3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \
75 { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }
77 #define COMMON_OPTS() \
78 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
79 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
82 const AVOption ff_rtsp_options[] = {
83 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
84 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
85 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
86 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
88 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
89 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
90 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
91 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
92 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
93 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
94 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
95 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
96 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
97 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
99 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
103 static const AVOption sdp_options[] = {
104 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
105 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
106 { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
107 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
112 static const AVOption rtp_options[] = {
113 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
119 static AVDictionary *map_to_opts(RTSPState *rt)
121 AVDictionary *opts = NULL;
124 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
125 av_dict_set(&opts, "buffer_size", buf, 0);
130 static void get_word_until_chars(char *buf, int buf_size,
131 const char *sep, const char **pp)
137 p += strspn(p, SPACE_CHARS);
139 while (!strchr(sep, *p) && *p != '\0') {
140 if ((q - buf) < buf_size - 1)
149 static void get_word_sep(char *buf, int buf_size, const char *sep,
152 if (**pp == '/') (*pp)++;
153 get_word_until_chars(buf, buf_size, sep, pp);
156 static void get_word(char *buf, int buf_size, const char **pp)
158 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
161 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
163 * Used for seeking in the rtp stream.
165 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
169 p += strspn(p, SPACE_CHARS);
170 if (!av_stristart(p, "npt=", &p))
173 *start = AV_NOPTS_VALUE;
174 *end = AV_NOPTS_VALUE;
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 if (av_parse_time(start, buf, 1) < 0)
181 get_word_sep(buf, sizeof(buf), "-", &p);
182 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(const char *buf, struct sockaddr_storage *sock)
189 struct addrinfo hints = { 0 }, *ai = NULL;
190 hints.ai_flags = AI_NUMERICHOST;
191 if (getaddrinfo(buf, NULL, &hints, &ai))
193 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
199 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
200 RTSPStream *rtsp_st, AVStream *st)
202 AVCodecContext *codec = st ? st->codec : NULL;
206 codec->codec_id = handler->codec_id;
207 rtsp_st->dynamic_handler = handler;
209 st->need_parsing = handler->need_parsing;
210 if (handler->priv_data_size) {
211 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
212 if (!rtsp_st->dynamic_protocol_context)
213 rtsp_st->dynamic_handler = NULL;
217 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
220 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
221 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
222 rtsp_st->dynamic_protocol_context);
224 if (rtsp_st->dynamic_protocol_context) {
225 if (rtsp_st->dynamic_handler->close)
226 rtsp_st->dynamic_handler->close(
227 rtsp_st->dynamic_protocol_context);
228 av_free(rtsp_st->dynamic_protocol_context);
230 rtsp_st->dynamic_protocol_context = NULL;
231 rtsp_st->dynamic_handler = NULL;
236 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
237 static int sdp_parse_rtpmap(AVFormatContext *s,
238 AVStream *st, RTSPStream *rtsp_st,
239 int payload_type, const char *p)
241 AVCodecContext *codec = st->codec;
247 /* See if we can handle this kind of payload.
248 * The space should normally not be there but some Real streams or
249 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
250 * have a trailing space. */
251 get_word_sep(buf, sizeof(buf), "/ ", &p);
252 if (payload_type < RTP_PT_PRIVATE) {
253 /* We are in a standard case
254 * (from http://www.iana.org/assignments/rtp-parameters). */
255 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
258 if (codec->codec_id == AV_CODEC_ID_NONE) {
259 RTPDynamicProtocolHandler *handler =
260 ff_rtp_handler_find_by_name(buf, codec->codec_type);
261 init_rtp_handler(handler, rtsp_st, st);
262 /* If no dynamic handler was found, check with the list of standard
263 * allocated types, if such a stream for some reason happens to
264 * use a private payload type. This isn't handled in rtpdec.c, since
265 * the format name from the rtpmap line never is passed into rtpdec. */
266 if (!rtsp_st->dynamic_handler)
267 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
270 c = avcodec_find_decoder(codec->codec_id);
276 get_word_sep(buf, sizeof(buf), "/", &p);
278 switch (codec->codec_type) {
279 case AVMEDIA_TYPE_AUDIO:
280 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
281 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
282 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
284 codec->sample_rate = i;
285 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
286 get_word_sep(buf, sizeof(buf), "/", &p);
291 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
293 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
296 case AVMEDIA_TYPE_VIDEO:
297 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
299 avpriv_set_pts_info(st, 32, 1, i);
304 finalize_rtp_handler_init(s, rtsp_st, st);
308 /* parse the attribute line from the fmtp a line of an sdp response. This
309 * is broken out as a function because it is used in rtp_h264.c, which is
311 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
312 char *value, int value_size)
314 *p += strspn(*p, SPACE_CHARS);
316 get_word_sep(attr, attr_size, "=", p);
319 get_word_sep(value, value_size, ";", p);
327 typedef struct SDPParseState {
329 struct sockaddr_storage default_ip;
331 int skip_media; ///< set if an unknown m= line occurs
332 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
333 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
334 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
335 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
338 char delayed_fmtp[2048];
341 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
342 struct RTSPSource ***dest, int *dest_count)
344 RTSPSource *rtsp_src, *rtsp_src2;
346 for (i = 0; i < count; i++) {
348 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
351 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
352 dynarray_add(dest, dest_count, rtsp_src2);
356 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
357 int payload_type, const char *line)
361 for (i = 0; i < rt->nb_rtsp_streams; i++) {
362 RTSPStream *rtsp_st = rt->rtsp_streams[i];
363 if (rtsp_st->sdp_payload_type == payload_type &&
364 rtsp_st->dynamic_handler &&
365 rtsp_st->dynamic_handler->parse_sdp_a_line) {
366 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
367 rtsp_st->dynamic_protocol_context, line);
372 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
373 int letter, const char *buf)
375 RTSPState *rt = s->priv_data;
376 char buf1[64], st_type[64];
378 enum AVMediaType codec_type;
382 RTSPSource *rtsp_src;
383 struct sockaddr_storage sdp_ip;
386 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
389 if (s1->skip_media && letter != 'm')
393 get_word(buf1, sizeof(buf1), &p);
394 if (strcmp(buf1, "IN") != 0)
396 get_word(buf1, sizeof(buf1), &p);
397 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
399 get_word_sep(buf1, sizeof(buf1), "/", &p);
400 if (get_sockaddr(buf1, &sdp_ip))
405 get_word_sep(buf1, sizeof(buf1), "/", &p);
408 if (s->nb_streams == 0) {
409 s1->default_ip = sdp_ip;
410 s1->default_ttl = ttl;
412 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
413 rtsp_st->sdp_ip = sdp_ip;
414 rtsp_st->sdp_ttl = ttl;
418 av_dict_set(&s->metadata, "title", p, 0);
421 if (s->nb_streams == 0) {
422 av_dict_set(&s->metadata, "comment", p, 0);
431 codec_type = AVMEDIA_TYPE_UNKNOWN;
432 get_word(st_type, sizeof(st_type), &p);
433 if (!strcmp(st_type, "audio")) {
434 codec_type = AVMEDIA_TYPE_AUDIO;
435 } else if (!strcmp(st_type, "video")) {
436 codec_type = AVMEDIA_TYPE_VIDEO;
437 } else if (!strcmp(st_type, "application")) {
438 codec_type = AVMEDIA_TYPE_DATA;
439 } else if (!strcmp(st_type, "text")) {
440 codec_type = AVMEDIA_TYPE_SUBTITLE;
442 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
446 rtsp_st = av_mallocz(sizeof(RTSPStream));
449 rtsp_st->stream_index = -1;
450 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
452 rtsp_st->sdp_ip = s1->default_ip;
453 rtsp_st->sdp_ttl = s1->default_ttl;
455 copy_default_source_addrs(s1->default_include_source_addrs,
456 s1->nb_default_include_source_addrs,
457 &rtsp_st->include_source_addrs,
458 &rtsp_st->nb_include_source_addrs);
459 copy_default_source_addrs(s1->default_exclude_source_addrs,
460 s1->nb_default_exclude_source_addrs,
461 &rtsp_st->exclude_source_addrs,
462 &rtsp_st->nb_exclude_source_addrs);
464 get_word(buf1, sizeof(buf1), &p); /* port */
465 rtsp_st->sdp_port = atoi(buf1);
467 get_word(buf1, sizeof(buf1), &p); /* protocol */
468 if (!strcmp(buf1, "udp"))
469 rt->transport = RTSP_TRANSPORT_RAW;
470 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
471 rtsp_st->feedback = 1;
473 /* XXX: handle list of formats */
474 get_word(buf1, sizeof(buf1), &p); /* format list */
475 rtsp_st->sdp_payload_type = atoi(buf1);
477 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
478 /* no corresponding stream */
479 if (rt->transport == RTSP_TRANSPORT_RAW) {
480 if (CONFIG_RTPDEC && !rt->ts)
481 rt->ts = avpriv_mpegts_parse_open(s);
483 RTPDynamicProtocolHandler *handler;
484 handler = ff_rtp_handler_find_by_id(
485 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
486 init_rtp_handler(handler, rtsp_st, NULL);
487 finalize_rtp_handler_init(s, rtsp_st, NULL);
489 } else if (rt->server_type == RTSP_SERVER_WMS &&
490 codec_type == AVMEDIA_TYPE_DATA) {
491 /* RTX stream, a stream that carries all the other actual
492 * audio/video streams. Don't expose this to the callers. */
494 st = avformat_new_stream(s, NULL);
497 st->id = rt->nb_rtsp_streams - 1;
498 rtsp_st->stream_index = st->index;
499 st->codec->codec_type = codec_type;
500 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
501 RTPDynamicProtocolHandler *handler;
502 /* if standard payload type, we can find the codec right now */
503 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
504 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
505 st->codec->sample_rate > 0)
506 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
507 /* Even static payload types may need a custom depacketizer */
508 handler = ff_rtp_handler_find_by_id(
509 rtsp_st->sdp_payload_type, st->codec->codec_type);
510 init_rtp_handler(handler, rtsp_st, st);
511 finalize_rtp_handler_init(s, rtsp_st, st);
513 if (rt->default_lang[0])
514 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
516 /* put a default control url */
517 av_strlcpy(rtsp_st->control_url, rt->control_uri,
518 sizeof(rtsp_st->control_url));
521 if (av_strstart(p, "control:", &p)) {
522 if (s->nb_streams == 0) {
523 if (!strncmp(p, "rtsp://", 7))
524 av_strlcpy(rt->control_uri, p,
525 sizeof(rt->control_uri));
528 /* get the control url */
529 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
531 /* XXX: may need to add full url resolution */
532 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
534 if (proto[0] == '\0') {
535 /* relative control URL */
536 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
537 av_strlcat(rtsp_st->control_url, "/",
538 sizeof(rtsp_st->control_url));
539 av_strlcat(rtsp_st->control_url, p,
540 sizeof(rtsp_st->control_url));
542 av_strlcpy(rtsp_st->control_url, p,
543 sizeof(rtsp_st->control_url));
545 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
546 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
547 get_word(buf1, sizeof(buf1), &p);
548 payload_type = atoi(buf1);
549 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
550 if (rtsp_st->stream_index >= 0) {
551 st = s->streams[rtsp_st->stream_index];
552 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
556 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
558 } else if (av_strstart(p, "fmtp:", &p) ||
559 av_strstart(p, "framesize:", &p)) {
560 // let dynamic protocol handlers have a stab at the line.
561 get_word(buf1, sizeof(buf1), &p);
562 payload_type = atoi(buf1);
563 if (s1->seen_rtpmap) {
564 parse_fmtp(s, rt, payload_type, buf);
567 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
569 } else if (av_strstart(p, "range:", &p)) {
572 // this is so that seeking on a streamed file can work.
573 rtsp_parse_range_npt(p, &start, &end);
574 s->start_time = start;
575 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
576 s->duration = (end == AV_NOPTS_VALUE) ?
577 AV_NOPTS_VALUE : end - start;
578 } else if (av_strstart(p, "lang:", &p)) {
579 if (s->nb_streams > 0) {
580 get_word(buf1, sizeof(buf1), &p);
581 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
582 if (rtsp_st->stream_index >= 0) {
583 st = s->streams[rtsp_st->stream_index];
584 av_dict_set(&st->metadata, "language", buf1, 0);
587 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
588 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
590 rt->transport = RTSP_TRANSPORT_RDT;
591 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
593 st = s->streams[s->nb_streams - 1];
594 st->codec->sample_rate = atoi(p);
595 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
597 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
598 get_word(buf1, sizeof(buf1), &p); // ignore tag
599 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
600 p += strspn(p, SPACE_CHARS);
601 if (av_strstart(p, "inline:", &p))
602 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
603 } else if (av_strstart(p, "source-filter:", &p)) {
605 get_word(buf1, sizeof(buf1), &p);
606 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
608 exclude = !strcmp(buf1, "excl");
610 get_word(buf1, sizeof(buf1), &p);
611 if (strcmp(buf1, "IN") != 0)
613 get_word(buf1, sizeof(buf1), &p);
614 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
616 // not checking that the destination address actually matches or is wildcard
617 get_word(buf1, sizeof(buf1), &p);
620 rtsp_src = av_mallocz(sizeof(*rtsp_src));
623 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
625 if (s->nb_streams == 0) {
626 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
628 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
629 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
632 if (s->nb_streams == 0) {
633 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
635 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
636 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
641 if (rt->server_type == RTSP_SERVER_WMS)
642 ff_wms_parse_sdp_a_line(s, p);
643 if (s->nb_streams > 0) {
644 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
646 if (rt->server_type == RTSP_SERVER_REAL)
647 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
649 if (rtsp_st->dynamic_handler &&
650 rtsp_st->dynamic_handler->parse_sdp_a_line)
651 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
652 rtsp_st->stream_index,
653 rtsp_st->dynamic_protocol_context, buf);
660 int ff_sdp_parse(AVFormatContext *s, const char *content)
662 RTSPState *rt = s->priv_data;
665 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
666 * contain long SDP lines containing complete ASF Headers (several
667 * kB) or arrays of MDPR (RM stream descriptor) headers plus
668 * "rulebooks" describing their properties. Therefore, the SDP line
671 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
672 * in rtpdec_xiph.c. */
674 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
678 p += strspn(p, SPACE_CHARS);
686 /* get the content */
688 while (*p != '\n' && *p != '\r' && *p != '\0') {
689 if ((q - buf) < sizeof(buf) - 1)
694 sdp_parse_line(s, s1, letter, buf);
696 while (*p != '\n' && *p != '\0')
702 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
703 av_freep(&s1->default_include_source_addrs[i]);
704 av_freep(&s1->default_include_source_addrs);
705 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
706 av_freep(&s1->default_exclude_source_addrs[i]);
707 av_freep(&s1->default_exclude_source_addrs);
709 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
710 if (!rt->p) return AVERROR(ENOMEM);
713 #endif /* CONFIG_RTPDEC */
715 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
717 RTSPState *rt = s->priv_data;
720 for (i = 0; i < rt->nb_rtsp_streams; i++) {
721 RTSPStream *rtsp_st = rt->rtsp_streams[i];
724 if (rtsp_st->transport_priv) {
726 AVFormatContext *rtpctx = rtsp_st->transport_priv;
727 av_write_trailer(rtpctx);
728 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
729 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
730 ff_rtsp_tcp_write_packet(s, rtsp_st);
731 ffio_free_dyn_buf(&rtpctx->pb);
733 avio_closep(&rtpctx->pb);
735 avformat_free_context(rtpctx);
736 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
737 ff_rdt_parse_close(rtsp_st->transport_priv);
738 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
739 ff_rtp_parse_close(rtsp_st->transport_priv);
741 rtsp_st->transport_priv = NULL;
742 if (rtsp_st->rtp_handle)
743 ffurl_close(rtsp_st->rtp_handle);
744 rtsp_st->rtp_handle = NULL;
748 /* close and free RTSP streams */
749 void ff_rtsp_close_streams(AVFormatContext *s)
751 RTSPState *rt = s->priv_data;
755 ff_rtsp_undo_setup(s, 0);
756 for (i = 0; i < rt->nb_rtsp_streams; i++) {
757 rtsp_st = rt->rtsp_streams[i];
759 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
760 if (rtsp_st->dynamic_handler->close)
761 rtsp_st->dynamic_handler->close(
762 rtsp_st->dynamic_protocol_context);
763 av_free(rtsp_st->dynamic_protocol_context);
765 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
766 av_freep(&rtsp_st->include_source_addrs[j]);
767 av_freep(&rtsp_st->include_source_addrs);
768 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
769 av_freep(&rtsp_st->exclude_source_addrs[j]);
770 av_freep(&rtsp_st->exclude_source_addrs);
775 av_freep(&rt->rtsp_streams);
777 avformat_close_input(&rt->asf_ctx);
779 if (CONFIG_RTPDEC && rt->ts)
780 avpriv_mpegts_parse_close(rt->ts);
782 av_freep(&rt->recvbuf);
785 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
787 RTSPState *rt = s->priv_data;
789 int reordering_queue_size = rt->reordering_queue_size;
790 if (reordering_queue_size < 0) {
791 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
792 reordering_queue_size = 0;
794 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
797 /* open the RTP context */
798 if (rtsp_st->stream_index >= 0)
799 st = s->streams[rtsp_st->stream_index];
801 s->ctx_flags |= AVFMTCTX_NOHEADER;
803 if (CONFIG_RTSP_MUXER && s->oformat && st) {
804 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
805 s, st, rtsp_st->rtp_handle,
806 RTSP_TCP_MAX_PACKET_SIZE,
807 rtsp_st->stream_index);
808 /* Ownership of rtp_handle is passed to the rtp mux context */
809 rtsp_st->rtp_handle = NULL;
812 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
813 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
814 return 0; // Don't need to open any parser here
815 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT && st)
816 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
817 rtsp_st->dynamic_protocol_context,
818 rtsp_st->dynamic_handler);
819 else if (CONFIG_RTPDEC)
820 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
821 rtsp_st->sdp_payload_type,
822 reordering_queue_size);
824 if (!rtsp_st->transport_priv) {
825 return AVERROR(ENOMEM);
826 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
827 if (rtsp_st->dynamic_handler) {
828 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
829 rtsp_st->dynamic_protocol_context,
830 rtsp_st->dynamic_handler);
832 if (rtsp_st->crypto_suite[0])
833 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
834 rtsp_st->crypto_suite,
835 rtsp_st->crypto_params);
841 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
842 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
849 q += strspn(q, SPACE_CHARS);
850 v = strtol(q, &p, 10);
854 v = strtol(p, &p, 10);
863 /* XXX: only one transport specification is parsed */
864 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
866 char transport_protocol[16];
868 char lower_transport[16];
870 RTSPTransportField *th;
873 reply->nb_transports = 0;
876 p += strspn(p, SPACE_CHARS);
880 th = &reply->transports[reply->nb_transports];
882 get_word_sep(transport_protocol, sizeof(transport_protocol),
884 if (!av_strcasecmp (transport_protocol, "rtp")) {
885 get_word_sep(profile, sizeof(profile), "/;,", &p);
886 lower_transport[0] = '\0';
887 /* rtp/avp/<protocol> */
889 get_word_sep(lower_transport, sizeof(lower_transport),
892 th->transport = RTSP_TRANSPORT_RTP;
893 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
894 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
895 /* x-pn-tng/<protocol> */
896 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
898 th->transport = RTSP_TRANSPORT_RDT;
899 } else if (!av_strcasecmp(transport_protocol, "raw")) {
900 get_word_sep(profile, sizeof(profile), "/;,", &p);
901 lower_transport[0] = '\0';
902 /* raw/raw/<protocol> */
904 get_word_sep(lower_transport, sizeof(lower_transport),
907 th->transport = RTSP_TRANSPORT_RAW;
909 if (!av_strcasecmp(lower_transport, "TCP"))
910 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
912 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
916 /* get each parameter */
917 while (*p != '\0' && *p != ',') {
918 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
919 if (!strcmp(parameter, "port")) {
922 rtsp_parse_range(&th->port_min, &th->port_max, &p);
924 } else if (!strcmp(parameter, "client_port")) {
927 rtsp_parse_range(&th->client_port_min,
928 &th->client_port_max, &p);
930 } else if (!strcmp(parameter, "server_port")) {
933 rtsp_parse_range(&th->server_port_min,
934 &th->server_port_max, &p);
936 } else if (!strcmp(parameter, "interleaved")) {
939 rtsp_parse_range(&th->interleaved_min,
940 &th->interleaved_max, &p);
942 } else if (!strcmp(parameter, "multicast")) {
943 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
944 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
945 } else if (!strcmp(parameter, "ttl")) {
949 th->ttl = strtol(p, &end, 10);
952 } else if (!strcmp(parameter, "destination")) {
955 get_word_sep(buf, sizeof(buf), ";,", &p);
956 get_sockaddr(buf, &th->destination);
958 } else if (!strcmp(parameter, "source")) {
961 get_word_sep(buf, sizeof(buf), ";,", &p);
962 av_strlcpy(th->source, buf, sizeof(th->source));
964 } else if (!strcmp(parameter, "mode")) {
967 get_word_sep(buf, sizeof(buf), ";, ", &p);
968 if (!strcmp(buf, "record") ||
969 !strcmp(buf, "receive"))
974 while (*p != ';' && *p != '\0' && *p != ',')
982 reply->nb_transports++;
983 if (reply->nb_transports >= RTSP_MAX_TRANSPORTS)
988 static void handle_rtp_info(RTSPState *rt, const char *url,
989 uint32_t seq, uint32_t rtptime)
992 if (!rtptime || !url[0])
994 if (rt->transport != RTSP_TRANSPORT_RTP)
996 for (i = 0; i < rt->nb_rtsp_streams; i++) {
997 RTSPStream *rtsp_st = rt->rtsp_streams[i];
998 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1001 if (!strcmp(rtsp_st->control_url, url)) {
1002 rtpctx->base_timestamp = rtptime;
1008 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1011 char key[20], value[1024], url[1024] = "";
1012 uint32_t seq = 0, rtptime = 0;
1015 p += strspn(p, SPACE_CHARS);
1018 get_word_sep(key, sizeof(key), "=", &p);
1022 get_word_sep(value, sizeof(value), ";, ", &p);
1024 if (!strcmp(key, "url"))
1025 av_strlcpy(url, value, sizeof(url));
1026 else if (!strcmp(key, "seq"))
1027 seq = strtoul(value, NULL, 10);
1028 else if (!strcmp(key, "rtptime"))
1029 rtptime = strtoul(value, NULL, 10);
1031 handle_rtp_info(rt, url, seq, rtptime);
1040 handle_rtp_info(rt, url, seq, rtptime);
1043 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1044 RTSPState *rt, const char *method)
1048 /* NOTE: we do case independent match for broken servers */
1050 if (av_stristart(p, "Session:", &p)) {
1052 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1053 if (av_stristart(p, ";timeout=", &p) &&
1054 (t = strtol(p, NULL, 10)) > 0) {
1057 } else if (av_stristart(p, "Content-Length:", &p)) {
1058 reply->content_length = strtol(p, NULL, 10);
1059 } else if (av_stristart(p, "Transport:", &p)) {
1060 rtsp_parse_transport(reply, p);
1061 } else if (av_stristart(p, "CSeq:", &p)) {
1062 reply->seq = strtol(p, NULL, 10);
1063 } else if (av_stristart(p, "Range:", &p)) {
1064 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1065 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1066 p += strspn(p, SPACE_CHARS);
1067 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1068 } else if (av_stristart(p, "Server:", &p)) {
1069 p += strspn(p, SPACE_CHARS);
1070 av_strlcpy(reply->server, p, sizeof(reply->server));
1071 } else if (av_stristart(p, "Notice:", &p) ||
1072 av_stristart(p, "X-Notice:", &p)) {
1073 reply->notice = strtol(p, NULL, 10);
1074 } else if (av_stristart(p, "Location:", &p)) {
1075 p += strspn(p, SPACE_CHARS);
1076 av_strlcpy(reply->location, p , sizeof(reply->location));
1077 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1078 p += strspn(p, SPACE_CHARS);
1079 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1080 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1081 p += strspn(p, SPACE_CHARS);
1082 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1083 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1084 p += strspn(p, SPACE_CHARS);
1085 if (method && !strcmp(method, "DESCRIBE"))
1086 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1087 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1088 p += strspn(p, SPACE_CHARS);
1089 if (method && !strcmp(method, "PLAY"))
1090 rtsp_parse_rtp_info(rt, p);
1091 } else if (av_stristart(p, "Public:", &p) && rt) {
1092 if (strstr(p, "GET_PARAMETER") &&
1093 method && !strcmp(method, "OPTIONS"))
1094 rt->get_parameter_supported = 1;
1095 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1096 p += strspn(p, SPACE_CHARS);
1097 rt->accept_dynamic_rate = atoi(p);
1098 } else if (av_stristart(p, "Content-Type:", &p)) {
1099 p += strspn(p, SPACE_CHARS);
1100 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1104 /* skip a RTP/TCP interleaved packet */
1105 void ff_rtsp_skip_packet(AVFormatContext *s)
1107 RTSPState *rt = s->priv_data;
1111 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1114 len = AV_RB16(buf + 1);
1116 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1121 if (len1 > sizeof(buf))
1123 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1130 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1131 unsigned char **content_ptr,
1132 int return_on_interleaved_data, const char *method)
1134 RTSPState *rt = s->priv_data;
1135 char buf[4096], buf1[1024], *q;
1138 int ret, content_length, line_count = 0, request = 0;
1140 unsigned char *content = NULL;
1146 memset(reply, 0, sizeof(*reply));
1148 /* parse reply (XXX: use buffers) */
1149 rt->last_reply[0] = '\0';
1153 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1154 av_log(s, AV_LOG_TRACE, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1159 if (ch == '$' && first_line && q == buf) {
1160 if (return_on_interleaved_data) {
1163 ff_rtsp_skip_packet(s);
1164 } else if (ch != '\r') {
1165 if ((q - buf) < sizeof(buf) - 1)
1172 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1174 /* test if last line */
1178 if (line_count == 0) {
1179 /* get reply code */
1180 get_word(buf1, sizeof(buf1), &p);
1181 if (!strncmp(buf1, "RTSP/", 5)) {
1182 get_word(buf1, sizeof(buf1), &p);
1183 reply->status_code = atoi(buf1);
1184 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1186 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1187 get_word(buf1, sizeof(buf1), &p); // object
1191 ff_rtsp_parse_line(reply, p, rt, method);
1192 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1193 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1198 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1199 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1201 content_length = reply->content_length;
1202 if (content_length > 0) {
1203 /* leave some room for a trailing '\0' (useful for simple parsing) */
1204 content = av_malloc(content_length + 1);
1206 return AVERROR(ENOMEM);
1207 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1208 content[content_length] = '\0';
1211 *content_ptr = content;
1217 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1218 const char* ptr = buf;
1220 if (!strcmp(reply->reason, "OPTIONS")) {
1221 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1223 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1224 if (reply->session_id[0])
1225 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1228 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1230 av_strlcat(buf, "\r\n", sizeof(buf));
1232 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1233 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1236 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1238 rt->last_cmd_time = av_gettime_relative();
1239 /* Even if the request from the server had data, it is not the data
1240 * that the caller wants or expects. The memory could also be leaked
1241 * if the actual following reply has content data. */
1243 av_freep(content_ptr);
1244 /* If method is set, this is called from ff_rtsp_send_cmd,
1245 * where a reply to exactly this request is awaited. For
1246 * callers from within packet receiving, we just want to
1247 * return to the caller and go back to receiving packets. */
1253 if (rt->seq != reply->seq) {
1254 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1255 rt->seq, reply->seq);
1259 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1260 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1261 reply->notice == 2306 /* Continuous Feed Terminated */) {
1262 rt->state = RTSP_STATE_IDLE;
1263 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1264 return AVERROR(EIO); /* data or server error */
1265 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1266 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1267 return AVERROR(EPERM);
1273 * Send a command to the RTSP server without waiting for the reply.
1275 * @param s RTSP (de)muxer context
1276 * @param method the method for the request
1277 * @param url the target url for the request
1278 * @param headers extra header lines to include in the request
1279 * @param send_content if non-null, the data to send as request body content
1280 * @param send_content_length the length of the send_content data, or 0 if
1281 * send_content is null
1283 * @return zero if success, nonzero otherwise
1285 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1286 const char *method, const char *url,
1287 const char *headers,
1288 const unsigned char *send_content,
1289 int send_content_length)
1291 RTSPState *rt = s->priv_data;
1292 char buf[4096], *out_buf;
1293 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1295 /* Add in RTSP headers */
1298 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1300 av_strlcat(buf, headers, sizeof(buf));
1301 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1302 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1303 if (rt->session_id[0] != '\0' && (!headers ||
1304 !strstr(headers, "\nIf-Match:"))) {
1305 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1308 char *str = ff_http_auth_create_response(&rt->auth_state,
1309 rt->auth, url, method);
1311 av_strlcat(buf, str, sizeof(buf));
1314 if (send_content_length > 0 && send_content)
1315 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1316 av_strlcat(buf, "\r\n", sizeof(buf));
1318 /* base64 encode rtsp if tunneling */
1319 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1320 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1321 out_buf = base64buf;
1324 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1326 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1327 if (send_content_length > 0 && send_content) {
1328 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1329 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1330 "with content data not supported\n");
1331 return AVERROR_PATCHWELCOME;
1333 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1335 rt->last_cmd_time = av_gettime_relative();
1340 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1341 const char *url, const char *headers)
1343 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1346 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1347 const char *headers, RTSPMessageHeader *reply,
1348 unsigned char **content_ptr)
1350 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1351 content_ptr, NULL, 0);
1354 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1355 const char *method, const char *url,
1357 RTSPMessageHeader *reply,
1358 unsigned char **content_ptr,
1359 const unsigned char *send_content,
1360 int send_content_length)
1362 RTSPState *rt = s->priv_data;
1363 HTTPAuthType cur_auth_type;
1364 int ret, attempts = 0;
1367 cur_auth_type = rt->auth_state.auth_type;
1368 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1370 send_content_length)))
1373 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1377 if (reply->status_code == 401 &&
1378 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1379 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1382 if (reply->status_code > 400){
1383 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1387 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1393 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1394 int lower_transport, const char *real_challenge)
1396 RTSPState *rt = s->priv_data;
1397 int rtx = 0, j, i, err, interleave = 0, port_off;
1398 RTSPStream *rtsp_st;
1399 RTSPMessageHeader reply1, *reply = &reply1;
1401 const char *trans_pref;
1403 if (rt->transport == RTSP_TRANSPORT_RDT)
1404 trans_pref = "x-pn-tng";
1405 else if (rt->transport == RTSP_TRANSPORT_RAW)
1406 trans_pref = "RAW/RAW";
1408 trans_pref = "RTP/AVP";
1410 /* default timeout: 1 minute */
1413 /* Choose a random starting offset within the first half of the
1414 * port range, to allow for a number of ports to try even if the offset
1415 * happens to be at the end of the random range. */
1416 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1417 /* even random offset */
1418 port_off -= port_off & 0x01;
1420 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1421 char transport[2048];
1424 * WMS serves all UDP data over a single connection, the RTX, which
1425 * isn't necessarily the first in the SDP but has to be the first
1426 * to be set up, else the second/third SETUP will fail with a 461.
1428 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1429 rt->server_type == RTSP_SERVER_WMS) {
1432 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1433 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1435 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1439 if (rtx == rt->nb_rtsp_streams)
1440 return -1; /* no RTX found */
1441 rtsp_st = rt->rtsp_streams[rtx];
1443 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1445 rtsp_st = rt->rtsp_streams[i];
1448 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1451 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1452 port = reply->transports[0].client_port_min;
1456 /* first try in specified port range */
1457 while (j <= rt->rtp_port_max) {
1458 AVDictionary *opts = map_to_opts(rt);
1460 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1461 "?localport=%d", j);
1462 /* we will use two ports per rtp stream (rtp and rtcp) */
1464 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1465 &s->interrupt_callback, &opts);
1467 av_dict_free(&opts);
1472 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1477 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1479 snprintf(transport, sizeof(transport) - 1,
1480 "%s/UDP;", trans_pref);
1481 if (rt->server_type != RTSP_SERVER_REAL)
1482 av_strlcat(transport, "unicast;", sizeof(transport));
1483 av_strlcatf(transport, sizeof(transport),
1484 "client_port=%d", port);
1485 if (rt->transport == RTSP_TRANSPORT_RTP &&
1486 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1487 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1491 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1492 /* For WMS streams, the application streams are only used for
1493 * UDP. When trying to set it up for TCP streams, the server
1494 * will return an error. Therefore, we skip those streams. */
1495 if (rt->server_type == RTSP_SERVER_WMS &&
1496 (rtsp_st->stream_index < 0 ||
1497 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1500 snprintf(transport, sizeof(transport) - 1,
1501 "%s/TCP;", trans_pref);
1502 if (rt->transport != RTSP_TRANSPORT_RDT)
1503 av_strlcat(transport, "unicast;", sizeof(transport));
1504 av_strlcatf(transport, sizeof(transport),
1505 "interleaved=%d-%d",
1506 interleave, interleave + 1);
1510 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1511 snprintf(transport, sizeof(transport) - 1,
1512 "%s/UDP;multicast", trans_pref);
1515 av_strlcat(transport, ";mode=record", sizeof(transport));
1516 } else if (rt->server_type == RTSP_SERVER_REAL ||
1517 rt->server_type == RTSP_SERVER_WMS)
1518 av_strlcat(transport, ";mode=play", sizeof(transport));
1519 snprintf(cmd, sizeof(cmd),
1520 "Transport: %s\r\n",
1522 if (rt->accept_dynamic_rate)
1523 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1524 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1525 char real_res[41], real_csum[9];
1526 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1528 av_strlcatf(cmd, sizeof(cmd),
1530 "RealChallenge2: %s, sd=%s\r\n",
1531 rt->session_id, real_res, real_csum);
1533 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1534 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1537 } else if (reply->status_code != RTSP_STATUS_OK ||
1538 reply->nb_transports != 1) {
1539 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1543 /* XXX: same protocol for all streams is required */
1545 if (reply->transports[0].lower_transport != rt->lower_transport ||
1546 reply->transports[0].transport != rt->transport) {
1547 err = AVERROR_INVALIDDATA;
1551 rt->lower_transport = reply->transports[0].lower_transport;
1552 rt->transport = reply->transports[0].transport;
1555 /* Fail if the server responded with another lower transport mode
1556 * than what we requested. */
1557 if (reply->transports[0].lower_transport != lower_transport) {
1558 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1559 err = AVERROR_INVALIDDATA;
1563 switch(reply->transports[0].lower_transport) {
1564 case RTSP_LOWER_TRANSPORT_TCP:
1565 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1566 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1569 case RTSP_LOWER_TRANSPORT_UDP: {
1570 char url[1024], options[30] = "";
1571 const char *peer = host;
1573 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1574 av_strlcpy(options, "?connect=1", sizeof(options));
1575 /* Use source address if specified */
1576 if (reply->transports[0].source[0])
1577 peer = reply->transports[0].source;
1578 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1579 reply->transports[0].server_port_min, "%s", options);
1580 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1581 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1582 err = AVERROR_INVALIDDATA;
1587 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1588 char url[1024], namebuf[50], optbuf[20] = "";
1589 struct sockaddr_storage addr;
1592 if (reply->transports[0].destination.ss_family) {
1593 addr = reply->transports[0].destination;
1594 port = reply->transports[0].port_min;
1595 ttl = reply->transports[0].ttl;
1597 addr = rtsp_st->sdp_ip;
1598 port = rtsp_st->sdp_port;
1599 ttl = rtsp_st->sdp_ttl;
1602 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1603 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1604 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1605 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1606 port, "%s", optbuf);
1607 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1608 &s->interrupt_callback, NULL) < 0) {
1609 err = AVERROR_INVALIDDATA;
1616 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1620 if (rt->nb_rtsp_streams && reply->timeout > 0)
1621 rt->timeout = reply->timeout;
1623 if (rt->server_type == RTSP_SERVER_REAL)
1624 rt->need_subscription = 1;
1629 ff_rtsp_undo_setup(s, 0);
1633 void ff_rtsp_close_connections(AVFormatContext *s)
1635 RTSPState *rt = s->priv_data;
1636 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1637 ffurl_close(rt->rtsp_hd);
1638 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1641 int ff_rtsp_connect(AVFormatContext *s)
1643 RTSPState *rt = s->priv_data;
1644 char proto[128], host[1024], path[1024];
1645 char tcpname[1024], cmd[2048], auth[128];
1646 const char *lower_rtsp_proto = "tcp";
1647 int port, err, tcp_fd;
1648 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1649 int lower_transport_mask = 0;
1650 int default_port = RTSP_DEFAULT_PORT;
1651 char real_challenge[64] = "";
1652 struct sockaddr_storage peer;
1653 socklen_t peer_len = sizeof(peer);
1655 if (rt->rtp_port_max < rt->rtp_port_min) {
1656 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1657 "than min port %d\n", rt->rtp_port_max,
1659 return AVERROR(EINVAL);
1662 if (!ff_network_init())
1663 return AVERROR(EIO);
1665 if (s->max_delay < 0) /* Not set by the caller */
1666 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1668 rt->control_transport = RTSP_MODE_PLAIN;
1669 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1670 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1671 rt->control_transport = RTSP_MODE_TUNNEL;
1673 /* Only pass through valid flags from here */
1674 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1677 /* extract hostname and port */
1678 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1679 host, sizeof(host), &port, path, sizeof(path), s->filename);
1681 if (!strcmp(proto, "rtsps")) {
1682 lower_rtsp_proto = "tls";
1683 default_port = RTSPS_DEFAULT_PORT;
1684 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1688 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1691 port = default_port;
1693 lower_transport_mask = rt->lower_transport_mask;
1695 if (!lower_transport_mask)
1696 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1699 /* Only UDP or TCP - UDP multicast isn't supported. */
1700 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1701 (1 << RTSP_LOWER_TRANSPORT_TCP);
1702 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1703 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1704 "only UDP and TCP are supported for output.\n");
1705 err = AVERROR(EINVAL);
1710 /* Construct the URI used in request; this is similar to s->filename,
1711 * but with authentication credentials removed and RTSP specific options
1713 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1714 host, port, "%s", path);
1716 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1717 /* set up initial handshake for tunneling */
1718 char httpname[1024];
1719 char sessioncookie[17];
1722 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1723 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1724 av_get_random_seed(), av_get_random_seed());
1727 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1728 &s->interrupt_callback) < 0) {
1733 /* generate GET headers */
1734 snprintf(headers, sizeof(headers),
1735 "x-sessioncookie: %s\r\n"
1736 "Accept: application/x-rtsp-tunnelled\r\n"
1737 "Pragma: no-cache\r\n"
1738 "Cache-Control: no-cache\r\n",
1740 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1742 /* complete the connection */
1743 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1749 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1750 &s->interrupt_callback) < 0 ) {
1755 /* generate POST headers */
1756 snprintf(headers, sizeof(headers),
1757 "x-sessioncookie: %s\r\n"
1758 "Content-Type: application/x-rtsp-tunnelled\r\n"
1759 "Pragma: no-cache\r\n"
1760 "Cache-Control: no-cache\r\n"
1761 "Content-Length: 32767\r\n"
1762 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1764 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1765 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1767 /* Initialize the authentication state for the POST session. The HTTP
1768 * protocol implementation doesn't properly handle multi-pass
1769 * authentication for POST requests, since it would require one of
1771 * - implementing Expect: 100-continue, which many HTTP servers
1772 * don't support anyway, even less the RTSP servers that do HTTP
1774 * - sending the whole POST data until getting a 401 reply specifying
1775 * what authentication method to use, then resending all that data
1776 * - waiting for potential 401 replies directly after sending the
1777 * POST header (waiting for some unspecified time)
1778 * Therefore, we copy the full auth state, which works for both basic
1779 * and digest. (For digest, we would have to synchronize the nonce
1780 * count variable between the two sessions, if we'd do more requests
1781 * with the original session, though.)
1783 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1785 /* complete the connection */
1786 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1792 /* open the tcp connection */
1793 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1795 "?timeout=%d", rt->stimeout);
1796 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1797 &s->interrupt_callback, NULL)) < 0) {
1801 rt->rtsp_hd_out = rt->rtsp_hd;
1805 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1810 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1811 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1812 NULL, 0, NI_NUMERICHOST);
1815 /* request options supported by the server; this also detects server
1817 for (rt->server_type = RTSP_SERVER_RTP;;) {
1819 if (rt->server_type == RTSP_SERVER_REAL)
1822 * The following entries are required for proper
1823 * streaming from a Realmedia server. They are
1824 * interdependent in some way although we currently
1825 * don't quite understand how. Values were copied
1826 * from mplayer SVN r23589.
1827 * ClientChallenge is a 16-byte ID in hex
1828 * CompanyID is a 16-byte ID in base64
1830 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1831 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1832 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1833 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1835 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1836 if (reply->status_code != RTSP_STATUS_OK) {
1837 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1841 /* detect server type if not standard-compliant RTP */
1842 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1843 rt->server_type = RTSP_SERVER_REAL;
1845 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1846 rt->server_type = RTSP_SERVER_WMS;
1847 } else if (rt->server_type == RTSP_SERVER_REAL)
1848 strcpy(real_challenge, reply->real_challenge);
1852 if (CONFIG_RTSP_DEMUXER && s->iformat)
1853 err = ff_rtsp_setup_input_streams(s, reply);
1854 else if (CONFIG_RTSP_MUXER)
1855 err = ff_rtsp_setup_output_streams(s, host);
1862 int lower_transport = ff_log2_tab[lower_transport_mask &
1863 ~(lower_transport_mask - 1)];
1865 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1866 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1867 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1869 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1870 rt->server_type == RTSP_SERVER_REAL ?
1871 real_challenge : NULL);
1874 lower_transport_mask &= ~(1 << lower_transport);
1875 if (lower_transport_mask == 0 && err == 1) {
1876 err = AVERROR(EPROTONOSUPPORT);
1881 rt->lower_transport_mask = lower_transport_mask;
1882 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1883 rt->state = RTSP_STATE_IDLE;
1884 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1887 ff_rtsp_close_streams(s);
1888 ff_rtsp_close_connections(s);
1889 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1890 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1891 rt->session_id[0] = '\0';
1892 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1900 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1903 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1904 uint8_t *buf, int buf_size, int64_t wait_end)
1906 RTSPState *rt = s->priv_data;
1907 RTSPStream *rtsp_st;
1908 int n, i, ret, tcp_fd, timeout_cnt = 0;
1910 struct pollfd *p = rt->p;
1911 int *fds = NULL, fdsnum, fdsidx;
1914 if (ff_check_interrupt(&s->interrupt_callback))
1915 return AVERROR_EXIT;
1916 if (wait_end && wait_end - av_gettime_relative() < 0)
1917 return AVERROR(EAGAIN);
1920 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1921 p[max_p].fd = tcp_fd;
1922 p[max_p++].events = POLLIN;
1926 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1927 rtsp_st = rt->rtsp_streams[i];
1928 if (rtsp_st->rtp_handle) {
1929 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1931 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1935 av_log(s, AV_LOG_ERROR,
1936 "Number of fds %d not supported\n", fdsnum);
1937 return AVERROR_INVALIDDATA;
1939 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1940 p[max_p].fd = fds[fdsidx];
1941 p[max_p++].events = POLLIN;
1946 n = poll(p, max_p, POLL_TIMEOUT_MS);
1948 int j = 1 - (tcp_fd == -1);
1950 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1951 rtsp_st = rt->rtsp_streams[i];
1952 if (rtsp_st->rtp_handle) {
1953 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1954 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1956 *prtsp_st = rtsp_st;
1963 #if CONFIG_RTSP_DEMUXER
1964 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1965 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1966 if (rt->state == RTSP_STATE_STREAMING) {
1967 if (!ff_rtsp_parse_streaming_commands(s))
1970 av_log(s, AV_LOG_WARNING,
1971 "Unable to answer to TEARDOWN\n");
1975 RTSPMessageHeader reply;
1976 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1979 /* XXX: parse message */
1980 if (rt->state != RTSP_STATE_STREAMING)
1985 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1986 return AVERROR(ETIMEDOUT);
1987 } else if (n < 0 && errno != EINTR)
1988 return AVERROR(errno);
1992 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1993 const uint8_t *buf, int len)
1995 RTSPState *rt = s->priv_data;
1999 if (rt->nb_rtsp_streams == 1) {
2000 *rtsp_st = rt->rtsp_streams[0];
2003 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2004 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2006 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2007 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2010 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2011 *rtsp_st = rt->rtsp_streams[i];
2018 av_log(s, AV_LOG_WARNING,
2019 "Unable to pick stream for packet - SSRC not known for "
2021 return AVERROR(EAGAIN);
2024 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2025 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2026 *rtsp_st = rt->rtsp_streams[i];
2032 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2033 return AVERROR(EAGAIN);
2036 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2038 RTSPState *rt = s->priv_data;
2040 RTSPStream *rtsp_st, *first_queue_st = NULL;
2041 int64_t wait_end = 0;
2043 if (rt->nb_byes == rt->nb_rtsp_streams)
2046 /* get next frames from the same RTP packet */
2047 if (rt->cur_transport_priv) {
2048 if (rt->transport == RTSP_TRANSPORT_RDT) {
2049 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2050 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2051 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2052 } else if (CONFIG_RTPDEC && rt->ts) {
2053 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2055 rt->recvbuf_pos += ret;
2056 ret = rt->recvbuf_pos < rt->recvbuf_len;
2061 rt->cur_transport_priv = NULL;
2063 } else if (ret == 1) {
2066 rt->cur_transport_priv = NULL;
2070 if (rt->transport == RTSP_TRANSPORT_RTP) {
2072 int64_t first_queue_time = 0;
2073 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2074 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2078 queue_time = ff_rtp_queued_packet_time(rtpctx);
2079 if (queue_time && (queue_time - first_queue_time < 0 ||
2080 !first_queue_time)) {
2081 first_queue_time = queue_time;
2082 first_queue_st = rt->rtsp_streams[i];
2085 if (first_queue_time) {
2086 wait_end = first_queue_time + s->max_delay;
2089 first_queue_st = NULL;
2093 /* read next RTP packet */
2095 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2097 return AVERROR(ENOMEM);
2100 switch(rt->lower_transport) {
2102 #if CONFIG_RTSP_DEMUXER
2103 case RTSP_LOWER_TRANSPORT_TCP:
2104 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2107 case RTSP_LOWER_TRANSPORT_UDP:
2108 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2109 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2110 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2111 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2113 case RTSP_LOWER_TRANSPORT_CUSTOM:
2114 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2115 wait_end && wait_end < av_gettime_relative())
2116 len = AVERROR(EAGAIN);
2118 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2119 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
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, NULL, s->pb, len);
2124 if (len == AVERROR(EAGAIN) && first_queue_st &&
2125 rt->transport == RTSP_TRANSPORT_RTP) {
2126 av_log(s, AV_LOG_WARNING,
2127 "max delay reached. need to consume packet\n");
2128 rtsp_st = first_queue_st;
2129 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2136 if (rt->transport == RTSP_TRANSPORT_RDT) {
2137 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2138 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2139 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2140 if (rtsp_st->feedback) {
2141 AVIOContext *pb = NULL;
2142 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2144 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2147 /* Either bad packet, or a RTCP packet. Check if the
2148 * first_rtcp_ntp_time field was initialized. */
2149 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2150 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2151 /* first_rtcp_ntp_time has been initialized for this stream,
2152 * copy the same value to all other uninitialized streams,
2153 * in order to map their timestamp origin to the same ntp time
2156 AVStream *st = NULL;
2157 if (rtsp_st->stream_index >= 0)
2158 st = s->streams[rtsp_st->stream_index];
2159 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2160 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2161 AVStream *st2 = NULL;
2162 if (rt->rtsp_streams[i]->stream_index >= 0)
2163 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2164 if (rtpctx2 && st && st2 &&
2165 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2166 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2167 rtpctx2->rtcp_ts_offset = av_rescale_q(
2168 rtpctx->rtcp_ts_offset, st->time_base,
2172 // Make real NTP start time available in AVFormatContext
2173 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2174 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2176 s->start_time_realtime -=
2177 av_rescale (rtpctx->rtcp_ts_offset,
2178 (uint64_t) rtpctx->st->time_base.num * 1000000,
2179 rtpctx->st->time_base.den);
2183 if (ret == -RTCP_BYE) {
2186 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2187 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2189 if (rt->nb_byes == rt->nb_rtsp_streams)
2193 } else if (CONFIG_RTPDEC && rt->ts) {
2194 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2197 rt->recvbuf_len = len;
2198 rt->recvbuf_pos = ret;
2199 rt->cur_transport_priv = rt->ts;
2206 return AVERROR_INVALIDDATA;
2212 /* more packets may follow, so we save the RTP context */
2213 rt->cur_transport_priv = rtsp_st->transport_priv;
2217 #endif /* CONFIG_RTPDEC */
2219 #if CONFIG_SDP_DEMUXER
2220 static int sdp_probe(AVProbeData *p1)
2222 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2224 /* we look for a line beginning "c=IN IP" */
2225 while (p < p_end && *p != '\0') {
2226 if (sizeof("c=IN IP") - 1 < p_end - p &&
2227 av_strstart(p, "c=IN IP", NULL))
2228 return AVPROBE_SCORE_EXTENSION;
2230 while (p < p_end - 1 && *p != '\n') p++;
2239 static void append_source_addrs(char *buf, int size, const char *name,
2240 int count, struct RTSPSource **addrs)
2245 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2246 for (i = 1; i < count; i++)
2247 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2250 static int sdp_read_header(AVFormatContext *s)
2252 RTSPState *rt = s->priv_data;
2253 RTSPStream *rtsp_st;
2258 if (!ff_network_init())
2259 return AVERROR(EIO);
2261 if (s->max_delay < 0) /* Not set by the caller */
2262 s->max_delay = DEFAULT_REORDERING_DELAY;
2263 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2264 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2266 /* read the whole sdp file */
2267 /* XXX: better loading */
2268 content = av_malloc(SDP_MAX_SIZE);
2270 return AVERROR(ENOMEM);
2271 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2274 return AVERROR_INVALIDDATA;
2276 content[size] ='\0';
2278 err = ff_sdp_parse(s, content);
2282 /* open each RTP stream */
2283 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2285 rtsp_st = rt->rtsp_streams[i];
2287 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2288 AVDictionary *opts = map_to_opts(rt);
2290 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2291 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2292 ff_url_join(url, sizeof(url), "rtp", NULL,
2293 namebuf, rtsp_st->sdp_port,
2294 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2295 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2296 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2297 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2299 append_source_addrs(url, sizeof(url), "sources",
2300 rtsp_st->nb_include_source_addrs,
2301 rtsp_st->include_source_addrs);
2302 append_source_addrs(url, sizeof(url), "block",
2303 rtsp_st->nb_exclude_source_addrs,
2304 rtsp_st->exclude_source_addrs);
2305 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2306 &s->interrupt_callback, &opts);
2308 av_dict_free(&opts);
2311 err = AVERROR_INVALIDDATA;
2315 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2320 ff_rtsp_close_streams(s);
2325 static int sdp_read_close(AVFormatContext *s)
2327 ff_rtsp_close_streams(s);
2332 static const AVClass sdp_demuxer_class = {
2333 .class_name = "SDP demuxer",
2334 .item_name = av_default_item_name,
2335 .option = sdp_options,
2336 .version = LIBAVUTIL_VERSION_INT,
2339 AVInputFormat ff_sdp_demuxer = {
2341 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2342 .priv_data_size = sizeof(RTSPState),
2343 .read_probe = sdp_probe,
2344 .read_header = sdp_read_header,
2345 .read_packet = ff_rtsp_fetch_packet,
2346 .read_close = sdp_read_close,
2347 .priv_class = &sdp_demuxer_class,
2349 #endif /* CONFIG_SDP_DEMUXER */
2351 #if CONFIG_RTP_DEMUXER
2352 static int rtp_probe(AVProbeData *p)
2354 if (av_strstart(p->filename, "rtp:", NULL))
2355 return AVPROBE_SCORE_MAX;
2359 static int rtp_read_header(AVFormatContext *s)
2361 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2362 char host[500], sdp[500];
2364 URLContext* in = NULL;
2366 AVCodecContext codec = { 0 };
2367 struct sockaddr_storage addr;
2369 socklen_t addrlen = sizeof(addr);
2370 RTSPState *rt = s->priv_data;
2372 if (!ff_network_init())
2373 return AVERROR(EIO);
2375 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2376 &s->interrupt_callback, NULL);
2381 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2382 if (ret == AVERROR(EAGAIN))
2387 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2391 if ((recvbuf[0] & 0xc0) != 0x80) {
2392 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2397 if (RTP_PT_IS_RTCP(recvbuf[1]))
2400 payload_type = recvbuf[1] & 0x7f;
2403 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2407 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2408 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2409 "without an SDP file describing it\n",
2413 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2414 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2415 "properly you need an SDP file "
2419 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2420 NULL, 0, s->filename);
2422 snprintf(sdp, sizeof(sdp),
2423 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2424 addr.ss_family == AF_INET ? 4 : 6, host,
2425 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2426 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2427 port, payload_type);
2428 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2430 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2433 /* sdp_read_header initializes this again */
2436 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2438 ret = sdp_read_header(s);
2449 static const AVClass rtp_demuxer_class = {
2450 .class_name = "RTP demuxer",
2451 .item_name = av_default_item_name,
2452 .option = rtp_options,
2453 .version = LIBAVUTIL_VERSION_INT,
2456 AVInputFormat ff_rtp_demuxer = {
2458 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2459 .priv_data_size = sizeof(RTSPState),
2460 .read_probe = rtp_probe,
2461 .read_header = rtp_read_header,
2462 .read_packet = ff_rtsp_fetch_packet,
2463 .read_close = sdp_read_close,
2464 .flags = AVFMT_NOFILE,
2465 .priv_class = &rtp_demuxer_class,
2467 #endif /* CONFIG_RTP_DEMUXER */