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 micro seconds) 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 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
302 struct RTSPSource ***dest, int *dest_count)
304 RTSPSource *rtsp_src, *rtsp_src2;
306 for (i = 0; i < count; i++) {
308 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
311 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
312 dynarray_add(dest, dest_count, rtsp_src2);
316 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
317 int letter, const char *buf)
319 RTSPState *rt = s->priv_data;
320 char buf1[64], st_type[64];
322 enum AVMediaType codec_type;
326 RTSPSource *rtsp_src;
327 struct sockaddr_storage sdp_ip;
330 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
333 if (s1->skip_media && letter != 'm')
337 get_word(buf1, sizeof(buf1), &p);
338 if (strcmp(buf1, "IN") != 0)
340 get_word(buf1, sizeof(buf1), &p);
341 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
343 get_word_sep(buf1, sizeof(buf1), "/", &p);
344 if (get_sockaddr(buf1, &sdp_ip))
349 get_word_sep(buf1, sizeof(buf1), "/", &p);
352 if (s->nb_streams == 0) {
353 s1->default_ip = sdp_ip;
354 s1->default_ttl = ttl;
356 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
357 rtsp_st->sdp_ip = sdp_ip;
358 rtsp_st->sdp_ttl = ttl;
362 av_dict_set(&s->metadata, "title", p, 0);
365 if (s->nb_streams == 0) {
366 av_dict_set(&s->metadata, "comment", p, 0);
373 codec_type = AVMEDIA_TYPE_UNKNOWN;
374 get_word(st_type, sizeof(st_type), &p);
375 if (!strcmp(st_type, "audio")) {
376 codec_type = AVMEDIA_TYPE_AUDIO;
377 } else if (!strcmp(st_type, "video")) {
378 codec_type = AVMEDIA_TYPE_VIDEO;
379 } else if (!strcmp(st_type, "application")) {
380 codec_type = AVMEDIA_TYPE_DATA;
382 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
386 rtsp_st = av_mallocz(sizeof(RTSPStream));
389 rtsp_st->stream_index = -1;
390 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
392 rtsp_st->sdp_ip = s1->default_ip;
393 rtsp_st->sdp_ttl = s1->default_ttl;
395 copy_default_source_addrs(s1->default_include_source_addrs,
396 s1->nb_default_include_source_addrs,
397 &rtsp_st->include_source_addrs,
398 &rtsp_st->nb_include_source_addrs);
399 copy_default_source_addrs(s1->default_exclude_source_addrs,
400 s1->nb_default_exclude_source_addrs,
401 &rtsp_st->exclude_source_addrs,
402 &rtsp_st->nb_exclude_source_addrs);
404 get_word(buf1, sizeof(buf1), &p); /* port */
405 rtsp_st->sdp_port = atoi(buf1);
407 get_word(buf1, sizeof(buf1), &p); /* protocol */
408 if (!strcmp(buf1, "udp"))
409 rt->transport = RTSP_TRANSPORT_RAW;
410 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
411 rtsp_st->feedback = 1;
413 /* XXX: handle list of formats */
414 get_word(buf1, sizeof(buf1), &p); /* format list */
415 rtsp_st->sdp_payload_type = atoi(buf1);
417 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
418 /* no corresponding stream */
419 if (rt->transport == RTSP_TRANSPORT_RAW) {
420 if (!rt->ts && CONFIG_RTPDEC)
421 rt->ts = ff_mpegts_parse_open(s);
423 RTPDynamicProtocolHandler *handler;
424 handler = ff_rtp_handler_find_by_id(
425 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
426 init_rtp_handler(handler, rtsp_st, NULL);
427 if (handler && handler->init)
428 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
430 } else if (rt->server_type == RTSP_SERVER_WMS &&
431 codec_type == AVMEDIA_TYPE_DATA) {
432 /* RTX stream, a stream that carries all the other actual
433 * audio/video streams. Don't expose this to the callers. */
435 st = avformat_new_stream(s, NULL);
438 st->id = rt->nb_rtsp_streams - 1;
439 rtsp_st->stream_index = st->index;
440 st->codec->codec_type = codec_type;
441 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
442 RTPDynamicProtocolHandler *handler;
443 /* if standard payload type, we can find the codec right now */
444 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
445 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
446 st->codec->sample_rate > 0)
447 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
448 /* Even static payload types may need a custom depacketizer */
449 handler = ff_rtp_handler_find_by_id(
450 rtsp_st->sdp_payload_type, st->codec->codec_type);
451 init_rtp_handler(handler, rtsp_st, st->codec);
452 if (handler && handler->init)
453 handler->init(s, st->index,
454 rtsp_st->dynamic_protocol_context);
457 /* put a default control url */
458 av_strlcpy(rtsp_st->control_url, rt->control_uri,
459 sizeof(rtsp_st->control_url));
462 if (av_strstart(p, "control:", &p)) {
463 if (s->nb_streams == 0) {
464 if (!strncmp(p, "rtsp://", 7))
465 av_strlcpy(rt->control_uri, p,
466 sizeof(rt->control_uri));
469 /* get the control url */
470 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
472 /* XXX: may need to add full url resolution */
473 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
475 if (proto[0] == '\0') {
476 /* relative control URL */
477 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
478 av_strlcat(rtsp_st->control_url, "/",
479 sizeof(rtsp_st->control_url));
480 av_strlcat(rtsp_st->control_url, p,
481 sizeof(rtsp_st->control_url));
483 av_strlcpy(rtsp_st->control_url, p,
484 sizeof(rtsp_st->control_url));
486 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
487 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
488 get_word(buf1, sizeof(buf1), &p);
489 payload_type = atoi(buf1);
490 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
491 if (rtsp_st->stream_index >= 0) {
492 st = s->streams[rtsp_st->stream_index];
493 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
495 } else if (av_strstart(p, "fmtp:", &p) ||
496 av_strstart(p, "framesize:", &p)) {
497 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
498 // let dynamic protocol handlers have a stab at the line.
499 get_word(buf1, sizeof(buf1), &p);
500 payload_type = atoi(buf1);
501 for (i = 0; i < rt->nb_rtsp_streams; i++) {
502 rtsp_st = rt->rtsp_streams[i];
503 if (rtsp_st->sdp_payload_type == payload_type &&
504 rtsp_st->dynamic_handler &&
505 rtsp_st->dynamic_handler->parse_sdp_a_line)
506 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
507 rtsp_st->dynamic_protocol_context, buf);
509 } else if (av_strstart(p, "range:", &p)) {
512 // this is so that seeking on a streamed file can work.
513 rtsp_parse_range_npt(p, &start, &end);
514 s->start_time = start;
515 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
516 s->duration = (end == AV_NOPTS_VALUE) ?
517 AV_NOPTS_VALUE : end - start;
518 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
520 rt->transport = RTSP_TRANSPORT_RDT;
521 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
523 st = s->streams[s->nb_streams - 1];
524 st->codec->sample_rate = atoi(p);
525 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
527 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
528 get_word(buf1, sizeof(buf1), &p); // ignore tag
529 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
530 p += strspn(p, SPACE_CHARS);
531 if (av_strstart(p, "inline:", &p))
532 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
533 } else if (av_strstart(p, "source-filter:", &p)) {
535 get_word(buf1, sizeof(buf1), &p);
536 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
538 exclude = !strcmp(buf1, "excl");
540 get_word(buf1, sizeof(buf1), &p);
541 if (strcmp(buf1, "IN") != 0)
543 get_word(buf1, sizeof(buf1), &p);
544 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
546 // not checking that the destination address actually matches or is wildcard
547 get_word(buf1, sizeof(buf1), &p);
550 rtsp_src = av_mallocz(sizeof(*rtsp_src));
553 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
555 if (s->nb_streams == 0) {
556 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
558 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
559 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
562 if (s->nb_streams == 0) {
563 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
565 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
566 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
571 if (rt->server_type == RTSP_SERVER_WMS)
572 ff_wms_parse_sdp_a_line(s, p);
573 if (s->nb_streams > 0) {
574 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
576 if (rt->server_type == RTSP_SERVER_REAL)
577 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
579 if (rtsp_st->dynamic_handler &&
580 rtsp_st->dynamic_handler->parse_sdp_a_line)
581 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
582 rtsp_st->stream_index,
583 rtsp_st->dynamic_protocol_context, buf);
590 int ff_sdp_parse(AVFormatContext *s, const char *content)
592 RTSPState *rt = s->priv_data;
595 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
596 * contain long SDP lines containing complete ASF Headers (several
597 * kB) or arrays of MDPR (RM stream descriptor) headers plus
598 * "rulebooks" describing their properties. Therefore, the SDP line
601 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
602 * in rtpdec_xiph.c. */
604 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
608 p += strspn(p, SPACE_CHARS);
616 /* get the content */
618 while (*p != '\n' && *p != '\r' && *p != '\0') {
619 if ((q - buf) < sizeof(buf) - 1)
624 sdp_parse_line(s, s1, letter, buf);
626 while (*p != '\n' && *p != '\0')
632 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
633 av_free(s1->default_include_source_addrs[i]);
634 av_freep(&s1->default_include_source_addrs);
635 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
636 av_free(s1->default_exclude_source_addrs[i]);
637 av_freep(&s1->default_exclude_source_addrs);
639 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
640 if (!rt->p) return AVERROR(ENOMEM);
643 #endif /* CONFIG_RTPDEC */
645 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
647 RTSPState *rt = s->priv_data;
650 for (i = 0; i < rt->nb_rtsp_streams; i++) {
651 RTSPStream *rtsp_st = rt->rtsp_streams[i];
654 if (rtsp_st->transport_priv) {
656 AVFormatContext *rtpctx = rtsp_st->transport_priv;
657 av_write_trailer(rtpctx);
658 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
660 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
661 ff_rtsp_tcp_write_packet(s, rtsp_st);
662 avio_close_dyn_buf(rtpctx->pb, &ptr);
665 avio_close(rtpctx->pb);
667 avformat_free_context(rtpctx);
668 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
669 ff_rdt_parse_close(rtsp_st->transport_priv);
670 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
671 ff_rtp_parse_close(rtsp_st->transport_priv);
673 rtsp_st->transport_priv = NULL;
674 if (rtsp_st->rtp_handle)
675 ffurl_close(rtsp_st->rtp_handle);
676 rtsp_st->rtp_handle = NULL;
680 /* close and free RTSP streams */
681 void ff_rtsp_close_streams(AVFormatContext *s)
683 RTSPState *rt = s->priv_data;
687 ff_rtsp_undo_setup(s, 0);
688 for (i = 0; i < rt->nb_rtsp_streams; i++) {
689 rtsp_st = rt->rtsp_streams[i];
691 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
692 rtsp_st->dynamic_handler->free(
693 rtsp_st->dynamic_protocol_context);
694 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
695 av_free(rtsp_st->include_source_addrs[j]);
696 av_freep(&rtsp_st->include_source_addrs);
697 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
698 av_free(rtsp_st->exclude_source_addrs[j]);
699 av_freep(&rtsp_st->exclude_source_addrs);
704 av_free(rt->rtsp_streams);
706 avformat_close_input(&rt->asf_ctx);
708 if (rt->ts && CONFIG_RTPDEC)
709 ff_mpegts_parse_close(rt->ts);
711 av_free(rt->recvbuf);
714 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
716 RTSPState *rt = s->priv_data;
718 int reordering_queue_size = rt->reordering_queue_size;
719 if (reordering_queue_size < 0) {
720 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
721 reordering_queue_size = 0;
723 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
726 /* open the RTP context */
727 if (rtsp_st->stream_index >= 0)
728 st = s->streams[rtsp_st->stream_index];
730 s->ctx_flags |= AVFMTCTX_NOHEADER;
732 if (s->oformat && CONFIG_RTSP_MUXER) {
733 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
734 s, st, rtsp_st->rtp_handle,
735 RTSP_TCP_MAX_PACKET_SIZE,
736 rtsp_st->stream_index);
737 /* Ownership of rtp_handle is passed to the rtp mux context */
738 rtsp_st->rtp_handle = NULL;
741 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
742 return 0; // Don't need to open any parser here
743 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
744 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
745 rtsp_st->dynamic_protocol_context,
746 rtsp_st->dynamic_handler);
747 else if (CONFIG_RTPDEC)
748 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
749 rtsp_st->sdp_payload_type,
750 reordering_queue_size);
752 if (!rtsp_st->transport_priv) {
753 return AVERROR(ENOMEM);
754 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
755 if (rtsp_st->dynamic_handler) {
756 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
757 rtsp_st->dynamic_protocol_context,
758 rtsp_st->dynamic_handler);
760 if (rtsp_st->crypto_suite[0])
761 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
762 rtsp_st->crypto_suite,
763 rtsp_st->crypto_params);
769 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
770 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
777 q += strspn(q, SPACE_CHARS);
778 v = strtol(q, &p, 10);
782 v = strtol(p, &p, 10);
791 /* XXX: only one transport specification is parsed */
792 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
794 char transport_protocol[16];
796 char lower_transport[16];
798 RTSPTransportField *th;
801 reply->nb_transports = 0;
804 p += strspn(p, SPACE_CHARS);
808 th = &reply->transports[reply->nb_transports];
810 get_word_sep(transport_protocol, sizeof(transport_protocol),
812 if (!av_strcasecmp (transport_protocol, "rtp")) {
813 get_word_sep(profile, sizeof(profile), "/;,", &p);
814 lower_transport[0] = '\0';
815 /* rtp/avp/<protocol> */
817 get_word_sep(lower_transport, sizeof(lower_transport),
820 th->transport = RTSP_TRANSPORT_RTP;
821 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
822 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
823 /* x-pn-tng/<protocol> */
824 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
826 th->transport = RTSP_TRANSPORT_RDT;
827 } else if (!av_strcasecmp(transport_protocol, "raw")) {
828 get_word_sep(profile, sizeof(profile), "/;,", &p);
829 lower_transport[0] = '\0';
830 /* raw/raw/<protocol> */
832 get_word_sep(lower_transport, sizeof(lower_transport),
835 th->transport = RTSP_TRANSPORT_RAW;
837 if (!av_strcasecmp(lower_transport, "TCP"))
838 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
840 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
844 /* get each parameter */
845 while (*p != '\0' && *p != ',') {
846 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
847 if (!strcmp(parameter, "port")) {
850 rtsp_parse_range(&th->port_min, &th->port_max, &p);
852 } else if (!strcmp(parameter, "client_port")) {
855 rtsp_parse_range(&th->client_port_min,
856 &th->client_port_max, &p);
858 } else if (!strcmp(parameter, "server_port")) {
861 rtsp_parse_range(&th->server_port_min,
862 &th->server_port_max, &p);
864 } else if (!strcmp(parameter, "interleaved")) {
867 rtsp_parse_range(&th->interleaved_min,
868 &th->interleaved_max, &p);
870 } else if (!strcmp(parameter, "multicast")) {
871 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
872 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
873 } else if (!strcmp(parameter, "ttl")) {
877 th->ttl = strtol(p, &end, 10);
880 } else if (!strcmp(parameter, "destination")) {
883 get_word_sep(buf, sizeof(buf), ";,", &p);
884 get_sockaddr(buf, &th->destination);
886 } else if (!strcmp(parameter, "source")) {
889 get_word_sep(buf, sizeof(buf), ";,", &p);
890 av_strlcpy(th->source, buf, sizeof(th->source));
892 } else if (!strcmp(parameter, "mode")) {
895 get_word_sep(buf, sizeof(buf), ";, ", &p);
896 if (!strcmp(buf, "record") ||
897 !strcmp(buf, "receive"))
902 while (*p != ';' && *p != '\0' && *p != ',')
910 reply->nb_transports++;
914 static void handle_rtp_info(RTSPState *rt, const char *url,
915 uint32_t seq, uint32_t rtptime)
918 if (!rtptime || !url[0])
920 if (rt->transport != RTSP_TRANSPORT_RTP)
922 for (i = 0; i < rt->nb_rtsp_streams; i++) {
923 RTSPStream *rtsp_st = rt->rtsp_streams[i];
924 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
927 if (!strcmp(rtsp_st->control_url, url)) {
928 rtpctx->base_timestamp = rtptime;
934 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
937 char key[20], value[1024], url[1024] = "";
938 uint32_t seq = 0, rtptime = 0;
941 p += strspn(p, SPACE_CHARS);
944 get_word_sep(key, sizeof(key), "=", &p);
948 get_word_sep(value, sizeof(value), ";, ", &p);
950 if (!strcmp(key, "url"))
951 av_strlcpy(url, value, sizeof(url));
952 else if (!strcmp(key, "seq"))
953 seq = strtoul(value, NULL, 10);
954 else if (!strcmp(key, "rtptime"))
955 rtptime = strtoul(value, NULL, 10);
957 handle_rtp_info(rt, url, seq, rtptime);
966 handle_rtp_info(rt, url, seq, rtptime);
969 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
970 RTSPState *rt, const char *method)
974 /* NOTE: we do case independent match for broken servers */
976 if (av_stristart(p, "Session:", &p)) {
978 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
979 if (av_stristart(p, ";timeout=", &p) &&
980 (t = strtol(p, NULL, 10)) > 0) {
983 } else if (av_stristart(p, "Content-Length:", &p)) {
984 reply->content_length = strtol(p, NULL, 10);
985 } else if (av_stristart(p, "Transport:", &p)) {
986 rtsp_parse_transport(reply, p);
987 } else if (av_stristart(p, "CSeq:", &p)) {
988 reply->seq = strtol(p, NULL, 10);
989 } else if (av_stristart(p, "Range:", &p)) {
990 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
991 } else if (av_stristart(p, "RealChallenge1:", &p)) {
992 p += strspn(p, SPACE_CHARS);
993 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
994 } else if (av_stristart(p, "Server:", &p)) {
995 p += strspn(p, SPACE_CHARS);
996 av_strlcpy(reply->server, p, sizeof(reply->server));
997 } else if (av_stristart(p, "Notice:", &p) ||
998 av_stristart(p, "X-Notice:", &p)) {
999 reply->notice = strtol(p, NULL, 10);
1000 } else if (av_stristart(p, "Location:", &p)) {
1001 p += strspn(p, SPACE_CHARS);
1002 av_strlcpy(reply->location, p , sizeof(reply->location));
1003 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1004 p += strspn(p, SPACE_CHARS);
1005 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1006 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1007 p += strspn(p, SPACE_CHARS);
1008 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1009 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1010 p += strspn(p, SPACE_CHARS);
1011 if (method && !strcmp(method, "DESCRIBE"))
1012 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1013 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1014 p += strspn(p, SPACE_CHARS);
1015 if (method && !strcmp(method, "PLAY"))
1016 rtsp_parse_rtp_info(rt, p);
1017 } else if (av_stristart(p, "Public:", &p) && rt) {
1018 if (strstr(p, "GET_PARAMETER") &&
1019 method && !strcmp(method, "OPTIONS"))
1020 rt->get_parameter_supported = 1;
1021 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1022 p += strspn(p, SPACE_CHARS);
1023 rt->accept_dynamic_rate = atoi(p);
1024 } else if (av_stristart(p, "Content-Type:", &p)) {
1025 p += strspn(p, SPACE_CHARS);
1026 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1030 /* skip a RTP/TCP interleaved packet */
1031 void ff_rtsp_skip_packet(AVFormatContext *s)
1033 RTSPState *rt = s->priv_data;
1037 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1040 len = AV_RB16(buf + 1);
1042 av_dlog(s, "skipping RTP packet len=%d\n", len);
1047 if (len1 > sizeof(buf))
1049 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1056 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1057 unsigned char **content_ptr,
1058 int return_on_interleaved_data, const char *method)
1060 RTSPState *rt = s->priv_data;
1061 char buf[4096], buf1[1024], *q;
1064 int ret, content_length, line_count = 0, request = 0;
1065 unsigned char *content = NULL;
1071 memset(reply, 0, sizeof(*reply));
1073 /* parse reply (XXX: use buffers) */
1074 rt->last_reply[0] = '\0';
1078 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1079 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1085 /* XXX: only parse it if first char on line ? */
1086 if (return_on_interleaved_data) {
1089 ff_rtsp_skip_packet(s);
1090 } else if (ch != '\r') {
1091 if ((q - buf) < sizeof(buf) - 1)
1097 av_dlog(s, "line='%s'\n", buf);
1099 /* test if last line */
1103 if (line_count == 0) {
1104 /* get reply code */
1105 get_word(buf1, sizeof(buf1), &p);
1106 if (!strncmp(buf1, "RTSP/", 5)) {
1107 get_word(buf1, sizeof(buf1), &p);
1108 reply->status_code = atoi(buf1);
1109 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1111 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1112 get_word(buf1, sizeof(buf1), &p); // object
1116 ff_rtsp_parse_line(reply, p, rt, method);
1117 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1118 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1123 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1124 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1126 content_length = reply->content_length;
1127 if (content_length > 0) {
1128 /* leave some room for a trailing '\0' (useful for simple parsing) */
1129 content = av_malloc(content_length + 1);
1130 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1131 content[content_length] = '\0';
1134 *content_ptr = content;
1140 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1141 const char* ptr = buf;
1143 if (!strcmp(reply->reason, "OPTIONS")) {
1144 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1146 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1147 if (reply->session_id[0])
1148 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1151 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1153 av_strlcat(buf, "\r\n", sizeof(buf));
1155 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1156 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1159 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1161 rt->last_cmd_time = av_gettime();
1162 /* Even if the request from the server had data, it is not the data
1163 * that the caller wants or expects. The memory could also be leaked
1164 * if the actual following reply has content data. */
1166 av_freep(content_ptr);
1167 /* If method is set, this is called from ff_rtsp_send_cmd,
1168 * where a reply to exactly this request is awaited. For
1169 * callers from within packet receiving, we just want to
1170 * return to the caller and go back to receiving packets. */
1176 if (rt->seq != reply->seq) {
1177 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1178 rt->seq, reply->seq);
1182 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1183 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1184 reply->notice == 2306 /* Continuous Feed Terminated */) {
1185 rt->state = RTSP_STATE_IDLE;
1186 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1187 return AVERROR(EIO); /* data or server error */
1188 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1189 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1190 return AVERROR(EPERM);
1196 * Send a command to the RTSP server without waiting for the reply.
1198 * @param s RTSP (de)muxer context
1199 * @param method the method for the request
1200 * @param url the target url for the request
1201 * @param headers extra header lines to include in the request
1202 * @param send_content if non-null, the data to send as request body content
1203 * @param send_content_length the length of the send_content data, or 0 if
1204 * send_content is null
1206 * @return zero if success, nonzero otherwise
1208 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1209 const char *method, const char *url,
1210 const char *headers,
1211 const unsigned char *send_content,
1212 int send_content_length)
1214 RTSPState *rt = s->priv_data;
1215 char buf[4096], *out_buf;
1216 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1218 /* Add in RTSP headers */
1221 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1223 av_strlcat(buf, headers, sizeof(buf));
1224 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1225 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1226 if (rt->session_id[0] != '\0' && (!headers ||
1227 !strstr(headers, "\nIf-Match:"))) {
1228 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1231 char *str = ff_http_auth_create_response(&rt->auth_state,
1232 rt->auth, url, method);
1234 av_strlcat(buf, str, sizeof(buf));
1237 if (send_content_length > 0 && send_content)
1238 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1239 av_strlcat(buf, "\r\n", sizeof(buf));
1241 /* base64 encode rtsp if tunneling */
1242 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1243 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1244 out_buf = base64buf;
1247 av_dlog(s, "Sending:\n%s--\n", buf);
1249 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1250 if (send_content_length > 0 && send_content) {
1251 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1252 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1253 "with content data not supported\n");
1254 return AVERROR_PATCHWELCOME;
1256 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1258 rt->last_cmd_time = av_gettime();
1263 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1264 const char *url, const char *headers)
1266 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1269 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1270 const char *headers, RTSPMessageHeader *reply,
1271 unsigned char **content_ptr)
1273 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1274 content_ptr, NULL, 0);
1277 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1278 const char *method, const char *url,
1280 RTSPMessageHeader *reply,
1281 unsigned char **content_ptr,
1282 const unsigned char *send_content,
1283 int send_content_length)
1285 RTSPState *rt = s->priv_data;
1286 HTTPAuthType cur_auth_type;
1287 int ret, attempts = 0;
1290 cur_auth_type = rt->auth_state.auth_type;
1291 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1293 send_content_length)))
1296 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1300 if (reply->status_code == 401 &&
1301 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1302 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1305 if (reply->status_code > 400){
1306 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1310 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1316 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1317 int lower_transport, const char *real_challenge)
1319 RTSPState *rt = s->priv_data;
1320 int rtx = 0, j, i, err, interleave = 0, port_off;
1321 RTSPStream *rtsp_st;
1322 RTSPMessageHeader reply1, *reply = &reply1;
1324 const char *trans_pref;
1326 if (rt->transport == RTSP_TRANSPORT_RDT)
1327 trans_pref = "x-pn-tng";
1328 else if (rt->transport == RTSP_TRANSPORT_RAW)
1329 trans_pref = "RAW/RAW";
1331 trans_pref = "RTP/AVP";
1333 /* default timeout: 1 minute */
1336 /* Choose a random starting offset within the first half of the
1337 * port range, to allow for a number of ports to try even if the offset
1338 * happens to be at the end of the random range. */
1339 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1340 /* even random offset */
1341 port_off -= port_off & 0x01;
1343 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1344 char transport[2048];
1347 * WMS serves all UDP data over a single connection, the RTX, which
1348 * isn't necessarily the first in the SDP but has to be the first
1349 * to be set up, else the second/third SETUP will fail with a 461.
1351 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1352 rt->server_type == RTSP_SERVER_WMS) {
1355 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1356 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1358 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1362 if (rtx == rt->nb_rtsp_streams)
1363 return -1; /* no RTX found */
1364 rtsp_st = rt->rtsp_streams[rtx];
1366 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1368 rtsp_st = rt->rtsp_streams[i];
1371 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1374 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1375 port = reply->transports[0].client_port_min;
1379 /* first try in specified port range */
1380 while (j <= rt->rtp_port_max) {
1381 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1382 "?localport=%d", j);
1383 /* we will use two ports per rtp stream (rtp and rtcp) */
1385 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1386 &s->interrupt_callback, NULL))
1389 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1394 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1396 snprintf(transport, sizeof(transport) - 1,
1397 "%s/UDP;", trans_pref);
1398 if (rt->server_type != RTSP_SERVER_REAL)
1399 av_strlcat(transport, "unicast;", sizeof(transport));
1400 av_strlcatf(transport, sizeof(transport),
1401 "client_port=%d", port);
1402 if (rt->transport == RTSP_TRANSPORT_RTP &&
1403 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1404 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1408 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1409 /* For WMS streams, the application streams are only used for
1410 * UDP. When trying to set it up for TCP streams, the server
1411 * will return an error. Therefore, we skip those streams. */
1412 if (rt->server_type == RTSP_SERVER_WMS &&
1413 (rtsp_st->stream_index < 0 ||
1414 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1417 snprintf(transport, sizeof(transport) - 1,
1418 "%s/TCP;", trans_pref);
1419 if (rt->transport != RTSP_TRANSPORT_RDT)
1420 av_strlcat(transport, "unicast;", sizeof(transport));
1421 av_strlcatf(transport, sizeof(transport),
1422 "interleaved=%d-%d",
1423 interleave, interleave + 1);
1427 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1428 snprintf(transport, sizeof(transport) - 1,
1429 "%s/UDP;multicast", trans_pref);
1432 av_strlcat(transport, ";mode=record", sizeof(transport));
1433 } else if (rt->server_type == RTSP_SERVER_REAL ||
1434 rt->server_type == RTSP_SERVER_WMS)
1435 av_strlcat(transport, ";mode=play", sizeof(transport));
1436 snprintf(cmd, sizeof(cmd),
1437 "Transport: %s\r\n",
1439 if (rt->accept_dynamic_rate)
1440 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1441 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1442 char real_res[41], real_csum[9];
1443 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1445 av_strlcatf(cmd, sizeof(cmd),
1447 "RealChallenge2: %s, sd=%s\r\n",
1448 rt->session_id, real_res, real_csum);
1450 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1451 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1454 } else if (reply->status_code != RTSP_STATUS_OK ||
1455 reply->nb_transports != 1) {
1456 err = AVERROR_INVALIDDATA;
1460 /* XXX: same protocol for all streams is required */
1462 if (reply->transports[0].lower_transport != rt->lower_transport ||
1463 reply->transports[0].transport != rt->transport) {
1464 err = AVERROR_INVALIDDATA;
1468 rt->lower_transport = reply->transports[0].lower_transport;
1469 rt->transport = reply->transports[0].transport;
1472 /* Fail if the server responded with another lower transport mode
1473 * than what we requested. */
1474 if (reply->transports[0].lower_transport != lower_transport) {
1475 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1476 err = AVERROR_INVALIDDATA;
1480 switch(reply->transports[0].lower_transport) {
1481 case RTSP_LOWER_TRANSPORT_TCP:
1482 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1483 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1486 case RTSP_LOWER_TRANSPORT_UDP: {
1487 char url[1024], options[30] = "";
1488 const char *peer = host;
1490 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1491 av_strlcpy(options, "?connect=1", sizeof(options));
1492 /* Use source address if specified */
1493 if (reply->transports[0].source[0])
1494 peer = reply->transports[0].source;
1495 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1496 reply->transports[0].server_port_min, "%s", options);
1497 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1498 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1499 err = AVERROR_INVALIDDATA;
1502 /* Try to initialize the connection state in a
1503 * potential NAT router by sending dummy packets.
1504 * RTP/RTCP dummy packets are used for RDT, too.
1506 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1508 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1511 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1512 char url[1024], namebuf[50], optbuf[20] = "";
1513 struct sockaddr_storage addr;
1516 if (reply->transports[0].destination.ss_family) {
1517 addr = reply->transports[0].destination;
1518 port = reply->transports[0].port_min;
1519 ttl = reply->transports[0].ttl;
1521 addr = rtsp_st->sdp_ip;
1522 port = rtsp_st->sdp_port;
1523 ttl = rtsp_st->sdp_ttl;
1526 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1527 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1528 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1529 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1530 port, "%s", optbuf);
1531 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1532 &s->interrupt_callback, NULL) < 0) {
1533 err = AVERROR_INVALIDDATA;
1540 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1544 if (rt->nb_rtsp_streams && reply->timeout > 0)
1545 rt->timeout = reply->timeout;
1547 if (rt->server_type == RTSP_SERVER_REAL)
1548 rt->need_subscription = 1;
1553 ff_rtsp_undo_setup(s, 0);
1557 void ff_rtsp_close_connections(AVFormatContext *s)
1559 RTSPState *rt = s->priv_data;
1560 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1561 ffurl_close(rt->rtsp_hd);
1562 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1565 int ff_rtsp_connect(AVFormatContext *s)
1567 RTSPState *rt = s->priv_data;
1568 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1569 int port, err, tcp_fd;
1570 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1571 int lower_transport_mask = 0;
1572 char real_challenge[64] = "";
1573 struct sockaddr_storage peer;
1574 socklen_t peer_len = sizeof(peer);
1576 if (rt->rtp_port_max < rt->rtp_port_min) {
1577 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1578 "than min port %d\n", rt->rtp_port_max,
1580 return AVERROR(EINVAL);
1583 if (!ff_network_init())
1584 return AVERROR(EIO);
1586 if (s->max_delay < 0) /* Not set by the caller */
1587 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1589 rt->control_transport = RTSP_MODE_PLAIN;
1590 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1591 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1592 rt->control_transport = RTSP_MODE_TUNNEL;
1594 /* Only pass through valid flags from here */
1595 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1598 lower_transport_mask = rt->lower_transport_mask;
1599 /* extract hostname and port */
1600 av_url_split(NULL, 0, auth, sizeof(auth),
1601 host, sizeof(host), &port, path, sizeof(path), s->filename);
1603 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1606 port = RTSP_DEFAULT_PORT;
1608 if (!lower_transport_mask)
1609 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1612 /* Only UDP or TCP - UDP multicast isn't supported. */
1613 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1614 (1 << RTSP_LOWER_TRANSPORT_TCP);
1615 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1616 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1617 "only UDP and TCP are supported for output.\n");
1618 err = AVERROR(EINVAL);
1623 /* Construct the URI used in request; this is similar to s->filename,
1624 * but with authentication credentials removed and RTSP specific options
1626 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1627 host, port, "%s", path);
1629 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1630 /* set up initial handshake for tunneling */
1631 char httpname[1024];
1632 char sessioncookie[17];
1635 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1636 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1637 av_get_random_seed(), av_get_random_seed());
1640 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1641 &s->interrupt_callback) < 0) {
1646 /* generate GET headers */
1647 snprintf(headers, sizeof(headers),
1648 "x-sessioncookie: %s\r\n"
1649 "Accept: application/x-rtsp-tunnelled\r\n"
1650 "Pragma: no-cache\r\n"
1651 "Cache-Control: no-cache\r\n",
1653 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1655 /* complete the connection */
1656 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1662 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1663 &s->interrupt_callback) < 0 ) {
1668 /* generate POST headers */
1669 snprintf(headers, sizeof(headers),
1670 "x-sessioncookie: %s\r\n"
1671 "Content-Type: application/x-rtsp-tunnelled\r\n"
1672 "Pragma: no-cache\r\n"
1673 "Cache-Control: no-cache\r\n"
1674 "Content-Length: 32767\r\n"
1675 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1677 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1678 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1680 /* Initialize the authentication state for the POST session. The HTTP
1681 * protocol implementation doesn't properly handle multi-pass
1682 * authentication for POST requests, since it would require one of
1684 * - implementing Expect: 100-continue, which many HTTP servers
1685 * don't support anyway, even less the RTSP servers that do HTTP
1687 * - sending the whole POST data until getting a 401 reply specifying
1688 * what authentication method to use, then resending all that data
1689 * - waiting for potential 401 replies directly after sending the
1690 * POST header (waiting for some unspecified time)
1691 * Therefore, we copy the full auth state, which works for both basic
1692 * and digest. (For digest, we would have to synchronize the nonce
1693 * count variable between the two sessions, if we'd do more requests
1694 * with the original session, though.)
1696 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1698 /* complete the connection */
1699 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1704 /* open the tcp connection */
1705 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1706 "?timeout=%d", rt->stimeout);
1707 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1708 &s->interrupt_callback, NULL) < 0) {
1712 rt->rtsp_hd_out = rt->rtsp_hd;
1716 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1717 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1718 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1719 NULL, 0, NI_NUMERICHOST);
1722 /* request options supported by the server; this also detects server
1724 for (rt->server_type = RTSP_SERVER_RTP;;) {
1726 if (rt->server_type == RTSP_SERVER_REAL)
1729 * The following entries are required for proper
1730 * streaming from a Realmedia server. They are
1731 * interdependent in some way although we currently
1732 * don't quite understand how. Values were copied
1733 * from mplayer SVN r23589.
1734 * ClientChallenge is a 16-byte ID in hex
1735 * CompanyID is a 16-byte ID in base64
1737 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1738 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1739 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1740 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1742 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1743 if (reply->status_code != RTSP_STATUS_OK) {
1744 err = AVERROR_INVALIDDATA;
1748 /* detect server type if not standard-compliant RTP */
1749 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1750 rt->server_type = RTSP_SERVER_REAL;
1752 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1753 rt->server_type = RTSP_SERVER_WMS;
1754 } else if (rt->server_type == RTSP_SERVER_REAL)
1755 strcpy(real_challenge, reply->real_challenge);
1759 if (s->iformat && CONFIG_RTSP_DEMUXER)
1760 err = ff_rtsp_setup_input_streams(s, reply);
1761 else if (CONFIG_RTSP_MUXER)
1762 err = ff_rtsp_setup_output_streams(s, host);
1767 int lower_transport = ff_log2_tab[lower_transport_mask &
1768 ~(lower_transport_mask - 1)];
1770 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1771 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1772 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1774 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1775 rt->server_type == RTSP_SERVER_REAL ?
1776 real_challenge : NULL);
1779 lower_transport_mask &= ~(1 << lower_transport);
1780 if (lower_transport_mask == 0 && err == 1) {
1781 err = AVERROR(EPROTONOSUPPORT);
1786 rt->lower_transport_mask = lower_transport_mask;
1787 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1788 rt->state = RTSP_STATE_IDLE;
1789 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1792 ff_rtsp_close_streams(s);
1793 ff_rtsp_close_connections(s);
1794 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1795 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1796 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1804 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1807 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1808 uint8_t *buf, int buf_size, int64_t wait_end)
1810 RTSPState *rt = s->priv_data;
1811 RTSPStream *rtsp_st;
1812 int n, i, ret, tcp_fd, timeout_cnt = 0;
1814 struct pollfd *p = rt->p;
1815 int *fds = NULL, fdsnum, fdsidx;
1818 if (ff_check_interrupt(&s->interrupt_callback))
1819 return AVERROR_EXIT;
1820 if (wait_end && wait_end - av_gettime() < 0)
1821 return AVERROR(EAGAIN);
1824 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1825 p[max_p].fd = tcp_fd;
1826 p[max_p++].events = POLLIN;
1830 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1831 rtsp_st = rt->rtsp_streams[i];
1832 if (rtsp_st->rtp_handle) {
1833 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1835 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1839 av_log(s, AV_LOG_ERROR,
1840 "Number of fds %d not supported\n", fdsnum);
1841 return AVERROR_INVALIDDATA;
1843 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1844 p[max_p].fd = fds[fdsidx];
1845 p[max_p++].events = POLLIN;
1850 n = poll(p, max_p, POLL_TIMEOUT_MS);
1852 int j = 1 - (tcp_fd == -1);
1854 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1855 rtsp_st = rt->rtsp_streams[i];
1856 if (rtsp_st->rtp_handle) {
1857 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1858 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1860 *prtsp_st = rtsp_st;
1867 #if CONFIG_RTSP_DEMUXER
1868 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1869 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1870 if (rt->state == RTSP_STATE_STREAMING) {
1871 if (!ff_rtsp_parse_streaming_commands(s))
1874 av_log(s, AV_LOG_WARNING,
1875 "Unable to answer to TEARDOWN\n");
1879 RTSPMessageHeader reply;
1880 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1883 /* XXX: parse message */
1884 if (rt->state != RTSP_STATE_STREAMING)
1889 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1890 return AVERROR(ETIMEDOUT);
1891 } else if (n < 0 && errno != EINTR)
1892 return AVERROR(errno);
1896 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1897 const uint8_t *buf, int len)
1899 RTSPState *rt = s->priv_data;
1903 if (rt->nb_rtsp_streams == 1) {
1904 *rtsp_st = rt->rtsp_streams[0];
1907 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1908 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1910 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1911 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1914 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1915 *rtsp_st = rt->rtsp_streams[i];
1922 av_log(s, AV_LOG_WARNING,
1923 "Unable to pick stream for packet - SSRC not known for "
1925 return AVERROR(EAGAIN);
1928 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1929 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1930 *rtsp_st = rt->rtsp_streams[i];
1936 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1937 return AVERROR(EAGAIN);
1940 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1942 RTSPState *rt = s->priv_data;
1944 RTSPStream *rtsp_st, *first_queue_st = NULL;
1945 int64_t wait_end = 0;
1947 if (rt->nb_byes == rt->nb_rtsp_streams)
1950 /* get next frames from the same RTP packet */
1951 if (rt->cur_transport_priv) {
1952 if (rt->transport == RTSP_TRANSPORT_RDT) {
1953 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1954 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1955 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1956 } else if (rt->ts && CONFIG_RTPDEC) {
1957 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1959 rt->recvbuf_pos += ret;
1960 ret = rt->recvbuf_pos < rt->recvbuf_len;
1965 rt->cur_transport_priv = NULL;
1967 } else if (ret == 1) {
1970 rt->cur_transport_priv = NULL;
1974 if (rt->transport == RTSP_TRANSPORT_RTP) {
1976 int64_t first_queue_time = 0;
1977 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1978 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1982 queue_time = ff_rtp_queued_packet_time(rtpctx);
1983 if (queue_time && (queue_time - first_queue_time < 0 ||
1984 !first_queue_time)) {
1985 first_queue_time = queue_time;
1986 first_queue_st = rt->rtsp_streams[i];
1989 if (first_queue_time) {
1990 wait_end = first_queue_time + s->max_delay;
1993 first_queue_st = NULL;
1997 /* read next RTP packet */
1999 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2001 return AVERROR(ENOMEM);
2004 switch(rt->lower_transport) {
2006 #if CONFIG_RTSP_DEMUXER
2007 case RTSP_LOWER_TRANSPORT_TCP:
2008 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2011 case RTSP_LOWER_TRANSPORT_UDP:
2012 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2013 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2014 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2015 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2017 case RTSP_LOWER_TRANSPORT_CUSTOM:
2018 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2019 wait_end && wait_end < av_gettime())
2020 len = AVERROR(EAGAIN);
2022 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2023 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2024 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2025 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2028 if (len == AVERROR(EAGAIN) && first_queue_st &&
2029 rt->transport == RTSP_TRANSPORT_RTP) {
2030 rtsp_st = first_queue_st;
2031 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2038 if (rt->transport == RTSP_TRANSPORT_RDT) {
2039 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2040 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2041 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2042 if (rtsp_st->feedback) {
2043 AVIOContext *pb = NULL;
2044 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2046 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2049 /* Either bad packet, or a RTCP packet. Check if the
2050 * first_rtcp_ntp_time field was initialized. */
2051 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2052 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2053 /* first_rtcp_ntp_time has been initialized for this stream,
2054 * copy the same value to all other uninitialized streams,
2055 * in order to map their timestamp origin to the same ntp time
2058 AVStream *st = NULL;
2059 if (rtsp_st->stream_index >= 0)
2060 st = s->streams[rtsp_st->stream_index];
2061 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2062 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2063 AVStream *st2 = NULL;
2064 if (rt->rtsp_streams[i]->stream_index >= 0)
2065 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2066 if (rtpctx2 && st && st2 &&
2067 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2068 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2069 rtpctx2->rtcp_ts_offset = av_rescale_q(
2070 rtpctx->rtcp_ts_offset, st->time_base,
2075 if (ret == -RTCP_BYE) {
2078 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2079 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2081 if (rt->nb_byes == rt->nb_rtsp_streams)
2085 } else if (rt->ts && CONFIG_RTPDEC) {
2086 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2089 rt->recvbuf_len = len;
2090 rt->recvbuf_pos = ret;
2091 rt->cur_transport_priv = rt->ts;
2098 return AVERROR_INVALIDDATA;
2104 /* more packets may follow, so we save the RTP context */
2105 rt->cur_transport_priv = rtsp_st->transport_priv;
2109 #endif /* CONFIG_RTPDEC */
2111 #if CONFIG_SDP_DEMUXER
2112 static int sdp_probe(AVProbeData *p1)
2114 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2116 /* we look for a line beginning "c=IN IP" */
2117 while (p < p_end && *p != '\0') {
2118 if (p + sizeof("c=IN IP") - 1 < p_end &&
2119 av_strstart(p, "c=IN IP", NULL))
2120 return AVPROBE_SCORE_EXTENSION;
2122 while (p < p_end - 1 && *p != '\n') p++;
2131 static void append_source_addrs(char *buf, int size, const char *name,
2132 int count, struct RTSPSource **addrs)
2137 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2138 for (i = 1; i < count; i++)
2139 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2142 static int sdp_read_header(AVFormatContext *s)
2144 RTSPState *rt = s->priv_data;
2145 RTSPStream *rtsp_st;
2150 if (!ff_network_init())
2151 return AVERROR(EIO);
2153 if (s->max_delay < 0) /* Not set by the caller */
2154 s->max_delay = DEFAULT_REORDERING_DELAY;
2155 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2156 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2158 /* read the whole sdp file */
2159 /* XXX: better loading */
2160 content = av_malloc(SDP_MAX_SIZE);
2161 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2164 return AVERROR_INVALIDDATA;
2166 content[size] ='\0';
2168 err = ff_sdp_parse(s, content);
2172 /* open each RTP stream */
2173 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2175 rtsp_st = rt->rtsp_streams[i];
2177 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2178 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2179 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2180 ff_url_join(url, sizeof(url), "rtp", NULL,
2181 namebuf, rtsp_st->sdp_port,
2182 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2183 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2184 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2185 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2187 append_source_addrs(url, sizeof(url), "sources",
2188 rtsp_st->nb_include_source_addrs,
2189 rtsp_st->include_source_addrs);
2190 append_source_addrs(url, sizeof(url), "block",
2191 rtsp_st->nb_exclude_source_addrs,
2192 rtsp_st->exclude_source_addrs);
2193 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2194 &s->interrupt_callback, NULL) < 0) {
2195 err = AVERROR_INVALIDDATA;
2199 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2204 ff_rtsp_close_streams(s);
2209 static int sdp_read_close(AVFormatContext *s)
2211 ff_rtsp_close_streams(s);
2216 static const AVClass sdp_demuxer_class = {
2217 .class_name = "SDP demuxer",
2218 .item_name = av_default_item_name,
2219 .option = sdp_options,
2220 .version = LIBAVUTIL_VERSION_INT,
2223 AVInputFormat ff_sdp_demuxer = {
2225 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2226 .priv_data_size = sizeof(RTSPState),
2227 .read_probe = sdp_probe,
2228 .read_header = sdp_read_header,
2229 .read_packet = ff_rtsp_fetch_packet,
2230 .read_close = sdp_read_close,
2231 .priv_class = &sdp_demuxer_class,
2233 #endif /* CONFIG_SDP_DEMUXER */
2235 #if CONFIG_RTP_DEMUXER
2236 static int rtp_probe(AVProbeData *p)
2238 if (av_strstart(p->filename, "rtp:", NULL))
2239 return AVPROBE_SCORE_MAX;
2243 static int rtp_read_header(AVFormatContext *s)
2245 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2246 char host[500], sdp[500];
2248 URLContext* in = NULL;
2250 AVCodecContext codec = { 0 };
2251 struct sockaddr_storage addr;
2253 socklen_t addrlen = sizeof(addr);
2254 RTSPState *rt = s->priv_data;
2256 if (!ff_network_init())
2257 return AVERROR(EIO);
2259 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2260 &s->interrupt_callback, NULL);
2265 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2266 if (ret == AVERROR(EAGAIN))
2271 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2275 if ((recvbuf[0] & 0xc0) != 0x80) {
2276 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2281 if (RTP_PT_IS_RTCP(recvbuf[1]))
2284 payload_type = recvbuf[1] & 0x7f;
2287 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2291 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2292 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2293 "without an SDP file describing it\n",
2297 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2298 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2299 "properly you need an SDP file "
2303 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2304 NULL, 0, s->filename);
2306 snprintf(sdp, sizeof(sdp),
2307 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2308 addr.ss_family == AF_INET ? 4 : 6, host,
2309 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2310 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2311 port, payload_type);
2312 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2314 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2317 /* sdp_read_header initializes this again */
2320 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2322 ret = sdp_read_header(s);
2333 static const AVClass rtp_demuxer_class = {
2334 .class_name = "RTP demuxer",
2335 .item_name = av_default_item_name,
2336 .option = rtp_options,
2337 .version = LIBAVUTIL_VERSION_INT,
2340 AVInputFormat ff_rtp_demuxer = {
2342 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2343 .priv_data_size = sizeof(RTSPState),
2344 .read_probe = rtp_probe,
2345 .read_header = rtp_read_header,
2346 .read_packet = ff_rtsp_fetch_packet,
2347 .read_close = sdp_read_close,
2348 .flags = AVFMT_NOFILE,
2349 .priv_class = &rtp_demuxer_class,
2351 #endif /* CONFIG_RTP_DEMUXER */