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 RTSP_REORDERING_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 }
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "do not 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", "set 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", "set RTSP flags"),
89 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
90 { "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" },
91 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
92 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
93 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
94 { "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 },
95 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
96 RTSP_REORDERING_OPTS(),
97 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
101 static const AVOption sdp_options[] = {
102 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
103 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
104 { "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" },
105 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
106 RTSP_REORDERING_OPTS(),
110 static const AVOption rtp_options[] = {
111 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
112 RTSP_REORDERING_OPTS(),
116 static void get_word_until_chars(char *buf, int buf_size,
117 const char *sep, const char **pp)
123 p += strspn(p, SPACE_CHARS);
125 while (!strchr(sep, *p) && *p != '\0') {
126 if ((q - buf) < buf_size - 1)
135 static void get_word_sep(char *buf, int buf_size, const char *sep,
138 if (**pp == '/') (*pp)++;
139 get_word_until_chars(buf, buf_size, sep, pp);
142 static void get_word(char *buf, int buf_size, const char **pp)
144 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
147 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
149 * Used for seeking in the rtp stream.
151 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
155 p += strspn(p, SPACE_CHARS);
156 if (!av_stristart(p, "npt=", &p))
159 *start = AV_NOPTS_VALUE;
160 *end = AV_NOPTS_VALUE;
162 get_word_sep(buf, sizeof(buf), "-", &p);
163 av_parse_time(start, buf, 1);
166 get_word_sep(buf, sizeof(buf), "-", &p);
167 av_parse_time(end, buf, 1);
171 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
173 struct addrinfo hints = { 0 }, *ai = NULL;
174 hints.ai_flags = AI_NUMERICHOST;
175 if (getaddrinfo(buf, NULL, &hints, &ai))
177 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
183 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
184 RTSPStream *rtsp_st, AVCodecContext *codec)
189 codec->codec_id = handler->codec_id;
190 rtsp_st->dynamic_handler = handler;
191 if (handler->alloc) {
192 rtsp_st->dynamic_protocol_context = handler->alloc();
193 if (!rtsp_st->dynamic_protocol_context)
194 rtsp_st->dynamic_handler = NULL;
198 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
199 static int sdp_parse_rtpmap(AVFormatContext *s,
200 AVStream *st, RTSPStream *rtsp_st,
201 int payload_type, const char *p)
203 AVCodecContext *codec = st->codec;
209 /* See if we can handle this kind of payload.
210 * The space should normally not be there but some Real streams or
211 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
212 * have a trailing space. */
213 get_word_sep(buf, sizeof(buf), "/ ", &p);
214 if (payload_type < RTP_PT_PRIVATE) {
215 /* We are in a standard case
216 * (from http://www.iana.org/assignments/rtp-parameters). */
217 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
220 if (codec->codec_id == AV_CODEC_ID_NONE) {
221 RTPDynamicProtocolHandler *handler =
222 ff_rtp_handler_find_by_name(buf, codec->codec_type);
223 init_rtp_handler(handler, rtsp_st, codec);
224 /* If no dynamic handler was found, check with the list of standard
225 * allocated types, if such a stream for some reason happens to
226 * use a private payload type. This isn't handled in rtpdec.c, since
227 * the format name from the rtpmap line never is passed into rtpdec. */
228 if (!rtsp_st->dynamic_handler)
229 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
232 c = avcodec_find_decoder(codec->codec_id);
238 get_word_sep(buf, sizeof(buf), "/", &p);
240 switch (codec->codec_type) {
241 case AVMEDIA_TYPE_AUDIO:
242 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
243 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
244 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
246 codec->sample_rate = i;
247 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
248 get_word_sep(buf, sizeof(buf), "/", &p);
253 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
255 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
258 case AVMEDIA_TYPE_VIDEO:
259 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
261 avpriv_set_pts_info(st, 32, 1, i);
266 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
267 rtsp_st->dynamic_handler->init(s, st->index,
268 rtsp_st->dynamic_protocol_context);
272 /* parse the attribute line from the fmtp a line of an sdp response. This
273 * is broken out as a function because it is used in rtp_h264.c, which is
275 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
276 char *value, int value_size)
278 *p += strspn(*p, SPACE_CHARS);
280 get_word_sep(attr, attr_size, "=", p);
283 get_word_sep(value, value_size, ";", p);
291 typedef struct SDPParseState {
293 struct sockaddr_storage default_ip;
295 int skip_media; ///< set if an unknown m= line occurs
296 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
297 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
298 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
299 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
302 char delayed_fmtp[2048];
305 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
306 struct RTSPSource ***dest, int *dest_count)
308 RTSPSource *rtsp_src, *rtsp_src2;
310 for (i = 0; i < count; i++) {
312 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
315 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
316 dynarray_add(dest, dest_count, rtsp_src2);
320 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
321 int payload_type, const char *line)
325 for (i = 0; i < rt->nb_rtsp_streams; i++) {
326 RTSPStream *rtsp_st = rt->rtsp_streams[i];
327 if (rtsp_st->sdp_payload_type == payload_type &&
328 rtsp_st->dynamic_handler &&
329 rtsp_st->dynamic_handler->parse_sdp_a_line) {
330 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
331 rtsp_st->dynamic_protocol_context, line);
336 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
337 int letter, const char *buf)
339 RTSPState *rt = s->priv_data;
340 char buf1[64], st_type[64];
342 enum AVMediaType codec_type;
346 RTSPSource *rtsp_src;
347 struct sockaddr_storage sdp_ip;
350 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
353 if (s1->skip_media && letter != 'm')
357 get_word(buf1, sizeof(buf1), &p);
358 if (strcmp(buf1, "IN") != 0)
360 get_word(buf1, sizeof(buf1), &p);
361 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
363 get_word_sep(buf1, sizeof(buf1), "/", &p);
364 if (get_sockaddr(buf1, &sdp_ip))
369 get_word_sep(buf1, sizeof(buf1), "/", &p);
372 if (s->nb_streams == 0) {
373 s1->default_ip = sdp_ip;
374 s1->default_ttl = ttl;
376 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
377 rtsp_st->sdp_ip = sdp_ip;
378 rtsp_st->sdp_ttl = ttl;
382 av_dict_set(&s->metadata, "title", p, 0);
385 if (s->nb_streams == 0) {
386 av_dict_set(&s->metadata, "comment", p, 0);
395 codec_type = AVMEDIA_TYPE_UNKNOWN;
396 get_word(st_type, sizeof(st_type), &p);
397 if (!strcmp(st_type, "audio")) {
398 codec_type = AVMEDIA_TYPE_AUDIO;
399 } else if (!strcmp(st_type, "video")) {
400 codec_type = AVMEDIA_TYPE_VIDEO;
401 } else if (!strcmp(st_type, "application")) {
402 codec_type = AVMEDIA_TYPE_DATA;
403 } else if (!strcmp(st_type, "text")) {
404 codec_type = AVMEDIA_TYPE_SUBTITLE;
406 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
410 rtsp_st = av_mallocz(sizeof(RTSPStream));
413 rtsp_st->stream_index = -1;
414 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
416 rtsp_st->sdp_ip = s1->default_ip;
417 rtsp_st->sdp_ttl = s1->default_ttl;
419 copy_default_source_addrs(s1->default_include_source_addrs,
420 s1->nb_default_include_source_addrs,
421 &rtsp_st->include_source_addrs,
422 &rtsp_st->nb_include_source_addrs);
423 copy_default_source_addrs(s1->default_exclude_source_addrs,
424 s1->nb_default_exclude_source_addrs,
425 &rtsp_st->exclude_source_addrs,
426 &rtsp_st->nb_exclude_source_addrs);
428 get_word(buf1, sizeof(buf1), &p); /* port */
429 rtsp_st->sdp_port = atoi(buf1);
431 get_word(buf1, sizeof(buf1), &p); /* protocol */
432 if (!strcmp(buf1, "udp"))
433 rt->transport = RTSP_TRANSPORT_RAW;
434 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
435 rtsp_st->feedback = 1;
437 /* XXX: handle list of formats */
438 get_word(buf1, sizeof(buf1), &p); /* format list */
439 rtsp_st->sdp_payload_type = atoi(buf1);
441 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
442 /* no corresponding stream */
443 if (rt->transport == RTSP_TRANSPORT_RAW) {
444 if (CONFIG_RTPDEC && !rt->ts)
445 rt->ts = avpriv_mpegts_parse_open(s);
447 RTPDynamicProtocolHandler *handler;
448 handler = ff_rtp_handler_find_by_id(
449 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
450 init_rtp_handler(handler, rtsp_st, NULL);
451 if (handler && handler->init)
452 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
454 } else if (rt->server_type == RTSP_SERVER_WMS &&
455 codec_type == AVMEDIA_TYPE_DATA) {
456 /* RTX stream, a stream that carries all the other actual
457 * audio/video streams. Don't expose this to the callers. */
459 st = avformat_new_stream(s, NULL);
462 st->id = rt->nb_rtsp_streams - 1;
463 rtsp_st->stream_index = st->index;
464 st->codec->codec_type = codec_type;
465 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
466 RTPDynamicProtocolHandler *handler;
467 /* if standard payload type, we can find the codec right now */
468 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
469 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
470 st->codec->sample_rate > 0)
471 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
472 /* Even static payload types may need a custom depacketizer */
473 handler = ff_rtp_handler_find_by_id(
474 rtsp_st->sdp_payload_type, st->codec->codec_type);
475 init_rtp_handler(handler, rtsp_st, st->codec);
476 if (handler && handler->init)
477 handler->init(s, st->index,
478 rtsp_st->dynamic_protocol_context);
480 if (rt->default_lang[0])
481 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
483 /* put a default control url */
484 av_strlcpy(rtsp_st->control_url, rt->control_uri,
485 sizeof(rtsp_st->control_url));
488 if (av_strstart(p, "control:", &p)) {
489 if (s->nb_streams == 0) {
490 if (!strncmp(p, "rtsp://", 7))
491 av_strlcpy(rt->control_uri, p,
492 sizeof(rt->control_uri));
495 /* get the control url */
496 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
498 /* XXX: may need to add full url resolution */
499 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
501 if (proto[0] == '\0') {
502 /* relative control URL */
503 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
504 av_strlcat(rtsp_st->control_url, "/",
505 sizeof(rtsp_st->control_url));
506 av_strlcat(rtsp_st->control_url, p,
507 sizeof(rtsp_st->control_url));
509 av_strlcpy(rtsp_st->control_url, p,
510 sizeof(rtsp_st->control_url));
512 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
513 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
514 get_word(buf1, sizeof(buf1), &p);
515 payload_type = atoi(buf1);
516 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
517 if (rtsp_st->stream_index >= 0) {
518 st = s->streams[rtsp_st->stream_index];
519 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
523 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
525 } else if (av_strstart(p, "fmtp:", &p) ||
526 av_strstart(p, "framesize:", &p)) {
527 // let dynamic protocol handlers have a stab at the line.
528 get_word(buf1, sizeof(buf1), &p);
529 payload_type = atoi(buf1);
530 if (s1->seen_rtpmap) {
531 parse_fmtp(s, rt, payload_type, buf);
534 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
536 } else if (av_strstart(p, "range:", &p)) {
539 // this is so that seeking on a streamed file can work.
540 rtsp_parse_range_npt(p, &start, &end);
541 s->start_time = start;
542 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
543 s->duration = (end == AV_NOPTS_VALUE) ?
544 AV_NOPTS_VALUE : end - start;
545 } else if (av_strstart(p, "lang:", &p)) {
546 if (s->nb_streams > 0) {
547 get_word(buf1, sizeof(buf1), &p);
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 av_dict_set(&st->metadata, "language", buf1, 0);
554 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
555 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
557 rt->transport = RTSP_TRANSPORT_RDT;
558 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
560 st = s->streams[s->nb_streams - 1];
561 st->codec->sample_rate = atoi(p);
562 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
564 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
565 get_word(buf1, sizeof(buf1), &p); // ignore tag
566 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
567 p += strspn(p, SPACE_CHARS);
568 if (av_strstart(p, "inline:", &p))
569 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
570 } else if (av_strstart(p, "source-filter:", &p)) {
572 get_word(buf1, sizeof(buf1), &p);
573 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
575 exclude = !strcmp(buf1, "excl");
577 get_word(buf1, sizeof(buf1), &p);
578 if (strcmp(buf1, "IN") != 0)
580 get_word(buf1, sizeof(buf1), &p);
581 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
583 // not checking that the destination address actually matches or is wildcard
584 get_word(buf1, sizeof(buf1), &p);
587 rtsp_src = av_mallocz(sizeof(*rtsp_src));
590 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
592 if (s->nb_streams == 0) {
593 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
595 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
596 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
599 if (s->nb_streams == 0) {
600 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
602 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
603 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
608 if (rt->server_type == RTSP_SERVER_WMS)
609 ff_wms_parse_sdp_a_line(s, p);
610 if (s->nb_streams > 0) {
611 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
613 if (rt->server_type == RTSP_SERVER_REAL)
614 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
616 if (rtsp_st->dynamic_handler &&
617 rtsp_st->dynamic_handler->parse_sdp_a_line)
618 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
619 rtsp_st->stream_index,
620 rtsp_st->dynamic_protocol_context, buf);
627 int ff_sdp_parse(AVFormatContext *s, const char *content)
629 RTSPState *rt = s->priv_data;
632 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
633 * contain long SDP lines containing complete ASF Headers (several
634 * kB) or arrays of MDPR (RM stream descriptor) headers plus
635 * "rulebooks" describing their properties. Therefore, the SDP line
638 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
639 * in rtpdec_xiph.c. */
641 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
645 p += strspn(p, SPACE_CHARS);
653 /* get the content */
655 while (*p != '\n' && *p != '\r' && *p != '\0') {
656 if ((q - buf) < sizeof(buf) - 1)
661 sdp_parse_line(s, s1, letter, buf);
663 while (*p != '\n' && *p != '\0')
669 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
670 av_freep(&s1->default_include_source_addrs[i]);
671 av_freep(&s1->default_include_source_addrs);
672 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
673 av_freep(&s1->default_exclude_source_addrs[i]);
674 av_freep(&s1->default_exclude_source_addrs);
676 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
677 if (!rt->p) return AVERROR(ENOMEM);
680 #endif /* CONFIG_RTPDEC */
682 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
684 RTSPState *rt = s->priv_data;
687 for (i = 0; i < rt->nb_rtsp_streams; i++) {
688 RTSPStream *rtsp_st = rt->rtsp_streams[i];
691 if (rtsp_st->transport_priv) {
693 AVFormatContext *rtpctx = rtsp_st->transport_priv;
694 av_write_trailer(rtpctx);
695 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
697 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
698 ff_rtsp_tcp_write_packet(s, rtsp_st);
699 avio_close_dyn_buf(rtpctx->pb, &ptr);
702 avio_closep(&rtpctx->pb);
704 avformat_free_context(rtpctx);
705 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
706 ff_rdt_parse_close(rtsp_st->transport_priv);
707 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
708 ff_rtp_parse_close(rtsp_st->transport_priv);
710 rtsp_st->transport_priv = NULL;
711 if (rtsp_st->rtp_handle)
712 ffurl_close(rtsp_st->rtp_handle);
713 rtsp_st->rtp_handle = NULL;
717 /* close and free RTSP streams */
718 void ff_rtsp_close_streams(AVFormatContext *s)
720 RTSPState *rt = s->priv_data;
724 ff_rtsp_undo_setup(s, 0);
725 for (i = 0; i < rt->nb_rtsp_streams; i++) {
726 rtsp_st = rt->rtsp_streams[i];
728 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
729 rtsp_st->dynamic_handler->free(
730 rtsp_st->dynamic_protocol_context);
731 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
732 av_freep(&rtsp_st->include_source_addrs[j]);
733 av_freep(&rtsp_st->include_source_addrs);
734 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
735 av_freep(&rtsp_st->exclude_source_addrs[j]);
736 av_freep(&rtsp_st->exclude_source_addrs);
741 av_freep(&rt->rtsp_streams);
743 avformat_close_input(&rt->asf_ctx);
745 if (CONFIG_RTPDEC && rt->ts)
746 avpriv_mpegts_parse_close(rt->ts);
748 av_freep(&rt->recvbuf);
751 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
753 RTSPState *rt = s->priv_data;
755 int reordering_queue_size = rt->reordering_queue_size;
756 if (reordering_queue_size < 0) {
757 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
758 reordering_queue_size = 0;
760 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
763 /* open the RTP context */
764 if (rtsp_st->stream_index >= 0)
765 st = s->streams[rtsp_st->stream_index];
767 s->ctx_flags |= AVFMTCTX_NOHEADER;
769 if (CONFIG_RTSP_MUXER && s->oformat) {
770 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
771 s, st, rtsp_st->rtp_handle,
772 RTSP_TCP_MAX_PACKET_SIZE,
773 rtsp_st->stream_index);
774 /* Ownership of rtp_handle is passed to the rtp mux context */
775 rtsp_st->rtp_handle = NULL;
778 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
779 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
780 return 0; // Don't need to open any parser here
781 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
782 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
783 rtsp_st->dynamic_protocol_context,
784 rtsp_st->dynamic_handler);
785 else if (CONFIG_RTPDEC)
786 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
787 rtsp_st->sdp_payload_type,
788 reordering_queue_size);
790 if (!rtsp_st->transport_priv) {
791 return AVERROR(ENOMEM);
792 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
793 if (rtsp_st->dynamic_handler) {
794 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
795 rtsp_st->dynamic_protocol_context,
796 rtsp_st->dynamic_handler);
798 if (rtsp_st->crypto_suite[0])
799 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
800 rtsp_st->crypto_suite,
801 rtsp_st->crypto_params);
807 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
808 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
815 q += strspn(q, SPACE_CHARS);
816 v = strtol(q, &p, 10);
820 v = strtol(p, &p, 10);
829 /* XXX: only one transport specification is parsed */
830 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
832 char transport_protocol[16];
834 char lower_transport[16];
836 RTSPTransportField *th;
839 reply->nb_transports = 0;
842 p += strspn(p, SPACE_CHARS);
846 th = &reply->transports[reply->nb_transports];
848 get_word_sep(transport_protocol, sizeof(transport_protocol),
850 if (!av_strcasecmp (transport_protocol, "rtp")) {
851 get_word_sep(profile, sizeof(profile), "/;,", &p);
852 lower_transport[0] = '\0';
853 /* rtp/avp/<protocol> */
855 get_word_sep(lower_transport, sizeof(lower_transport),
858 th->transport = RTSP_TRANSPORT_RTP;
859 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
860 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
861 /* x-pn-tng/<protocol> */
862 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
864 th->transport = RTSP_TRANSPORT_RDT;
865 } else if (!av_strcasecmp(transport_protocol, "raw")) {
866 get_word_sep(profile, sizeof(profile), "/;,", &p);
867 lower_transport[0] = '\0';
868 /* raw/raw/<protocol> */
870 get_word_sep(lower_transport, sizeof(lower_transport),
873 th->transport = RTSP_TRANSPORT_RAW;
875 if (!av_strcasecmp(lower_transport, "TCP"))
876 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
878 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
882 /* get each parameter */
883 while (*p != '\0' && *p != ',') {
884 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
885 if (!strcmp(parameter, "port")) {
888 rtsp_parse_range(&th->port_min, &th->port_max, &p);
890 } else if (!strcmp(parameter, "client_port")) {
893 rtsp_parse_range(&th->client_port_min,
894 &th->client_port_max, &p);
896 } else if (!strcmp(parameter, "server_port")) {
899 rtsp_parse_range(&th->server_port_min,
900 &th->server_port_max, &p);
902 } else if (!strcmp(parameter, "interleaved")) {
905 rtsp_parse_range(&th->interleaved_min,
906 &th->interleaved_max, &p);
908 } else if (!strcmp(parameter, "multicast")) {
909 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
910 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
911 } else if (!strcmp(parameter, "ttl")) {
915 th->ttl = strtol(p, &end, 10);
918 } else if (!strcmp(parameter, "destination")) {
921 get_word_sep(buf, sizeof(buf), ";,", &p);
922 get_sockaddr(buf, &th->destination);
924 } else if (!strcmp(parameter, "source")) {
927 get_word_sep(buf, sizeof(buf), ";,", &p);
928 av_strlcpy(th->source, buf, sizeof(th->source));
930 } else if (!strcmp(parameter, "mode")) {
933 get_word_sep(buf, sizeof(buf), ";, ", &p);
934 if (!strcmp(buf, "record") ||
935 !strcmp(buf, "receive"))
940 while (*p != ';' && *p != '\0' && *p != ',')
948 reply->nb_transports++;
952 static void handle_rtp_info(RTSPState *rt, const char *url,
953 uint32_t seq, uint32_t rtptime)
956 if (!rtptime || !url[0])
958 if (rt->transport != RTSP_TRANSPORT_RTP)
960 for (i = 0; i < rt->nb_rtsp_streams; i++) {
961 RTSPStream *rtsp_st = rt->rtsp_streams[i];
962 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
965 if (!strcmp(rtsp_st->control_url, url)) {
966 rtpctx->base_timestamp = rtptime;
972 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
975 char key[20], value[1024], url[1024] = "";
976 uint32_t seq = 0, rtptime = 0;
979 p += strspn(p, SPACE_CHARS);
982 get_word_sep(key, sizeof(key), "=", &p);
986 get_word_sep(value, sizeof(value), ";, ", &p);
988 if (!strcmp(key, "url"))
989 av_strlcpy(url, value, sizeof(url));
990 else if (!strcmp(key, "seq"))
991 seq = strtoul(value, NULL, 10);
992 else if (!strcmp(key, "rtptime"))
993 rtptime = strtoul(value, NULL, 10);
995 handle_rtp_info(rt, url, seq, rtptime);
1004 handle_rtp_info(rt, url, seq, rtptime);
1007 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1008 RTSPState *rt, const char *method)
1012 /* NOTE: we do case independent match for broken servers */
1014 if (av_stristart(p, "Session:", &p)) {
1016 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1017 if (av_stristart(p, ";timeout=", &p) &&
1018 (t = strtol(p, NULL, 10)) > 0) {
1021 } else if (av_stristart(p, "Content-Length:", &p)) {
1022 reply->content_length = strtol(p, NULL, 10);
1023 } else if (av_stristart(p, "Transport:", &p)) {
1024 rtsp_parse_transport(reply, p);
1025 } else if (av_stristart(p, "CSeq:", &p)) {
1026 reply->seq = strtol(p, NULL, 10);
1027 } else if (av_stristart(p, "Range:", &p)) {
1028 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1029 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1030 p += strspn(p, SPACE_CHARS);
1031 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1032 } else if (av_stristart(p, "Server:", &p)) {
1033 p += strspn(p, SPACE_CHARS);
1034 av_strlcpy(reply->server, p, sizeof(reply->server));
1035 } else if (av_stristart(p, "Notice:", &p) ||
1036 av_stristart(p, "X-Notice:", &p)) {
1037 reply->notice = strtol(p, NULL, 10);
1038 } else if (av_stristart(p, "Location:", &p)) {
1039 p += strspn(p, SPACE_CHARS);
1040 av_strlcpy(reply->location, p , sizeof(reply->location));
1041 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1042 p += strspn(p, SPACE_CHARS);
1043 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1044 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1045 p += strspn(p, SPACE_CHARS);
1046 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1047 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1048 p += strspn(p, SPACE_CHARS);
1049 if (method && !strcmp(method, "DESCRIBE"))
1050 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1051 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1052 p += strspn(p, SPACE_CHARS);
1053 if (method && !strcmp(method, "PLAY"))
1054 rtsp_parse_rtp_info(rt, p);
1055 } else if (av_stristart(p, "Public:", &p) && rt) {
1056 if (strstr(p, "GET_PARAMETER") &&
1057 method && !strcmp(method, "OPTIONS"))
1058 rt->get_parameter_supported = 1;
1059 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1060 p += strspn(p, SPACE_CHARS);
1061 rt->accept_dynamic_rate = atoi(p);
1062 } else if (av_stristart(p, "Content-Type:", &p)) {
1063 p += strspn(p, SPACE_CHARS);
1064 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1068 /* skip a RTP/TCP interleaved packet */
1069 void ff_rtsp_skip_packet(AVFormatContext *s)
1071 RTSPState *rt = s->priv_data;
1075 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1078 len = AV_RB16(buf + 1);
1080 av_dlog(s, "skipping RTP packet len=%d\n", len);
1085 if (len1 > sizeof(buf))
1087 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1094 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1095 unsigned char **content_ptr,
1096 int return_on_interleaved_data, const char *method)
1098 RTSPState *rt = s->priv_data;
1099 char buf[4096], buf1[1024], *q;
1102 int ret, content_length, line_count = 0, request = 0;
1103 unsigned char *content = NULL;
1109 memset(reply, 0, sizeof(*reply));
1111 /* parse reply (XXX: use buffers) */
1112 rt->last_reply[0] = '\0';
1116 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1117 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1123 /* XXX: only parse it if first char on line ? */
1124 if (return_on_interleaved_data) {
1127 ff_rtsp_skip_packet(s);
1128 } else if (ch != '\r') {
1129 if ((q - buf) < sizeof(buf) - 1)
1135 av_dlog(s, "line='%s'\n", buf);
1137 /* test if last line */
1141 if (line_count == 0) {
1142 /* get reply code */
1143 get_word(buf1, sizeof(buf1), &p);
1144 if (!strncmp(buf1, "RTSP/", 5)) {
1145 get_word(buf1, sizeof(buf1), &p);
1146 reply->status_code = atoi(buf1);
1147 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1149 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1150 get_word(buf1, sizeof(buf1), &p); // object
1154 ff_rtsp_parse_line(reply, p, rt, method);
1155 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1156 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1161 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1162 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1164 content_length = reply->content_length;
1165 if (content_length > 0) {
1166 /* leave some room for a trailing '\0' (useful for simple parsing) */
1167 content = av_malloc(content_length + 1);
1169 return AVERROR(ENOMEM);
1170 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1171 content[content_length] = '\0';
1174 *content_ptr = content;
1180 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1181 const char* ptr = buf;
1183 if (!strcmp(reply->reason, "OPTIONS")) {
1184 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1186 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1187 if (reply->session_id[0])
1188 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1191 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1193 av_strlcat(buf, "\r\n", sizeof(buf));
1195 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1196 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1199 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1201 rt->last_cmd_time = av_gettime_relative();
1202 /* Even if the request from the server had data, it is not the data
1203 * that the caller wants or expects. The memory could also be leaked
1204 * if the actual following reply has content data. */
1206 av_freep(content_ptr);
1207 /* If method is set, this is called from ff_rtsp_send_cmd,
1208 * where a reply to exactly this request is awaited. For
1209 * callers from within packet receiving, we just want to
1210 * return to the caller and go back to receiving packets. */
1216 if (rt->seq != reply->seq) {
1217 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1218 rt->seq, reply->seq);
1222 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1223 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1224 reply->notice == 2306 /* Continuous Feed Terminated */) {
1225 rt->state = RTSP_STATE_IDLE;
1226 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1227 return AVERROR(EIO); /* data or server error */
1228 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1229 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1230 return AVERROR(EPERM);
1236 * Send a command to the RTSP server without waiting for the reply.
1238 * @param s RTSP (de)muxer context
1239 * @param method the method for the request
1240 * @param url the target url for the request
1241 * @param headers extra header lines to include in the request
1242 * @param send_content if non-null, the data to send as request body content
1243 * @param send_content_length the length of the send_content data, or 0 if
1244 * send_content is null
1246 * @return zero if success, nonzero otherwise
1248 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1249 const char *method, const char *url,
1250 const char *headers,
1251 const unsigned char *send_content,
1252 int send_content_length)
1254 RTSPState *rt = s->priv_data;
1255 char buf[4096], *out_buf;
1256 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1258 /* Add in RTSP headers */
1261 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1263 av_strlcat(buf, headers, sizeof(buf));
1264 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1265 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1266 if (rt->session_id[0] != '\0' && (!headers ||
1267 !strstr(headers, "\nIf-Match:"))) {
1268 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1271 char *str = ff_http_auth_create_response(&rt->auth_state,
1272 rt->auth, url, method);
1274 av_strlcat(buf, str, sizeof(buf));
1277 if (send_content_length > 0 && send_content)
1278 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1279 av_strlcat(buf, "\r\n", sizeof(buf));
1281 /* base64 encode rtsp if tunneling */
1282 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1283 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1284 out_buf = base64buf;
1287 av_dlog(s, "Sending:\n%s--\n", buf);
1289 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1290 if (send_content_length > 0 && send_content) {
1291 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1292 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1293 "with content data not supported\n");
1294 return AVERROR_PATCHWELCOME;
1296 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1298 rt->last_cmd_time = av_gettime_relative();
1303 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1304 const char *url, const char *headers)
1306 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1309 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1310 const char *headers, RTSPMessageHeader *reply,
1311 unsigned char **content_ptr)
1313 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1314 content_ptr, NULL, 0);
1317 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1318 const char *method, const char *url,
1320 RTSPMessageHeader *reply,
1321 unsigned char **content_ptr,
1322 const unsigned char *send_content,
1323 int send_content_length)
1325 RTSPState *rt = s->priv_data;
1326 HTTPAuthType cur_auth_type;
1327 int ret, attempts = 0;
1330 cur_auth_type = rt->auth_state.auth_type;
1331 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1333 send_content_length)))
1336 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1340 if (reply->status_code == 401 &&
1341 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1342 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1345 if (reply->status_code > 400){
1346 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1350 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1356 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1357 int lower_transport, const char *real_challenge)
1359 RTSPState *rt = s->priv_data;
1360 int rtx = 0, j, i, err, interleave = 0, port_off;
1361 RTSPStream *rtsp_st;
1362 RTSPMessageHeader reply1, *reply = &reply1;
1364 const char *trans_pref;
1366 if (rt->transport == RTSP_TRANSPORT_RDT)
1367 trans_pref = "x-pn-tng";
1368 else if (rt->transport == RTSP_TRANSPORT_RAW)
1369 trans_pref = "RAW/RAW";
1371 trans_pref = "RTP/AVP";
1373 /* default timeout: 1 minute */
1376 /* Choose a random starting offset within the first half of the
1377 * port range, to allow for a number of ports to try even if the offset
1378 * happens to be at the end of the random range. */
1379 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1380 /* even random offset */
1381 port_off -= port_off & 0x01;
1383 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1384 char transport[2048];
1387 * WMS serves all UDP data over a single connection, the RTX, which
1388 * isn't necessarily the first in the SDP but has to be the first
1389 * to be set up, else the second/third SETUP will fail with a 461.
1391 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1392 rt->server_type == RTSP_SERVER_WMS) {
1395 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1396 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1398 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1402 if (rtx == rt->nb_rtsp_streams)
1403 return -1; /* no RTX found */
1404 rtsp_st = rt->rtsp_streams[rtx];
1406 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1408 rtsp_st = rt->rtsp_streams[i];
1411 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1414 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1415 port = reply->transports[0].client_port_min;
1419 /* first try in specified port range */
1420 while (j <= rt->rtp_port_max) {
1421 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1422 "?localport=%d", j);
1423 /* we will use two ports per rtp stream (rtp and rtcp) */
1425 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1426 &s->interrupt_callback, NULL))
1429 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1434 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1436 snprintf(transport, sizeof(transport) - 1,
1437 "%s/UDP;", trans_pref);
1438 if (rt->server_type != RTSP_SERVER_REAL)
1439 av_strlcat(transport, "unicast;", sizeof(transport));
1440 av_strlcatf(transport, sizeof(transport),
1441 "client_port=%d", port);
1442 if (rt->transport == RTSP_TRANSPORT_RTP &&
1443 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1444 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1448 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1449 /* For WMS streams, the application streams are only used for
1450 * UDP. When trying to set it up for TCP streams, the server
1451 * will return an error. Therefore, we skip those streams. */
1452 if (rt->server_type == RTSP_SERVER_WMS &&
1453 (rtsp_st->stream_index < 0 ||
1454 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1457 snprintf(transport, sizeof(transport) - 1,
1458 "%s/TCP;", trans_pref);
1459 if (rt->transport != RTSP_TRANSPORT_RDT)
1460 av_strlcat(transport, "unicast;", sizeof(transport));
1461 av_strlcatf(transport, sizeof(transport),
1462 "interleaved=%d-%d",
1463 interleave, interleave + 1);
1467 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1468 snprintf(transport, sizeof(transport) - 1,
1469 "%s/UDP;multicast", trans_pref);
1472 av_strlcat(transport, ";mode=record", sizeof(transport));
1473 } else if (rt->server_type == RTSP_SERVER_REAL ||
1474 rt->server_type == RTSP_SERVER_WMS)
1475 av_strlcat(transport, ";mode=play", sizeof(transport));
1476 snprintf(cmd, sizeof(cmd),
1477 "Transport: %s\r\n",
1479 if (rt->accept_dynamic_rate)
1480 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1481 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1482 char real_res[41], real_csum[9];
1483 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1485 av_strlcatf(cmd, sizeof(cmd),
1487 "RealChallenge2: %s, sd=%s\r\n",
1488 rt->session_id, real_res, real_csum);
1490 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1491 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1494 } else if (reply->status_code != RTSP_STATUS_OK ||
1495 reply->nb_transports != 1) {
1496 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1500 /* XXX: same protocol for all streams is required */
1502 if (reply->transports[0].lower_transport != rt->lower_transport ||
1503 reply->transports[0].transport != rt->transport) {
1504 err = AVERROR_INVALIDDATA;
1508 rt->lower_transport = reply->transports[0].lower_transport;
1509 rt->transport = reply->transports[0].transport;
1512 /* Fail if the server responded with another lower transport mode
1513 * than what we requested. */
1514 if (reply->transports[0].lower_transport != lower_transport) {
1515 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1516 err = AVERROR_INVALIDDATA;
1520 switch(reply->transports[0].lower_transport) {
1521 case RTSP_LOWER_TRANSPORT_TCP:
1522 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1523 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1526 case RTSP_LOWER_TRANSPORT_UDP: {
1527 char url[1024], options[30] = "";
1528 const char *peer = host;
1530 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1531 av_strlcpy(options, "?connect=1", sizeof(options));
1532 /* Use source address if specified */
1533 if (reply->transports[0].source[0])
1534 peer = reply->transports[0].source;
1535 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1536 reply->transports[0].server_port_min, "%s", options);
1537 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1538 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1539 err = AVERROR_INVALIDDATA;
1542 /* Try to initialize the connection state in a
1543 * potential NAT router by sending dummy packets.
1544 * RTP/RTCP dummy packets are used for RDT, too.
1546 if (CONFIG_RTPDEC &&
1547 !(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1548 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1551 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1552 char url[1024], namebuf[50], optbuf[20] = "";
1553 struct sockaddr_storage addr;
1556 if (reply->transports[0].destination.ss_family) {
1557 addr = reply->transports[0].destination;
1558 port = reply->transports[0].port_min;
1559 ttl = reply->transports[0].ttl;
1561 addr = rtsp_st->sdp_ip;
1562 port = rtsp_st->sdp_port;
1563 ttl = rtsp_st->sdp_ttl;
1566 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1567 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1568 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1569 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1570 port, "%s", optbuf);
1571 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1572 &s->interrupt_callback, NULL) < 0) {
1573 err = AVERROR_INVALIDDATA;
1580 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1584 if (rt->nb_rtsp_streams && reply->timeout > 0)
1585 rt->timeout = reply->timeout;
1587 if (rt->server_type == RTSP_SERVER_REAL)
1588 rt->need_subscription = 1;
1593 ff_rtsp_undo_setup(s, 0);
1597 void ff_rtsp_close_connections(AVFormatContext *s)
1599 RTSPState *rt = s->priv_data;
1600 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1601 ffurl_close(rt->rtsp_hd);
1602 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1605 int ff_rtsp_connect(AVFormatContext *s)
1607 RTSPState *rt = s->priv_data;
1608 char proto[128], host[1024], path[1024];
1609 char tcpname[1024], cmd[2048], auth[128];
1610 const char *lower_rtsp_proto = "tcp";
1611 int port, err, tcp_fd;
1612 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1613 int lower_transport_mask = 0;
1614 int default_port = RTSP_DEFAULT_PORT;
1615 char real_challenge[64] = "";
1616 struct sockaddr_storage peer;
1617 socklen_t peer_len = sizeof(peer);
1619 if (rt->rtp_port_max < rt->rtp_port_min) {
1620 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1621 "than min port %d\n", rt->rtp_port_max,
1623 return AVERROR(EINVAL);
1626 if (!ff_network_init())
1627 return AVERROR(EIO);
1629 if (s->max_delay < 0) /* Not set by the caller */
1630 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1632 rt->control_transport = RTSP_MODE_PLAIN;
1633 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1634 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1635 rt->control_transport = RTSP_MODE_TUNNEL;
1637 /* Only pass through valid flags from here */
1638 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1641 /* extract hostname and port */
1642 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1643 host, sizeof(host), &port, path, sizeof(path), s->filename);
1645 if (!strcmp(proto, "rtsps")) {
1646 lower_rtsp_proto = "tls";
1647 default_port = RTSPS_DEFAULT_PORT;
1648 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1652 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1655 port = default_port;
1657 lower_transport_mask = rt->lower_transport_mask;
1659 if (!lower_transport_mask)
1660 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1663 /* Only UDP or TCP - UDP multicast isn't supported. */
1664 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1665 (1 << RTSP_LOWER_TRANSPORT_TCP);
1666 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1667 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1668 "only UDP and TCP are supported for output.\n");
1669 err = AVERROR(EINVAL);
1674 /* Construct the URI used in request; this is similar to s->filename,
1675 * but with authentication credentials removed and RTSP specific options
1677 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1678 host, port, "%s", path);
1680 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1681 /* set up initial handshake for tunneling */
1682 char httpname[1024];
1683 char sessioncookie[17];
1686 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1687 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1688 av_get_random_seed(), av_get_random_seed());
1691 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1692 &s->interrupt_callback) < 0) {
1697 /* generate GET headers */
1698 snprintf(headers, sizeof(headers),
1699 "x-sessioncookie: %s\r\n"
1700 "Accept: application/x-rtsp-tunnelled\r\n"
1701 "Pragma: no-cache\r\n"
1702 "Cache-Control: no-cache\r\n",
1704 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1706 /* complete the connection */
1707 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1713 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1714 &s->interrupt_callback) < 0 ) {
1719 /* generate POST headers */
1720 snprintf(headers, sizeof(headers),
1721 "x-sessioncookie: %s\r\n"
1722 "Content-Type: application/x-rtsp-tunnelled\r\n"
1723 "Pragma: no-cache\r\n"
1724 "Cache-Control: no-cache\r\n"
1725 "Content-Length: 32767\r\n"
1726 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1728 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1729 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1731 /* Initialize the authentication state for the POST session. The HTTP
1732 * protocol implementation doesn't properly handle multi-pass
1733 * authentication for POST requests, since it would require one of
1735 * - implementing Expect: 100-continue, which many HTTP servers
1736 * don't support anyway, even less the RTSP servers that do HTTP
1738 * - sending the whole POST data until getting a 401 reply specifying
1739 * what authentication method to use, then resending all that data
1740 * - waiting for potential 401 replies directly after sending the
1741 * POST header (waiting for some unspecified time)
1742 * Therefore, we copy the full auth state, which works for both basic
1743 * and digest. (For digest, we would have to synchronize the nonce
1744 * count variable between the two sessions, if we'd do more requests
1745 * with the original session, though.)
1747 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1749 /* complete the connection */
1750 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1756 /* open the tcp connection */
1757 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1759 "?timeout=%d", rt->stimeout);
1760 if ((ret = ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1761 &s->interrupt_callback, NULL)) < 0) {
1765 rt->rtsp_hd_out = rt->rtsp_hd;
1769 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1774 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1775 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1776 NULL, 0, NI_NUMERICHOST);
1779 /* request options supported by the server; this also detects server
1781 for (rt->server_type = RTSP_SERVER_RTP;;) {
1783 if (rt->server_type == RTSP_SERVER_REAL)
1786 * The following entries are required for proper
1787 * streaming from a Realmedia server. They are
1788 * interdependent in some way although we currently
1789 * don't quite understand how. Values were copied
1790 * from mplayer SVN r23589.
1791 * ClientChallenge is a 16-byte ID in hex
1792 * CompanyID is a 16-byte ID in base64
1794 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1795 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1796 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1797 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1799 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1800 if (reply->status_code != RTSP_STATUS_OK) {
1801 err = ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
1805 /* detect server type if not standard-compliant RTP */
1806 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1807 rt->server_type = RTSP_SERVER_REAL;
1809 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1810 rt->server_type = RTSP_SERVER_WMS;
1811 } else if (rt->server_type == RTSP_SERVER_REAL)
1812 strcpy(real_challenge, reply->real_challenge);
1816 if (CONFIG_RTSP_DEMUXER && s->iformat)
1817 err = ff_rtsp_setup_input_streams(s, reply);
1818 else if (CONFIG_RTSP_MUXER)
1819 err = ff_rtsp_setup_output_streams(s, host);
1826 int lower_transport = ff_log2_tab[lower_transport_mask &
1827 ~(lower_transport_mask - 1)];
1829 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1830 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1831 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1833 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1834 rt->server_type == RTSP_SERVER_REAL ?
1835 real_challenge : NULL);
1838 lower_transport_mask &= ~(1 << lower_transport);
1839 if (lower_transport_mask == 0 && err == 1) {
1840 err = AVERROR(EPROTONOSUPPORT);
1845 rt->lower_transport_mask = lower_transport_mask;
1846 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1847 rt->state = RTSP_STATE_IDLE;
1848 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1851 ff_rtsp_close_streams(s);
1852 ff_rtsp_close_connections(s);
1853 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1854 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1855 rt->session_id[0] = '\0';
1856 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1864 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1867 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1868 uint8_t *buf, int buf_size, int64_t wait_end)
1870 RTSPState *rt = s->priv_data;
1871 RTSPStream *rtsp_st;
1872 int n, i, ret, tcp_fd, timeout_cnt = 0;
1874 struct pollfd *p = rt->p;
1875 int *fds = NULL, fdsnum, fdsidx;
1878 if (ff_check_interrupt(&s->interrupt_callback))
1879 return AVERROR_EXIT;
1880 if (wait_end && wait_end - av_gettime_relative() < 0)
1881 return AVERROR(EAGAIN);
1884 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1885 p[max_p].fd = tcp_fd;
1886 p[max_p++].events = POLLIN;
1890 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1891 rtsp_st = rt->rtsp_streams[i];
1892 if (rtsp_st->rtp_handle) {
1893 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1895 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1899 av_log(s, AV_LOG_ERROR,
1900 "Number of fds %d not supported\n", fdsnum);
1901 return AVERROR_INVALIDDATA;
1903 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1904 p[max_p].fd = fds[fdsidx];
1905 p[max_p++].events = POLLIN;
1910 n = poll(p, max_p, POLL_TIMEOUT_MS);
1912 int j = 1 - (tcp_fd == -1);
1914 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1915 rtsp_st = rt->rtsp_streams[i];
1916 if (rtsp_st->rtp_handle) {
1917 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1918 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1920 *prtsp_st = rtsp_st;
1927 #if CONFIG_RTSP_DEMUXER
1928 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1929 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1930 if (rt->state == RTSP_STATE_STREAMING) {
1931 if (!ff_rtsp_parse_streaming_commands(s))
1934 av_log(s, AV_LOG_WARNING,
1935 "Unable to answer to TEARDOWN\n");
1939 RTSPMessageHeader reply;
1940 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1943 /* XXX: parse message */
1944 if (rt->state != RTSP_STATE_STREAMING)
1949 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1950 return AVERROR(ETIMEDOUT);
1951 } else if (n < 0 && errno != EINTR)
1952 return AVERROR(errno);
1956 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1957 const uint8_t *buf, int len)
1959 RTSPState *rt = s->priv_data;
1963 if (rt->nb_rtsp_streams == 1) {
1964 *rtsp_st = rt->rtsp_streams[0];
1967 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1968 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1970 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1971 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1974 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1975 *rtsp_st = rt->rtsp_streams[i];
1982 av_log(s, AV_LOG_WARNING,
1983 "Unable to pick stream for packet - SSRC not known for "
1985 return AVERROR(EAGAIN);
1988 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1989 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1990 *rtsp_st = rt->rtsp_streams[i];
1996 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1997 return AVERROR(EAGAIN);
2000 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2002 RTSPState *rt = s->priv_data;
2004 RTSPStream *rtsp_st, *first_queue_st = NULL;
2005 int64_t wait_end = 0;
2007 if (rt->nb_byes == rt->nb_rtsp_streams)
2010 /* get next frames from the same RTP packet */
2011 if (rt->cur_transport_priv) {
2012 if (rt->transport == RTSP_TRANSPORT_RDT) {
2013 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2014 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2015 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2016 } else if (CONFIG_RTPDEC && rt->ts) {
2017 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2019 rt->recvbuf_pos += ret;
2020 ret = rt->recvbuf_pos < rt->recvbuf_len;
2025 rt->cur_transport_priv = NULL;
2027 } else if (ret == 1) {
2030 rt->cur_transport_priv = NULL;
2034 if (rt->transport == RTSP_TRANSPORT_RTP) {
2036 int64_t first_queue_time = 0;
2037 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2038 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2042 queue_time = ff_rtp_queued_packet_time(rtpctx);
2043 if (queue_time && (queue_time - first_queue_time < 0 ||
2044 !first_queue_time)) {
2045 first_queue_time = queue_time;
2046 first_queue_st = rt->rtsp_streams[i];
2049 if (first_queue_time) {
2050 wait_end = first_queue_time + s->max_delay;
2053 first_queue_st = NULL;
2057 /* read next RTP packet */
2059 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2061 return AVERROR(ENOMEM);
2064 switch(rt->lower_transport) {
2066 #if CONFIG_RTSP_DEMUXER
2067 case RTSP_LOWER_TRANSPORT_TCP:
2068 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2071 case RTSP_LOWER_TRANSPORT_UDP:
2072 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2073 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2074 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2075 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2077 case RTSP_LOWER_TRANSPORT_CUSTOM:
2078 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2079 wait_end && wait_end < av_gettime_relative())
2080 len = AVERROR(EAGAIN);
2082 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2083 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2084 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2085 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2088 if (len == AVERROR(EAGAIN) && first_queue_st &&
2089 rt->transport == RTSP_TRANSPORT_RTP) {
2090 rtsp_st = first_queue_st;
2091 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2098 if (rt->transport == RTSP_TRANSPORT_RDT) {
2099 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2100 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2101 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2102 if (rtsp_st->feedback) {
2103 AVIOContext *pb = NULL;
2104 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2106 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2109 /* Either bad packet, or a RTCP packet. Check if the
2110 * first_rtcp_ntp_time field was initialized. */
2111 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2112 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2113 /* first_rtcp_ntp_time has been initialized for this stream,
2114 * copy the same value to all other uninitialized streams,
2115 * in order to map their timestamp origin to the same ntp time
2118 AVStream *st = NULL;
2119 if (rtsp_st->stream_index >= 0)
2120 st = s->streams[rtsp_st->stream_index];
2121 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2122 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2123 AVStream *st2 = NULL;
2124 if (rt->rtsp_streams[i]->stream_index >= 0)
2125 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2126 if (rtpctx2 && st && st2 &&
2127 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2128 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2129 rtpctx2->rtcp_ts_offset = av_rescale_q(
2130 rtpctx->rtcp_ts_offset, st->time_base,
2134 // Make real NTP start time available in AVFormatContext
2135 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2136 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2138 s->start_time_realtime -=
2139 av_rescale (rtpctx->rtcp_ts_offset,
2140 (uint64_t) rtpctx->st->time_base.num * 1000000,
2141 rtpctx->st->time_base.den);
2145 if (ret == -RTCP_BYE) {
2148 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2149 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2151 if (rt->nb_byes == rt->nb_rtsp_streams)
2155 } else if (CONFIG_RTPDEC && rt->ts) {
2156 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2159 rt->recvbuf_len = len;
2160 rt->recvbuf_pos = ret;
2161 rt->cur_transport_priv = rt->ts;
2168 return AVERROR_INVALIDDATA;
2174 /* more packets may follow, so we save the RTP context */
2175 rt->cur_transport_priv = rtsp_st->transport_priv;
2179 #endif /* CONFIG_RTPDEC */
2181 #if CONFIG_SDP_DEMUXER
2182 static int sdp_probe(AVProbeData *p1)
2184 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2186 /* we look for a line beginning "c=IN IP" */
2187 while (p < p_end && *p != '\0') {
2188 if (p + sizeof("c=IN IP") - 1 < p_end &&
2189 av_strstart(p, "c=IN IP", NULL))
2190 return AVPROBE_SCORE_EXTENSION;
2192 while (p < p_end - 1 && *p != '\n') p++;
2201 static void append_source_addrs(char *buf, int size, const char *name,
2202 int count, struct RTSPSource **addrs)
2207 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2208 for (i = 1; i < count; i++)
2209 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2212 static int sdp_read_header(AVFormatContext *s)
2214 RTSPState *rt = s->priv_data;
2215 RTSPStream *rtsp_st;
2220 if (!ff_network_init())
2221 return AVERROR(EIO);
2223 if (s->max_delay < 0) /* Not set by the caller */
2224 s->max_delay = DEFAULT_REORDERING_DELAY;
2225 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2226 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2228 /* read the whole sdp file */
2229 /* XXX: better loading */
2230 content = av_malloc(SDP_MAX_SIZE);
2231 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2234 return AVERROR_INVALIDDATA;
2236 content[size] ='\0';
2238 err = ff_sdp_parse(s, content);
2242 /* open each RTP stream */
2243 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2245 rtsp_st = rt->rtsp_streams[i];
2247 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2248 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2249 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2250 ff_url_join(url, sizeof(url), "rtp", NULL,
2251 namebuf, rtsp_st->sdp_port,
2252 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2253 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2254 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2255 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2257 append_source_addrs(url, sizeof(url), "sources",
2258 rtsp_st->nb_include_source_addrs,
2259 rtsp_st->include_source_addrs);
2260 append_source_addrs(url, sizeof(url), "block",
2261 rtsp_st->nb_exclude_source_addrs,
2262 rtsp_st->exclude_source_addrs);
2263 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2264 &s->interrupt_callback, NULL) < 0) {
2265 err = AVERROR_INVALIDDATA;
2269 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2274 ff_rtsp_close_streams(s);
2279 static int sdp_read_close(AVFormatContext *s)
2281 ff_rtsp_close_streams(s);
2286 static const AVClass sdp_demuxer_class = {
2287 .class_name = "SDP demuxer",
2288 .item_name = av_default_item_name,
2289 .option = sdp_options,
2290 .version = LIBAVUTIL_VERSION_INT,
2293 AVInputFormat ff_sdp_demuxer = {
2295 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2296 .priv_data_size = sizeof(RTSPState),
2297 .read_probe = sdp_probe,
2298 .read_header = sdp_read_header,
2299 .read_packet = ff_rtsp_fetch_packet,
2300 .read_close = sdp_read_close,
2301 .priv_class = &sdp_demuxer_class,
2303 #endif /* CONFIG_SDP_DEMUXER */
2305 #if CONFIG_RTP_DEMUXER
2306 static int rtp_probe(AVProbeData *p)
2308 if (av_strstart(p->filename, "rtp:", NULL))
2309 return AVPROBE_SCORE_MAX;
2313 static int rtp_read_header(AVFormatContext *s)
2315 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2316 char host[500], sdp[500];
2318 URLContext* in = NULL;
2320 AVCodecContext codec = { 0 };
2321 struct sockaddr_storage addr;
2323 socklen_t addrlen = sizeof(addr);
2324 RTSPState *rt = s->priv_data;
2326 if (!ff_network_init())
2327 return AVERROR(EIO);
2329 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2330 &s->interrupt_callback, NULL);
2335 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2336 if (ret == AVERROR(EAGAIN))
2341 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2345 if ((recvbuf[0] & 0xc0) != 0x80) {
2346 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2351 if (RTP_PT_IS_RTCP(recvbuf[1]))
2354 payload_type = recvbuf[1] & 0x7f;
2357 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2361 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2362 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2363 "without an SDP file describing it\n",
2367 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2368 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2369 "properly you need an SDP file "
2373 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2374 NULL, 0, s->filename);
2376 snprintf(sdp, sizeof(sdp),
2377 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2378 addr.ss_family == AF_INET ? 4 : 6, host,
2379 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2380 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2381 port, payload_type);
2382 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2384 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2387 /* sdp_read_header initializes this again */
2390 rt->media_type_mask = (1 << (AVMEDIA_TYPE_SUBTITLE+1)) - 1;
2392 ret = sdp_read_header(s);
2403 static const AVClass rtp_demuxer_class = {
2404 .class_name = "RTP demuxer",
2405 .item_name = av_default_item_name,
2406 .option = rtp_options,
2407 .version = LIBAVUTIL_VERSION_INT,
2410 AVInputFormat ff_rtp_demuxer = {
2412 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2413 .priv_data_size = sizeof(RTSPState),
2414 .read_probe = rtp_probe,
2415 .read_header = rtp_read_header,
2416 .read_packet = ff_rtsp_fetch_packet,
2417 .read_close = sdp_read_close,
2418 .flags = AVFMT_NOFILE,
2419 .priv_class = &rtp_demuxer_class,
2421 #endif /* CONFIG_RTP_DEMUXER */