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 st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;
742 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
743 return 0; // Don't need to open any parser here
744 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
745 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
746 rtsp_st->dynamic_protocol_context,
747 rtsp_st->dynamic_handler);
748 else if (CONFIG_RTPDEC)
749 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
750 rtsp_st->sdp_payload_type,
751 reordering_queue_size);
753 if (!rtsp_st->transport_priv) {
754 return AVERROR(ENOMEM);
755 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
756 if (rtsp_st->dynamic_handler) {
757 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
758 rtsp_st->dynamic_protocol_context,
759 rtsp_st->dynamic_handler);
761 if (rtsp_st->crypto_suite[0])
762 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
763 rtsp_st->crypto_suite,
764 rtsp_st->crypto_params);
770 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
771 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
778 q += strspn(q, SPACE_CHARS);
779 v = strtol(q, &p, 10);
783 v = strtol(p, &p, 10);
792 /* XXX: only one transport specification is parsed */
793 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
795 char transport_protocol[16];
797 char lower_transport[16];
799 RTSPTransportField *th;
802 reply->nb_transports = 0;
805 p += strspn(p, SPACE_CHARS);
809 th = &reply->transports[reply->nb_transports];
811 get_word_sep(transport_protocol, sizeof(transport_protocol),
813 if (!av_strcasecmp (transport_protocol, "rtp")) {
814 get_word_sep(profile, sizeof(profile), "/;,", &p);
815 lower_transport[0] = '\0';
816 /* rtp/avp/<protocol> */
818 get_word_sep(lower_transport, sizeof(lower_transport),
821 th->transport = RTSP_TRANSPORT_RTP;
822 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
823 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
824 /* x-pn-tng/<protocol> */
825 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
827 th->transport = RTSP_TRANSPORT_RDT;
828 } else if (!av_strcasecmp(transport_protocol, "raw")) {
829 get_word_sep(profile, sizeof(profile), "/;,", &p);
830 lower_transport[0] = '\0';
831 /* raw/raw/<protocol> */
833 get_word_sep(lower_transport, sizeof(lower_transport),
836 th->transport = RTSP_TRANSPORT_RAW;
838 if (!av_strcasecmp(lower_transport, "TCP"))
839 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
841 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
845 /* get each parameter */
846 while (*p != '\0' && *p != ',') {
847 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
848 if (!strcmp(parameter, "port")) {
851 rtsp_parse_range(&th->port_min, &th->port_max, &p);
853 } else if (!strcmp(parameter, "client_port")) {
856 rtsp_parse_range(&th->client_port_min,
857 &th->client_port_max, &p);
859 } else if (!strcmp(parameter, "server_port")) {
862 rtsp_parse_range(&th->server_port_min,
863 &th->server_port_max, &p);
865 } else if (!strcmp(parameter, "interleaved")) {
868 rtsp_parse_range(&th->interleaved_min,
869 &th->interleaved_max, &p);
871 } else if (!strcmp(parameter, "multicast")) {
872 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
873 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
874 } else if (!strcmp(parameter, "ttl")) {
878 th->ttl = strtol(p, &end, 10);
881 } else if (!strcmp(parameter, "destination")) {
884 get_word_sep(buf, sizeof(buf), ";,", &p);
885 get_sockaddr(buf, &th->destination);
887 } else if (!strcmp(parameter, "source")) {
890 get_word_sep(buf, sizeof(buf), ";,", &p);
891 av_strlcpy(th->source, buf, sizeof(th->source));
893 } else if (!strcmp(parameter, "mode")) {
896 get_word_sep(buf, sizeof(buf), ";, ", &p);
897 if (!strcmp(buf, "record") ||
898 !strcmp(buf, "receive"))
903 while (*p != ';' && *p != '\0' && *p != ',')
911 reply->nb_transports++;
915 static void handle_rtp_info(RTSPState *rt, const char *url,
916 uint32_t seq, uint32_t rtptime)
919 if (!rtptime || !url[0])
921 if (rt->transport != RTSP_TRANSPORT_RTP)
923 for (i = 0; i < rt->nb_rtsp_streams; i++) {
924 RTSPStream *rtsp_st = rt->rtsp_streams[i];
925 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
928 if (!strcmp(rtsp_st->control_url, url)) {
929 rtpctx->base_timestamp = rtptime;
935 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
938 char key[20], value[1024], url[1024] = "";
939 uint32_t seq = 0, rtptime = 0;
942 p += strspn(p, SPACE_CHARS);
945 get_word_sep(key, sizeof(key), "=", &p);
949 get_word_sep(value, sizeof(value), ";, ", &p);
951 if (!strcmp(key, "url"))
952 av_strlcpy(url, value, sizeof(url));
953 else if (!strcmp(key, "seq"))
954 seq = strtoul(value, NULL, 10);
955 else if (!strcmp(key, "rtptime"))
956 rtptime = strtoul(value, NULL, 10);
958 handle_rtp_info(rt, url, seq, rtptime);
967 handle_rtp_info(rt, url, seq, rtptime);
970 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
971 RTSPState *rt, const char *method)
975 /* NOTE: we do case independent match for broken servers */
977 if (av_stristart(p, "Session:", &p)) {
979 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
980 if (av_stristart(p, ";timeout=", &p) &&
981 (t = strtol(p, NULL, 10)) > 0) {
984 } else if (av_stristart(p, "Content-Length:", &p)) {
985 reply->content_length = strtol(p, NULL, 10);
986 } else if (av_stristart(p, "Transport:", &p)) {
987 rtsp_parse_transport(reply, p);
988 } else if (av_stristart(p, "CSeq:", &p)) {
989 reply->seq = strtol(p, NULL, 10);
990 } else if (av_stristart(p, "Range:", &p)) {
991 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
992 } else if (av_stristart(p, "RealChallenge1:", &p)) {
993 p += strspn(p, SPACE_CHARS);
994 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
995 } else if (av_stristart(p, "Server:", &p)) {
996 p += strspn(p, SPACE_CHARS);
997 av_strlcpy(reply->server, p, sizeof(reply->server));
998 } else if (av_stristart(p, "Notice:", &p) ||
999 av_stristart(p, "X-Notice:", &p)) {
1000 reply->notice = strtol(p, NULL, 10);
1001 } else if (av_stristart(p, "Location:", &p)) {
1002 p += strspn(p, SPACE_CHARS);
1003 av_strlcpy(reply->location, p , sizeof(reply->location));
1004 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1005 p += strspn(p, SPACE_CHARS);
1006 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1007 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1008 p += strspn(p, SPACE_CHARS);
1009 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1010 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1011 p += strspn(p, SPACE_CHARS);
1012 if (method && !strcmp(method, "DESCRIBE"))
1013 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1014 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1015 p += strspn(p, SPACE_CHARS);
1016 if (method && !strcmp(method, "PLAY"))
1017 rtsp_parse_rtp_info(rt, p);
1018 } else if (av_stristart(p, "Public:", &p) && rt) {
1019 if (strstr(p, "GET_PARAMETER") &&
1020 method && !strcmp(method, "OPTIONS"))
1021 rt->get_parameter_supported = 1;
1022 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1023 p += strspn(p, SPACE_CHARS);
1024 rt->accept_dynamic_rate = atoi(p);
1025 } else if (av_stristart(p, "Content-Type:", &p)) {
1026 p += strspn(p, SPACE_CHARS);
1027 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1031 /* skip a RTP/TCP interleaved packet */
1032 void ff_rtsp_skip_packet(AVFormatContext *s)
1034 RTSPState *rt = s->priv_data;
1038 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1041 len = AV_RB16(buf + 1);
1043 av_dlog(s, "skipping RTP packet len=%d\n", len);
1048 if (len1 > sizeof(buf))
1050 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1057 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1058 unsigned char **content_ptr,
1059 int return_on_interleaved_data, const char *method)
1061 RTSPState *rt = s->priv_data;
1062 char buf[4096], buf1[1024], *q;
1065 int ret, content_length, line_count = 0, request = 0;
1066 unsigned char *content = NULL;
1072 memset(reply, 0, sizeof(*reply));
1074 /* parse reply (XXX: use buffers) */
1075 rt->last_reply[0] = '\0';
1079 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1080 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1086 /* XXX: only parse it if first char on line ? */
1087 if (return_on_interleaved_data) {
1090 ff_rtsp_skip_packet(s);
1091 } else if (ch != '\r') {
1092 if ((q - buf) < sizeof(buf) - 1)
1098 av_dlog(s, "line='%s'\n", buf);
1100 /* test if last line */
1104 if (line_count == 0) {
1105 /* get reply code */
1106 get_word(buf1, sizeof(buf1), &p);
1107 if (!strncmp(buf1, "RTSP/", 5)) {
1108 get_word(buf1, sizeof(buf1), &p);
1109 reply->status_code = atoi(buf1);
1110 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1112 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1113 get_word(buf1, sizeof(buf1), &p); // object
1117 ff_rtsp_parse_line(reply, p, rt, method);
1118 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1119 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1124 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1125 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1127 content_length = reply->content_length;
1128 if (content_length > 0) {
1129 /* leave some room for a trailing '\0' (useful for simple parsing) */
1130 content = av_malloc(content_length + 1);
1131 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1132 content[content_length] = '\0';
1135 *content_ptr = content;
1141 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1142 const char* ptr = buf;
1144 if (!strcmp(reply->reason, "OPTIONS")) {
1145 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1147 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1148 if (reply->session_id[0])
1149 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1152 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1154 av_strlcat(buf, "\r\n", sizeof(buf));
1156 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1157 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1160 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1162 rt->last_cmd_time = av_gettime();
1163 /* Even if the request from the server had data, it is not the data
1164 * that the caller wants or expects. The memory could also be leaked
1165 * if the actual following reply has content data. */
1167 av_freep(content_ptr);
1168 /* If method is set, this is called from ff_rtsp_send_cmd,
1169 * where a reply to exactly this request is awaited. For
1170 * callers from within packet receiving, we just want to
1171 * return to the caller and go back to receiving packets. */
1177 if (rt->seq != reply->seq) {
1178 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1179 rt->seq, reply->seq);
1183 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1184 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1185 reply->notice == 2306 /* Continuous Feed Terminated */) {
1186 rt->state = RTSP_STATE_IDLE;
1187 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1188 return AVERROR(EIO); /* data or server error */
1189 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1190 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1191 return AVERROR(EPERM);
1197 * Send a command to the RTSP server without waiting for the reply.
1199 * @param s RTSP (de)muxer context
1200 * @param method the method for the request
1201 * @param url the target url for the request
1202 * @param headers extra header lines to include in the request
1203 * @param send_content if non-null, the data to send as request body content
1204 * @param send_content_length the length of the send_content data, or 0 if
1205 * send_content is null
1207 * @return zero if success, nonzero otherwise
1209 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1210 const char *method, const char *url,
1211 const char *headers,
1212 const unsigned char *send_content,
1213 int send_content_length)
1215 RTSPState *rt = s->priv_data;
1216 char buf[4096], *out_buf;
1217 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1219 /* Add in RTSP headers */
1222 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1224 av_strlcat(buf, headers, sizeof(buf));
1225 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1226 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1227 if (rt->session_id[0] != '\0' && (!headers ||
1228 !strstr(headers, "\nIf-Match:"))) {
1229 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1232 char *str = ff_http_auth_create_response(&rt->auth_state,
1233 rt->auth, url, method);
1235 av_strlcat(buf, str, sizeof(buf));
1238 if (send_content_length > 0 && send_content)
1239 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1240 av_strlcat(buf, "\r\n", sizeof(buf));
1242 /* base64 encode rtsp if tunneling */
1243 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1244 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1245 out_buf = base64buf;
1248 av_dlog(s, "Sending:\n%s--\n", buf);
1250 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1251 if (send_content_length > 0 && send_content) {
1252 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1253 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1254 "with content data not supported\n");
1255 return AVERROR_PATCHWELCOME;
1257 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1259 rt->last_cmd_time = av_gettime();
1264 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1265 const char *url, const char *headers)
1267 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1270 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1271 const char *headers, RTSPMessageHeader *reply,
1272 unsigned char **content_ptr)
1274 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1275 content_ptr, NULL, 0);
1278 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1279 const char *method, const char *url,
1281 RTSPMessageHeader *reply,
1282 unsigned char **content_ptr,
1283 const unsigned char *send_content,
1284 int send_content_length)
1286 RTSPState *rt = s->priv_data;
1287 HTTPAuthType cur_auth_type;
1288 int ret, attempts = 0;
1291 cur_auth_type = rt->auth_state.auth_type;
1292 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1294 send_content_length)))
1297 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1301 if (reply->status_code == 401 &&
1302 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1303 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1306 if (reply->status_code > 400){
1307 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1311 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1317 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1318 int lower_transport, const char *real_challenge)
1320 RTSPState *rt = s->priv_data;
1321 int rtx = 0, j, i, err, interleave = 0, port_off;
1322 RTSPStream *rtsp_st;
1323 RTSPMessageHeader reply1, *reply = &reply1;
1325 const char *trans_pref;
1327 if (rt->transport == RTSP_TRANSPORT_RDT)
1328 trans_pref = "x-pn-tng";
1329 else if (rt->transport == RTSP_TRANSPORT_RAW)
1330 trans_pref = "RAW/RAW";
1332 trans_pref = "RTP/AVP";
1334 /* default timeout: 1 minute */
1337 /* Choose a random starting offset within the first half of the
1338 * port range, to allow for a number of ports to try even if the offset
1339 * happens to be at the end of the random range. */
1340 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1341 /* even random offset */
1342 port_off -= port_off & 0x01;
1344 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1345 char transport[2048];
1348 * WMS serves all UDP data over a single connection, the RTX, which
1349 * isn't necessarily the first in the SDP but has to be the first
1350 * to be set up, else the second/third SETUP will fail with a 461.
1352 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1353 rt->server_type == RTSP_SERVER_WMS) {
1356 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1357 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1359 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1363 if (rtx == rt->nb_rtsp_streams)
1364 return -1; /* no RTX found */
1365 rtsp_st = rt->rtsp_streams[rtx];
1367 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1369 rtsp_st = rt->rtsp_streams[i];
1372 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1375 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1376 port = reply->transports[0].client_port_min;
1380 /* first try in specified port range */
1381 while (j <= rt->rtp_port_max) {
1382 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1383 "?localport=%d", j);
1384 /* we will use two ports per rtp stream (rtp and rtcp) */
1386 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1387 &s->interrupt_callback, NULL))
1390 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1395 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1397 snprintf(transport, sizeof(transport) - 1,
1398 "%s/UDP;", trans_pref);
1399 if (rt->server_type != RTSP_SERVER_REAL)
1400 av_strlcat(transport, "unicast;", sizeof(transport));
1401 av_strlcatf(transport, sizeof(transport),
1402 "client_port=%d", port);
1403 if (rt->transport == RTSP_TRANSPORT_RTP &&
1404 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1405 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1409 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1410 /* For WMS streams, the application streams are only used for
1411 * UDP. When trying to set it up for TCP streams, the server
1412 * will return an error. Therefore, we skip those streams. */
1413 if (rt->server_type == RTSP_SERVER_WMS &&
1414 (rtsp_st->stream_index < 0 ||
1415 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1418 snprintf(transport, sizeof(transport) - 1,
1419 "%s/TCP;", trans_pref);
1420 if (rt->transport != RTSP_TRANSPORT_RDT)
1421 av_strlcat(transport, "unicast;", sizeof(transport));
1422 av_strlcatf(transport, sizeof(transport),
1423 "interleaved=%d-%d",
1424 interleave, interleave + 1);
1428 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1429 snprintf(transport, sizeof(transport) - 1,
1430 "%s/UDP;multicast", trans_pref);
1433 av_strlcat(transport, ";mode=record", sizeof(transport));
1434 } else if (rt->server_type == RTSP_SERVER_REAL ||
1435 rt->server_type == RTSP_SERVER_WMS)
1436 av_strlcat(transport, ";mode=play", sizeof(transport));
1437 snprintf(cmd, sizeof(cmd),
1438 "Transport: %s\r\n",
1440 if (rt->accept_dynamic_rate)
1441 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1442 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1443 char real_res[41], real_csum[9];
1444 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1446 av_strlcatf(cmd, sizeof(cmd),
1448 "RealChallenge2: %s, sd=%s\r\n",
1449 rt->session_id, real_res, real_csum);
1451 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1452 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1455 } else if (reply->status_code != RTSP_STATUS_OK ||
1456 reply->nb_transports != 1) {
1457 err = AVERROR_INVALIDDATA;
1461 /* XXX: same protocol for all streams is required */
1463 if (reply->transports[0].lower_transport != rt->lower_transport ||
1464 reply->transports[0].transport != rt->transport) {
1465 err = AVERROR_INVALIDDATA;
1469 rt->lower_transport = reply->transports[0].lower_transport;
1470 rt->transport = reply->transports[0].transport;
1473 /* Fail if the server responded with another lower transport mode
1474 * than what we requested. */
1475 if (reply->transports[0].lower_transport != lower_transport) {
1476 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1477 err = AVERROR_INVALIDDATA;
1481 switch(reply->transports[0].lower_transport) {
1482 case RTSP_LOWER_TRANSPORT_TCP:
1483 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1484 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1487 case RTSP_LOWER_TRANSPORT_UDP: {
1488 char url[1024], options[30] = "";
1489 const char *peer = host;
1491 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1492 av_strlcpy(options, "?connect=1", sizeof(options));
1493 /* Use source address if specified */
1494 if (reply->transports[0].source[0])
1495 peer = reply->transports[0].source;
1496 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1497 reply->transports[0].server_port_min, "%s", options);
1498 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1499 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1500 err = AVERROR_INVALIDDATA;
1503 /* Try to initialize the connection state in a
1504 * potential NAT router by sending dummy packets.
1505 * RTP/RTCP dummy packets are used for RDT, too.
1507 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1509 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1512 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1513 char url[1024], namebuf[50], optbuf[20] = "";
1514 struct sockaddr_storage addr;
1517 if (reply->transports[0].destination.ss_family) {
1518 addr = reply->transports[0].destination;
1519 port = reply->transports[0].port_min;
1520 ttl = reply->transports[0].ttl;
1522 addr = rtsp_st->sdp_ip;
1523 port = rtsp_st->sdp_port;
1524 ttl = rtsp_st->sdp_ttl;
1527 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1528 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1529 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1530 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1531 port, "%s", optbuf);
1532 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1533 &s->interrupt_callback, NULL) < 0) {
1534 err = AVERROR_INVALIDDATA;
1541 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1545 if (rt->nb_rtsp_streams && reply->timeout > 0)
1546 rt->timeout = reply->timeout;
1548 if (rt->server_type == RTSP_SERVER_REAL)
1549 rt->need_subscription = 1;
1554 ff_rtsp_undo_setup(s, 0);
1558 void ff_rtsp_close_connections(AVFormatContext *s)
1560 RTSPState *rt = s->priv_data;
1561 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1562 ffurl_close(rt->rtsp_hd);
1563 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1566 int ff_rtsp_connect(AVFormatContext *s)
1568 RTSPState *rt = s->priv_data;
1569 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1570 int port, err, tcp_fd;
1571 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1572 int lower_transport_mask = 0;
1573 char real_challenge[64] = "";
1574 struct sockaddr_storage peer;
1575 socklen_t peer_len = sizeof(peer);
1577 if (rt->rtp_port_max < rt->rtp_port_min) {
1578 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1579 "than min port %d\n", rt->rtp_port_max,
1581 return AVERROR(EINVAL);
1584 if (!ff_network_init())
1585 return AVERROR(EIO);
1587 if (s->max_delay < 0) /* Not set by the caller */
1588 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1590 rt->control_transport = RTSP_MODE_PLAIN;
1591 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1592 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1593 rt->control_transport = RTSP_MODE_TUNNEL;
1595 /* Only pass through valid flags from here */
1596 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1599 lower_transport_mask = rt->lower_transport_mask;
1600 /* extract hostname and port */
1601 av_url_split(NULL, 0, auth, sizeof(auth),
1602 host, sizeof(host), &port, path, sizeof(path), s->filename);
1604 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1607 port = RTSP_DEFAULT_PORT;
1609 if (!lower_transport_mask)
1610 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1613 /* Only UDP or TCP - UDP multicast isn't supported. */
1614 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1615 (1 << RTSP_LOWER_TRANSPORT_TCP);
1616 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1617 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1618 "only UDP and TCP are supported for output.\n");
1619 err = AVERROR(EINVAL);
1624 /* Construct the URI used in request; this is similar to s->filename,
1625 * but with authentication credentials removed and RTSP specific options
1627 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1628 host, port, "%s", path);
1630 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1631 /* set up initial handshake for tunneling */
1632 char httpname[1024];
1633 char sessioncookie[17];
1636 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1637 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1638 av_get_random_seed(), av_get_random_seed());
1641 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1642 &s->interrupt_callback) < 0) {
1647 /* generate GET headers */
1648 snprintf(headers, sizeof(headers),
1649 "x-sessioncookie: %s\r\n"
1650 "Accept: application/x-rtsp-tunnelled\r\n"
1651 "Pragma: no-cache\r\n"
1652 "Cache-Control: no-cache\r\n",
1654 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1656 /* complete the connection */
1657 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1663 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1664 &s->interrupt_callback) < 0 ) {
1669 /* generate POST headers */
1670 snprintf(headers, sizeof(headers),
1671 "x-sessioncookie: %s\r\n"
1672 "Content-Type: application/x-rtsp-tunnelled\r\n"
1673 "Pragma: no-cache\r\n"
1674 "Cache-Control: no-cache\r\n"
1675 "Content-Length: 32767\r\n"
1676 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1678 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1679 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1681 /* Initialize the authentication state for the POST session. The HTTP
1682 * protocol implementation doesn't properly handle multi-pass
1683 * authentication for POST requests, since it would require one of
1685 * - implementing Expect: 100-continue, which many HTTP servers
1686 * don't support anyway, even less the RTSP servers that do HTTP
1688 * - sending the whole POST data until getting a 401 reply specifying
1689 * what authentication method to use, then resending all that data
1690 * - waiting for potential 401 replies directly after sending the
1691 * POST header (waiting for some unspecified time)
1692 * Therefore, we copy the full auth state, which works for both basic
1693 * and digest. (For digest, we would have to synchronize the nonce
1694 * count variable between the two sessions, if we'd do more requests
1695 * with the original session, though.)
1697 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1699 /* complete the connection */
1700 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1705 /* open the tcp connection */
1706 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1707 "?timeout=%d", rt->stimeout);
1708 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1709 &s->interrupt_callback, NULL) < 0) {
1713 rt->rtsp_hd_out = rt->rtsp_hd;
1717 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1718 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1719 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1720 NULL, 0, NI_NUMERICHOST);
1723 /* request options supported by the server; this also detects server
1725 for (rt->server_type = RTSP_SERVER_RTP;;) {
1727 if (rt->server_type == RTSP_SERVER_REAL)
1730 * The following entries are required for proper
1731 * streaming from a Realmedia server. They are
1732 * interdependent in some way although we currently
1733 * don't quite understand how. Values were copied
1734 * from mplayer SVN r23589.
1735 * ClientChallenge is a 16-byte ID in hex
1736 * CompanyID is a 16-byte ID in base64
1738 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1739 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1740 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1741 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1743 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1744 if (reply->status_code != RTSP_STATUS_OK) {
1745 err = AVERROR_INVALIDDATA;
1749 /* detect server type if not standard-compliant RTP */
1750 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1751 rt->server_type = RTSP_SERVER_REAL;
1753 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1754 rt->server_type = RTSP_SERVER_WMS;
1755 } else if (rt->server_type == RTSP_SERVER_REAL)
1756 strcpy(real_challenge, reply->real_challenge);
1760 if (s->iformat && CONFIG_RTSP_DEMUXER)
1761 err = ff_rtsp_setup_input_streams(s, reply);
1762 else if (CONFIG_RTSP_MUXER)
1763 err = ff_rtsp_setup_output_streams(s, host);
1768 int lower_transport = ff_log2_tab[lower_transport_mask &
1769 ~(lower_transport_mask - 1)];
1771 if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
1772 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
1773 lower_transport = RTSP_LOWER_TRANSPORT_TCP;
1775 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1776 rt->server_type == RTSP_SERVER_REAL ?
1777 real_challenge : NULL);
1780 lower_transport_mask &= ~(1 << lower_transport);
1781 if (lower_transport_mask == 0 && err == 1) {
1782 err = AVERROR(EPROTONOSUPPORT);
1787 rt->lower_transport_mask = lower_transport_mask;
1788 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1789 rt->state = RTSP_STATE_IDLE;
1790 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1793 ff_rtsp_close_streams(s);
1794 ff_rtsp_close_connections(s);
1795 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1796 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1797 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1805 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1808 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1809 uint8_t *buf, int buf_size, int64_t wait_end)
1811 RTSPState *rt = s->priv_data;
1812 RTSPStream *rtsp_st;
1813 int n, i, ret, tcp_fd, timeout_cnt = 0;
1815 struct pollfd *p = rt->p;
1816 int *fds = NULL, fdsnum, fdsidx;
1819 if (ff_check_interrupt(&s->interrupt_callback))
1820 return AVERROR_EXIT;
1821 if (wait_end && wait_end - av_gettime() < 0)
1822 return AVERROR(EAGAIN);
1825 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1826 p[max_p].fd = tcp_fd;
1827 p[max_p++].events = POLLIN;
1831 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1832 rtsp_st = rt->rtsp_streams[i];
1833 if (rtsp_st->rtp_handle) {
1834 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1836 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1840 av_log(s, AV_LOG_ERROR,
1841 "Number of fds %d not supported\n", fdsnum);
1842 return AVERROR_INVALIDDATA;
1844 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1845 p[max_p].fd = fds[fdsidx];
1846 p[max_p++].events = POLLIN;
1851 n = poll(p, max_p, POLL_TIMEOUT_MS);
1853 int j = 1 - (tcp_fd == -1);
1855 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1856 rtsp_st = rt->rtsp_streams[i];
1857 if (rtsp_st->rtp_handle) {
1858 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1859 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1861 *prtsp_st = rtsp_st;
1868 #if CONFIG_RTSP_DEMUXER
1869 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1870 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1871 if (rt->state == RTSP_STATE_STREAMING) {
1872 if (!ff_rtsp_parse_streaming_commands(s))
1875 av_log(s, AV_LOG_WARNING,
1876 "Unable to answer to TEARDOWN\n");
1880 RTSPMessageHeader reply;
1881 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1884 /* XXX: parse message */
1885 if (rt->state != RTSP_STATE_STREAMING)
1890 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1891 return AVERROR(ETIMEDOUT);
1892 } else if (n < 0 && errno != EINTR)
1893 return AVERROR(errno);
1897 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1898 const uint8_t *buf, int len)
1900 RTSPState *rt = s->priv_data;
1904 if (rt->nb_rtsp_streams == 1) {
1905 *rtsp_st = rt->rtsp_streams[0];
1908 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1909 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1911 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1912 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1915 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1916 *rtsp_st = rt->rtsp_streams[i];
1923 av_log(s, AV_LOG_WARNING,
1924 "Unable to pick stream for packet - SSRC not known for "
1926 return AVERROR(EAGAIN);
1929 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1930 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1931 *rtsp_st = rt->rtsp_streams[i];
1937 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1938 return AVERROR(EAGAIN);
1941 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1943 RTSPState *rt = s->priv_data;
1945 RTSPStream *rtsp_st, *first_queue_st = NULL;
1946 int64_t wait_end = 0;
1948 if (rt->nb_byes == rt->nb_rtsp_streams)
1951 /* get next frames from the same RTP packet */
1952 if (rt->cur_transport_priv) {
1953 if (rt->transport == RTSP_TRANSPORT_RDT) {
1954 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1955 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1956 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1957 } else if (rt->ts && CONFIG_RTPDEC) {
1958 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1960 rt->recvbuf_pos += ret;
1961 ret = rt->recvbuf_pos < rt->recvbuf_len;
1966 rt->cur_transport_priv = NULL;
1968 } else if (ret == 1) {
1971 rt->cur_transport_priv = NULL;
1975 if (rt->transport == RTSP_TRANSPORT_RTP) {
1977 int64_t first_queue_time = 0;
1978 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1979 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1983 queue_time = ff_rtp_queued_packet_time(rtpctx);
1984 if (queue_time && (queue_time - first_queue_time < 0 ||
1985 !first_queue_time)) {
1986 first_queue_time = queue_time;
1987 first_queue_st = rt->rtsp_streams[i];
1990 if (first_queue_time) {
1991 wait_end = first_queue_time + s->max_delay;
1994 first_queue_st = NULL;
1998 /* read next RTP packet */
2000 rt->recvbuf = av_malloc(RECVBUF_SIZE);
2002 return AVERROR(ENOMEM);
2005 switch(rt->lower_transport) {
2007 #if CONFIG_RTSP_DEMUXER
2008 case RTSP_LOWER_TRANSPORT_TCP:
2009 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2012 case RTSP_LOWER_TRANSPORT_UDP:
2013 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2014 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2015 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2016 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2018 case RTSP_LOWER_TRANSPORT_CUSTOM:
2019 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2020 wait_end && wait_end < av_gettime())
2021 len = AVERROR(EAGAIN);
2023 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2024 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2025 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2026 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2029 if (len == AVERROR(EAGAIN) && first_queue_st &&
2030 rt->transport == RTSP_TRANSPORT_RTP) {
2031 rtsp_st = first_queue_st;
2032 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2039 if (rt->transport == RTSP_TRANSPORT_RDT) {
2040 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2041 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2042 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2043 if (rtsp_st->feedback) {
2044 AVIOContext *pb = NULL;
2045 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2047 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2050 /* Either bad packet, or a RTCP packet. Check if the
2051 * first_rtcp_ntp_time field was initialized. */
2052 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2053 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2054 /* first_rtcp_ntp_time has been initialized for this stream,
2055 * copy the same value to all other uninitialized streams,
2056 * in order to map their timestamp origin to the same ntp time
2059 AVStream *st = NULL;
2060 if (rtsp_st->stream_index >= 0)
2061 st = s->streams[rtsp_st->stream_index];
2062 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2063 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2064 AVStream *st2 = NULL;
2065 if (rt->rtsp_streams[i]->stream_index >= 0)
2066 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2067 if (rtpctx2 && st && st2 &&
2068 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2069 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2070 rtpctx2->rtcp_ts_offset = av_rescale_q(
2071 rtpctx->rtcp_ts_offset, st->time_base,
2075 // Make real NTP start time available in AVFormatContext
2076 if (s->start_time_realtime == AV_NOPTS_VALUE) {
2077 s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
2079 s->start_time_realtime -=
2080 av_rescale (rtpctx->rtcp_ts_offset,
2081 (uint64_t) rtpctx->st->time_base.num * 1000000,
2082 rtpctx->st->time_base.den);
2086 if (ret == -RTCP_BYE) {
2089 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2090 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2092 if (rt->nb_byes == rt->nb_rtsp_streams)
2096 } else if (rt->ts && CONFIG_RTPDEC) {
2097 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2100 rt->recvbuf_len = len;
2101 rt->recvbuf_pos = ret;
2102 rt->cur_transport_priv = rt->ts;
2109 return AVERROR_INVALIDDATA;
2115 /* more packets may follow, so we save the RTP context */
2116 rt->cur_transport_priv = rtsp_st->transport_priv;
2120 #endif /* CONFIG_RTPDEC */
2122 #if CONFIG_SDP_DEMUXER
2123 static int sdp_probe(AVProbeData *p1)
2125 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2127 /* we look for a line beginning "c=IN IP" */
2128 while (p < p_end && *p != '\0') {
2129 if (p + sizeof("c=IN IP") - 1 < p_end &&
2130 av_strstart(p, "c=IN IP", NULL))
2131 return AVPROBE_SCORE_EXTENSION;
2133 while (p < p_end - 1 && *p != '\n') p++;
2142 static void append_source_addrs(char *buf, int size, const char *name,
2143 int count, struct RTSPSource **addrs)
2148 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2149 for (i = 1; i < count; i++)
2150 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2153 static int sdp_read_header(AVFormatContext *s)
2155 RTSPState *rt = s->priv_data;
2156 RTSPStream *rtsp_st;
2161 if (!ff_network_init())
2162 return AVERROR(EIO);
2164 if (s->max_delay < 0) /* Not set by the caller */
2165 s->max_delay = DEFAULT_REORDERING_DELAY;
2166 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2167 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2169 /* read the whole sdp file */
2170 /* XXX: better loading */
2171 content = av_malloc(SDP_MAX_SIZE);
2172 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2175 return AVERROR_INVALIDDATA;
2177 content[size] ='\0';
2179 err = ff_sdp_parse(s, content);
2183 /* open each RTP stream */
2184 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2186 rtsp_st = rt->rtsp_streams[i];
2188 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2189 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2190 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2191 ff_url_join(url, sizeof(url), "rtp", NULL,
2192 namebuf, rtsp_st->sdp_port,
2193 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2194 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2195 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2196 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2198 append_source_addrs(url, sizeof(url), "sources",
2199 rtsp_st->nb_include_source_addrs,
2200 rtsp_st->include_source_addrs);
2201 append_source_addrs(url, sizeof(url), "block",
2202 rtsp_st->nb_exclude_source_addrs,
2203 rtsp_st->exclude_source_addrs);
2204 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2205 &s->interrupt_callback, NULL) < 0) {
2206 err = AVERROR_INVALIDDATA;
2210 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2215 ff_rtsp_close_streams(s);
2220 static int sdp_read_close(AVFormatContext *s)
2222 ff_rtsp_close_streams(s);
2227 static const AVClass sdp_demuxer_class = {
2228 .class_name = "SDP demuxer",
2229 .item_name = av_default_item_name,
2230 .option = sdp_options,
2231 .version = LIBAVUTIL_VERSION_INT,
2234 AVInputFormat ff_sdp_demuxer = {
2236 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2237 .priv_data_size = sizeof(RTSPState),
2238 .read_probe = sdp_probe,
2239 .read_header = sdp_read_header,
2240 .read_packet = ff_rtsp_fetch_packet,
2241 .read_close = sdp_read_close,
2242 .priv_class = &sdp_demuxer_class,
2244 #endif /* CONFIG_SDP_DEMUXER */
2246 #if CONFIG_RTP_DEMUXER
2247 static int rtp_probe(AVProbeData *p)
2249 if (av_strstart(p->filename, "rtp:", NULL))
2250 return AVPROBE_SCORE_MAX;
2254 static int rtp_read_header(AVFormatContext *s)
2256 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2257 char host[500], sdp[500];
2259 URLContext* in = NULL;
2261 AVCodecContext codec = { 0 };
2262 struct sockaddr_storage addr;
2264 socklen_t addrlen = sizeof(addr);
2265 RTSPState *rt = s->priv_data;
2267 if (!ff_network_init())
2268 return AVERROR(EIO);
2270 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2271 &s->interrupt_callback, NULL);
2276 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2277 if (ret == AVERROR(EAGAIN))
2282 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2286 if ((recvbuf[0] & 0xc0) != 0x80) {
2287 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2292 if (RTP_PT_IS_RTCP(recvbuf[1]))
2295 payload_type = recvbuf[1] & 0x7f;
2298 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2302 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2303 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2304 "without an SDP file describing it\n",
2308 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2309 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2310 "properly you need an SDP file "
2314 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2315 NULL, 0, s->filename);
2317 snprintf(sdp, sizeof(sdp),
2318 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2319 addr.ss_family == AF_INET ? 4 : 6, host,
2320 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2321 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2322 port, payload_type);
2323 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2325 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2328 /* sdp_read_header initializes this again */
2331 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2333 ret = sdp_read_header(s);
2344 static const AVClass rtp_demuxer_class = {
2345 .class_name = "RTP demuxer",
2346 .item_name = av_default_item_name,
2347 .option = rtp_options,
2348 .version = LIBAVUTIL_VERSION_INT,
2351 AVInputFormat ff_rtp_demuxer = {
2353 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2354 .priv_data_size = sizeof(RTSPState),
2355 .read_probe = rtp_probe,
2356 .read_header = rtp_read_header,
2357 .read_packet = ff_rtsp_fetch_packet,
2358 .read_close = sdp_read_close,
2359 .flags = AVFMT_NOFILE,
2360 .priv_class = &rtp_demuxer_class,
2362 #endif /* CONFIG_RTP_DEMUXER */