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) {
173 av_log(NULL, AV_LOG_ERROR, "Invalid interval start specification '%s'\n", buf);
178 get_word_sep(buf, sizeof(buf), "-", &p);
179 if (av_parse_time(end, buf, 1) < 0)
180 av_log(NULL, AV_LOG_ERROR, "Invalid interval end specification '%s'\n", buf);
184 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
186 struct addrinfo hints = { 0 }, *ai = NULL;
187 hints.ai_flags = AI_NUMERICHOST;
188 if (getaddrinfo(buf, NULL, &hints, &ai))
190 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
196 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
197 RTSPStream *rtsp_st, AVStream *st)
199 AVCodecContext *codec = st ? st->codec : NULL;
203 codec->codec_id = handler->codec_id;
204 rtsp_st->dynamic_handler = handler;
206 st->need_parsing = handler->need_parsing;
207 if (handler->priv_data_size) {
208 rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);
209 if (!rtsp_st->dynamic_protocol_context)
210 rtsp_st->dynamic_handler = NULL;
214 static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,
217 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {
218 int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,
219 rtsp_st->dynamic_protocol_context);
221 if (rtsp_st->dynamic_protocol_context) {
222 if (rtsp_st->dynamic_handler->close)
223 rtsp_st->dynamic_handler->close(
224 rtsp_st->dynamic_protocol_context);
225 av_free(rtsp_st->dynamic_protocol_context);
227 rtsp_st->dynamic_protocol_context = NULL;
228 rtsp_st->dynamic_handler = NULL;
233 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
234 static int sdp_parse_rtpmap(AVFormatContext *s,
235 AVStream *st, RTSPStream *rtsp_st,
236 int payload_type, const char *p)
238 AVCodecContext *codec = st->codec;
244 /* See if we can handle this kind of payload.
245 * The space should normally not be there but some Real streams or
246 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
247 * have a trailing space. */
248 get_word_sep(buf, sizeof(buf), "/ ", &p);
249 if (payload_type < RTP_PT_PRIVATE) {
250 /* We are in a standard case
251 * (from http://www.iana.org/assignments/rtp-parameters). */
252 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
255 if (codec->codec_id == AV_CODEC_ID_NONE) {
256 RTPDynamicProtocolHandler *handler =
257 ff_rtp_handler_find_by_name(buf, codec->codec_type);
258 init_rtp_handler(handler, rtsp_st, st);
259 /* If no dynamic handler was found, check with the list of standard
260 * allocated types, if such a stream for some reason happens to
261 * use a private payload type. This isn't handled in rtpdec.c, since
262 * the format name from the rtpmap line never is passed into rtpdec. */
263 if (!rtsp_st->dynamic_handler)
264 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
267 c = avcodec_find_decoder(codec->codec_id);
273 get_word_sep(buf, sizeof(buf), "/", &p);
275 switch (codec->codec_type) {
276 case AVMEDIA_TYPE_AUDIO:
277 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
278 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
279 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
281 codec->sample_rate = i;
282 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
283 get_word_sep(buf, sizeof(buf), "/", &p);
288 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
290 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
293 case AVMEDIA_TYPE_VIDEO:
294 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
296 avpriv_set_pts_info(st, 32, 1, i);
301 finalize_rtp_handler_init(s, rtsp_st, st);
305 /* parse the attribute line from the fmtp a line of an sdp response. This
306 * is broken out as a function because it is used in rtp_h264.c, which is
308 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
309 char *value, int value_size)
311 *p += strspn(*p, SPACE_CHARS);
313 get_word_sep(attr, attr_size, "=", p);
316 get_word_sep(value, value_size, ";", p);
324 typedef struct SDPParseState {
326 struct sockaddr_storage default_ip;
328 int skip_media; ///< set if an unknown m= line occurs
329 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
330 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
331 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
332 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
335 char delayed_fmtp[2048];
338 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
339 struct RTSPSource ***dest, int *dest_count)
341 RTSPSource *rtsp_src, *rtsp_src2;
343 for (i = 0; i < count; i++) {
345 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
348 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
349 dynarray_add(dest, dest_count, rtsp_src2);
353 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
354 int payload_type, const char *line)
358 for (i = 0; i < rt->nb_rtsp_streams; i++) {
359 RTSPStream *rtsp_st = rt->rtsp_streams[i];
360 if (rtsp_st->sdp_payload_type == payload_type &&
361 rtsp_st->dynamic_handler &&
362 rtsp_st->dynamic_handler->parse_sdp_a_line) {
363 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
364 rtsp_st->dynamic_protocol_context, line);
369 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
370 int letter, const char *buf)
372 RTSPState *rt = s->priv_data;
373 char buf1[64], st_type[64];
375 enum AVMediaType codec_type;
379 RTSPSource *rtsp_src;
380 struct sockaddr_storage sdp_ip;
383 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
386 if (s1->skip_media && letter != 'm')
390 get_word(buf1, sizeof(buf1), &p);
391 if (strcmp(buf1, "IN") != 0)
393 get_word(buf1, sizeof(buf1), &p);
394 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
396 get_word_sep(buf1, sizeof(buf1), "/", &p);
397 if (get_sockaddr(buf1, &sdp_ip))
402 get_word_sep(buf1, sizeof(buf1), "/", &p);
405 if (s->nb_streams == 0) {
406 s1->default_ip = sdp_ip;
407 s1->default_ttl = ttl;
409 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
410 rtsp_st->sdp_ip = sdp_ip;
411 rtsp_st->sdp_ttl = ttl;
415 av_dict_set(&s->metadata, "title", p, 0);
418 if (s->nb_streams == 0) {
419 av_dict_set(&s->metadata, "comment", p, 0);
428 codec_type = AVMEDIA_TYPE_UNKNOWN;
429 get_word(st_type, sizeof(st_type), &p);
430 if (!strcmp(st_type, "audio")) {
431 codec_type = AVMEDIA_TYPE_AUDIO;
432 } else if (!strcmp(st_type, "video")) {
433 codec_type = AVMEDIA_TYPE_VIDEO;
434 } else if (!strcmp(st_type, "application") || !strcmp(st_type, "text")) {
435 codec_type = AVMEDIA_TYPE_DATA;
437 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
441 rtsp_st = av_mallocz(sizeof(RTSPStream));
444 rtsp_st->stream_index = -1;
445 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
447 rtsp_st->sdp_ip = s1->default_ip;
448 rtsp_st->sdp_ttl = s1->default_ttl;
450 copy_default_source_addrs(s1->default_include_source_addrs,
451 s1->nb_default_include_source_addrs,
452 &rtsp_st->include_source_addrs,
453 &rtsp_st->nb_include_source_addrs);
454 copy_default_source_addrs(s1->default_exclude_source_addrs,
455 s1->nb_default_exclude_source_addrs,
456 &rtsp_st->exclude_source_addrs,
457 &rtsp_st->nb_exclude_source_addrs);
459 get_word(buf1, sizeof(buf1), &p); /* port */
460 rtsp_st->sdp_port = atoi(buf1);
462 get_word(buf1, sizeof(buf1), &p); /* protocol */
463 if (!strcmp(buf1, "udp"))
464 rt->transport = RTSP_TRANSPORT_RAW;
465 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
466 rtsp_st->feedback = 1;
468 /* XXX: handle list of formats */
469 get_word(buf1, sizeof(buf1), &p); /* format list */
470 rtsp_st->sdp_payload_type = atoi(buf1);
472 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
473 /* no corresponding stream */
474 if (rt->transport == RTSP_TRANSPORT_RAW) {
475 if (CONFIG_RTPDEC && !rt->ts)
476 rt->ts = ff_mpegts_parse_open(s);
478 RTPDynamicProtocolHandler *handler;
479 handler = ff_rtp_handler_find_by_id(
480 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
481 init_rtp_handler(handler, rtsp_st, NULL);
482 finalize_rtp_handler_init(s, rtsp_st, NULL);
484 } else if (rt->server_type == RTSP_SERVER_WMS &&
485 codec_type == AVMEDIA_TYPE_DATA) {
486 /* RTX stream, a stream that carries all the other actual
487 * audio/video streams. Don't expose this to the callers. */
489 st = avformat_new_stream(s, NULL);
492 st->id = rt->nb_rtsp_streams - 1;
493 rtsp_st->stream_index = st->index;
494 st->codec->codec_type = codec_type;
495 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
496 RTPDynamicProtocolHandler *handler;
497 /* if standard payload type, we can find the codec right now */
498 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
499 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
500 st->codec->sample_rate > 0)
501 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
502 /* Even static payload types may need a custom depacketizer */
503 handler = ff_rtp_handler_find_by_id(
504 rtsp_st->sdp_payload_type, st->codec->codec_type);
505 init_rtp_handler(handler, rtsp_st, st);
506 finalize_rtp_handler_init(s, rtsp_st, st);
508 if (rt->default_lang[0])
509 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
511 /* put a default control url */
512 av_strlcpy(rtsp_st->control_url, rt->control_uri,
513 sizeof(rtsp_st->control_url));
516 if (av_strstart(p, "control:", &p)) {
517 if (s->nb_streams == 0) {
518 if (!strncmp(p, "rtsp://", 7))
519 av_strlcpy(rt->control_uri, p,
520 sizeof(rt->control_uri));
523 /* get the control url */
524 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
526 /* XXX: may need to add full url resolution */
527 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
529 if (proto[0] == '\0') {
530 /* relative control URL */
531 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
532 av_strlcat(rtsp_st->control_url, "/",
533 sizeof(rtsp_st->control_url));
534 av_strlcat(rtsp_st->control_url, p,
535 sizeof(rtsp_st->control_url));
537 av_strlcpy(rtsp_st->control_url, p,
538 sizeof(rtsp_st->control_url));
540 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
541 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
542 get_word(buf1, sizeof(buf1), &p);
543 payload_type = atoi(buf1);
544 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
545 if (rtsp_st->stream_index >= 0) {
546 st = s->streams[rtsp_st->stream_index];
547 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
551 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
553 } else if (av_strstart(p, "fmtp:", &p) ||
554 av_strstart(p, "framesize:", &p)) {
555 // let dynamic protocol handlers have a stab at the line.
556 get_word(buf1, sizeof(buf1), &p);
557 payload_type = atoi(buf1);
558 if (s1->seen_rtpmap) {
559 parse_fmtp(s, rt, payload_type, buf);
562 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
564 } else if (av_strstart(p, "range:", &p)) {
567 // this is so that seeking on a streamed file can work.
568 rtsp_parse_range_npt(p, &start, &end);
569 s->start_time = start;
570 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
571 s->duration = (end == AV_NOPTS_VALUE) ?
572 AV_NOPTS_VALUE : end - start;
573 } else if (av_strstart(p, "lang:", &p)) {
574 if (s->nb_streams > 0) {
575 get_word(buf1, sizeof(buf1), &p);
576 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
577 if (rtsp_st->stream_index >= 0) {
578 st = s->streams[rtsp_st->stream_index];
579 av_dict_set(&st->metadata, "language", buf1, 0);
582 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
583 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
585 rt->transport = RTSP_TRANSPORT_RDT;
586 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
588 st = s->streams[s->nb_streams - 1];
589 st->codec->sample_rate = atoi(p);
590 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
592 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
593 get_word(buf1, sizeof(buf1), &p); // ignore tag
594 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
595 p += strspn(p, SPACE_CHARS);
596 if (av_strstart(p, "inline:", &p))
597 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
598 } else if (av_strstart(p, "source-filter:", &p)) {
600 get_word(buf1, sizeof(buf1), &p);
601 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
603 exclude = !strcmp(buf1, "excl");
605 get_word(buf1, sizeof(buf1), &p);
606 if (strcmp(buf1, "IN") != 0)
608 get_word(buf1, sizeof(buf1), &p);
609 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
611 // not checking that the destination address actually matches or is wildcard
612 get_word(buf1, sizeof(buf1), &p);
615 rtsp_src = av_mallocz(sizeof(*rtsp_src));
618 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
620 if (s->nb_streams == 0) {
621 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
623 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
624 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
627 if (s->nb_streams == 0) {
628 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
630 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
631 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
636 if (rt->server_type == RTSP_SERVER_WMS)
637 ff_wms_parse_sdp_a_line(s, p);
638 if (s->nb_streams > 0) {
639 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
641 if (rt->server_type == RTSP_SERVER_REAL)
642 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
644 if (rtsp_st->dynamic_handler &&
645 rtsp_st->dynamic_handler->parse_sdp_a_line)
646 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
647 rtsp_st->stream_index,
648 rtsp_st->dynamic_protocol_context, buf);
655 int ff_sdp_parse(AVFormatContext *s, const char *content)
657 RTSPState *rt = s->priv_data;
660 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
661 * contain long SDP lines containing complete ASF Headers (several
662 * kB) or arrays of MDPR (RM stream descriptor) headers plus
663 * "rulebooks" describing their properties. Therefore, the SDP line
666 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
667 * in rtpdec_xiph.c. */
669 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
673 p += strspn(p, SPACE_CHARS);
681 /* get the content */
683 while (*p != '\n' && *p != '\r' && *p != '\0') {
684 if ((q - buf) < sizeof(buf) - 1)
689 sdp_parse_line(s, s1, letter, buf);
691 while (*p != '\n' && *p != '\0')
697 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
698 av_free(s1->default_include_source_addrs[i]);
699 av_freep(&s1->default_include_source_addrs);
700 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
701 av_free(s1->default_exclude_source_addrs[i]);
702 av_freep(&s1->default_exclude_source_addrs);
704 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
705 if (!rt->p) return AVERROR(ENOMEM);
708 #endif /* CONFIG_RTPDEC */
710 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
712 RTSPState *rt = s->priv_data;
715 for (i = 0; i < rt->nb_rtsp_streams; i++) {
716 RTSPStream *rtsp_st = rt->rtsp_streams[i];
719 if (rtsp_st->transport_priv) {
721 AVFormatContext *rtpctx = rtsp_st->transport_priv;
722 av_write_trailer(rtpctx);
723 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
724 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
725 ff_rtsp_tcp_write_packet(s, rtsp_st);
726 ffio_free_dyn_buf(&rtpctx->pb);
728 avio_close(rtpctx->pb);
730 avformat_free_context(rtpctx);
731 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
732 ff_rdt_parse_close(rtsp_st->transport_priv);
733 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
734 ff_rtp_parse_close(rtsp_st->transport_priv);
736 rtsp_st->transport_priv = NULL;
737 if (rtsp_st->rtp_handle)
738 ffurl_close(rtsp_st->rtp_handle);
739 rtsp_st->rtp_handle = NULL;
743 /* close and free RTSP streams */
744 void ff_rtsp_close_streams(AVFormatContext *s)
746 RTSPState *rt = s->priv_data;
750 ff_rtsp_undo_setup(s, 0);
751 for (i = 0; i < rt->nb_rtsp_streams; i++) {
752 rtsp_st = rt->rtsp_streams[i];
754 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) {
755 if (rtsp_st->dynamic_handler->close)
756 rtsp_st->dynamic_handler->close(
757 rtsp_st->dynamic_protocol_context);
758 av_free(rtsp_st->dynamic_protocol_context);
760 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
761 av_free(rtsp_st->include_source_addrs[j]);
762 av_freep(&rtsp_st->include_source_addrs);
763 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
764 av_free(rtsp_st->exclude_source_addrs[j]);
765 av_freep(&rtsp_st->exclude_source_addrs);
770 av_free(rt->rtsp_streams);
772 avformat_close_input(&rt->asf_ctx);
774 if (CONFIG_RTPDEC && rt->ts)
775 ff_mpegts_parse_close(rt->ts);
777 av_free(rt->recvbuf);
780 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
782 RTSPState *rt = s->priv_data;
784 int reordering_queue_size = rt->reordering_queue_size;
785 if (reordering_queue_size < 0) {
786 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
787 reordering_queue_size = 0;
789 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
792 /* open the RTP context */
793 if (rtsp_st->stream_index >= 0)
794 st = s->streams[rtsp_st->stream_index];
796 s->ctx_flags |= AVFMTCTX_NOHEADER;
798 if (CONFIG_RTSP_MUXER && s->oformat) {
799 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
800 s, st, rtsp_st->rtp_handle,
801 RTSP_TCP_MAX_PACKET_SIZE,
802 rtsp_st->stream_index);
803 /* Ownership of rtp_handle is passed to the rtp mux context */
804 rtsp_st->rtp_handle = NULL;
807 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
808 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
809 return 0; // Don't need to open any parser here
810 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
811 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
812 rtsp_st->dynamic_protocol_context,
813 rtsp_st->dynamic_handler);
814 else if (CONFIG_RTPDEC)
815 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
816 rtsp_st->sdp_payload_type,
817 reordering_queue_size);
819 if (!rtsp_st->transport_priv) {
820 return AVERROR(ENOMEM);
821 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
822 if (rtsp_st->dynamic_handler) {
823 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
824 rtsp_st->dynamic_protocol_context,
825 rtsp_st->dynamic_handler);
827 if (rtsp_st->crypto_suite[0])
828 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
829 rtsp_st->crypto_suite,
830 rtsp_st->crypto_params);
836 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
837 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
844 q += strspn(q, SPACE_CHARS);
845 v = strtol(q, &p, 10);
849 v = strtol(p, &p, 10);
858 /* XXX: only one transport specification is parsed */
859 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
861 char transport_protocol[16];
863 char lower_transport[16];
865 RTSPTransportField *th;
868 reply->nb_transports = 0;
871 p += strspn(p, SPACE_CHARS);
875 th = &reply->transports[reply->nb_transports];
877 get_word_sep(transport_protocol, sizeof(transport_protocol),
879 if (!av_strcasecmp (transport_protocol, "rtp")) {
880 get_word_sep(profile, sizeof(profile), "/;,", &p);
881 lower_transport[0] = '\0';
882 /* rtp/avp/<protocol> */
884 get_word_sep(lower_transport, sizeof(lower_transport),
887 th->transport = RTSP_TRANSPORT_RTP;
888 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
889 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
890 /* x-pn-tng/<protocol> */
891 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
893 th->transport = RTSP_TRANSPORT_RDT;
894 } else if (!av_strcasecmp(transport_protocol, "raw")) {
895 get_word_sep(profile, sizeof(profile), "/;,", &p);
896 lower_transport[0] = '\0';
897 /* raw/raw/<protocol> */
899 get_word_sep(lower_transport, sizeof(lower_transport),
902 th->transport = RTSP_TRANSPORT_RAW;
904 if (!av_strcasecmp(lower_transport, "TCP"))
905 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
907 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
911 /* get each parameter */
912 while (*p != '\0' && *p != ',') {
913 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
914 if (!strcmp(parameter, "port")) {
917 rtsp_parse_range(&th->port_min, &th->port_max, &p);
919 } else if (!strcmp(parameter, "client_port")) {
922 rtsp_parse_range(&th->client_port_min,
923 &th->client_port_max, &p);
925 } else if (!strcmp(parameter, "server_port")) {
928 rtsp_parse_range(&th->server_port_min,
929 &th->server_port_max, &p);
931 } else if (!strcmp(parameter, "interleaved")) {
934 rtsp_parse_range(&th->interleaved_min,
935 &th->interleaved_max, &p);
937 } else if (!strcmp(parameter, "multicast")) {
938 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
939 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
940 } else if (!strcmp(parameter, "ttl")) {
944 th->ttl = strtol(p, &end, 10);
947 } else if (!strcmp(parameter, "destination")) {
950 get_word_sep(buf, sizeof(buf), ";,", &p);
951 get_sockaddr(buf, &th->destination);
953 } else if (!strcmp(parameter, "source")) {
956 get_word_sep(buf, sizeof(buf), ";,", &p);
957 av_strlcpy(th->source, buf, sizeof(th->source));
959 } else if (!strcmp(parameter, "mode")) {
962 get_word_sep(buf, sizeof(buf), ";, ", &p);
963 if (!strcmp(buf, "record") ||
964 !strcmp(buf, "receive"))
969 while (*p != ';' && *p != '\0' && *p != ',')
977 reply->nb_transports++;
981 static void handle_rtp_info(RTSPState *rt, const char *url,
982 uint32_t seq, uint32_t rtptime)
985 if (!rtptime || !url[0])
987 if (rt->transport != RTSP_TRANSPORT_RTP)
989 for (i = 0; i < rt->nb_rtsp_streams; i++) {
990 RTSPStream *rtsp_st = rt->rtsp_streams[i];
991 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
994 if (!strcmp(rtsp_st->control_url, url)) {
995 rtpctx->base_timestamp = rtptime;
1001 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
1004 char key[20], value[1024], url[1024] = "";
1005 uint32_t seq = 0, rtptime = 0;
1008 p += strspn(p, SPACE_CHARS);
1011 get_word_sep(key, sizeof(key), "=", &p);
1015 get_word_sep(value, sizeof(value), ";, ", &p);
1017 if (!strcmp(key, "url"))
1018 av_strlcpy(url, value, sizeof(url));
1019 else if (!strcmp(key, "seq"))
1020 seq = strtoul(value, NULL, 10);
1021 else if (!strcmp(key, "rtptime"))
1022 rtptime = strtoul(value, NULL, 10);
1024 handle_rtp_info(rt, url, seq, rtptime);
1033 handle_rtp_info(rt, url, seq, rtptime);
1036 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1037 RTSPState *rt, const char *method)
1041 /* NOTE: we do case independent match for broken servers */
1043 if (av_stristart(p, "Session:", &p)) {
1045 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1046 if (av_stristart(p, ";timeout=", &p) &&
1047 (t = strtol(p, NULL, 10)) > 0) {
1050 } else if (av_stristart(p, "Content-Length:", &p)) {
1051 reply->content_length = strtol(p, NULL, 10);
1052 } else if (av_stristart(p, "Transport:", &p)) {
1053 rtsp_parse_transport(reply, p);
1054 } else if (av_stristart(p, "CSeq:", &p)) {
1055 reply->seq = strtol(p, NULL, 10);
1056 } else if (av_stristart(p, "Range:", &p)) {
1057 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1058 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1059 p += strspn(p, SPACE_CHARS);
1060 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1061 } else if (av_stristart(p, "Server:", &p)) {
1062 p += strspn(p, SPACE_CHARS);
1063 av_strlcpy(reply->server, p, sizeof(reply->server));
1064 } else if (av_stristart(p, "Notice:", &p) ||
1065 av_stristart(p, "X-Notice:", &p)) {
1066 reply->notice = strtol(p, NULL, 10);
1067 } else if (av_stristart(p, "Location:", &p)) {
1068 p += strspn(p, SPACE_CHARS);
1069 av_strlcpy(reply->location, p , sizeof(reply->location));
1070 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1071 p += strspn(p, SPACE_CHARS);
1072 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1073 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1074 p += strspn(p, SPACE_CHARS);
1075 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1076 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1077 p += strspn(p, SPACE_CHARS);
1078 if (method && !strcmp(method, "DESCRIBE"))
1079 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1080 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1081 p += strspn(p, SPACE_CHARS);
1082 if (method && !strcmp(method, "PLAY"))
1083 rtsp_parse_rtp_info(rt, p);
1084 } else if (av_stristart(p, "Public:", &p) && rt) {
1085 if (strstr(p, "GET_PARAMETER") &&
1086 method && !strcmp(method, "OPTIONS"))
1087 rt->get_parameter_supported = 1;
1088 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1089 p += strspn(p, SPACE_CHARS);
1090 rt->accept_dynamic_rate = atoi(p);
1091 } else if (av_stristart(p, "Content-Type:", &p)) {
1092 p += strspn(p, SPACE_CHARS);
1093 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1097 /* skip a RTP/TCP interleaved packet */
1098 void ff_rtsp_skip_packet(AVFormatContext *s)
1100 RTSPState *rt = s->priv_data;
1104 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1107 len = AV_RB16(buf + 1);
1109 av_dlog(s, "skipping RTP packet len=%d\n", len);
1114 if (len1 > sizeof(buf))
1116 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1123 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1124 unsigned char **content_ptr,
1125 int return_on_interleaved_data, const char *method)
1127 RTSPState *rt = s->priv_data;
1128 char buf[4096], buf1[1024], *q;
1131 int ret, content_length, line_count = 0, request = 0;
1132 unsigned char *content = NULL;
1138 memset(reply, 0, sizeof(*reply));
1140 /* parse reply (XXX: use buffers) */
1141 rt->last_reply[0] = '\0';
1145 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1146 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1152 /* XXX: only parse it if first char on line ? */
1153 if (return_on_interleaved_data) {
1156 ff_rtsp_skip_packet(s);
1157 } else if (ch != '\r') {
1158 if ((q - buf) < sizeof(buf) - 1)
1164 av_dlog(s, "line='%s'\n", buf);
1166 /* test if last line */
1170 if (line_count == 0) {
1171 /* get reply code */
1172 get_word(buf1, sizeof(buf1), &p);
1173 if (!strncmp(buf1, "RTSP/", 5)) {
1174 get_word(buf1, sizeof(buf1), &p);
1175 reply->status_code = atoi(buf1);
1176 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1178 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1179 get_word(buf1, sizeof(buf1), &p); // object
1183 ff_rtsp_parse_line(reply, p, rt, method);
1184 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1185 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1190 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1191 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1193 content_length = reply->content_length;
1194 if (content_length > 0) {
1195 /* leave some room for a trailing '\0' (useful for simple parsing) */
1196 content = av_malloc(content_length + 1);
1198 return AVERROR(ENOMEM);
1199 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1200 content[content_length] = '\0';
1203 *content_ptr = content;
1209 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1210 const char* ptr = buf;
1212 if (!strcmp(reply->reason, "OPTIONS")) {
1213 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1215 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1216 if (reply->session_id[0])
1217 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1220 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1222 av_strlcat(buf, "\r\n", sizeof(buf));
1224 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1225 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1228 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1230 rt->last_cmd_time = av_gettime_relative();
1231 /* Even if the request from the server had data, it is not the data
1232 * that the caller wants or expects. The memory could also be leaked
1233 * if the actual following reply has content data. */
1235 av_freep(content_ptr);
1236 /* If method is set, this is called from ff_rtsp_send_cmd,
1237 * where a reply to exactly this request is awaited. For
1238 * callers from within packet receiving, we just want to
1239 * return to the caller and go back to receiving packets. */
1245 if (rt->seq != reply->seq) {
1246 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1247 rt->seq, reply->seq);
1251 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1252 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1253 reply->notice == 2306 /* Continuous Feed Terminated */) {
1254 rt->state = RTSP_STATE_IDLE;
1255 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1256 return AVERROR(EIO); /* data or server error */
1257 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1258 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1259 return AVERROR(EPERM);
1265 * Send a command to the RTSP server without waiting for the reply.
1267 * @param s RTSP (de)muxer context
1268 * @param method the method for the request
1269 * @param url the target url for the request
1270 * @param headers extra header lines to include in the request
1271 * @param send_content if non-null, the data to send as request body content
1272 * @param send_content_length the length of the send_content data, or 0 if
1273 * send_content is null
1275 * @return zero if success, nonzero otherwise
1277 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1278 const char *method, const char *url,
1279 const char *headers,
1280 const unsigned char *send_content,
1281 int send_content_length)
1283 RTSPState *rt = s->priv_data;
1284 char buf[4096], *out_buf;
1285 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1287 /* Add in RTSP headers */
1290 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1292 av_strlcat(buf, headers, sizeof(buf));
1293 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1294 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1295 if (rt->session_id[0] != '\0' && (!headers ||
1296 !strstr(headers, "\nIf-Match:"))) {
1297 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1300 char *str = ff_http_auth_create_response(&rt->auth_state,
1301 rt->auth, url, method);
1303 av_strlcat(buf, str, sizeof(buf));
1306 if (send_content_length > 0 && send_content)
1307 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1308 av_strlcat(buf, "\r\n", sizeof(buf));
1310 /* base64 encode rtsp if tunneling */
1311 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1312 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1313 out_buf = base64buf;
1316 av_dlog(s, "Sending:\n%s--\n", buf);
1318 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1319 if (send_content_length > 0 && send_content) {
1320 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1321 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1322 "with content data not supported\n");
1323 return AVERROR_PATCHWELCOME;
1325 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1327 rt->last_cmd_time = av_gettime_relative();
1332 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1333 const char *url, const char *headers)
1335 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1338 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1339 const char *headers, RTSPMessageHeader *reply,
1340 unsigned char **content_ptr)
1342 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1343 content_ptr, NULL, 0);
1346 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1347 const char *method, const char *url,
1349 RTSPMessageHeader *reply,
1350 unsigned char **content_ptr,
1351 const unsigned char *send_content,
1352 int send_content_length)
1354 RTSPState *rt = s->priv_data;
1355 HTTPAuthType cur_auth_type;
1356 int ret, attempts = 0;
1359 cur_auth_type = rt->auth_state.auth_type;
1360 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1362 send_content_length)))
1365 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1369 if (reply->status_code == 401 &&
1370 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1371 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1374 if (reply->status_code > 400){
1375 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1379 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1385 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1386 int lower_transport, const char *real_challenge)
1388 RTSPState *rt = s->priv_data;
1389 int rtx = 0, j, i, err, interleave = 0, port_off;
1390 RTSPStream *rtsp_st;
1391 RTSPMessageHeader reply1, *reply = &reply1;
1393 const char *trans_pref;
1395 if (rt->transport == RTSP_TRANSPORT_RDT)
1396 trans_pref = "x-pn-tng";
1397 else if (rt->transport == RTSP_TRANSPORT_RAW)
1398 trans_pref = "RAW/RAW";
1400 trans_pref = "RTP/AVP";
1402 /* default timeout: 1 minute */
1405 /* for each stream, make the setup request */
1406 /* XXX: we assume the same server is used for the control of each
1409 /* Choose a random starting offset within the first half of the
1410 * port range, to allow for a number of ports to try even if the offset
1411 * happens to be at the end of the random range. */
1412 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1413 /* even random offset */
1414 port_off -= port_off & 0x01;
1416 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1417 char transport[2048];
1420 * WMS serves all UDP data over a single connection, the RTX, which
1421 * isn't necessarily the first in the SDP but has to be the first
1422 * to be set up, else the second/third SETUP will fail with a 461.
1424 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1425 rt->server_type == RTSP_SERVER_WMS) {
1428 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1429 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1431 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1435 if (rtx == rt->nb_rtsp_streams)
1436 return -1; /* no RTX found */
1437 rtsp_st = rt->rtsp_streams[rtx];
1439 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1441 rtsp_st = rt->rtsp_streams[i];
1444 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1447 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1448 port = reply->transports[0].client_port_min;
1452 /* first try in specified port range */
1453 while (j <= rt->rtp_port_max) {
1454 AVDictionary *opts = map_to_opts(rt);
1456 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1457 "?localport=%d", j);
1458 /* we will use two ports per rtp stream (rtp and rtcp) */
1460 err = ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1461 &s->interrupt_callback, &opts);
1463 av_dict_free(&opts);
1469 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1474 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1476 snprintf(transport, sizeof(transport) - 1,
1477 "%s/UDP;", trans_pref);
1478 if (rt->server_type != RTSP_SERVER_REAL)
1479 av_strlcat(transport, "unicast;", sizeof(transport));
1480 av_strlcatf(transport, sizeof(transport),
1481 "client_port=%d", port);
1482 if (rt->transport == RTSP_TRANSPORT_RTP &&
1483 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1484 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1488 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1489 /* For WMS streams, the application streams are only used for
1490 * UDP. When trying to set it up for TCP streams, the server
1491 * will return an error. Therefore, we skip those streams. */
1492 if (rt->server_type == RTSP_SERVER_WMS &&
1493 (rtsp_st->stream_index < 0 ||
1494 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1497 snprintf(transport, sizeof(transport) - 1,
1498 "%s/TCP;", trans_pref);
1499 if (rt->transport != RTSP_TRANSPORT_RDT)
1500 av_strlcat(transport, "unicast;", sizeof(transport));
1501 av_strlcatf(transport, sizeof(transport),
1502 "interleaved=%d-%d",
1503 interleave, interleave + 1);
1507 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1508 snprintf(transport, sizeof(transport) - 1,
1509 "%s/UDP;multicast", trans_pref);
1512 av_strlcat(transport, ";mode=record", sizeof(transport));
1513 } else if (rt->server_type == RTSP_SERVER_REAL ||
1514 rt->server_type == RTSP_SERVER_WMS)
1515 av_strlcat(transport, ";mode=play", sizeof(transport));
1516 snprintf(cmd, sizeof(cmd),
1517 "Transport: %s\r\n",
1519 if (rt->accept_dynamic_rate)
1520 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1521 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1522 char real_res[41], real_csum[9];
1523 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1525 av_strlcatf(cmd, sizeof(cmd),
1527 "RealChallenge2: %s, sd=%s\r\n",
1528 rt->session_id, real_res, real_csum);
1530 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1531 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1534 } else if (reply->status_code != RTSP_STATUS_OK ||
1535 reply->nb_transports != 1) {
1536 err = AVERROR_INVALIDDATA;
1540 /* XXX: same protocol for all streams is required */
1542 if (reply->transports[0].lower_transport != rt->lower_transport ||
1543 reply->transports[0].transport != rt->transport) {
1544 err = AVERROR_INVALIDDATA;
1548 rt->lower_transport = reply->transports[0].lower_transport;
1549 rt->transport = reply->transports[0].transport;
1552 /* Fail if the server responded with another lower transport mode
1553 * than what we requested. */
1554 if (reply->transports[0].lower_transport != lower_transport) {
1555 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1556 err = AVERROR_INVALIDDATA;
1560 switch(reply->transports[0].lower_transport) {
1561 case RTSP_LOWER_TRANSPORT_TCP:
1562 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1563 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1566 case RTSP_LOWER_TRANSPORT_UDP: {
1567 char url[1024], options[30] = "";
1568 const char *peer = host;
1570 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1571 av_strlcpy(options, "?connect=1", sizeof(options));
1572 /* Use source address if specified */
1573 if (reply->transports[0].source[0])
1574 peer = reply->transports[0].source;
1575 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1576 reply->transports[0].server_port_min, "%s", options);
1577 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1578 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1579 err = AVERROR_INVALIDDATA;
1584 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1585 char url[1024], namebuf[50], optbuf[20] = "";
1586 struct sockaddr_storage addr;
1589 if (reply->transports[0].destination.ss_family) {
1590 addr = reply->transports[0].destination;
1591 port = reply->transports[0].port_min;
1592 ttl = reply->transports[0].ttl;
1594 addr = rtsp_st->sdp_ip;
1595 port = rtsp_st->sdp_port;
1596 ttl = rtsp_st->sdp_ttl;
1599 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1600 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1601 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1602 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1603 port, "%s", optbuf);
1604 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1605 &s->interrupt_callback, NULL) < 0) {
1606 err = AVERROR_INVALIDDATA;
1613 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1617 if (rt->nb_rtsp_streams && reply->timeout > 0)
1618 rt->timeout = reply->timeout;
1620 if (rt->server_type == RTSP_SERVER_REAL)
1621 rt->need_subscription = 1;
1626 ff_rtsp_undo_setup(s, 0);
1630 void ff_rtsp_close_connections(AVFormatContext *s)
1632 RTSPState *rt = s->priv_data;
1633 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1634 ffurl_close(rt->rtsp_hd);
1635 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1638 int ff_rtsp_connect(AVFormatContext *s)
1640 RTSPState *rt = s->priv_data;
1641 char proto[128], host[1024], path[1024];
1642 char tcpname[1024], cmd[2048], auth[128];
1643 const char *lower_rtsp_proto = "tcp";
1644 int port, err, tcp_fd;
1645 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1646 int lower_transport_mask = 0;
1647 int default_port = RTSP_DEFAULT_PORT;
1648 char real_challenge[64] = "";
1649 struct sockaddr_storage peer;
1650 socklen_t peer_len = sizeof(peer);
1652 if (rt->rtp_port_max < rt->rtp_port_min) {
1653 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1654 "than min port %d\n", rt->rtp_port_max,
1656 return AVERROR(EINVAL);
1659 if (!ff_network_init())
1660 return AVERROR(EIO);
1662 if (s->max_delay < 0) /* Not set by the caller */
1663 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1665 rt->control_transport = RTSP_MODE_PLAIN;
1666 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1667 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1668 rt->control_transport = RTSP_MODE_TUNNEL;
1670 /* Only pass through valid flags from here */
1671 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1674 /* extract hostname and port */
1675 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1676 host, sizeof(host), &port, path, sizeof(path), s->filename);
1678 if (!strcmp(proto, "rtsps")) {
1679 lower_rtsp_proto = "tls";
1680 default_port = RTSPS_DEFAULT_PORT;
1681 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1685 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1688 port = default_port;
1690 lower_transport_mask = rt->lower_transport_mask;
1692 if (!lower_transport_mask)
1693 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1696 /* Only UDP or TCP - UDP multicast isn't supported. */
1697 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1698 (1 << RTSP_LOWER_TRANSPORT_TCP);
1699 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1700 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1701 "only UDP and TCP are supported for output.\n");
1702 err = AVERROR(EINVAL);
1707 /* Construct the URI used in request; this is similar to s->filename,
1708 * but with authentication credentials removed and RTSP specific options
1710 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1711 host, port, "%s", path);
1713 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1714 /* set up initial handshake for tunneling */
1715 char httpname[1024];
1716 char sessioncookie[17];
1719 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1720 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1721 av_get_random_seed(), av_get_random_seed());
1724 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1725 &s->interrupt_callback) < 0) {
1730 /* generate GET headers */
1731 snprintf(headers, sizeof(headers),
1732 "x-sessioncookie: %s\r\n"
1733 "Accept: application/x-rtsp-tunnelled\r\n"
1734 "Pragma: no-cache\r\n"
1735 "Cache-Control: no-cache\r\n",
1737 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1739 /* complete the connection */
1740 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1746 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1747 &s->interrupt_callback) < 0 ) {
1752 /* generate POST headers */
1753 snprintf(headers, sizeof(headers),
1754 "x-sessioncookie: %s\r\n"
1755 "Content-Type: application/x-rtsp-tunnelled\r\n"
1756 "Pragma: no-cache\r\n"
1757 "Cache-Control: no-cache\r\n"
1758 "Content-Length: 32767\r\n"
1759 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1761 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1762 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1764 /* Initialize the authentication state for the POST session. The HTTP
1765 * protocol implementation doesn't properly handle multi-pass
1766 * authentication for POST requests, since it would require one of
1768 * - implementing Expect: 100-continue, which many HTTP servers
1769 * don't support anyway, even less the RTSP servers that do HTTP
1771 * - sending the whole POST data until getting a 401 reply specifying
1772 * what authentication method to use, then resending all that data
1773 * - waiting for potential 401 replies directly after sending the
1774 * POST header (waiting for some unspecified time)
1775 * Therefore, we copy the full auth state, which works for both basic
1776 * and digest. (For digest, we would have to synchronize the nonce
1777 * count variable between the two sessions, if we'd do more requests
1778 * with the original session, though.)
1780 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1782 /* complete the connection */
1783 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1788 /* open the tcp connection */
1789 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1791 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1792 &s->interrupt_callback, NULL) < 0) {
1796 rt->rtsp_hd_out = rt->rtsp_hd;
1800 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1805 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1806 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1807 NULL, 0, NI_NUMERICHOST);
1810 /* request options supported by the server; this also detects server
1812 for (rt->server_type = RTSP_SERVER_RTP;;) {
1814 if (rt->server_type == RTSP_SERVER_REAL)
1817 * The following entries are required for proper
1818 * streaming from a Realmedia server. They are
1819 * interdependent in some way although we currently
1820 * don't quite understand how. Values were copied
1821 * from mplayer SVN r23589.
1822 * ClientChallenge is a 16-byte ID in hex
1823 * CompanyID is a 16-byte ID in base64
1825 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1826 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1827 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1828 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1830 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1831 if (reply->status_code != RTSP_STATUS_OK) {
1832 err = AVERROR_INVALIDDATA;
1836 /* detect server type if not standard-compliant RTP */
1837 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1838 rt->server_type = RTSP_SERVER_REAL;
1840 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1841 rt->server_type = RTSP_SERVER_WMS;
1842 } else if (rt->server_type == RTSP_SERVER_REAL)
1843 strcpy(real_challenge, reply->real_challenge);
1847 if (CONFIG_RTSP_DEMUXER && s->iformat)
1848 err = ff_rtsp_setup_input_streams(s, reply);
1849 else if (CONFIG_RTSP_MUXER)
1850 err = ff_rtsp_setup_output_streams(s, host);
1855 int lower_transport = ff_log2_tab[lower_transport_mask &
1856 ~(lower_transport_mask - 1)];
1858 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1859 rt->server_type == RTSP_SERVER_REAL ?
1860 real_challenge : NULL);
1863 lower_transport_mask &= ~(1 << lower_transport);
1864 if (lower_transport_mask == 0 && err == 1) {
1865 err = AVERROR(EPROTONOSUPPORT);
1870 rt->lower_transport_mask = lower_transport_mask;
1871 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1872 rt->state = RTSP_STATE_IDLE;
1873 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1876 ff_rtsp_close_streams(s);
1877 ff_rtsp_close_connections(s);
1878 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1879 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1880 rt->session_id[0] = '\0';
1881 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1889 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1892 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1893 uint8_t *buf, int buf_size, int64_t wait_end)
1895 RTSPState *rt = s->priv_data;
1896 RTSPStream *rtsp_st;
1897 int n, i, ret, tcp_fd, timeout_cnt = 0;
1899 struct pollfd *p = rt->p;
1900 int *fds = NULL, fdsnum, fdsidx;
1903 if (ff_check_interrupt(&s->interrupt_callback))
1904 return AVERROR_EXIT;
1905 if (wait_end && wait_end - av_gettime_relative() < 0)
1906 return AVERROR(EAGAIN);
1909 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1910 p[max_p].fd = tcp_fd;
1911 p[max_p++].events = POLLIN;
1915 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1916 rtsp_st = rt->rtsp_streams[i];
1917 if (rtsp_st->rtp_handle) {
1918 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1920 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1924 av_log(s, AV_LOG_ERROR,
1925 "Number of fds %d not supported\n", fdsnum);
1926 return AVERROR_INVALIDDATA;
1928 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1929 p[max_p].fd = fds[fdsidx];
1930 p[max_p++].events = POLLIN;
1935 n = poll(p, max_p, POLL_TIMEOUT_MS);
1937 int j = 1 - (tcp_fd == -1);
1939 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1940 rtsp_st = rt->rtsp_streams[i];
1941 if (rtsp_st->rtp_handle) {
1942 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1943 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1945 *prtsp_st = rtsp_st;
1952 #if CONFIG_RTSP_DEMUXER
1953 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1954 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1955 if (rt->state == RTSP_STATE_STREAMING) {
1956 if (!ff_rtsp_parse_streaming_commands(s))
1959 av_log(s, AV_LOG_WARNING,
1960 "Unable to answer to TEARDOWN\n");
1964 RTSPMessageHeader reply;
1965 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1968 /* XXX: parse message */
1969 if (rt->state != RTSP_STATE_STREAMING)
1974 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1975 return AVERROR(ETIMEDOUT);
1976 } else if (n < 0 && errno != EINTR)
1977 return AVERROR(errno);
1981 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1982 const uint8_t *buf, int len)
1984 RTSPState *rt = s->priv_data;
1988 if (rt->nb_rtsp_streams == 1) {
1989 *rtsp_st = rt->rtsp_streams[0];
1992 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1993 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1995 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1996 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1999 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
2000 *rtsp_st = rt->rtsp_streams[i];
2007 av_log(s, AV_LOG_WARNING,
2008 "Unable to pick stream for packet - SSRC not known for "
2010 return AVERROR(EAGAIN);
2013 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2014 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
2015 *rtsp_st = rt->rtsp_streams[i];
2021 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
2022 return AVERROR(EAGAIN);
2025 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
2027 RTSPState *rt = s->priv_data;
2029 RTSPStream *rtsp_st, *first_queue_st = NULL;
2030 int64_t wait_end = 0;
2032 if (rt->nb_byes == rt->nb_rtsp_streams)
2035 /* get next frames from the same RTP packet */
2036 if (rt->cur_transport_priv) {
2037 if (rt->transport == RTSP_TRANSPORT_RDT) {
2038 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2039 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2040 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2041 } else if (CONFIG_RTPDEC && rt->ts) {
2042 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2044 rt->recvbuf_pos += ret;
2045 ret = rt->recvbuf_pos < rt->recvbuf_len;
2050 rt->cur_transport_priv = NULL;
2052 } else if (ret == 1) {
2055 rt->cur_transport_priv = NULL;
2059 if (rt->transport == RTSP_TRANSPORT_RTP) {
2061 int64_t first_queue_time = 0;
2062 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2063 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2067 queue_time = ff_rtp_queued_packet_time(rtpctx);
2068 if (queue_time && (queue_time - first_queue_time < 0 ||
2069 !first_queue_time)) {
2070 first_queue_time = queue_time;
2071 first_queue_st = rt->rtsp_streams[i];
2074 if (first_queue_time) {
2075 wait_end = first_queue_time + s->max_delay;
2078 first_queue_st = NULL;
2082 /* read next RTP packet */
2084 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2086 return AVERROR(ENOMEM);
2089 switch(rt->lower_transport) {
2091 #if CONFIG_RTSP_DEMUXER
2092 case RTSP_LOWER_TRANSPORT_TCP:
2093 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2096 case RTSP_LOWER_TRANSPORT_UDP:
2097 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2098 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2099 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2100 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2102 case RTSP_LOWER_TRANSPORT_CUSTOM:
2103 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2104 wait_end && wait_end < av_gettime_relative())
2105 len = AVERROR(EAGAIN);
2107 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2108 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2109 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2110 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2113 if (len == AVERROR(EAGAIN) && first_queue_st &&
2114 rt->transport == RTSP_TRANSPORT_RTP) {
2115 rtsp_st = first_queue_st;
2116 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2123 if (rt->transport == RTSP_TRANSPORT_RDT) {
2124 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2125 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2126 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2127 if (rtsp_st->feedback) {
2128 AVIOContext *pb = NULL;
2129 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2131 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2134 /* Either bad packet, or a RTCP packet. Check if the
2135 * first_rtcp_ntp_time field was initialized. */
2136 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2137 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2138 /* first_rtcp_ntp_time has been initialized for this stream,
2139 * copy the same value to all other uninitialized streams,
2140 * in order to map their timestamp origin to the same ntp time
2143 AVStream *st = NULL;
2144 if (rtsp_st->stream_index >= 0)
2145 st = s->streams[rtsp_st->stream_index];
2146 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2147 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2148 AVStream *st2 = NULL;
2149 if (rt->rtsp_streams[i]->stream_index >= 0)
2150 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2151 if (rtpctx2 && st && st2 &&
2152 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2153 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2154 rtpctx2->rtcp_ts_offset = av_rescale_q(
2155 rtpctx->rtcp_ts_offset, st->time_base,
2160 if (ret == -RTCP_BYE) {
2163 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2164 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2166 if (rt->nb_byes == rt->nb_rtsp_streams)
2170 } else if (CONFIG_RTPDEC && rt->ts) {
2171 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2174 rt->recvbuf_len = len;
2175 rt->recvbuf_pos = ret;
2176 rt->cur_transport_priv = rt->ts;
2183 return AVERROR_INVALIDDATA;
2189 /* more packets may follow, so we save the RTP context */
2190 rt->cur_transport_priv = rtsp_st->transport_priv;
2194 #endif /* CONFIG_RTPDEC */
2196 #if CONFIG_SDP_DEMUXER
2197 static int sdp_probe(AVProbeData *p1)
2199 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2201 /* we look for a line beginning "c=IN IP" */
2202 while (p < p_end && *p != '\0') {
2203 if (p + sizeof("c=IN IP") - 1 < p_end &&
2204 av_strstart(p, "c=IN IP", NULL))
2205 return AVPROBE_SCORE_EXTENSION;
2207 while (p < p_end - 1 && *p != '\n') p++;
2216 static void append_source_addrs(char *buf, int size, const char *name,
2217 int count, struct RTSPSource **addrs)
2222 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2223 for (i = 1; i < count; i++)
2224 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2227 static int sdp_read_header(AVFormatContext *s)
2229 RTSPState *rt = s->priv_data;
2230 RTSPStream *rtsp_st;
2235 if (!ff_network_init())
2236 return AVERROR(EIO);
2238 if (s->max_delay < 0) /* Not set by the caller */
2239 s->max_delay = DEFAULT_REORDERING_DELAY;
2240 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2241 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2243 /* read the whole sdp file */
2244 /* XXX: better loading */
2245 content = av_malloc(SDP_MAX_SIZE);
2247 return AVERROR(ENOMEM);
2248 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2251 return AVERROR_INVALIDDATA;
2253 content[size] ='\0';
2255 err = ff_sdp_parse(s, content);
2259 /* open each RTP stream */
2260 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2262 rtsp_st = rt->rtsp_streams[i];
2264 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2265 AVDictionary *opts = map_to_opts(rt);
2267 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2268 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2269 ff_url_join(url, sizeof(url), "rtp", NULL,
2270 namebuf, rtsp_st->sdp_port,
2271 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2272 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2273 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2274 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2276 append_source_addrs(url, sizeof(url), "sources",
2277 rtsp_st->nb_include_source_addrs,
2278 rtsp_st->include_source_addrs);
2279 append_source_addrs(url, sizeof(url), "block",
2280 rtsp_st->nb_exclude_source_addrs,
2281 rtsp_st->exclude_source_addrs);
2282 err = ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2283 &s->interrupt_callback, &opts);
2285 av_dict_free(&opts);
2288 err = AVERROR_INVALIDDATA;
2292 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2297 ff_rtsp_close_streams(s);
2302 static int sdp_read_close(AVFormatContext *s)
2304 ff_rtsp_close_streams(s);
2309 static const AVClass sdp_demuxer_class = {
2310 .class_name = "SDP demuxer",
2311 .item_name = av_default_item_name,
2312 .option = sdp_options,
2313 .version = LIBAVUTIL_VERSION_INT,
2316 AVInputFormat ff_sdp_demuxer = {
2318 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2319 .priv_data_size = sizeof(RTSPState),
2320 .read_probe = sdp_probe,
2321 .read_header = sdp_read_header,
2322 .read_packet = ff_rtsp_fetch_packet,
2323 .read_close = sdp_read_close,
2324 .priv_class = &sdp_demuxer_class,
2326 #endif /* CONFIG_SDP_DEMUXER */
2328 #if CONFIG_RTP_DEMUXER
2329 static int rtp_probe(AVProbeData *p)
2331 if (av_strstart(p->filename, "rtp:", NULL))
2332 return AVPROBE_SCORE_MAX;
2336 static int rtp_read_header(AVFormatContext *s)
2338 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2339 char host[500], sdp[500];
2341 URLContext* in = NULL;
2343 AVCodecContext codec = { 0 };
2344 struct sockaddr_storage addr;
2346 socklen_t addrlen = sizeof(addr);
2347 RTSPState *rt = s->priv_data;
2349 if (!ff_network_init())
2350 return AVERROR(EIO);
2352 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2353 &s->interrupt_callback, NULL);
2358 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2359 if (ret == AVERROR(EAGAIN))
2364 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2368 if ((recvbuf[0] & 0xc0) != 0x80) {
2369 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2374 if (RTP_PT_IS_RTCP(recvbuf[1]))
2377 payload_type = recvbuf[1] & 0x7f;
2380 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2384 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2385 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2386 "without an SDP file describing it\n",
2390 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2391 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2392 "properly you need an SDP file "
2396 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2397 NULL, 0, s->filename);
2399 snprintf(sdp, sizeof(sdp),
2400 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2401 addr.ss_family == AF_INET ? 4 : 6, host,
2402 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2403 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2404 port, payload_type);
2405 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2407 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2410 /* sdp_read_header initializes this again */
2413 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2415 ret = sdp_read_header(s);
2426 static const AVClass rtp_demuxer_class = {
2427 .class_name = "RTP demuxer",
2428 .item_name = av_default_item_name,
2429 .option = rtp_options,
2430 .version = LIBAVUTIL_VERSION_INT,
2433 AVInputFormat ff_rtp_demuxer = {
2435 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2436 .priv_data_size = sizeof(RTSPState),
2437 .read_probe = rtp_probe,
2438 .read_header = rtp_read_header,
2439 .read_packet = ff_rtsp_fetch_packet,
2440 .read_close = sdp_read_close,
2441 .flags = AVFMT_NOFILE,
2442 .priv_class = &rtp_demuxer_class,
2444 #endif /* CONFIG_RTP_DEMUXER */