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 ffurl_closep(&rtsp_st->rtp_handle);
770 /* close and free RTSP streams */
771 void ff_rtsp_close_streams(AVFormatContext *s)
773 RTSPState *rt = s->priv_data;
777 ff_rtsp_undo_setup(s, 0);
778 for (i = 0; i < rt->nb_rtsp_streams; i++) {
779 rtsp_st = rt->rtsp_streams[i];
781 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
782 if (rtsp_st->dynamic_handler->close)
783 rtsp_st->dynamic_handler->close(
784 rtsp_st->dynamic_protocol_context);
785 av_free(rtsp_st->dynamic_protocol_context);
787 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
788 av_freep(&rtsp_st->include_source_addrs[j]);
789 av_freep(&rtsp_st->include_source_addrs);
790 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
791 av_freep(&rtsp_st->exclude_source_addrs[j]);
792 av_freep(&rtsp_st->exclude_source_addrs);
797 av_freep(&rt->rtsp_streams);
799 avformat_close_input(&rt->asf_ctx);
801 if (CONFIG_RTPDEC && rt->ts)
802 avpriv_mpegts_parse_close(rt->ts);
804 av_freep(&rt->recvbuf);
807 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
809 RTSPState *rt = s->priv_data;
811 int reordering_queue_size = rt->reordering_queue_size;
812 if (reordering_queue_size < 0) {
813 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
814 reordering_queue_size = 0;
816 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
819 /* open the RTP context */
820 if (rtsp_st->stream_index >= 0)
821 st = s->streams[rtsp_st->stream_index];
823 s->ctx_flags |= AVFMTCTX_NOHEADER;
825 if (CONFIG_RTSP_MUXER && s->oformat && st) {
826 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
827 s, st, rtsp_st->rtp_handle,
828 RTSP_TCP_MAX_PACKET_SIZE,
829 rtsp_st->stream_index);
830 /* Ownership of rtp_handle is passed to the rtp mux context */
831 rtsp_st->rtp_handle = NULL;
834 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
835 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
836 return 0; // Don't need to open any parser here
837 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
838 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
839 rtsp_st->dynamic_protocol_context,
840 rtsp_st->dynamic_handler);
841 else if (CONFIG_RTPDEC)
842 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
843 rtsp_st->sdp_payload_type,
844 reordering_queue_size);
846 if (!rtsp_st->transport_priv) {
847 return AVERROR(ENOMEM);
848 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP &&
850 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
851 rtpctx->ssrc = rtsp_st->ssrc;
852 if (rtsp_st->dynamic_handler) {
853 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
854 rtsp_st->dynamic_protocol_context,
855 rtsp_st->dynamic_handler);
857 if (rtsp_st->crypto_suite[0])
858 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
859 rtsp_st->crypto_suite,
860 rtsp_st->crypto_params);
866 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
867 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
874 q += strspn(q, SPACE_CHARS);
875 v = strtol(q, &p, 10);
879 v = strtol(p, &p, 10);
888 /* XXX: only one transport specification is parsed */
889 static void rtsp_parse_transport(AVFormatContext *s,
890 RTSPMessageHeader *reply, const char *p)
892 char transport_protocol[16];
894 char lower_transport[16];
896 RTSPTransportField *th;
899 reply->nb_transports = 0;
902 p += strspn(p, SPACE_CHARS);
906 th = &reply->transports[reply->nb_transports];
908 get_word_sep(transport_protocol, sizeof(transport_protocol),
910 if (!av_strcasecmp (transport_protocol, "rtp")) {
911 get_word_sep(profile, sizeof(profile), "/;,", &p);
912 lower_transport[0] = '\0';
913 /* rtp/avp/<protocol> */
915 get_word_sep(lower_transport, sizeof(lower_transport),
918 th->transport = RTSP_TRANSPORT_RTP;
919 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
920 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
921 /* x-pn-tng/<protocol> */
922 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
924 th->transport = RTSP_TRANSPORT_RDT;
925 } else if (!av_strcasecmp(transport_protocol, "raw")) {
926 get_word_sep(profile, sizeof(profile), "/;,", &p);
927 lower_transport[0] = '\0';
928 /* raw/raw/<protocol> */
930 get_word_sep(lower_transport, sizeof(lower_transport),
933 th->transport = RTSP_TRANSPORT_RAW;
935 if (!av_strcasecmp(lower_transport, "TCP"))
936 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
938 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
942 /* get each parameter */
943 while (*p != '\0' && *p != ',') {
944 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
945 if (!strcmp(parameter, "port")) {
948 rtsp_parse_range(&th->port_min, &th->port_max, &p);
950 } else if (!strcmp(parameter, "client_port")) {
953 rtsp_parse_range(&th->client_port_min,
954 &th->client_port_max, &p);
956 } else if (!strcmp(parameter, "server_port")) {
959 rtsp_parse_range(&th->server_port_min,
960 &th->server_port_max, &p);
962 } else if (!strcmp(parameter, "interleaved")) {
965 rtsp_parse_range(&th->interleaved_min,
966 &th->interleaved_max, &p);
968 } else if (!strcmp(parameter, "multicast")) {
969 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
970 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
971 } else if (!strcmp(parameter, "ttl")) {
975 th->ttl = strtol(p, &end, 10);
978 } else if (!strcmp(parameter, "destination")) {
981 get_word_sep(buf, sizeof(buf), ";,", &p);
982 get_sockaddr(s, buf, &th->destination);
984 } else if (!strcmp(parameter, "source")) {
987 get_word_sep(buf, sizeof(buf), ";,", &p);
988 av_strlcpy(th->source, buf, sizeof(th->source));
990 } else if (!strcmp(parameter, "mode")) {
993 get_word_sep(buf, sizeof(buf), ";, ", &p);
994 if (!strcmp(buf, "record") ||
995 !strcmp(buf, "receive"))
1000 while (*p != ';' && *p != '\0' && *p != ',')
1008 reply->nb_transports++;
1009 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1014 static void handle_rtp_info(RTSPState *rt, const char *url,
1015 uint32_t seq, uint32_t rtptime)
1018 if (!rtptime || !url[0])
1020 if (rt->transport != RTSP_TRANSPORT_RTP)
1022 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1023 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1024 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1027 if (!strcmp(rtsp_st->control_url, url)) {
1028 rtpctx->base_timestamp = rtptime;
1034 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1037 char key[20], value[1024], url[1024] = "";
1038 uint32_t seq = 0, rtptime = 0;
1041 p += strspn(p, SPACE_CHARS);
1044 get_word_sep(key, sizeof(key), "=", &p);
1048 get_word_sep(value, sizeof(value), ";, ", &p);
1050 if (!strcmp(key, "url"))
1051 av_strlcpy(url, value, sizeof(url));
1052 else if (!strcmp(key, "seq"))
1053 seq = strtoul(value, NULL, 10);
1054 else if (!strcmp(key, "rtptime"))
1055 rtptime = strtoul(value, NULL, 10);
1057 handle_rtp_info(rt, url, seq, rtptime);
1066 handle_rtp_info(rt, url, seq, rtptime);
1069 void ff_rtsp_parse_line(AVFormatContext *s,
1070 RTSPMessageHeader *reply, const char *buf,
1071 RTSPState *rt, const char *method)
1075 /* NOTE: we do case independent match for broken servers */
1077 if (av_stristart(p, "Session:", &p)) {
1079 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1080 if (av_stristart(p, ";timeout=", &p) &&
1081 (t = strtol(p, NULL, 10)) > 0) {
1084 } else if (av_stristart(p, "Content-Length:", &p)) {
1085 reply->content_length = strtol(p, NULL, 10);
1086 } else if (av_stristart(p, "Transport:", &p)) {
1087 rtsp_parse_transport(s, reply, p);
1088 } else if (av_stristart(p, "CSeq:", &p)) {
1089 reply->seq = strtol(p, NULL, 10);
1090 } else if (av_stristart(p, "Range:", &p)) {
1091 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1092 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1093 p += strspn(p, SPACE_CHARS);
1094 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1095 } else if (av_stristart(p, "Server:", &p)) {
1096 p += strspn(p, SPACE_CHARS);
1097 av_strlcpy(reply->server, p, sizeof(reply->server));
1098 } else if (av_stristart(p, "Notice:", &p) ||
1099 av_stristart(p, "X-Notice:", &p)) {
1100 reply->notice = strtol(p, NULL, 10);
1101 } else if (av_stristart(p, "Location:", &p)) {
1102 p += strspn(p, SPACE_CHARS);
1103 av_strlcpy(reply->location, p , sizeof(reply->location));
1104 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1105 p += strspn(p, SPACE_CHARS);
1106 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1107 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1108 p += strspn(p, SPACE_CHARS);
1109 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1110 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1111 p += strspn(p, SPACE_CHARS);
1112 if (method && !strcmp(method, "DESCRIBE"))
1113 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1114 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1115 p += strspn(p, SPACE_CHARS);
1116 if (method && !strcmp(method, "PLAY"))
1117 rtsp_parse_rtp_info(rt, p);
1118 } else if (av_stristart(p, "Public:", &p) && rt) {
1119 if (strstr(p, "GET_PARAMETER") &&
1120 method && !strcmp(method, "OPTIONS"))
1121 rt->get_parameter_supported = 1;
1122 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1123 p += strspn(p, SPACE_CHARS);
1124 rt->accept_dynamic_rate = atoi(p);
1125 } else if (av_stristart(p, "Content-Type:", &p)) {
1126 p += strspn(p, SPACE_CHARS);
1127 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1131 /* skip a RTP/TCP interleaved packet */
1132 void ff_rtsp_skip_packet(AVFormatContext *s)
1134 RTSPState *rt = s->priv_data;
1138 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1141 len = AV_RB16(buf + 1);
1143 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1148 if (len1 > sizeof(buf))
1150 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1157 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1158 unsigned char **content_ptr,
1159 int return_on_interleaved_data, const char *method)
1161 RTSPState *rt = s->priv_data;
1162 char buf[4096], buf1[1024], *q;
1165 int ret, content_length, line_count = 0, request = 0;
1166 unsigned char *content = NULL;
1172 memset(reply, 0, sizeof(*reply));
1174 /* parse reply (XXX: use buffers) */
1175 rt->last_reply[0] = '\0';
1179 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1180 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1185 if (ch == '$' && q == buf) {
1186 if (return_on_interleaved_data) {
1189 ff_rtsp_skip_packet(s);
1190 } else if (ch != '\r') {
1191 if ((q - buf) < sizeof(buf) - 1)
1197 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1199 /* test if last line */
1203 if (line_count == 0) {
1204 /* get reply code */
1205 get_word(buf1, sizeof(buf1), &p);
1206 if (!strncmp(buf1, "RTSP/", 5)) {
1207 get_word(buf1, sizeof(buf1), &p);
1208 reply->status_code = atoi(buf1);
1209 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1211 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1212 get_word(buf1, sizeof(buf1), &p); // object
1216 ff_rtsp_parse_line(s, reply, p, rt, method);
1217 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1218 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1223 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1224 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1226 content_length = reply->content_length;
1227 if (content_length > 0) {
1228 /* leave some room for a trailing '\0' (useful for simple parsing) */
1229 content = av_malloc(content_length + 1);
1231 return AVERROR(ENOMEM);
1232 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1233 content[content_length] = '\0';
1236 *content_ptr = content;
1242 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1243 const char* ptr = buf;
1245 if (!strcmp(reply->reason, "OPTIONS")) {
1246 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1248 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1249 if (reply->session_id[0])
1250 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1253 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1255 av_strlcat(buf, "\r\n", sizeof(buf));
1257 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1258 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1261 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1263 rt->last_cmd_time = av_gettime_relative();
1264 /* Even if the request from the server had data, it is not the data
1265 * that the caller wants or expects. The memory could also be leaked
1266 * if the actual following reply has content data. */
1268 av_freep(content_ptr);
1269 /* If method is set, this is called from ff_rtsp_send_cmd,
1270 * where a reply to exactly this request is awaited. For
1271 * callers from within packet receiving, we just want to
1272 * return to the caller and go back to receiving packets. */
1278 if (rt->seq != reply->seq) {
1279 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1280 rt->seq, reply->seq);
1284 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1285 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1286 reply->notice == 2306 /* Continuous Feed Terminated */) {
1287 rt->state = RTSP_STATE_IDLE;
1288 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1289 return AVERROR(EIO); /* data or server error */
1290 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1291 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1292 return AVERROR(EPERM);
1298 * Send a command to the RTSP server without waiting for the reply.
1300 * @param s RTSP (de)muxer context
1301 * @param method the method for the request
1302 * @param url the target url for the request
1303 * @param headers extra header lines to include in the request
1304 * @param send_content if non-null, the data to send as request body content
1305 * @param send_content_length the length of the send_content data, or 0 if
1306 * send_content is null
1308 * @return zero if success, nonzero otherwise
1310 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1311 const char *method, const char *url,
1312 const char *headers,
1313 const unsigned char *send_content,
1314 int send_content_length)
1316 RTSPState *rt = s->priv_data;
1317 char buf[4096], *out_buf;
1318 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1320 if (!rt->rtsp_hd_out)
1321 return AVERROR(ENOTCONN);
1323 /* Add in RTSP headers */
1326 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1328 av_strlcat(buf, headers, sizeof(buf));
1329 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1330 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1331 if (rt->session_id[0] != '\0' && (!headers ||
1332 !strstr(headers, "\nIf-Match:"))) {
1333 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1336 char *str = ff_http_auth_create_response(&rt->auth_state,
1337 rt->auth, url, method);
1339 av_strlcat(buf, str, sizeof(buf));
1342 if (send_content_length > 0 && send_content)
1343 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1344 av_strlcat(buf, "\r\n", sizeof(buf));
1346 /* base64 encode rtsp if tunneling */
1347 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1348 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1349 out_buf = base64buf;
1352 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1354 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1355 if (send_content_length > 0 && send_content) {
1356 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1357 avpriv_report_missing_feature(s, "Tunneling of RTSP requests with content data");
1358 return AVERROR_PATCHWELCOME;
1360 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1362 rt->last_cmd_time = av_gettime_relative();
1367 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1368 const char *url, const char *headers)
1370 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1373 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1374 const char *headers, RTSPMessageHeader *reply,
1375 unsigned char **content_ptr)
1377 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1378 content_ptr, NULL, 0);
1381 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1382 const char *method, const char *url,
1384 RTSPMessageHeader *reply,
1385 unsigned char **content_ptr,
1386 const unsigned char *send_content,
1387 int send_content_length)
1389 RTSPState *rt = s->priv_data;
1390 HTTPAuthType cur_auth_type;
1391 int ret, attempts = 0;
1394 cur_auth_type = rt->auth_state.auth_type;
1395 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1397 send_content_length)))
1400 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1404 if (reply->status_code == 401 &&
1405 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1406 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1409 if (reply->status_code > 400){
1410 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1414 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1420 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1421 int lower_transport, const char *real_challenge)
1423 RTSPState *rt = s->priv_data;
1424 int rtx = 0, j, i, err, interleave = 0, port_off;
1425 RTSPStream *rtsp_st;
1426 RTSPMessageHeader reply1, *reply = &reply1;
1428 const char *trans_pref;
1430 if (rt->transport == RTSP_TRANSPORT_RDT)
1431 trans_pref = "x-pn-tng";
1432 else if (rt->transport == RTSP_TRANSPORT_RAW)
1433 trans_pref = "RAW/RAW";
1435 trans_pref = "RTP/AVP";
1437 /* default timeout: 1 minute */
1440 /* Choose a random starting offset within the first half of the
1441 * port range, to allow for a number of ports to try even if the offset
1442 * happens to be at the end of the random range. */
1443 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1444 /* even random offset */
1445 port_off -= port_off & 0x01;
1447 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1448 char transport[2048];
1451 * WMS serves all UDP data over a single connection, the RTX, which
1452 * isn't necessarily the first in the SDP but has to be the first
1453 * to be set up, else the second/third SETUP will fail with a 461.
1455 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1456 rt->server_type == RTSP_SERVER_WMS) {
1459 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1460 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1462 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1466 if (rtx == rt->nb_rtsp_streams)
1467 return -1; /* no RTX found */
1468 rtsp_st = rt->rtsp_streams[rtx];
1470 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1472 rtsp_st = rt->rtsp_streams[i];
1475 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1478 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1479 port = reply->transports[0].client_port_min;
1483 /* first try in specified port range */
1484 while (j <= rt->rtp_port_max) {
1485 AVDictionary *opts = map_to_opts(rt);
1487 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1488 "?localport=%d", j);
1489 /* we will use two ports per rtp stream (rtp and rtcp) */
1491 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1492 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1494 av_dict_free(&opts);
1499 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1504 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1506 snprintf(transport, sizeof(transport) - 1,
1507 "%s/UDP;", trans_pref);
1508 if (rt->server_type != RTSP_SERVER_REAL)
1509 av_strlcat(transport, "unicast;", sizeof(transport));
1510 av_strlcatf(transport, sizeof(transport),
1511 "client_port=%d", port);
1512 if (rt->transport == RTSP_TRANSPORT_RTP &&
1513 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1514 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1518 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1519 /* For WMS streams, the application streams are only used for
1520 * UDP. When trying to set it up for TCP streams, the server
1521 * will return an error. Therefore, we skip those streams. */
1522 if (rt->server_type == RTSP_SERVER_WMS &&
1523 (rtsp_st->stream_index < 0 ||
1524 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1527 snprintf(transport, sizeof(transport) - 1,
1528 "%s/TCP;", trans_pref);
1529 if (rt->transport != RTSP_TRANSPORT_RDT)
1530 av_strlcat(transport, "unicast;", sizeof(transport));
1531 av_strlcatf(transport, sizeof(transport),
1532 "interleaved=%d-%d",
1533 interleave, interleave + 1);
1537 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1538 snprintf(transport, sizeof(transport) - 1,
1539 "%s/UDP;multicast", trans_pref);
1542 av_strlcat(transport, ";mode=record", sizeof(transport));
1543 } else if (rt->server_type == RTSP_SERVER_REAL ||
1544 rt->server_type == RTSP_SERVER_WMS)
1545 av_strlcat(transport, ";mode=play", sizeof(transport));
1546 snprintf(cmd, sizeof(cmd),
1547 "Transport: %s\r\n",
1549 if (rt->accept_dynamic_rate)
1550 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1551 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1552 char real_res[41], real_csum[9];
1553 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1555 av_strlcatf(cmd, sizeof(cmd),
1557 "RealChallenge2: %s, sd=%s\r\n",
1558 rt->session_id, real_res, real_csum);
1560 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1561 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1564 } else if (reply->status_code != RTSP_STATUS_OK ||
1565 reply->nb_transports != 1) {
1566 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1570 /* XXX: same protocol for all streams is required */
1572 if (reply->transports[0].lower_transport != rt->lower_transport ||
1573 reply->transports[0].transport != rt->transport) {
1574 err = AVERROR_INVALIDDATA;
1578 rt->lower_transport = reply->transports[0].lower_transport;
1579 rt->transport = reply->transports[0].transport;
1582 /* Fail if the server responded with another lower transport mode
1583 * than what we requested. */
1584 if (reply->transports[0].lower_transport != lower_transport) {
1585 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1586 err = AVERROR_INVALIDDATA;
1590 switch(reply->transports[0].lower_transport) {
1591 case RTSP_LOWER_TRANSPORT_TCP:
1592 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1593 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1596 case RTSP_LOWER_TRANSPORT_UDP: {
1597 char url[1024], options[30] = "";
1598 const char *peer = host;
1600 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1601 av_strlcpy(options, "?connect=1", sizeof(options));
1602 /* Use source address if specified */
1603 if (reply->transports[0].source[0])
1604 peer = reply->transports[0].source;
1605 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1606 reply->transports[0].server_port_min, "%s", options);
1607 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1608 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1609 err = AVERROR_INVALIDDATA;
1614 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1615 char url[1024], namebuf[50], optbuf[20] = "";
1616 struct sockaddr_storage addr;
1618 AVDictionary *opts = map_to_opts(rt);
1620 if (reply->transports[0].destination.ss_family) {
1621 addr = reply->transports[0].destination;
1622 port = reply->transports[0].port_min;
1623 ttl = reply->transports[0].ttl;
1625 addr = rtsp_st->sdp_ip;
1626 port = rtsp_st->sdp_port;
1627 ttl = rtsp_st->sdp_ttl;
1630 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1631 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1632 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1633 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1634 port, "%s", optbuf);
1635 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1636 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1637 av_dict_free(&opts);
1640 err = AVERROR_INVALIDDATA;
1647 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1651 if (rt->nb_rtsp_streams && reply->timeout > 0)
1652 rt->timeout = reply->timeout;
1654 if (rt->server_type == RTSP_SERVER_REAL)
1655 rt->need_subscription = 1;
1660 ff_rtsp_undo_setup(s, 0);
1664 void ff_rtsp_close_connections(AVFormatContext *s)
1666 RTSPState *rt = s->priv_data;
1667 if (rt->rtsp_hd_out != rt->rtsp_hd)
1668 ffurl_closep(&rt->rtsp_hd_out);
1669 rt->rtsp_hd_out = NULL;
1670 ffurl_closep(&rt->rtsp_hd);
1673 int ff_rtsp_connect(AVFormatContext *s)
1675 RTSPState *rt = s->priv_data;
1676 char proto[128], host[1024], path[1024];
1677 char tcpname[1024], cmd[2048], auth[128];
1678 const char *lower_rtsp_proto = "tcp";
1679 int port, err, tcp_fd;
1680 RTSPMessageHeader reply1, *reply = &reply1;
1681 int lower_transport_mask = 0;
1682 int default_port = RTSP_DEFAULT_PORT;
1683 int https_tunnel = 0;
1684 char real_challenge[64] = "";
1685 struct sockaddr_storage peer;
1686 socklen_t peer_len = sizeof(peer);
1688 if (rt->rtp_port_max < rt->rtp_port_min) {
1689 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1690 "than min port %d\n", rt->rtp_port_max,
1692 return AVERROR(EINVAL);
1695 if (!ff_network_init())
1696 return AVERROR(EIO);
1698 if (s->max_delay < 0) /* Not set by the caller */
1699 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1701 rt->control_transport = RTSP_MODE_PLAIN;
1702 if (rt->lower_transport_mask & ((1 << RTSP_LOWER_TRANSPORT_HTTP) |
1703 (1 << RTSP_LOWER_TRANSPORT_HTTPS))) {
1704 https_tunnel = !!(rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTPS));
1705 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1706 rt->control_transport = RTSP_MODE_TUNNEL;
1708 /* Only pass through valid flags from here */
1709 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1712 memset(&reply1, 0, sizeof(reply1));
1713 /* extract hostname and port */
1714 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1715 host, sizeof(host), &port, path, sizeof(path), s->url);
1717 if (!strcmp(proto, "rtsps")) {
1718 lower_rtsp_proto = "tls";
1719 default_port = RTSPS_DEFAULT_PORT;
1720 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1724 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1727 port = default_port;
1729 lower_transport_mask = rt->lower_transport_mask;
1731 if (!lower_transport_mask)
1732 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1735 /* Only UDP or TCP - UDP multicast isn't supported. */
1736 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1737 (1 << RTSP_LOWER_TRANSPORT_TCP);
1738 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1739 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1740 "only UDP and TCP are supported for output.\n");
1741 err = AVERROR(EINVAL);
1746 /* Construct the URI used in request; this is similar to s->url,
1747 * but with authentication credentials removed and RTSP specific options
1749 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1750 host, port, "%s", path);
1752 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1753 /* set up initial handshake for tunneling */
1754 char httpname[1024];
1755 char sessioncookie[17];
1757 AVDictionary *options = NULL;
1759 av_dict_set_int(&options, "timeout", rt->stimeout, 0);
1761 ff_url_join(httpname, sizeof(httpname), https_tunnel ? "https" : "http", auth, host, port, "%s", path);
1762 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1763 av_get_random_seed(), av_get_random_seed());
1766 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1767 &s->interrupt_callback) < 0) {
1772 /* generate GET headers */
1773 snprintf(headers, sizeof(headers),
1774 "x-sessioncookie: %s\r\n"
1775 "Accept: application/x-rtsp-tunnelled\r\n"
1776 "Pragma: no-cache\r\n"
1777 "Cache-Control: no-cache\r\n",
1779 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1781 if (!rt->rtsp_hd->protocol_whitelist && s->protocol_whitelist) {
1782 rt->rtsp_hd->protocol_whitelist = av_strdup(s->protocol_whitelist);
1783 if (!rt->rtsp_hd->protocol_whitelist) {
1784 err = AVERROR(ENOMEM);
1789 /* complete the connection */
1790 if (ffurl_connect(rt->rtsp_hd, &options)) {
1791 av_dict_free(&options);
1797 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1798 &s->interrupt_callback) < 0 ) {
1803 /* generate POST headers */
1804 snprintf(headers, sizeof(headers),
1805 "x-sessioncookie: %s\r\n"
1806 "Content-Type: application/x-rtsp-tunnelled\r\n"
1807 "Pragma: no-cache\r\n"
1808 "Cache-Control: no-cache\r\n"
1809 "Content-Length: 32767\r\n"
1810 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1812 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1813 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1814 av_opt_set(rt->rtsp_hd_out->priv_data, "send_expect_100", "0", 0);
1816 /* Initialize the authentication state for the POST session. The HTTP
1817 * protocol implementation doesn't properly handle multi-pass
1818 * authentication for POST requests, since it would require one of
1820 * - implementing Expect: 100-continue, which many HTTP servers
1821 * don't support anyway, even less the RTSP servers that do HTTP
1823 * - sending the whole POST data until getting a 401 reply specifying
1824 * what authentication method to use, then resending all that data
1825 * - waiting for potential 401 replies directly after sending the
1826 * POST header (waiting for some unspecified time)
1827 * Therefore, we copy the full auth state, which works for both basic
1828 * and digest. (For digest, we would have to synchronize the nonce
1829 * count variable between the two sessions, if we'd do more requests
1830 * with the original session, though.)
1832 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1834 /* complete the connection */
1835 if (ffurl_connect(rt->rtsp_hd_out, &options)) {
1836 av_dict_free(&options);
1840 av_dict_free(&options);
1843 /* open the tcp connection */
1844 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1846 "?timeout=%d", rt->stimeout);
1847 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1848 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1852 rt->rtsp_hd_out = rt->rtsp_hd;
1856 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1861 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1862 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1863 NULL, 0, NI_NUMERICHOST);
1866 /* request options supported by the server; this also detects server
1868 for (rt->server_type = RTSP_SERVER_RTP;;) {
1870 if (rt->server_type == RTSP_SERVER_REAL)
1873 * The following entries are required for proper
1874 * streaming from a Realmedia server. They are
1875 * interdependent in some way although we currently
1876 * don't quite understand how. Values were copied
1877 * from mplayer SVN r23589.
1878 * ClientChallenge is a 16-byte ID in hex
1879 * CompanyID is a 16-byte ID in base64
1881 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1882 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1883 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1884 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1886 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1887 if (reply->status_code != RTSP_STATUS_OK) {
1888 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1892 /* detect server type if not standard-compliant RTP */
1893 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1894 rt->server_type = RTSP_SERVER_REAL;
1896 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1897 rt->server_type = RTSP_SERVER_WMS;
1898 } else if (rt->server_type == RTSP_SERVER_REAL)
1899 strcpy(real_challenge, reply->real_challenge);
1903 if (CONFIG_RTSP_DEMUXER && s->iformat)
1904 err = ff_rtsp_setup_input_streams(s, reply);
1905 else if (CONFIG_RTSP_MUXER)
1906 err = ff_rtsp_setup_output_streams(s, host);
1913 int lower_transport = ff_log2_tab[lower_transport_mask &
1914 ~(lower_transport_mask - 1)];
1916 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1917 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1918 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1920 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1921 rt->server_type == RTSP_SERVER_REAL ?
1922 real_challenge : NULL);
1925 lower_transport_mask &= ~(1 << lower_transport);
1926 if (lower_transport_mask == 0 && err == 1) {
1927 err = AVERROR(EPROTONOSUPPORT);
1932 rt->lower_transport_mask = lower_transport_mask;
1933 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1934 rt->state = RTSP_STATE_IDLE;
1935 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1938 ff_rtsp_close_streams(s);
1939 ff_rtsp_close_connections(s);
1940 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1941 char *new_url = av_strdup(reply->location);
1943 err = AVERROR(ENOMEM);
1946 ff_format_set_url(s, new_url);
1947 rt->session_id[0] = '\0';
1948 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1957 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1960 static int parse_rtsp_message(AVFormatContext *s)
1962 RTSPState *rt = s->priv_data;
1965 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1966 if (rt->state == RTSP_STATE_STREAMING) {
1967 return ff_rtsp_parse_streaming_commands(s);
1971 RTSPMessageHeader reply;
1972 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1975 /* XXX: parse message */
1976 if (rt->state != RTSP_STATE_STREAMING)
1983 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1984 uint8_t *buf, int buf_size, int64_t wait_end)
1986 RTSPState *rt = s->priv_data;
1987 RTSPStream *rtsp_st;
1988 int n, i, ret, timeout_cnt = 0;
1989 struct pollfd *p = rt->p;
1990 int *fds = NULL, fdsnum, fdsidx;
1993 p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(struct pollfd));
1995 return AVERROR(ENOMEM);
1998 p[rt->max_p].fd = ffurl_get_file_handle(rt->rtsp_hd);
1999 p[rt->max_p++].events = POLLIN;
2001 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2002 rtsp_st = rt->rtsp_streams[i];
2003 if (rtsp_st->rtp_handle) {
2004 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
2006 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
2010 av_log(s, AV_LOG_ERROR,
2011 "Number of fds %d not supported\n", fdsnum);
2012 return AVERROR_INVALIDDATA;
2014 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
2015 p[rt->max_p].fd = fds[fdsidx];
2016 p[rt->max_p++].events = POLLIN;
2024 if (ff_check_interrupt(&s->interrupt_callback))
2025 return AVERROR_EXIT;
2026 if (wait_end && wait_end - av_gettime_relative() < 0)
2027 return AVERROR(EAGAIN);
2028 n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
2030 int j = rt->rtsp_hd ? 1 : 0;
2032 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2033 rtsp_st = rt->rtsp_streams[i];
2034 if (rtsp_st->rtp_handle) {
2035 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
2036 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
2038 *prtsp_st = rtsp_st;
2045 #if CONFIG_RTSP_DEMUXER
2046 if (rt->rtsp_hd && p[0].revents & POLLIN) {
2047 if ((ret = parse_rtsp_message(s)) < 0) {
2052 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
2053 return AVERROR(ETIMEDOUT);
2054 } else if (n < 0 && errno != EINTR)
2055 return AVERROR(errno);
2059 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2060 const uint8_t *buf, int len)
2062 RTSPState *rt = s->priv_data;
2066 if (rt->nb_rtsp_streams == 1) {
2067 *rtsp_st = rt->rtsp_streams[0];
2070 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2071 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2073 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2077 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2078 *rtsp_st = rt->rtsp_streams[i];
2085 av_log(s, AV_LOG_WARNING,
2086 "Unable to pick stream for packet - SSRC not known for "
2088 return AVERROR(EAGAIN);
2091 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2092 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2093 *rtsp_st = rt->rtsp_streams[i];
2099 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2100 return AVERROR(EAGAIN);
2103 static int read_packet(AVFormatContext *s,
2104 RTSPStream **rtsp_st, RTSPStream *first_queue_st,
2107 RTSPState *rt = s->priv_data;
2110 switch(rt->lower_transport) {
2112 #if CONFIG_RTSP_DEMUXER
2113 case RTSP_LOWER_TRANSPORT_TCP:
2114 len = ff_rtsp_tcp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2117 case RTSP_LOWER_TRANSPORT_UDP:
2118 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2119 len = udp_read_packet(s, rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2120 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2121 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, (*rtsp_st)->rtp_handle, NULL, len);
2123 case RTSP_LOWER_TRANSPORT_CUSTOM:
2124 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2125 wait_end && wait_end < av_gettime_relative())
2126 len = AVERROR(EAGAIN);
2128 len = avio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2129 len = pick_stream(s, rtsp_st, rt->recvbuf, len);
2130 if (len > 0 && (*rtsp_st)->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2131 ff_rtp_check_and_send_back_rr((*rtsp_st)->transport_priv, NULL, s->pb, len);
2141 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2143 RTSPState *rt = s->priv_data;
2145 RTSPStream *rtsp_st, *first_queue_st = NULL;
2146 int64_t wait_end = 0;
2148 if (rt->nb_byes == rt->nb_rtsp_streams)
2151 /* get next frames from the same RTP packet */
2152 if (rt->cur_transport_priv) {
2153 if (rt->transport == RTSP_TRANSPORT_RDT) {
2154 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2155 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2156 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2157 } else if (CONFIG_RTPDEC && rt->ts) {
2158 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2160 rt->recvbuf_pos += ret;
2161 ret = rt->recvbuf_pos < rt->recvbuf_len;
2166 rt->cur_transport_priv = NULL;
2168 } else if (ret == 1) {
2171 rt->cur_transport_priv = NULL;
2175 if (rt->transport == RTSP_TRANSPORT_RTP) {
2177 int64_t first_queue_time = 0;
2178 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2179 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2183 queue_time = ff_rtp_queued_packet_time(rtpctx);
2184 if (queue_time && (queue_time - first_queue_time < 0 ||
2185 !first_queue_time)) {
2186 first_queue_time = queue_time;
2187 first_queue_st = rt->rtsp_streams[i];
2190 if (first_queue_time) {
2191 wait_end = first_queue_time + s->max_delay;
2194 first_queue_st = NULL;
2198 /* read next RTP packet */
2200 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2202 return AVERROR(ENOMEM);
2205 len = read_packet(s, &rtsp_st, first_queue_st, wait_end);
2206 if (len == AVERROR(EAGAIN) && first_queue_st &&
2207 rt->transport == RTSP_TRANSPORT_RTP) {
2208 av_log(s, AV_LOG_WARNING,
2209 "max delay reached. need to consume packet\n");
2210 rtsp_st = first_queue_st;
2211 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2217 if (rt->transport == RTSP_TRANSPORT_RDT) {
2218 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2219 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2220 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2221 if (rtsp_st->feedback) {
2222 AVIOContext *pb = NULL;
2223 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2225 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2228 /* Either bad packet, or a RTCP packet. Check if the
2229 * first_rtcp_ntp_time field was initialized. */
2230 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2231 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2232 /* first_rtcp_ntp_time has been initialized for this stream,
2233 * copy the same value to all other uninitialized streams,
2234 * in order to map their timestamp origin to the same ntp time
2237 AVStream *st = NULL;
2238 if (rtsp_st->stream_index >= 0)
2239 st = s->streams[rtsp_st->stream_index];
2240 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2241 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2242 AVStream *st2 = NULL;
2243 if (rt->rtsp_streams[i]->stream_index >= 0)
2244 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2245 if (rtpctx2 && st && st2 &&
2246 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2247 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2248 rtpctx2->rtcp_ts_offset = av_rescale_q(
2249 rtpctx->rtcp_ts_offset, st->time_base,
2253 // Make real NTP start time available in AVFormatContext
2254 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2255 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2257 s->start_time_realtime -=
2258 av_rescale (rtpctx->rtcp_ts_offset,
2259 (uint64_t) rtpctx->st->time_base.num * 1000000,
2260 rtpctx->st->time_base.den);
2264 if (ret == -RTCP_BYE) {
2267 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2268 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2270 if (rt->nb_byes == rt->nb_rtsp_streams)
2274 } else if (CONFIG_RTPDEC && rt->ts) {
2275 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2278 rt->recvbuf_len = len;
2279 rt->recvbuf_pos = ret;
2280 rt->cur_transport_priv = rt->ts;
2287 return AVERROR_INVALIDDATA;
2293 /* more packets may follow, so we save the RTP context */
2294 rt->cur_transport_priv = rtsp_st->transport_priv;
2298 #endif /* CONFIG_RTPDEC */
2300 #if CONFIG_SDP_DEMUXER
2301 static int sdp_probe(const AVProbeData *p1)
2303 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2305 /* we look for a line beginning "c=IN IP" */
2306 while (p < p_end && *p != '\0') {
2307 if (sizeof("c=IN IP") - 1 < p_end - p &&
2308 av_strstart(p, "c=IN IP", NULL))
2309 return AVPROBE_SCORE_EXTENSION;
2311 while (p < p_end - 1 && *p != '\n') p++;
2320 static void append_source_addrs(char *buf, int size, const char *name,
2321 int count, struct RTSPSource **addrs)
2326 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2327 for (i = 1; i < count; i++)
2328 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2331 static int sdp_read_header(AVFormatContext *s)
2333 RTSPState *rt = s->priv_data;
2334 RTSPStream *rtsp_st;
2339 if (!ff_network_init())
2340 return AVERROR(EIO);
2342 if (s->max_delay < 0) /* Not set by the caller */
2343 s->max_delay = DEFAULT_REORDERING_DELAY;
2344 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2345 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2347 /* read the whole sdp file */
2348 /* XXX: better loading */
2349 content = av_malloc(SDP_MAX_SIZE);
2351 return AVERROR(ENOMEM);
2352 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2355 return AVERROR_INVALIDDATA;
2357 content[size] ='\0';
2359 err = ff_sdp_parse(s, content);
2363 /* open each RTP stream */
2364 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2366 rtsp_st = rt->rtsp_streams[i];
2368 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2369 AVDictionary *opts = map_to_opts(rt);
2371 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2372 sizeof(rtsp_st->sdp_ip),
2373 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2375 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2377 av_dict_free(&opts);
2380 ff_url_join(url, sizeof(url), "rtp", NULL,
2381 namebuf, rtsp_st->sdp_port,
2382 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2383 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2384 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2385 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2387 append_source_addrs(url, sizeof(url), "sources",
2388 rtsp_st->nb_include_source_addrs,
2389 rtsp_st->include_source_addrs);
2390 append_source_addrs(url, sizeof(url), "block",
2391 rtsp_st->nb_exclude_source_addrs,
2392 rtsp_st->exclude_source_addrs);
2393 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2394 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2396 av_dict_free(&opts);
2399 err = AVERROR_INVALIDDATA;
2403 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2408 ff_rtsp_close_streams(s);
2413 static int sdp_read_close(AVFormatContext *s)
2415 ff_rtsp_close_streams(s);
2420 static const AVClass sdp_demuxer_class = {
2421 .class_name = "SDP demuxer",
2422 .item_name = av_default_item_name,
2423 .option = sdp_options,
2424 .version = LIBAVUTIL_VERSION_INT,
2427 AVInputFormat ff_sdp_demuxer = {
2429 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2430 .priv_data_size = sizeof(RTSPState),
2431 .read_probe = sdp_probe,
2432 .read_header = sdp_read_header,
2433 .read_packet = ff_rtsp_fetch_packet,
2434 .read_close = sdp_read_close,
2435 .priv_class = &sdp_demuxer_class,
2437 #endif /* CONFIG_SDP_DEMUXER */
2439 #if CONFIG_RTP_DEMUXER
2440 static int rtp_probe(const AVProbeData *p)
2442 if (av_strstart(p->filename, "rtp:", NULL))
2443 return AVPROBE_SCORE_MAX;
2447 static int rtp_read_header(AVFormatContext *s)
2449 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2450 char host[500], filters_buf[1000];
2452 URLContext* in = NULL;
2454 AVCodecParameters *par = NULL;
2455 struct sockaddr_storage addr;
2457 socklen_t addrlen = sizeof(addr);
2458 RTSPState *rt = s->priv_data;
2462 if (!ff_network_init())
2463 return AVERROR(EIO);
2465 ret = ffurl_open_whitelist(&in, s->url, AVIO_FLAG_READ,
2466 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2471 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2472 if (ret == AVERROR(EAGAIN))
2477 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2481 if ((recvbuf[0] & 0xc0) != 0x80) {
2482 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2487 if (RTP_PT_IS_RTCP(recvbuf[1]))
2490 payload_type = recvbuf[1] & 0x7f;
2493 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2496 par = avcodec_parameters_alloc();
2498 ret = AVERROR(ENOMEM);
2502 if (ff_rtp_get_codec_info(par, payload_type)) {
2503 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2504 "without an SDP file describing it\n",
2508 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2509 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2510 "properly you need an SDP file "
2514 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2517 av_bprint_init(&sdp, 0, AV_BPRINT_SIZE_UNLIMITED);
2518 av_bprintf(&sdp, "v=0\r\nc=IN IP%d %s\r\n",
2519 addr.ss_family == AF_INET ? 4 : 6, host);
2521 p = strchr(s->url, '?');
2523 static const char filters[][2][8] = { { "sources", "incl" },
2524 { "block", "excl" } };
2527 for (i = 0; i < FF_ARRAY_ELEMS(filters); i++) {
2528 if (av_find_info_tag(filters_buf, sizeof(filters_buf), filters[i][0], p)) {
2530 while ((q = strchr(q, ',')) != NULL)
2532 av_bprintf(&sdp, "a=source-filter:%s IN IP%d %s %s\r\n",
2534 addr.ss_family == AF_INET ? 4 : 6, host,
2540 av_bprintf(&sdp, "m=%s %d RTP/AVP %d\r\n",
2541 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2542 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2543 port, payload_type);
2544 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp.str);
2545 if (!av_bprint_is_complete(&sdp))
2547 avcodec_parameters_free(&par);
2549 ffio_init_context(&pb, sdp.str, sdp.len, 0, NULL, NULL, NULL, NULL);
2552 /* sdp_read_header initializes this again */
2555 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2557 ret = sdp_read_header(s);
2559 av_bprint_finalize(&sdp, NULL);
2563 ret = AVERROR(ENOMEM);
2564 av_log(s, AV_LOG_ERROR, "rtp_read_header(): not enough buffer space for sdp-headers\n");
2565 av_bprint_finalize(&sdp, NULL);
2567 avcodec_parameters_free(&par);
2573 static const AVClass rtp_demuxer_class = {
2574 .class_name = "RTP demuxer",
2575 .item_name = av_default_item_name,
2576 .option = rtp_options,
2577 .version = LIBAVUTIL_VERSION_INT,
2580 AVInputFormat ff_rtp_demuxer = {
2582 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2583 .priv_data_size = sizeof(RTSPState),
2584 .read_probe = rtp_probe,
2585 .read_header = rtp_read_header,
2586 .read_packet = ff_rtsp_fetch_packet,
2587 .read_close = sdp_read_close,
2588 .flags = AVFMT_NOFILE,
2589 .priv_class = &rtp_demuxer_class,
2591 #endif /* CONFIG_RTP_DEMUXER */