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 char delayed_fmtp[2048];
300 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
301 struct RTSPSource ***dest, int *dest_count)
303 RTSPSource *rtsp_src, *rtsp_src2;
305 for (i = 0; i < count; i++) {
307 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
310 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
311 dynarray_add(dest, dest_count, rtsp_src2);
315 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
316 int payload_type, const char *line)
320 for (i = 0; i < rt->nb_rtsp_streams; i++) {
321 RTSPStream *rtsp_st = rt->rtsp_streams[i];
322 if (rtsp_st->sdp_payload_type == payload_type &&
323 rtsp_st->dynamic_handler &&
324 rtsp_st->dynamic_handler->parse_sdp_a_line) {
325 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
326 rtsp_st->dynamic_protocol_context, line);
331 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
332 int letter, const char *buf)
334 RTSPState *rt = s->priv_data;
335 char buf1[64], st_type[64];
337 enum AVMediaType codec_type;
341 RTSPSource *rtsp_src;
342 struct sockaddr_storage sdp_ip;
345 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
348 if (s1->skip_media && letter != 'm')
352 get_word(buf1, sizeof(buf1), &p);
353 if (strcmp(buf1, "IN") != 0)
355 get_word(buf1, sizeof(buf1), &p);
356 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
358 get_word_sep(buf1, sizeof(buf1), "/", &p);
359 if (get_sockaddr(buf1, &sdp_ip))
364 get_word_sep(buf1, sizeof(buf1), "/", &p);
367 if (s->nb_streams == 0) {
368 s1->default_ip = sdp_ip;
369 s1->default_ttl = ttl;
371 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
372 rtsp_st->sdp_ip = sdp_ip;
373 rtsp_st->sdp_ttl = ttl;
377 av_dict_set(&s->metadata, "title", p, 0);
380 if (s->nb_streams == 0) {
381 av_dict_set(&s->metadata, "comment", p, 0);
390 codec_type = AVMEDIA_TYPE_UNKNOWN;
391 get_word(st_type, sizeof(st_type), &p);
392 if (!strcmp(st_type, "audio")) {
393 codec_type = AVMEDIA_TYPE_AUDIO;
394 } else if (!strcmp(st_type, "video")) {
395 codec_type = AVMEDIA_TYPE_VIDEO;
396 } else if (!strcmp(st_type, "application")) {
397 codec_type = AVMEDIA_TYPE_DATA;
399 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
403 rtsp_st = av_mallocz(sizeof(RTSPStream));
406 rtsp_st->stream_index = -1;
407 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
409 rtsp_st->sdp_ip = s1->default_ip;
410 rtsp_st->sdp_ttl = s1->default_ttl;
412 copy_default_source_addrs(s1->default_include_source_addrs,
413 s1->nb_default_include_source_addrs,
414 &rtsp_st->include_source_addrs,
415 &rtsp_st->nb_include_source_addrs);
416 copy_default_source_addrs(s1->default_exclude_source_addrs,
417 s1->nb_default_exclude_source_addrs,
418 &rtsp_st->exclude_source_addrs,
419 &rtsp_st->nb_exclude_source_addrs);
421 get_word(buf1, sizeof(buf1), &p); /* port */
422 rtsp_st->sdp_port = atoi(buf1);
424 get_word(buf1, sizeof(buf1), &p); /* protocol */
425 if (!strcmp(buf1, "udp"))
426 rt->transport = RTSP_TRANSPORT_RAW;
427 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
428 rtsp_st->feedback = 1;
430 /* XXX: handle list of formats */
431 get_word(buf1, sizeof(buf1), &p); /* format list */
432 rtsp_st->sdp_payload_type = atoi(buf1);
434 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
435 /* no corresponding stream */
436 if (rt->transport == RTSP_TRANSPORT_RAW) {
437 if (CONFIG_RTPDEC && !rt->ts)
438 rt->ts = ff_mpegts_parse_open(s);
440 RTPDynamicProtocolHandler *handler;
441 handler = ff_rtp_handler_find_by_id(
442 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
443 init_rtp_handler(handler, rtsp_st, NULL);
444 if (handler && handler->init)
445 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
447 } else if (rt->server_type == RTSP_SERVER_WMS &&
448 codec_type == AVMEDIA_TYPE_DATA) {
449 /* RTX stream, a stream that carries all the other actual
450 * audio/video streams. Don't expose this to the callers. */
452 st = avformat_new_stream(s, NULL);
455 st->id = rt->nb_rtsp_streams - 1;
456 rtsp_st->stream_index = st->index;
457 st->codec->codec_type = codec_type;
458 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
459 RTPDynamicProtocolHandler *handler;
460 /* if standard payload type, we can find the codec right now */
461 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
462 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
463 st->codec->sample_rate > 0)
464 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
465 /* Even static payload types may need a custom depacketizer */
466 handler = ff_rtp_handler_find_by_id(
467 rtsp_st->sdp_payload_type, st->codec->codec_type);
468 init_rtp_handler(handler, rtsp_st, st->codec);
469 if (handler && handler->init)
470 handler->init(s, st->index,
471 rtsp_st->dynamic_protocol_context);
473 if (rt->default_lang[0])
474 av_dict_set(&st->metadata, "language", rt->default_lang, 0);
476 /* put a default control url */
477 av_strlcpy(rtsp_st->control_url, rt->control_uri,
478 sizeof(rtsp_st->control_url));
481 if (av_strstart(p, "control:", &p)) {
482 if (s->nb_streams == 0) {
483 if (!strncmp(p, "rtsp://", 7))
484 av_strlcpy(rt->control_uri, p,
485 sizeof(rt->control_uri));
488 /* get the control url */
489 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
491 /* XXX: may need to add full url resolution */
492 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
494 if (proto[0] == '\0') {
495 /* relative control URL */
496 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
497 av_strlcat(rtsp_st->control_url, "/",
498 sizeof(rtsp_st->control_url));
499 av_strlcat(rtsp_st->control_url, p,
500 sizeof(rtsp_st->control_url));
502 av_strlcpy(rtsp_st->control_url, p,
503 sizeof(rtsp_st->control_url));
505 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
506 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
507 get_word(buf1, sizeof(buf1), &p);
508 payload_type = atoi(buf1);
509 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
510 if (rtsp_st->stream_index >= 0) {
511 st = s->streams[rtsp_st->stream_index];
512 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
516 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
518 } else if (av_strstart(p, "fmtp:", &p) ||
519 av_strstart(p, "framesize:", &p)) {
520 // let dynamic protocol handlers have a stab at the line.
521 get_word(buf1, sizeof(buf1), &p);
522 payload_type = atoi(buf1);
523 if (s1->seen_rtpmap) {
524 parse_fmtp(s, rt, payload_type, buf);
527 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
529 } else if (av_strstart(p, "range:", &p)) {
532 // this is so that seeking on a streamed file can work.
533 rtsp_parse_range_npt(p, &start, &end);
534 s->start_time = start;
535 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
536 s->duration = (end == AV_NOPTS_VALUE) ?
537 AV_NOPTS_VALUE : end - start;
538 } else if (av_strstart(p, "lang:", &p)) {
539 if (s->nb_streams > 0) {
540 get_word(buf1, sizeof(buf1), &p);
541 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
542 if (rtsp_st->stream_index >= 0) {
543 st = s->streams[rtsp_st->stream_index];
544 av_dict_set(&st->metadata, "language", buf1, 0);
547 get_word(rt->default_lang, sizeof(rt->default_lang), &p);
548 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
550 rt->transport = RTSP_TRANSPORT_RDT;
551 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
553 st = s->streams[s->nb_streams - 1];
554 st->codec->sample_rate = atoi(p);
555 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
557 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
558 get_word(buf1, sizeof(buf1), &p); // ignore tag
559 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
560 p += strspn(p, SPACE_CHARS);
561 if (av_strstart(p, "inline:", &p))
562 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
563 } else if (av_strstart(p, "source-filter:", &p)) {
565 get_word(buf1, sizeof(buf1), &p);
566 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
568 exclude = !strcmp(buf1, "excl");
570 get_word(buf1, sizeof(buf1), &p);
571 if (strcmp(buf1, "IN") != 0)
573 get_word(buf1, sizeof(buf1), &p);
574 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
576 // not checking that the destination address actually matches or is wildcard
577 get_word(buf1, sizeof(buf1), &p);
580 rtsp_src = av_mallocz(sizeof(*rtsp_src));
583 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
585 if (s->nb_streams == 0) {
586 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
588 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
589 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
592 if (s->nb_streams == 0) {
593 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
595 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
596 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
601 if (rt->server_type == RTSP_SERVER_WMS)
602 ff_wms_parse_sdp_a_line(s, p);
603 if (s->nb_streams > 0) {
604 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
606 if (rt->server_type == RTSP_SERVER_REAL)
607 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
609 if (rtsp_st->dynamic_handler &&
610 rtsp_st->dynamic_handler->parse_sdp_a_line)
611 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
612 rtsp_st->stream_index,
613 rtsp_st->dynamic_protocol_context, buf);
620 int ff_sdp_parse(AVFormatContext *s, const char *content)
622 RTSPState *rt = s->priv_data;
625 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
626 * contain long SDP lines containing complete ASF Headers (several
627 * kB) or arrays of MDPR (RM stream descriptor) headers plus
628 * "rulebooks" describing their properties. Therefore, the SDP line
631 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
632 * in rtpdec_xiph.c. */
634 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
638 p += strspn(p, SPACE_CHARS);
646 /* get the content */
648 while (*p != '\n' && *p != '\r' && *p != '\0') {
649 if ((q - buf) < sizeof(buf) - 1)
654 sdp_parse_line(s, s1, letter, buf);
656 while (*p != '\n' && *p != '\0')
662 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
663 av_free(s1->default_include_source_addrs[i]);
664 av_freep(&s1->default_include_source_addrs);
665 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
666 av_free(s1->default_exclude_source_addrs[i]);
667 av_freep(&s1->default_exclude_source_addrs);
669 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
670 if (!rt->p) return AVERROR(ENOMEM);
673 #endif /* CONFIG_RTPDEC */
675 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
677 RTSPState *rt = s->priv_data;
680 for (i = 0; i < rt->nb_rtsp_streams; i++) {
681 RTSPStream *rtsp_st = rt->rtsp_streams[i];
684 if (rtsp_st->transport_priv) {
686 AVFormatContext *rtpctx = rtsp_st->transport_priv;
687 av_write_trailer(rtpctx);
688 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
690 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
691 ff_rtsp_tcp_write_packet(s, rtsp_st);
692 avio_close_dyn_buf(rtpctx->pb, &ptr);
695 avio_close(rtpctx->pb);
697 avformat_free_context(rtpctx);
698 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
699 ff_rdt_parse_close(rtsp_st->transport_priv);
700 else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP)
701 ff_rtp_parse_close(rtsp_st->transport_priv);
703 rtsp_st->transport_priv = NULL;
704 if (rtsp_st->rtp_handle)
705 ffurl_close(rtsp_st->rtp_handle);
706 rtsp_st->rtp_handle = NULL;
710 /* close and free RTSP streams */
711 void ff_rtsp_close_streams(AVFormatContext *s)
713 RTSPState *rt = s->priv_data;
717 ff_rtsp_undo_setup(s, 0);
718 for (i = 0; i < rt->nb_rtsp_streams; i++) {
719 rtsp_st = rt->rtsp_streams[i];
721 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
722 rtsp_st->dynamic_handler->free(
723 rtsp_st->dynamic_protocol_context);
724 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
725 av_free(rtsp_st->include_source_addrs[j]);
726 av_freep(&rtsp_st->include_source_addrs);
727 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
728 av_free(rtsp_st->exclude_source_addrs[j]);
729 av_freep(&rtsp_st->exclude_source_addrs);
734 av_free(rt->rtsp_streams);
736 avformat_close_input(&rt->asf_ctx);
738 if (CONFIG_RTPDEC && rt->ts)
739 ff_mpegts_parse_close(rt->ts);
741 av_free(rt->recvbuf);
744 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
746 RTSPState *rt = s->priv_data;
748 int reordering_queue_size = rt->reordering_queue_size;
749 if (reordering_queue_size < 0) {
750 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
751 reordering_queue_size = 0;
753 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
756 /* open the RTP context */
757 if (rtsp_st->stream_index >= 0)
758 st = s->streams[rtsp_st->stream_index];
760 s->ctx_flags |= AVFMTCTX_NOHEADER;
762 if (CONFIG_RTSP_MUXER && s->oformat) {
763 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
764 s, st, rtsp_st->rtp_handle,
765 RTSP_TCP_MAX_PACKET_SIZE,
766 rtsp_st->stream_index);
767 /* Ownership of rtp_handle is passed to the rtp mux context */
768 rtsp_st->rtp_handle = NULL;
771 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
772 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
773 return 0; // Don't need to open any parser here
774 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RDT)
775 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
776 rtsp_st->dynamic_protocol_context,
777 rtsp_st->dynamic_handler);
778 else if (CONFIG_RTPDEC)
779 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
780 rtsp_st->sdp_payload_type,
781 reordering_queue_size);
783 if (!rtsp_st->transport_priv) {
784 return AVERROR(ENOMEM);
785 } else if (CONFIG_RTPDEC && rt->transport == RTSP_TRANSPORT_RTP) {
786 if (rtsp_st->dynamic_handler) {
787 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
788 rtsp_st->dynamic_protocol_context,
789 rtsp_st->dynamic_handler);
791 if (rtsp_st->crypto_suite[0])
792 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
793 rtsp_st->crypto_suite,
794 rtsp_st->crypto_params);
800 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
801 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
808 q += strspn(q, SPACE_CHARS);
809 v = strtol(q, &p, 10);
813 v = strtol(p, &p, 10);
822 /* XXX: only one transport specification is parsed */
823 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
825 char transport_protocol[16];
827 char lower_transport[16];
829 RTSPTransportField *th;
832 reply->nb_transports = 0;
835 p += strspn(p, SPACE_CHARS);
839 th = &reply->transports[reply->nb_transports];
841 get_word_sep(transport_protocol, sizeof(transport_protocol),
843 if (!av_strcasecmp (transport_protocol, "rtp")) {
844 get_word_sep(profile, sizeof(profile), "/;,", &p);
845 lower_transport[0] = '\0';
846 /* rtp/avp/<protocol> */
848 get_word_sep(lower_transport, sizeof(lower_transport),
851 th->transport = RTSP_TRANSPORT_RTP;
852 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
853 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
854 /* x-pn-tng/<protocol> */
855 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
857 th->transport = RTSP_TRANSPORT_RDT;
858 } else if (!av_strcasecmp(transport_protocol, "raw")) {
859 get_word_sep(profile, sizeof(profile), "/;,", &p);
860 lower_transport[0] = '\0';
861 /* raw/raw/<protocol> */
863 get_word_sep(lower_transport, sizeof(lower_transport),
866 th->transport = RTSP_TRANSPORT_RAW;
868 if (!av_strcasecmp(lower_transport, "TCP"))
869 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
871 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
875 /* get each parameter */
876 while (*p != '\0' && *p != ',') {
877 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
878 if (!strcmp(parameter, "port")) {
881 rtsp_parse_range(&th->port_min, &th->port_max, &p);
883 } else if (!strcmp(parameter, "client_port")) {
886 rtsp_parse_range(&th->client_port_min,
887 &th->client_port_max, &p);
889 } else if (!strcmp(parameter, "server_port")) {
892 rtsp_parse_range(&th->server_port_min,
893 &th->server_port_max, &p);
895 } else if (!strcmp(parameter, "interleaved")) {
898 rtsp_parse_range(&th->interleaved_min,
899 &th->interleaved_max, &p);
901 } else if (!strcmp(parameter, "multicast")) {
902 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
903 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
904 } else if (!strcmp(parameter, "ttl")) {
908 th->ttl = strtol(p, &end, 10);
911 } else if (!strcmp(parameter, "destination")) {
914 get_word_sep(buf, sizeof(buf), ";,", &p);
915 get_sockaddr(buf, &th->destination);
917 } else if (!strcmp(parameter, "source")) {
920 get_word_sep(buf, sizeof(buf), ";,", &p);
921 av_strlcpy(th->source, buf, sizeof(th->source));
923 } else if (!strcmp(parameter, "mode")) {
926 get_word_sep(buf, sizeof(buf), ";, ", &p);
927 if (!strcmp(buf, "record") ||
928 !strcmp(buf, "receive"))
933 while (*p != ';' && *p != '\0' && *p != ',')
941 reply->nb_transports++;
945 static void handle_rtp_info(RTSPState *rt, const char *url,
946 uint32_t seq, uint32_t rtptime)
949 if (!rtptime || !url[0])
951 if (rt->transport != RTSP_TRANSPORT_RTP)
953 for (i = 0; i < rt->nb_rtsp_streams; i++) {
954 RTSPStream *rtsp_st = rt->rtsp_streams[i];
955 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
958 if (!strcmp(rtsp_st->control_url, url)) {
959 rtpctx->base_timestamp = rtptime;
965 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
968 char key[20], value[1024], url[1024] = "";
969 uint32_t seq = 0, rtptime = 0;
972 p += strspn(p, SPACE_CHARS);
975 get_word_sep(key, sizeof(key), "=", &p);
979 get_word_sep(value, sizeof(value), ";, ", &p);
981 if (!strcmp(key, "url"))
982 av_strlcpy(url, value, sizeof(url));
983 else if (!strcmp(key, "seq"))
984 seq = strtoul(value, NULL, 10);
985 else if (!strcmp(key, "rtptime"))
986 rtptime = strtoul(value, NULL, 10);
988 handle_rtp_info(rt, url, seq, rtptime);
997 handle_rtp_info(rt, url, seq, rtptime);
1000 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
1001 RTSPState *rt, const char *method)
1005 /* NOTE: we do case independent match for broken servers */
1007 if (av_stristart(p, "Session:", &p)) {
1009 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1010 if (av_stristart(p, ";timeout=", &p) &&
1011 (t = strtol(p, NULL, 10)) > 0) {
1014 } else if (av_stristart(p, "Content-Length:", &p)) {
1015 reply->content_length = strtol(p, NULL, 10);
1016 } else if (av_stristart(p, "Transport:", &p)) {
1017 rtsp_parse_transport(reply, p);
1018 } else if (av_stristart(p, "CSeq:", &p)) {
1019 reply->seq = strtol(p, NULL, 10);
1020 } else if (av_stristart(p, "Range:", &p)) {
1021 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1022 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1023 p += strspn(p, SPACE_CHARS);
1024 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1025 } else if (av_stristart(p, "Server:", &p)) {
1026 p += strspn(p, SPACE_CHARS);
1027 av_strlcpy(reply->server, p, sizeof(reply->server));
1028 } else if (av_stristart(p, "Notice:", &p) ||
1029 av_stristart(p, "X-Notice:", &p)) {
1030 reply->notice = strtol(p, NULL, 10);
1031 } else if (av_stristart(p, "Location:", &p)) {
1032 p += strspn(p, SPACE_CHARS);
1033 av_strlcpy(reply->location, p , sizeof(reply->location));
1034 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1035 p += strspn(p, SPACE_CHARS);
1036 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1037 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1038 p += strspn(p, SPACE_CHARS);
1039 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1040 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1041 p += strspn(p, SPACE_CHARS);
1042 if (method && !strcmp(method, "DESCRIBE"))
1043 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1044 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1045 p += strspn(p, SPACE_CHARS);
1046 if (method && !strcmp(method, "PLAY"))
1047 rtsp_parse_rtp_info(rt, p);
1048 } else if (av_stristart(p, "Public:", &p) && rt) {
1049 if (strstr(p, "GET_PARAMETER") &&
1050 method && !strcmp(method, "OPTIONS"))
1051 rt->get_parameter_supported = 1;
1052 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1053 p += strspn(p, SPACE_CHARS);
1054 rt->accept_dynamic_rate = atoi(p);
1055 } else if (av_stristart(p, "Content-Type:", &p)) {
1056 p += strspn(p, SPACE_CHARS);
1057 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1061 /* skip a RTP/TCP interleaved packet */
1062 void ff_rtsp_skip_packet(AVFormatContext *s)
1064 RTSPState *rt = s->priv_data;
1068 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1071 len = AV_RB16(buf + 1);
1073 av_dlog(s, "skipping RTP packet len=%d\n", len);
1078 if (len1 > sizeof(buf))
1080 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1087 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1088 unsigned char **content_ptr,
1089 int return_on_interleaved_data, const char *method)
1091 RTSPState *rt = s->priv_data;
1092 char buf[4096], buf1[1024], *q;
1095 int ret, content_length, line_count = 0, request = 0;
1096 unsigned char *content = NULL;
1102 memset(reply, 0, sizeof(*reply));
1104 /* parse reply (XXX: use buffers) */
1105 rt->last_reply[0] = '\0';
1109 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1110 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1116 /* XXX: only parse it if first char on line ? */
1117 if (return_on_interleaved_data) {
1120 ff_rtsp_skip_packet(s);
1121 } else if (ch != '\r') {
1122 if ((q - buf) < sizeof(buf) - 1)
1128 av_dlog(s, "line='%s'\n", buf);
1130 /* test if last line */
1134 if (line_count == 0) {
1135 /* get reply code */
1136 get_word(buf1, sizeof(buf1), &p);
1137 if (!strncmp(buf1, "RTSP/", 5)) {
1138 get_word(buf1, sizeof(buf1), &p);
1139 reply->status_code = atoi(buf1);
1140 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1142 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1143 get_word(buf1, sizeof(buf1), &p); // object
1147 ff_rtsp_parse_line(reply, p, rt, method);
1148 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1149 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1154 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1155 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1157 content_length = reply->content_length;
1158 if (content_length > 0) {
1159 /* leave some room for a trailing '\0' (useful for simple parsing) */
1160 content = av_malloc(content_length + 1);
1162 return AVERROR(ENOMEM);
1163 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1164 content[content_length] = '\0';
1167 *content_ptr = content;
1173 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1174 const char* ptr = buf;
1176 if (!strcmp(reply->reason, "OPTIONS")) {
1177 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1179 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1180 if (reply->session_id[0])
1181 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1184 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1186 av_strlcat(buf, "\r\n", sizeof(buf));
1188 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1189 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1192 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1194 rt->last_cmd_time = av_gettime_relative();
1195 /* Even if the request from the server had data, it is not the data
1196 * that the caller wants or expects. The memory could also be leaked
1197 * if the actual following reply has content data. */
1199 av_freep(content_ptr);
1200 /* If method is set, this is called from ff_rtsp_send_cmd,
1201 * where a reply to exactly this request is awaited. For
1202 * callers from within packet receiving, we just want to
1203 * return to the caller and go back to receiving packets. */
1209 if (rt->seq != reply->seq) {
1210 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1211 rt->seq, reply->seq);
1215 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1216 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1217 reply->notice == 2306 /* Continuous Feed Terminated */) {
1218 rt->state = RTSP_STATE_IDLE;
1219 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1220 return AVERROR(EIO); /* data or server error */
1221 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1222 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1223 return AVERROR(EPERM);
1229 * Send a command to the RTSP server without waiting for the reply.
1231 * @param s RTSP (de)muxer context
1232 * @param method the method for the request
1233 * @param url the target url for the request
1234 * @param headers extra header lines to include in the request
1235 * @param send_content if non-null, the data to send as request body content
1236 * @param send_content_length the length of the send_content data, or 0 if
1237 * send_content is null
1239 * @return zero if success, nonzero otherwise
1241 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1242 const char *method, const char *url,
1243 const char *headers,
1244 const unsigned char *send_content,
1245 int send_content_length)
1247 RTSPState *rt = s->priv_data;
1248 char buf[4096], *out_buf;
1249 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1251 /* Add in RTSP headers */
1254 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1256 av_strlcat(buf, headers, sizeof(buf));
1257 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1258 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
1259 if (rt->session_id[0] != '\0' && (!headers ||
1260 !strstr(headers, "\nIf-Match:"))) {
1261 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1264 char *str = ff_http_auth_create_response(&rt->auth_state,
1265 rt->auth, url, method);
1267 av_strlcat(buf, str, sizeof(buf));
1270 if (send_content_length > 0 && send_content)
1271 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1272 av_strlcat(buf, "\r\n", sizeof(buf));
1274 /* base64 encode rtsp if tunneling */
1275 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1276 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1277 out_buf = base64buf;
1280 av_dlog(s, "Sending:\n%s--\n", buf);
1282 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1283 if (send_content_length > 0 && send_content) {
1284 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1285 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1286 "with content data not supported\n");
1287 return AVERROR_PATCHWELCOME;
1289 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1291 rt->last_cmd_time = av_gettime_relative();
1296 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1297 const char *url, const char *headers)
1299 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1302 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1303 const char *headers, RTSPMessageHeader *reply,
1304 unsigned char **content_ptr)
1306 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1307 content_ptr, NULL, 0);
1310 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1311 const char *method, const char *url,
1313 RTSPMessageHeader *reply,
1314 unsigned char **content_ptr,
1315 const unsigned char *send_content,
1316 int send_content_length)
1318 RTSPState *rt = s->priv_data;
1319 HTTPAuthType cur_auth_type;
1320 int ret, attempts = 0;
1323 cur_auth_type = rt->auth_state.auth_type;
1324 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1326 send_content_length)))
1329 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1333 if (reply->status_code == 401 &&
1334 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1335 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1338 if (reply->status_code > 400){
1339 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1343 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1349 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1350 int lower_transport, const char *real_challenge)
1352 RTSPState *rt = s->priv_data;
1353 int rtx = 0, j, i, err, interleave = 0, port_off;
1354 RTSPStream *rtsp_st;
1355 RTSPMessageHeader reply1, *reply = &reply1;
1357 const char *trans_pref;
1359 if (rt->transport == RTSP_TRANSPORT_RDT)
1360 trans_pref = "x-pn-tng";
1361 else if (rt->transport == RTSP_TRANSPORT_RAW)
1362 trans_pref = "RAW/RAW";
1364 trans_pref = "RTP/AVP";
1366 /* default timeout: 1 minute */
1369 /* for each stream, make the setup request */
1370 /* XXX: we assume the same server is used for the control of each
1373 /* Choose a random starting offset within the first half of the
1374 * port range, to allow for a number of ports to try even if the offset
1375 * happens to be at the end of the random range. */
1376 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1377 /* even random offset */
1378 port_off -= port_off & 0x01;
1380 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1381 char transport[2048];
1384 * WMS serves all UDP data over a single connection, the RTX, which
1385 * isn't necessarily the first in the SDP but has to be the first
1386 * to be set up, else the second/third SETUP will fail with a 461.
1388 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1389 rt->server_type == RTSP_SERVER_WMS) {
1392 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1393 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1395 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1399 if (rtx == rt->nb_rtsp_streams)
1400 return -1; /* no RTX found */
1401 rtsp_st = rt->rtsp_streams[rtx];
1403 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1405 rtsp_st = rt->rtsp_streams[i];
1408 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1411 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1412 port = reply->transports[0].client_port_min;
1416 /* first try in specified port range */
1417 while (j <= rt->rtp_port_max) {
1418 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1419 "?localport=%d", j);
1420 /* we will use two ports per rtp stream (rtp and rtcp) */
1422 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1423 &s->interrupt_callback, NULL))
1427 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1432 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1434 snprintf(transport, sizeof(transport) - 1,
1435 "%s/UDP;", trans_pref);
1436 if (rt->server_type != RTSP_SERVER_REAL)
1437 av_strlcat(transport, "unicast;", sizeof(transport));
1438 av_strlcatf(transport, sizeof(transport),
1439 "client_port=%d", port);
1440 if (rt->transport == RTSP_TRANSPORT_RTP &&
1441 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1442 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1446 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1447 /* For WMS streams, the application streams are only used for
1448 * UDP. When trying to set it up for TCP streams, the server
1449 * will return an error. Therefore, we skip those streams. */
1450 if (rt->server_type == RTSP_SERVER_WMS &&
1451 (rtsp_st->stream_index < 0 ||
1452 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1455 snprintf(transport, sizeof(transport) - 1,
1456 "%s/TCP;", trans_pref);
1457 if (rt->transport != RTSP_TRANSPORT_RDT)
1458 av_strlcat(transport, "unicast;", sizeof(transport));
1459 av_strlcatf(transport, sizeof(transport),
1460 "interleaved=%d-%d",
1461 interleave, interleave + 1);
1465 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1466 snprintf(transport, sizeof(transport) - 1,
1467 "%s/UDP;multicast", trans_pref);
1470 av_strlcat(transport, ";mode=record", sizeof(transport));
1471 } else if (rt->server_type == RTSP_SERVER_REAL ||
1472 rt->server_type == RTSP_SERVER_WMS)
1473 av_strlcat(transport, ";mode=play", sizeof(transport));
1474 snprintf(cmd, sizeof(cmd),
1475 "Transport: %s\r\n",
1477 if (rt->accept_dynamic_rate)
1478 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1479 if (CONFIG_RTPDEC && i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1480 char real_res[41], real_csum[9];
1481 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1483 av_strlcatf(cmd, sizeof(cmd),
1485 "RealChallenge2: %s, sd=%s\r\n",
1486 rt->session_id, real_res, real_csum);
1488 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1489 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1492 } else if (reply->status_code != RTSP_STATUS_OK ||
1493 reply->nb_transports != 1) {
1494 err = AVERROR_INVALIDDATA;
1498 /* XXX: same protocol for all streams is required */
1500 if (reply->transports[0].lower_transport != rt->lower_transport ||
1501 reply->transports[0].transport != rt->transport) {
1502 err = AVERROR_INVALIDDATA;
1506 rt->lower_transport = reply->transports[0].lower_transport;
1507 rt->transport = reply->transports[0].transport;
1510 /* Fail if the server responded with another lower transport mode
1511 * than what we requested. */
1512 if (reply->transports[0].lower_transport != lower_transport) {
1513 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1514 err = AVERROR_INVALIDDATA;
1518 switch(reply->transports[0].lower_transport) {
1519 case RTSP_LOWER_TRANSPORT_TCP:
1520 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1521 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1524 case RTSP_LOWER_TRANSPORT_UDP: {
1525 char url[1024], options[30] = "";
1526 const char *peer = host;
1528 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1529 av_strlcpy(options, "?connect=1", sizeof(options));
1530 /* Use source address if specified */
1531 if (reply->transports[0].source[0])
1532 peer = reply->transports[0].source;
1533 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1534 reply->transports[0].server_port_min, "%s", options);
1535 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1536 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1537 err = AVERROR_INVALIDDATA;
1540 /* Try to initialize the connection state in a
1541 * potential NAT router by sending dummy packets.
1542 * RTP/RTCP dummy packets are used for RDT, too.
1544 if (CONFIG_RTPDEC &&
1545 !(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1546 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1549 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1550 char url[1024], namebuf[50], optbuf[20] = "";
1551 struct sockaddr_storage addr;
1554 if (reply->transports[0].destination.ss_family) {
1555 addr = reply->transports[0].destination;
1556 port = reply->transports[0].port_min;
1557 ttl = reply->transports[0].ttl;
1559 addr = rtsp_st->sdp_ip;
1560 port = rtsp_st->sdp_port;
1561 ttl = rtsp_st->sdp_ttl;
1564 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1565 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1566 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1567 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1568 port, "%s", optbuf);
1569 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1570 &s->interrupt_callback, NULL) < 0) {
1571 err = AVERROR_INVALIDDATA;
1578 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1582 if (rt->nb_rtsp_streams && reply->timeout > 0)
1583 rt->timeout = reply->timeout;
1585 if (rt->server_type == RTSP_SERVER_REAL)
1586 rt->need_subscription = 1;
1591 ff_rtsp_undo_setup(s, 0);
1595 void ff_rtsp_close_connections(AVFormatContext *s)
1597 RTSPState *rt = s->priv_data;
1598 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1599 ffurl_close(rt->rtsp_hd);
1600 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1603 int ff_rtsp_connect(AVFormatContext *s)
1605 RTSPState *rt = s->priv_data;
1606 char proto[128], host[1024], path[1024];
1607 char tcpname[1024], cmd[2048], auth[128];
1608 const char *lower_rtsp_proto = "tcp";
1609 int port, err, tcp_fd;
1610 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1611 int lower_transport_mask = 0;
1612 int default_port = RTSP_DEFAULT_PORT;
1613 char real_challenge[64] = "";
1614 struct sockaddr_storage peer;
1615 socklen_t peer_len = sizeof(peer);
1617 if (rt->rtp_port_max < rt->rtp_port_min) {
1618 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1619 "than min port %d\n", rt->rtp_port_max,
1621 return AVERROR(EINVAL);
1624 if (!ff_network_init())
1625 return AVERROR(EIO);
1627 if (s->max_delay < 0) /* Not set by the caller */
1628 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1630 rt->control_transport = RTSP_MODE_PLAIN;
1631 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1632 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1633 rt->control_transport = RTSP_MODE_TUNNEL;
1635 /* Only pass through valid flags from here */
1636 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1639 /* extract hostname and port */
1640 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1641 host, sizeof(host), &port, path, sizeof(path), s->filename);
1643 if (!strcmp(proto, "rtsps")) {
1644 lower_rtsp_proto = "tls";
1645 default_port = RTSPS_DEFAULT_PORT;
1646 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1650 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1653 port = default_port;
1655 lower_transport_mask = rt->lower_transport_mask;
1657 if (!lower_transport_mask)
1658 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1661 /* Only UDP or TCP - UDP multicast isn't supported. */
1662 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1663 (1 << RTSP_LOWER_TRANSPORT_TCP);
1664 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1665 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1666 "only UDP and TCP are supported for output.\n");
1667 err = AVERROR(EINVAL);
1672 /* Construct the URI used in request; this is similar to s->filename,
1673 * but with authentication credentials removed and RTSP specific options
1675 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1676 host, port, "%s", path);
1678 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1679 /* set up initial handshake for tunneling */
1680 char httpname[1024];
1681 char sessioncookie[17];
1684 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1685 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1686 av_get_random_seed(), av_get_random_seed());
1689 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1690 &s->interrupt_callback) < 0) {
1695 /* generate GET headers */
1696 snprintf(headers, sizeof(headers),
1697 "x-sessioncookie: %s\r\n"
1698 "Accept: application/x-rtsp-tunnelled\r\n"
1699 "Pragma: no-cache\r\n"
1700 "Cache-Control: no-cache\r\n",
1702 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1704 /* complete the connection */
1705 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1711 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1712 &s->interrupt_callback) < 0 ) {
1717 /* generate POST headers */
1718 snprintf(headers, sizeof(headers),
1719 "x-sessioncookie: %s\r\n"
1720 "Content-Type: application/x-rtsp-tunnelled\r\n"
1721 "Pragma: no-cache\r\n"
1722 "Cache-Control: no-cache\r\n"
1723 "Content-Length: 32767\r\n"
1724 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1726 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1727 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1729 /* Initialize the authentication state for the POST session. The HTTP
1730 * protocol implementation doesn't properly handle multi-pass
1731 * authentication for POST requests, since it would require one of
1733 * - implementing Expect: 100-continue, which many HTTP servers
1734 * don't support anyway, even less the RTSP servers that do HTTP
1736 * - sending the whole POST data until getting a 401 reply specifying
1737 * what authentication method to use, then resending all that data
1738 * - waiting for potential 401 replies directly after sending the
1739 * POST header (waiting for some unspecified time)
1740 * Therefore, we copy the full auth state, which works for both basic
1741 * and digest. (For digest, we would have to synchronize the nonce
1742 * count variable between the two sessions, if we'd do more requests
1743 * with the original session, though.)
1745 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1747 /* complete the connection */
1748 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1753 /* open the tcp connection */
1754 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1756 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1757 &s->interrupt_callback, NULL) < 0) {
1761 rt->rtsp_hd_out = rt->rtsp_hd;
1765 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1770 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1771 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1772 NULL, 0, NI_NUMERICHOST);
1775 /* request options supported by the server; this also detects server
1777 for (rt->server_type = RTSP_SERVER_RTP;;) {
1779 if (rt->server_type == RTSP_SERVER_REAL)
1782 * The following entries are required for proper
1783 * streaming from a Realmedia server. They are
1784 * interdependent in some way although we currently
1785 * don't quite understand how. Values were copied
1786 * from mplayer SVN r23589.
1787 * ClientChallenge is a 16-byte ID in hex
1788 * CompanyID is a 16-byte ID in base64
1790 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1791 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1792 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1793 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1795 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1796 if (reply->status_code != RTSP_STATUS_OK) {
1797 err = AVERROR_INVALIDDATA;
1801 /* detect server type if not standard-compliant RTP */
1802 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1803 rt->server_type = RTSP_SERVER_REAL;
1805 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1806 rt->server_type = RTSP_SERVER_WMS;
1807 } else if (rt->server_type == RTSP_SERVER_REAL)
1808 strcpy(real_challenge, reply->real_challenge);
1812 if (CONFIG_RTSP_DEMUXER && s->iformat)
1813 err = ff_rtsp_setup_input_streams(s, reply);
1814 else if (CONFIG_RTSP_MUXER)
1815 err = ff_rtsp_setup_output_streams(s, host);
1820 int lower_transport = ff_log2_tab[lower_transport_mask &
1821 ~(lower_transport_mask - 1)];
1823 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1824 rt->server_type == RTSP_SERVER_REAL ?
1825 real_challenge : NULL);
1828 lower_transport_mask &= ~(1 << lower_transport);
1829 if (lower_transport_mask == 0 && err == 1) {
1830 err = AVERROR(EPROTONOSUPPORT);
1835 rt->lower_transport_mask = lower_transport_mask;
1836 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1837 rt->state = RTSP_STATE_IDLE;
1838 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1841 ff_rtsp_close_streams(s);
1842 ff_rtsp_close_connections(s);
1843 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1844 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1845 rt->session_id[0] = '\0';
1846 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1854 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1857 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1858 uint8_t *buf, int buf_size, int64_t wait_end)
1860 RTSPState *rt = s->priv_data;
1861 RTSPStream *rtsp_st;
1862 int n, i, ret, tcp_fd, timeout_cnt = 0;
1864 struct pollfd *p = rt->p;
1865 int *fds = NULL, fdsnum, fdsidx;
1868 if (ff_check_interrupt(&s->interrupt_callback))
1869 return AVERROR_EXIT;
1870 if (wait_end && wait_end - av_gettime_relative() < 0)
1871 return AVERROR(EAGAIN);
1874 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1875 p[max_p].fd = tcp_fd;
1876 p[max_p++].events = POLLIN;
1880 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1881 rtsp_st = rt->rtsp_streams[i];
1882 if (rtsp_st->rtp_handle) {
1883 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1885 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1889 av_log(s, AV_LOG_ERROR,
1890 "Number of fds %d not supported\n", fdsnum);
1891 return AVERROR_INVALIDDATA;
1893 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1894 p[max_p].fd = fds[fdsidx];
1895 p[max_p++].events = POLLIN;
1900 n = poll(p, max_p, POLL_TIMEOUT_MS);
1902 int j = 1 - (tcp_fd == -1);
1904 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1905 rtsp_st = rt->rtsp_streams[i];
1906 if (rtsp_st->rtp_handle) {
1907 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1908 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1910 *prtsp_st = rtsp_st;
1917 #if CONFIG_RTSP_DEMUXER
1918 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1919 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1920 if (rt->state == RTSP_STATE_STREAMING) {
1921 if (!ff_rtsp_parse_streaming_commands(s))
1924 av_log(s, AV_LOG_WARNING,
1925 "Unable to answer to TEARDOWN\n");
1929 RTSPMessageHeader reply;
1930 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1933 /* XXX: parse message */
1934 if (rt->state != RTSP_STATE_STREAMING)
1939 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1940 return AVERROR(ETIMEDOUT);
1941 } else if (n < 0 && errno != EINTR)
1942 return AVERROR(errno);
1946 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1947 const uint8_t *buf, int len)
1949 RTSPState *rt = s->priv_data;
1953 if (rt->nb_rtsp_streams == 1) {
1954 *rtsp_st = rt->rtsp_streams[0];
1957 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1958 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1960 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1961 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1964 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1965 *rtsp_st = rt->rtsp_streams[i];
1972 av_log(s, AV_LOG_WARNING,
1973 "Unable to pick stream for packet - SSRC not known for "
1975 return AVERROR(EAGAIN);
1978 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1979 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1980 *rtsp_st = rt->rtsp_streams[i];
1986 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1987 return AVERROR(EAGAIN);
1990 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1992 RTSPState *rt = s->priv_data;
1994 RTSPStream *rtsp_st, *first_queue_st = NULL;
1995 int64_t wait_end = 0;
1997 if (rt->nb_byes == rt->nb_rtsp_streams)
2000 /* get next frames from the same RTP packet */
2001 if (rt->cur_transport_priv) {
2002 if (rt->transport == RTSP_TRANSPORT_RDT) {
2003 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2004 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2005 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
2006 } else if (CONFIG_RTPDEC && rt->ts) {
2007 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
2009 rt->recvbuf_pos += ret;
2010 ret = rt->recvbuf_pos < rt->recvbuf_len;
2015 rt->cur_transport_priv = NULL;
2017 } else if (ret == 1) {
2020 rt->cur_transport_priv = NULL;
2024 if (rt->transport == RTSP_TRANSPORT_RTP) {
2026 int64_t first_queue_time = 0;
2027 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2028 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2032 queue_time = ff_rtp_queued_packet_time(rtpctx);
2033 if (queue_time && (queue_time - first_queue_time < 0 ||
2034 !first_queue_time)) {
2035 first_queue_time = queue_time;
2036 first_queue_st = rt->rtsp_streams[i];
2039 if (first_queue_time) {
2040 wait_end = first_queue_time + s->max_delay;
2043 first_queue_st = NULL;
2047 /* read next RTP packet */
2049 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2051 return AVERROR(ENOMEM);
2054 switch(rt->lower_transport) {
2056 #if CONFIG_RTSP_DEMUXER
2057 case RTSP_LOWER_TRANSPORT_TCP:
2058 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2061 case RTSP_LOWER_TRANSPORT_UDP:
2062 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2063 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2064 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2065 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2067 case RTSP_LOWER_TRANSPORT_CUSTOM:
2068 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2069 wait_end && wait_end < av_gettime_relative())
2070 len = AVERROR(EAGAIN);
2072 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2073 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2074 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2075 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2078 if (len == AVERROR(EAGAIN) && first_queue_st &&
2079 rt->transport == RTSP_TRANSPORT_RTP) {
2080 rtsp_st = first_queue_st;
2081 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2088 if (rt->transport == RTSP_TRANSPORT_RDT) {
2089 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2090 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2091 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2092 if (rtsp_st->feedback) {
2093 AVIOContext *pb = NULL;
2094 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2096 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2099 /* Either bad packet, or a RTCP packet. Check if the
2100 * first_rtcp_ntp_time field was initialized. */
2101 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2102 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2103 /* first_rtcp_ntp_time has been initialized for this stream,
2104 * copy the same value to all other uninitialized streams,
2105 * in order to map their timestamp origin to the same ntp time
2108 AVStream *st = NULL;
2109 if (rtsp_st->stream_index >= 0)
2110 st = s->streams[rtsp_st->stream_index];
2111 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2112 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2113 AVStream *st2 = NULL;
2114 if (rt->rtsp_streams[i]->stream_index >= 0)
2115 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2116 if (rtpctx2 && st && st2 &&
2117 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2118 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2119 rtpctx2->rtcp_ts_offset = av_rescale_q(
2120 rtpctx->rtcp_ts_offset, st->time_base,
2125 if (ret == -RTCP_BYE) {
2128 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2129 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2131 if (rt->nb_byes == rt->nb_rtsp_streams)
2135 } else if (CONFIG_RTPDEC && rt->ts) {
2136 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2139 rt->recvbuf_len = len;
2140 rt->recvbuf_pos = ret;
2141 rt->cur_transport_priv = rt->ts;
2148 return AVERROR_INVALIDDATA;
2154 /* more packets may follow, so we save the RTP context */
2155 rt->cur_transport_priv = rtsp_st->transport_priv;
2159 #endif /* CONFIG_RTPDEC */
2161 #if CONFIG_SDP_DEMUXER
2162 static int sdp_probe(AVProbeData *p1)
2164 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2166 /* we look for a line beginning "c=IN IP" */
2167 while (p < p_end && *p != '\0') {
2168 if (p + sizeof("c=IN IP") - 1 < p_end &&
2169 av_strstart(p, "c=IN IP", NULL))
2170 return AVPROBE_SCORE_EXTENSION;
2172 while (p < p_end - 1 && *p != '\n') p++;
2181 static void append_source_addrs(char *buf, int size, const char *name,
2182 int count, struct RTSPSource **addrs)
2187 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2188 for (i = 1; i < count; i++)
2189 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2192 static int sdp_read_header(AVFormatContext *s)
2194 RTSPState *rt = s->priv_data;
2195 RTSPStream *rtsp_st;
2200 if (!ff_network_init())
2201 return AVERROR(EIO);
2203 if (s->max_delay < 0) /* Not set by the caller */
2204 s->max_delay = DEFAULT_REORDERING_DELAY;
2205 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2206 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2208 /* read the whole sdp file */
2209 /* XXX: better loading */
2210 content = av_malloc(SDP_MAX_SIZE);
2211 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2214 return AVERROR_INVALIDDATA;
2216 content[size] ='\0';
2218 err = ff_sdp_parse(s, content);
2222 /* open each RTP stream */
2223 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2225 rtsp_st = rt->rtsp_streams[i];
2227 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2228 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2229 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2230 ff_url_join(url, sizeof(url), "rtp", NULL,
2231 namebuf, rtsp_st->sdp_port,
2232 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2233 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2234 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2235 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2237 append_source_addrs(url, sizeof(url), "sources",
2238 rtsp_st->nb_include_source_addrs,
2239 rtsp_st->include_source_addrs);
2240 append_source_addrs(url, sizeof(url), "block",
2241 rtsp_st->nb_exclude_source_addrs,
2242 rtsp_st->exclude_source_addrs);
2243 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2244 &s->interrupt_callback, NULL) < 0) {
2245 err = AVERROR_INVALIDDATA;
2249 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2254 ff_rtsp_close_streams(s);
2259 static int sdp_read_close(AVFormatContext *s)
2261 ff_rtsp_close_streams(s);
2266 static const AVClass sdp_demuxer_class = {
2267 .class_name = "SDP demuxer",
2268 .item_name = av_default_item_name,
2269 .option = sdp_options,
2270 .version = LIBAVUTIL_VERSION_INT,
2273 AVInputFormat ff_sdp_demuxer = {
2275 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2276 .priv_data_size = sizeof(RTSPState),
2277 .read_probe = sdp_probe,
2278 .read_header = sdp_read_header,
2279 .read_packet = ff_rtsp_fetch_packet,
2280 .read_close = sdp_read_close,
2281 .priv_class = &sdp_demuxer_class,
2283 #endif /* CONFIG_SDP_DEMUXER */
2285 #if CONFIG_RTP_DEMUXER
2286 static int rtp_probe(AVProbeData *p)
2288 if (av_strstart(p->filename, "rtp:", NULL))
2289 return AVPROBE_SCORE_MAX;
2293 static int rtp_read_header(AVFormatContext *s)
2295 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2296 char host[500], sdp[500];
2298 URLContext* in = NULL;
2300 AVCodecContext codec = { 0 };
2301 struct sockaddr_storage addr;
2303 socklen_t addrlen = sizeof(addr);
2304 RTSPState *rt = s->priv_data;
2306 if (!ff_network_init())
2307 return AVERROR(EIO);
2309 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2310 &s->interrupt_callback, NULL);
2315 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2316 if (ret == AVERROR(EAGAIN))
2321 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2325 if ((recvbuf[0] & 0xc0) != 0x80) {
2326 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2331 if (RTP_PT_IS_RTCP(recvbuf[1]))
2334 payload_type = recvbuf[1] & 0x7f;
2337 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2341 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2342 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2343 "without an SDP file describing it\n",
2347 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2348 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2349 "properly you need an SDP file "
2353 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2354 NULL, 0, s->filename);
2356 snprintf(sdp, sizeof(sdp),
2357 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2358 addr.ss_family == AF_INET ? 4 : 6, host,
2359 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2360 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2361 port, payload_type);
2362 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2364 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2367 /* sdp_read_header initializes this again */
2370 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2372 ret = sdp_read_header(s);
2383 static const AVClass rtp_demuxer_class = {
2384 .class_name = "RTP demuxer",
2385 .item_name = av_default_item_name,
2386 .option = rtp_options,
2387 .version = LIBAVUTIL_VERSION_INT,
2390 AVInputFormat ff_rtp_demuxer = {
2392 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2393 .priv_data_size = sizeof(RTSPState),
2394 .read_probe = rtp_probe,
2395 .read_header = rtp_read_header,
2396 .read_packet = ff_rtsp_fetch_packet,
2397 .read_close = sdp_read_close,
2398 .flags = AVFMT_NOFILE,
2399 .priv_class = &rtp_demuxer_class,
2401 #endif /* CONFIG_RTP_DEMUXER */