3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/bprint.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/parseutils.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/dict.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/time.h"
34 #include "avio_internal.h"
41 #include "os_support.h"
48 #include "rtpdec_formats.h"
49 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "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" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { 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" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
76 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
78 #define COMMON_OPTS() \
79 { "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 }, \
80 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC }, \
81 { "pkt_size", "Underlying protocol send packet size", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC } \
84 const AVOption ff_rtsp_options[] = {
85 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
86 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
87 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
88 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
89 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
90 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
91 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
92 { "https", "HTTPS tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTPS )}, 0, 0, DEC, "rtsp_transport" },
93 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
94 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
95 { "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" },
96 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
97 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
98 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
99 { "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 },
100 #if FF_API_OLD_RTSP_OPTIONS
101 { "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 },
102 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
104 { "timeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
107 { "user_agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
108 #if FF_API_OLD_RTSP_OPTIONS
109 { "user-agent", "override User-Agent header (deprecated, use user_agent)", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
114 static const AVOption sdp_options[] = {
115 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
116 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
117 { "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" },
118 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
123 static const AVOption rtp_options[] = {
124 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
130 static AVDictionary *map_to_opts(RTSPState *rt)
132 AVDictionary *opts = NULL;
135 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
136 av_dict_set(&opts, "buffer_size", buf, 0);
137 snprintf(buf, sizeof(buf), "%d", rt->pkt_size);
138 av_dict_set(&opts, "pkt_size", buf, 0);
143 static void get_word_until_chars(char *buf, int buf_size,
144 const char *sep, const char **pp)
150 p += strspn(p, SPACE_CHARS);
152 while (!strchr(sep, *p) && *p != '\0') {
153 if ((q - buf) < buf_size - 1)
162 static void get_word_sep(char *buf, int buf_size, const char *sep,
165 if (**pp == '/') (*pp)++;
166 get_word_until_chars(buf, buf_size, sep, pp);
169 static void get_word(char *buf, int buf_size, const char **pp)
171 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
174 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
176 * Used for seeking in the rtp stream.
178 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
182 p += strspn(p, SPACE_CHARS);
183 if (!av_stristart(p, "npt=", &p))
186 *start = AV_NOPTS_VALUE;
187 *end = AV_NOPTS_VALUE;
189 get_word_sep(buf, sizeof(buf), "-", &p);
190 if (av_parse_time(start, buf, 1) < 0)
194 get_word_sep(buf, sizeof(buf), "-", &p);
195 if (av_parse_time(end, buf, 1) < 0)
196 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
200 static int get_sockaddr(AVFormatContext *s,
201 const char *buf, struct sockaddr_storage *sock)
203 struct addrinfo hints = { 0 }, *ai = NULL;
206 hints.ai_flags = AI_NUMERICHOST;
207 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
208 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
213 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
219 static void init_rtp_handler(const RTPDynamicProtocolHandler *handler,
220 RTSPStream *rtsp_st, AVStream *st)
222 AVCodecParameters *par = st ? st->codecpar : NULL;
226 par->codec_id = handler->codec_id;
227 rtsp_st->dynamic_handler = handler;
229 st->need_parsing = handler->need_parsing;
230 if (handler->priv_data_size) {
231 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
232 if (!rtsp_st->dynamic_protocol_context)
233 rtsp_st->dynamic_handler = NULL;
237 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
240 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
241 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
242 rtsp_st->dynamic_protocol_context);
244 if (rtsp_st->dynamic_protocol_context) {
245 if (rtsp_st->dynamic_handler->close)
246 rtsp_st->dynamic_handler->close(
247 rtsp_st->dynamic_protocol_context);
248 av_free(rtsp_st->dynamic_protocol_context);
250 rtsp_st->dynamic_protocol_context = NULL;
251 rtsp_st->dynamic_handler = NULL;
256 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
257 static int sdp_parse_rtpmap(AVFormatContext *s,
258 AVStream *st, RTSPStream *rtsp_st,
259 int payload_type, const char *p)
261 AVCodecParameters *par = st->codecpar;
264 const AVCodecDescriptor *desc;
267 /* See if we can handle this kind of payload.
268 * The space should normally not be there but some Real streams or
269 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
270 * have a trailing space. */
271 get_word_sep(buf, sizeof(buf), "/ ", &p);
272 if (payload_type < RTP_PT_PRIVATE) {
273 /* We are in a standard case
274 * (from http://www.iana.org/assignments/rtp-parameters). */
275 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
278 if (par->codec_id == AV_CODEC_ID_NONE) {
279 const RTPDynamicProtocolHandler *handler =
280 ff_rtp_handler_find_by_name(buf, par->codec_type);
281 init_rtp_handler(handler, rtsp_st, st);
282 /* If no dynamic handler was found, check with the list of standard
283 * allocated types, if such a stream for some reason happens to
284 * use a private payload type. This isn't handled in rtpdec.c, since
285 * the format name from the rtpmap line never is passed into rtpdec. */
286 if (!rtsp_st->dynamic_handler)
287 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
290 desc = avcodec_descriptor_get(par->codec_id);
291 if (desc && desc->name)
296 get_word_sep(buf, sizeof(buf), "/", &p);
298 switch (par->codec_type) {
299 case AVMEDIA_TYPE_AUDIO:
300 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
301 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
302 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
304 par->sample_rate = i;
305 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
306 get_word_sep(buf, sizeof(buf), "/", &p);
311 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
313 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
316 case AVMEDIA_TYPE_VIDEO:
317 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
319 avpriv_set_pts_info(st, 32, 1, i);
324 finalize_rtp_handler_init(s, rtsp_st, st);
328 /* parse the attribute line from the fmtp a line of an sdp response. This
329 * is broken out as a function because it is used in rtp_h264.c, which is
331 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
332 char *value, int value_size)
334 *p += strspn(*p, SPACE_CHARS);
336 get_word_sep(attr, attr_size, "=", p);
339 get_word_sep(value, value_size, ";", p);
347 typedef struct SDPParseState {
349 struct sockaddr_storage default_ip;
351 int skip_media; ///< set if an unknown m= line occurs
352 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
353 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
354 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
355 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
358 char delayed_fmtp[2048];
361 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
362 struct RTSPSource ***dest, int *dest_count)
364 RTSPSource *rtsp_src, *rtsp_src2;
366 for (i = 0; i < count; i++) {
368 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
371 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
372 dynarray_add(dest, dest_count, rtsp_src2);
376 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
377 int payload_type, const char *line)
381 for (i = 0; i < rt->nb_rtsp_streams; i++) {
382 RTSPStream *rtsp_st = rt->rtsp_streams[i];
383 if (rtsp_st->sdp_payload_type == payload_type &&
384 rtsp_st->dynamic_handler &&
385 rtsp_st->dynamic_handler->parse_sdp_a_line) {
386 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
387 rtsp_st->dynamic_protocol_context, line);
392 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
393 int letter, const char *buf)
395 RTSPState *rt = s->priv_data;
396 char buf1[64], st_type[64];
398 enum AVMediaType codec_type;
402 RTSPSource *rtsp_src;
403 struct sockaddr_storage sdp_ip;
406 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
409 if (s1->skip_media && letter != 'm')
413 get_word(buf1, sizeof(buf1), &p);
414 if (strcmp(buf1, "IN") != 0)
416 get_word(buf1, sizeof(buf1), &p);
417 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
419 get_word_sep(buf1, sizeof(buf1), "/", &p);
420 if (get_sockaddr(s, buf1, &sdp_ip))
425 get_word_sep(buf1, sizeof(buf1), "/", &p);
428 if (s->nb_streams == 0) {
429 s1->default_ip = sdp_ip;
430 s1->default_ttl = ttl;
432 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433 rtsp_st->sdp_ip = sdp_ip;
434 rtsp_st->sdp_ttl = ttl;
438 av_dict_set(&s->metadata, "title", p, 0);
441 if (s->nb_streams == 0) {
442 av_dict_set(&s->metadata, "comment", p, 0);
451 codec_type = AVMEDIA_TYPE_UNKNOWN;
452 get_word(st_type, sizeof(st_type), &p);
453 if (!strcmp(st_type, "audio")) {
454 codec_type = AVMEDIA_TYPE_AUDIO;
455 } else if (!strcmp(st_type, "video")) {
456 codec_type = AVMEDIA_TYPE_VIDEO;
457 } else if (!strcmp(st_type, "application")) {
458 codec_type = AVMEDIA_TYPE_DATA;
459 } else if (!strcmp(st_type, "text")) {
460 codec_type = AVMEDIA_TYPE_SUBTITLE;
462 if (codec_type == AVMEDIA_TYPE_UNKNOWN ||
463 !(rt->media_type_mask & (1 << codec_type)) ||
464 rt->nb_rtsp_streams >= s->max_streams
469 rtsp_st = av_mallocz(sizeof(RTSPStream));
472 rtsp_st->stream_index = -1;
473 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
475 rtsp_st->sdp_ip = s1->default_ip;
476 rtsp_st->sdp_ttl = s1->default_ttl;
478 copy_default_source_addrs(s1->default_include_source_addrs,
479 s1->nb_default_include_source_addrs,
480 &rtsp_st->include_source_addrs,
481 &rtsp_st->nb_include_source_addrs);
482 copy_default_source_addrs(s1->default_exclude_source_addrs,
483 s1->nb_default_exclude_source_addrs,
484 &rtsp_st->exclude_source_addrs,
485 &rtsp_st->nb_exclude_source_addrs);
487 get_word(buf1, sizeof(buf1), &p); /* port */
488 rtsp_st->sdp_port = atoi(buf1);
490 get_word(buf1, sizeof(buf1), &p); /* protocol */
491 if (!strcmp(buf1, "udp"))
492 rt->transport = RTSP_TRANSPORT_RAW;
493 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
494 rtsp_st->feedback = 1;
496 /* XXX: handle list of formats */
497 get_word(buf1, sizeof(buf1), &p); /* format list */
498 rtsp_st->sdp_payload_type = atoi(buf1);
500 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
501 /* no corresponding stream */
502 if (rt->transport == RTSP_TRANSPORT_RAW) {
503 if (CONFIG_RTPDEC && !rt->ts)
504 rt->ts = avpriv_mpegts_parse_open(s);
506 const RTPDynamicProtocolHandler *handler;
507 handler = ff_rtp_handler_find_by_id(
508 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
509 init_rtp_handler(handler, rtsp_st, NULL);
510 finalize_rtp_handler_init(s, rtsp_st, NULL);
512 } else if (rt->server_type == RTSP_SERVER_WMS &&
513 codec_type == AVMEDIA_TYPE_DATA) {
514 /* RTX stream, a stream that carries all the other actual
515 * audio/video streams. Don't expose this to the callers. */
517 st = avformat_new_stream(s, NULL);
520 st->id = rt->nb_rtsp_streams - 1;
521 rtsp_st->stream_index = st->index;
522 st->codecpar->codec_type = codec_type;
523 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
524 const RTPDynamicProtocolHandler *handler;
525 /* if standard payload type, we can find the codec right now */
526 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
527 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
528 st->codecpar->sample_rate > 0)
529 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
530 /* Even static payload types may need a custom depacketizer */
531 handler = ff_rtp_handler_find_by_id(
532 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
533 init_rtp_handler(handler, rtsp_st, st);
534 finalize_rtp_handler_init(s, rtsp_st, st);
536 if (rt->default_lang[0])
537 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
539 /* put a default control url */
540 av_strlcpy(rtsp_st->control_url, rt->control_uri,
541 sizeof(rtsp_st->control_url));
544 if (av_strstart(p, "control:", &p)) {
545 if (s->nb_streams == 0) {
546 if (!strncmp(p, "rtsp://", 7))
547 av_strlcpy(rt->control_uri, p,
548 sizeof(rt->control_uri));
551 /* get the control url */
552 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
554 /* XXX: may need to add full url resolution */
555 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
557 if (proto[0] == '\0') {
558 /* relative control URL */
559 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
560 av_strlcat(rtsp_st->control_url, "/",
561 sizeof(rtsp_st->control_url));
562 av_strlcat(rtsp_st->control_url, p,
563 sizeof(rtsp_st->control_url));
565 av_strlcpy(rtsp_st->control_url, p,
566 sizeof(rtsp_st->control_url));
568 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
569 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
570 get_word(buf1, sizeof(buf1), &p);
571 payload_type = atoi(buf1);
572 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
573 if (rtsp_st->stream_index >= 0) {
574 st = s->streams[rtsp_st->stream_index];
575 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
579 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
581 } else if (av_strstart(p, "fmtp:", &p) ||
582 av_strstart(p, "framesize:", &p)) {
583 // let dynamic protocol handlers have a stab at the line.
584 get_word(buf1, sizeof(buf1), &p);
585 payload_type = atoi(buf1);
586 if (s1->seen_rtpmap) {
587 parse_fmtp(s, rt, payload_type, buf);
590 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
592 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
593 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
594 get_word(buf1, sizeof(buf1), &p);
595 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
596 } else if (av_strstart(p, "range:", &p)) {
599 // this is so that seeking on a streamed file can work.
600 rtsp_parse_range_npt(p, &start, &end);
601 s->start_time = start;
602 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
603 s->duration = (end == AV_NOPTS_VALUE) ?
604 AV_NOPTS_VALUE : end - start;
605 } else if (av_strstart(p, "lang:", &p)) {
606 if (s->nb_streams > 0) {
607 get_word(buf1, sizeof(buf1), &p);
608 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609 if (rtsp_st->stream_index >= 0) {
610 st = s->streams[rtsp_st->stream_index];
611 av_dict_set(&st->metadata, "language", buf1, 0);
614 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
615 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
617 rt->transport = RTSP_TRANSPORT_RDT;
618 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
620 st = s->streams[s->nb_streams - 1];
621 st->codecpar->sample_rate = atoi(p);
622 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
624 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
625 get_word(buf1, sizeof(buf1), &p); // ignore tag
626 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
627 p += strspn(p, SPACE_CHARS);
628 if (av_strstart(p, "inline:", &p))
629 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
630 } else if (av_strstart(p, "source-filter:", &p)) {
632 get_word(buf1, sizeof(buf1), &p);
633 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
635 exclude = !strcmp(buf1, "excl");
637 get_word(buf1, sizeof(buf1), &p);
638 if (strcmp(buf1, "IN") != 0)
640 get_word(buf1, sizeof(buf1), &p);
641 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
643 // not checking that the destination address actually matches or is wildcard
644 get_word(buf1, sizeof(buf1), &p);
647 rtsp_src = av_mallocz(sizeof(*rtsp_src));
650 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
652 if (s->nb_streams == 0) {
653 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
655 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
656 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
659 if (s->nb_streams == 0) {
660 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
662 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
663 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
668 if (rt->server_type == RTSP_SERVER_WMS)
669 ff_wms_parse_sdp_a_line(s, p);
670 if (s->nb_streams > 0) {
671 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
673 if (rt->server_type == RTSP_SERVER_REAL)
674 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
676 if (rtsp_st->dynamic_handler &&
677 rtsp_st->dynamic_handler->parse_sdp_a_line)
678 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
679 rtsp_st->stream_index,
680 rtsp_st->dynamic_protocol_context, buf);
687 int ff_sdp_parse(AVFormatContext *s, const char *content)
691 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
692 * contain long SDP lines containing complete ASF Headers (several
693 * kB) or arrays of MDPR (RM stream descriptor) headers plus
694 * "rulebooks" describing their properties. Therefore, the SDP line
697 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
698 * in rtpdec_xiph.c. */
700 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
704 p += strspn(p, SPACE_CHARS);
712 /* get the content */
714 while (*p != '\n' && *p != '\r' && *p != '\0') {
715 if ((q - buf) < sizeof(buf) - 1)
720 sdp_parse_line(s, s1, letter, buf);
722 while (*p != '\n' && *p != '\0')
728 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
729 av_freep(&s1->default_include_source_addrs[i]);
730 av_freep(&s1->default_include_source_addrs);
731 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
732 av_freep(&s1->default_exclude_source_addrs[i]);
733 av_freep(&s1->default_exclude_source_addrs);
737 #endif /* CONFIG_RTPDEC */
739 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
741 RTSPState *rt = s->priv_data;
744 for (i = 0; i < rt->nb_rtsp_streams; i++) {
745 RTSPStream *rtsp_st = rt->rtsp_streams[i];
748 if (rtsp_st->transport_priv) {
750 AVFormatContext *rtpctx = rtsp_st->transport_priv;
751 av_write_trailer(rtpctx);
752 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
753 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
754 ff_rtsp_tcp_write_packet(s, rtsp_st);
755 ffio_free_dyn_buf(&rtpctx->pb);
757 avio_closep(&rtpctx->pb);
759 avformat_free_context(rtpctx);
760 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
761 ff_rdt_parse_close(rtsp_st->transport_priv);
762 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
763 ff_rtp_parse_close(rtsp_st->transport_priv);
765 rtsp_st->transport_priv = NULL;
766 if (rtsp_st->rtp_handle)
767 ffurl_close(rtsp_st->rtp_handle);
768 rtsp_st->rtp_handle = NULL;
772 /* close and free RTSP streams */
773 void ff_rtsp_close_streams(AVFormatContext *s)
775 RTSPState *rt = s->priv_data;
779 ff_rtsp_undo_setup(s, 0);
780 for (i = 0; i < rt->nb_rtsp_streams; i++) {
781 rtsp_st = rt->rtsp_streams[i];
783 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
784 if (rtsp_st->dynamic_handler->close)
785 rtsp_st->dynamic_handler->close(
786 rtsp_st->dynamic_protocol_context);
787 av_free(rtsp_st->dynamic_protocol_context);
789 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
790 av_freep(&rtsp_st->include_source_addrs[j]);
791 av_freep(&rtsp_st->include_source_addrs);
792 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
793 av_freep(&rtsp_st->exclude_source_addrs[j]);
794 av_freep(&rtsp_st->exclude_source_addrs);
799 av_freep(&rt->rtsp_streams);
801 avformat_close_input(&rt->asf_ctx);
803 if (CONFIG_RTPDEC && rt->ts)
804 avpriv_mpegts_parse_close(rt->ts);
806 av_freep(&rt->recvbuf);
809 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
811 RTSPState *rt = s->priv_data;
813 int reordering_queue_size = rt->reordering_queue_size;
814 if (reordering_queue_size < 0) {
815 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
816 reordering_queue_size = 0;
818 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
821 /* open the RTP context */
822 if (rtsp_st->stream_index >= 0)
823 st = s->streams[rtsp_st->stream_index];
825 s->ctx_flags |= AVFMTCTX_NOHEADER;
827 if (CONFIG_RTSP_MUXER && s->oformat && st) {
828 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
829 s, st, rtsp_st->rtp_handle,
830 RTSP_TCP_MAX_PACKET_SIZE,
831 rtsp_st->stream_index);
832 /* Ownership of rtp_handle is passed to the rtp mux context */
833 rtsp_st->rtp_handle = NULL;
836 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
837 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
838 return 0; // Don't need to open any parser here
839 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
840 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
841 rtsp_st->dynamic_protocol_context,
842 rtsp_st->dynamic_handler);
843 else if (CONFIG_RTPDEC)
844 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
845 rtsp_st->sdp_payload_type,
846 reordering_queue_size);
848 if (!rtsp_st->transport_priv) {
849 return AVERROR(ENOMEM);
850 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
852 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
853 rtpctx->ssrc = rtsp_st->ssrc;
854 if (rtsp_st->dynamic_handler) {
855 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
856 rtsp_st->dynamic_protocol_context,
857 rtsp_st->dynamic_handler);
859 if (rtsp_st->crypto_suite[0])
860 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
861 rtsp_st->crypto_suite,
862 rtsp_st->crypto_params);
868 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
869 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
876 q += strspn(q, SPACE_CHARS);
877 v = strtol(q, &p, 10);
881 v = strtol(p, &p, 10);
890 /* XXX: only one transport specification is parsed */
891 static void rtsp_parse_transport(AVFormatContext *s,
892 RTSPMessageHeader *reply, const char *p)
894 char transport_protocol[16];
896 char lower_transport[16];
898 RTSPTransportField *th;
901 reply->nb_transports = 0;
904 p += strspn(p, SPACE_CHARS);
908 th = &reply->transports[reply->nb_transports];
910 get_word_sep(transport_protocol, sizeof(transport_protocol),
912 if (!av_strcasecmp (transport_protocol, "rtp")) {
913 get_word_sep(profile, sizeof(profile), "/;,", &p);
914 lower_transport[0] = '\0';
915 /* rtp/avp/<protocol> */
917 get_word_sep(lower_transport, sizeof(lower_transport),
920 th->transport = RTSP_TRANSPORT_RTP;
921 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
922 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
923 /* x-pn-tng/<protocol> */
924 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
926 th->transport = RTSP_TRANSPORT_RDT;
927 } else if (!av_strcasecmp(transport_protocol, "raw")) {
928 get_word_sep(profile, sizeof(profile), "/;,", &p);
929 lower_transport[0] = '\0';
930 /* raw/raw/<protocol> */
932 get_word_sep(lower_transport, sizeof(lower_transport),
935 th->transport = RTSP_TRANSPORT_RAW;
937 if (!av_strcasecmp(lower_transport, "TCP"))
938 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
940 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
944 /* get each parameter */
945 while (*p != '\0' && *p != ',') {
946 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
947 if (!strcmp(parameter, "port")) {
950 rtsp_parse_range(&th->port_min, &th->port_max, &p);
952 } else if (!strcmp(parameter, "client_port")) {
955 rtsp_parse_range(&th->client_port_min,
956 &th->client_port_max, &p);
958 } else if (!strcmp(parameter, "server_port")) {
961 rtsp_parse_range(&th->server_port_min,
962 &th->server_port_max, &p);
964 } else if (!strcmp(parameter, "interleaved")) {
967 rtsp_parse_range(&th->interleaved_min,
968 &th->interleaved_max, &p);
970 } else if (!strcmp(parameter, "multicast")) {
971 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
972 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
973 } else if (!strcmp(parameter, "ttl")) {
977 th->ttl = strtol(p, &end, 10);
980 } else if (!strcmp(parameter, "destination")) {
983 get_word_sep(buf, sizeof(buf), ";,", &p);
984 get_sockaddr(s, buf, &th->destination);
986 } else if (!strcmp(parameter, "source")) {
989 get_word_sep(buf, sizeof(buf), ";,", &p);
990 av_strlcpy(th->source, buf, sizeof(th->source));
992 } else if (!strcmp(parameter, "mode")) {
995 get_word_sep(buf, sizeof(buf), ";, ", &p);
996 if (!strcmp(buf, "record") ||
997 !strcmp(buf, "receive"))
1002 while (*p != ';' && *p != '\0' && *p != ',')
1010 reply->nb_transports++;
1011 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1016 static void handle_rtp_info(RTSPState *rt, const char *url,
1017 uint32_t seq, uint32_t rtptime)
1020 if (!rtptime || !url[0])
1022 if (rt->transport != RTSP_TRANSPORT_RTP)
1024 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1025 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1026 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1029 if (!strcmp(rtsp_st->control_url, url)) {
1030 rtpctx->base_timestamp = rtptime;
1036 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1039 char key[20], value[1024], url[1024] = "";
1040 uint32_t seq = 0, rtptime = 0;
1043 p += strspn(p, SPACE_CHARS);
1046 get_word_sep(key, sizeof(key), "=", &p);
1050 get_word_sep(value, sizeof(value), ";, ", &p);
1052 if (!strcmp(key, "url"))
1053 av_strlcpy(url, value, sizeof(url));
1054 else if (!strcmp(key, "seq"))
1055 seq = strtoul(value, NULL, 10);
1056 else if (!strcmp(key, "rtptime"))
1057 rtptime = strtoul(value, NULL, 10);
1059 handle_rtp_info(rt, url, seq, rtptime);
1068 handle_rtp_info(rt, url, seq, rtptime);
1071 void ff_rtsp_parse_line(AVFormatContext *s,
1072 RTSPMessageHeader *reply, const char *buf,
1073 RTSPState *rt, const char *method)
1077 /* NOTE: we do case independent match for broken servers */
1079 if (av_stristart(p, "Session:", &p)) {
1081 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1082 if (av_stristart(p, ";timeout=", &p) &&
1083 (t = strtol(p, NULL, 10)) > 0) {
1086 } else if (av_stristart(p, "Content-Length:", &p)) {
1087 reply->content_length = strtol(p, NULL, 10);
1088 } else if (av_stristart(p, "Transport:", &p)) {
1089 rtsp_parse_transport(s, reply, p);
1090 } else if (av_stristart(p, "CSeq:", &p)) {
1091 reply->seq = strtol(p, NULL, 10);
1092 } else if (av_stristart(p, "Range:", &p)) {
1093 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1094 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1095 p += strspn(p, SPACE_CHARS);
1096 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1097 } else if (av_stristart(p, "Server:", &p)) {
1098 p += strspn(p, SPACE_CHARS);
1099 av_strlcpy(reply->server, p, sizeof(reply->server));
1100 } else if (av_stristart(p, "Notice:", &p) ||
1101 av_stristart(p, "X-Notice:", &p)) {
1102 reply->notice = strtol(p, NULL, 10);
1103 } else if (av_stristart(p, "Location:", &p)) {
1104 p += strspn(p, SPACE_CHARS);
1105 av_strlcpy(reply->location, p , sizeof(reply->location));
1106 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1107 p += strspn(p, SPACE_CHARS);
1108 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1109 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1110 p += strspn(p, SPACE_CHARS);
1111 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1112 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1113 p += strspn(p, SPACE_CHARS);
1114 if (method && !strcmp(method, "DESCRIBE"))
1115 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1116 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1117 p += strspn(p, SPACE_CHARS);
1118 if (method && !strcmp(method, "PLAY"))
1119 rtsp_parse_rtp_info(rt, p);
1120 } else if (av_stristart(p, "Public:", &p) && rt) {
1121 if (strstr(p, "GET_PARAMETER") &&
1122 method && !strcmp(method, "OPTIONS"))
1123 rt->get_parameter_supported = 1;
1124 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1125 p += strspn(p, SPACE_CHARS);
1126 rt->accept_dynamic_rate = atoi(p);
1127 } else if (av_stristart(p, "Content-Type:", &p)) {
1128 p += strspn(p, SPACE_CHARS);
1129 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1133 /* skip a RTP/TCP interleaved packet */
1134 void ff_rtsp_skip_packet(AVFormatContext *s)
1136 RTSPState *rt = s->priv_data;
1140 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1143 len = AV_RB16(buf + 1);
1145 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1150 if (len1 > sizeof(buf))
1152 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1159 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1160 unsigned char **content_ptr,
1161 int return_on_interleaved_data, const char *method)
1163 RTSPState *rt = s->priv_data;
1164 char buf[4096], buf1[1024], *q;
1167 int ret, content_length, line_count = 0, request = 0;
1168 unsigned char *content = NULL;
1174 memset(reply, 0, sizeof(*reply));
1176 /* parse reply (XXX: use buffers) */
1177 rt->last_reply[0] = '\0';
1181 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1182 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1187 if (ch == '$' && q == buf) {
1188 if (return_on_interleaved_data) {
1191 ff_rtsp_skip_packet(s);
1192 } else if (ch != '\r') {
1193 if ((q - buf) < sizeof(buf) - 1)
1199 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1201 /* test if last line */
1205 if (line_count == 0) {
1206 /* get reply code */
1207 get_word(buf1, sizeof(buf1), &p);
1208 if (!strncmp(buf1, "RTSP/", 5)) {
1209 get_word(buf1, sizeof(buf1), &p);
1210 reply->status_code = atoi(buf1);
1211 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1213 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1214 get_word(buf1, sizeof(buf1), &p); // object
1218 ff_rtsp_parse_line(s, reply, p, rt, method);
1219 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1220 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1225 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1226 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1228 content_length = reply->content_length;
1229 if (content_length > 0) {
1230 /* leave some room for a trailing '\0' (useful for simple parsing) */
1231 content = av_malloc(content_length + 1);
1233 return AVERROR(ENOMEM);
1234 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1235 content[content_length] = '\0';
1238 *content_ptr = content;
1244 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1245 const char* ptr = buf;
1247 if (!strcmp(reply->reason, "OPTIONS")) {
1248 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1250 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1251 if (reply->session_id[0])
1252 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1255 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1257 av_strlcat(buf, "\r\n", sizeof(buf));
1259 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1260 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1263 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1265 rt->last_cmd_time = av_gettime_relative();
1266 /* Even if the request from the server had data, it is not the data
1267 * that the caller wants or expects. The memory could also be leaked
1268 * if the actual following reply has content data. */
1270 av_freep(content_ptr);
1271 /* If method is set, this is called from ff_rtsp_send_cmd,
1272 * where a reply to exactly this request is awaited. For
1273 * callers from within packet receiving, we just want to
1274 * return to the caller and go back to receiving packets. */
1280 if (rt->seq != reply->seq) {
1281 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1282 rt->seq, reply->seq);
1286 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1287 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1288 reply->notice == 2306 /* Continuous Feed Terminated */) {
1289 rt->state = RTSP_STATE_IDLE;
1290 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1291 return AVERROR(EIO); /* data or server error */
1292 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1293 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1294 return AVERROR(EPERM);
1300 * Send a command to the RTSP server without waiting for the reply.
1302 * @param s RTSP (de)muxer context
1303 * @param method the method for the request
1304 * @param url the target url for the request
1305 * @param headers extra header lines to include in the request
1306 * @param send_content if non-null, the data to send as request body content
1307 * @param send_content_length the length of the send_content data, or 0 if
1308 * send_content is null
1310 * @return zero if success, nonzero otherwise
1312 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1313 const char *method, const char *url,
1314 const char *headers,
1315 const unsigned char *send_content,
1316 int send_content_length)
1318 RTSPState *rt = s->priv_data;
1319 char buf[4096], *out_buf;
1320 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1322 if (!rt->rtsp_hd_out)
1323 return AVERROR(ENOTCONN);
1325 /* Add in RTSP headers */
1328 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1330 av_strlcat(buf, headers, sizeof(buf));
1331 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1332 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1333 if (rt->session_id[0] != '\0' && (!headers ||
1334 !strstr(headers, "\nIf-Match:"))) {
1335 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1338 char *str = ff_http_auth_create_response(&rt->auth_state,
1339 rt->auth, url, method);
1341 av_strlcat(buf, str, sizeof(buf));
1344 if (send_content_length > 0 && send_content)
1345 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1346 av_strlcat(buf, "\r\n", sizeof(buf));
1348 /* base64 encode rtsp if tunneling */
1349 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1350 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1351 out_buf = base64buf;
1354 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1356 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1357 if (send_content_length > 0 && send_content) {
1358 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1359 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1360 return AVERROR_PATCHWELCOME;
1362 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1364 rt->last_cmd_time = av_gettime_relative();
1369 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1370 const char *url, const char *headers)
1372 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1375 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1376 const char *headers, RTSPMessageHeader *reply,
1377 unsigned char **content_ptr)
1379 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1380 content_ptr, NULL, 0);
1383 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1384 const char *method, const char *url,
1386 RTSPMessageHeader *reply,
1387 unsigned char **content_ptr,
1388 const unsigned char *send_content,
1389 int send_content_length)
1391 RTSPState *rt = s->priv_data;
1392 HTTPAuthType cur_auth_type;
1393 int ret, attempts = 0;
1396 cur_auth_type = rt->auth_state.auth_type;
1397 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1399 send_content_length)))
1402 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1406 if (reply->status_code == 401 &&
1407 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1408 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1411 if (reply->status_code > 400){
1412 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1416 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1422 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1423 int lower_transport, const char *real_challenge)
1425 RTSPState *rt = s->priv_data;
1426 int rtx = 0, j, i, err, interleave = 0, port_off;
1427 RTSPStream *rtsp_st;
1428 RTSPMessageHeader reply1, *reply = &reply1;
1430 const char *trans_pref;
1432 if (rt->transport == RTSP_TRANSPORT_RDT)
1433 trans_pref = "x-pn-tng";
1434 else if (rt->transport == RTSP_TRANSPORT_RAW)
1435 trans_pref = "RAW/RAW";
1437 trans_pref = "RTP/AVP";
1439 /* default timeout: 1 minute */
1442 /* Choose a random starting offset within the first half of the
1443 * port range, to allow for a number of ports to try even if the offset
1444 * happens to be at the end of the random range. */
1445 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1446 /* even random offset */
1447 port_off -= port_off & 0x01;
1449 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1450 char transport[2048];
1453 * WMS serves all UDP data over a single connection, the RTX, which
1454 * isn't necessarily the first in the SDP but has to be the first
1455 * to be set up, else the second/third SETUP will fail with a 461.
1457 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1458 rt->server_type == RTSP_SERVER_WMS) {
1461 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1462 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1464 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1468 if (rtx == rt->nb_rtsp_streams)
1469 return -1; /* no RTX found */
1470 rtsp_st = rt->rtsp_streams[rtx];
1472 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1474 rtsp_st = rt->rtsp_streams[i];
1477 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1480 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1481 port = reply->transports[0].client_port_min;
1485 /* first try in specified port range */
1486 while (j <= rt->rtp_port_max) {
1487 AVDictionary *opts = map_to_opts(rt);
1489 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1490 "?localport=%d", j);
1491 /* we will use two ports per rtp stream (rtp and rtcp) */
1493 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1494 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1496 av_dict_free(&opts);
1501 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1506 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1508 snprintf(transport, sizeof(transport) - 1,
1509 "%s/UDP;", trans_pref);
1510 if (rt->server_type != RTSP_SERVER_REAL)
1511 av_strlcat(transport, "unicast;", sizeof(transport));
1512 av_strlcatf(transport, sizeof(transport),
1513 "client_port=%d", port);
1514 if (rt->transport == RTSP_TRANSPORT_RTP &&
1515 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1516 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1520 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1521 /* For WMS streams, the application streams are only used for
1522 * UDP. When trying to set it up for TCP streams, the server
1523 * will return an error. Therefore, we skip those streams. */
1524 if (rt->server_type == RTSP_SERVER_WMS &&
1525 (rtsp_st->stream_index < 0 ||
1526 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1529 snprintf(transport, sizeof(transport) - 1,
1530 "%s/TCP;", trans_pref);
1531 if (rt->transport != RTSP_TRANSPORT_RDT)
1532 av_strlcat(transport, "unicast;", sizeof(transport));
1533 av_strlcatf(transport, sizeof(transport),
1534 "interleaved=%d-%d",
1535 interleave, interleave + 1);
1539 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1540 snprintf(transport, sizeof(transport) - 1,
1541 "%s/UDP;multicast", trans_pref);
1544 av_strlcat(transport, ";mode=record", sizeof(transport));
1545 } else if (rt->server_type == RTSP_SERVER_REAL ||
1546 rt->server_type == RTSP_SERVER_WMS)
1547 av_strlcat(transport, ";mode=play", sizeof(transport));
1548 snprintf(cmd, sizeof(cmd),
1549 "Transport: %s\r\n",
1551 if (rt->accept_dynamic_rate)
1552 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1553 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1554 char real_res[41], real_csum[9];
1555 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1557 av_strlcatf(cmd, sizeof(cmd),
1559 "RealChallenge2: %s, sd=%s\r\n",
1560 rt->session_id, real_res, real_csum);
1562 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1563 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1566 } else if (reply->status_code != RTSP_STATUS_OK ||
1567 reply->nb_transports != 1) {
1568 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1572 /* XXX: same protocol for all streams is required */
1574 if (reply->transports[0].lower_transport != rt->lower_transport ||
1575 reply->transports[0].transport != rt->transport) {
1576 err = AVERROR_INVALIDDATA;
1580 rt->lower_transport = reply->transports[0].lower_transport;
1581 rt->transport = reply->transports[0].transport;
1584 /* Fail if the server responded with another lower transport mode
1585 * than what we requested. */
1586 if (reply->transports[0].lower_transport != lower_transport) {
1587 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1588 err = AVERROR_INVALIDDATA;
1592 switch(reply->transports[0].lower_transport) {
1593 case RTSP_LOWER_TRANSPORT_TCP:
1594 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1595 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1598 case RTSP_LOWER_TRANSPORT_UDP: {
1599 char url[1024], options[30] = "";
1600 const char *peer = host;
1602 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1603 av_strlcpy(options, "?connect=1", sizeof(options));
1604 /* Use source address if specified */
1605 if (reply->transports[0].source[0])
1606 peer = reply->transports[0].source;
1607 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1608 reply->transports[0].server_port_min, "%s", options);
1609 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1610 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1611 err = AVERROR_INVALIDDATA;
1616 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1617 char url[1024], namebuf[50], optbuf[20] = "";
1618 struct sockaddr_storage addr;
1620 AVDictionary *opts = map_to_opts(rt);
1622 if (reply->transports[0].destination.ss_family) {
1623 addr = reply->transports[0].destination;
1624 port = reply->transports[0].port_min;
1625 ttl = reply->transports[0].ttl;
1627 addr = rtsp_st->sdp_ip;
1628 port = rtsp_st->sdp_port;
1629 ttl = rtsp_st->sdp_ttl;
1632 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1633 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1634 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1635 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1636 port, "%s", optbuf);
1637 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1638 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1639 av_dict_free(&opts);
1642 err = AVERROR_INVALIDDATA;
1649 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1653 if (rt->nb_rtsp_streams && reply->timeout > 0)
1654 rt->timeout = reply->timeout;
1656 if (rt->server_type == RTSP_SERVER_REAL)
1657 rt->need_subscription = 1;
1662 ff_rtsp_undo_setup(s, 0);
1666 void ff_rtsp_close_connections(AVFormatContext *s)
1668 RTSPState *rt = s->priv_data;
1669 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1670 ffurl_close(rt->rtsp_hd);
1671 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1674 int ff_rtsp_connect(AVFormatContext *s)
1676 RTSPState *rt = s->priv_data;
1677 char proto[128], host[1024], path[1024];
1678 char tcpname[1024], cmd[2048], auth[128];
1679 const char *lower_rtsp_proto = "tcp";
1680 int port, err, tcp_fd;
1681 RTSPMessageHeader reply1, *reply = &reply1;
1682 int lower_transport_mask = 0;
1683 int default_port = RTSP_DEFAULT_PORT;
1684 int https_tunnel = 0;
1685 char real_challenge[64] = "";
1686 struct sockaddr_storage peer;
1687 socklen_t peer_len = sizeof(peer);
1689 if (rt->rtp_port_max < rt->rtp_port_min) {
1690 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1691 "than min port %d\n", rt->rtp_port_max,
1693 return AVERROR(EINVAL);
1696 if (!ff_network_init())
1697 return AVERROR(EIO);
1699 if (s->max_delay < 0) /* Not set by the caller */
1700 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1702 rt->control_transport = RTSP_MODE_PLAIN;
1703 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1704 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1705 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1706 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1707 rt->control_transport = RTSP_MODE_TUNNEL;
1709 /* Only pass through valid flags from here */
1710 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1713 memset(&reply1, 0, sizeof(reply1));
1714 /* extract hostname and port */
1715 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1716 host, sizeof(host), &port, path, sizeof(path), s->url);
1718 if (!strcmp(proto, "rtsps")) {
1719 lower_rtsp_proto = "tls";
1720 default_port = RTSPS_DEFAULT_PORT;
1721 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1725 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1728 port = default_port;
1730 lower_transport_mask = rt->lower_transport_mask;
1732 if (!lower_transport_mask)
1733 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1736 /* Only UDP or TCP - UDP multicast isn't supported. */
1737 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1738 (1 << RTSP_LOWER_TRANSPORT_TCP);
1739 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1740 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1741 "only UDP and TCP are supported for output.\n");
1742 err = AVERROR(EINVAL);
1747 /* Construct the URI used in request; this is similar to s->url,
1748 * but with authentication credentials removed and RTSP specific options
1750 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1751 host, port, "%s", path);
1753 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1754 /* set up initial handshake for tunneling */
1755 char httpname[1024];
1756 char sessioncookie[17];
1758 AVDictionary *options = NULL;
1760 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1762 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1763 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1764 av_get_random_seed(), av_get_random_seed());
1767 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1768 &s->interrupt_callback) < 0) {
1773 /* generate GET headers */
1774 snprintf(headers, sizeof(headers),
1775 "x-sessioncookie: %s\r\n"
1776 "Accept: application/x-rtsp-tunnelled\r\n"
1777 "Pragma: no-cache\r\n"
1778 "Cache-Control: no-cache\r\n",
1780 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1782 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1783 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1784 if (!rt->rtsp_hd->protocol_whitelist) {
1785 err = AVERROR(ENOMEM);
1790 /* complete the connection */
1791 if (ffurl_connect(rt->rtsp_hd, &options)) {
1792 av_dict_free(&options);
1798 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1799 &s->interrupt_callback) < 0 ) {
1804 /* generate POST headers */
1805 snprintf(headers, sizeof(headers),
1806 "x-sessioncookie: %s\r\n"
1807 "Content-Type: application/x-rtsp-tunnelled\r\n"
1808 "Pragma: no-cache\r\n"
1809 "Cache-Control: no-cache\r\n"
1810 "Content-Length: 32767\r\n"
1811 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1813 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1814 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1815 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1817 /* Initialize the authentication state for the POST session. The HTTP
1818 * protocol implementation doesn't properly handle multi-pass
1819 * authentication for POST requests, since it would require one of
1821 * - implementing Expect: 100-continue, which many HTTP servers
1822 * don't support anyway, even less the RTSP servers that do HTTP
1824 * - sending the whole POST data until getting a 401 reply specifying
1825 * what authentication method to use, then resending all that data
1826 * - waiting for potential 401 replies directly after sending the
1827 * POST header (waiting for some unspecified time)
1828 * Therefore, we copy the full auth state, which works for both basic
1829 * and digest. (For digest, we would have to synchronize the nonce
1830 * count variable between the two sessions, if we'd do more requests
1831 * with the original session, though.)
1833 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1835 /* complete the connection */
1836 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1837 av_dict_free(&options);
1841 av_dict_free(&options);
1844 /* open the tcp connection */
1845 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1847 "?timeout=%d", rt->stimeout);
1848 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1849 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1853 rt->rtsp_hd_out = rt->rtsp_hd;
1857 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1862 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1863 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1864 NULL, 0, NI_NUMERICHOST);
1867 /* request options supported by the server; this also detects server
1869 for (rt->server_type = RTSP_SERVER_RTP;;) {
1871 if (rt->server_type == RTSP_SERVER_REAL)
1874 * The following entries are required for proper
1875 * streaming from a Realmedia server. They are
1876 * interdependent in some way although we currently
1877 * don't quite understand how. Values were copied
1878 * from mplayer SVN r23589.
1879 * ClientChallenge is a 16-byte ID in hex
1880 * CompanyID is a 16-byte ID in base64
1882 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1883 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1884 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1885 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1887 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1888 if (reply->status_code != RTSP_STATUS_OK) {
1889 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1893 /* detect server type if not standard-compliant RTP */
1894 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1895 rt->server_type = RTSP_SERVER_REAL;
1897 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1898 rt->server_type = RTSP_SERVER_WMS;
1899 } else if (rt->server_type == RTSP_SERVER_REAL)
1900 strcpy(real_challenge, reply->real_challenge);
1904 if (CONFIG_RTSP_DEMUXER && s->iformat)
1905 err = ff_rtsp_setup_input_streams(s, reply);
1906 else if (CONFIG_RTSP_MUXER)
1907 err = ff_rtsp_setup_output_streams(s, host);
1914 int lower_transport = ff_log2_tab[lower_transport_mask &
1915 ~(lower_transport_mask - 1)];
1917 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1918 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1919 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1921 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1922 rt->server_type == RTSP_SERVER_REAL ?
1923 real_challenge : NULL);
1926 lower_transport_mask &= ~(1 << lower_transport);
1927 if (lower_transport_mask == 0 && err == 1) {
1928 err = AVERROR(EPROTONOSUPPORT);
1933 rt->lower_transport_mask = lower_transport_mask;
1934 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1935 rt->state = RTSP_STATE_IDLE;
1936 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1939 ff_rtsp_close_streams(s);
1940 ff_rtsp_close_connections(s);
1941 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1942 char *new_url = av_strdup(reply->location);
1944 err = AVERROR(ENOMEM);
1947 ff_format_set_url(s, new_url);
1948 rt->session_id[0] = '\0';
1949 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1958 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1961 static int parse_rtsp_message(AVFormatContext *s)
1963 RTSPState *rt = s->priv_data;
1966 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1967 if (rt->state == RTSP_STATE_STREAMING) {
1968 if (!ff_rtsp_parse_streaming_commands(s))
1971 av_log(s, AV_LOG_WARNING,
1972 "Unable to answer to TEARDOWN\n");
1976 RTSPMessageHeader reply;
1977 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1980 /* XXX: parse message */
1981 if (rt->state != RTSP_STATE_STREAMING)
1988 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1989 uint8_t *buf, int buf_size, int64_t wait_end)
1991 RTSPState *rt = s->priv_data;
1992 RTSPStream *rtsp_st;
1993 int n, i, ret, timeout_cnt = 0;
1994 struct pollfd *p = rt->p;
1995 int *fds = NULL, fdsnum, fdsidx;
1998 p = rt->p = av_malloc_array(2 * (rt->nb_rtsp_streams + 1), sizeof(struct pollfd));
2000 return AVERROR(ENOMEM);
2003 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
2004 p[rt->max_p++].events = POLLIN;
2006 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2007 rtsp_st = rt->rtsp_streams[i];
2008 if (rtsp_st->rtp_handle) {
2009 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2011 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2015 av_log(s, AV_LOG_ERROR,
2016 "Number of fds %d not supported\n", fdsnum);
2017 return AVERROR_INVALIDDATA;
2019 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2020 p[rt->max_p].fd = fds[fdsidx];
2021 p[rt->max_p++].events = POLLIN;
2029 if (ff_check_interrupt(&s->interrupt_callback))
2030 return AVERROR_EXIT;
2031 if (wait_end && wait_end - av_gettime_relative() < 0)
2032 return AVERROR(EAGAIN);
2033 n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2035 int j = rt->rtsp_hd ? 1 : 0;
2037 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2038 rtsp_st = rt->rtsp_streams[i];
2039 if (rtsp_st->rtp_handle) {
2040 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2041 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2043 *prtsp_st = rtsp_st;
2050 #if CONFIG_RTSP_DEMUXER
2051 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2052 if ((ret = parse_rtsp_message(s)) < 0) {
2057 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2058 return AVERROR(ETIMEDOUT);
2059 } else if (n < 0 && errno != EINTR)
2060 return AVERROR(errno);
2064 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2065 const uint8_t *buf, int len)
2067 RTSPState *rt = s->priv_data;
2071 if (rt->nb_rtsp_streams == 1) {
2072 *rtsp_st = rt->rtsp_streams[0];
2075 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2076 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2078 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2079 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2082 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2083 *rtsp_st = rt->rtsp_streams[i];
2090 av_log(s, AV_LOG_WARNING,
2091 "Unable to pick stream for packet - SSRC not known for "
2093 return AVERROR(EAGAIN);
2096 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2097 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2098 *rtsp_st = rt->rtsp_streams[i];
2104 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2105 return AVERROR(EAGAIN);
2108 static int read_packet(AVFormatContext *s,
2109 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2112 RTSPState *rt = s->priv_data;
2115 switch(rt->lower_transport) {
2117 #if CONFIG_RTSP_DEMUXER
2118 case RTSP_LOWER_TRANSPORT_TCP:
2119 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2122 case RTSP_LOWER_TRANSPORT_UDP:
2123 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2124 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2125 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2126 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2128 case RTSP_LOWER_TRANSPORT_CUSTOM:
2129 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2130 wait_end && wait_end < av_gettime_relative())
2131 len = AVERROR(EAGAIN);
2133 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2134 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2135 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2136 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2146 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2148 RTSPState *rt = s->priv_data;
2150 RTSPStream *rtsp_st, *first_queue_st = NULL;
2151 int64_t wait_end = 0;
2153 if (rt->nb_byes == rt->nb_rtsp_streams)
2156 /* get next frames from the same RTP packet */
2157 if (rt->cur_transport_priv) {
2158 if (rt->transport == RTSP_TRANSPORT_RDT) {
2159 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2160 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2161 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2162 } else if (CONFIG_RTPDEC && rt->ts) {
2163 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2165 rt->recvbuf_pos += ret;
2166 ret = rt->recvbuf_pos < rt->recvbuf_len;
2171 rt->cur_transport_priv = NULL;
2173 } else if (ret == 1) {
2176 rt->cur_transport_priv = NULL;
2180 if (rt->transport == RTSP_TRANSPORT_RTP) {
2182 int64_t first_queue_time = 0;
2183 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2184 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2188 queue_time = ff_rtp_queued_packet_time(rtpctx);
2189 if (queue_time && (queue_time - first_queue_time < 0 ||
2190 !first_queue_time)) {
2191 first_queue_time = queue_time;
2192 first_queue_st = rt->rtsp_streams[i];
2195 if (first_queue_time) {
2196 wait_end = first_queue_time + s->max_delay;
2199 first_queue_st = NULL;
2203 /* read next RTP packet */
2205 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2207 return AVERROR(ENOMEM);
2210 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2211 if (len == AVERROR(EAGAIN) && first_queue_st &&
2212 rt->transport == RTSP_TRANSPORT_RTP) {
2213 av_log(s, AV_LOG_WARNING,
2214 "max delay reached. need to consume packet\n");
2215 rtsp_st = first_queue_st;
2216 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2222 if (rt->transport == RTSP_TRANSPORT_RDT) {
2223 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2224 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2225 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2226 if (rtsp_st->feedback) {
2227 AVIOContext *pb = NULL;
2228 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2230 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2233 /* Either bad packet, or a RTCP packet. Check if the
2234 * first_rtcp_ntp_time field was initialized. */
2235 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2236 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2237 /* first_rtcp_ntp_time has been initialized for this stream,
2238 * copy the same value to all other uninitialized streams,
2239 * in order to map their timestamp origin to the same ntp time
2242 AVStream *st = NULL;
2243 if (rtsp_st->stream_index >= 0)
2244 st = s->streams[rtsp_st->stream_index];
2245 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2246 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2247 AVStream *st2 = NULL;
2248 if (rt->rtsp_streams[i]->stream_index >= 0)
2249 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2250 if (rtpctx2 && st && st2 &&
2251 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2252 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2253 rtpctx2->rtcp_ts_offset = av_rescale_q(
2254 rtpctx->rtcp_ts_offset, st->time_base,
2258 // Make real NTP start time available in AVFormatContext
2259 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2260 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2262 s->start_time_realtime -=
2263 av_rescale (rtpctx->rtcp_ts_offset,
2264 (uint64_t) rtpctx->st->time_base.num * 1000000,
2265 rtpctx->st->time_base.den);
2269 if (ret == -RTCP_BYE) {
2272 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2273 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2275 if (rt->nb_byes == rt->nb_rtsp_streams)
2279 } else if (CONFIG_RTPDEC && rt->ts) {
2280 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2283 rt->recvbuf_len = len;
2284 rt->recvbuf_pos = ret;
2285 rt->cur_transport_priv = rt->ts;
2292 return AVERROR_INVALIDDATA;
2298 /* more packets may follow, so we save the RTP context */
2299 rt->cur_transport_priv = rtsp_st->transport_priv;
2303 #endif /* CONFIG_RTPDEC */
2305 #if CONFIG_SDP_DEMUXER
2306 static int sdp_probe(const AVProbeData *p1)
2308 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2310 /* we look for a line beginning "c=IN IP" */
2311 while (p < p_end && *p != '\0') {
2312 if (sizeof("c=IN IP") - 1 < p_end - p &&
2313 av_strstart(p, "c=IN IP", NULL))
2314 return AVPROBE_SCORE_EXTENSION;
2316 while (p < p_end - 1 && *p != '\n') p++;
2325 static void append_source_addrs(char *buf, int size, const char *name,
2326 int count, struct RTSPSource **addrs)
2331 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2332 for (i = 1; i < count; i++)
2333 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2336 static int sdp_read_header(AVFormatContext *s)
2338 RTSPState *rt = s->priv_data;
2339 RTSPStream *rtsp_st;
2344 if (!ff_network_init())
2345 return AVERROR(EIO);
2347 if (s->max_delay < 0) /* Not set by the caller */
2348 s->max_delay = DEFAULT_REORDERING_DELAY;
2349 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2350 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2352 /* read the whole sdp file */
2353 /* XXX: better loading */
2354 content = av_malloc(SDP_MAX_SIZE);
2356 return AVERROR(ENOMEM);
2357 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2360 return AVERROR_INVALIDDATA;
2362 content[size] ='\0';
2364 err = ff_sdp_parse(s, content);
2368 /* open each RTP stream */
2369 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2371 rtsp_st = rt->rtsp_streams[i];
2373 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2374 AVDictionary *opts = map_to_opts(rt);
2376 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2377 sizeof(rtsp_st->sdp_ip),
2378 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2380 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2382 av_dict_free(&opts);
2385 ff_url_join(url, sizeof(url), "rtp", NULL,
2386 namebuf, rtsp_st->sdp_port,
2387 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2388 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2389 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2390 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2392 append_source_addrs(url, sizeof(url), "sources",
2393 rtsp_st->nb_include_source_addrs,
2394 rtsp_st->include_source_addrs);
2395 append_source_addrs(url, sizeof(url), "block",
2396 rtsp_st->nb_exclude_source_addrs,
2397 rtsp_st->exclude_source_addrs);
2398 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2399 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2401 av_dict_free(&opts);
2404 err = AVERROR_INVALIDDATA;
2408 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2413 ff_rtsp_close_streams(s);
2418 static int sdp_read_close(AVFormatContext *s)
2420 ff_rtsp_close_streams(s);
2425 static const AVClass sdp_demuxer_class = {
2426 .class_name = "SDP demuxer",
2427 .item_name = av_default_item_name,
2428 .option = sdp_options,
2429 .version = LIBAVUTIL_VERSION_INT,
2432 AVInputFormat ff_sdp_demuxer = {
2434 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2435 .priv_data_size = sizeof(RTSPState),
2436 .read_probe = sdp_probe,
2437 .read_header = sdp_read_header,
2438 .read_packet = ff_rtsp_fetch_packet,
2439 .read_close = sdp_read_close,
2440 .priv_class = &sdp_demuxer_class,
2442 #endif /* CONFIG_SDP_DEMUXER */
2444 #if CONFIG_RTP_DEMUXER
2445 static int rtp_probe(const AVProbeData *p)
2447 if (av_strstart(p->filename, "rtp:", NULL))
2448 return AVPROBE_SCORE_MAX;
2452 static int rtp_read_header(AVFormatContext *s)
2454 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2455 char host[500], filters_buf[1000];
2457 URLContext* in = NULL;
2459 AVCodecParameters *par = NULL;
2460 struct sockaddr_storage addr;
2462 socklen_t addrlen = sizeof(addr);
2463 RTSPState *rt = s->priv_data;
2467 if (!ff_network_init())
2468 return AVERROR(EIO);
2470 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2471 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2476 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2477 if (ret == AVERROR(EAGAIN))
2482 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2486 if ((recvbuf[0] & 0xc0) != 0x80) {
2487 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2492 if (RTP_PT_IS_RTCP(recvbuf[1]))
2495 payload_type = recvbuf[1] & 0x7f;
2498 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2502 par = avcodec_parameters_alloc();
2504 ret = AVERROR(ENOMEM);
2508 if (ff_rtp_get_codec_info(par, payload_type)) {
2509 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2510 "without an SDP file describing it\n",
2514 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2515 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2516 "properly you need an SDP file "
2520 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2523 av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
2524 av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2525 addr.ss_family == AF_INET ? 4 : 6, host);
2527 p = strchr(s->url, '?');
2529 static const char filters[][2][8] = { { "sources", "incl" },
2530 { "block", "excl" } };
2533 for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2534 if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2536 while ((q = strchr(q, ',')) != NULL)
2538 av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2540 addr.ss_family == AF_INET ? 4 : 6, host,
2546 av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2547 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2548 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2549 port, payload_type);
2550 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2551 if (!av_bprint_is_complete(&sdp))
2553 avcodec_parameters_free(&par);
2555 ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2558 /* sdp_read_header initializes this again */
2561 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2563 ret = sdp_read_header(s);
2565 av_bprint_finalize(&sdp, NULL);
2569 ret = AVERROR(ENOMEM);
2570 av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2571 av_bprint_finalize(&sdp, NULL);
2573 avcodec_parameters_free(&par);
2580 static const AVClass rtp_demuxer_class = {
2581 .class_name = "RTP demuxer",
2582 .item_name = av_default_item_name,
2583 .option = rtp_options,
2584 .version = LIBAVUTIL_VERSION_INT,
2587 AVInputFormat ff_rtp_demuxer = {
2589 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2590 .priv_data_size = sizeof(RTSPState),
2591 .read_probe = rtp_probe,
2592 .read_header = rtp_read_header,
2593 .read_packet = ff_rtsp_fetch_packet,
2594 .read_close = sdp_read_close,
2595 .flags = AVFMT_NOFILE,
2596 .priv_class = &rtp_demuxer_class,
2598 #endif /* CONFIG_RTP_DEMUXER */