARS);
1069 av_strlcpy(reply->server, p, sizeof(reply->server));
1070 } else if (av_stristart(p, "Notice:", &p) ||
1071 av_stristart(p, "X-Notice:", &p)) {
1072 reply->notice = strtol(p, NULL, 10);
1073 } else if (av_stristart(p, "Location:", &p)) {
1074 p += strspn(p, SPACE_CHARS);
1075 av_strlcpy(reply->location, p , sizeof(reply->location));
1076 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1077 p += strspn(p, SPACE_CHARS);
1078 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1079 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1080 p += strspn(p, SPACE_CHARS);
1081 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1082 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1083 p += strspn(p, SPACE_CHARS);
1084 if (method && !strcmp(method, "DESCRIBE"))
1085 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1086 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1087 p += strspn(p, SPACE_CHARS);
1088 if (method && !strcmp(method, "PLAY"))
1089 rtsp_parse_rtp_info(rt, p);
1090 } else if (av_stristart(p, "Public:", &p) && rt) {
1091 if (strstr(p, "GET_PARAMETER") &&
1092 method && !strcmp(method, "OPTIONS"))
1093 rt->get_parameter_supported = 1;
1094 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1095 p += strspn(p, SPACE_CHARS);
1096 rt->accept_dynamic_rate = atoi(p);
1097 } else if (av_stristart(p, "Content-Type:", &p)) {
1098 p += strspn(p, SPACE_CHARS);
1099 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1103 /* skip a RTP/TCP interleaved packet */
1104 void ff_rtsp_skip_packet(AVFormatContext *s)
1106 RTSPState *rt = s->priv_data;
1110 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1113 len = AV_RB16(buf + 1);
1115 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1120 if (len1 > sizeof(buf))
1122 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1129 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1130 unsigned char **content_ptr,
1131 int return_on_interleaved_data, const char *method)
1133 RTSPState *rt = s->priv_data;
1134 char buf[4096], buf1[1024], *q;
1137 int ret, content_length, line_count = 0, request = 0;
1138 unsigned char *content = NULL;
1144 memset(reply, 0, sizeof(*reply));
1146 /* parse reply (XXX: use buffers) */
1147 rt->last_reply[0] = '\0';
1151 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1152 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1158 /* XXX: only parse it if first char on line ? */
1159 if (return_on_interleaved_data) {
1162 ff_rtsp_skip_packet(s);
1163 } else if (ch != '\r') {
1164 if ((q - buf) < sizeof(buf) - 1)
1170 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1172 /* test if last line */
1176 if (line_count == 0) {
1177 /* get reply code */
1178 get_word(buf1, sizeof(buf1), &p);
1179 if (!strncmp(buf1, "RTSP/", 5)) {
1180 get_word(buf1, sizeof(buf1), &p);
1181 reply->status_code = atoi(buf1);
1182 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1184 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1185 get_word(buf1, sizeof(buf1), &p); // object
1189 ff_rtsp_parse_line(reply, p, rt, method);
1190 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1191 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1196 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1197 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1199 content_length = reply->content_length;
1200 if (content_length > 0) {
1201 /* leave some room for a trailing '\0' (useful for simple parsing) */
1202 content = av_malloc(content_length + 1);
1204 return AVERROR(ENOMEM);
1205 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1206 content[content_length] = '\0';
1209 *content_ptr = content;
1215 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1216 const char* ptr = buf;
1218 if (!strcmp(reply->reason, "OPTIONS")) {
1219 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1221 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1222 if (reply->session_id[0])
1223 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1226 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1228 av_strlcat(buf, "\r\n", sizeof(buf));
1230 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1231 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1234 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1236 rt->last_cmd_time = av_gettime_relative();
1237 /* Even if the request from the server had data, it is not the data
1238 * that the caller wants or expects. The memory could also be leaked
1239 * if the actual following reply has content data. */
1241 av_freep(content_ptr);
1242 /* If method is set, this is called from ff_rtsp_send_cmd,
1243 * where a reply to exactly this request is awaited. For
1244 * callers from within packet receiving, we just want to
1245 * return to the caller and go back to receiving packets. */
1251 if (rt->seq != reply->seq) {
1252 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1253 rt->seq, reply->seq);
1257 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1258 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1259 reply->notice == 2306 /* Continuous Feed Terminated */) {
1260 rt->state = RTSP_STATE_IDLE;
1261 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1262 return AVERROR(EIO); /* data or server error */
1263 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1264 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1265 return AVERROR(EPERM);
1271 * Send a command to the RTSP server without waiting for the reply.
1273 * @param s RTSP (de)muxer context
1274 * @param method the method for the request
1275 * @param url the target url for the request
1276 * @param headers extra header lines to include in the request
1277 * @param send_content if non-null, the data to send as request body content
1278 * @param send_content_length the length of the send_content data, or 0 if
1279 * send_content is null
1281 * @return zero if success, nonzero otherwise
1283 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1284 const char *method, const char *url,
1285 const char *headers,
1286 const unsigned char *send_content,
1287 int send_content_length)
1289 RTSPState *rt = s->priv_data;
1290 char buf[4096], *out_buf;
1291 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1293 /* Add in RTSP headers */
1296 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1298 av_strlcat(buf, headers, sizeof(buf));
1299 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1300 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1301 if (rt->session_id[0] != '\0' && (!headers ||
1302 !strstr(headers, "\nIf-Match:"))) {
1303 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1306 char *str = ff_http_auth_create_response(&rt->auth_state,
1307 rt->auth, url, method);
1309 av_strlcat(buf, str, sizeof(buf));
1312 if (send_content_length > 0 && send_content)
1313 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1314 av_strlcat(buf, "\r\n", sizeof(buf));
1316 /* base64 encode rtsp if tunneling */
1317 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1318 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1319 out_buf = base64buf;
1322 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1324 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1325 if (send_content_length > 0 && send_content) {
1326 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1327 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1328 "with content data not supported\n");
1329 return AVERROR_PATCHWELCOME;
1331 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1333 rt->last_cmd_time = av_gettime_relative();
1338 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1339 const char *url, const char *headers)
1341 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1344 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1345 const char *headers, RTSPMessageHeader *reply,
1346 unsigned char **content_ptr)
1348 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1349 content_ptr, NULL, 0);
1352 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1353 const char *method, const char *url,
1355 RTSPMessageHeader *reply,
1356 unsigned char **content_ptr,
1357 const unsigned char *send_content,
1358 int send_content_length)
1360 RTSPState *rt = s->priv_data;
1361 HTTPAuthType cur_auth_type;
1362 int ret, attempts = 0;
1365 cur_auth_type = rt->auth_state.auth_type;
1366 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1368 send_content_length)))
1371 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1375 if (reply->status_code == 401 &&
1376 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1377 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1380 if (reply->status_code > 400){
1381 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1385 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1391 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1392 int lower_transport, const char *real_challenge)
1394 RTSPState *rt = s->priv_data;
1395 int rtx = 0, j, i, err, interleave = 0, port_off;
1396 RTSPStream *rtsp_st;
1397 RTSPMessageHeader reply1, *reply = &reply1;
1399 const char *trans_pref;
1401 if (rt->transport == RTSP_TRANSPORT_RDT)
1402 trans_pref = "x-pn-tng";
1403 else if (rt->transport == RTSP_TRANSPORT_RAW)
1404 trans_pref = "RAW/RAW";
1406 trans_pref = "RTP/AVP";
1408 /* default timeout: 1 minute */
1411 /* Choose a random starting offset within the first half of the
1412 * port range, to allow for a number of ports to try even if the offset
1413 * happens to be at the end of the random range. */
1414 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1415 /* even random offset */
1416 port_off -= port_off & 0x01;
1418 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1419 char transport[2048];
1422 * WMS serves all UDP data over a single connection, the RTX, which
1423 * isn't necessarily the first in the SDP but has to be the first
1424 * to be set up, else the second/third SETUP will fail with a 461.
1426 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1427 rt->server_type == RTSP_SERVER_WMS) {
1430 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1431 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1433 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1437 if (rtx == rt->nb_rtsp_streams)
1438 return -1; /* no RTX found */
1439 rtsp_st = rt->rtsp_streams[rtx];
1441 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1443 rtsp_st = rt->rtsp_streams[i];
1446 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1449 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1450 port = reply->transports[0].client_port_min;
1454 /* first try in specified port range */
1455 while (j <= rt->rtp_port_max) {
1456 AVDictionary *opts = map_to_opts(rt);
1458 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1459 "?localport=%d", j);
1460 /* we will use two ports per rtp stream (rtp and rtcp) */
1462 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1463 &s->interrupt_callback, &opts);
1465 av_dict_free(&opts);
1470 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1475 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1477 snprintf(transport, sizeof(transport) - 1,
1478 "%s/UDP;", trans_pref);
1479 if (rt->server_type != RTSP_SERVER_REAL)
1480 av_strlcat(transport, "unicast;", sizeof(transport));
1481 av_strlcatf(transport, sizeof(transport),
1482 "client_port=%d", port);
1483 if (rt->transport == RTSP_TRANSPORT_RTP &&
1484 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1485 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1489 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1490 /* For WMS streams, the application streams are only used for
1491 * UDP. When trying to set it up for TCP streams, the server
1492 * will return an error. Therefore, we skip those streams. */
1493 if (rt->server_type == RTSP_SERVER_WMS &&
1494 (rtsp_st->stream_index < 0 ||
1495 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1498 snprintf(transport, sizeof(transport) - 1,
1499 "%s/TCP;", trans_pref);
1500 if (rt->transport != RTSP_TRANSPORT_RDT)
1501 av_strlcat(transport, "unicast;", sizeof(transport));
1502 av_strlcatf(transport, sizeof(transport),
1503 "interleaved=%d-%d",
1504 interleave, interleave + 1);
1508 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1509 snprintf(transport, sizeof(transport) - 1,
1510 "%s/UDP;multicast", trans_pref);
1513 av_strlcat(transport, ";mode=record", sizeof(transport));
1514 } else if (rt->server_type == RTSP_SERVER_REAL ||
1515 rt->server_type == RTSP_SERVER_WMS)
1516 av_strlcat(transport, ";mode=play", sizeof(transport));
1517 snprintf(cmd, sizeof(cmd),
1518 "Transport: %s\r\n",
1520 if (rt->accept_dynamic_rate)
1521 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1522 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1523 char real_res[41], real_csum[9];
1524 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1526 av_strlcatf(cmd, sizeof(cmd),
1528 "RealChallenge2: %s, sd=%s\r\n",
1529 rt->session_id, real_res, real_csum);
1531 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1532 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1535 } else if (reply->status_code != RTSP_STATUS_OK ||
1536 reply->nb_transports != 1) {
1537 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1541 /* XXX: same protocol for all streams is required */
1543 if (reply->transports[0].lower_transport != rt->lower_transport ||
1544 reply->transports[0].transport != rt->transport) {
1545 err = AVERROR_INVALIDDATA;
1549 rt->lower_transport = reply->transports[0].lower_transport;
1550 rt->transport = reply->transports[0].transport;
1553 /* Fail if the server responded with another lower transport mode
1554 * than what we requested. */
1555 if (reply->transports[0].lower_transport != lower_transport) {
1556 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1557 err = AVERROR_INVALIDDATA;
1561 switch(reply->transports[0].lower_transport) {
1562 case RTSP_LOWER_TRANSPORT_TCP:
1563 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1564 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1567 case RTSP_LOWER_TRANSPORT_UDP: {
1568 char url[1024], options[30] = "";
1569 const char *peer = host;
1571 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1572 av_strlcpy(options, "?connect=1", sizeof(options));
1573 /* Use source address if specified */
1574 if (reply->transports[0].source[0])
1575 peer = reply->transports[0].source;
1576 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1577 reply->transports[0].server_port_min, "%s", options);
1578 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1579 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1580 err = AVERROR_INVALIDDATA;
1585 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1586 char url[1024], namebuf[50], optbuf[20] = "";
1587 struct sockaddr_storage addr;
1590 if (reply->transports[0].destination.ss_family) {
1591 addr = reply->transports[0].destination;
1592 port = reply->transports[0].port_min;
1593 ttl = reply->transports[0].ttl;
1595 addr = rtsp_st->sdp_ip;
1596 port = rtsp_st->sdp_port;
1597 ttl = rtsp_st->sdp_ttl;
1600 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1601 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1602 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1603 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1604 port, "%s", optbuf);
1605 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1606 &s->interrupt_callback, NULL) < 0) {
1607 err = AVERROR_INVALIDDATA;
1614 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1618 if (rt->nb_rtsp_streams && reply->timeout > 0)
1619 rt->timeout = reply->timeout;
1621 if (rt->server_type == RTSP_SERVER_REAL)
1622 rt->need_subscription = 1;
1627 ff_rtsp_undo_setup(s, 0);
1631 void ff_rtsp_close_connections(AVFormatContext *s)
1633 RTSPState *rt = s->priv_data;
1634 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1635 ffurl_close(rt->rtsp_hd);
1636 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1639 int ff_rtsp_connect(AVFormatContext *s)
1641 RTSPState *rt = s->priv_data;
1642 char proto[128], host[1024], path[1024];
1643 char tcpname[1024], cmd[2048], auth[128];
1644 const char *lower_rtsp_proto = "tcp";
1645 int port, err, tcp_fd;
1646 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1647 int lower_transport_mask = 0;
1648 int default_port = RTSP_DEFAULT_PORT;
1649 char real_challenge[64] = "";
1650 struct sockaddr_storage peer;
1651 socklen_t peer_len = sizeof(peer);
1653 if (rt->rtp_port_max < rt->rtp_port_min) {
1654 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1655 "than min port %d\n", rt->rtp_port_max,
1657 return AVERROR(EINVAL);
1660 if (!ff_network_init())
1661 return AVERROR(EIO);
1663 if (s->max_delay < 0) /* Not set by the caller */
1664 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1666 rt->control_transport = RTSP_MODE_PLAIN;
1667 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1668 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1669 rt->control_transport = RTSP_MODE_TUNNEL;
1671 /* Only pass through valid flags from here */
1672 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1675 /* extract hostname and port */
1676 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1677 host, sizeof(host), &port, path, sizeof(path), s->filename);
1679 if (!strcmp(proto, "rtsps")) {
1680 lower_rtsp_proto = "tls";
1681 default_port = RTSPS_DEFAULT_PORT;
1682 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1686 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1689 port = default_port;
1691 lower_transport_mask = rt->lower_transport_mask;
1693 if (!lower_transport_mask)
1694 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1697 /* Only UDP or TCP - UDP multicast isn't supported. */
1698 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1699 (1 << RTSP_LOWER_TRANSPORT_TCP);
1700 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1701 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1702 "only UDP and TCP are supported for output.\n");
1703 err = AVERROR(EINVAL);
1708 /* Construct the URI used in request; this is similar to s->filename,
1709 * but with authentication credentials removed and RTSP specific options
1711 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1712 host, port, "%s", path);
1714 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1715 /* set up initial handshake for tunneling */
1716 char httpname[1024];
1717 char sessioncookie[17];
1720 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1721 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1722 av_get_random_seed(), av_get_random_seed());
1725 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1726 &s->interrupt_callback) < 0) {
1731 /* generate GET headers */
1732 snprintf(headers, sizeof(headers),
1733 "x-sessioncookie: %s\r\n"
1734 "Accept: application/x-rtsp-tunnelled\r\n"
1735 "Pragma: no-cache\r\n"
1736 "Cache-Control: no-cache\r\n",
1738 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1740 /* complete the connection */
1741 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1747 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1748 &s->interrupt_callback) < 0 ) {
1753 /* generate POST headers */
1754 snprintf(headers, sizeof(headers),
1755 "x-sessioncookie: %s\r\n"
1756 "Content-Type: application/x-rtsp-tunnelled\r\n"
1757 "Pragma: no-cache\r\n"
1758 "Cache-Control: no-cache\r\n"
1759 "Content-Length: 32767\r\n"
1760 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1762 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1763 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1765 /* Initialize the authentication state for the POST session. The HTTP
1766 * protocol implementation doesn't properly handle multi-pass
1767 * authentication for POST requests, since it would require one of
1769 * - implementing Expect: 100-continue, which many HTTP servers
1770 * don't support anyway, even less the RTSP servers that do HTTP
1772 * - sending the whole POST data until getting a 401 reply specifying
1773 * what authentication method to use, then resending all that data
1774 * - waiting for potential 401 replies directly after sending the
1775 * POST header (waiting for some unspecified time)
1776 * Therefore, we copy the full auth state, which works for both basic
1777 * and digest. (For digest, we would have to synchronize the nonce
1778 * count variable between the two sessions, if we'd do more requests
1779 * with the original session, though.)
1781 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1783 /* complete the connection */
1784 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1790 /* open the tcp connection */
1791 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1793 "?timeout=%d", rt->stimeout);
1794 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1795 &s->interrupt_callback, NULL)) < 0) {
1799 rt->rtsp_hd_out = rt->rtsp_hd;
1803 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1808 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1809 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1810 NULL, 0, NI_NUMERICHOST);
1813 /* request options supported by the server; this also detects server
1815 for (rt->server_type = RTSP_SERVER_RTP;;) {
1817 if (rt->server_type == RTSP_SERVER_REAL)
1820 * The following entries are required for proper
1821 * streaming from a Realmedia server. They are
1822 * interdependent in some way although we currently
1823 * don't quite understand how. Values were copied
1824 * from mplayer SVN r23589.
1825 * ClientChallenge is a 16-byte ID in hex
1826 * CompanyID is a 16-byte ID in base64
1828 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1829 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1830 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1831 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1833 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1834 if (reply->status_code != RTSP_STATUS_OK) {
1835 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1839 /* detect server type if not standard-compliant RTP */
1840 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1841 rt->server_type = RTSP_SERVER_REAL;
1843 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1844 rt->server_type = RTSP_SERVER_WMS;
1845 } else if (rt->server_type == RTSP_SERVER_REAL)
1846 strcpy(real_challenge, reply->real_challenge);
1850 if (CONFIG_RTSP_DEMUXER && s->iformat)
1851 err = ff_rtsp_setup_input_streams(s, reply);
1852 else if (CONFIG_RTSP_MUXER)
1853 err = ff_rtsp_setup_output_streams(s, host);
1860 int lower_transport = ff_log2_tab[lower_transport_mask &
1861 ~(lower_transport_mask - 1)];
1863 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1864 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1865 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1867 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1868 rt->server_type == RTSP_SERVER_REAL ?
1869 real_challenge : NULL);
1872 lower_transport_mask &= ~(1 << lower_transport);
1873 if (lower_transport_mask == 0 && err == 1) {
1874 err = AVERROR(EPROTONOSUPPORT);
1879 rt->lower_transport_mask = lower_transport_mask;
1880 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1881 rt->state = RTSP_STATE_IDLE;
1882 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1885 ff_rtsp_close_streams(s);
1886 ff_rtsp_close_connections(s);
1887 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1888 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1889 rt->session_id[0] = '\0';
1890 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1898 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1901 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1902 uint8_t *buf, int buf_size, int64_t wait_end)
1904 RTSPState *rt = s->priv_data;
1905 RTSPStream *rtsp_st;
1906 int n, i, ret, tcp_fd, timeout_cnt = 0;
1908 struct pollfd *p = rt->p;
1909 int *fds = NULL, fdsnum, fdsidx;
1912 if (ff_check_interrupt(&s->interrupt_callback))
1913 return AVERROR_EXIT;
1914 if (wait_end && wait_end - av_gettime_relative() < 0)
1915 return AVERROR(EAGAIN);
1918 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1919 p[max_p].fd = tcp_fd;
1920 p[max_p++].events = POLLIN;
1924 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1925 rtsp_st = rt->rtsp_streams[i];
1926 if (rtsp_st->rtp_handle) {
1927 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1929 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1933 av_log(s, AV_LOG_ERROR,
1934 "Number of fds %d not supported\n", fdsnum);
1935 return AVERROR_INVALIDDATA;
1937 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1938 p[max_p].fd = fds[fdsidx];
1939 p[max_p++].events = POLLIN;
1944 n = poll(p, max_p, POLL_TIMEOUT_MS);
1946 int j = 1 - (tcp_fd == -1);
1948 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1949 rtsp_st = rt->rtsp_streams[i];
1950 if (rtsp_st->rtp_handle) {
1951 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1952 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1954 *prtsp_st = rtsp_st;
1961 #if CONFIG_RTSP_DEMUXER
1962 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1963 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1964 if (rt->state == RTSP_STATE_STREAMING) {
1965 if (!ff_rtsp_parse_streaming_commands(s))
1968 av_log(s, AV_LOG_WARNING,
1969 "Unable to answer to TEARDOWN\n");
1973 RTSPMessageHeader reply;
1974 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1977 /* XXX: parse message */
1978 if (rt->state != RTSP_STATE_STREAMING)
1983 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1984 return AVERROR(ETIMEDOUT);
1985 } else if (n < 0 && errno != EINTR)
1986 return AVERROR(errno);
1990 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1991 const uint8_t *buf, int len)
1993 RTSPState *rt = s->priv_data;
1997 if (rt->nb_rtsp_streams == 1) {
1998 *rtsp_st = rt->rtsp_streams[0];
2001 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2002 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2004 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2005 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2008 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2009 *rtsp_st = rt->rtsp_streams[i];
2016 av_log(s, AV_LOG_WARNING,
2017 "Unable to pick stream for packet - SSRC not known for "
2019 return AVERROR(EAGAIN);
2022 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2023 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2024 *rtsp_st = rt->rtsp_streams[i];
2030 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2031 return AVERROR(EAGAIN);
2034 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2036 RTSPState *rt = s->priv_data;
2038 RTSPStream *rtsp_st, *first_queue_st = NULL;
2039 int64_t wait_end = 0;
2041 if (rt->nb_byes == rt->nb_rtsp_streams)
2044 /* get next frames from the same RTP packet */
2045 if (rt->cur_transport_priv) {
2046 if (rt->transport == RTSP_TRANSPORT_RDT) {
2047 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2048 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2049 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2050 } else if (CONFIG_RTPDEC && rt->ts) {
2051 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2053 rt->recvbuf_pos += ret;
2054 ret = rt->recvbuf_pos < rt->recvbuf_len;
2059 rt->cur_transport_priv = NULL;
2061 } else if (ret == 1) {
2064 rt->cur_transport_priv = NULL;
2068 if (rt->transport == RTSP_TRANSPORT_RTP) {
2070 int64_t first_queue_time = 0;
2071 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2072 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2076 queue_time = ff_rtp_queued_packet_time(rtpctx);
2077 if (queue_time && (queue_time - first_queue_time < 0 ||
2078 !first_queue_time)) {
2079 first_queue_time = queue_time;
2080 first_queue_st = rt->rtsp_streams[i];
2083 if (first_queue_time) {
2084 wait_end = first_queue_time + s->max_delay;
2087 first_queue_st = NULL;
2091 /* read next RTP packet */
2093 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2095 return AVERROR(ENOMEM);
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 = ffio_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);
2122 if (len == AVERROR(EAGAIN) && first_queue_st &&
2123 rt->transport == RTSP_TRANSPORT_RTP) {
2124 rtsp_st = first_queue_st;
2125 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2132 if (rt->transport == RTSP_TRANSPORT_RDT) {
2133 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2134 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2135 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2136 if (rtsp_st->feedback) {
2137 AVIOContext *pb = NULL;
2138 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2140 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2143 /* Either bad packet, or a RTCP packet. Check if the
2144 * first_rtcp_ntp_time field was initialized. */
2145 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2146 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2147 /* first_rtcp_ntp_time has been initialized for this stream,
2148 * copy the same value to all other uninitialized streams,
2149 * in order to map their timestamp origin to the same ntp time
2152 AVStream *st = NULL;
2153 if (rtsp_st->stream_index >= 0)
2154 st = s->streams[rtsp_st->stream_index];
2155 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2156 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2157 AVStream *st2 = NULL;
2158 if (rt->rtsp_streams[i]->stream_index >= 0)
2159 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2160 if (rtpctx2 && st && st2 &&
2161 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2162 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2163 rtpctx2->rtcp_ts_offset = av_rescale_q(
2164 rtpctx->rtcp_ts_offset, st->time_base,
2168 // Make real NTP start time available in AVFormatContext
2169 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2170 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2172 s->start_time_realtime -=
2173 av_rescale (rtpctx->rtcp_ts_offset,
2174 (uint64_t) rtpctx->st->time_base.num * 1000000,
2175 rtpctx->st->time_base.den);
2179 if (ret == -RTCP_BYE) {
2182 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2183 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2185 if (rt->nb_byes == rt->nb_rtsp_streams)
2189 } else if (CONFIG_RTPDEC && rt->ts) {
2190 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2193 rt->recvbuf_len = len;
2194 rt->recvbuf_pos = ret;
2195 rt->cur_transport_priv = rt->ts;
2202 return AVERROR_INVALIDDATA;
2208 /* more packets may follow, so we save the RTP context */
2209 rt->cur_transport_priv = rtsp_st->transport_priv;
2213 #endif /* CONFIG_RTPDEC */
2215 #if CONFIG_SDP_DEMUXER
2216 static int sdp_probe(AVProbeData *p1)
2218 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2220 /* we look for a line beginning "c=IN IP" */
2221 while (p < p_end && *p != '\0') {
2222 if (sizeof("c=IN IP") - 1 < p_end - p &&
2223 av_strstart(p, "c=IN IP", NULL))
2224 return AVPROBE_SCORE_EXTENSION;
2226 while (p < p_end - 1 && *p != '\n') p++;
2235 static void append_source_addrs(char *buf, int size, const char *name,
2236 int count, struct RTSPSource **addrs)
2241 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2242 for (i = 1; i < count; i++)
2243 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2246 static int sdp_read_header(AVFormatContext *s)
2248 RTSPState *rt = s->priv_data;
2249 RTSPStream *rtsp_st;
2254 if (!ff_network_init())
2255 return AVERROR(EIO);
2257 if (s->max_delay < 0) /* Not set by the caller */
2258 s->max_delay = DEFAULT_REORDERING_DELAY;
2259 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2260 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2262 /* read the whole sdp file */
2263 /* XXX: better loading */
2264 content = av_malloc(SDP_MAX_SIZE);
2266 return AVERROR(ENOMEM);
2267 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2270 return AVERROR_INVALIDDATA;
2272 content[size] ='\0';
2274 err = ff_sdp_parse(s, content);
2278 /* open each RTP stream */
2279 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2281 rtsp_st = rt->rtsp_streams[i];
2283 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2284 AVDictionary *opts = map_to_opts(rt);
2286 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2287 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2288 ff_url_join(url, sizeof(url), "rtp", NULL,
2289 namebuf, rtsp_st->sdp_port,
2290 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2291 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2292 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2293 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2295 append_source_addrs(url, sizeof(url), "sources",
2296 rtsp_st->nb_include_source_addrs,
2297 rtsp_st->include_source_addrs);
2298 append_source_addrs(url, sizeof(url), "block",
2299 rtsp_st->nb_exclude_source_addrs,
2300 rtsp_st->exclude_source_addrs);
2301 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2302 &s->interrupt_callback, &opts);
2304 av_dict_free(&opts);
2307 err = AVERROR_INVALIDDATA;
2311 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2316 ff_rtsp_close_streams(s);
2321 static int sdp_read_close(AVFormatContext *s)
2323 ff_rtsp_close_streams(s);
2328 static const AVClass sdp_demuxer_class = {
2329 .class_name = "SDP demuxer",
2330 .item_name = av_default_item_name,
2331 .option = sdp_options,
2332 .version = LIBAVUTIL_VERSION_INT,
2335 AVInputFormat ff_sdp_demuxer = {
2337 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2338 .priv_data_size = sizeof(RTSPState),
2339 .read_probe = sdp_probe,
2340 .read_header = sdp_read_header,
2341 .read_packet = ff_rtsp_fetch_packet,
2342 .read_close = sdp_read_close,
2343 .priv_class = &sdp_demuxer_class,
2345 #endif /* CONFIG_SDP_DEMUXER */
2347 #if CONFIG_RTP_DEMUXER
2348 static int rtp_probe(AVProbeData *p)
2350 if (av_strstart(p->filename, "rtp:", NULL))
2351 return AVPROBE_SCORE_MAX;
2355 static int rtp_read_header(AVFormatContext *s)
2357 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2358 char host[500], sdp[500];
2360 URLContext* in = NULL;
2362 AVCodecContext codec = { 0 };
2363 struct sockaddr_storage addr;
2365 socklen_t addrlen = sizeof(addr);
2366 RTSPState *rt = s->priv_data;
2368 if (!ff_network_init())
2369 return AVERROR(EIO);
2371 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2372 &s->interrupt_callback, NULL);
2377 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2378 if (ret == AVERROR(EAGAIN))
2383 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2387 if ((recvbuf[0] & 0xc0) != 0x80) {
2388 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2393 if (RTP_PT_IS_RTCP(recvbuf[1]))
2396 payload_type = recvbuf[1] & 0x7f;
2399 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2403 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2404 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2405 "without an SDP file describing it\n",
2409 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2410 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2411 "properly you need an SDP file "
2415 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2416 NULL, 0, s->filename);
2418 snprintf(sdp, sizeof(sdp),
2419 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2420 addr.ss_family == AF_INET ? 4 : 6, host,
2421 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2422 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2423 port, payload_type);
2424 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2426 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2429 /* sdp_read_header initializes this again */
2432 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2434 ret = sdp_read_header(s);
2445 static const AVClass rtp_demuxer_class = {
2446 .class_name = "RTP demuxer",
2447 .item_name = av_default_item_name,
2448 .option = rtp_options,
2449 .version = LIBAVUTIL_VERSION_INT,
2452 AVInputFormat ff_rtp_demuxer = {
2454 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2455 .priv_data_size = sizeof(RTSPState),
2456 .read_probe = rtp_probe,
2457 .read_header = rtp_read_header,
2458 .read_packet = ff_rtsp_fetch_packet,
2459 .read_close = sdp_read_close,
2460 .flags = AVFMT_NOFILE,
2461 .priv_class = &rtp_demuxer_class,
2463 #endif /* CONFIG_RTP_DEMUXER */