3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "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" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 #define COMMON_OPTS() \
76 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }, \
77 { "buffer_size", "Underlying protocol send/receive buffer size", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
89 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101 { "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" },
102 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
107 static const AVOption rtp_options[] = {
108 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
114 static AVDictionary *map_to_opts(RTSPState *rt)
116 AVDictionary *opts = NULL;
119 snprintf(buf, sizeof(buf), "%d", rt->buffer_size);
120 av_dict_set(&opts, "buffer_size", buf, 0);
125 static void get_word_until_chars(char *buf, int buf_size,
126 const char *sep, const char **pp)
132 p += strspn(p, SPACE_CHARS);
134 while (!strchr(sep, *p) && *p != '\0') {
135 if ((q - buf) < buf_size - 1)
144 static void get_word_sep(char *buf, int buf_size, const char *sep,
147 if (**pp == '/') (*pp)++;
148 get_word_until_chars(buf, buf_size, sep, pp);
151 static void get_word(char *buf, int buf_size, const char **pp)
153 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
156 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
158 * Used for seeking in the rtp stream.
160 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
164 p += strspn(p, SPACE_CHARS);
165 if (!av_stristart(p, "npt=", &p))
168 *start = AV_NOPTS_VALUE;
169 *end = AV_NOPTS_VALUE;
171 get_word_sep(buf, sizeof(buf), "-", &p);
172 if (av_parse_time(start, buf, 1) < 0)
176 get_word_sep(buf, sizeof(buf), "-", &p);
177 av_parse_time(end, buf, 1);
181 static int get_sockaddr(AVFormatContext *s,
182 const char *buf, struct sockaddr_storage *sock)
184 struct addrinfo hints = { 0 }, *ai = NULL;
187 hints.ai_flags = AI_NUMERICHOST;
188 if ((ret = getaddrinfo(buf, NULL, &hints, &ai))) {
189 av_log(s, AV_LOG_ERROR, "getaddrinfo(%s): %s\n",
194 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
200 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
201 RTSPStream *rtsp_st, AVStream *st)
203 AVCodecContext *codec = st ? st->codec : NULL;
207 codec->codec_id = handler->codec_id;
208 rtsp_st->dynamic_handler = handler;
210 st->need_parsing = handler->need_parsing;
211 if (handler->priv_data_size) {
212 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
213 if (!rtsp_st->dynamic_protocol_context)
214 rtsp_st->dynamic_handler = NULL;
218 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
221 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
222 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
223 rtsp_st->dynamic_protocol_context);
225 if (rtsp_st->dynamic_protocol_context) {
226 if (rtsp_st->dynamic_handler->close)
227 rtsp_st->dynamic_handler->close(
228 rtsp_st->dynamic_protocol_context);
229 av_free(rtsp_st->dynamic_protocol_context);
231 rtsp_st->dynamic_protocol_context = NULL;
232 rtsp_st->dynamic_handler = NULL;
237 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
238 static int sdp_parse_rtpmap(AVFormatContext *s,
239 AVStream *st, RTSPStream *rtsp_st,
240 int payload_type, const char *p)
242 AVCodecContext *codec = st->codec;
248 /* See if we can handle this kind of payload.
249 * The space should normally not be there but some Real streams or
250 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
251 * have a trailing space. */
252 get_word_sep(buf, sizeof(buf), "/ ", &p);
253 if (payload_type < RTP_PT_PRIVATE) {
254 /* We are in a standard case
255 * (from http://www.iana.org/assignments/rtp-parameters). */
256 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
259 if (codec->codec_id == AV_CODEC_ID_NONE) {
260 RTPDynamicProtocolHandler *handler =
261 ff_rtp_handler_find_by_name(buf, codec->codec_type);
262 init_rtp_handler(handler, rtsp_st, st);
263 /* If no dynamic handler was found, check with the list of standard
264 * allocated types, if such a stream for some reason happens to
265 * use a private payload type. This isn't handled in rtpdec.c, since
266 * the format name from the rtpmap line never is passed into rtpdec. */
267 if (!rtsp_st->dynamic_handler)
268 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
271 c = avcodec_find_decoder(codec->codec_id);
277 get_word_sep(buf, sizeof(buf), "/", &p);
279 switch (codec->codec_type) {
280 case AVMEDIA_TYPE_AUDIO:
281 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
282 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
283 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
285 codec->sample_rate = i;
286 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
287 get_word_sep(buf, sizeof(buf), "/", &p);
292 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
294 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
297 case AVMEDIA_TYPE_VIDEO:
298 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
300 avpriv_set_pts_info(st, 32, 1, i);
305 finalize_rtp_handler_init(s, rtsp_st, st);
309 /* parse the attribute line from the fmtp a line of an sdp response. This
310 * is broken out as a function because it is used in rtp_h264.c, which is
312 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
313 char *value, int value_size)
315 *p += strspn(*p, SPACE_CHARS);
317 get_word_sep(attr, attr_size, "=", p);
320 get_word_sep(value, value_size, ";", p);
328 typedef struct SDPParseState {
330 struct sockaddr_storage default_ip;
332 int skip_media; ///< set if an unknown m= line occurs
333 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
334 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
335 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
336 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
339 char delayed_fmtp[2048];
342 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
343 struct RTSPSource ***dest, int *dest_count)
345 RTSPSource *rtsp_src, *rtsp_src2;
347 for (i = 0; i < count; i++) {
349 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
352 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
353 dynarray_add(dest, dest_count, rtsp_src2);
357 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
358 int payload_type, const char *line)
362 for (i = 0; i < rt->nb_rtsp_streams; i++) {
363 RTSPStream *rtsp_st = rt->rtsp_streams[i];
364 if (rtsp_st->sdp_payload_type == payload_type &&
365 rtsp_st->dynamic_handler &&
366 rtsp_st->dynamic_handler->parse_sdp_a_line) {
367 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
368 rtsp_st->dynamic_protocol_context, line);
373 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
374 int letter, const char *buf)
376 RTSPState *rt = s->priv_data;
377 char buf1[64], st_type[64];
379 enum AVMediaType codec_type;
383 RTSPSource *rtsp_src;
384 struct sockaddr_storage sdp_ip;
387 av_log(s, AV_LOG_TRACE, "sdp: %c='%s'\n", letter, buf);
390 if (s1->skip_media && letter != 'm')
394 get_word(buf1, sizeof(buf1), &p);
395 if (strcmp(buf1, "IN") != 0)
397 get_word(buf1, sizeof(buf1), &p);
398 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
400 get_word_sep(buf1, sizeof(buf1), "/", &p);
401 if (get_sockaddr(s, buf1, &sdp_ip))
406 get_word_sep(buf1, sizeof(buf1), "/", &p);
409 if (s->nb_streams == 0) {
410 s1->default_ip = sdp_ip;
411 s1->default_ttl = ttl;
413 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
414 rtsp_st->sdp_ip = sdp_ip;
415 rtsp_st->sdp_ttl = ttl;
419 av_dict_set(&s->metadata, "title", p, 0);
422 if (s->nb_streams == 0) {
423 av_dict_set(&s->metadata, "comment", p, 0);
432 codec_type = AVMEDIA_TYPE_UNKNOWN;
433 get_word(st_type, sizeof(st_type), &p);
434 if (!strcmp(st_type, "audio")) {
435 codec_type = AVMEDIA_TYPE_AUDIO;
436 } else if (!strcmp(st_type, "video")) {
437 codec_type = AVMEDIA_TYPE_VIDEO;
438 } else if (!strcmp(st_type, "application") || !strcmp(st_type, "text")) {
439 codec_type = AVMEDIA_TYPE_DATA;
441 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
445 rtsp_st = av_mallocz(sizeof(RTSPStream));
448 rtsp_st->stream_index = -1;
449 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
451 rtsp_st->sdp_ip = s1->default_ip;
452 rtsp_st->sdp_ttl = s1->default_ttl;
454 copy_default_source_addrs(s1->default_include_source_addrs,
455 s1->nb_default_include_source_addrs,
456 &rtsp_st->include_source_addrs,
457 &rtsp_st->nb_include_source_addrs);
458 copy_default_source_addrs(s1->default_exclude_source_addrs,
459 s1->nb_default_exclude_source_addrs,
460 &rtsp_st->exclude_source_addrs,
461 &rtsp_st->nb_exclude_source_addrs);
463 get_word(buf1, sizeof(buf1), &p); /* port */
464 rtsp_st->sdp_port = atoi(buf1);
466 get_word(buf1, sizeof(buf1), &p); /* protocol */
467 if (!strcmp(buf1, "udp"))
468 rt->transport = RTSP_TRANSPORT_RAW;
469 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
470 rtsp_st->feedback = 1;
472 /* XXX: handle list of formats */
473 get_word(buf1, sizeof(buf1), &p); /* format list */
474 rtsp_st->sdp_payload_type = atoi(buf1);
476 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
477 /* no corresponding stream */
478 if (rt->transport == RTSP_TRANSPORT_RAW) {
479 if (CONFIG_RTPDEC && !rt->ts)
480 rt->ts = ff_mpegts_parse_open(s);
482 RTPDynamicProtocolHandler *handler;
483 handler = ff_rtp_handler_find_by_id(
484 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
485 init_rtp_handler(handler, rtsp_st, NULL);
486 finalize_rtp_handler_init(s, rtsp_st, NULL);
488 } else if (rt->server_type == RTSP_SERVER_WMS &&
489 codec_type == AVMEDIA_TYPE_DATA) {
490 /* RTX stream, a stream that carries all the other actual
491 * audio/video streams. Don't expose this to the callers. */
493 st = avformat_new_stream(s, NULL);
496 st->id = rt->nb_rtsp_streams - 1;
497 rtsp_st->stream_index = st->index;
498 st->codec->codec_type = codec_type;
499 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
500 RTPDynamicProtocolHandler *handler;
501 /* if standard payload type, we can find the codec right now */
502 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
503 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
504 st->codec->sample_rate > 0)
505 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
506 /* Even static payload types may need a custom depacketizer */
507 handler = ff_rtp_handler_find_by_id(
508 rtsp_st->sdp_payload_type, st->codec->codec_type);
509 init_rtp_handler(handler, rtsp_st, st);
510 finalize_rtp_handler_init(s, rtsp_st, st);
512 if (rt->default_lang[0])
513 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
515 /* put a default control url */
516 av_strlcpy(rtsp_st->control_url, rt->control_uri,
517 sizeof(rtsp_st->control_url));
520 if (av_strstart(p, "control:", &p)) {
521 if (s->nb_streams == 0) {
522 if (!strncmp(p, "rtsp://", 7))
523 av_strlcpy(rt->control_uri, p,
524 sizeof(rt->control_uri));
527 /* get the control url */
528 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
530 /* XXX: may need to add full url resolution */
531 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
533 if (proto[0] == '\0') {
534 /* relative control URL */
535 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
536 av_strlcat(rtsp_st->control_url, "/",
537 sizeof(rtsp_st->control_url));
538 av_strlcat(rtsp_st->control_url, p,
539 sizeof(rtsp_st->control_url));
541 av_strlcpy(rtsp_st->control_url, p,
542 sizeof(rtsp_st->control_url));
544 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
545 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
546 get_word(buf1, sizeof(buf1), &p);
547 payload_type = atoi(buf1);
548 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
549 if (rtsp_st->stream_index >= 0) {
550 st = s->streams[rtsp_st->stream_index];
551 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
555 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
557 } else if (av_strstart(p, "fmtp:", &p) ||
558 av_strstart(p, "framesize:", &p)) {
559 // let dynamic protocol handlers have a stab at the line.
560 get_word(buf1, sizeof(buf1), &p);
561 payload_type = atoi(buf1);
562 if (s1->seen_rtpmap) {
563 parse_fmtp(s, rt, payload_type, buf);
566 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
568 } else if (av_strstart(p, "range:", &p)) {
571 // this is so that seeking on a streamed file can work.
572 rtsp_parse_range_npt(p, &start, &end);
573 s->start_time = start;
574 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
575 s->duration = (end == AV_NOPTS_VALUE) ?
576 AV_NOPTS_VALUE : end - start;
577 } else if (av_strstart(p, "lang:", &p)) {
578 if (s->nb_streams > 0) {
579 get_word(buf1, sizeof(buf1), &p);
580 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
581 if (rtsp_st->stream_index >= 0) {
582 st = s->streams[rtsp_st->stream_index];
583 av_dict_set(&st->metadata, "language", buf1, 0);
586 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
587 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
589 rt->transport = RTSP_TRANSPORT_RDT;
590 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
592 st = s->streams[s->nb_streams - 1];
593 st->codec->sample_rate = atoi(p);
594 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
596 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
597 get_word(buf1, sizeof(buf1), &p); // ignore tag
598 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
599 p += strspn(p, SPACE_CHARS);
600 if (av_strstart(p, "inline:", &p))
601 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
602 } else if (av_strstart(p, "source-filter:", &p)) {
604 get_word(buf1, sizeof(buf1), &p);
605 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
607 exclude = !strcmp(buf1, "excl");
609 get_word(buf1, sizeof(buf1), &p);
610 if (strcmp(buf1, "IN") != 0)
612 get_word(buf1, sizeof(buf1), &p);
613 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
615 // not checking that the destination address actually matches or is wildcard
616 get_word(buf1, sizeof(buf1), &p);
619 rtsp_src = av_mallocz(sizeof(*rtsp_src));
622 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
624 if (s->nb_streams == 0) {
625 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
627 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
628 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
631 if (s->nb_streams == 0) {
632 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
634 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
635 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
640 if (rt->server_type == RTSP_SERVER_WMS)
641 ff_wms_parse_sdp_a_line(s, p);
642 if (s->nb_streams > 0) {
643 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
645 if (rt->server_type == RTSP_SERVER_REAL)
646 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
648 if (rtsp_st->dynamic_handler &&
649 rtsp_st->dynamic_handler->parse_sdp_a_line)
650 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
651 rtsp_st->stream_index,
652 rtsp_st->dynamic_protocol_context, buf);
659 int ff_sdp_parse(AVFormatContext *s, const char *content)
661 RTSPState *rt = s->priv_data;
664 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
665 * contain long SDP lines containing complete ASF Headers (several
666 * kB) or arrays of MDPR (RM stream descriptor) headers plus
667 * "rulebooks" describing their properties. Therefore, the SDP line
670 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
671 * in rtpdec_xiph.c. */
673 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
677 p += strspn(p, SPACE_CHARS);
685 /* get the content */
687 while (*p != '\n' && *p != '\r' && *p != '\0') {
688 if ((q - buf) < sizeof(buf) - 1)
693 sdp_parse_line(s, s1, letter, buf);
695 while (*p != '\n' && *p != '\0')
701 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
702 av_free(s1->default_include_source_addrs[i]);
703 av_freep(&s1->default_include_source_addrs);
704 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
705 av_free(s1->default_exclude_source_addrs[i]);
706 av_freep(&s1->default_exclude_source_addrs);
708 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
709 if (!rt->p) return AVERROR(ENOMEM);
712 #endif /* CONFIG_RTPDEC */
714 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
716 RTSPState *rt = s->priv_data;
719 for (i = 0; i < rt->nb_rtsp_streams; i++) {
720 RTSPStream *rtsp_st = rt->rtsp_streams[i];
723 if (rtsp_st->transport_priv) {
725 AVFormatContext *rtpctx = rtsp_st->transport_priv;
726 av_write_trailer(rtpctx);
727 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
728 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
729 ff_rtsp_tcp_write_packet(s, rtsp_st);
730 ffio_free_dyn_buf(&rtpctx->pb);
732 avio_close(rtpctx->pb);
734 avformat_free_context(rtpctx);
735 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
736 ff_rdt_parse_close(rtsp_st->transport_priv);
737 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
738 ff_rtp_parse_close(rtsp_st->transport_priv);
740 rtsp_st->transport_priv = NULL;
741 if (rtsp_st->rtp_handle)
742 ffurl_close(rtsp_st->rtp_handle);
743 rtsp_st->rtp_handle = NULL;
747 /* close and free RTSP streams */
748 void ff_rtsp_close_streams(AVFormatContext *s)
750 RTSPState *rt = s->priv_data;
754 ff_rtsp_undo_setup(s, 0);
755 for (i = 0; i < rt->nb_rtsp_streams; i++) {
756 rtsp_st = rt->rtsp_streams[i];
758 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
759 if (rtsp_st->dynamic_handler->close)
760 rtsp_st->dynamic_handler->close(
761 rtsp_st->dynamic_protocol_context);
762 av_free(rtsp_st->dynamic_protocol_context);
764 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
765 av_free(rtsp_st->include_source_addrs[j]);
766 av_freep(&rtsp_st->include_source_addrs);
767 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
768 av_free(rtsp_st->exclude_source_addrs[j]);
769 av_freep(&rtsp_st->exclude_source_addrs);
774 av_free(rt->rtsp_streams);
776 avformat_close_input(&rt->asf_ctx);
778 if (CONFIG_RTPDEC && rt->ts)
779 ff_mpegts_parse_close(rt->ts);
781 av_free(rt->recvbuf);
784 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
786 RTSPState *rt = s->priv_data;
788 int reordering_queue_size = rt->reordering_queue_size;
789 if (reordering_queue_size < 0) {
790 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
791 reordering_queue_size = 0;
793 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
796 /* open the RTP context */
797 if (rtsp_st->stream_index >= 0)
798 st = s->streams[rtsp_st->stream_index];
800 s->ctx_flags |= AVFMTCTX_NOHEADER;
802 if (CONFIG_RTSP_MUXER && s->oformat) {
803 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
804 s, st, rtsp_st->rtp_handle,
805 RTSP_TCP_MAX_PACKET_SIZE,
806 rtsp_st->stream_index);
807 /* Ownership of rtp_handle is passed to the rtp mux context */
808 rtsp_st->rtp_handle = NULL;
811 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
812 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
813 return 0; // Don't need to open any parser here
814 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
815 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
816 rtsp_st->dynamic_protocol_context,
817 rtsp_st->dynamic_handler);
818 else if (CONFIG_RTPDEC)
819 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
820 rtsp_st->sdp_payload_type,
821 reordering_queue_size);
823 if (!rtsp_st->transport_priv) {
824 return AVERROR(ENOMEM);
825 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
826 if (rtsp_st->dynamic_handler) {
827 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
828 rtsp_st->dynamic_protocol_context,
829 rtsp_st->dynamic_handler);
831 if (rtsp_st->crypto_suite[0])
832 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
833 rtsp_st->crypto_suite,
834 rtsp_st->crypto_params);
840 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
841 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
848 q += strspn(q, SPACE_CHARS);
849 v = strtol(q, &p, 10);
853 v = strtol(p, &p, 10);
862 /* XXX: only one transport specification is parsed */
863 static void rtsp_parse_transport(AVFormatContext *s,
864 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(s, 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(AVFormatContext *s,
1044 RTSPMessageHeader *reply, const char *buf,
1045 RTSPState *rt, const char *method)
1049 /* NOTE: we do case independent match for broken servers */
1051 if (av_stristart(p, "Session:", &p)) {
1053 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1054 if (av_stristart(p, ";timeout=", &p) &&
1055 (t = strtol(p, NULL, 10)) > 0) {
1058 } else if (av_stristart(p, "Content-Length:", &p)) {
1059 reply->content_length = strtol(p, NULL, 10);
1060 } else if (av_stristart(p, "Transport:", &p)) {
1061 rtsp_parse_transport(s, reply, p);
1062 } else if (av_stristart(p, "CSeq:", &p)) {
1063 reply->seq = strtol(p, NULL, 10);
1064 } else if (av_stristart(p, "Range:", &p)) {
1065 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1066 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1067 p += strspn(p, SPACE_CHARS);
1068 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1069 } else if (av_stristart(p, "Server:", &p)) {
1070 p += strspn(p, SPACE_CHARS);
1071 av_strlcpy(reply->server, p, sizeof(reply->server));
1072 } else if (av_stristart(p, "Notice:", &p) ||
1073 av_stristart(p, "X-Notice:", &p)) {
1074 reply->notice = strtol(p, NULL, 10);
1075 } else if (av_stristart(p, "Location:", &p)) {
1076 p += strspn(p, SPACE_CHARS);
1077 av_strlcpy(reply->location, p , sizeof(reply->location));
1078 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1079 p += strspn(p, SPACE_CHARS);
1080 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1081 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1082 p += strspn(p, SPACE_CHARS);
1083 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1084 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1085 p += strspn(p, SPACE_CHARS);
1086 if (method && !strcmp(method, "DESCRIBE"))
1087 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1088 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1089 p += strspn(p, SPACE_CHARS);
1090 if (method && !strcmp(method, "PLAY"))
1091 rtsp_parse_rtp_info(rt, p);
1092 } else if (av_stristart(p, "Public:", &p) && rt) {
1093 if (strstr(p, "GET_PARAMETER") &&
1094 method && !strcmp(method, "OPTIONS"))
1095 rt->get_parameter_supported = 1;
1096 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1097 p += strspn(p, SPACE_CHARS);
1098 rt->accept_dynamic_rate = atoi(p);
1099 } else if (av_stristart(p, "Content-Type:", &p)) {
1100 p += strspn(p, SPACE_CHARS);
1101 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1105 /* skip a RTP/TCP interleaved packet */
1106 void ff_rtsp_skip_packet(AVFormatContext *s)
1108 RTSPState *rt = s->priv_data;
1112 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1115 len = AV_RB16(buf + 1);
1117 av_log(s, AV_LOG_TRACE, "skipping RTP packet len=%d\n", len);
1122 if (len1 > sizeof(buf))
1124 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1131 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1132 unsigned char **content_ptr,
1133 int return_on_interleaved_data, const char *method)
1135 RTSPState *rt = s->priv_data;
1136 char buf[4096], buf1[1024], *q;
1139 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 == '$' && 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)
1171 av_log(s, AV_LOG_TRACE, "line='%s'\n", buf);
1173 /* test if last line */
1177 if (line_count == 0) {
1178 /* get reply code */
1179 get_word(buf1, sizeof(buf1), &p);
1180 if (!strncmp(buf1, "RTSP/", 5)) {
1181 get_word(buf1, sizeof(buf1), &p);
1182 reply->status_code = atoi(buf1);
1183 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1185 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1186 get_word(buf1, sizeof(buf1), &p); // object
1190 ff_rtsp_parse_line(s, reply, p, rt, method);
1191 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1192 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1197 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1198 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1200 content_length = reply->content_length;
1201 if (content_length > 0) {
1202 /* leave some room for a trailing '\0' (useful for simple parsing) */
1203 content = av_malloc(content_length + 1);
1205 return AVERROR(ENOMEM);
1206 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1207 content[content_length] = '\0';
1210 *content_ptr = content;
1216 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1217 const char* ptr = buf;
1219 if (!strcmp(reply->reason, "OPTIONS")) {
1220 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1222 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1223 if (reply->session_id[0])
1224 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1227 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1229 av_strlcat(buf, "\r\n", sizeof(buf));
1231 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1232 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1235 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1237 rt->last_cmd_time = av_gettime_relative();
1238 /* Even if the request from the server had data, it is not the data
1239 * that the caller wants or expects. The memory could also be leaked
1240 * if the actual following reply has content data. */
1242 av_freep(content_ptr);
1243 /* If method is set, this is called from ff_rtsp_send_cmd,
1244 * where a reply to exactly this request is awaited. For
1245 * callers from within packet receiving, we just want to
1246 * return to the caller and go back to receiving packets. */
1252 if (rt->seq != reply->seq) {
1253 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1254 rt->seq, reply->seq);
1258 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1259 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1260 reply->notice == 2306 /* Continuous Feed Terminated */) {
1261 rt->state = RTSP_STATE_IDLE;
1262 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1263 return AVERROR(EIO); /* data or server error */
1264 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1265 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1266 return AVERROR(EPERM);
1272 * Send a command to the RTSP server without waiting for the reply.
1274 * @param s RTSP (de)muxer context
1275 * @param method the method for the request
1276 * @param url the target url for the request
1277 * @param headers extra header lines to include in the request
1278 * @param send_content if non-null, the data to send as request body content
1279 * @param send_content_length the length of the send_content data, or 0 if
1280 * send_content is null
1282 * @return zero if success, nonzero otherwise
1284 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1285 const char *method, const char *url,
1286 const char *headers,
1287 const unsigned char *send_content,
1288 int send_content_length)
1290 RTSPState *rt = s->priv_data;
1291 char buf[4096], *out_buf;
1292 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1294 /* Add in RTSP headers */
1297 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1299 av_strlcat(buf, headers, sizeof(buf));
1300 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1301 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1302 if (rt->session_id[0] != '\0' && (!headers ||
1303 !strstr(headers, "\nIf-Match:"))) {
1304 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1307 char *str = ff_http_auth_create_response(&rt->auth_state,
1308 rt->auth, url, method);
1310 av_strlcat(buf, str, sizeof(buf));
1313 if (send_content_length > 0 && send_content)
1314 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1315 av_strlcat(buf, "\r\n", sizeof(buf));
1317 /* base64 encode rtsp if tunneling */
1318 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1319 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1320 out_buf = base64buf;
1323 av_log(s, AV_LOG_TRACE, "Sending:\n%s--\n", buf);
1325 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1326 if (send_content_length > 0 && send_content) {
1327 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1328 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1329 "with content data not supported\n");
1330 return AVERROR_PATCHWELCOME;
1332 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1334 rt->last_cmd_time = av_gettime_relative();
1339 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1340 const char *url, const char *headers)
1342 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1345 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1346 const char *headers, RTSPMessageHeader *reply,
1347 unsigned char **content_ptr)
1349 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1350 content_ptr, NULL, 0);
1353 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1354 const char *method, const char *url,
1356 RTSPMessageHeader *reply,
1357 unsigned char **content_ptr,
1358 const unsigned char *send_content,
1359 int send_content_length)
1361 RTSPState *rt = s->priv_data;
1362 HTTPAuthType cur_auth_type;
1363 int ret, attempts = 0;
1366 cur_auth_type = rt->auth_state.auth_type;
1367 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1369 send_content_length)))
1372 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1376 if (reply->status_code == 401 &&
1377 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1378 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1381 if (reply->status_code > 400){
1382 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1386 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1392 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1393 int lower_transport, const char *real_challenge)
1395 RTSPState *rt = s->priv_data;
1396 int rtx = 0, j, i, err, interleave = 0, port_off;
1397 RTSPStream *rtsp_st;
1398 RTSPMessageHeader reply1, *reply = &reply1;
1400 const char *trans_pref;
1402 if (rt->transport == RTSP_TRANSPORT_RDT)
1403 trans_pref = "x-pn-tng";
1404 else if (rt->transport == RTSP_TRANSPORT_RAW)
1405 trans_pref = "RAW/RAW";
1407 trans_pref = "RTP/AVP";
1409 /* default timeout: 1 minute */
1412 /* for each stream, make the setup request */
1413 /* XXX: we assume the same server is used for the control of each
1416 /* Choose a random starting offset within the first half of the
1417 * port range, to allow for a number of ports to try even if the offset
1418 * happens to be at the end of the random range. */
1419 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1420 /* even random offset */
1421 port_off -= port_off & 0x01;
1423 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1424 char transport[2048];
1427 * WMS serves all UDP data over a single connection, the RTX, which
1428 * isn't necessarily the first in the SDP but has to be the first
1429 * to be set up, else the second/third SETUP will fail with a 461.
1431 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1432 rt->server_type == RTSP_SERVER_WMS) {
1435 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1436 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1438 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1442 if (rtx == rt->nb_rtsp_streams)
1443 return -1; /* no RTX found */
1444 rtsp_st = rt->rtsp_streams[rtx];
1446 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1448 rtsp_st = rt->rtsp_streams[i];
1451 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1454 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1455 port = reply->transports[0].client_port_min;
1459 /* first try in specified port range */
1460 while (j <= rt->rtp_port_max) {
1461 AVDictionary *opts = map_to_opts(rt);
1463 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1464 "?localport=%d", j);
1465 /* we will use two ports per rtp stream (rtp and rtcp) */
1467 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1468 &s->interrupt_callback, &opts);
1470 av_dict_free(&opts);
1476 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1481 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1483 snprintf(transport, sizeof(transport) - 1,
1484 "%s/UDP;", trans_pref);
1485 if (rt->server_type != RTSP_SERVER_REAL)
1486 av_strlcat(transport, "unicast;", sizeof(transport));
1487 av_strlcatf(transport, sizeof(transport),
1488 "client_port=%d", port);
1489 if (rt->transport == RTSP_TRANSPORT_RTP &&
1490 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1491 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1495 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1496 /* For WMS streams, the application streams are only used for
1497 * UDP. When trying to set it up for TCP streams, the server
1498 * will return an error. Therefore, we skip those streams. */
1499 if (rt->server_type == RTSP_SERVER_WMS &&
1500 (rtsp_st->stream_index < 0 ||
1501 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1504 snprintf(transport, sizeof(transport) - 1,
1505 "%s/TCP;", trans_pref);
1506 if (rt->transport != RTSP_TRANSPORT_RDT)
1507 av_strlcat(transport, "unicast;", sizeof(transport));
1508 av_strlcatf(transport, sizeof(transport),
1509 "interleaved=%d-%d",
1510 interleave, interleave + 1);
1514 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1515 snprintf(transport, sizeof(transport) - 1,
1516 "%s/UDP;multicast", trans_pref);
1519 av_strlcat(transport, ";mode=record", sizeof(transport));
1520 } else if (rt->server_type == RTSP_SERVER_REAL ||
1521 rt->server_type == RTSP_SERVER_WMS)
1522 av_strlcat(transport, ";mode=play", sizeof(transport));
1523 snprintf(cmd, sizeof(cmd),
1524 "Transport: %s\r\n",
1526 if (rt->accept_dynamic_rate)
1527 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1528 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1529 char real_res[41], real_csum[9];
1530 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1532 av_strlcatf(cmd, sizeof(cmd),
1534 "RealChallenge2: %s, sd=%s\r\n",
1535 rt->session_id, real_res, real_csum);
1537 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1538 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1541 } else if (reply->status_code != RTSP_STATUS_OK ||
1542 reply->nb_transports != 1) {
1543 err = AVERROR_INVALIDDATA;
1547 /* XXX: same protocol for all streams is required */
1549 if (reply->transports[0].lower_transport != rt->lower_transport ||
1550 reply->transports[0].transport != rt->transport) {
1551 err = AVERROR_INVALIDDATA;
1555 rt->lower_transport = reply->transports[0].lower_transport;
1556 rt->transport = reply->transports[0].transport;
1559 /* Fail if the server responded with another lower transport mode
1560 * than what we requested. */
1561 if (reply->transports[0].lower_transport != lower_transport) {
1562 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1563 err = AVERROR_INVALIDDATA;
1567 switch(reply->transports[0].lower_transport) {
1568 case RTSP_LOWER_TRANSPORT_TCP:
1569 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1570 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1573 case RTSP_LOWER_TRANSPORT_UDP: {
1574 char url[1024], options[30] = "";
1575 const char *peer = host;
1577 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1578 av_strlcpy(options, "?connect=1", sizeof(options));
1579 /* Use source address if specified */
1580 if (reply->transports[0].source[0])
1581 peer = reply->transports[0].source;
1582 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1583 reply->transports[0].server_port_min, "%s", options);
1584 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1585 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1586 err = AVERROR_INVALIDDATA;
1591 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1592 char url[1024], namebuf[50], optbuf[20] = "";
1593 struct sockaddr_storage addr;
1596 if (reply->transports[0].destination.ss_family) {
1597 addr = reply->transports[0].destination;
1598 port = reply->transports[0].port_min;
1599 ttl = reply->transports[0].ttl;
1601 addr = rtsp_st->sdp_ip;
1602 port = rtsp_st->sdp_port;
1603 ttl = rtsp_st->sdp_ttl;
1606 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1607 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1608 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1609 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1610 port, "%s", optbuf);
1611 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1612 &s->interrupt_callback, NULL) < 0) {
1613 err = AVERROR_INVALIDDATA;
1620 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1624 if (rt->nb_rtsp_streams && reply->timeout > 0)
1625 rt->timeout = reply->timeout;
1627 if (rt->server_type == RTSP_SERVER_REAL)
1628 rt->need_subscription = 1;
1633 ff_rtsp_undo_setup(s, 0);
1637 void ff_rtsp_close_connections(AVFormatContext *s)
1639 RTSPState *rt = s->priv_data;
1640 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1641 ffurl_close(rt->rtsp_hd);
1642 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1645 int ff_rtsp_connect(AVFormatContext *s)
1647 RTSPState *rt = s->priv_data;
1648 char proto[128], host[1024], path[1024];
1649 char tcpname[1024], cmd[2048], auth[128];
1650 const char *lower_rtsp_proto = "tcp";
1651 int port, err, tcp_fd;
1652 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1653 int lower_transport_mask = 0;
1654 int default_port = RTSP_DEFAULT_PORT;
1655 char real_challenge[64] = "";
1656 struct sockaddr_storage peer;
1657 socklen_t peer_len = sizeof(peer);
1659 if (rt->rtp_port_max < rt->rtp_port_min) {
1660 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1661 "than min port %d\n", rt->rtp_port_max,
1663 return AVERROR(EINVAL);
1666 if (!ff_network_init())
1667 return AVERROR(EIO);
1669 if (s->max_delay < 0) /* Not set by the caller */
1670 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1672 rt->control_transport = RTSP_MODE_PLAIN;
1673 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1674 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1675 rt->control_transport = RTSP_MODE_TUNNEL;
1677 /* Only pass through valid flags from here */
1678 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1681 /* extract hostname and port */
1682 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1683 host, sizeof(host), &port, path, sizeof(path), s->filename);
1685 if (!strcmp(proto, "rtsps")) {
1686 lower_rtsp_proto = "tls";
1687 default_port = RTSPS_DEFAULT_PORT;
1688 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1692 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1695 port = default_port;
1697 lower_transport_mask = rt->lower_transport_mask;
1699 if (!lower_transport_mask)
1700 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1703 /* Only UDP or TCP - UDP multicast isn't supported. */
1704 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1705 (1 << RTSP_LOWER_TRANSPORT_TCP);
1706 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1707 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1708 "only UDP and TCP are supported for output.\n");
1709 err = AVERROR(EINVAL);
1714 /* Construct the URI used in request; this is similar to s->filename,
1715 * but with authentication credentials removed and RTSP specific options
1717 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1718 host, port, "%s", path);
1720 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1721 /* set up initial handshake for tunneling */
1722 char httpname[1024];
1723 char sessioncookie[17];
1726 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1727 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1728 av_get_random_seed(), av_get_random_seed());
1731 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1732 &s->interrupt_callback) < 0) {
1737 /* generate GET headers */
1738 snprintf(headers, sizeof(headers),
1739 "x-sessioncookie: %s\r\n"
1740 "Accept: application/x-rtsp-tunnelled\r\n"
1741 "Pragma: no-cache\r\n"
1742 "Cache-Control: no-cache\r\n",
1744 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1746 /* complete the connection */
1747 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1753 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1754 &s->interrupt_callback) < 0 ) {
1759 /* generate POST headers */
1760 snprintf(headers, sizeof(headers),
1761 "x-sessioncookie: %s\r\n"
1762 "Content-Type: application/x-rtsp-tunnelled\r\n"
1763 "Pragma: no-cache\r\n"
1764 "Cache-Control: no-cache\r\n"
1765 "Content-Length: 32767\r\n"
1766 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1768 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1769 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1771 /* Initialize the authentication state for the POST session. The HTTP
1772 * protocol implementation doesn't properly handle multi-pass
1773 * authentication for POST requests, since it would require one of
1775 * - implementing Expect: 100-continue, which many HTTP servers
1776 * don't support anyway, even less the RTSP servers that do HTTP
1778 * - sending the whole POST data until getting a 401 reply specifying
1779 * what authentication method to use, then resending all that data
1780 * - waiting for potential 401 replies directly after sending the
1781 * POST header (waiting for some unspecified time)
1782 * Therefore, we copy the full auth state, which works for both basic
1783 * and digest. (For digest, we would have to synchronize the nonce
1784 * count variable between the two sessions, if we'd do more requests
1785 * with the original session, though.)
1787 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1789 /* complete the connection */
1790 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1795 /* open the tcp connection */
1796 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1798 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1799 &s->interrupt_callback, NULL) < 0) {
1803 rt->rtsp_hd_out = rt->rtsp_hd;
1807 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1812 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1813 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1814 NULL, 0, NI_NUMERICHOST);
1817 /* request options supported by the server; this also detects server
1819 for (rt->server_type = RTSP_SERVER_RTP;;) {
1821 if (rt->server_type == RTSP_SERVER_REAL)
1824 * The following entries are required for proper
1825 * streaming from a Realmedia server. They are
1826 * interdependent in some way although we currently
1827 * don't quite understand how. Values were copied
1828 * from mplayer SVN r23589.
1829 * ClientChallenge is a 16-byte ID in hex
1830 * CompanyID is a 16-byte ID in base64
1832 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1833 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1834 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1835 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1837 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1838 if (reply->status_code != RTSP_STATUS_OK) {
1839 err = AVERROR_INVALIDDATA;
1843 /* detect server type if not standard-compliant RTP */
1844 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1845 rt->server_type = RTSP_SERVER_REAL;
1847 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1848 rt->server_type = RTSP_SERVER_WMS;
1849 } else if (rt->server_type == RTSP_SERVER_REAL)
1850 strcpy(real_challenge, reply->real_challenge);
1854 if (CONFIG_RTSP_DEMUXER && s->iformat)
1855 err = ff_rtsp_setup_input_streams(s, reply);
1856 else if (CONFIG_RTSP_MUXER)
1857 err = ff_rtsp_setup_output_streams(s, host);
1862 int lower_transport = ff_log2_tab[lower_transport_mask &
1863 ~(lower_transport_mask - 1)];
1865 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1866 rt->server_type == RTSP_SERVER_REAL ?
1867 real_challenge : NULL);
1870 lower_transport_mask &= ~(1 << lower_transport);
1871 if (lower_transport_mask == 0 && err == 1) {
1872 err = AVERROR(EPROTONOSUPPORT);
1877 rt->lower_transport_mask = lower_transport_mask;
1878 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1879 rt->state = RTSP_STATE_IDLE;
1880 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1883 ff_rtsp_close_streams(s);
1884 ff_rtsp_close_connections(s);
1885 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1886 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1887 rt->session_id[0] = '\0';
1888 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1896 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1899 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1900 uint8_t *buf, int buf_size, int64_t wait_end)
1902 RTSPState *rt = s->priv_data;
1903 RTSPStream *rtsp_st;
1904 int n, i, ret, tcp_fd, timeout_cnt = 0;
1906 struct pollfd *p = rt->p;
1907 int *fds = NULL, fdsnum, fdsidx;
1910 if (ff_check_interrupt(&s->interrupt_callback))
1911 return AVERROR_EXIT;
1912 if (wait_end && wait_end - av_gettime_relative() < 0)
1913 return AVERROR(EAGAIN);
1916 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1917 p[max_p].fd = tcp_fd;
1918 p[max_p++].events = POLLIN;
1922 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1923 rtsp_st = rt->rtsp_streams[i];
1924 if (rtsp_st->rtp_handle) {
1925 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1927 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1931 av_log(s, AV_LOG_ERROR,
1932 "Number of fds %d not supported\n", fdsnum);
1933 return AVERROR_INVALIDDATA;
1935 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1936 p[max_p].fd = fds[fdsidx];
1937 p[max_p++].events = POLLIN;
1942 n = poll(p, max_p, POLL_TIMEOUT_MS);
1944 int j = 1 - (tcp_fd == -1);
1946 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1947 rtsp_st = rt->rtsp_streams[i];
1948 if (rtsp_st->rtp_handle) {
1949 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1950 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1952 *prtsp_st = rtsp_st;
1959 #if CONFIG_RTSP_DEMUXER
1960 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1961 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1962 if (rt->state == RTSP_STATE_STREAMING) {
1963 if (!ff_rtsp_parse_streaming_commands(s))
1966 av_log(s, AV_LOG_WARNING,
1967 "Unable to answer to TEARDOWN\n");
1971 RTSPMessageHeader reply;
1972 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1975 /* XXX: parse message */
1976 if (rt->state != RTSP_STATE_STREAMING)
1981 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1982 return AVERROR(ETIMEDOUT);
1983 } else if (n < 0 && errno != EINTR)
1984 return AVERROR(errno);
1988 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1989 const uint8_t *buf, int len)
1991 RTSPState *rt = s->priv_data;
1995 if (rt->nb_rtsp_streams == 1) {
1996 *rtsp_st = rt->rtsp_streams[0];
1999 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
2000 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
2002 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2003 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2006 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2007 *rtsp_st = rt->rtsp_streams[i];
2014 av_log(s, AV_LOG_WARNING,
2015 "Unable to pick stream for packet - SSRC not known for "
2017 return AVERROR(EAGAIN);
2020 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2021 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2022 *rtsp_st = rt->rtsp_streams[i];
2028 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2029 return AVERROR(EAGAIN);
2032 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2034 RTSPState *rt = s->priv_data;
2036 RTSPStream *rtsp_st, *first_queue_st = NULL;
2037 int64_t wait_end = 0;
2039 if (rt->nb_byes == rt->nb_rtsp_streams)
2042 /* get next frames from the same RTP packet */
2043 if (rt->cur_transport_priv) {
2044 if (rt->transport == RTSP_TRANSPORT_RDT) {
2045 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2046 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2047 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2048 } else if (CONFIG_RTPDEC && rt->ts) {
2049 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2051 rt->recvbuf_pos += ret;
2052 ret = rt->recvbuf_pos < rt->recvbuf_len;
2057 rt->cur_transport_priv = NULL;
2059 } else if (ret == 1) {
2062 rt->cur_transport_priv = NULL;
2066 if (rt->transport == RTSP_TRANSPORT_RTP) {
2068 int64_t first_queue_time = 0;
2069 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2070 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2074 queue_time = ff_rtp_queued_packet_time(rtpctx);
2075 if (queue_time && (queue_time - first_queue_time < 0 ||
2076 !first_queue_time)) {
2077 first_queue_time = queue_time;
2078 first_queue_st = rt->rtsp_streams[i];
2081 if (first_queue_time) {
2082 wait_end = first_queue_time + s->max_delay;
2085 first_queue_st = NULL;
2089 /* read next RTP packet */
2091 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2093 return AVERROR(ENOMEM);
2096 switch(rt->lower_transport) {
2098 #if CONFIG_RTSP_DEMUXER
2099 case RTSP_LOWER_TRANSPORT_TCP:
2100 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2103 case RTSP_LOWER_TRANSPORT_UDP:
2104 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2105 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2106 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2107 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2109 case RTSP_LOWER_TRANSPORT_CUSTOM:
2110 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2111 wait_end && wait_end < av_gettime_relative())
2112 len = AVERROR(EAGAIN);
2114 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2115 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2116 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2117 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2120 if (len == AVERROR(EAGAIN) && first_queue_st &&
2121 rt->transport == RTSP_TRANSPORT_RTP) {
2122 av_log(s, AV_LOG_WARNING,
2123 "max delay reached. need to consume packet\n");
2124 rtsp_st = first_queue_st;
2125 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2132 if (rt->transport == RTSP_TRANSPORT_RDT) {
2133 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2134 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2135 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2136 if (rtsp_st->feedback) {
2137 AVIOContext *pb = NULL;
2138 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2140 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2143 /* Either bad packet, or a RTCP packet. Check if the
2144 * first_rtcp_ntp_time field was initialized. */
2145 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2146 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2147 /* first_rtcp_ntp_time has been initialized for this stream,
2148 * copy the same value to all other uninitialized streams,
2149 * in order to map their timestamp origin to the same ntp time
2152 AVStream *st = NULL;
2153 if (rtsp_st->stream_index >= 0)
2154 st = s->streams[rtsp_st->stream_index];
2155 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2156 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2157 AVStream *st2 = NULL;
2158 if (rt->rtsp_streams[i]->stream_index >= 0)
2159 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2160 if (rtpctx2 && st && st2 &&
2161 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2162 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2163 rtpctx2->rtcp_ts_offset = av_rescale_q(
2164 rtpctx->rtcp_ts_offset, st->time_base,
2169 if (ret == -RTCP_BYE) {
2172 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2173 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2175 if (rt->nb_byes == rt->nb_rtsp_streams)
2179 } else if (CONFIG_RTPDEC && rt->ts) {
2180 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2183 rt->recvbuf_len = len;
2184 rt->recvbuf_pos = ret;
2185 rt->cur_transport_priv = rt->ts;
2192 return AVERROR_INVALIDDATA;
2198 /* more packets may follow, so we save the RTP context */
2199 rt->cur_transport_priv = rtsp_st->transport_priv;
2203 #endif /* CONFIG_RTPDEC */
2205 #if CONFIG_SDP_DEMUXER
2206 static int sdp_probe(AVProbeData *p1)
2208 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2210 /* we look for a line beginning "c=IN IP" */
2211 while (p < p_end && *p != '\0') {
2212 if (p + sizeof("c=IN IP") - 1 < p_end &&
2213 av_strstart(p, "c=IN IP", NULL))
2214 return AVPROBE_SCORE_EXTENSION;
2216 while (p < p_end - 1 && *p != '\n') p++;
2225 static void append_source_addrs(char *buf, int size, const char *name,
2226 int count, struct RTSPSource **addrs)
2231 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2232 for (i = 1; i < count; i++)
2233 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2236 static int sdp_read_header(AVFormatContext *s)
2238 RTSPState *rt = s->priv_data;
2239 RTSPStream *rtsp_st;
2244 if (!ff_network_init())
2245 return AVERROR(EIO);
2247 if (s->max_delay < 0) /* Not set by the caller */
2248 s->max_delay = DEFAULT_REORDERING_DELAY;
2249 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2250 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2252 /* read the whole sdp file */
2253 /* XXX: better loading */
2254 content = av_malloc(SDP_MAX_SIZE);
2256 return AVERROR(ENOMEM);
2257 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2260 return AVERROR_INVALIDDATA;
2262 content[size] ='\0';
2264 err = ff_sdp_parse(s, content);
2268 /* open each RTP stream */
2269 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2271 rtsp_st = rt->rtsp_streams[i];
2273 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2274 AVDictionary *opts = map_to_opts(rt);
2276 err = getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip,
2277 sizeof(rtsp_st->sdp_ip),
2278 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2280 av_log(s, AV_LOG_ERROR, "getnameinfo: %s\n", gai_strerror(err));
2282 av_dict_free(&opts);
2285 ff_url_join(url, sizeof(url), "rtp", NULL,
2286 namebuf, rtsp_st->sdp_port,
2287 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2288 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2289 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2290 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2292 append_source_addrs(url, sizeof(url), "sources",
2293 rtsp_st->nb_include_source_addrs,
2294 rtsp_st->include_source_addrs);
2295 append_source_addrs(url, sizeof(url), "block",
2296 rtsp_st->nb_exclude_source_addrs,
2297 rtsp_st->exclude_source_addrs);
2298 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2299 &s->interrupt_callback, &opts);
2301 av_dict_free(&opts);
2304 err = AVERROR_INVALIDDATA;
2308 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2313 ff_rtsp_close_streams(s);
2318 static int sdp_read_close(AVFormatContext *s)
2320 ff_rtsp_close_streams(s);
2325 static const AVClass sdp_demuxer_class = {
2326 .class_name = "SDP demuxer",
2327 .item_name = av_default_item_name,
2328 .option = sdp_options,
2329 .version = LIBAVUTIL_VERSION_INT,
2332 AVInputFormat ff_sdp_demuxer = {
2334 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2335 .priv_data_size = sizeof(RTSPState),
2336 .read_probe = sdp_probe,
2337 .read_header = sdp_read_header,
2338 .read_packet = ff_rtsp_fetch_packet,
2339 .read_close = sdp_read_close,
2340 .priv_class = &sdp_demuxer_class,
2342 #endif /* CONFIG_SDP_DEMUXER */
2344 #if CONFIG_RTP_DEMUXER
2345 static int rtp_probe(AVProbeData *p)
2347 if (av_strstart(p->filename, "rtp:", NULL))
2348 return AVPROBE_SCORE_MAX;
2352 static int rtp_read_header(AVFormatContext *s)
2354 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2355 char host[500], sdp[500];
2357 URLContext* in = NULL;
2359 AVCodecContext codec = { 0 };
2360 struct sockaddr_storage addr;
2362 socklen_t addrlen = sizeof(addr);
2363 RTSPState *rt = s->priv_data;
2365 if (!ff_network_init())
2366 return AVERROR(EIO);
2368 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2369 &s->interrupt_callback, NULL);
2374 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2375 if (ret == AVERROR(EAGAIN))
2380 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2384 if ((recvbuf[0] & 0xc0) != 0x80) {
2385 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2390 if (RTP_PT_IS_RTCP(recvbuf[1]))
2393 payload_type = recvbuf[1] & 0x7f;
2396 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2400 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2401 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2402 "without an SDP file describing it\n",
2406 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2407 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2408 "properly you need an SDP file "
2412 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2413 NULL, 0, s->filename);
2415 snprintf(sdp, sizeof(sdp),
2416 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2417 addr.ss_family == AF_INET ? 4 : 6, host,
2418 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2419 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2420 port, payload_type);
2421 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2423 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2426 /* sdp_read_header initializes this again */
2429 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2431 ret = sdp_read_header(s);
2442 static const AVClass rtp_demuxer_class = {
2443 .class_name = "RTP demuxer",
2444 .item_name = av_default_item_name,
2445 .option = rtp_options,
2446 .version = LIBAVUTIL_VERSION_INT,
2449 AVInputFormat ff_rtp_demuxer = {
2451 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2452 .priv_data_size = sizeof(RTSPState),
2453 .read_probe = rtp_probe,
2454 .read_header = rtp_read_header,
2455 .read_packet = ff_rtsp_fetch_packet,
2456 .read_close = sdp_read_close,
2457 .flags = AVFMT_NOFILE,
2458 .priv_class = &rtp_demuxer_class,
2460 #endif /* CONFIG_RTP_DEMUXER */