3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
77 #define COMMON_OPTS() \
78 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
82 const AVOption ff_rtsp_options[] = {
83 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC },
84 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
99 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103 static const AVOption sdp_options[] = {
104 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
107 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
112 static const AVOption rtp_options[] = {
113 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
119 static AVDictionary *map_to_opts(RTSPState *rt)
121 AVDictionary *opts = NULL;
124 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125 av_dict_set(&opts, "buffer_size", buf, 0);
130 static void get_word_until_chars(char *buf, int buf_size,
131 const char *sep, const char **pp)
137 p += strspn(p, SPACE_CHARS);
139 while (!strchr(sep, *p) && *p != '\0') {
140 if ((q - buf) < buf_size - 1)
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
152 if (**pp == '/') (*pp)++;
153 get_word_until_chars(buf, buf_size, sep, pp);
156 static void get_word(char *buf, int buf_size, const char **pp)
158 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
163 * Used for seeking in the rtp stream.
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
169 p += strspn(p, SPACE_CHARS);
170 if (!av_stristart(p, "npt=", &p))
173 *start = AV_NOPTS_VALUE;
174 *end = AV_NOPTS_VALUE;
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 if (av_parse_time(start, buf, 1) < 0)
181 get_word_sep(buf, sizeof(buf), "-", &p);
182 if (av_parse_time(end, buf, 1) < 0)
183 av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);
187 static int get_sockaddr(AVFormatContext *s,
188 const char *buf, struct sockaddr_storage *sock)
190 struct addrinfo hints = { 0 }, *ai = NULL;
193 hints.ai_flags = AI_NUMERICHOST;
194 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
195 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
200 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
206 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
207 RTSPStream *rtsp_st, AVStream *st)
209 AVCodecParameters *par = st ? st->codecpar : NULL;
213 par->codec_id = handler->codec_id;
214 rtsp_st->dynamic_handler = handler;
216 st->need_parsing = handler->need_parsing;
217 if (handler->priv_data_size) {
218 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
219 if (!rtsp_st->dynamic_protocol_context)
220 rtsp_st->dynamic_handler = NULL;
224 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
227 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
228 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
229 rtsp_st->dynamic_protocol_context);
231 if (rtsp_st->dynamic_protocol_context) {
232 if (rtsp_st->dynamic_handler->close)
233 rtsp_st->dynamic_handler->close(
234 rtsp_st->dynamic_protocol_context);
235 av_free(rtsp_st->dynamic_protocol_context);
237 rtsp_st->dynamic_protocol_context = NULL;
238 rtsp_st->dynamic_handler = NULL;
243 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
244 static int sdp_parse_rtpmap(AVFormatContext *s,
245 AVStream *st, RTSPStream *rtsp_st,
246 int payload_type, const char *p)
248 AVCodecParameters *par = st->codecpar;
251 const AVCodecDescriptor *desc;
254 /* See if we can handle this kind of payload.
255 * The space should normally not be there but some Real streams or
256 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
257 * have a trailing space. */
258 get_word_sep(buf, sizeof(buf), "/ ", &p);
259 if (payload_type < RTP_PT_PRIVATE) {
260 /* We are in a standard case
261 * (from http://www.iana.org/assignments/rtp-parameters). */
262 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
265 if (par->codec_id == AV_CODEC_ID_NONE) {
266 RTPDynamicProtocolHandler *handler =
267 ff_rtp_handler_find_by_name(buf, par->codec_type);
268 init_rtp_handler(handler, rtsp_st, st);
269 /* If no dynamic handler was found, check with the list of standard
270 * allocated types, if such a stream for some reason happens to
271 * use a private payload type. This isn't handled in rtpdec.c, since
272 * the format name from the rtpmap line never is passed into rtpdec. */
273 if (!rtsp_st->dynamic_handler)
274 par->codec_id = ff_rtp_codec_id(buf, par->codec_type);
277 desc = avcodec_descriptor_get(par->codec_id);
278 if (desc && desc->name)
283 get_word_sep(buf, sizeof(buf), "/", &p);
285 switch (par->codec_type) {
286 case AVMEDIA_TYPE_AUDIO:
287 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
288 par->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
289 par->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
291 par->sample_rate = i;
292 avpriv_set_pts_info(st, 32, 1, par->sample_rate);
293 get_word_sep(buf, sizeof(buf), "/", &p);
298 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
300 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
303 case AVMEDIA_TYPE_VIDEO:
304 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
306 avpriv_set_pts_info(st, 32, 1, i);
311 finalize_rtp_handler_init(s, rtsp_st, st);
315 /* parse the attribute line from the fmtp a line of an sdp response. This
316 * is broken out as a function because it is used in rtp_h264.c, which is
318 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
319 char *value, int value_size)
321 *p += strspn(*p, SPACE_CHARS);
323 get_word_sep(attr, attr_size, "=", p);
326 get_word_sep(value, value_size, ";", p);
334 typedef struct SDPParseState {
336 struct sockaddr_storage default_ip;
338 int skip_media; ///< set if an unknown m= line occurs
339 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
340 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
341 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
342 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
345 char delayed_fmtp[2048];
348 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
349 struct RTSPSource ***dest, int *dest_count)
351 RTSPSource *rtsp_src, *rtsp_src2;
353 for (i = 0; i < count; i++) {
355 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
358 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
359 dynarray_add(dest, dest_count, rtsp_src2);
363 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
364 int payload_type, const char *line)
368 for (i = 0; i < rt->nb_rtsp_streams; i++) {
369 RTSPStream *rtsp_st = rt->rtsp_streams[i];
370 if (rtsp_st->sdp_payload_type == payload_type &&
371 rtsp_st->dynamic_handler &&
372 rtsp_st->dynamic_handler->parse_sdp_a_line) {
373 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
374 rtsp_st->dynamic_protocol_context, line);
379 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
380 int letter, const char *buf)
382 RTSPState *rt = s->priv_data;
383 char buf1[64], st_type[64];
385 enum AVMediaType codec_type;
389 RTSPSource *rtsp_src;
390 struct sockaddr_storage sdp_ip;
393 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
396 if (s1->skip_media && letter != 'm')
400 get_word(buf1, sizeof(buf1), &p);
401 if (strcmp(buf1, "IN") != 0)
403 get_word(buf1, sizeof(buf1), &p);
404 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
406 get_word_sep(buf1, sizeof(buf1), "/", &p);
407 if (get_sockaddr(s, buf1, &sdp_ip))
412 get_word_sep(buf1, sizeof(buf1), "/", &p);
415 if (s->nb_streams == 0) {
416 s1->default_ip = sdp_ip;
417 s1->default_ttl = ttl;
419 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420 rtsp_st->sdp_ip = sdp_ip;
421 rtsp_st->sdp_ttl = ttl;
425 av_dict_set(&s->metadata, "title", p, 0);
428 if (s->nb_streams == 0) {
429 av_dict_set(&s->metadata, "comment", p, 0);
438 codec_type = AVMEDIA_TYPE_UNKNOWN;
439 get_word(st_type, sizeof(st_type), &p);
440 if (!strcmp(st_type, "audio")) {
441 codec_type = AVMEDIA_TYPE_AUDIO;
442 } else if (!strcmp(st_type, "video")) {
443 codec_type = AVMEDIA_TYPE_VIDEO;
444 } else if (!strcmp(st_type, "application")) {
445 codec_type = AVMEDIA_TYPE_DATA;
446 } else if (!strcmp(st_type, "text")) {
447 codec_type = AVMEDIA_TYPE_SUBTITLE;
449 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
453 rtsp_st = av_mallocz(sizeof(RTSPStream));
456 rtsp_st->stream_index = -1;
457 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
459 rtsp_st->sdp_ip = s1->default_ip;
460 rtsp_st->sdp_ttl = s1->default_ttl;
462 copy_default_source_addrs(s1->default_include_source_addrs,
463 s1->nb_default_include_source_addrs,
464 &rtsp_st->include_source_addrs,
465 &rtsp_st->nb_include_source_addrs);
466 copy_default_source_addrs(s1->default_exclude_source_addrs,
467 s1->nb_default_exclude_source_addrs,
468 &rtsp_st->exclude_source_addrs,
469 &rtsp_st->nb_exclude_source_addrs);
471 get_word(buf1, sizeof(buf1), &p); /* port */
472 rtsp_st->sdp_port = atoi(buf1);
474 get_word(buf1, sizeof(buf1), &p); /* protocol */
475 if (!strcmp(buf1, "udp"))
476 rt->transport = RTSP_TRANSPORT_RAW;
477 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
478 rtsp_st->feedback = 1;
480 /* XXX: handle list of formats */
481 get_word(buf1, sizeof(buf1), &p); /* format list */
482 rtsp_st->sdp_payload_type = atoi(buf1);
484 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
485 /* no corresponding stream */
486 if (rt->transport == RTSP_TRANSPORT_RAW) {
487 if (CONFIG_RTPDEC && !rt->ts)
488 rt->ts = avpriv_mpegts_parse_open(s);
490 RTPDynamicProtocolHandler *handler;
491 handler = ff_rtp_handler_find_by_id(
492 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
493 init_rtp_handler(handler, rtsp_st, NULL);
494 finalize_rtp_handler_init(s, rtsp_st, NULL);
496 } else if (rt->server_type == RTSP_SERVER_WMS &&
497 codec_type == AVMEDIA_TYPE_DATA) {
498 /* RTX stream, a stream that carries all the other actual
499 * audio/video streams. Don't expose this to the callers. */
501 st = avformat_new_stream(s, NULL);
504 st->id = rt->nb_rtsp_streams - 1;
505 rtsp_st->stream_index = st->index;
506 st->codecpar->codec_type = codec_type;
507 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
508 RTPDynamicProtocolHandler *handler;
509 /* if standard payload type, we can find the codec right now */
510 ff_rtp_get_codec_info(st->codecpar, rtsp_st->sdp_payload_type);
511 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
512 st->codecpar->sample_rate > 0)
513 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
514 /* Even static payload types may need a custom depacketizer */
515 handler = ff_rtp_handler_find_by_id(
516 rtsp_st->sdp_payload_type, st->codecpar->codec_type);
517 init_rtp_handler(handler, rtsp_st, st);
518 finalize_rtp_handler_init(s, rtsp_st, st);
520 if (rt->default_lang[0])
521 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
523 /* put a default control url */
524 av_strlcpy(rtsp_st->control_url, rt->control_uri,
525 sizeof(rtsp_st->control_url));
528 if (av_strstart(p, "control:", &p)) {
529 if (s->nb_streams == 0) {
530 if (!strncmp(p, "rtsp://", 7))
531 av_strlcpy(rt->control_uri, p,
532 sizeof(rt->control_uri));
535 /* get the control url */
536 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
538 /* XXX: may need to add full url resolution */
539 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
541 if (proto[0] == '\0') {
542 /* relative control URL */
543 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
544 av_strlcat(rtsp_st->control_url, "/",
545 sizeof(rtsp_st->control_url));
546 av_strlcat(rtsp_st->control_url, p,
547 sizeof(rtsp_st->control_url));
549 av_strlcpy(rtsp_st->control_url, p,
550 sizeof(rtsp_st->control_url));
552 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
553 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
554 get_word(buf1, sizeof(buf1), &p);
555 payload_type = atoi(buf1);
556 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
557 if (rtsp_st->stream_index >= 0) {
558 st = s->streams[rtsp_st->stream_index];
559 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
563 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
565 } else if (av_strstart(p, "fmtp:", &p) ||
566 av_strstart(p, "framesize:", &p)) {
567 // let dynamic protocol handlers have a stab at the line.
568 get_word(buf1, sizeof(buf1), &p);
569 payload_type = atoi(buf1);
570 if (s1->seen_rtpmap) {
571 parse_fmtp(s, rt, payload_type, buf);
574 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
576 } else if (av_strstart(p, "ssrc:", &p) && s->nb_streams > 0) {
577 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
578 get_word(buf1, sizeof(buf1), &p);
579 rtsp_st->ssrc = strtoll(buf1, NULL, 10);
580 } else if (av_strstart(p, "range:", &p)) {
583 // this is so that seeking on a streamed file can work.
584 rtsp_parse_range_npt(p, &start, &end);
585 s->start_time = start;
586 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
587 s->duration = (end == AV_NOPTS_VALUE) ?
588 AV_NOPTS_VALUE : end - start;
589 } else if (av_strstart(p, "lang:", &p)) {
590 if (s->nb_streams > 0) {
591 get_word(buf1, sizeof(buf1), &p);
592 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593 if (rtsp_st->stream_index >= 0) {
594 st = s->streams[rtsp_st->stream_index];
595 av_dict_set(&st->metadata, "language", buf1, 0);
598 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
599 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
601 rt->transport = RTSP_TRANSPORT_RDT;
602 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
604 st = s->streams[s->nb_streams - 1];
605 st->codecpar->sample_rate = atoi(p);
606 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
608 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
609 get_word(buf1, sizeof(buf1), &p); // ignore tag
610 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
611 p += strspn(p, SPACE_CHARS);
612 if (av_strstart(p, "inline:", &p))
613 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
614 } else if (av_strstart(p, "source-filter:", &p)) {
616 get_word(buf1, sizeof(buf1), &p);
617 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
619 exclude = !strcmp(buf1, "excl");
621 get_word(buf1, sizeof(buf1), &p);
622 if (strcmp(buf1, "IN") != 0)
624 get_word(buf1, sizeof(buf1), &p);
625 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
627 // not checking that the destination address actually matches or is wildcard
628 get_word(buf1, sizeof(buf1), &p);
631 rtsp_src = av_mallocz(sizeof(*rtsp_src));
634 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
636 if (s->nb_streams == 0) {
637 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
639 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
640 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
643 if (s->nb_streams == 0) {
644 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
646 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
647 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
652 if (rt->server_type == RTSP_SERVER_WMS)
653 ff_wms_parse_sdp_a_line(s, p);
654 if (s->nb_streams > 0) {
655 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
657 if (rt->server_type == RTSP_SERVER_REAL)
658 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
660 if (rtsp_st->dynamic_handler &&
661 rtsp_st->dynamic_handler->parse_sdp_a_line)
662 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
663 rtsp_st->stream_index,
664 rtsp_st->dynamic_protocol_context, buf);
671 int ff_sdp_parse(AVFormatContext *s, const char *content)
673 RTSPState *rt = s->priv_data;
676 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
677 * contain long SDP lines containing complete ASF Headers (several
678 * kB) or arrays of MDPR (RM stream descriptor) headers plus
679 * "rulebooks" describing their properties. Therefore, the SDP line
682 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
683 * in rtpdec_xiph.c. */
685 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
689 p += strspn(p, SPACE_CHARS);
697 /* get the content */
699 while (*p != '\n' && *p != '\r' && *p != '\0') {
700 if ((q - buf) < sizeof(buf) - 1)
705 sdp_parse_line(s, s1, letter, buf);
707 while (*p != '\n' && *p != '\0')
713 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
714 av_freep(&s1->default_include_source_addrs[i]);
715 av_freep(&s1->default_include_source_addrs);
716 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
717 av_freep(&s1->default_exclude_source_addrs[i]);
718 av_freep(&s1->default_exclude_source_addrs);
720 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
721 if (!rt->p) return AVERROR(ENOMEM);
724 #endif /* CONFIG_RTPDEC */
726 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
728 RTSPState *rt = s->priv_data;
731 for (i = 0; i < rt->nb_rtsp_streams; i++) {
732 RTSPStream *rtsp_st = rt->rtsp_streams[i];
735 if (rtsp_st->transport_priv) {
737 AVFormatContext *rtpctx = rtsp_st->transport_priv;
738 av_write_trailer(rtpctx);
739 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
740 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
741 ff_rtsp_tcp_write_packet(s, rtsp_st);
742 ffio_free_dyn_buf(&rtpctx->pb);
744 avio_closep(&rtpctx->pb);
746 avformat_free_context(rtpctx);
747 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
748 ff_rdt_parse_close(rtsp_st->transport_priv);
749 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
750 ff_rtp_parse_close(rtsp_st->transport_priv);
752 rtsp_st->transport_priv = NULL;
753 if (rtsp_st->rtp_handle)
754 ffurl_close(rtsp_st->rtp_handle);
755 rtsp_st->rtp_handle = NULL;
759 /* close and free RTSP streams */
760 void ff_rtsp_close_streams(AVFormatContext *s)
762 RTSPState *rt = s->priv_data;
766 ff_rtsp_undo_setup(s, 0);
767 for (i = 0; i < rt->nb_rtsp_streams; i++) {
768 rtsp_st = rt->rtsp_streams[i];
770 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
771 if (rtsp_st->dynamic_handler->close)
772 rtsp_st->dynamic_handler->close(
773 rtsp_st->dynamic_protocol_context);
774 av_free(rtsp_st->dynamic_protocol_context);
776 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
777 av_freep(&rtsp_st->include_source_addrs[j]);
778 av_freep(&rtsp_st->include_source_addrs);
779 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
780 av_freep(&rtsp_st->exclude_source_addrs[j]);
781 av_freep(&rtsp_st->exclude_source_addrs);
786 av_freep(&rt->rtsp_streams);
788 avformat_close_input(&rt->asf_ctx);
790 if (CONFIG_RTPDEC && rt->ts)
791 avpriv_mpegts_parse_close(rt->ts);
793 av_freep(&rt->recvbuf);
796 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
798 RTSPState *rt = s->priv_data;
800 int reordering_queue_size = rt->reordering_queue_size;
801 if (reordering_queue_size < 0) {
802 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
803 reordering_queue_size = 0;
805 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
808 /* open the RTP context */
809 if (rtsp_st->stream_index >= 0)
810 st = s->streams[rtsp_st->stream_index];
812 s->ctx_flags |= AVFMTCTX_NOHEADER;
814 if (CONFIG_RTSP_MUXER && s->oformat && st) {
815 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
816 s, st, rtsp_st->rtp_handle,
817 RTSP_TCP_MAX_PACKET_SIZE,
818 rtsp_st->stream_index);
819 /* Ownership of rtp_handle is passed to the rtp mux context */
820 rtsp_st->rtp_handle = NULL;
823 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
824 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
825 return 0; // Don't need to open any parser here
826 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
827 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
828 rtsp_st->dynamic_protocol_context,
829 rtsp_st->dynamic_handler);
830 else if (CONFIG_RTPDEC)
831 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
832 rtsp_st->sdp_payload_type,
833 reordering_queue_size);
835 if (!rtsp_st->transport_priv) {
836 return AVERROR(ENOMEM);
837 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
838 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
839 rtpctx->ssrc = rtsp_st->ssrc;
840 if (rtsp_st->dynamic_handler) {
841 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
842 rtsp_st->dynamic_protocol_context,
843 rtsp_st->dynamic_handler);
845 if (rtsp_st->crypto_suite[0])
846 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
847 rtsp_st->crypto_suite,
848 rtsp_st->crypto_params);
854 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
855 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
862 q += strspn(q, SPACE_CHARS);
863 v = strtol(q, &p, 10);
867 v = strtol(p, &p, 10);
876 /* XXX: only one transport specification is parsed */
877 static void rtsp_parse_transport(AVFormatContext *s,
878 RTSPMessageHeader *reply, const char *p)
880 char transport_protocol[16];
882 char lower_transport[16];
884 RTSPTransportField *th;
887 reply->nb_transports = 0;
890 p += strspn(p, SPACE_CHARS);
894 th = &reply->transports[reply->nb_transports];
896 get_word_sep(transport_protocol, sizeof(transport_protocol),
898 if (!av_strcasecmp (transport_protocol, "rtp")) {
899 get_word_sep(profile, sizeof(profile), "/;,", &p);
900 lower_transport[0] = '\0';
901 /* rtp/avp/<protocol> */
903 get_word_sep(lower_transport, sizeof(lower_transport),
906 th->transport = RTSP_TRANSPORT_RTP;
907 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
908 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
909 /* x-pn-tng/<protocol> */
910 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
912 th->transport = RTSP_TRANSPORT_RDT;
913 } else if (!av_strcasecmp(transport_protocol, "raw")) {
914 get_word_sep(profile, sizeof(profile), "/;,", &p);
915 lower_transport[0] = '\0';
916 /* raw/raw/<protocol> */
918 get_word_sep(lower_transport, sizeof(lower_transport),
921 th->transport = RTSP_TRANSPORT_RAW;
923 if (!av_strcasecmp(lower_transport, "TCP"))
924 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
926 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
930 /* get each parameter */
931 while (*p != '\0' && *p != ',') {
932 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
933 if (!strcmp(parameter, "port")) {
936 rtsp_parse_range(&th->port_min, &th->port_max, &p);
938 } else if (!strcmp(parameter, "client_port")) {
941 rtsp_parse_range(&th->client_port_min,
942 &th->client_port_max, &p);
944 } else if (!strcmp(parameter, "server_port")) {
947 rtsp_parse_range(&th->server_port_min,
948 &th->server_port_max, &p);
950 } else if (!strcmp(parameter, "interleaved")) {
953 rtsp_parse_range(&th->interleaved_min,
954 &th->interleaved_max, &p);
956 } else if (!strcmp(parameter, "multicast")) {
957 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
958 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
959 } else if (!strcmp(parameter, "ttl")) {
963 th->ttl = strtol(p, &end, 10);
966 } else if (!strcmp(parameter, "destination")) {
969 get_word_sep(buf, sizeof(buf), ";,", &p);
970 get_sockaddr(s, buf, &th->destination);
972 } else if (!strcmp(parameter, "source")) {
975 get_word_sep(buf, sizeof(buf), ";,", &p);
976 av_strlcpy(th->source, buf, sizeof(th->source));
978 } else if (!strcmp(parameter, "mode")) {
981 get_word_sep(buf, sizeof(buf), ";, ", &p);
982 if (!strcmp(buf, "record") ||
983 !strcmp(buf, "receive"))
988 while (*p != ';' && *p != '\0' && *p != ',')
996 reply->nb_transports++;
997 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
1002 static void handle_rtp_info(RTSPState *rt, const char *url,
1003 uint32_t seq, uint32_t rtptime)
1006 if (!rtptime || !url[0])
1008 if (rt->transport != RTSP_TRANSPORT_RTP)
1010 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1011 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1012 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1015 if (!strcmp(rtsp_st->control_url, url)) {
1016 rtpctx->base_timestamp = rtptime;
1022 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1025 char key[20], value[1024], url[1024] = "";
1026 uint32_t seq = 0, rtptime = 0;
1029 p += strspn(p, SPACE_CHARS);
1032 get_word_sep(key, sizeof(key), "=", &p);
1036 get_word_sep(value, sizeof(value), ";, ", &p);
1038 if (!strcmp(key, "url"))
1039 av_strlcpy(url, value, sizeof(url));
1040 else if (!strcmp(key, "seq"))
1041 seq = strtoul(value, NULL, 10);
1042 else if (!strcmp(key, "rtptime"))
1043 rtptime = strtoul(value, NULL, 10);
1045 handle_rtp_info(rt, url, seq, rtptime);
1054 handle_rtp_info(rt, url, seq, rtptime);
1057 void ff_rtsp_parse_line(AVFormatContext *s,
1058 RTSPMessageHeader *reply, const char *buf,
1059 RTSPState *rt, const char *method)
1063 /* NOTE: we do case independent match for broken servers */
1065 if (av_stristart(p, "Session:", &p)) {
1067 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1068 if (av_stristart(p, ";timeout=", &p) &&
1069 (t = strtol(p, NULL, 10)) > 0) {
1072 } else if (av_stristart(p, "Content-Length:", &p)) {
1073 reply->content_length = strtol(p, NULL, 10);
1074 } else if (av_stristart(p, "Transport:", &p)) {
1075 rtsp_parse_transport(s, reply, p);
1076 } else if (av_stristart(p, "CSeq:", &p)) {
1077 reply->seq = strtol(p, NULL, 10);
1078 } else if (av_stristart(p, "Range:", &p)) {
1079 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1080 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1081 p += strspn(p, SPACE_CHARS);
1082 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1083 } else if (av_stristart(p, "Server:", &p)) {
1084 p += strspn(p, SPACE_CHARS);
1085 av_strlcpy(reply->server, p, sizeof(reply->server));
1086 } else if (av_stristart(p, "Notice:", &p) ||
1087 av_stristart(p, "X-Notice:", &p)) {
1088 reply->notice = strtol(p, NULL, 10);
1089 } else if (av_stristart(p, "Location:", &p)) {
1090 p += strspn(p, SPACE_CHARS);
1091 av_strlcpy(reply->location, p , sizeof(reply->location));
1092 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1093 p += strspn(p, SPACE_CHARS);
1094 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1095 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1096 p += strspn(p, SPACE_CHARS);
1097 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1098 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1099 p += strspn(p, SPACE_CHARS);
1100 if (method && !strcmp(method, "DESCRIBE"))
1101 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1102 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1103 p += strspn(p, SPACE_CHARS);
1104 if (method && !strcmp(method, "PLAY"))
1105 rtsp_parse_rtp_info(rt, p);
1106 } else if (av_stristart(p, "Public:", &p) && rt) {
1107 if (strstr(p, "GET_PARAMETER") &&
1108 method && !strcmp(method, "OPTIONS"))
1109 rt->get_parameter_supported = 1;
1110 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1111 p += strspn(p, SPACE_CHARS);
1112 rt->accept_dynamic_rate = atoi(p);
1113 } else if (av_stristart(p, "Content-Type:", &p)) {
1114 p += strspn(p, SPACE_CHARS);
1115 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1119 /* skip a RTP/TCP interleaved packet */
1120 void ff_rtsp_skip_packet(AVFormatContext *s)
1122 RTSPState *rt = s->priv_data;
1126 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1129 len = AV_RB16(buf + 1);
1131 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1136 if (len1 > sizeof(buf))
1138 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1145 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1146 unsigned char **content_ptr,
1147 int return_on_interleaved_data, const char *method)
1149 RTSPState *rt = s->priv_data;
1150 char buf[4096], buf1[1024], *q;
1153 int ret, content_length, line_count = 0, request = 0;
1154 unsigned char *content = NULL;
1160 memset(reply, 0, sizeof(*reply));
1162 /* parse reply (XXX: use buffers) */
1163 rt->last_reply[0] = '\0';
1167 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1168 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1173 if (ch == '$' && q == buf) {
1174 if (return_on_interleaved_data) {
1177 ff_rtsp_skip_packet(s);
1178 } else if (ch != '\r') {
1179 if ((q - buf) < sizeof(buf) - 1)
1185 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1187 /* test if last line */
1191 if (line_count == 0) {
1192 /* get reply code */
1193 get_word(buf1, sizeof(buf1), &p);
1194 if (!strncmp(buf1, "RTSP/", 5)) {
1195 get_word(buf1, sizeof(buf1), &p);
1196 reply->status_code = atoi(buf1);
1197 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1199 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1200 get_word(buf1, sizeof(buf1), &p); // object
1204 ff_rtsp_parse_line(s, reply, p, rt, method);
1205 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1206 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1211 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1212 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1214 content_length = reply->content_length;
1215 if (content_length > 0) {
1216 /* leave some room for a trailing '\0' (useful for simple parsing) */
1217 content = av_malloc(content_length + 1);
1219 return AVERROR(ENOMEM);
1220 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1221 content[content_length] = '\0';
1224 *content_ptr = content;
1230 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1231 const char* ptr = buf;
1233 if (!strcmp(reply->reason, "OPTIONS")) {
1234 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1236 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1237 if (reply->session_id[0])
1238 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1241 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1243 av_strlcat(buf, "\r\n", sizeof(buf));
1245 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1246 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1249 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1251 rt->last_cmd_time = av_gettime_relative();
1252 /* Even if the request from the server had data, it is not the data
1253 * that the caller wants or expects. The memory could also be leaked
1254 * if the actual following reply has content data. */
1256 av_freep(content_ptr);
1257 /* If method is set, this is called from ff_rtsp_send_cmd,
1258 * where a reply to exactly this request is awaited. For
1259 * callers from within packet receiving, we just want to
1260 * return to the caller and go back to receiving packets. */
1266 if (rt->seq != reply->seq) {
1267 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1268 rt->seq, reply->seq);
1272 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1273 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1274 reply->notice == 2306 /* Continuous Feed Terminated */) {
1275 rt->state = RTSP_STATE_IDLE;
1276 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1277 return AVERROR(EIO); /* data or server error */
1278 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1279 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1280 return AVERROR(EPERM);
1286 * Send a command to the RTSP server without waiting for the reply.
1288 * @param s RTSP (de)muxer context
1289 * @param method the method for the request
1290 * @param url the target url for the request
1291 * @param headers extra header lines to include in the request
1292 * @param send_content if non-null, the data to send as request body content
1293 * @param send_content_length the length of the send_content data, or 0 if
1294 * send_content is null
1296 * @return zero if success, nonzero otherwise
1298 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1299 const char *method, const char *url,
1300 const char *headers,
1301 const unsigned char *send_content,
1302 int send_content_length)
1304 RTSPState *rt = s->priv_data;
1305 char buf[4096], *out_buf;
1306 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1308 /* Add in RTSP headers */
1311 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1313 av_strlcat(buf, headers, sizeof(buf));
1314 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1315 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1316 if (rt->session_id[0] != '\0' && (!headers ||
1317 !strstr(headers, "\nIf-Match:"))) {
1318 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1321 char *str = ff_http_auth_create_response(&rt->auth_state,
1322 rt->auth, url, method);
1324 av_strlcat(buf, str, sizeof(buf));
1327 if (send_content_length > 0 && send_content)
1328 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1329 av_strlcat(buf, "\r\n", sizeof(buf));
1331 /* base64 encode rtsp if tunneling */
1332 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1333 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1334 out_buf = base64buf;
1337 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1339 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1340 if (send_content_length > 0 && send_content) {
1341 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1342 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1343 "with content data not supported\n");
1344 return AVERROR_PATCHWELCOME;
1346 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1348 rt->last_cmd_time = av_gettime_relative();
1353 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1354 const char *url, const char *headers)
1356 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1359 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1360 const char *headers, RTSPMessageHeader *reply,
1361 unsigned char **content_ptr)
1363 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1364 content_ptr, NULL, 0);
1367 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1368 const char *method, const char *url,
1370 RTSPMessageHeader *reply,
1371 unsigned char **content_ptr,
1372 const unsigned char *send_content,
1373 int send_content_length)
1375 RTSPState *rt = s->priv_data;
1376 HTTPAuthType cur_auth_type;
1377 int ret, attempts = 0;
1380 cur_auth_type = rt->auth_state.auth_type;
1381 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1383 send_content_length)))
1386 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1390 if (reply->status_code == 401 &&
1391 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1392 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1395 if (reply->status_code > 400){
1396 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1400 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1406 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1407 int lower_transport, const char *real_challenge)
1409 RTSPState *rt = s->priv_data;
1410 int rtx = 0, j, i, err, interleave = 0, port_off;
1411 RTSPStream *rtsp_st;
1412 RTSPMessageHeader reply1, *reply = &reply1;
1414 const char *trans_pref;
1416 if (rt->transport == RTSP_TRANSPORT_RDT)
1417 trans_pref = "x-pn-tng";
1418 else if (rt->transport == RTSP_TRANSPORT_RAW)
1419 trans_pref = "RAW/RAW";
1421 trans_pref = "RTP/AVP";
1423 /* default timeout: 1 minute */
1426 /* Choose a random starting offset within the first half of the
1427 * port range, to allow for a number of ports to try even if the offset
1428 * happens to be at the end of the random range. */
1429 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1430 /* even random offset */
1431 port_off -= port_off & 0x01;
1433 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1434 char transport[2048];
1437 * WMS serves all UDP data over a single connection, the RTX, which
1438 * isn't necessarily the first in the SDP but has to be the first
1439 * to be set up, else the second/third SETUP will fail with a 461.
1441 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1442 rt->server_type == RTSP_SERVER_WMS) {
1445 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1446 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1448 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1452 if (rtx == rt->nb_rtsp_streams)
1453 return -1; /* no RTX found */
1454 rtsp_st = rt->rtsp_streams[rtx];
1456 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1458 rtsp_st = rt->rtsp_streams[i];
1461 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1464 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1465 port = reply->transports[0].client_port_min;
1469 /* first try in specified port range */
1470 while (j <= rt->rtp_port_max) {
1471 AVDictionary *opts = map_to_opts(rt);
1473 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1474 "?localport=%d", j);
1475 /* we will use two ports per rtp stream (rtp and rtcp) */
1477 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1478 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
1480 av_dict_free(&opts);
1485 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1490 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1492 snprintf(transport, sizeof(transport) - 1,
1493 "%s/UDP;", trans_pref);
1494 if (rt->server_type != RTSP_SERVER_REAL)
1495 av_strlcat(transport, "unicast;", sizeof(transport));
1496 av_strlcatf(transport, sizeof(transport),
1497 "client_port=%d", port);
1498 if (rt->transport == RTSP_TRANSPORT_RTP &&
1499 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1500 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1504 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1505 /* For WMS streams, the application streams are only used for
1506 * UDP. When trying to set it up for TCP streams, the server
1507 * will return an error. Therefore, we skip those streams. */
1508 if (rt->server_type == RTSP_SERVER_WMS &&
1509 (rtsp_st->stream_index < 0 ||
1510 s->streams[rtsp_st->stream_index]->codecpar->codec_type ==
1513 snprintf(transport, sizeof(transport) - 1,
1514 "%s/TCP;", trans_pref);
1515 if (rt->transport != RTSP_TRANSPORT_RDT)
1516 av_strlcat(transport, "unicast;", sizeof(transport));
1517 av_strlcatf(transport, sizeof(transport),
1518 "interleaved=%d-%d",
1519 interleave, interleave + 1);
1523 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1524 snprintf(transport, sizeof(transport) - 1,
1525 "%s/UDP;multicast", trans_pref);
1528 av_strlcat(transport, ";mode=record", sizeof(transport));
1529 } else if (rt->server_type == RTSP_SERVER_REAL ||
1530 rt->server_type == RTSP_SERVER_WMS)
1531 av_strlcat(transport, ";mode=play", sizeof(transport));
1532 snprintf(cmd, sizeof(cmd),
1533 "Transport: %s\r\n",
1535 if (rt->accept_dynamic_rate)
1536 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1537 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1538 char real_res[41], real_csum[9];
1539 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1541 av_strlcatf(cmd, sizeof(cmd),
1543 "RealChallenge2: %s, sd=%s\r\n",
1544 rt->session_id, real_res, real_csum);
1546 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1547 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1550 } else if (reply->status_code != RTSP_STATUS_OK ||
1551 reply->nb_transports != 1) {
1552 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1556 /* XXX: same protocol for all streams is required */
1558 if (reply->transports[0].lower_transport != rt->lower_transport ||
1559 reply->transports[0].transport != rt->transport) {
1560 err = AVERROR_INVALIDDATA;
1564 rt->lower_transport = reply->transports[0].lower_transport;
1565 rt->transport = reply->transports[0].transport;
1568 /* Fail if the server responded with another lower transport mode
1569 * than what we requested. */
1570 if (reply->transports[0].lower_transport != lower_transport) {
1571 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1572 err = AVERROR_INVALIDDATA;
1576 switch(reply->transports[0].lower_transport) {
1577 case RTSP_LOWER_TRANSPORT_TCP:
1578 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1579 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1582 case RTSP_LOWER_TRANSPORT_UDP: {
1583 char url[1024], options[30] = "";
1584 const char *peer = host;
1586 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1587 av_strlcpy(options, "?connect=1", sizeof(options));
1588 /* Use source address if specified */
1589 if (reply->transports[0].source[0])
1590 peer = reply->transports[0].source;
1591 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1592 reply->transports[0].server_port_min, "%s", options);
1593 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1594 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1595 err = AVERROR_INVALIDDATA;
1600 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1601 char url[1024], namebuf[50], optbuf[20] = "";
1602 struct sockaddr_storage addr;
1605 if (reply->transports[0].destination.ss_family) {
1606 addr = reply->transports[0].destination;
1607 port = reply->transports[0].port_min;
1608 ttl = reply->transports[0].ttl;
1610 addr = rtsp_st->sdp_ip;
1611 port = rtsp_st->sdp_port;
1612 ttl = rtsp_st->sdp_ttl;
1615 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1616 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1617 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1618 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1619 port, "%s", optbuf);
1620 if (ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1621 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL) < 0) {
1622 err = AVERROR_INVALIDDATA;
1629 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1633 if (rt->nb_rtsp_streams && reply->timeout > 0)
1634 rt->timeout = reply->timeout;
1636 if (rt->server_type == RTSP_SERVER_REAL)
1637 rt->need_subscription = 1;
1642 ff_rtsp_undo_setup(s, 0);
1646 void ff_rtsp_close_connections(AVFormatContext *s)
1648 RTSPState *rt = s->priv_data;
1649 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1650 ffurl_close(rt->rtsp_hd);
1651 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1654 int ff_rtsp_connect(AVFormatContext *s)
1656 RTSPState *rt = s->priv_data;
1657 char proto[128], host[1024], path[1024];
1658 char tcpname[1024], cmd[2048], auth[128];
1659 const char *lower_rtsp_proto = "tcp";
1660 int port, err, tcp_fd;
1661 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1662 int lower_transport_mask = 0;
1663 int default_port = RTSP_DEFAULT_PORT;
1664 char real_challenge[64] = "";
1665 struct sockaddr_storage peer;
1666 socklen_t peer_len = sizeof(peer);
1668 if (rt->rtp_port_max < rt->rtp_port_min) {
1669 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1670 "than min port %d\n", rt->rtp_port_max,
1672 return AVERROR(EINVAL);
1675 if (!ff_network_init())
1676 return AVERROR(EIO);
1678 if (s->max_delay < 0) /* Not set by the caller */
1679 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1681 rt->control_transport = RTSP_MODE_PLAIN;
1682 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1683 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1684 rt->control_transport = RTSP_MODE_TUNNEL;
1686 /* Only pass through valid flags from here */
1687 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1690 /* extract hostname and port */
1691 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1692 host, sizeof(host), &port, path, sizeof(path), s->filename);
1694 if (!strcmp(proto, "rtsps")) {
1695 lower_rtsp_proto = "tls";
1696 default_port = RTSPS_DEFAULT_PORT;
1697 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1701 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1704 port = default_port;
1706 lower_transport_mask = rt->lower_transport_mask;
1708 if (!lower_transport_mask)
1709 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1712 /* Only UDP or TCP - UDP multicast isn't supported. */
1713 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1714 (1 << RTSP_LOWER_TRANSPORT_TCP);
1715 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1716 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1717 "only UDP and TCP are supported for output.\n");
1718 err = AVERROR(EINVAL);
1723 /* Construct the URI used in request; this is similar to s->filename,
1724 * but with authentication credentials removed and RTSP specific options
1726 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1727 host, port, "%s", path);
1729 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1730 /* set up initial handshake for tunneling */
1731 char httpname[1024];
1732 char sessioncookie[17];
1735 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1736 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1737 av_get_random_seed(), av_get_random_seed());
1740 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1741 &s->interrupt_callback) < 0) {
1746 /* generate GET headers */
1747 snprintf(headers, sizeof(headers),
1748 "x-sessioncookie: %s\r\n"
1749 "Accept: application/x-rtsp-tunnelled\r\n"
1750 "Pragma: no-cache\r\n"
1751 "Cache-Control: no-cache\r\n",
1753 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1755 /* complete the connection */
1756 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1762 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1763 &s->interrupt_callback) < 0 ) {
1768 /* generate POST headers */
1769 snprintf(headers, sizeof(headers),
1770 "x-sessioncookie: %s\r\n"
1771 "Content-Type: application/x-rtsp-tunnelled\r\n"
1772 "Pragma: no-cache\r\n"
1773 "Cache-Control: no-cache\r\n"
1774 "Content-Length: 32767\r\n"
1775 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1777 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1778 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1780 /* Initialize the authentication state for the POST session. The HTTP
1781 * protocol implementation doesn't properly handle multi-pass
1782 * authentication for POST requests, since it would require one of
1784 * - implementing Expect: 100-continue, which many HTTP servers
1785 * don't support anyway, even less the RTSP servers that do HTTP
1787 * - sending the whole POST data until getting a 401 reply specifying
1788 * what authentication method to use, then resending all that data
1789 * - waiting for potential 401 replies directly after sending the
1790 * POST header (waiting for some unspecified time)
1791 * Therefore, we copy the full auth state, which works for both basic
1792 * and digest. (For digest, we would have to synchronize the nonce
1793 * count variable between the two sessions, if we'd do more requests
1794 * with the original session, though.)
1796 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1798 /* complete the connection */
1799 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1805 /* open the tcp connection */
1806 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1808 "?timeout=%d", rt->stimeout);
1809 if ((ret = ffurl_open_whitelist(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1810 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL)) < 0) {
1814 rt->rtsp_hd_out = rt->rtsp_hd;
1818 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1823 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1824 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1825 NULL, 0, NI_NUMERICHOST);
1828 /* request options supported by the server; this also detects server
1830 for (rt->server_type = RTSP_SERVER_RTP;;) {
1832 if (rt->server_type == RTSP_SERVER_REAL)
1835 * The following entries are required for proper
1836 * streaming from a Realmedia server. They are
1837 * interdependent in some way although we currently
1838 * don't quite understand how. Values were copied
1839 * from mplayer SVN r23589.
1840 * ClientChallenge is a 16-byte ID in hex
1841 * CompanyID is a 16-byte ID in base64
1843 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1844 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1845 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1846 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1848 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1849 if (reply->status_code != RTSP_STATUS_OK) {
1850 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1854 /* detect server type if not standard-compliant RTP */
1855 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1856 rt->server_type = RTSP_SERVER_REAL;
1858 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1859 rt->server_type = RTSP_SERVER_WMS;
1860 } else if (rt->server_type == RTSP_SERVER_REAL)
1861 strcpy(real_challenge, reply->real_challenge);
1865 if (CONFIG_RTSP_DEMUXER && s->iformat)
1866 err = ff_rtsp_setup_input_streams(s, reply);
1867 else if (CONFIG_RTSP_MUXER)
1868 err = ff_rtsp_setup_output_streams(s, host);
1875 int lower_transport = ff_log2_tab[lower_transport_mask &
1876 ~(lower_transport_mask - 1)];
1878 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1879 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1880 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1882 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1883 rt->server_type == RTSP_SERVER_REAL ?
1884 real_challenge : NULL);
1887 lower_transport_mask &= ~(1 << lower_transport);
1888 if (lower_transport_mask == 0 && err == 1) {
1889 err = AVERROR(EPROTONOSUPPORT);
1894 rt->lower_transport_mask = lower_transport_mask;
1895 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1896 rt->state = RTSP_STATE_IDLE;
1897 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1900 ff_rtsp_close_streams(s);
1901 ff_rtsp_close_connections(s);
1902 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1903 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1904 rt->session_id[0] = '\0';
1905 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1913 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1916 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1917 uint8_t *buf, int buf_size, int64_t wait_end)
1919 RTSPState *rt = s->priv_data;
1920 RTSPStream *rtsp_st;
1921 int n, i, ret, tcp_fd, timeout_cnt = 0;
1923 struct pollfd *p = rt->p;
1924 int *fds = NULL, fdsnum, fdsidx;
1927 if (ff_check_interrupt(&s->interrupt_callback))
1928 return AVERROR_EXIT;
1929 if (wait_end && wait_end - av_gettime_relative() < 0)
1930 return AVERROR(EAGAIN);
1933 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1934 p[max_p].fd = tcp_fd;
1935 p[max_p++].events = POLLIN;
1939 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1940 rtsp_st = rt->rtsp_streams[i];
1941 if (rtsp_st->rtp_handle) {
1942 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1944 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1948 av_log(s, AV_LOG_ERROR,
1949 "Number of fds %d not supported\n", fdsnum);
1950 return AVERROR_INVALIDDATA;
1952 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1953 p[max_p].fd = fds[fdsidx];
1954 p[max_p++].events = POLLIN;
1959 n = poll(p, max_p, POLL_TIMEOUT_MS);
1961 int j = 1 - (tcp_fd == -1);
1963 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1964 rtsp_st = rt->rtsp_streams[i];
1965 if (rtsp_st->rtp_handle) {
1966 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1967 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1969 *prtsp_st = rtsp_st;
1976 #if CONFIG_RTSP_DEMUXER
1977 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1978 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1979 if (rt->state == RTSP_STATE_STREAMING) {
1980 if (!ff_rtsp_parse_streaming_commands(s))
1983 av_log(s, AV_LOG_WARNING,
1984 "Unable to answer to TEARDOWN\n");
1988 RTSPMessageHeader reply;
1989 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1992 /* XXX: parse message */
1993 if (rt->state != RTSP_STATE_STREAMING)
1998 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1999 return AVERROR(ETIMEDOUT);
2000 } else if (n < 0 && errno != EINTR)
2001 return AVERROR(errno);
2005 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
2006 const uint8_t *buf, int len)
2008 RTSPState *rt = s->priv_data;
2012 if (rt->nb_rtsp_streams == 1) {
2013 *rtsp_st = rt->rtsp_streams[0];
2016 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2017 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2019 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2020 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2023 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2024 *rtsp_st = rt->rtsp_streams[i];
2031 av_log(s, AV_LOG_WARNING,
2032 "Unable to pick stream for packet - SSRC not known for "
2034 return AVERROR(EAGAIN);
2037 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2038 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2039 *rtsp_st = rt->rtsp_streams[i];
2045 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2046 return AVERROR(EAGAIN);
2049 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2051 RTSPState *rt = s->priv_data;
2053 RTSPStream *rtsp_st, *first_queue_st = NULL;
2054 int64_t wait_end = 0;
2056 if (rt->nb_byes == rt->nb_rtsp_streams)
2059 /* get next frames from the same RTP packet */
2060 if (rt->cur_transport_priv) {
2061 if (rt->transport == RTSP_TRANSPORT_RDT) {
2062 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2063 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2064 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2065 } else if (CONFIG_RTPDEC && rt->ts) {
2066 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2068 rt->recvbuf_pos += ret;
2069 ret = rt->recvbuf_pos < rt->recvbuf_len;
2074 rt->cur_transport_priv = NULL;
2076 } else if (ret == 1) {
2079 rt->cur_transport_priv = NULL;
2083 if (rt->transport == RTSP_TRANSPORT_RTP) {
2085 int64_t first_queue_time = 0;
2086 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2087 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2091 queue_time = ff_rtp_queued_packet_time(rtpctx);
2092 if (queue_time && (queue_time - first_queue_time < 0 ||
2093 !first_queue_time)) {
2094 first_queue_time = queue_time;
2095 first_queue_st = rt->rtsp_streams[i];
2098 if (first_queue_time) {
2099 wait_end = first_queue_time + s->max_delay;
2102 first_queue_st = NULL;
2106 /* read next RTP packet */
2108 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2110 return AVERROR(ENOMEM);
2113 switch(rt->lower_transport) {
2115 #if CONFIG_RTSP_DEMUXER
2116 case RTSP_LOWER_TRANSPORT_TCP:
2117 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2120 case RTSP_LOWER_TRANSPORT_UDP:
2121 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2122 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2123 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2124 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2126 case RTSP_LOWER_TRANSPORT_CUSTOM:
2127 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2128 wait_end && wait_end < av_gettime_relative())
2129 len = AVERROR(EAGAIN);
2131 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2132 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2133 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2134 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2137 if (len == AVERROR(EAGAIN) && first_queue_st &&
2138 rt->transport == RTSP_TRANSPORT_RTP) {
2139 av_log(s, AV_LOG_WARNING,
2140 "max delay reached. need to consume packet\n");
2141 rtsp_st = first_queue_st;
2142 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2149 if (rt->transport == RTSP_TRANSPORT_RDT) {
2150 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2151 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2152 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2153 if (rtsp_st->feedback) {
2154 AVIOContext *pb = NULL;
2155 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2157 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2160 /* Either bad packet, or a RTCP packet. Check if the
2161 * first_rtcp_ntp_time field was initialized. */
2162 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2163 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2164 /* first_rtcp_ntp_time has been initialized for this stream,
2165 * copy the same value to all other uninitialized streams,
2166 * in order to map their timestamp origin to the same ntp time
2169 AVStream *st = NULL;
2170 if (rtsp_st->stream_index >= 0)
2171 st = s->streams[rtsp_st->stream_index];
2172 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2173 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2174 AVStream *st2 = NULL;
2175 if (rt->rtsp_streams[i]->stream_index >= 0)
2176 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2177 if (rtpctx2 && st && st2 &&
2178 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2179 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2180 rtpctx2->rtcp_ts_offset = av_rescale_q(
2181 rtpctx->rtcp_ts_offset, st->time_base,
2185 // Make real NTP start time available in AVFormatContext
2186 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2187 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2189 s->start_time_realtime -=
2190 av_rescale (rtpctx->rtcp_ts_offset,
2191 (uint64_t) rtpctx->st->time_base.num * 1000000,
2192 rtpctx->st->time_base.den);
2196 if (ret == -RTCP_BYE) {
2199 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2200 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2202 if (rt->nb_byes == rt->nb_rtsp_streams)
2206 } else if (CONFIG_RTPDEC && rt->ts) {
2207 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2210 rt->recvbuf_len = len;
2211 rt->recvbuf_pos = ret;
2212 rt->cur_transport_priv = rt->ts;
2219 return AVERROR_INVALIDDATA;
2225 /* more packets may follow, so we save the RTP context */
2226 rt->cur_transport_priv = rtsp_st->transport_priv;
2230 #endif /* CONFIG_RTPDEC */
2232 #if CONFIG_SDP_DEMUXER
2233 static int sdp_probe(AVProbeData *p1)
2235 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2237 /* we look for a line beginning "c=IN IP" */
2238 while (p < p_end && *p != '\0') {
2239 if (sizeof("c=IN IP") - 1 < p_end - p &&
2240 av_strstart(p, "c=IN IP", NULL))
2241 return AVPROBE_SCORE_EXTENSION;
2243 while (p < p_end - 1 && *p != '\n') p++;
2252 static void append_source_addrs(char *buf, int size, const char *name,
2253 int count, struct RTSPSource **addrs)
2258 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2259 for (i = 1; i < count; i++)
2260 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2263 static int sdp_read_header(AVFormatContext *s)
2265 RTSPState *rt = s->priv_data;
2266 RTSPStream *rtsp_st;
2271 if (!ff_network_init())
2272 return AVERROR(EIO);
2274 if (s->max_delay < 0) /* Not set by the caller */
2275 s->max_delay = DEFAULT_REORDERING_DELAY;
2276 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2277 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2279 /* read the whole sdp file */
2280 /* XXX: better loading */
2281 content = av_malloc(SDP_MAX_SIZE);
2283 return AVERROR(ENOMEM);
2284 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2287 return AVERROR_INVALIDDATA;
2289 content[size] ='\0';
2291 err = ff_sdp_parse(s, content);
2295 /* open each RTP stream */
2296 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2298 rtsp_st = rt->rtsp_streams[i];
2300 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2301 AVDictionary *opts = map_to_opts(rt);
2303 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2304 sizeof(rtsp_st->sdp_ip),
2305 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2307 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2309 av_dict_free(&opts);
2312 ff_url_join(url, sizeof(url), "rtp", NULL,
2313 namebuf, rtsp_st->sdp_port,
2314 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2315 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2316 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2317 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2319 append_source_addrs(url, sizeof(url), "sources",
2320 rtsp_st->nb_include_source_addrs,
2321 rtsp_st->include_source_addrs);
2322 append_source_addrs(url, sizeof(url), "block",
2323 rtsp_st->nb_exclude_source_addrs,
2324 rtsp_st->exclude_source_addrs);
2325 err = ffurl_open_whitelist(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ,
2326 &s->interrupt_callback, &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
2328 av_dict_free(&opts);
2331 err = AVERROR_INVALIDDATA;
2335 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2340 ff_rtsp_close_streams(s);
2345 static int sdp_read_close(AVFormatContext *s)
2347 ff_rtsp_close_streams(s);
2352 static const AVClass sdp_demuxer_class = {
2353 .class_name = "SDP demuxer",
2354 .item_name = av_default_item_name,
2355 .option = sdp_options,
2356 .version = LIBAVUTIL_VERSION_INT,
2359 AVInputFormat ff_sdp_demuxer = {
2361 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2362 .priv_data_size = sizeof(RTSPState),
2363 .read_probe = sdp_probe,
2364 .read_header = sdp_read_header,
2365 .read_packet = ff_rtsp_fetch_packet,
2366 .read_close = sdp_read_close,
2367 .priv_class = &sdp_demuxer_class,
2369 #endif /* CONFIG_SDP_DEMUXER */
2371 #if CONFIG_RTP_DEMUXER
2372 static int rtp_probe(AVProbeData *p)
2374 if (av_strstart(p->filename, "rtp:", NULL))
2375 return AVPROBE_SCORE_MAX;
2379 static int rtp_read_header(AVFormatContext *s)
2381 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2382 char host[500], sdp[500];
2384 URLContext* in = NULL;
2386 AVCodecParameters *par = NULL;
2387 struct sockaddr_storage addr;
2389 socklen_t addrlen = sizeof(addr);
2390 RTSPState *rt = s->priv_data;
2392 if (!ff_network_init())
2393 return AVERROR(EIO);
2395 ret = ffurl_open_whitelist(&in, s->filename, AVIO_FLAG_READ,
2396 &s->interrupt_callback, NULL, s->protocol_whitelist, s->protocol_blacklist, NULL);
2401 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2402 if (ret == AVERROR(EAGAIN))
2407 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2411 if ((recvbuf[0] & 0xc0) != 0x80) {
2412 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2417 if (RTP_PT_IS_RTCP(recvbuf[1]))
2420 payload_type = recvbuf[1] & 0x7f;
2423 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2427 par = avcodec_parameters_alloc();
2429 ret = AVERROR(ENOMEM);
2433 if (ff_rtp_get_codec_info(par, payload_type)) {
2434 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2435 "without an SDP file describing it\n",
2439 if (par->codec_type != AVMEDIA_TYPE_DATA) {
2440 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2441 "properly you need an SDP file "
2445 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2446 NULL, 0, s->filename);
2448 snprintf(sdp, sizeof(sdp),
2449 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2450 addr.ss_family == AF_INET ? 4 : 6, host,
2451 par->codec_type == AVMEDIA_TYPE_DATA ? "application" :
2452 par->codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2453 port, payload_type);
2454 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2455 avcodec_parameters_free(&par);
2457 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2460 /* sdp_read_header initializes this again */
2463 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2465 ret = sdp_read_header(s);
2470 avcodec_parameters_free(&par);
2477 static const AVClass rtp_demuxer_class = {
2478 .class_name = "RTP demuxer",
2479 .item_name = av_default_item_name,
2480 .option = rtp_options,
2481 .version = LIBAVUTIL_VERSION_INT,
2484 AVInputFormat ff_rtp_demuxer = {
2486 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2487 .priv_data_size = sizeof(RTSPState),
2488 .read_probe = rtp_probe,
2489 .read_header = rtp_read_header,
2490 .read_packet = ff_rtsp_fetch_packet,
2491 .read_close = sdp_read_close,
2492 .flags = AVFMT_NOFILE,
2493 .priv_class = &rtp_demuxer_class,
2495 #endif /* CONFIG_RTP_DEMUXER */