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", "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", "Don't 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", "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", "RTSP flags"),
88 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
90 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
91 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
92 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
93 { "stimeout", "timeout (in micro seconds) of socket i/o operations.", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
94 RTSP_REORDERING_OPTS(),
95 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },
99 static const AVOption sdp_options[] = {
100 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
101 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
102 { "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" },
103 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
104 RTSP_REORDERING_OPTS(),
108 static const AVOption rtp_options[] = {
109 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
110 RTSP_REORDERING_OPTS(),
114 static void get_word_until_chars(char *buf, int buf_size,
115 const char *sep, const char **pp)
121 p += strspn(p, SPACE_CHARS);
123 while (!strchr(sep, *p) && *p != '\0') {
124 if ((q - buf) < buf_size - 1)
133 static void get_word_sep(char *buf, int buf_size, const char *sep,
136 if (**pp == '/') (*pp)++;
137 get_word_until_chars(buf, buf_size, sep, pp);
140 static void get_word(char *buf, int buf_size, const char **pp)
142 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
145 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
147 * Used for seeking in the rtp stream.
149 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
153 p += strspn(p, SPACE_CHARS);
154 if (!av_stristart(p, "npt=", &p))
157 *start = AV_NOPTS_VALUE;
158 *end = AV_NOPTS_VALUE;
160 get_word_sep(buf, sizeof(buf), "-", &p);
161 av_parse_time(start, buf, 1);
164 get_word_sep(buf, sizeof(buf), "-", &p);
165 av_parse_time(end, buf, 1);
169 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
171 struct addrinfo hints = { 0 }, *ai = NULL;
172 hints.ai_flags = AI_NUMERICHOST;
173 if (getaddrinfo(buf, NULL, &hints, &ai))
175 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
181 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
182 RTSPStream *rtsp_st, AVCodecContext *codec)
187 codec->codec_id = handler->codec_id;
188 rtsp_st->dynamic_handler = handler;
189 if (handler->alloc) {
190 rtsp_st->dynamic_protocol_context = handler->alloc();
191 if (!rtsp_st->dynamic_protocol_context)
192 rtsp_st->dynamic_handler = NULL;
196 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
197 static int sdp_parse_rtpmap(AVFormatContext *s,
198 AVStream *st, RTSPStream *rtsp_st,
199 int payload_type, const char *p)
201 AVCodecContext *codec = st->codec;
207 /* See if we can handle this kind of payload.
208 * The space should normally not be there but some Real streams or
209 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
210 * have a trailing space. */
211 get_word_sep(buf, sizeof(buf), "/ ", &p);
212 if (payload_type < RTP_PT_PRIVATE) {
213 /* We are in a standard case
214 * (from http://www.iana.org/assignments/rtp-parameters). */
215 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
218 if (codec->codec_id == AV_CODEC_ID_NONE) {
219 RTPDynamicProtocolHandler *handler =
220 ff_rtp_handler_find_by_name(buf, codec->codec_type);
221 init_rtp_handler(handler, rtsp_st, codec);
222 /* If no dynamic handler was found, check with the list of standard
223 * allocated types, if such a stream for some reason happens to
224 * use a private payload type. This isn't handled in rtpdec.c, since
225 * the format name from the rtpmap line never is passed into rtpdec. */
226 if (!rtsp_st->dynamic_handler)
227 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
230 c = avcodec_find_decoder(codec->codec_id);
236 get_word_sep(buf, sizeof(buf), "/", &p);
238 switch (codec->codec_type) {
239 case AVMEDIA_TYPE_AUDIO:
240 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
241 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
242 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
244 codec->sample_rate = i;
245 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
246 get_word_sep(buf, sizeof(buf), "/", &p);
251 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
253 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
256 case AVMEDIA_TYPE_VIDEO:
257 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
259 avpriv_set_pts_info(st, 32, 1, i);
264 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
265 rtsp_st->dynamic_handler->init(s, st->index,
266 rtsp_st->dynamic_protocol_context);
270 /* parse the attribute line from the fmtp a line of an sdp response. This
271 * is broken out as a function because it is used in rtp_h264.c, which is
273 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
274 char *value, int value_size)
276 *p += strspn(*p, SPACE_CHARS);
278 get_word_sep(attr, attr_size, "=", p);
281 get_word_sep(value, value_size, ";", p);
289 typedef struct SDPParseState {
291 struct sockaddr_storage default_ip;
293 int skip_media; ///< set if an unknown m= line occurs
294 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
295 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
296 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
297 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
300 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
301 struct RTSPSource ***dest, int *dest_count)
303 RTSPSource *rtsp_src, *rtsp_src2;
305 for (i = 0; i < count; i++) {
307 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
310 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
311 dynarray_add(dest, dest_count, rtsp_src2);
315 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
316 int letter, const char *buf)
318 RTSPState *rt = s->priv_data;
319 char buf1[64], st_type[64];
321 enum AVMediaType codec_type;
325 RTSPSource *rtsp_src;
326 struct sockaddr_storage sdp_ip;
329 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
332 if (s1->skip_media && letter != 'm')
336 get_word(buf1, sizeof(buf1), &p);
337 if (strcmp(buf1, "IN") != 0)
339 get_word(buf1, sizeof(buf1), &p);
340 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
342 get_word_sep(buf1, sizeof(buf1), "/", &p);
343 if (get_sockaddr(buf1, &sdp_ip))
348 get_word_sep(buf1, sizeof(buf1), "/", &p);
351 if (s->nb_streams == 0) {
352 s1->default_ip = sdp_ip;
353 s1->default_ttl = ttl;
355 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
356 rtsp_st->sdp_ip = sdp_ip;
357 rtsp_st->sdp_ttl = ttl;
361 av_dict_set(&s->metadata, "title", p, 0);
364 if (s->nb_streams == 0) {
365 av_dict_set(&s->metadata, "comment", p, 0);
372 codec_type = AVMEDIA_TYPE_UNKNOWN;
373 get_word(st_type, sizeof(st_type), &p);
374 if (!strcmp(st_type, "audio")) {
375 codec_type = AVMEDIA_TYPE_AUDIO;
376 } else if (!strcmp(st_type, "video")) {
377 codec_type = AVMEDIA_TYPE_VIDEO;
378 } else if (!strcmp(st_type, "application")) {
379 codec_type = AVMEDIA_TYPE_DATA;
381 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
385 rtsp_st = av_mallocz(sizeof(RTSPStream));
388 rtsp_st->stream_index = -1;
389 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
391 rtsp_st->sdp_ip = s1->default_ip;
392 rtsp_st->sdp_ttl = s1->default_ttl;
394 copy_default_source_addrs(s1->default_include_source_addrs,
395 s1->nb_default_include_source_addrs,
396 &rtsp_st->include_source_addrs,
397 &rtsp_st->nb_include_source_addrs);
398 copy_default_source_addrs(s1->default_exclude_source_addrs,
399 s1->nb_default_exclude_source_addrs,
400 &rtsp_st->exclude_source_addrs,
401 &rtsp_st->nb_exclude_source_addrs);
403 get_word(buf1, sizeof(buf1), &p); /* port */
404 rtsp_st->sdp_port = atoi(buf1);
406 get_word(buf1, sizeof(buf1), &p); /* protocol */
407 if (!strcmp(buf1, "udp"))
408 rt->transport = RTSP_TRANSPORT_RAW;
409 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
410 rtsp_st->feedback = 1;
412 /* XXX: handle list of formats */
413 get_word(buf1, sizeof(buf1), &p); /* format list */
414 rtsp_st->sdp_payload_type = atoi(buf1);
416 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
417 /* no corresponding stream */
418 if (rt->transport == RTSP_TRANSPORT_RAW) {
419 if (!rt->ts && CONFIG_RTPDEC)
420 rt->ts = ff_mpegts_parse_open(s);
422 RTPDynamicProtocolHandler *handler;
423 handler = ff_rtp_handler_find_by_id(
424 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
425 init_rtp_handler(handler, rtsp_st, NULL);
426 if (handler && handler->init)
427 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
429 } else if (rt->server_type == RTSP_SERVER_WMS &&
430 codec_type == AVMEDIA_TYPE_DATA) {
431 /* RTX stream, a stream that carries all the other actual
432 * audio/video streams. Don't expose this to the callers. */
434 st = avformat_new_stream(s, NULL);
437 st->id = rt->nb_rtsp_streams - 1;
438 rtsp_st->stream_index = st->index;
439 st->codec->codec_type = codec_type;
440 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
441 RTPDynamicProtocolHandler *handler;
442 /* if standard payload type, we can find the codec right now */
443 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
444 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
445 st->codec->sample_rate > 0)
446 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
447 /* Even static payload types may need a custom depacketizer */
448 handler = ff_rtp_handler_find_by_id(
449 rtsp_st->sdp_payload_type, st->codec->codec_type);
450 init_rtp_handler(handler, rtsp_st, st->codec);
451 if (handler && handler->init)
452 handler->init(s, st->index,
453 rtsp_st->dynamic_protocol_context);
456 /* put a default control url */
457 av_strlcpy(rtsp_st->control_url, rt->control_uri,
458 sizeof(rtsp_st->control_url));
461 if (av_strstart(p, "control:", &p)) {
462 if (s->nb_streams == 0) {
463 if (!strncmp(p, "rtsp://", 7))
464 av_strlcpy(rt->control_uri, p,
465 sizeof(rt->control_uri));
468 /* get the control url */
469 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
471 /* XXX: may need to add full url resolution */
472 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
474 if (proto[0] == '\0') {
475 /* relative control URL */
476 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
477 av_strlcat(rtsp_st->control_url, "/",
478 sizeof(rtsp_st->control_url));
479 av_strlcat(rtsp_st->control_url, p,
480 sizeof(rtsp_st->control_url));
482 av_strlcpy(rtsp_st->control_url, p,
483 sizeof(rtsp_st->control_url));
485 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
486 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
487 get_word(buf1, sizeof(buf1), &p);
488 payload_type = atoi(buf1);
489 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
490 if (rtsp_st->stream_index >= 0) {
491 st = s->streams[rtsp_st->stream_index];
492 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
494 } else if (av_strstart(p, "fmtp:", &p) ||
495 av_strstart(p, "framesize:", &p)) {
496 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
497 // let dynamic protocol handlers have a stab at the line.
498 get_word(buf1, sizeof(buf1), &p);
499 payload_type = atoi(buf1);
500 for (i = 0; i < rt->nb_rtsp_streams; i++) {
501 rtsp_st = rt->rtsp_streams[i];
502 if (rtsp_st->sdp_payload_type == payload_type &&
503 rtsp_st->dynamic_handler &&
504 rtsp_st->dynamic_handler->parse_sdp_a_line)
505 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
506 rtsp_st->dynamic_protocol_context, buf);
508 } else if (av_strstart(p, "range:", &p)) {
511 // this is so that seeking on a streamed file can work.
512 rtsp_parse_range_npt(p, &start, &end);
513 s->start_time = start;
514 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
515 s->duration = (end == AV_NOPTS_VALUE) ?
516 AV_NOPTS_VALUE : end - start;
517 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
519 rt->transport = RTSP_TRANSPORT_RDT;
520 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
522 st = s->streams[s->nb_streams - 1];
523 st->codec->sample_rate = atoi(p);
524 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
526 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
527 get_word(buf1, sizeof(buf1), &p); // ignore tag
528 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
529 p += strspn(p, SPACE_CHARS);
530 if (av_strstart(p, "inline:", &p))
531 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
532 } else if (av_strstart(p, "source-filter:", &p)) {
534 get_word(buf1, sizeof(buf1), &p);
535 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
537 exclude = !strcmp(buf1, "excl");
539 get_word(buf1, sizeof(buf1), &p);
540 if (strcmp(buf1, "IN") != 0)
542 get_word(buf1, sizeof(buf1), &p);
543 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
545 // not checking that the destination address actually matches or is wildcard
546 get_word(buf1, sizeof(buf1), &p);
549 rtsp_src = av_mallocz(sizeof(*rtsp_src));
552 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
554 if (s->nb_streams == 0) {
555 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
557 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
558 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
561 if (s->nb_streams == 0) {
562 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
564 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
565 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
570 if (rt->server_type == RTSP_SERVER_WMS)
571 ff_wms_parse_sdp_a_line(s, p);
572 if (s->nb_streams > 0) {
573 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
575 if (rt->server_type == RTSP_SERVER_REAL)
576 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
578 if (rtsp_st->dynamic_handler &&
579 rtsp_st->dynamic_handler->parse_sdp_a_line)
580 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
581 rtsp_st->stream_index,
582 rtsp_st->dynamic_protocol_context, buf);
589 int ff_sdp_parse(AVFormatContext *s, const char *content)
591 RTSPState *rt = s->priv_data;
594 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
595 * contain long SDP lines containing complete ASF Headers (several
596 * kB) or arrays of MDPR (RM stream descriptor) headers plus
597 * "rulebooks" describing their properties. Therefore, the SDP line
600 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
601 * in rtpdec_xiph.c. */
603 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
607 p += strspn(p, SPACE_CHARS);
615 /* get the content */
617 while (*p != '\n' && *p != '\r' && *p != '\0') {
618 if ((q - buf) < sizeof(buf) - 1)
623 sdp_parse_line(s, s1, letter, buf);
625 while (*p != '\n' && *p != '\0')
631 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
632 av_free(s1->default_include_source_addrs[i]);
633 av_freep(&s1->default_include_source_addrs);
634 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
635 av_free(s1->default_exclude_source_addrs[i]);
636 av_freep(&s1->default_exclude_source_addrs);
638 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
639 if (!rt->p) return AVERROR(ENOMEM);
642 #endif /* CONFIG_RTPDEC */
644 void ff_rtsp_undo_setup(AVFormatContext *s)
646 RTSPState *rt = s->priv_data;
649 for (i = 0; i < rt->nb_rtsp_streams; i++) {
650 RTSPStream *rtsp_st = rt->rtsp_streams[i];
653 if (rtsp_st->transport_priv) {
655 AVFormatContext *rtpctx = rtsp_st->transport_priv;
656 av_write_trailer(rtpctx);
657 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
659 avio_close_dyn_buf(rtpctx->pb, &ptr);
662 avio_close(rtpctx->pb);
664 avformat_free_context(rtpctx);
665 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
666 ff_rdt_parse_close(rtsp_st->transport_priv);
667 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
668 ff_rtp_parse_close(rtsp_st->transport_priv);
670 rtsp_st->transport_priv = NULL;
671 if (rtsp_st->rtp_handle)
672 ffurl_close(rtsp_st->rtp_handle);
673 rtsp_st->rtp_handle = NULL;
677 /* close and free RTSP streams */
678 void ff_rtsp_close_streams(AVFormatContext *s)
680 RTSPState *rt = s->priv_data;
684 ff_rtsp_undo_setup(s);
685 for (i = 0; i < rt->nb_rtsp_streams; i++) {
686 rtsp_st = rt->rtsp_streams[i];
688 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
689 rtsp_st->dynamic_handler->free(
690 rtsp_st->dynamic_protocol_context);
691 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
692 av_free(rtsp_st->include_source_addrs[j]);
693 av_freep(&rtsp_st->include_source_addrs);
694 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
695 av_free(rtsp_st->exclude_source_addrs[j]);
696 av_freep(&rtsp_st->exclude_source_addrs);
701 av_free(rt->rtsp_streams);
703 avformat_close_input(&rt->asf_ctx);
705 if (rt->ts && CONFIG_RTPDEC)
706 ff_mpegts_parse_close(rt->ts);
708 av_free(rt->recvbuf);
711 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
713 RTSPState *rt = s->priv_data;
715 int reordering_queue_size = rt->reordering_queue_size;
716 if (reordering_queue_size < 0) {
717 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
718 reordering_queue_size = 0;
720 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
723 /* open the RTP context */
724 if (rtsp_st->stream_index >= 0)
725 st = s->streams[rtsp_st->stream_index];
727 s->ctx_flags |= AVFMTCTX_NOHEADER;
729 if (s->oformat && CONFIG_RTSP_MUXER) {
730 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
732 RTSP_TCP_MAX_PACKET_SIZE,
733 rtsp_st->stream_index);
734 /* Ownership of rtp_handle is passed to the rtp mux context */
735 rtsp_st->rtp_handle = NULL;
738 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
739 return 0; // Don't need to open any parser here
740 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
741 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
742 rtsp_st->dynamic_protocol_context,
743 rtsp_st->dynamic_handler);
744 else if (CONFIG_RTPDEC)
745 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
746 rtsp_st->sdp_payload_type,
747 reordering_queue_size);
749 if (!rtsp_st->transport_priv) {
750 return AVERROR(ENOMEM);
751 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
752 if (rtsp_st->dynamic_handler) {
753 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
754 rtsp_st->dynamic_protocol_context,
755 rtsp_st->dynamic_handler);
757 if (rtsp_st->crypto_suite[0])
758 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
759 rtsp_st->crypto_suite,
760 rtsp_st->crypto_params);
766 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
767 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
774 q += strspn(q, SPACE_CHARS);
775 v = strtol(q, &p, 10);
779 v = strtol(p, &p, 10);
788 /* XXX: only one transport specification is parsed */
789 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
791 char transport_protocol[16];
793 char lower_transport[16];
795 RTSPTransportField *th;
798 reply->nb_transports = 0;
801 p += strspn(p, SPACE_CHARS);
805 th = &reply->transports[reply->nb_transports];
807 get_word_sep(transport_protocol, sizeof(transport_protocol),
809 if (!av_strcasecmp (transport_protocol, "rtp")) {
810 get_word_sep(profile, sizeof(profile), "/;,", &p);
811 lower_transport[0] = '\0';
812 /* rtp/avp/<protocol> */
814 get_word_sep(lower_transport, sizeof(lower_transport),
817 th->transport = RTSP_TRANSPORT_RTP;
818 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
819 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
820 /* x-pn-tng/<protocol> */
821 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
823 th->transport = RTSP_TRANSPORT_RDT;
824 } else if (!av_strcasecmp(transport_protocol, "raw")) {
825 get_word_sep(profile, sizeof(profile), "/;,", &p);
826 lower_transport[0] = '\0';
827 /* raw/raw/<protocol> */
829 get_word_sep(lower_transport, sizeof(lower_transport),
832 th->transport = RTSP_TRANSPORT_RAW;
834 if (!av_strcasecmp(lower_transport, "TCP"))
835 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
837 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
841 /* get each parameter */
842 while (*p != '\0' && *p != ',') {
843 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
844 if (!strcmp(parameter, "port")) {
847 rtsp_parse_range(&th->port_min, &th->port_max, &p);
849 } else if (!strcmp(parameter, "client_port")) {
852 rtsp_parse_range(&th->client_port_min,
853 &th->client_port_max, &p);
855 } else if (!strcmp(parameter, "server_port")) {
858 rtsp_parse_range(&th->server_port_min,
859 &th->server_port_max, &p);
861 } else if (!strcmp(parameter, "interleaved")) {
864 rtsp_parse_range(&th->interleaved_min,
865 &th->interleaved_max, &p);
867 } else if (!strcmp(parameter, "multicast")) {
868 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
869 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
870 } else if (!strcmp(parameter, "ttl")) {
874 th->ttl = strtol(p, &end, 10);
877 } else if (!strcmp(parameter, "destination")) {
880 get_word_sep(buf, sizeof(buf), ";,", &p);
881 get_sockaddr(buf, &th->destination);
883 } else if (!strcmp(parameter, "source")) {
886 get_word_sep(buf, sizeof(buf), ";,", &p);
887 av_strlcpy(th->source, buf, sizeof(th->source));
889 } else if (!strcmp(parameter, "mode")) {
892 get_word_sep(buf, sizeof(buf), ";, ", &p);
893 if (!strcmp(buf, "record") ||
894 !strcmp(buf, "receive"))
899 while (*p != ';' && *p != '\0' && *p != ',')
907 reply->nb_transports++;
911 static void handle_rtp_info(RTSPState *rt, const char *url,
912 uint32_t seq, uint32_t rtptime)
915 if (!rtptime || !url[0])
917 if (rt->transport != RTSP_TRANSPORT_RTP)
919 for (i = 0; i < rt->nb_rtsp_streams; i++) {
920 RTSPStream *rtsp_st = rt->rtsp_streams[i];
921 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
924 if (!strcmp(rtsp_st->control_url, url)) {
925 rtpctx->base_timestamp = rtptime;
931 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
934 char key[20], value[1024], url[1024] = "";
935 uint32_t seq = 0, rtptime = 0;
938 p += strspn(p, SPACE_CHARS);
941 get_word_sep(key, sizeof(key), "=", &p);
945 get_word_sep(value, sizeof(value), ";, ", &p);
947 if (!strcmp(key, "url"))
948 av_strlcpy(url, value, sizeof(url));
949 else if (!strcmp(key, "seq"))
950 seq = strtoul(value, NULL, 10);
951 else if (!strcmp(key, "rtptime"))
952 rtptime = strtoul(value, NULL, 10);
954 handle_rtp_info(rt, url, seq, rtptime);
963 handle_rtp_info(rt, url, seq, rtptime);
966 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
967 RTSPState *rt, const char *method)
971 /* NOTE: we do case independent match for broken servers */
973 if (av_stristart(p, "Session:", &p)) {
975 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
976 if (av_stristart(p, ";timeout=", &p) &&
977 (t = strtol(p, NULL, 10)) > 0) {
980 } else if (av_stristart(p, "Content-Length:", &p)) {
981 reply->content_length = strtol(p, NULL, 10);
982 } else if (av_stristart(p, "Transport:", &p)) {
983 rtsp_parse_transport(reply, p);
984 } else if (av_stristart(p, "CSeq:", &p)) {
985 reply->seq = strtol(p, NULL, 10);
986 } else if (av_stristart(p, "Range:", &p)) {
987 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
988 } else if (av_stristart(p, "RealChallenge1:", &p)) {
989 p += strspn(p, SPACE_CHARS);
990 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
991 } else if (av_stristart(p, "Server:", &p)) {
992 p += strspn(p, SPACE_CHARS);
993 av_strlcpy(reply->server, p, sizeof(reply->server));
994 } else if (av_stristart(p, "Notice:", &p) ||
995 av_stristart(p, "X-Notice:", &p)) {
996 reply->notice = strtol(p, NULL, 10);
997 } else if (av_stristart(p, "Location:", &p)) {
998 p += strspn(p, SPACE_CHARS);
999 av_strlcpy(reply->location, p , sizeof(reply->location));
1000 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1001 p += strspn(p, SPACE_CHARS);
1002 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1003 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1004 p += strspn(p, SPACE_CHARS);
1005 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1006 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1007 p += strspn(p, SPACE_CHARS);
1008 if (method && !strcmp(method, "DESCRIBE"))
1009 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1010 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1011 p += strspn(p, SPACE_CHARS);
1012 if (method && !strcmp(method, "PLAY"))
1013 rtsp_parse_rtp_info(rt, p);
1014 } else if (av_stristart(p, "Public:", &p) && rt) {
1015 if (strstr(p, "GET_PARAMETER") &&
1016 method && !strcmp(method, "OPTIONS"))
1017 rt->get_parameter_supported = 1;
1018 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1019 p += strspn(p, SPACE_CHARS);
1020 rt->accept_dynamic_rate = atoi(p);
1021 } else if (av_stristart(p, "Content-Type:", &p)) {
1022 p += strspn(p, SPACE_CHARS);
1023 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1027 /* skip a RTP/TCP interleaved packet */
1028 void ff_rtsp_skip_packet(AVFormatContext *s)
1030 RTSPState *rt = s->priv_data;
1034 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1037 len = AV_RB16(buf + 1);
1039 av_dlog(s, "skipping RTP packet len=%d\n", len);
1044 if (len1 > sizeof(buf))
1046 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1053 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1054 unsigned char **content_ptr,
1055 int return_on_interleaved_data, const char *method)
1057 RTSPState *rt = s->priv_data;
1058 char buf[4096], buf1[1024], *q;
1061 int ret, content_length, line_count = 0, request = 0;
1062 unsigned char *content = NULL;
1068 memset(reply, 0, sizeof(*reply));
1070 /* parse reply (XXX: use buffers) */
1071 rt->last_reply[0] = '\0';
1075 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1076 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1082 /* XXX: only parse it if first char on line ? */
1083 if (return_on_interleaved_data) {
1086 ff_rtsp_skip_packet(s);
1087 } else if (ch != '\r') {
1088 if ((q - buf) < sizeof(buf) - 1)
1094 av_dlog(s, "line='%s'\n", buf);
1096 /* test if last line */
1100 if (line_count == 0) {
1101 /* get reply code */
1102 get_word(buf1, sizeof(buf1), &p);
1103 if (!strncmp(buf1, "RTSP/", 5)) {
1104 get_word(buf1, sizeof(buf1), &p);
1105 reply->status_code = atoi(buf1);
1106 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1108 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1109 get_word(buf1, sizeof(buf1), &p); // object
1113 ff_rtsp_parse_line(reply, p, rt, method);
1114 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1115 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1120 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1121 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1123 content_length = reply->content_length;
1124 if (content_length > 0) {
1125 /* leave some room for a trailing '\0' (useful for simple parsing) */
1126 content = av_malloc(content_length + 1);
1127 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1128 content[content_length] = '\0';
1131 *content_ptr = content;
1137 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1138 const char* ptr = buf;
1140 if (!strcmp(reply->reason, "OPTIONS")) {
1141 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1143 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1144 if (reply->session_id[0])
1145 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1148 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1150 av_strlcat(buf, "\r\n", sizeof(buf));
1152 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1153 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1156 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1158 rt->last_cmd_time = av_gettime();
1159 /* Even if the request from the server had data, it is not the data
1160 * that the caller wants or expects. The memory could also be leaked
1161 * if the actual following reply has content data. */
1163 av_freep(content_ptr);
1164 /* If method is set, this is called from ff_rtsp_send_cmd,
1165 * where a reply to exactly this request is awaited. For
1166 * callers from within packet receiving, we just want to
1167 * return to the caller and go back to receiving packets. */
1173 if (rt->seq != reply->seq) {
1174 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1175 rt->seq, reply->seq);
1179 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1180 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1181 reply->notice == 2306 /* Continuous Feed Terminated */) {
1182 rt->state = RTSP_STATE_IDLE;
1183 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1184 return AVERROR(EIO); /* data or server error */
1185 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1186 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1187 return AVERROR(EPERM);
1193 * Send a command to the RTSP server without waiting for the reply.
1195 * @param s RTSP (de)muxer context
1196 * @param method the method for the request
1197 * @param url the target url for the request
1198 * @param headers extra header lines to include in the request
1199 * @param send_content if non-null, the data to send as request body content
1200 * @param send_content_length the length of the send_content data, or 0 if
1201 * send_content is null
1203 * @return zero if success, nonzero otherwise
1205 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1206 const char *method, const char *url,
1207 const char *headers,
1208 const unsigned char *send_content,
1209 int send_content_length)
1211 RTSPState *rt = s->priv_data;
1212 char buf[4096], *out_buf;
1213 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1215 /* Add in RTSP headers */
1218 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1220 av_strlcat(buf, headers, sizeof(buf));
1221 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1222 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1223 if (rt->session_id[0] != '\0' && (!headers ||
1224 !strstr(headers, "\nIf-Match:"))) {
1225 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1228 char *str = ff_http_auth_create_response(&rt->auth_state,
1229 rt->auth, url, method);
1231 av_strlcat(buf, str, sizeof(buf));
1234 if (send_content_length > 0 && send_content)
1235 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1236 av_strlcat(buf, "\r\n", sizeof(buf));
1238 /* base64 encode rtsp if tunneling */
1239 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1240 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1241 out_buf = base64buf;
1244 av_dlog(s, "Sending:\n%s--\n", buf);
1246 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1247 if (send_content_length > 0 && send_content) {
1248 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1249 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1250 "with content data not supported\n");
1251 return AVERROR_PATCHWELCOME;
1253 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1255 rt->last_cmd_time = av_gettime();
1260 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1261 const char *url, const char *headers)
1263 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1266 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1267 const char *headers, RTSPMessageHeader *reply,
1268 unsigned char **content_ptr)
1270 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1271 content_ptr, NULL, 0);
1274 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1275 const char *method, const char *url,
1277 RTSPMessageHeader *reply,
1278 unsigned char **content_ptr,
1279 const unsigned char *send_content,
1280 int send_content_length)
1282 RTSPState *rt = s->priv_data;
1283 HTTPAuthType cur_auth_type;
1284 int ret, attempts = 0;
1287 cur_auth_type = rt->auth_state.auth_type;
1288 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1290 send_content_length)))
1293 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1297 if (reply->status_code == 401 &&
1298 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1299 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1302 if (reply->status_code > 400){
1303 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1307 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1313 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1314 int lower_transport, const char *real_challenge)
1316 RTSPState *rt = s->priv_data;
1317 int rtx = 0, j, i, err, interleave = 0, port_off;
1318 RTSPStream *rtsp_st;
1319 RTSPMessageHeader reply1, *reply = &reply1;
1321 const char *trans_pref;
1323 if (rt->transport == RTSP_TRANSPORT_RDT)
1324 trans_pref = "x-pn-tng";
1325 else if (rt->transport == RTSP_TRANSPORT_RAW)
1326 trans_pref = "RAW/RAW";
1328 trans_pref = "RTP/AVP";
1330 /* default timeout: 1 minute */
1333 /* Choose a random starting offset within the first half of the
1334 * port range, to allow for a number of ports to try even if the offset
1335 * happens to be at the end of the random range. */
1336 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1337 /* even random offset */
1338 port_off -= port_off & 0x01;
1340 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1341 char transport[2048];
1344 * WMS serves all UDP data over a single connection, the RTX, which
1345 * isn't necessarily the first in the SDP but has to be the first
1346 * to be set up, else the second/third SETUP will fail with a 461.
1348 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1349 rt->server_type == RTSP_SERVER_WMS) {
1352 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1353 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1355 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1359 if (rtx == rt->nb_rtsp_streams)
1360 return -1; /* no RTX found */
1361 rtsp_st = rt->rtsp_streams[rtx];
1363 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1365 rtsp_st = rt->rtsp_streams[i];
1368 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1371 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1372 port = reply->transports[0].client_port_min;
1376 /* first try in specified port range */
1377 while (j <= rt->rtp_port_max) {
1378 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1379 "?localport=%d", j);
1380 /* we will use two ports per rtp stream (rtp and rtcp) */
1382 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1383 &s->interrupt_callback, NULL))
1386 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1391 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1393 snprintf(transport, sizeof(transport) - 1,
1394 "%s/UDP;", trans_pref);
1395 if (rt->server_type != RTSP_SERVER_REAL)
1396 av_strlcat(transport, "unicast;", sizeof(transport));
1397 av_strlcatf(transport, sizeof(transport),
1398 "client_port=%d", port);
1399 if (rt->transport == RTSP_TRANSPORT_RTP &&
1400 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1401 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1405 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1406 /* For WMS streams, the application streams are only used for
1407 * UDP. When trying to set it up for TCP streams, the server
1408 * will return an error. Therefore, we skip those streams. */
1409 if (rt->server_type == RTSP_SERVER_WMS &&
1410 (rtsp_st->stream_index < 0 ||
1411 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1414 snprintf(transport, sizeof(transport) - 1,
1415 "%s/TCP;", trans_pref);
1416 if (rt->transport != RTSP_TRANSPORT_RDT)
1417 av_strlcat(transport, "unicast;", sizeof(transport));
1418 av_strlcatf(transport, sizeof(transport),
1419 "interleaved=%d-%d",
1420 interleave, interleave + 1);
1424 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1425 snprintf(transport, sizeof(transport) - 1,
1426 "%s/UDP;multicast", trans_pref);
1429 av_strlcat(transport, ";mode=record", sizeof(transport));
1430 } else if (rt->server_type == RTSP_SERVER_REAL ||
1431 rt->server_type == RTSP_SERVER_WMS)
1432 av_strlcat(transport, ";mode=play", sizeof(transport));
1433 snprintf(cmd, sizeof(cmd),
1434 "Transport: %s\r\n",
1436 if (rt->accept_dynamic_rate)
1437 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1438 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1439 char real_res[41], real_csum[9];
1440 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1442 av_strlcatf(cmd, sizeof(cmd),
1444 "RealChallenge2: %s, sd=%s\r\n",
1445 rt->session_id, real_res, real_csum);
1447 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1448 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1451 } else if (reply->status_code != RTSP_STATUS_OK ||
1452 reply->nb_transports != 1) {
1453 err = AVERROR_INVALIDDATA;
1457 /* XXX: same protocol for all streams is required */
1459 if (reply->transports[0].lower_transport != rt->lower_transport ||
1460 reply->transports[0].transport != rt->transport) {
1461 err = AVERROR_INVALIDDATA;
1465 rt->lower_transport = reply->transports[0].lower_transport;
1466 rt->transport = reply->transports[0].transport;
1469 /* Fail if the server responded with another lower transport mode
1470 * than what we requested. */
1471 if (reply->transports[0].lower_transport != lower_transport) {
1472 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1473 err = AVERROR_INVALIDDATA;
1477 switch(reply->transports[0].lower_transport) {
1478 case RTSP_LOWER_TRANSPORT_TCP:
1479 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1480 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1483 case RTSP_LOWER_TRANSPORT_UDP: {
1484 char url[1024], options[30] = "";
1485 const char *peer = host;
1487 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1488 av_strlcpy(options, "?connect=1", sizeof(options));
1489 /* Use source address if specified */
1490 if (reply->transports[0].source[0])
1491 peer = reply->transports[0].source;
1492 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1493 reply->transports[0].server_port_min, "%s", options);
1494 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1495 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1496 err = AVERROR_INVALIDDATA;
1499 /* Try to initialize the connection state in a
1500 * potential NAT router by sending dummy packets.
1501 * RTP/RTCP dummy packets are used for RDT, too.
1503 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1505 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1508 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1509 char url[1024], namebuf[50], optbuf[20] = "";
1510 struct sockaddr_storage addr;
1513 if (reply->transports[0].destination.ss_family) {
1514 addr = reply->transports[0].destination;
1515 port = reply->transports[0].port_min;
1516 ttl = reply->transports[0].ttl;
1518 addr = rtsp_st->sdp_ip;
1519 port = rtsp_st->sdp_port;
1520 ttl = rtsp_st->sdp_ttl;
1523 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1524 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1525 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1526 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1527 port, "%s", optbuf);
1528 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1529 &s->interrupt_callback, NULL) < 0) {
1530 err = AVERROR_INVALIDDATA;
1537 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1541 if (rt->nb_rtsp_streams && reply->timeout > 0)
1542 rt->timeout = reply->timeout;
1544 if (rt->server_type == RTSP_SERVER_REAL)
1545 rt->need_subscription = 1;
1550 ff_rtsp_undo_setup(s);
1554 void ff_rtsp_close_connections(AVFormatContext *s)
1556 RTSPState *rt = s->priv_data;
1557 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1558 ffurl_close(rt->rtsp_hd);
1559 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1562 int ff_rtsp_connect(AVFormatContext *s)
1564 RTSPState *rt = s->priv_data;
1565 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1566 int port, err, tcp_fd;
1567 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1568 int lower_transport_mask = 0;
1569 char real_challenge[64] = "";
1570 struct sockaddr_storage peer;
1571 socklen_t peer_len = sizeof(peer);
1573 if (rt->rtp_port_max < rt->rtp_port_min) {
1574 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1575 "than min port %d\n", rt->rtp_port_max,
1577 return AVERROR(EINVAL);
1580 if (!ff_network_init())
1581 return AVERROR(EIO);
1583 if (s->max_delay < 0) /* Not set by the caller */
1584 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1586 rt->control_transport = RTSP_MODE_PLAIN;
1587 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1588 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1589 rt->control_transport = RTSP_MODE_TUNNEL;
1591 /* Only pass through valid flags from here */
1592 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1595 lower_transport_mask = rt->lower_transport_mask;
1596 /* extract hostname and port */
1597 av_url_split(NULL, 0, auth, sizeof(auth),
1598 host, sizeof(host), &port, path, sizeof(path), s->filename);
1600 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1603 port = RTSP_DEFAULT_PORT;
1605 if (!lower_transport_mask)
1606 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1609 /* Only UDP or TCP - UDP multicast isn't supported. */
1610 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1611 (1 << RTSP_LOWER_TRANSPORT_TCP);
1612 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1613 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1614 "only UDP and TCP are supported for output.\n");
1615 err = AVERROR(EINVAL);
1620 /* Construct the URI used in request; this is similar to s->filename,
1621 * but with authentication credentials removed and RTSP specific options
1623 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1624 host, port, "%s", path);
1626 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1627 /* set up initial handshake for tunneling */
1628 char httpname[1024];
1629 char sessioncookie[17];
1632 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1633 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1634 av_get_random_seed(), av_get_random_seed());
1637 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1638 &s->interrupt_callback) < 0) {
1643 /* generate GET headers */
1644 snprintf(headers, sizeof(headers),
1645 "x-sessioncookie: %s\r\n"
1646 "Accept: application/x-rtsp-tunnelled\r\n"
1647 "Pragma: no-cache\r\n"
1648 "Cache-Control: no-cache\r\n",
1650 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1652 /* complete the connection */
1653 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1659 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1660 &s->interrupt_callback) < 0 ) {
1665 /* generate POST headers */
1666 snprintf(headers, sizeof(headers),
1667 "x-sessioncookie: %s\r\n"
1668 "Content-Type: application/x-rtsp-tunnelled\r\n"
1669 "Pragma: no-cache\r\n"
1670 "Cache-Control: no-cache\r\n"
1671 "Content-Length: 32767\r\n"
1672 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1674 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1675 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1677 /* Initialize the authentication state for the POST session. The HTTP
1678 * protocol implementation doesn't properly handle multi-pass
1679 * authentication for POST requests, since it would require one of
1681 * - implementing Expect: 100-continue, which many HTTP servers
1682 * don't support anyway, even less the RTSP servers that do HTTP
1684 * - sending the whole POST data until getting a 401 reply specifying
1685 * what authentication method to use, then resending all that data
1686 * - waiting for potential 401 replies directly after sending the
1687 * POST header (waiting for some unspecified time)
1688 * Therefore, we copy the full auth state, which works for both basic
1689 * and digest. (For digest, we would have to synchronize the nonce
1690 * count variable between the two sessions, if we'd do more requests
1691 * with the original session, though.)
1693 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1695 /* complete the connection */
1696 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1701 /* open the tcp connection */
1702 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1703 "?timeout=%d", rt->stimeout);
1704 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1705 &s->interrupt_callback, NULL) < 0) {
1709 rt->rtsp_hd_out = rt->rtsp_hd;
1713 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1714 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1715 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1716 NULL, 0, NI_NUMERICHOST);
1719 /* request options supported by the server; this also detects server
1721 for (rt->server_type = RTSP_SERVER_RTP;;) {
1723 if (rt->server_type == RTSP_SERVER_REAL)
1726 * The following entries are required for proper
1727 * streaming from a Realmedia server. They are
1728 * interdependent in some way although we currently
1729 * don't quite understand how. Values were copied
1730 * from mplayer SVN r23589.
1731 * ClientChallenge is a 16-byte ID in hex
1732 * CompanyID is a 16-byte ID in base64
1734 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1735 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1736 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1737 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1739 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1740 if (reply->status_code != RTSP_STATUS_OK) {
1741 err = AVERROR_INVALIDDATA;
1745 /* detect server type if not standard-compliant RTP */
1746 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1747 rt->server_type = RTSP_SERVER_REAL;
1749 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1750 rt->server_type = RTSP_SERVER_WMS;
1751 } else if (rt->server_type == RTSP_SERVER_REAL)
1752 strcpy(real_challenge, reply->real_challenge);
1756 if (s->iformat && CONFIG_RTSP_DEMUXER)
1757 err = ff_rtsp_setup_input_streams(s, reply);
1758 else if (CONFIG_RTSP_MUXER)
1759 err = ff_rtsp_setup_output_streams(s, host);
1764 int lower_transport = ff_log2_tab[lower_transport_mask &
1765 ~(lower_transport_mask - 1)];
1767 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1768 rt->server_type == RTSP_SERVER_REAL ?
1769 real_challenge : NULL);
1772 lower_transport_mask &= ~(1 << lower_transport);
1773 if (lower_transport_mask == 0 && err == 1) {
1774 err = AVERROR(EPROTONOSUPPORT);
1779 rt->lower_transport_mask = lower_transport_mask;
1780 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1781 rt->state = RTSP_STATE_IDLE;
1782 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1785 ff_rtsp_close_streams(s);
1786 ff_rtsp_close_connections(s);
1787 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1788 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1789 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1797 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1800 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1801 uint8_t *buf, int buf_size, int64_t wait_end)
1803 RTSPState *rt = s->priv_data;
1804 RTSPStream *rtsp_st;
1805 int n, i, ret, tcp_fd, timeout_cnt = 0;
1807 struct pollfd *p = rt->p;
1808 int *fds = NULL, fdsnum, fdsidx;
1811 if (ff_check_interrupt(&s->interrupt_callback))
1812 return AVERROR_EXIT;
1813 if (wait_end && wait_end - av_gettime() < 0)
1814 return AVERROR(EAGAIN);
1817 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1818 p[max_p].fd = tcp_fd;
1819 p[max_p++].events = POLLIN;
1823 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1824 rtsp_st = rt->rtsp_streams[i];
1825 if (rtsp_st->rtp_handle) {
1826 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1828 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1832 av_log(s, AV_LOG_ERROR,
1833 "Number of fds %d not supported\n", fdsnum);
1834 return AVERROR_INVALIDDATA;
1836 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1837 p[max_p].fd = fds[fdsidx];
1838 p[max_p++].events = POLLIN;
1843 n = poll(p, max_p, POLL_TIMEOUT_MS);
1845 int j = 1 - (tcp_fd == -1);
1847 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1848 rtsp_st = rt->rtsp_streams[i];
1849 if (rtsp_st->rtp_handle) {
1850 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1851 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1853 *prtsp_st = rtsp_st;
1860 #if CONFIG_RTSP_DEMUXER
1861 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1862 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1863 if (rt->state == RTSP_STATE_STREAMING) {
1864 if (!ff_rtsp_parse_streaming_commands(s))
1867 av_log(s, AV_LOG_WARNING,
1868 "Unable to answer to TEARDOWN\n");
1872 RTSPMessageHeader reply;
1873 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1876 /* XXX: parse message */
1877 if (rt->state != RTSP_STATE_STREAMING)
1882 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1883 return AVERROR(ETIMEDOUT);
1884 } else if (n < 0 && errno != EINTR)
1885 return AVERROR(errno);
1889 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1890 const uint8_t *buf, int len)
1892 RTSPState *rt = s->priv_data;
1896 if (rt->nb_rtsp_streams == 1) {
1897 *rtsp_st = rt->rtsp_streams[0];
1900 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1901 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1903 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1904 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1907 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1908 *rtsp_st = rt->rtsp_streams[i];
1915 av_log(s, AV_LOG_WARNING,
1916 "Unable to pick stream for packet - SSRC not known for "
1918 return AVERROR(EAGAIN);
1921 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1922 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1923 *rtsp_st = rt->rtsp_streams[i];
1929 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1930 return AVERROR(EAGAIN);
1933 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1935 RTSPState *rt = s->priv_data;
1937 RTSPStream *rtsp_st, *first_queue_st = NULL;
1938 int64_t wait_end = 0;
1940 if (rt->nb_byes == rt->nb_rtsp_streams)
1943 /* get next frames from the same RTP packet */
1944 if (rt->cur_transport_priv) {
1945 if (rt->transport == RTSP_TRANSPORT_RDT) {
1946 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1947 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1948 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1949 } else if (rt->ts && CONFIG_RTPDEC) {
1950 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1952 rt->recvbuf_pos += ret;
1953 ret = rt->recvbuf_pos < rt->recvbuf_len;
1958 rt->cur_transport_priv = NULL;
1960 } else if (ret == 1) {
1963 rt->cur_transport_priv = NULL;
1967 if (rt->transport == RTSP_TRANSPORT_RTP) {
1969 int64_t first_queue_time = 0;
1970 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1971 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1975 queue_time = ff_rtp_queued_packet_time(rtpctx);
1976 if (queue_time && (queue_time - first_queue_time < 0 ||
1977 !first_queue_time)) {
1978 first_queue_time = queue_time;
1979 first_queue_st = rt->rtsp_streams[i];
1982 if (first_queue_time) {
1983 wait_end = first_queue_time + s->max_delay;
1986 first_queue_st = NULL;
1990 /* read next RTP packet */
1992 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1994 return AVERROR(ENOMEM);
1997 switch(rt->lower_transport) {
1999 #if CONFIG_RTSP_DEMUXER
2000 case RTSP_LOWER_TRANSPORT_TCP:
2001 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2004 case RTSP_LOWER_TRANSPORT_UDP:
2005 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2006 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2007 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2008 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2010 case RTSP_LOWER_TRANSPORT_CUSTOM:
2011 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2012 wait_end && wait_end < av_gettime())
2013 len = AVERROR(EAGAIN);
2015 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2016 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2017 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2018 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2021 if (len == AVERROR(EAGAIN) && first_queue_st &&
2022 rt->transport == RTSP_TRANSPORT_RTP) {
2023 rtsp_st = first_queue_st;
2024 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2031 if (rt->transport == RTSP_TRANSPORT_RDT) {
2032 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2033 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2034 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2035 if (rtsp_st->feedback) {
2036 AVIOContext *pb = NULL;
2037 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2039 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2042 /* Either bad packet, or a RTCP packet. Check if the
2043 * first_rtcp_ntp_time field was initialized. */
2044 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2045 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2046 /* first_rtcp_ntp_time has been initialized for this stream,
2047 * copy the same value to all other uninitialized streams,
2048 * in order to map their timestamp origin to the same ntp time
2051 AVStream *st = NULL;
2052 if (rtsp_st->stream_index >= 0)
2053 st = s->streams[rtsp_st->stream_index];
2054 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2055 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2056 AVStream *st2 = NULL;
2057 if (rt->rtsp_streams[i]->stream_index >= 0)
2058 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2059 if (rtpctx2 && st && st2 &&
2060 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2061 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2062 rtpctx2->rtcp_ts_offset = av_rescale_q(
2063 rtpctx->rtcp_ts_offset, st->time_base,
2068 if (ret == -RTCP_BYE) {
2071 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2072 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2074 if (rt->nb_byes == rt->nb_rtsp_streams)
2078 } else if (rt->ts && CONFIG_RTPDEC) {
2079 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2082 rt->recvbuf_len = len;
2083 rt->recvbuf_pos = ret;
2084 rt->cur_transport_priv = rt->ts;
2091 return AVERROR_INVALIDDATA;
2097 /* more packets may follow, so we save the RTP context */
2098 rt->cur_transport_priv = rtsp_st->transport_priv;
2102 #endif /* CONFIG_RTPDEC */
2104 #if CONFIG_SDP_DEMUXER
2105 static int sdp_probe(AVProbeData *p1)
2107 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2109 /* we look for a line beginning "c=IN IP" */
2110 while (p < p_end && *p != '\0') {
2111 if (p + sizeof("c=IN IP") - 1 < p_end &&
2112 av_strstart(p, "c=IN IP", NULL))
2113 return AVPROBE_SCORE_EXTENSION;
2115 while (p < p_end - 1 && *p != '\n') p++;
2124 static void append_source_addrs(char *buf, int size, const char *name,
2125 int count, struct RTSPSource **addrs)
2130 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2131 for (i = 1; i < count; i++)
2132 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2135 static int sdp_read_header(AVFormatContext *s)
2137 RTSPState *rt = s->priv_data;
2138 RTSPStream *rtsp_st;
2143 if (!ff_network_init())
2144 return AVERROR(EIO);
2146 if (s->max_delay < 0) /* Not set by the caller */
2147 s->max_delay = DEFAULT_REORDERING_DELAY;
2148 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2149 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2151 /* read the whole sdp file */
2152 /* XXX: better loading */
2153 content = av_malloc(SDP_MAX_SIZE);
2154 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2157 return AVERROR_INVALIDDATA;
2159 content[size] ='\0';
2161 err = ff_sdp_parse(s, content);
2165 /* open each RTP stream */
2166 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2168 rtsp_st = rt->rtsp_streams[i];
2170 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2171 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2172 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2173 ff_url_join(url, sizeof(url), "rtp", NULL,
2174 namebuf, rtsp_st->sdp_port,
2175 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2176 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2177 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2178 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2180 append_source_addrs(url, sizeof(url), "sources",
2181 rtsp_st->nb_include_source_addrs,
2182 rtsp_st->include_source_addrs);
2183 append_source_addrs(url, sizeof(url), "block",
2184 rtsp_st->nb_exclude_source_addrs,
2185 rtsp_st->exclude_source_addrs);
2186 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2187 &s->interrupt_callback, NULL) < 0) {
2188 err = AVERROR_INVALIDDATA;
2192 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2197 ff_rtsp_close_streams(s);
2202 static int sdp_read_close(AVFormatContext *s)
2204 ff_rtsp_close_streams(s);
2209 static const AVClass sdp_demuxer_class = {
2210 .class_name = "SDP demuxer",
2211 .item_name = av_default_item_name,
2212 .option = sdp_options,
2213 .version = LIBAVUTIL_VERSION_INT,
2216 AVInputFormat ff_sdp_demuxer = {
2218 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2219 .priv_data_size = sizeof(RTSPState),
2220 .read_probe = sdp_probe,
2221 .read_header = sdp_read_header,
2222 .read_packet = ff_rtsp_fetch_packet,
2223 .read_close = sdp_read_close,
2224 .priv_class = &sdp_demuxer_class,
2226 #endif /* CONFIG_SDP_DEMUXER */
2228 #if CONFIG_RTP_DEMUXER
2229 static int rtp_probe(AVProbeData *p)
2231 if (av_strstart(p->filename, "rtp:", NULL))
2232 return AVPROBE_SCORE_MAX;
2236 static int rtp_read_header(AVFormatContext *s)
2238 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2239 char host[500], sdp[500];
2241 URLContext* in = NULL;
2243 AVCodecContext codec = { 0 };
2244 struct sockaddr_storage addr;
2246 socklen_t addrlen = sizeof(addr);
2247 RTSPState *rt = s->priv_data;
2249 if (!ff_network_init())
2250 return AVERROR(EIO);
2252 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2253 &s->interrupt_callback, NULL);
2258 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2259 if (ret == AVERROR(EAGAIN))
2264 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2268 if ((recvbuf[0] & 0xc0) != 0x80) {
2269 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2274 if (RTP_PT_IS_RTCP(recvbuf[1]))
2277 payload_type = recvbuf[1] & 0x7f;
2280 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2284 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2285 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2286 "without an SDP file describing it\n",
2290 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2291 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2292 "properly you need an SDP file "
2296 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2297 NULL, 0, s->filename);
2299 snprintf(sdp, sizeof(sdp),
2300 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2301 addr.ss_family == AF_INET ? 4 : 6, host,
2302 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2303 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2304 port, payload_type);
2305 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2307 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2310 /* sdp_read_header initializes this again */
2313 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2315 ret = sdp_read_header(s);
2326 static const AVClass rtp_demuxer_class = {
2327 .class_name = "RTP demuxer",
2328 .item_name = av_default_item_name,
2329 .option = rtp_options,
2330 .version = LIBAVUTIL_VERSION_INT,
2333 AVInputFormat ff_rtp_demuxer = {
2335 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2336 .priv_data_size = sizeof(RTSPState),
2337 .read_probe = rtp_probe,
2338 .read_header = rtp_read_header,
2339 .read_packet = ff_rtsp_fetch_packet,
2340 .read_close = sdp_read_close,
2341 .flags = AVFMT_NOFILE,
2342 .priv_class = &rtp_demuxer_class,
2344 #endif /* CONFIG_RTP_DEMUXER */