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 RTSP_REORDERING_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 }
78 const AVOption ff_rtsp_options[] = {
79 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
88 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91 { "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 },
92 RTSP_REORDERING_OPTS(),
96 static const AVOption sdp_options[] = {
97 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
98 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
99 { "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" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
101 RTSP_REORDERING_OPTS(),
105 static const AVOption rtp_options[] = {
106 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
107 RTSP_REORDERING_OPTS(),
111 static void get_word_until_chars(char *buf, int buf_size,
112 const char *sep, const char **pp)
118 p += strspn(p, SPACE_CHARS);
120 while (!strchr(sep, *p) && *p != '\0') {
121 if ((q - buf) < buf_size - 1)
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
133 if (**pp == '/') (*pp)++;
134 get_word_until_chars(buf, buf_size, sep, pp);
137 static void get_word(char *buf, int buf_size, const char **pp)
139 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144 * Used for seeking in the rtp stream.
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
150 p += strspn(p, SPACE_CHARS);
151 if (!av_stristart(p, "npt=", &p))
154 *start = AV_NOPTS_VALUE;
155 *end = AV_NOPTS_VALUE;
157 get_word_sep(buf, sizeof(buf), "-", &p);
158 av_parse_time(start, buf, 1);
161 get_word_sep(buf, sizeof(buf), "-", &p);
162 av_parse_time(end, buf, 1);
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
168 struct addrinfo hints = { 0 }, *ai = NULL;
169 hints.ai_flags = AI_NUMERICHOST;
170 if (getaddrinfo(buf, NULL, &hints, &ai))
172 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179 RTSPStream *rtsp_st, AVCodecContext *codec)
184 codec->codec_id = handler->codec_id;
185 rtsp_st->dynamic_handler = handler;
186 if (handler->alloc) {
187 rtsp_st->dynamic_protocol_context = handler->alloc();
188 if (!rtsp_st->dynamic_protocol_context)
189 rtsp_st->dynamic_handler = NULL;
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195 AVStream *st, RTSPStream *rtsp_st,
196 int payload_type, const char *p)
198 AVCodecContext *codec = st->codec;
204 /* See if we can handle this kind of payload.
205 * The space should normally not be there but some Real streams or
206 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207 * have a trailing space. */
208 get_word_sep(buf, sizeof(buf), "/ ", &p);
209 if (payload_type < RTP_PT_PRIVATE) {
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
215 if (codec->codec_id == AV_CODEC_ID_NONE) {
216 RTPDynamicProtocolHandler *handler =
217 ff_rtp_handler_find_by_name(buf, codec->codec_type);
218 init_rtp_handler(handler, rtsp_st, codec);
219 /* If no dynamic handler was found, check with the list of standard
220 * allocated types, if such a stream for some reason happens to
221 * use a private payload type. This isn't handled in rtpdec.c, since
222 * the format name from the rtpmap line never is passed into rtpdec. */
223 if (!rtsp_st->dynamic_handler)
224 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
227 c = avcodec_find_decoder(codec->codec_id);
233 get_word_sep(buf, sizeof(buf), "/", &p);
235 switch (codec->codec_type) {
236 case AVMEDIA_TYPE_AUDIO:
237 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
238 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
239 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
241 codec->sample_rate = i;
242 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243 get_word_sep(buf, sizeof(buf), "/", &p);
248 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
250 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
253 case AVMEDIA_TYPE_VIDEO:
254 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
256 avpriv_set_pts_info(st, 32, 1, i);
261 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262 rtsp_st->dynamic_handler->init(s, st->index,
263 rtsp_st->dynamic_protocol_context);
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271 char *value, int value_size)
273 *p += strspn(*p, SPACE_CHARS);
275 get_word_sep(attr, attr_size, "=", p);
278 get_word_sep(value, value_size, ";", p);
286 typedef struct SDPParseState {
288 struct sockaddr_storage default_ip;
290 int skip_media; ///< set if an unknown m= line occurs
291 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
292 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
293 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
294 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
297 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
298 struct RTSPSource ***dest, int *dest_count)
300 RTSPSource *rtsp_src, *rtsp_src2;
302 for (i = 0; i < count; i++) {
304 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
307 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
308 dynarray_add(dest, dest_count, rtsp_src2);
312 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
313 int letter, const char *buf)
315 RTSPState *rt = s->priv_data;
316 char buf1[64], st_type[64];
318 enum AVMediaType codec_type;
322 RTSPSource *rtsp_src;
323 struct sockaddr_storage sdp_ip;
326 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
329 if (s1->skip_media && letter != 'm')
333 get_word(buf1, sizeof(buf1), &p);
334 if (strcmp(buf1, "IN") != 0)
336 get_word(buf1, sizeof(buf1), &p);
337 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
339 get_word_sep(buf1, sizeof(buf1), "/", &p);
340 if (get_sockaddr(buf1, &sdp_ip))
345 get_word_sep(buf1, sizeof(buf1), "/", &p);
348 if (s->nb_streams == 0) {
349 s1->default_ip = sdp_ip;
350 s1->default_ttl = ttl;
352 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
353 rtsp_st->sdp_ip = sdp_ip;
354 rtsp_st->sdp_ttl = ttl;
358 av_dict_set(&s->metadata, "title", p, 0);
361 if (s->nb_streams == 0) {
362 av_dict_set(&s->metadata, "comment", p, 0);
369 codec_type = AVMEDIA_TYPE_UNKNOWN;
370 get_word(st_type, sizeof(st_type), &p);
371 if (!strcmp(st_type, "audio")) {
372 codec_type = AVMEDIA_TYPE_AUDIO;
373 } else if (!strcmp(st_type, "video")) {
374 codec_type = AVMEDIA_TYPE_VIDEO;
375 } else if (!strcmp(st_type, "application")) {
376 codec_type = AVMEDIA_TYPE_DATA;
378 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
382 rtsp_st = av_mallocz(sizeof(RTSPStream));
385 rtsp_st->stream_index = -1;
386 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
388 rtsp_st->sdp_ip = s1->default_ip;
389 rtsp_st->sdp_ttl = s1->default_ttl;
391 copy_default_source_addrs(s1->default_include_source_addrs,
392 s1->nb_default_include_source_addrs,
393 &rtsp_st->include_source_addrs,
394 &rtsp_st->nb_include_source_addrs);
395 copy_default_source_addrs(s1->default_exclude_source_addrs,
396 s1->nb_default_exclude_source_addrs,
397 &rtsp_st->exclude_source_addrs,
398 &rtsp_st->nb_exclude_source_addrs);
400 get_word(buf1, sizeof(buf1), &p); /* port */
401 rtsp_st->sdp_port = atoi(buf1);
403 get_word(buf1, sizeof(buf1), &p); /* protocol */
404 if (!strcmp(buf1, "udp"))
405 rt->transport = RTSP_TRANSPORT_RAW;
406 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
407 rtsp_st->feedback = 1;
409 /* XXX: handle list of formats */
410 get_word(buf1, sizeof(buf1), &p); /* format list */
411 rtsp_st->sdp_payload_type = atoi(buf1);
413 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
414 /* no corresponding stream */
415 if (rt->transport == RTSP_TRANSPORT_RAW) {
416 if (!rt->ts && CONFIG_RTPDEC)
417 rt->ts = ff_mpegts_parse_open(s);
419 RTPDynamicProtocolHandler *handler;
420 handler = ff_rtp_handler_find_by_id(
421 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
422 init_rtp_handler(handler, rtsp_st, NULL);
423 if (handler && handler->init)
424 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
426 } else if (rt->server_type == RTSP_SERVER_WMS &&
427 codec_type == AVMEDIA_TYPE_DATA) {
428 /* RTX stream, a stream that carries all the other actual
429 * audio/video streams. Don't expose this to the callers. */
431 st = avformat_new_stream(s, NULL);
434 st->id = rt->nb_rtsp_streams - 1;
435 rtsp_st->stream_index = st->index;
436 st->codec->codec_type = codec_type;
437 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
438 RTPDynamicProtocolHandler *handler;
439 /* if standard payload type, we can find the codec right now */
440 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
441 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
442 st->codec->sample_rate > 0)
443 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
444 /* Even static payload types may need a custom depacketizer */
445 handler = ff_rtp_handler_find_by_id(
446 rtsp_st->sdp_payload_type, st->codec->codec_type);
447 init_rtp_handler(handler, rtsp_st, st->codec);
448 if (handler && handler->init)
449 handler->init(s, st->index,
450 rtsp_st->dynamic_protocol_context);
453 /* put a default control url */
454 av_strlcpy(rtsp_st->control_url, rt->control_uri,
455 sizeof(rtsp_st->control_url));
458 if (av_strstart(p, "control:", &p)) {
459 if (s->nb_streams == 0) {
460 if (!strncmp(p, "rtsp://", 7))
461 av_strlcpy(rt->control_uri, p,
462 sizeof(rt->control_uri));
465 /* get the control url */
466 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
468 /* XXX: may need to add full url resolution */
469 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
471 if (proto[0] == '\0') {
472 /* relative control URL */
473 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
474 av_strlcat(rtsp_st->control_url, "/",
475 sizeof(rtsp_st->control_url));
476 av_strlcat(rtsp_st->control_url, p,
477 sizeof(rtsp_st->control_url));
479 av_strlcpy(rtsp_st->control_url, p,
480 sizeof(rtsp_st->control_url));
482 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
483 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
484 get_word(buf1, sizeof(buf1), &p);
485 payload_type = atoi(buf1);
486 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
487 if (rtsp_st->stream_index >= 0) {
488 st = s->streams[rtsp_st->stream_index];
489 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
491 } else if (av_strstart(p, "fmtp:", &p) ||
492 av_strstart(p, "framesize:", &p)) {
493 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
494 // let dynamic protocol handlers have a stab at the line.
495 get_word(buf1, sizeof(buf1), &p);
496 payload_type = atoi(buf1);
497 for (i = 0; i < rt->nb_rtsp_streams; i++) {
498 rtsp_st = rt->rtsp_streams[i];
499 if (rtsp_st->sdp_payload_type == payload_type &&
500 rtsp_st->dynamic_handler &&
501 rtsp_st->dynamic_handler->parse_sdp_a_line)
502 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
503 rtsp_st->dynamic_protocol_context, buf);
505 } else if (av_strstart(p, "range:", &p)) {
508 // this is so that seeking on a streamed file can work.
509 rtsp_parse_range_npt(p, &start, &end);
510 s->start_time = start;
511 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
512 s->duration = (end == AV_NOPTS_VALUE) ?
513 AV_NOPTS_VALUE : end - start;
514 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
516 rt->transport = RTSP_TRANSPORT_RDT;
517 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
519 st = s->streams[s->nb_streams - 1];
520 st->codec->sample_rate = atoi(p);
521 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
523 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
524 get_word(buf1, sizeof(buf1), &p); // ignore tag
525 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
526 p += strspn(p, SPACE_CHARS);
527 if (av_strstart(p, "inline:", &p))
528 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
529 } else if (av_strstart(p, "source-filter:", &p)) {
531 get_word(buf1, sizeof(buf1), &p);
532 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
534 exclude = !strcmp(buf1, "excl");
536 get_word(buf1, sizeof(buf1), &p);
537 if (strcmp(buf1, "IN") != 0)
539 get_word(buf1, sizeof(buf1), &p);
540 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
542 // not checking that the destination address actually matches or is wildcard
543 get_word(buf1, sizeof(buf1), &p);
546 rtsp_src = av_mallocz(sizeof(*rtsp_src));
549 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
551 if (s->nb_streams == 0) {
552 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
554 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
555 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
558 if (s->nb_streams == 0) {
559 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
561 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
562 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
567 if (rt->server_type == RTSP_SERVER_WMS)
568 ff_wms_parse_sdp_a_line(s, p);
569 if (s->nb_streams > 0) {
570 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
572 if (rt->server_type == RTSP_SERVER_REAL)
573 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
575 if (rtsp_st->dynamic_handler &&
576 rtsp_st->dynamic_handler->parse_sdp_a_line)
577 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
578 rtsp_st->stream_index,
579 rtsp_st->dynamic_protocol_context, buf);
586 int ff_sdp_parse(AVFormatContext *s, const char *content)
588 RTSPState *rt = s->priv_data;
591 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
592 * contain long SDP lines containing complete ASF Headers (several
593 * kB) or arrays of MDPR (RM stream descriptor) headers plus
594 * "rulebooks" describing their properties. Therefore, the SDP line
597 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
598 * in rtpdec_xiph.c. */
600 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
604 p += strspn(p, SPACE_CHARS);
612 /* get the content */
614 while (*p != '\n' && *p != '\r' && *p != '\0') {
615 if ((q - buf) < sizeof(buf) - 1)
620 sdp_parse_line(s, s1, letter, buf);
622 while (*p != '\n' && *p != '\0')
628 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
629 av_free(s1->default_include_source_addrs[i]);
630 av_freep(&s1->default_include_source_addrs);
631 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
632 av_free(s1->default_exclude_source_addrs[i]);
633 av_freep(&s1->default_exclude_source_addrs);
635 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
636 if (!rt->p) return AVERROR(ENOMEM);
639 #endif /* CONFIG_RTPDEC */
641 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
643 RTSPState *rt = s->priv_data;
646 for (i = 0; i < rt->nb_rtsp_streams; i++) {
647 RTSPStream *rtsp_st = rt->rtsp_streams[i];
650 if (rtsp_st->transport_priv) {
652 AVFormatContext *rtpctx = rtsp_st->transport_priv;
653 av_write_trailer(rtpctx);
654 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
656 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
657 ff_rtsp_tcp_write_packet(s, rtsp_st);
658 avio_close_dyn_buf(rtpctx->pb, &ptr);
661 avio_close(rtpctx->pb);
663 avformat_free_context(rtpctx);
664 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
665 ff_rdt_parse_close(rtsp_st->transport_priv);
666 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
667 ff_rtp_parse_close(rtsp_st->transport_priv);
669 rtsp_st->transport_priv = NULL;
670 if (rtsp_st->rtp_handle)
671 ffurl_close(rtsp_st->rtp_handle);
672 rtsp_st->rtp_handle = NULL;
676 /* close and free RTSP streams */
677 void ff_rtsp_close_streams(AVFormatContext *s)
679 RTSPState *rt = s->priv_data;
683 ff_rtsp_undo_setup(s, 0);
684 for (i = 0; i < rt->nb_rtsp_streams; i++) {
685 rtsp_st = rt->rtsp_streams[i];
687 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
688 rtsp_st->dynamic_handler->free(
689 rtsp_st->dynamic_protocol_context);
690 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
691 av_free(rtsp_st->include_source_addrs[j]);
692 av_freep(&rtsp_st->include_source_addrs);
693 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
694 av_free(rtsp_st->exclude_source_addrs[j]);
695 av_freep(&rtsp_st->exclude_source_addrs);
700 av_free(rt->rtsp_streams);
702 avformat_close_input(&rt->asf_ctx);
704 if (rt->ts && CONFIG_RTPDEC)
705 ff_mpegts_parse_close(rt->ts);
707 av_free(rt->recvbuf);
710 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
712 RTSPState *rt = s->priv_data;
714 int reordering_queue_size = rt->reordering_queue_size;
715 if (reordering_queue_size < 0) {
716 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
717 reordering_queue_size = 0;
719 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
722 /* open the RTP context */
723 if (rtsp_st->stream_index >= 0)
724 st = s->streams[rtsp_st->stream_index];
726 s->ctx_flags |= AVFMTCTX_NOHEADER;
728 if (s->oformat && CONFIG_RTSP_MUXER) {
729 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
730 s, st, rtsp_st->rtp_handle,
731 RTSP_TCP_MAX_PACKET_SIZE,
732 rtsp_st->stream_index);
733 /* Ownership of rtp_handle is passed to the rtp mux context */
734 rtsp_st->rtp_handle = NULL;
737 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
738 return 0; // Don't need to open any parser here
739 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
740 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
741 rtsp_st->dynamic_protocol_context,
742 rtsp_st->dynamic_handler);
743 else if (CONFIG_RTPDEC)
744 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
745 rtsp_st->sdp_payload_type,
746 reordering_queue_size);
748 if (!rtsp_st->transport_priv) {
749 return AVERROR(ENOMEM);
750 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
751 if (rtsp_st->dynamic_handler) {
752 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
753 rtsp_st->dynamic_protocol_context,
754 rtsp_st->dynamic_handler);
756 if (rtsp_st->crypto_suite[0])
757 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
758 rtsp_st->crypto_suite,
759 rtsp_st->crypto_params);
765 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
766 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
773 q += strspn(q, SPACE_CHARS);
774 v = strtol(q, &p, 10);
778 v = strtol(p, &p, 10);
787 /* XXX: only one transport specification is parsed */
788 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
790 char transport_protocol[16];
792 char lower_transport[16];
794 RTSPTransportField *th;
797 reply->nb_transports = 0;
800 p += strspn(p, SPACE_CHARS);
804 th = &reply->transports[reply->nb_transports];
806 get_word_sep(transport_protocol, sizeof(transport_protocol),
808 if (!av_strcasecmp (transport_protocol, "rtp")) {
809 get_word_sep(profile, sizeof(profile), "/;,", &p);
810 lower_transport[0] = '\0';
811 /* rtp/avp/<protocol> */
813 get_word_sep(lower_transport, sizeof(lower_transport),
816 th->transport = RTSP_TRANSPORT_RTP;
817 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
818 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
819 /* x-pn-tng/<protocol> */
820 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
822 th->transport = RTSP_TRANSPORT_RDT;
823 } else if (!av_strcasecmp(transport_protocol, "raw")) {
824 get_word_sep(profile, sizeof(profile), "/;,", &p);
825 lower_transport[0] = '\0';
826 /* raw/raw/<protocol> */
828 get_word_sep(lower_transport, sizeof(lower_transport),
831 th->transport = RTSP_TRANSPORT_RAW;
833 if (!av_strcasecmp(lower_transport, "TCP"))
834 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
836 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
840 /* get each parameter */
841 while (*p != '\0' && *p != ',') {
842 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
843 if (!strcmp(parameter, "port")) {
846 rtsp_parse_range(&th->port_min, &th->port_max, &p);
848 } else if (!strcmp(parameter, "client_port")) {
851 rtsp_parse_range(&th->client_port_min,
852 &th->client_port_max, &p);
854 } else if (!strcmp(parameter, "server_port")) {
857 rtsp_parse_range(&th->server_port_min,
858 &th->server_port_max, &p);
860 } else if (!strcmp(parameter, "interleaved")) {
863 rtsp_parse_range(&th->interleaved_min,
864 &th->interleaved_max, &p);
866 } else if (!strcmp(parameter, "multicast")) {
867 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
868 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
869 } else if (!strcmp(parameter, "ttl")) {
873 th->ttl = strtol(p, &end, 10);
876 } else if (!strcmp(parameter, "destination")) {
879 get_word_sep(buf, sizeof(buf), ";,", &p);
880 get_sockaddr(buf, &th->destination);
882 } else if (!strcmp(parameter, "source")) {
885 get_word_sep(buf, sizeof(buf), ";,", &p);
886 av_strlcpy(th->source, buf, sizeof(th->source));
888 } else if (!strcmp(parameter, "mode")) {
891 get_word_sep(buf, sizeof(buf), ";, ", &p);
892 if (!strcmp(buf, "record") ||
893 !strcmp(buf, "receive"))
898 while (*p != ';' && *p != '\0' && *p != ',')
906 reply->nb_transports++;
910 static void handle_rtp_info(RTSPState *rt, const char *url,
911 uint32_t seq, uint32_t rtptime)
914 if (!rtptime || !url[0])
916 if (rt->transport != RTSP_TRANSPORT_RTP)
918 for (i = 0; i < rt->nb_rtsp_streams; i++) {
919 RTSPStream *rtsp_st = rt->rtsp_streams[i];
920 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
923 if (!strcmp(rtsp_st->control_url, url)) {
924 rtpctx->base_timestamp = rtptime;
930 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
933 char key[20], value[1024], url[1024] = "";
934 uint32_t seq = 0, rtptime = 0;
937 p += strspn(p, SPACE_CHARS);
940 get_word_sep(key, sizeof(key), "=", &p);
944 get_word_sep(value, sizeof(value), ";, ", &p);
946 if (!strcmp(key, "url"))
947 av_strlcpy(url, value, sizeof(url));
948 else if (!strcmp(key, "seq"))
949 seq = strtoul(value, NULL, 10);
950 else if (!strcmp(key, "rtptime"))
951 rtptime = strtoul(value, NULL, 10);
953 handle_rtp_info(rt, url, seq, rtptime);
962 handle_rtp_info(rt, url, seq, rtptime);
965 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
966 RTSPState *rt, const char *method)
970 /* NOTE: we do case independent match for broken servers */
972 if (av_stristart(p, "Session:", &p)) {
974 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
975 if (av_stristart(p, ";timeout=", &p) &&
976 (t = strtol(p, NULL, 10)) > 0) {
979 } else if (av_stristart(p, "Content-Length:", &p)) {
980 reply->content_length = strtol(p, NULL, 10);
981 } else if (av_stristart(p, "Transport:", &p)) {
982 rtsp_parse_transport(reply, p);
983 } else if (av_stristart(p, "CSeq:", &p)) {
984 reply->seq = strtol(p, NULL, 10);
985 } else if (av_stristart(p, "Range:", &p)) {
986 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
987 } else if (av_stristart(p, "RealChallenge1:", &p)) {
988 p += strspn(p, SPACE_CHARS);
989 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
990 } else if (av_stristart(p, "Server:", &p)) {
991 p += strspn(p, SPACE_CHARS);
992 av_strlcpy(reply->server, p, sizeof(reply->server));
993 } else if (av_stristart(p, "Notice:", &p) ||
994 av_stristart(p, "X-Notice:", &p)) {
995 reply->notice = strtol(p, NULL, 10);
996 } else if (av_stristart(p, "Location:", &p)) {
997 p += strspn(p, SPACE_CHARS);
998 av_strlcpy(reply->location, p , sizeof(reply->location));
999 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1000 p += strspn(p, SPACE_CHARS);
1001 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1002 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1003 p += strspn(p, SPACE_CHARS);
1004 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1005 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1006 p += strspn(p, SPACE_CHARS);
1007 if (method && !strcmp(method, "DESCRIBE"))
1008 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1009 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1010 p += strspn(p, SPACE_CHARS);
1011 if (method && !strcmp(method, "PLAY"))
1012 rtsp_parse_rtp_info(rt, p);
1013 } else if (av_stristart(p, "Public:", &p) && rt) {
1014 if (strstr(p, "GET_PARAMETER") &&
1015 method && !strcmp(method, "OPTIONS"))
1016 rt->get_parameter_supported = 1;
1017 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1018 p += strspn(p, SPACE_CHARS);
1019 rt->accept_dynamic_rate = atoi(p);
1020 } else if (av_stristart(p, "Content-Type:", &p)) {
1021 p += strspn(p, SPACE_CHARS);
1022 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1026 /* skip a RTP/TCP interleaved packet */
1027 void ff_rtsp_skip_packet(AVFormatContext *s)
1029 RTSPState *rt = s->priv_data;
1033 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1036 len = AV_RB16(buf + 1);
1038 av_dlog(s, "skipping RTP packet len=%d\n", len);
1043 if (len1 > sizeof(buf))
1045 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1052 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1053 unsigned char **content_ptr,
1054 int return_on_interleaved_data, const char *method)
1056 RTSPState *rt = s->priv_data;
1057 char buf[4096], buf1[1024], *q;
1060 int ret, content_length, line_count = 0, request = 0;
1061 unsigned char *content = NULL;
1067 memset(reply, 0, sizeof(*reply));
1069 /* parse reply (XXX: use buffers) */
1070 rt->last_reply[0] = '\0';
1074 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1075 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1081 /* XXX: only parse it if first char on line ? */
1082 if (return_on_interleaved_data) {
1085 ff_rtsp_skip_packet(s);
1086 } else if (ch != '\r') {
1087 if ((q - buf) < sizeof(buf) - 1)
1093 av_dlog(s, "line='%s'\n", buf);
1095 /* test if last line */
1099 if (line_count == 0) {
1100 /* get reply code */
1101 get_word(buf1, sizeof(buf1), &p);
1102 if (!strncmp(buf1, "RTSP/", 5)) {
1103 get_word(buf1, sizeof(buf1), &p);
1104 reply->status_code = atoi(buf1);
1105 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1107 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1108 get_word(buf1, sizeof(buf1), &p); // object
1112 ff_rtsp_parse_line(reply, p, rt, method);
1113 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1114 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1119 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1120 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1122 content_length = reply->content_length;
1123 if (content_length > 0) {
1124 /* leave some room for a trailing '\0' (useful for simple parsing) */
1125 content = av_malloc(content_length + 1);
1126 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1127 content[content_length] = '\0';
1130 *content_ptr = content;
1136 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1137 const char* ptr = buf;
1139 if (!strcmp(reply->reason, "OPTIONS")) {
1140 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1142 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1143 if (reply->session_id[0])
1144 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1147 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1149 av_strlcat(buf, "\r\n", sizeof(buf));
1151 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1152 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1155 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1157 rt->last_cmd_time = av_gettime();
1158 /* Even if the request from the server had data, it is not the data
1159 * that the caller wants or expects. The memory could also be leaked
1160 * if the actual following reply has content data. */
1162 av_freep(content_ptr);
1163 /* If method is set, this is called from ff_rtsp_send_cmd,
1164 * where a reply to exactly this request is awaited. For
1165 * callers from within packet receiving, we just want to
1166 * return to the caller and go back to receiving packets. */
1172 if (rt->seq != reply->seq) {
1173 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1174 rt->seq, reply->seq);
1178 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1179 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1180 reply->notice == 2306 /* Continuous Feed Terminated */) {
1181 rt->state = RTSP_STATE_IDLE;
1182 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1183 return AVERROR(EIO); /* data or server error */
1184 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1185 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1186 return AVERROR(EPERM);
1192 * Send a command to the RTSP server without waiting for the reply.
1194 * @param s RTSP (de)muxer context
1195 * @param method the method for the request
1196 * @param url the target url for the request
1197 * @param headers extra header lines to include in the request
1198 * @param send_content if non-null, the data to send as request body content
1199 * @param send_content_length the length of the send_content data, or 0 if
1200 * send_content is null
1202 * @return zero if success, nonzero otherwise
1204 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1205 const char *method, const char *url,
1206 const char *headers,
1207 const unsigned char *send_content,
1208 int send_content_length)
1210 RTSPState *rt = s->priv_data;
1211 char buf[4096], *out_buf;
1212 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1214 /* Add in RTSP headers */
1217 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1219 av_strlcat(buf, headers, sizeof(buf));
1220 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1221 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1222 if (rt->session_id[0] != '\0' && (!headers ||
1223 !strstr(headers, "\nIf-Match:"))) {
1224 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1227 char *str = ff_http_auth_create_response(&rt->auth_state,
1228 rt->auth, url, method);
1230 av_strlcat(buf, str, sizeof(buf));
1233 if (send_content_length > 0 && send_content)
1234 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1235 av_strlcat(buf, "\r\n", sizeof(buf));
1237 /* base64 encode rtsp if tunneling */
1238 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1239 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1240 out_buf = base64buf;
1243 av_dlog(s, "Sending:\n%s--\n", buf);
1245 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1246 if (send_content_length > 0 && send_content) {
1247 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1248 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1249 "with content data not supported\n");
1250 return AVERROR_PATCHWELCOME;
1252 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1254 rt->last_cmd_time = av_gettime();
1259 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1260 const char *url, const char *headers)
1262 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1265 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1266 const char *headers, RTSPMessageHeader *reply,
1267 unsigned char **content_ptr)
1269 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1270 content_ptr, NULL, 0);
1273 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1274 const char *method, const char *url,
1276 RTSPMessageHeader *reply,
1277 unsigned char **content_ptr,
1278 const unsigned char *send_content,
1279 int send_content_length)
1281 RTSPState *rt = s->priv_data;
1282 HTTPAuthType cur_auth_type;
1283 int ret, attempts = 0;
1286 cur_auth_type = rt->auth_state.auth_type;
1287 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1289 send_content_length)))
1292 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1296 if (reply->status_code == 401 &&
1297 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1298 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1301 if (reply->status_code > 400){
1302 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1306 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1312 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1313 int lower_transport, const char *real_challenge)
1315 RTSPState *rt = s->priv_data;
1316 int rtx = 0, j, i, err, interleave = 0, port_off;
1317 RTSPStream *rtsp_st;
1318 RTSPMessageHeader reply1, *reply = &reply1;
1320 const char *trans_pref;
1322 if (rt->transport == RTSP_TRANSPORT_RDT)
1323 trans_pref = "x-pn-tng";
1324 else if (rt->transport == RTSP_TRANSPORT_RAW)
1325 trans_pref = "RAW/RAW";
1327 trans_pref = "RTP/AVP";
1329 /* default timeout: 1 minute */
1332 /* for each stream, make the setup request */
1333 /* XXX: we assume the same server is used for the control of each
1336 /* Choose a random starting offset within the first half of the
1337 * port range, to allow for a number of ports to try even if the offset
1338 * happens to be at the end of the random range. */
1339 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1340 /* even random offset */
1341 port_off -= port_off & 0x01;
1343 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1344 char transport[2048];
1347 * WMS serves all UDP data over a single connection, the RTX, which
1348 * isn't necessarily the first in the SDP but has to be the first
1349 * to be set up, else the second/third SETUP will fail with a 461.
1351 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1352 rt->server_type == RTSP_SERVER_WMS) {
1355 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1356 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1358 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1362 if (rtx == rt->nb_rtsp_streams)
1363 return -1; /* no RTX found */
1364 rtsp_st = rt->rtsp_streams[rtx];
1366 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1368 rtsp_st = rt->rtsp_streams[i];
1371 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1374 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1375 port = reply->transports[0].client_port_min;
1379 /* first try in specified port range */
1380 while (j <= rt->rtp_port_max) {
1381 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1382 "?localport=%d", j);
1383 /* we will use two ports per rtp stream (rtp and rtcp) */
1385 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1386 &s->interrupt_callback, NULL))
1390 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1395 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1397 snprintf(transport, sizeof(transport) - 1,
1398 "%s/UDP;", trans_pref);
1399 if (rt->server_type != RTSP_SERVER_REAL)
1400 av_strlcat(transport, "unicast;", sizeof(transport));
1401 av_strlcatf(transport, sizeof(transport),
1402 "client_port=%d", port);
1403 if (rt->transport == RTSP_TRANSPORT_RTP &&
1404 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1405 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1409 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1410 /* For WMS streams, the application streams are only used for
1411 * UDP. When trying to set it up for TCP streams, the server
1412 * will return an error. Therefore, we skip those streams. */
1413 if (rt->server_type == RTSP_SERVER_WMS &&
1414 (rtsp_st->stream_index < 0 ||
1415 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1418 snprintf(transport, sizeof(transport) - 1,
1419 "%s/TCP;", trans_pref);
1420 if (rt->transport != RTSP_TRANSPORT_RDT)
1421 av_strlcat(transport, "unicast;", sizeof(transport));
1422 av_strlcatf(transport, sizeof(transport),
1423 "interleaved=%d-%d",
1424 interleave, interleave + 1);
1428 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1429 snprintf(transport, sizeof(transport) - 1,
1430 "%s/UDP;multicast", trans_pref);
1433 av_strlcat(transport, ";mode=record", sizeof(transport));
1434 } else if (rt->server_type == RTSP_SERVER_REAL ||
1435 rt->server_type == RTSP_SERVER_WMS)
1436 av_strlcat(transport, ";mode=play", sizeof(transport));
1437 snprintf(cmd, sizeof(cmd),
1438 "Transport: %s\r\n",
1440 if (rt->accept_dynamic_rate)
1441 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1442 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1443 char real_res[41], real_csum[9];
1444 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1446 av_strlcatf(cmd, sizeof(cmd),
1448 "RealChallenge2: %s, sd=%s\r\n",
1449 rt->session_id, real_res, real_csum);
1451 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1452 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1455 } else if (reply->status_code != RTSP_STATUS_OK ||
1456 reply->nb_transports != 1) {
1457 err = AVERROR_INVALIDDATA;
1461 /* XXX: same protocol for all streams is required */
1463 if (reply->transports[0].lower_transport != rt->lower_transport ||
1464 reply->transports[0].transport != rt->transport) {
1465 err = AVERROR_INVALIDDATA;
1469 rt->lower_transport = reply->transports[0].lower_transport;
1470 rt->transport = reply->transports[0].transport;
1473 /* Fail if the server responded with another lower transport mode
1474 * than what we requested. */
1475 if (reply->transports[0].lower_transport != lower_transport) {
1476 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1477 err = AVERROR_INVALIDDATA;
1481 switch(reply->transports[0].lower_transport) {
1482 case RTSP_LOWER_TRANSPORT_TCP:
1483 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1484 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1487 case RTSP_LOWER_TRANSPORT_UDP: {
1488 char url[1024], options[30] = "";
1489 const char *peer = host;
1491 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1492 av_strlcpy(options, "?connect=1", sizeof(options));
1493 /* Use source address if specified */
1494 if (reply->transports[0].source[0])
1495 peer = reply->transports[0].source;
1496 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1497 reply->transports[0].server_port_min, "%s", options);
1498 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1499 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1500 err = AVERROR_INVALIDDATA;
1503 /* Try to initialize the connection state in a
1504 * potential NAT router by sending dummy packets.
1505 * RTP/RTCP dummy packets are used for RDT, too.
1507 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1509 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1512 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1513 char url[1024], namebuf[50], optbuf[20] = "";
1514 struct sockaddr_storage addr;
1517 if (reply->transports[0].destination.ss_family) {
1518 addr = reply->transports[0].destination;
1519 port = reply->transports[0].port_min;
1520 ttl = reply->transports[0].ttl;
1522 addr = rtsp_st->sdp_ip;
1523 port = rtsp_st->sdp_port;
1524 ttl = rtsp_st->sdp_ttl;
1527 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1528 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1529 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1530 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1531 port, "%s", optbuf);
1532 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1533 &s->interrupt_callback, NULL) < 0) {
1534 err = AVERROR_INVALIDDATA;
1541 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1545 if (rt->nb_rtsp_streams && reply->timeout > 0)
1546 rt->timeout = reply->timeout;
1548 if (rt->server_type == RTSP_SERVER_REAL)
1549 rt->need_subscription = 1;
1554 ff_rtsp_undo_setup(s, 0);
1558 void ff_rtsp_close_connections(AVFormatContext *s)
1560 RTSPState *rt = s->priv_data;
1561 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1562 ffurl_close(rt->rtsp_hd);
1563 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1566 int ff_rtsp_connect(AVFormatContext *s)
1568 RTSPState *rt = s->priv_data;
1569 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1570 int port, err, tcp_fd;
1571 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1572 int lower_transport_mask = 0;
1573 char real_challenge[64] = "";
1574 struct sockaddr_storage peer;
1575 socklen_t peer_len = sizeof(peer);
1577 if (rt->rtp_port_max < rt->rtp_port_min) {
1578 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1579 "than min port %d\n", rt->rtp_port_max,
1581 return AVERROR(EINVAL);
1584 if (!ff_network_init())
1585 return AVERROR(EIO);
1587 if (s->max_delay < 0) /* Not set by the caller */
1588 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1590 rt->control_transport = RTSP_MODE_PLAIN;
1591 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1592 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1593 rt->control_transport = RTSP_MODE_TUNNEL;
1595 /* Only pass through valid flags from here */
1596 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1599 lower_transport_mask = rt->lower_transport_mask;
1600 /* extract hostname and port */
1601 av_url_split(NULL, 0, auth, sizeof(auth),
1602 host, sizeof(host), &port, path, sizeof(path), s->filename);
1604 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1607 port = RTSP_DEFAULT_PORT;
1609 if (!lower_transport_mask)
1610 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1613 /* Only UDP or TCP - UDP multicast isn't supported. */
1614 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1615 (1 << RTSP_LOWER_TRANSPORT_TCP);
1616 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1617 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1618 "only UDP and TCP are supported for output.\n");
1619 err = AVERROR(EINVAL);
1624 /* Construct the URI used in request; this is similar to s->filename,
1625 * but with authentication credentials removed and RTSP specific options
1627 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1628 host, port, "%s", path);
1630 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1631 /* set up initial handshake for tunneling */
1632 char httpname[1024];
1633 char sessioncookie[17];
1636 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1637 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1638 av_get_random_seed(), av_get_random_seed());
1641 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1642 &s->interrupt_callback) < 0) {
1647 /* generate GET headers */
1648 snprintf(headers, sizeof(headers),
1649 "x-sessioncookie: %s\r\n"
1650 "Accept: application/x-rtsp-tunnelled\r\n"
1651 "Pragma: no-cache\r\n"
1652 "Cache-Control: no-cache\r\n",
1654 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1656 /* complete the connection */
1657 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1663 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1664 &s->interrupt_callback) < 0 ) {
1669 /* generate POST headers */
1670 snprintf(headers, sizeof(headers),
1671 "x-sessioncookie: %s\r\n"
1672 "Content-Type: application/x-rtsp-tunnelled\r\n"
1673 "Pragma: no-cache\r\n"
1674 "Cache-Control: no-cache\r\n"
1675 "Content-Length: 32767\r\n"
1676 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1678 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1679 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1681 /* Initialize the authentication state for the POST session. The HTTP
1682 * protocol implementation doesn't properly handle multi-pass
1683 * authentication for POST requests, since it would require one of
1685 * - implementing Expect: 100-continue, which many HTTP servers
1686 * don't support anyway, even less the RTSP servers that do HTTP
1688 * - sending the whole POST data until getting a 401 reply specifying
1689 * what authentication method to use, then resending all that data
1690 * - waiting for potential 401 replies directly after sending the
1691 * POST header (waiting for some unspecified time)
1692 * Therefore, we copy the full auth state, which works for both basic
1693 * and digest. (For digest, we would have to synchronize the nonce
1694 * count variable between the two sessions, if we'd do more requests
1695 * with the original session, though.)
1697 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1699 /* complete the connection */
1700 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1705 /* open the tcp connection */
1706 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1707 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1708 &s->interrupt_callback, NULL) < 0) {
1712 rt->rtsp_hd_out = rt->rtsp_hd;
1716 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1717 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1718 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1719 NULL, 0, NI_NUMERICHOST);
1722 /* request options supported by the server; this also detects server
1724 for (rt->server_type = RTSP_SERVER_RTP;;) {
1726 if (rt->server_type == RTSP_SERVER_REAL)
1729 * The following entries are required for proper
1730 * streaming from a Realmedia server. They are
1731 * interdependent in some way although we currently
1732 * don't quite understand how. Values were copied
1733 * from mplayer SVN r23589.
1734 * ClientChallenge is a 16-byte ID in hex
1735 * CompanyID is a 16-byte ID in base64
1737 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1738 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1739 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1740 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1742 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1743 if (reply->status_code != RTSP_STATUS_OK) {
1744 err = AVERROR_INVALIDDATA;
1748 /* detect server type if not standard-compliant RTP */
1749 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1750 rt->server_type = RTSP_SERVER_REAL;
1752 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1753 rt->server_type = RTSP_SERVER_WMS;
1754 } else if (rt->server_type == RTSP_SERVER_REAL)
1755 strcpy(real_challenge, reply->real_challenge);
1759 if (s->iformat && CONFIG_RTSP_DEMUXER)
1760 err = ff_rtsp_setup_input_streams(s, reply);
1761 else if (CONFIG_RTSP_MUXER)
1762 err = ff_rtsp_setup_output_streams(s, host);
1767 int lower_transport = ff_log2_tab[lower_transport_mask &
1768 ~(lower_transport_mask - 1)];
1770 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1771 rt->server_type == RTSP_SERVER_REAL ?
1772 real_challenge : NULL);
1775 lower_transport_mask &= ~(1 << lower_transport);
1776 if (lower_transport_mask == 0 && err == 1) {
1777 err = AVERROR(EPROTONOSUPPORT);
1782 rt->lower_transport_mask = lower_transport_mask;
1783 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1784 rt->state = RTSP_STATE_IDLE;
1785 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1788 ff_rtsp_close_streams(s);
1789 ff_rtsp_close_connections(s);
1790 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1791 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1792 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1800 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1803 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1804 uint8_t *buf, int buf_size, int64_t wait_end)
1806 RTSPState *rt = s->priv_data;
1807 RTSPStream *rtsp_st;
1808 int n, i, ret, tcp_fd, timeout_cnt = 0;
1810 struct pollfd *p = rt->p;
1811 int *fds = NULL, fdsnum, fdsidx;
1814 if (ff_check_interrupt(&s->interrupt_callback))
1815 return AVERROR_EXIT;
1816 if (wait_end && wait_end - av_gettime() < 0)
1817 return AVERROR(EAGAIN);
1820 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1821 p[max_p].fd = tcp_fd;
1822 p[max_p++].events = POLLIN;
1826 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1827 rtsp_st = rt->rtsp_streams[i];
1828 if (rtsp_st->rtp_handle) {
1829 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1831 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1835 av_log(s, AV_LOG_ERROR,
1836 "Number of fds %d not supported\n", fdsnum);
1837 return AVERROR_INVALIDDATA;
1839 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1840 p[max_p].fd = fds[fdsidx];
1841 p[max_p++].events = POLLIN;
1846 n = poll(p, max_p, POLL_TIMEOUT_MS);
1848 int j = 1 - (tcp_fd == -1);
1850 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1851 rtsp_st = rt->rtsp_streams[i];
1852 if (rtsp_st->rtp_handle) {
1853 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1854 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1856 *prtsp_st = rtsp_st;
1863 #if CONFIG_RTSP_DEMUXER
1864 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1865 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1866 if (rt->state == RTSP_STATE_STREAMING) {
1867 if (!ff_rtsp_parse_streaming_commands(s))
1870 av_log(s, AV_LOG_WARNING,
1871 "Unable to answer to TEARDOWN\n");
1875 RTSPMessageHeader reply;
1876 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1879 /* XXX: parse message */
1880 if (rt->state != RTSP_STATE_STREAMING)
1885 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1886 return AVERROR(ETIMEDOUT);
1887 } else if (n < 0 && errno != EINTR)
1888 return AVERROR(errno);
1892 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1893 const uint8_t *buf, int len)
1895 RTSPState *rt = s->priv_data;
1899 if (rt->nb_rtsp_streams == 1) {
1900 *rtsp_st = rt->rtsp_streams[0];
1903 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1904 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1906 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1907 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1910 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1911 *rtsp_st = rt->rtsp_streams[i];
1918 av_log(s, AV_LOG_WARNING,
1919 "Unable to pick stream for packet - SSRC not known for "
1921 return AVERROR(EAGAIN);
1924 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1925 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1926 *rtsp_st = rt->rtsp_streams[i];
1932 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1933 return AVERROR(EAGAIN);
1936 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1938 RTSPState *rt = s->priv_data;
1940 RTSPStream *rtsp_st, *first_queue_st = NULL;
1941 int64_t wait_end = 0;
1943 if (rt->nb_byes == rt->nb_rtsp_streams)
1946 /* get next frames from the same RTP packet */
1947 if (rt->cur_transport_priv) {
1948 if (rt->transport == RTSP_TRANSPORT_RDT) {
1949 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1950 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1951 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1952 } else if (rt->ts && CONFIG_RTPDEC) {
1953 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1955 rt->recvbuf_pos += ret;
1956 ret = rt->recvbuf_pos < rt->recvbuf_len;
1961 rt->cur_transport_priv = NULL;
1963 } else if (ret == 1) {
1966 rt->cur_transport_priv = NULL;
1970 if (rt->transport == RTSP_TRANSPORT_RTP) {
1972 int64_t first_queue_time = 0;
1973 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1974 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1978 queue_time = ff_rtp_queued_packet_time(rtpctx);
1979 if (queue_time && (queue_time - first_queue_time < 0 ||
1980 !first_queue_time)) {
1981 first_queue_time = queue_time;
1982 first_queue_st = rt->rtsp_streams[i];
1985 if (first_queue_time) {
1986 wait_end = first_queue_time + s->max_delay;
1989 first_queue_st = NULL;
1993 /* read next RTP packet */
1995 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1997 return AVERROR(ENOMEM);
2000 switch(rt->lower_transport) {
2002 #if CONFIG_RTSP_DEMUXER
2003 case RTSP_LOWER_TRANSPORT_TCP:
2004 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2007 case RTSP_LOWER_TRANSPORT_UDP:
2008 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2009 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2010 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2011 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2013 case RTSP_LOWER_TRANSPORT_CUSTOM:
2014 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2015 wait_end && wait_end < av_gettime())
2016 len = AVERROR(EAGAIN);
2018 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2019 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2020 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2021 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2024 if (len == AVERROR(EAGAIN) && first_queue_st &&
2025 rt->transport == RTSP_TRANSPORT_RTP) {
2026 rtsp_st = first_queue_st;
2027 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2034 if (rt->transport == RTSP_TRANSPORT_RDT) {
2035 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2036 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2037 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2038 if (rtsp_st->feedback) {
2039 AVIOContext *pb = NULL;
2040 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2042 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2045 /* Either bad packet, or a RTCP packet. Check if the
2046 * first_rtcp_ntp_time field was initialized. */
2047 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2048 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2049 /* first_rtcp_ntp_time has been initialized for this stream,
2050 * copy the same value to all other uninitialized streams,
2051 * in order to map their timestamp origin to the same ntp time
2054 AVStream *st = NULL;
2055 if (rtsp_st->stream_index >= 0)
2056 st = s->streams[rtsp_st->stream_index];
2057 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2058 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2059 AVStream *st2 = NULL;
2060 if (rt->rtsp_streams[i]->stream_index >= 0)
2061 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2062 if (rtpctx2 && st && st2 &&
2063 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2064 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2065 rtpctx2->rtcp_ts_offset = av_rescale_q(
2066 rtpctx->rtcp_ts_offset, st->time_base,
2071 if (ret == -RTCP_BYE) {
2074 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2075 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2077 if (rt->nb_byes == rt->nb_rtsp_streams)
2081 } else if (rt->ts && CONFIG_RTPDEC) {
2082 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2085 rt->recvbuf_len = len;
2086 rt->recvbuf_pos = ret;
2087 rt->cur_transport_priv = rt->ts;
2094 return AVERROR_INVALIDDATA;
2100 /* more packets may follow, so we save the RTP context */
2101 rt->cur_transport_priv = rtsp_st->transport_priv;
2105 #endif /* CONFIG_RTPDEC */
2107 #if CONFIG_SDP_DEMUXER
2108 static int sdp_probe(AVProbeData *p1)
2110 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2112 /* we look for a line beginning "c=IN IP" */
2113 while (p < p_end && *p != '\0') {
2114 if (p + sizeof("c=IN IP") - 1 < p_end &&
2115 av_strstart(p, "c=IN IP", NULL))
2116 return AVPROBE_SCORE_EXTENSION;
2118 while (p < p_end - 1 && *p != '\n') p++;
2127 static void append_source_addrs(char *buf, int size, const char *name,
2128 int count, struct RTSPSource **addrs)
2133 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2134 for (i = 1; i < count; i++)
2135 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2138 static int sdp_read_header(AVFormatContext *s)
2140 RTSPState *rt = s->priv_data;
2141 RTSPStream *rtsp_st;
2146 if (!ff_network_init())
2147 return AVERROR(EIO);
2149 if (s->max_delay < 0) /* Not set by the caller */
2150 s->max_delay = DEFAULT_REORDERING_DELAY;
2151 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2152 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2154 /* read the whole sdp file */
2155 /* XXX: better loading */
2156 content = av_malloc(SDP_MAX_SIZE);
2157 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2160 return AVERROR_INVALIDDATA;
2162 content[size] ='\0';
2164 err = ff_sdp_parse(s, content);
2168 /* open each RTP stream */
2169 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2171 rtsp_st = rt->rtsp_streams[i];
2173 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2174 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2175 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2176 ff_url_join(url, sizeof(url), "rtp", NULL,
2177 namebuf, rtsp_st->sdp_port,
2178 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2179 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2180 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2181 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2183 append_source_addrs(url, sizeof(url), "sources",
2184 rtsp_st->nb_include_source_addrs,
2185 rtsp_st->include_source_addrs);
2186 append_source_addrs(url, sizeof(url), "block",
2187 rtsp_st->nb_exclude_source_addrs,
2188 rtsp_st->exclude_source_addrs);
2189 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2190 &s->interrupt_callback, NULL) < 0) {
2191 err = AVERROR_INVALIDDATA;
2195 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2200 ff_rtsp_close_streams(s);
2205 static int sdp_read_close(AVFormatContext *s)
2207 ff_rtsp_close_streams(s);
2212 static const AVClass sdp_demuxer_class = {
2213 .class_name = "SDP demuxer",
2214 .item_name = av_default_item_name,
2215 .option = sdp_options,
2216 .version = LIBAVUTIL_VERSION_INT,
2219 AVInputFormat ff_sdp_demuxer = {
2221 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2222 .priv_data_size = sizeof(RTSPState),
2223 .read_probe = sdp_probe,
2224 .read_header = sdp_read_header,
2225 .read_packet = ff_rtsp_fetch_packet,
2226 .read_close = sdp_read_close,
2227 .priv_class = &sdp_demuxer_class,
2229 #endif /* CONFIG_SDP_DEMUXER */
2231 #if CONFIG_RTP_DEMUXER
2232 static int rtp_probe(AVProbeData *p)
2234 if (av_strstart(p->filename, "rtp:", NULL))
2235 return AVPROBE_SCORE_MAX;
2239 static int rtp_read_header(AVFormatContext *s)
2241 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2242 char host[500], sdp[500];
2244 URLContext* in = NULL;
2246 AVCodecContext codec = { 0 };
2247 struct sockaddr_storage addr;
2249 socklen_t addrlen = sizeof(addr);
2250 RTSPState *rt = s->priv_data;
2252 if (!ff_network_init())
2253 return AVERROR(EIO);
2255 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2256 &s->interrupt_callback, NULL);
2261 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2262 if (ret == AVERROR(EAGAIN))
2267 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2271 if ((recvbuf[0] & 0xc0) != 0x80) {
2272 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2277 if (RTP_PT_IS_RTCP(recvbuf[1]))
2280 payload_type = recvbuf[1] & 0x7f;
2283 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2287 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2288 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2289 "without an SDP file describing it\n",
2293 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2294 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2295 "properly you need an SDP file "
2299 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2300 NULL, 0, s->filename);
2302 snprintf(sdp, sizeof(sdp),
2303 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2304 addr.ss_family == AF_INET ? 4 : 6, host,
2305 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2306 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2307 port, payload_type);
2308 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2310 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2313 /* sdp_read_header initializes this again */
2316 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2318 ret = sdp_read_header(s);
2329 static const AVClass rtp_demuxer_class = {
2330 .class_name = "RTP demuxer",
2331 .item_name = av_default_item_name,
2332 .option = rtp_options,
2333 .version = LIBAVUTIL_VERSION_INT,
2336 AVInputFormat ff_rtp_demuxer = {
2338 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2339 .priv_data_size = sizeof(RTSPState),
2340 .read_probe = rtp_probe,
2341 .read_header = rtp_read_header,
2342 .read_packet = ff_rtsp_fetch_packet,
2343 .read_close = sdp_read_close,
2344 .flags = AVFMT_NOFILE,
2345 .priv_class = &rtp_demuxer_class,
2347 #endif /* CONFIG_RTP_DEMUXER */