3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
47 #include "rtpdec_formats.h"
48 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 #define RTSP_REORDERING_OPTS() \
77 { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
79 const AVOption ff_rtsp_options[] = {
80 { "initial_pause", "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
81 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
82 { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
83 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
86 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
87 RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
88 { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
89 { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
91 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
94 { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
95 RTSP_REORDERING_OPTS(),
96 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
100 static const AVOption sdp_options[] = {
101 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
102 { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
103 { "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" },
104 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
105 RTSP_REORDERING_OPTS(),
109 static const AVOption rtp_options[] = {
110 RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
111 RTSP_REORDERING_OPTS(),
115 static void get_word_until_chars(char *buf, int buf_size,
116 const char *sep, const char **pp)
122 p += strspn(p, SPACE_CHARS);
124 while (!strchr(sep, *p) && *p != '\0') {
125 if ((q - buf) < buf_size - 1)
134 static void get_word_sep(char *buf, int buf_size, const char *sep,
137 if (**pp == '/') (*pp)++;
138 get_word_until_chars(buf, buf_size, sep, pp);
141 static void get_word(char *buf, int buf_size, const char **pp)
143 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
146 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
148 * Used for seeking in the rtp stream.
150 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
154 p += strspn(p, SPACE_CHARS);
155 if (!av_stristart(p, "npt=", &p))
158 *start = AV_NOPTS_VALUE;
159 *end = AV_NOPTS_VALUE;
161 get_word_sep(buf, sizeof(buf), "-", &p);
162 av_parse_time(start, buf, 1);
165 get_word_sep(buf, sizeof(buf), "-", &p);
166 av_parse_time(end, buf, 1);
170 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
172 struct addrinfo hints = { 0 }, *ai = NULL;
173 hints.ai_flags = AI_NUMERICHOST;
174 if (getaddrinfo(buf, NULL, &hints, &ai))
176 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
182 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
183 RTSPStream *rtsp_st, AVCodecContext *codec)
188 codec->codec_id = handler->codec_id;
189 rtsp_st->dynamic_handler = handler;
190 if (handler->alloc) {
191 rtsp_st->dynamic_protocol_context = handler->alloc();
192 if (!rtsp_st->dynamic_protocol_context)
193 rtsp_st->dynamic_handler = NULL;
197 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
198 static int sdp_parse_rtpmap(AVFormatContext *s,
199 AVStream *st, RTSPStream *rtsp_st,
200 int payload_type, const char *p)
202 AVCodecContext *codec = st->codec;
208 /* See if we can handle this kind of payload.
209 * The space should normally not be there but some Real streams or
210 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
211 * have a trailing space. */
212 get_word_sep(buf, sizeof(buf), "/ ", &p);
213 if (payload_type < RTP_PT_PRIVATE) {
214 /* We are in a standard case
215 * (from http://www.iana.org/assignments/rtp-parameters). */
216 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
219 if (codec->codec_id == AV_CODEC_ID_NONE) {
220 RTPDynamicProtocolHandler *handler =
221 ff_rtp_handler_find_by_name(buf, codec->codec_type);
222 init_rtp_handler(handler, rtsp_st, codec);
223 /* If no dynamic handler was found, check with the list of standard
224 * allocated types, if such a stream for some reason happens to
225 * use a private payload type. This isn't handled in rtpdec.c, since
226 * the format name from the rtpmap line never is passed into rtpdec. */
227 if (!rtsp_st->dynamic_handler)
228 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
231 c = avcodec_find_decoder(codec->codec_id);
237 get_word_sep(buf, sizeof(buf), "/", &p);
239 switch (codec->codec_type) {
240 case AVMEDIA_TYPE_AUDIO:
241 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
242 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
243 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
245 codec->sample_rate = i;
246 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
247 get_word_sep(buf, sizeof(buf), "/", &p);
252 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
254 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
257 case AVMEDIA_TYPE_VIDEO:
258 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
260 avpriv_set_pts_info(st, 32, 1, i);
265 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
266 rtsp_st->dynamic_handler->init(s, st->index,
267 rtsp_st->dynamic_protocol_context);
271 /* parse the attribute line from the fmtp a line of an sdp response. This
272 * is broken out as a function because it is used in rtp_h264.c, which is
274 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
275 char *value, int value_size)
277 *p += strspn(*p, SPACE_CHARS);
279 get_word_sep(attr, attr_size, "=", p);
282 get_word_sep(value, value_size, ";", p);
290 typedef struct SDPParseState {
292 struct sockaddr_storage default_ip;
294 int skip_media; ///< set if an unknown m= line occurs
295 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
296 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
297 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
298 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
301 char delayed_fmtp[2048];
304 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
305 struct RTSPSource ***dest, int *dest_count)
307 RTSPSource *rtsp_src, *rtsp_src2;
309 for (i = 0; i < count; i++) {
311 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
314 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
315 dynarray_add(dest, dest_count, rtsp_src2);
319 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
320 int payload_type, const char *line)
324 for (i = 0; i < rt->nb_rtsp_streams; i++) {
325 RTSPStream *rtsp_st = rt->rtsp_streams[i];
326 if (rtsp_st->sdp_payload_type == payload_type &&
327 rtsp_st->dynamic_handler &&
328 rtsp_st->dynamic_handler->parse_sdp_a_line) {
329 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
330 rtsp_st->dynamic_protocol_context, line);
335 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
336 int letter, const char *buf)
338 RTSPState *rt = s->priv_data;
339 char buf1[64], st_type[64];
341 enum AVMediaType codec_type;
345 RTSPSource *rtsp_src;
346 struct sockaddr_storage sdp_ip;
349 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
352 if (s1->skip_media && letter != 'm')
356 get_word(buf1, sizeof(buf1), &p);
357 if (strcmp(buf1, "IN") != 0)
359 get_word(buf1, sizeof(buf1), &p);
360 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
362 get_word_sep(buf1, sizeof(buf1), "/", &p);
363 if (get_sockaddr(buf1, &sdp_ip))
368 get_word_sep(buf1, sizeof(buf1), "/", &p);
371 if (s->nb_streams == 0) {
372 s1->default_ip = sdp_ip;
373 s1->default_ttl = ttl;
375 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
376 rtsp_st->sdp_ip = sdp_ip;
377 rtsp_st->sdp_ttl = ttl;
381 av_dict_set(&s->metadata, "title", p, 0);
384 if (s->nb_streams == 0) {
385 av_dict_set(&s->metadata, "comment", p, 0);
394 codec_type = AVMEDIA_TYPE_UNKNOWN;
395 get_word(st_type, sizeof(st_type), &p);
396 if (!strcmp(st_type, "audio")) {
397 codec_type = AVMEDIA_TYPE_AUDIO;
398 } else if (!strcmp(st_type, "video")) {
399 codec_type = AVMEDIA_TYPE_VIDEO;
400 } else if (!strcmp(st_type, "application")) {
401 codec_type = AVMEDIA_TYPE_DATA;
403 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
407 rtsp_st = av_mallocz(sizeof(RTSPStream));
410 rtsp_st->stream_index = -1;
411 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
413 rtsp_st->sdp_ip = s1->default_ip;
414 rtsp_st->sdp_ttl = s1->default_ttl;
416 copy_default_source_addrs(s1->default_include_source_addrs,
417 s1->nb_default_include_source_addrs,
418 &rtsp_st->include_source_addrs,
419 &rtsp_st->nb_include_source_addrs);
420 copy_default_source_addrs(s1->default_exclude_source_addrs,
421 s1->nb_default_exclude_source_addrs,
422 &rtsp_st->exclude_source_addrs,
423 &rtsp_st->nb_exclude_source_addrs);
425 get_word(buf1, sizeof(buf1), &p); /* port */
426 rtsp_st->sdp_port = atoi(buf1);
428 get_word(buf1, sizeof(buf1), &p); /* protocol */
429 if (!strcmp(buf1, "udp"))
430 rt->transport = RTSP_TRANSPORT_RAW;
431 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
432 rtsp_st->feedback = 1;
434 /* XXX: handle list of formats */
435 get_word(buf1, sizeof(buf1), &p); /* format list */
436 rtsp_st->sdp_payload_type = atoi(buf1);
438 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
439 /* no corresponding stream */
440 if (rt->transport == RTSP_TRANSPORT_RAW) {
441 if (!rt->ts && CONFIG_RTPDEC)
442 rt->ts = avpriv_mpegts_parse_open(s);
444 RTPDynamicProtocolHandler *handler;
445 handler = ff_rtp_handler_find_by_id(
446 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
447 init_rtp_handler(handler, rtsp_st, NULL);
448 if (handler && handler->init)
449 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
451 } else if (rt->server_type == RTSP_SERVER_WMS &&
452 codec_type == AVMEDIA_TYPE_DATA) {
453 /* RTX stream, a stream that carries all the other actual
454 * audio/video streams. Don't expose this to the callers. */
456 st = avformat_new_stream(s, NULL);
459 st->id = rt->nb_rtsp_streams - 1;
460 rtsp_st->stream_index = st->index;
461 st->codec->codec_type = codec_type;
462 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
463 RTPDynamicProtocolHandler *handler;
464 /* if standard payload type, we can find the codec right now */
465 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
466 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
467 st->codec->sample_rate > 0)
468 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
469 /* Even static payload types may need a custom depacketizer */
470 handler = ff_rtp_handler_find_by_id(
471 rtsp_st->sdp_payload_type, st->codec->codec_type);
472 init_rtp_handler(handler, rtsp_st, st->codec);
473 if (handler && handler->init)
474 handler->init(s, st->index,
475 rtsp_st->dynamic_protocol_context);
478 /* put a default control url */
479 av_strlcpy(rtsp_st->control_url, rt->control_uri,
480 sizeof(rtsp_st->control_url));
483 if (av_strstart(p, "control:", &p)) {
484 if (s->nb_streams == 0) {
485 if (!strncmp(p, "rtsp://", 7))
486 av_strlcpy(rt->control_uri, p,
487 sizeof(rt->control_uri));
490 /* get the control url */
491 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
493 /* XXX: may need to add full url resolution */
494 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
496 if (proto[0] == '\0') {
497 /* relative control URL */
498 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
499 av_strlcat(rtsp_st->control_url, "/",
500 sizeof(rtsp_st->control_url));
501 av_strlcat(rtsp_st->control_url, p,
502 sizeof(rtsp_st->control_url));
504 av_strlcpy(rtsp_st->control_url, p,
505 sizeof(rtsp_st->control_url));
507 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
508 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
509 get_word(buf1, sizeof(buf1), &p);
510 payload_type = atoi(buf1);
511 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
512 if (rtsp_st->stream_index >= 0) {
513 st = s->streams[rtsp_st->stream_index];
514 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
518 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
520 } else if (av_strstart(p, "fmtp:", &p) ||
521 av_strstart(p, "framesize:", &p)) {
522 // let dynamic protocol handlers have a stab at the line.
523 get_word(buf1, sizeof(buf1), &p);
524 payload_type = atoi(buf1);
525 if (s1->seen_rtpmap) {
526 parse_fmtp(s, rt, payload_type, buf);
529 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
531 } else if (av_strstart(p, "range:", &p)) {
534 // this is so that seeking on a streamed file can work.
535 rtsp_parse_range_npt(p, &start, &end);
536 s->start_time = start;
537 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
538 s->duration = (end == AV_NOPTS_VALUE) ?
539 AV_NOPTS_VALUE : end - start;
540 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
542 rt->transport = RTSP_TRANSPORT_RDT;
543 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
545 st = s->streams[s->nb_streams - 1];
546 st->codec->sample_rate = atoi(p);
547 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
549 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
550 get_word(buf1, sizeof(buf1), &p); // ignore tag
551 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
552 p += strspn(p, SPACE_CHARS);
553 if (av_strstart(p, "inline:", &p))
554 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
555 } else if (av_strstart(p, "source-filter:", &p)) {
557 get_word(buf1, sizeof(buf1), &p);
558 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
560 exclude = !strcmp(buf1, "excl");
562 get_word(buf1, sizeof(buf1), &p);
563 if (strcmp(buf1, "IN") != 0)
565 get_word(buf1, sizeof(buf1), &p);
566 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
568 // not checking that the destination address actually matches or is wildcard
569 get_word(buf1, sizeof(buf1), &p);
572 rtsp_src = av_mallocz(sizeof(*rtsp_src));
575 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
577 if (s->nb_streams == 0) {
578 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
580 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
581 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
584 if (s->nb_streams == 0) {
585 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
587 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
588 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
593 if (rt->server_type == RTSP_SERVER_WMS)
594 ff_wms_parse_sdp_a_line(s, p);
595 if (s->nb_streams > 0) {
596 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
598 if (rt->server_type == RTSP_SERVER_REAL)
599 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
601 if (rtsp_st->dynamic_handler &&
602 rtsp_st->dynamic_handler->parse_sdp_a_line)
603 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
604 rtsp_st->stream_index,
605 rtsp_st->dynamic_protocol_context, buf);
612 int ff_sdp_parse(AVFormatContext *s, const char *content)
614 RTSPState *rt = s->priv_data;
617 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
618 * contain long SDP lines containing complete ASF Headers (several
619 * kB) or arrays of MDPR (RM stream descriptor) headers plus
620 * "rulebooks" describing their properties. Therefore, the SDP line
623 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
624 * in rtpdec_xiph.c. */
626 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
630 p += strspn(p, SPACE_CHARS);
638 /* get the content */
640 while (*p != '\n' && *p != '\r' && *p != '\0') {
641 if ((q - buf) < sizeof(buf) - 1)
646 sdp_parse_line(s, s1, letter, buf);
648 while (*p != '\n' && *p != '\0')
654 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
655 av_free(s1->default_include_source_addrs[i]);
656 av_freep(&s1->default_include_source_addrs);
657 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
658 av_free(s1->default_exclude_source_addrs[i]);
659 av_freep(&s1->default_exclude_source_addrs);
661 rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);
662 if (!rt->p) return AVERROR(ENOMEM);
665 #endif /* CONFIG_RTPDEC */
667 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
669 RTSPState *rt = s->priv_data;
672 for (i = 0; i < rt->nb_rtsp_streams; i++) {
673 RTSPStream *rtsp_st = rt->rtsp_streams[i];
676 if (rtsp_st->transport_priv) {
678 AVFormatContext *rtpctx = rtsp_st->transport_priv;
679 av_write_trailer(rtpctx);
680 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
682 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
683 ff_rtsp_tcp_write_packet(s, rtsp_st);
684 avio_close_dyn_buf(rtpctx->pb, &ptr);
687 avio_close(rtpctx->pb);
689 avformat_free_context(rtpctx);
690 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
691 ff_rdt_parse_close(rtsp_st->transport_priv);
692 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
693 ff_rtp_parse_close(rtsp_st->transport_priv);
695 rtsp_st->transport_priv = NULL;
696 if (rtsp_st->rtp_handle)
697 ffurl_close(rtsp_st->rtp_handle);
698 rtsp_st->rtp_handle = NULL;
702 /* close and free RTSP streams */
703 void ff_rtsp_close_streams(AVFormatContext *s)
705 RTSPState *rt = s->priv_data;
709 ff_rtsp_undo_setup(s, 0);
710 for (i = 0; i < rt->nb_rtsp_streams; i++) {
711 rtsp_st = rt->rtsp_streams[i];
713 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
714 rtsp_st->dynamic_handler->free(
715 rtsp_st->dynamic_protocol_context);
716 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
717 av_free(rtsp_st->include_source_addrs[j]);
718 av_freep(&rtsp_st->include_source_addrs);
719 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
720 av_free(rtsp_st->exclude_source_addrs[j]);
721 av_freep(&rtsp_st->exclude_source_addrs);
726 av_free(rt->rtsp_streams);
728 avformat_close_input(&rt->asf_ctx);
730 if (rt->ts && CONFIG_RTPDEC)
731 avpriv_mpegts_parse_close(rt->ts);
733 av_free(rt->recvbuf);
736 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
738 RTSPState *rt = s->priv_data;
740 int reordering_queue_size = rt->reordering_queue_size;
741 if (reordering_queue_size < 0) {
742 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
743 reordering_queue_size = 0;
745 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
748 /* open the RTP context */
749 if (rtsp_st->stream_index >= 0)
750 st = s->streams[rtsp_st->stream_index];
752 s->ctx_flags |= AVFMTCTX_NOHEADER;
754 if (s->oformat && CONFIG_RTSP_MUXER) {
755 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
756 s, st, rtsp_st->rtp_handle,
757 RTSP_TCP_MAX_PACKET_SIZE,
758 rtsp_st->stream_index);
759 /* Ownership of rtp_handle is passed to the rtp mux context */
760 rtsp_st->rtp_handle = NULL;
763 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
764 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
765 return 0; // Don't need to open any parser here
766 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
767 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
768 rtsp_st->dynamic_protocol_context,
769 rtsp_st->dynamic_handler);
770 else if (CONFIG_RTPDEC)
771 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
772 rtsp_st->sdp_payload_type,
773 reordering_queue_size);
775 if (!rtsp_st->transport_priv) {
776 return AVERROR(ENOMEM);
777 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
778 if (rtsp_st->dynamic_handler) {
779 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
780 rtsp_st->dynamic_protocol_context,
781 rtsp_st->dynamic_handler);
783 if (rtsp_st->crypto_suite[0])
784 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
785 rtsp_st->crypto_suite,
786 rtsp_st->crypto_params);
792 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
793 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
800 q += strspn(q, SPACE_CHARS);
801 v = strtol(q, &p, 10);
805 v = strtol(p, &p, 10);
814 /* XXX: only one transport specification is parsed */
815 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
817 char transport_protocol[16];
819 char lower_transport[16];
821 RTSPTransportField *th;
824 reply->nb_transports = 0;
827 p += strspn(p, SPACE_CHARS);
831 th = &reply->transports[reply->nb_transports];
833 get_word_sep(transport_protocol, sizeof(transport_protocol),
835 if (!av_strcasecmp (transport_protocol, "rtp")) {
836 get_word_sep(profile, sizeof(profile), "/;,", &p);
837 lower_transport[0] = '\0';
838 /* rtp/avp/<protocol> */
840 get_word_sep(lower_transport, sizeof(lower_transport),
843 th->transport = RTSP_TRANSPORT_RTP;
844 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
845 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
846 /* x-pn-tng/<protocol> */
847 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
849 th->transport = RTSP_TRANSPORT_RDT;
850 } else if (!av_strcasecmp(transport_protocol, "raw")) {
851 get_word_sep(profile, sizeof(profile), "/;,", &p);
852 lower_transport[0] = '\0';
853 /* raw/raw/<protocol> */
855 get_word_sep(lower_transport, sizeof(lower_transport),
858 th->transport = RTSP_TRANSPORT_RAW;
860 if (!av_strcasecmp(lower_transport, "TCP"))
861 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
863 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
867 /* get each parameter */
868 while (*p != '\0' && *p != ',') {
869 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
870 if (!strcmp(parameter, "port")) {
873 rtsp_parse_range(&th->port_min, &th->port_max, &p);
875 } else if (!strcmp(parameter, "client_port")) {
878 rtsp_parse_range(&th->client_port_min,
879 &th->client_port_max, &p);
881 } else if (!strcmp(parameter, "server_port")) {
884 rtsp_parse_range(&th->server_port_min,
885 &th->server_port_max, &p);
887 } else if (!strcmp(parameter, "interleaved")) {
890 rtsp_parse_range(&th->interleaved_min,
891 &th->interleaved_max, &p);
893 } else if (!strcmp(parameter, "multicast")) {
894 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
895 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
896 } else if (!strcmp(parameter, "ttl")) {
900 th->ttl = strtol(p, &end, 10);
903 } else if (!strcmp(parameter, "destination")) {
906 get_word_sep(buf, sizeof(buf), ";,", &p);
907 get_sockaddr(buf, &th->destination);
909 } else if (!strcmp(parameter, "source")) {
912 get_word_sep(buf, sizeof(buf), ";,", &p);
913 av_strlcpy(th->source, buf, sizeof(th->source));
915 } else if (!strcmp(parameter, "mode")) {
918 get_word_sep(buf, sizeof(buf), ";, ", &p);
919 if (!strcmp(buf, "record") ||
920 !strcmp(buf, "receive"))
925 while (*p != ';' && *p != '\0' && *p != ',')
933 reply->nb_transports++;
937 static void handle_rtp_info(RTSPState *rt, const char *url,
938 uint32_t seq, uint32_t rtptime)
941 if (!rtptime || !url[0])
943 if (rt->transport != RTSP_TRANSPORT_RTP)
945 for (i = 0; i < rt->nb_rtsp_streams; i++) {
946 RTSPStream *rtsp_st = rt->rtsp_streams[i];
947 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
950 if (!strcmp(rtsp_st->control_url, url)) {
951 rtpctx->base_timestamp = rtptime;
957 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
960 char key[20], value[1024], url[1024] = "";
961 uint32_t seq = 0, rtptime = 0;
964 p += strspn(p, SPACE_CHARS);
967 get_word_sep(key, sizeof(key), "=", &p);
971 get_word_sep(value, sizeof(value), ";, ", &p);
973 if (!strcmp(key, "url"))
974 av_strlcpy(url, value, sizeof(url));
975 else if (!strcmp(key, "seq"))
976 seq = strtoul(value, NULL, 10);
977 else if (!strcmp(key, "rtptime"))
978 rtptime = strtoul(value, NULL, 10);
980 handle_rtp_info(rt, url, seq, rtptime);
989 handle_rtp_info(rt, url, seq, rtptime);
992 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
993 RTSPState *rt, const char *method)
997 /* NOTE: we do case independent match for broken servers */
999 if (av_stristart(p, "Session:", &p)) {
1001 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
1002 if (av_stristart(p, ";timeout=", &p) &&
1003 (t = strtol(p, NULL, 10)) > 0) {
1006 } else if (av_stristart(p, "Content-Length:", &p)) {
1007 reply->content_length = strtol(p, NULL, 10);
1008 } else if (av_stristart(p, "Transport:", &p)) {
1009 rtsp_parse_transport(reply, p);
1010 } else if (av_stristart(p, "CSeq:", &p)) {
1011 reply->seq = strtol(p, NULL, 10);
1012 } else if (av_stristart(p, "Range:", &p)) {
1013 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
1014 } else if (av_stristart(p, "RealChallenge1:", &p)) {
1015 p += strspn(p, SPACE_CHARS);
1016 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
1017 } else if (av_stristart(p, "Server:", &p)) {
1018 p += strspn(p, SPACE_CHARS);
1019 av_strlcpy(reply->server, p, sizeof(reply->server));
1020 } else if (av_stristart(p, "Notice:", &p) ||
1021 av_stristart(p, "X-Notice:", &p)) {
1022 reply->notice = strtol(p, NULL, 10);
1023 } else if (av_stristart(p, "Location:", &p)) {
1024 p += strspn(p, SPACE_CHARS);
1025 av_strlcpy(reply->location, p , sizeof(reply->location));
1026 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1027 p += strspn(p, SPACE_CHARS);
1028 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1029 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1030 p += strspn(p, SPACE_CHARS);
1031 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1032 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1033 p += strspn(p, SPACE_CHARS);
1034 if (method && !strcmp(method, "DESCRIBE"))
1035 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1036 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1037 p += strspn(p, SPACE_CHARS);
1038 if (method && !strcmp(method, "PLAY"))
1039 rtsp_parse_rtp_info(rt, p);
1040 } else if (av_stristart(p, "Public:", &p) && rt) {
1041 if (strstr(p, "GET_PARAMETER") &&
1042 method && !strcmp(method, "OPTIONS"))
1043 rt->get_parameter_supported = 1;
1044 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1045 p += strspn(p, SPACE_CHARS);
1046 rt->accept_dynamic_rate = atoi(p);
1047 } else if (av_stristart(p, "Content-Type:", &p)) {
1048 p += strspn(p, SPACE_CHARS);
1049 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1053 /* skip a RTP/TCP interleaved packet */
1054 void ff_rtsp_skip_packet(AVFormatContext *s)
1056 RTSPState *rt = s->priv_data;
1060 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1063 len = AV_RB16(buf + 1);
1065 av_dlog(s, "skipping RTP packet len=%d\n", len);
1070 if (len1 > sizeof(buf))
1072 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1079 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1080 unsigned char **content_ptr,
1081 int return_on_interleaved_data, const char *method)
1083 RTSPState *rt = s->priv_data;
1084 char buf[4096], buf1[1024], *q;
1087 int ret, content_length, line_count = 0, request = 0;
1088 unsigned char *content = NULL;
1094 memset(reply, 0, sizeof(*reply));
1096 /* parse reply (XXX: use buffers) */
1097 rt->last_reply[0] = '\0';
1101 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1102 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1108 /* XXX: only parse it if first char on line ? */
1109 if (return_on_interleaved_data) {
1112 ff_rtsp_skip_packet(s);
1113 } else if (ch != '\r') {
1114 if ((q - buf) < sizeof(buf) - 1)
1120 av_dlog(s, "line='%s'\n", buf);
1122 /* test if last line */
1126 if (line_count == 0) {
1127 /* get reply code */
1128 get_word(buf1, sizeof(buf1), &p);
1129 if (!strncmp(buf1, "RTSP/", 5)) {
1130 get_word(buf1, sizeof(buf1), &p);
1131 reply->status_code = atoi(buf1);
1132 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1134 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1135 get_word(buf1, sizeof(buf1), &p); // object
1139 ff_rtsp_parse_line(reply, p, rt, method);
1140 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1141 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1146 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1147 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1149 content_length = reply->content_length;
1150 if (content_length > 0) {
1151 /* leave some room for a trailing '\0' (useful for simple parsing) */
1152 content = av_malloc(content_length + 1);
1154 return AVERROR(ENOMEM);
1155 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1156 content[content_length] = '\0';
1159 *content_ptr = content;
1165 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1166 const char* ptr = buf;
1168 if (!strcmp(reply->reason, "OPTIONS")) {
1169 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1171 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1172 if (reply->session_id[0])
1173 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1176 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1178 av_strlcat(buf, "\r\n", sizeof(buf));
1180 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1181 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1184 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1186 rt->last_cmd_time = av_gettime();
1187 /* Even if the request from the server had data, it is not the data
1188 * that the caller wants or expects. The memory could also be leaked
1189 * if the actual following reply has content data. */
1191 av_freep(content_ptr);
1192 /* If method is set, this is called from ff_rtsp_send_cmd,
1193 * where a reply to exactly this request is awaited. For
1194 * callers from within packet receiving, we just want to
1195 * return to the caller and go back to receiving packets. */
1201 if (rt->seq != reply->seq) {
1202 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1203 rt->seq, reply->seq);
1207 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1208 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1209 reply->notice == 2306 /* Continuous Feed Terminated */) {
1210 rt->state = RTSP_STATE_IDLE;
1211 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1212 return AVERROR(EIO); /* data or server error */
1213 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1214 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1215 return AVERROR(EPERM);
1221 * Send a command to the RTSP server without waiting for the reply.
1223 * @param s RTSP (de)muxer context
1224 * @param method the method for the request
1225 * @param url the target url for the request
1226 * @param headers extra header lines to include in the request
1227 * @param send_content if non-null, the data to send as request body content
1228 * @param send_content_length the length of the send_content data, or 0 if
1229 * send_content is null
1231 * @return zero if success, nonzero otherwise
1233 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1234 const char *method, const char *url,
1235 const char *headers,
1236 const unsigned char *send_content,
1237 int send_content_length)
1239 RTSPState *rt = s->priv_data;
1240 char buf[4096], *out_buf;
1241 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1243 /* Add in RTSP headers */
1246 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1248 av_strlcat(buf, headers, sizeof(buf));
1249 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1250 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1251 if (rt->session_id[0] != '\0' && (!headers ||
1252 !strstr(headers, "\nIf-Match:"))) {
1253 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1256 char *str = ff_http_auth_create_response(&rt->auth_state,
1257 rt->auth, url, method);
1259 av_strlcat(buf, str, sizeof(buf));
1262 if (send_content_length > 0 && send_content)
1263 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1264 av_strlcat(buf, "\r\n", sizeof(buf));
1266 /* base64 encode rtsp if tunneling */
1267 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1268 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1269 out_buf = base64buf;
1272 av_dlog(s, "Sending:\n%s--\n", buf);
1274 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1275 if (send_content_length > 0 && send_content) {
1276 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1277 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1278 "with content data not supported\n");
1279 return AVERROR_PATCHWELCOME;
1281 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1283 rt->last_cmd_time = av_gettime();
1288 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1289 const char *url, const char *headers)
1291 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1294 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1295 const char *headers, RTSPMessageHeader *reply,
1296 unsigned char **content_ptr)
1298 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1299 content_ptr, NULL, 0);
1302 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1303 const char *method, const char *url,
1305 RTSPMessageHeader *reply,
1306 unsigned char **content_ptr,
1307 const unsigned char *send_content,
1308 int send_content_length)
1310 RTSPState *rt = s->priv_data;
1311 HTTPAuthType cur_auth_type;
1312 int ret, attempts = 0;
1315 cur_auth_type = rt->auth_state.auth_type;
1316 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1318 send_content_length)))
1321 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1325 if (reply->status_code == 401 &&
1326 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1327 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1330 if (reply->status_code > 400){
1331 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1335 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1341 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1342 int lower_transport, const char *real_challenge)
1344 RTSPState *rt = s->priv_data;
1345 int rtx = 0, j, i, err, interleave = 0, port_off;
1346 RTSPStream *rtsp_st;
1347 RTSPMessageHeader reply1, *reply = &reply1;
1349 const char *trans_pref;
1351 if (rt->transport == RTSP_TRANSPORT_RDT)
1352 trans_pref = "x-pn-tng";
1353 else if (rt->transport == RTSP_TRANSPORT_RAW)
1354 trans_pref = "RAW/RAW";
1356 trans_pref = "RTP/AVP";
1358 /* default timeout: 1 minute */
1361 /* Choose a random starting offset within the first half of the
1362 * port range, to allow for a number of ports to try even if the offset
1363 * happens to be at the end of the random range. */
1364 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1365 /* even random offset */
1366 port_off -= port_off & 0x01;
1368 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1369 char transport[2048];
1372 * WMS serves all UDP data over a single connection, the RTX, which
1373 * isn't necessarily the first in the SDP but has to be the first
1374 * to be set up, else the second/third SETUP will fail with a 461.
1376 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1377 rt->server_type == RTSP_SERVER_WMS) {
1380 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1381 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1383 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1387 if (rtx == rt->nb_rtsp_streams)
1388 return -1; /* no RTX found */
1389 rtsp_st = rt->rtsp_streams[rtx];
1391 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1393 rtsp_st = rt->rtsp_streams[i];
1396 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1399 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1400 port = reply->transports[0].client_port_min;
1404 /* first try in specified port range */
1405 while (j <= rt->rtp_port_max) {
1406 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1407 "?localport=%d", j);
1408 /* we will use two ports per rtp stream (rtp and rtcp) */
1410 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1411 &s->interrupt_callback, NULL))
1414 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1419 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1421 snprintf(transport, sizeof(transport) - 1,
1422 "%s/UDP;", trans_pref);
1423 if (rt->server_type != RTSP_SERVER_REAL)
1424 av_strlcat(transport, "unicast;", sizeof(transport));
1425 av_strlcatf(transport, sizeof(transport),
1426 "client_port=%d", port);
1427 if (rt->transport == RTSP_TRANSPORT_RTP &&
1428 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1429 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1433 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1434 /* For WMS streams, the application streams are only used for
1435 * UDP. When trying to set it up for TCP streams, the server
1436 * will return an error. Therefore, we skip those streams. */
1437 if (rt->server_type == RTSP_SERVER_WMS &&
1438 (rtsp_st->stream_index < 0 ||
1439 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1442 snprintf(transport, sizeof(transport) - 1,
1443 "%s/TCP;", trans_pref);
1444 if (rt->transport != RTSP_TRANSPORT_RDT)
1445 av_strlcat(transport, "unicast;", sizeof(transport));
1446 av_strlcatf(transport, sizeof(transport),
1447 "interleaved=%d-%d",
1448 interleave, interleave + 1);
1452 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1453 snprintf(transport, sizeof(transport) - 1,
1454 "%s/UDP;multicast", trans_pref);
1457 av_strlcat(transport, ";mode=record", sizeof(transport));
1458 } else if (rt->server_type == RTSP_SERVER_REAL ||
1459 rt->server_type == RTSP_SERVER_WMS)
1460 av_strlcat(transport, ";mode=play", sizeof(transport));
1461 snprintf(cmd, sizeof(cmd),
1462 "Transport: %s\r\n",
1464 if (rt->accept_dynamic_rate)
1465 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1466 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1467 char real_res[41], real_csum[9];
1468 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1470 av_strlcatf(cmd, sizeof(cmd),
1472 "RealChallenge2: %s, sd=%s\r\n",
1473 rt->session_id, real_res, real_csum);
1475 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1476 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1479 } else if (reply->status_code != RTSP_STATUS_OK ||
1480 reply->nb_transports != 1) {
1481 err = AVERROR_INVALIDDATA;
1485 /* XXX: same protocol for all streams is required */
1487 if (reply->transports[0].lower_transport != rt->lower_transport ||
1488 reply->transports[0].transport != rt->transport) {
1489 err = AVERROR_INVALIDDATA;
1493 rt->lower_transport = reply->transports[0].lower_transport;
1494 rt->transport = reply->transports[0].transport;
1497 /* Fail if the server responded with another lower transport mode
1498 * than what we requested. */
1499 if (reply->transports[0].lower_transport != lower_transport) {
1500 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1501 err = AVERROR_INVALIDDATA;
1505 switch(reply->transports[0].lower_transport) {
1506 case RTSP_LOWER_TRANSPORT_TCP:
1507 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1508 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1511 case RTSP_LOWER_TRANSPORT_UDP: {
1512 char url[1024], options[30] = "";
1513 const char *peer = host;
1515 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1516 av_strlcpy(options, "?connect=1", sizeof(options));
1517 /* Use source address if specified */
1518 if (reply->transports[0].source[0])
1519 peer = reply->transports[0].source;
1520 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1521 reply->transports[0].server_port_min, "%s", options);
1522 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1523 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1524 err = AVERROR_INVALIDDATA;
1527 /* Try to initialize the connection state in a
1528 * potential NAT router by sending dummy packets.
1529 * RTP/RTCP dummy packets are used for RDT, too.
1531 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1533 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1536 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1537 char url[1024], namebuf[50], optbuf[20] = "";
1538 struct sockaddr_storage addr;
1541 if (reply->transports[0].destination.ss_family) {
1542 addr = reply->transports[0].destination;
1543 port = reply->transports[0].port_min;
1544 ttl = reply->transports[0].ttl;
1546 addr = rtsp_st->sdp_ip;
1547 port = rtsp_st->sdp_port;
1548 ttl = rtsp_st->sdp_ttl;
1551 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1552 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1553 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1554 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1555 port, "%s", optbuf);
1556 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1557 &s->interrupt_callback, NULL) < 0) {
1558 err = AVERROR_INVALIDDATA;
1565 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1569 if (rt->nb_rtsp_streams && reply->timeout > 0)
1570 rt->timeout = reply->timeout;
1572 if (rt->server_type == RTSP_SERVER_REAL)
1573 rt->need_subscription = 1;
1578 ff_rtsp_undo_setup(s, 0);
1582 void ff_rtsp_close_connections(AVFormatContext *s)
1584 RTSPState *rt = s->priv_data;
1585 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1586 ffurl_close(rt->rtsp_hd);
1587 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1590 int ff_rtsp_connect(AVFormatContext *s)
1592 RTSPState *rt = s->priv_data;
1593 char proto[128], host[1024], path[1024];
1594 char tcpname[1024], cmd[2048], auth[128];
1595 const char *lower_rtsp_proto = "tcp";
1596 int port, err, tcp_fd;
1597 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1598 int lower_transport_mask = 0;
1599 int default_port = RTSP_DEFAULT_PORT;
1600 char real_challenge[64] = "";
1601 struct sockaddr_storage peer;
1602 socklen_t peer_len = sizeof(peer);
1604 if (rt->rtp_port_max < rt->rtp_port_min) {
1605 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1606 "than min port %d\n", rt->rtp_port_max,
1608 return AVERROR(EINVAL);
1611 if (!ff_network_init())
1612 return AVERROR(EIO);
1614 if (s->max_delay < 0) /* Not set by the caller */
1615 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1617 rt->control_transport = RTSP_MODE_PLAIN;
1618 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1619 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1620 rt->control_transport = RTSP_MODE_TUNNEL;
1622 /* Only pass through valid flags from here */
1623 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1626 /* extract hostname and port */
1627 av_url_split(proto, sizeof(proto), auth, sizeof(auth),
1628 host, sizeof(host), &port, path, sizeof(path), s->filename);
1630 if (!strcmp(proto, "rtsps")) {
1631 lower_rtsp_proto = "tls";
1632 default_port = RTSPS_DEFAULT_PORT;
1633 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1637 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1640 port = default_port;
1642 lower_transport_mask = rt->lower_transport_mask;
1644 if (!lower_transport_mask)
1645 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1648 /* Only UDP or TCP - UDP multicast isn't supported. */
1649 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1650 (1 << RTSP_LOWER_TRANSPORT_TCP);
1651 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1652 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1653 "only UDP and TCP are supported for output.\n");
1654 err = AVERROR(EINVAL);
1659 /* Construct the URI used in request; this is similar to s->filename,
1660 * but with authentication credentials removed and RTSP specific options
1662 ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
1663 host, port, "%s", path);
1665 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1666 /* set up initial handshake for tunneling */
1667 char httpname[1024];
1668 char sessioncookie[17];
1671 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1672 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1673 av_get_random_seed(), av_get_random_seed());
1676 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1677 &s->interrupt_callback) < 0) {
1682 /* generate GET headers */
1683 snprintf(headers, sizeof(headers),
1684 "x-sessioncookie: %s\r\n"
1685 "Accept: application/x-rtsp-tunnelled\r\n"
1686 "Pragma: no-cache\r\n"
1687 "Cache-Control: no-cache\r\n",
1689 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1691 /* complete the connection */
1692 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1698 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1699 &s->interrupt_callback) < 0 ) {
1704 /* generate POST headers */
1705 snprintf(headers, sizeof(headers),
1706 "x-sessioncookie: %s\r\n"
1707 "Content-Type: application/x-rtsp-tunnelled\r\n"
1708 "Pragma: no-cache\r\n"
1709 "Cache-Control: no-cache\r\n"
1710 "Content-Length: 32767\r\n"
1711 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1713 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1714 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1716 /* Initialize the authentication state for the POST session. The HTTP
1717 * protocol implementation doesn't properly handle multi-pass
1718 * authentication for POST requests, since it would require one of
1720 * - implementing Expect: 100-continue, which many HTTP servers
1721 * don't support anyway, even less the RTSP servers that do HTTP
1723 * - sending the whole POST data until getting a 401 reply specifying
1724 * what authentication method to use, then resending all that data
1725 * - waiting for potential 401 replies directly after sending the
1726 * POST header (waiting for some unspecified time)
1727 * Therefore, we copy the full auth state, which works for both basic
1728 * and digest. (For digest, we would have to synchronize the nonce
1729 * count variable between the two sessions, if we'd do more requests
1730 * with the original session, though.)
1732 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1734 /* complete the connection */
1735 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1740 /* open the tcp connection */
1741 ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
1743 "?timeout=%d", rt->stimeout);
1744 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1745 &s->interrupt_callback, NULL) < 0) {
1749 rt->rtsp_hd_out = rt->rtsp_hd;
1753 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1754 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1755 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1756 NULL, 0, NI_NUMERICHOST);
1759 /* request options supported by the server; this also detects server
1761 for (rt->server_type = RTSP_SERVER_RTP;;) {
1763 if (rt->server_type == RTSP_SERVER_REAL)
1766 * The following entries are required for proper
1767 * streaming from a Realmedia server. They are
1768 * interdependent in some way although we currently
1769 * don't quite understand how. Values were copied
1770 * from mplayer SVN r23589.
1771 * ClientChallenge is a 16-byte ID in hex
1772 * CompanyID is a 16-byte ID in base64
1774 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1775 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1776 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1777 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1779 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1780 if (reply->status_code != RTSP_STATUS_OK) {
1781 err = AVERROR_INVALIDDATA;
1785 /* detect server type if not standard-compliant RTP */
1786 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1787 rt->server_type = RTSP_SERVER_REAL;
1789 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1790 rt->server_type = RTSP_SERVER_WMS;
1791 } else if (rt->server_type == RTSP_SERVER_REAL)
1792 strcpy(real_challenge, reply->real_challenge);
1796 if (s->iformat && CONFIG_RTSP_DEMUXER)
1797 err = ff_rtsp_setup_input_streams(s, reply);
1798 else if (CONFIG_RTSP_MUXER)
1799 err = ff_rtsp_setup_output_streams(s, host);
1804 int lower_transport = ff_log2_tab[lower_transport_mask &
1805 ~(lower_transport_mask - 1)];
1807 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1808 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1809 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1811 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1812 rt->server_type == RTSP_SERVER_REAL ?
1813 real_challenge : NULL);
1816 lower_transport_mask &= ~(1 << lower_transport);
1817 if (lower_transport_mask == 0 && err == 1) {
1818 err = AVERROR(EPROTONOSUPPORT);
1823 rt->lower_transport_mask = lower_transport_mask;
1824 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1825 rt->state = RTSP_STATE_IDLE;
1826 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1829 ff_rtsp_close_streams(s);
1830 ff_rtsp_close_connections(s);
1831 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1832 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1833 rt->session_id[0] = '\0';
1834 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1842 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1845 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1846 uint8_t *buf, int buf_size, int64_t wait_end)
1848 RTSPState *rt = s->priv_data;
1849 RTSPStream *rtsp_st;
1850 int n, i, ret, tcp_fd, timeout_cnt = 0;
1852 struct pollfd *p = rt->p;
1853 int *fds = NULL, fdsnum, fdsidx;
1856 if (ff_check_interrupt(&s->interrupt_callback))
1857 return AVERROR_EXIT;
1858 if (wait_end && wait_end - av_gettime() < 0)
1859 return AVERROR(EAGAIN);
1862 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1863 p[max_p].fd = tcp_fd;
1864 p[max_p++].events = POLLIN;
1868 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1869 rtsp_st = rt->rtsp_streams[i];
1870 if (rtsp_st->rtp_handle) {
1871 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1873 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1877 av_log(s, AV_LOG_ERROR,
1878 "Number of fds %d not supported\n", fdsnum);
1879 return AVERROR_INVALIDDATA;
1881 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1882 p[max_p].fd = fds[fdsidx];
1883 p[max_p++].events = POLLIN;
1888 n = poll(p, max_p, POLL_TIMEOUT_MS);
1890 int j = 1 - (tcp_fd == -1);
1892 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1893 rtsp_st = rt->rtsp_streams[i];
1894 if (rtsp_st->rtp_handle) {
1895 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1896 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1898 *prtsp_st = rtsp_st;
1905 #if CONFIG_RTSP_DEMUXER
1906 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1907 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1908 if (rt->state == RTSP_STATE_STREAMING) {
1909 if (!ff_rtsp_parse_streaming_commands(s))
1912 av_log(s, AV_LOG_WARNING,
1913 "Unable to answer to TEARDOWN\n");
1917 RTSPMessageHeader reply;
1918 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1921 /* XXX: parse message */
1922 if (rt->state != RTSP_STATE_STREAMING)
1927 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1928 return AVERROR(ETIMEDOUT);
1929 } else if (n < 0 && errno != EINTR)
1930 return AVERROR(errno);
1934 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1935 const uint8_t *buf, int len)
1937 RTSPState *rt = s->priv_data;
1941 if (rt->nb_rtsp_streams == 1) {
1942 *rtsp_st = rt->rtsp_streams[0];
1945 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1946 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1948 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1949 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1952 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1953 *rtsp_st = rt->rtsp_streams[i];
1960 av_log(s, AV_LOG_WARNING,
1961 "Unable to pick stream for packet - SSRC not known for "
1963 return AVERROR(EAGAIN);
1966 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1967 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1968 *rtsp_st = rt->rtsp_streams[i];
1974 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1975 return AVERROR(EAGAIN);
1978 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1980 RTSPState *rt = s->priv_data;
1982 RTSPStream *rtsp_st, *first_queue_st = NULL;
1983 int64_t wait_end = 0;
1985 if (rt->nb_byes == rt->nb_rtsp_streams)
1988 /* get next frames from the same RTP packet */
1989 if (rt->cur_transport_priv) {
1990 if (rt->transport == RTSP_TRANSPORT_RDT) {
1991 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1992 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1993 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1994 } else if (rt->ts && CONFIG_RTPDEC) {
1995 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1997 rt->recvbuf_pos += ret;
1998 ret = rt->recvbuf_pos < rt->recvbuf_len;
2003 rt->cur_transport_priv = NULL;
2005 } else if (ret == 1) {
2008 rt->cur_transport_priv = NULL;
2012 if (rt->transport == RTSP_TRANSPORT_RTP) {
2014 int64_t first_queue_time = 0;
2015 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2016 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
2020 queue_time = ff_rtp_queued_packet_time(rtpctx);
2021 if (queue_time && (queue_time - first_queue_time < 0 ||
2022 !first_queue_time)) {
2023 first_queue_time = queue_time;
2024 first_queue_st = rt->rtsp_streams[i];
2027 if (first_queue_time) {
2028 wait_end = first_queue_time + s->max_delay;
2031 first_queue_st = NULL;
2035 /* read next RTP packet */
2037 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2039 return AVERROR(ENOMEM);
2042 switch(rt->lower_transport) {
2044 #if CONFIG_RTSP_DEMUXER
2045 case RTSP_LOWER_TRANSPORT_TCP:
2046 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2049 case RTSP_LOWER_TRANSPORT_UDP:
2050 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2051 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2052 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2053 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2055 case RTSP_LOWER_TRANSPORT_CUSTOM:
2056 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2057 wait_end && wait_end < av_gettime())
2058 len = AVERROR(EAGAIN);
2060 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2061 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2062 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2063 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2066 if (len == AVERROR(EAGAIN) && first_queue_st &&
2067 rt->transport == RTSP_TRANSPORT_RTP) {
2068 rtsp_st = first_queue_st;
2069 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2076 if (rt->transport == RTSP_TRANSPORT_RDT) {
2077 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2078 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2079 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2080 if (rtsp_st->feedback) {
2081 AVIOContext *pb = NULL;
2082 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2084 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2087 /* Either bad packet, or a RTCP packet. Check if the
2088 * first_rtcp_ntp_time field was initialized. */
2089 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2090 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2091 /* first_rtcp_ntp_time has been initialized for this stream,
2092 * copy the same value to all other uninitialized streams,
2093 * in order to map their timestamp origin to the same ntp time
2096 AVStream *st = NULL;
2097 if (rtsp_st->stream_index >= 0)
2098 st = s->streams[rtsp_st->stream_index];
2099 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2100 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2101 AVStream *st2 = NULL;
2102 if (rt->rtsp_streams[i]->stream_index >= 0)
2103 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2104 if (rtpctx2 && st && st2 &&
2105 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2106 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2107 rtpctx2->rtcp_ts_offset = av_rescale_q(
2108 rtpctx->rtcp_ts_offset, st->time_base,
2112 // Make real NTP start time available in AVFormatContext
2113 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2114 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2116 s->start_time_realtime -=
2117 av_rescale (rtpctx->rtcp_ts_offset,
2118 (uint64_t) rtpctx->st->time_base.num * 1000000,
2119 rtpctx->st->time_base.den);
2123 if (ret == -RTCP_BYE) {
2126 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2127 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2129 if (rt->nb_byes == rt->nb_rtsp_streams)
2133 } else if (rt->ts && CONFIG_RTPDEC) {
2134 ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2137 rt->recvbuf_len = len;
2138 rt->recvbuf_pos = ret;
2139 rt->cur_transport_priv = rt->ts;
2146 return AVERROR_INVALIDDATA;
2152 /* more packets may follow, so we save the RTP context */
2153 rt->cur_transport_priv = rtsp_st->transport_priv;
2157 #endif /* CONFIG_RTPDEC */
2159 #if CONFIG_SDP_DEMUXER
2160 static int sdp_probe(AVProbeData *p1)
2162 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2164 /* we look for a line beginning "c=IN IP" */
2165 while (p < p_end && *p != '\0') {
2166 if (p + sizeof("c=IN IP") - 1 < p_end &&
2167 av_strstart(p, "c=IN IP", NULL))
2168 return AVPROBE_SCORE_EXTENSION;
2170 while (p < p_end - 1 && *p != '\n') p++;
2179 static void append_source_addrs(char *buf, int size, const char *name,
2180 int count, struct RTSPSource **addrs)
2185 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2186 for (i = 1; i < count; i++)
2187 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2190 static int sdp_read_header(AVFormatContext *s)
2192 RTSPState *rt = s->priv_data;
2193 RTSPStream *rtsp_st;
2198 if (!ff_network_init())
2199 return AVERROR(EIO);
2201 if (s->max_delay < 0) /* Not set by the caller */
2202 s->max_delay = DEFAULT_REORDERING_DELAY;
2203 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2204 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2206 /* read the whole sdp file */
2207 /* XXX: better loading */
2208 content = av_malloc(SDP_MAX_SIZE);
2209 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2212 return AVERROR_INVALIDDATA;
2214 content[size] ='\0';
2216 err = ff_sdp_parse(s, content);
2220 /* open each RTP stream */
2221 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2223 rtsp_st = rt->rtsp_streams[i];
2225 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2226 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2227 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2228 ff_url_join(url, sizeof(url), "rtp", NULL,
2229 namebuf, rtsp_st->sdp_port,
2230 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2231 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2232 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2233 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2235 append_source_addrs(url, sizeof(url), "sources",
2236 rtsp_st->nb_include_source_addrs,
2237 rtsp_st->include_source_addrs);
2238 append_source_addrs(url, sizeof(url), "block",
2239 rtsp_st->nb_exclude_source_addrs,
2240 rtsp_st->exclude_source_addrs);
2241 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2242 &s->interrupt_callback, NULL) < 0) {
2243 err = AVERROR_INVALIDDATA;
2247 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2252 ff_rtsp_close_streams(s);
2257 static int sdp_read_close(AVFormatContext *s)
2259 ff_rtsp_close_streams(s);
2264 static const AVClass sdp_demuxer_class = {
2265 .class_name = "SDP demuxer",
2266 .item_name = av_default_item_name,
2267 .option = sdp_options,
2268 .version = LIBAVUTIL_VERSION_INT,
2271 AVInputFormat ff_sdp_demuxer = {
2273 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2274 .priv_data_size = sizeof(RTSPState),
2275 .read_probe = sdp_probe,
2276 .read_header = sdp_read_header,
2277 .read_packet = ff_rtsp_fetch_packet,
2278 .read_close = sdp_read_close,
2279 .priv_class = &sdp_demuxer_class,
2281 #endif /* CONFIG_SDP_DEMUXER */
2283 #if CONFIG_RTP_DEMUXER
2284 static int rtp_probe(AVProbeData *p)
2286 if (av_strstart(p->filename, "rtp:", NULL))
2287 return AVPROBE_SCORE_MAX;
2291 static int rtp_read_header(AVFormatContext *s)
2293 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2294 char host[500], sdp[500];
2296 URLContext* in = NULL;
2298 AVCodecContext codec = { 0 };
2299 struct sockaddr_storage addr;
2301 socklen_t addrlen = sizeof(addr);
2302 RTSPState *rt = s->priv_data;
2304 if (!ff_network_init())
2305 return AVERROR(EIO);
2307 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2308 &s->interrupt_callback, NULL);
2313 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2314 if (ret == AVERROR(EAGAIN))
2319 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2323 if ((recvbuf[0] & 0xc0) != 0x80) {
2324 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2329 if (RTP_PT_IS_RTCP(recvbuf[1]))
2332 payload_type = recvbuf[1] & 0x7f;
2335 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2339 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2340 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2341 "without an SDP file describing it\n",
2345 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2346 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2347 "properly you need an SDP file "
2351 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2352 NULL, 0, s->filename);
2354 snprintf(sdp, sizeof(sdp),
2355 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2356 addr.ss_family == AF_INET ? 4 : 6, host,
2357 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2358 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2359 port, payload_type);
2360 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2362 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2365 /* sdp_read_header initializes this again */
2368 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2370 ret = sdp_read_header(s);
2381 static const AVClass rtp_demuxer_class = {
2382 .class_name = "RTP demuxer",
2383 .item_name = av_default_item_name,
2384 .option = rtp_options,
2385 .version = LIBAVUTIL_VERSION_INT,
2388 AVInputFormat ff_rtp_demuxer = {
2390 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2391 .priv_data_size = sizeof(RTSPState),
2392 .read_probe = rtp_probe,
2393 .read_header = rtp_read_header,
2394 .read_packet = ff_rtsp_fetch_packet,
2395 .read_close = sdp_read_close,
2396 .flags = AVFMT_NOFILE,
2397 .priv_class = &rtp_demuxer_class,
2399 #endif /* CONFIG_RTP_DEMUXER */