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 RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
90 { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
91 { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
92 { "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 },
93 { "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 },
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 I/O", 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", "set media types to accept from the server"),
104 RTSP_REORDERING_OPTS(),
108 static const AVOption rtp_options[] = {
109 RTSP_FLAG_OPTS("rtp_flags", "set 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, int send_packets)
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 if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
660 ff_rtsp_tcp_write_packet(s, rtsp_st);
661 avio_close_dyn_buf(rtpctx->pb, &ptr);
664 avio_close(rtpctx->pb);
666 avformat_free_context(rtpctx);
667 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
668 ff_rdt_parse_close(rtsp_st->transport_priv);
669 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
670 ff_rtp_parse_close(rtsp_st->transport_priv);
672 rtsp_st->transport_priv = NULL;
673 if (rtsp_st->rtp_handle)
674 ffurl_close(rtsp_st->rtp_handle);
675 rtsp_st->rtp_handle = NULL;
679 /* close and free RTSP streams */
680 void ff_rtsp_close_streams(AVFormatContext *s)
682 RTSPState *rt = s->priv_data;
686 ff_rtsp_undo_setup(s, 0);
687 for (i = 0; i < rt->nb_rtsp_streams; i++) {
688 rtsp_st = rt->rtsp_streams[i];
690 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
691 rtsp_st->dynamic_handler->free(
692 rtsp_st->dynamic_protocol_context);
693 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
694 av_free(rtsp_st->include_source_addrs[j]);
695 av_freep(&rtsp_st->include_source_addrs);
696 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
697 av_free(rtsp_st->exclude_source_addrs[j]);
698 av_freep(&rtsp_st->exclude_source_addrs);
703 av_free(rt->rtsp_streams);
705 avformat_close_input(&rt->asf_ctx);
707 if (rt->ts && CONFIG_RTPDEC)
708 ff_mpegts_parse_close(rt->ts);
710 av_free(rt->recvbuf);
713 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
715 RTSPState *rt = s->priv_data;
717 int reordering_queue_size = rt->reordering_queue_size;
718 if (reordering_queue_size < 0) {
719 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
720 reordering_queue_size = 0;
722 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
725 /* open the RTP context */
726 if (rtsp_st->stream_index >= 0)
727 st = s->streams[rtsp_st->stream_index];
729 s->ctx_flags |= AVFMTCTX_NOHEADER;
731 if (s->oformat && CONFIG_RTSP_MUXER) {
732 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
733 s, st, rtsp_st->rtp_handle,
734 RTSP_TCP_MAX_PACKET_SIZE,
735 rtsp_st->stream_index);
736 /* Ownership of rtp_handle is passed to the rtp mux context */
737 rtsp_st->rtp_handle = NULL;
740 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
741 return 0; // Don't need to open any parser here
742 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
743 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
744 rtsp_st->dynamic_protocol_context,
745 rtsp_st->dynamic_handler);
746 else if (CONFIG_RTPDEC)
747 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
748 rtsp_st->sdp_payload_type,
749 reordering_queue_size);
751 if (!rtsp_st->transport_priv) {
752 return AVERROR(ENOMEM);
753 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
754 if (rtsp_st->dynamic_handler) {
755 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
756 rtsp_st->dynamic_protocol_context,
757 rtsp_st->dynamic_handler);
759 if (rtsp_st->crypto_suite[0])
760 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
761 rtsp_st->crypto_suite,
762 rtsp_st->crypto_params);
768 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
769 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
776 q += strspn(q, SPACE_CHARS);
777 v = strtol(q, &p, 10);
781 v = strtol(p, &p, 10);
790 /* XXX: only one transport specification is parsed */
791 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
793 char transport_protocol[16];
795 char lower_transport[16];
797 RTSPTransportField *th;
800 reply->nb_transports = 0;
803 p += strspn(p, SPACE_CHARS);
807 th = &reply->transports[reply->nb_transports];
809 get_word_sep(transport_protocol, sizeof(transport_protocol),
811 if (!av_strcasecmp (transport_protocol, "rtp")) {
812 get_word_sep(profile, sizeof(profile), "/;,", &p);
813 lower_transport[0] = '\0';
814 /* rtp/avp/<protocol> */
816 get_word_sep(lower_transport, sizeof(lower_transport),
819 th->transport = RTSP_TRANSPORT_RTP;
820 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
821 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
822 /* x-pn-tng/<protocol> */
823 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
825 th->transport = RTSP_TRANSPORT_RDT;
826 } else if (!av_strcasecmp(transport_protocol, "raw")) {
827 get_word_sep(profile, sizeof(profile), "/;,", &p);
828 lower_transport[0] = '\0';
829 /* raw/raw/<protocol> */
831 get_word_sep(lower_transport, sizeof(lower_transport),
834 th->transport = RTSP_TRANSPORT_RAW;
836 if (!av_strcasecmp(lower_transport, "TCP"))
837 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
839 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
843 /* get each parameter */
844 while (*p != '\0' && *p != ',') {
845 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
846 if (!strcmp(parameter, "port")) {
849 rtsp_parse_range(&th->port_min, &th->port_max, &p);
851 } else if (!strcmp(parameter, "client_port")) {
854 rtsp_parse_range(&th->client_port_min,
855 &th->client_port_max, &p);
857 } else if (!strcmp(parameter, "server_port")) {
860 rtsp_parse_range(&th->server_port_min,
861 &th->server_port_max, &p);
863 } else if (!strcmp(parameter, "interleaved")) {
866 rtsp_parse_range(&th->interleaved_min,
867 &th->interleaved_max, &p);
869 } else if (!strcmp(parameter, "multicast")) {
870 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
871 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
872 } else if (!strcmp(parameter, "ttl")) {
876 th->ttl = strtol(p, &end, 10);
879 } else if (!strcmp(parameter, "destination")) {
882 get_word_sep(buf, sizeof(buf), ";,", &p);
883 get_sockaddr(buf, &th->destination);
885 } else if (!strcmp(parameter, "source")) {
888 get_word_sep(buf, sizeof(buf), ";,", &p);
889 av_strlcpy(th->source, buf, sizeof(th->source));
891 } else if (!strcmp(parameter, "mode")) {
894 get_word_sep(buf, sizeof(buf), ";, ", &p);
895 if (!strcmp(buf, "record") ||
896 !strcmp(buf, "receive"))
901 while (*p != ';' && *p != '\0' && *p != ',')
909 reply->nb_transports++;
913 static void handle_rtp_info(RTSPState *rt, const char *url,
914 uint32_t seq, uint32_t rtptime)
917 if (!rtptime || !url[0])
919 if (rt->transport != RTSP_TRANSPORT_RTP)
921 for (i = 0; i < rt->nb_rtsp_streams; i++) {
922 RTSPStream *rtsp_st = rt->rtsp_streams[i];
923 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
926 if (!strcmp(rtsp_st->control_url, url)) {
927 rtpctx->base_timestamp = rtptime;
933 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
936 char key[20], value[1024], url[1024] = "";
937 uint32_t seq = 0, rtptime = 0;
940 p += strspn(p, SPACE_CHARS);
943 get_word_sep(key, sizeof(key), "=", &p);
947 get_word_sep(value, sizeof(value), ";, ", &p);
949 if (!strcmp(key, "url"))
950 av_strlcpy(url, value, sizeof(url));
951 else if (!strcmp(key, "seq"))
952 seq = strtoul(value, NULL, 10);
953 else if (!strcmp(key, "rtptime"))
954 rtptime = strtoul(value, NULL, 10);
956 handle_rtp_info(rt, url, seq, rtptime);
965 handle_rtp_info(rt, url, seq, rtptime);
968 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
969 RTSPState *rt, const char *method)
973 /* NOTE: we do case independent match for broken servers */
975 if (av_stristart(p, "Session:", &p)) {
977 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
978 if (av_stristart(p, ";timeout=", &p) &&
979 (t = strtol(p, NULL, 10)) > 0) {
982 } else if (av_stristart(p, "Content-Length:", &p)) {
983 reply->content_length = strtol(p, NULL, 10);
984 } else if (av_stristart(p, "Transport:", &p)) {
985 rtsp_parse_transport(reply, p);
986 } else if (av_stristart(p, "CSeq:", &p)) {
987 reply->seq = strtol(p, NULL, 10);
988 } else if (av_stristart(p, "Range:", &p)) {
989 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
990 } else if (av_stristart(p, "RealChallenge1:", &p)) {
991 p += strspn(p, SPACE_CHARS);
992 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
993 } else if (av_stristart(p, "Server:", &p)) {
994 p += strspn(p, SPACE_CHARS);
995 av_strlcpy(reply->server, p, sizeof(reply->server));
996 } else if (av_stristart(p, "Notice:", &p) ||
997 av_stristart(p, "X-Notice:", &p)) {
998 reply->notice = strtol(p, NULL, 10);
999 } else if (av_stristart(p, "Location:", &p)) {
1000 p += strspn(p, SPACE_CHARS);
1001 av_strlcpy(reply->location, p , sizeof(reply->location));
1002 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1003 p += strspn(p, SPACE_CHARS);
1004 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1005 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1006 p += strspn(p, SPACE_CHARS);
1007 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1008 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1009 p += strspn(p, SPACE_CHARS);
1010 if (method && !strcmp(method, "DESCRIBE"))
1011 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1012 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1013 p += strspn(p, SPACE_CHARS);
1014 if (method && !strcmp(method, "PLAY"))
1015 rtsp_parse_rtp_info(rt, p);
1016 } else if (av_stristart(p, "Public:", &p) && rt) {
1017 if (strstr(p, "GET_PARAMETER") &&
1018 method && !strcmp(method, "OPTIONS"))
1019 rt->get_parameter_supported = 1;
1020 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1021 p += strspn(p, SPACE_CHARS);
1022 rt->accept_dynamic_rate = atoi(p);
1023 } else if (av_stristart(p, "Content-Type:", &p)) {
1024 p += strspn(p, SPACE_CHARS);
1025 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1029 /* skip a RTP/TCP interleaved packet */
1030 void ff_rtsp_skip_packet(AVFormatContext *s)
1032 RTSPState *rt = s->priv_data;
1036 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1039 len = AV_RB16(buf + 1);
1041 av_dlog(s, "skipping RTP packet len=%d\n", len);
1046 if (len1 > sizeof(buf))
1048 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1055 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1056 unsigned char **content_ptr,
1057 int return_on_interleaved_data, const char *method)
1059 RTSPState *rt = s->priv_data;
1060 char buf[4096], buf1[1024], *q;
1063 int ret, content_length, line_count = 0, request = 0;
1064 unsigned char *content = NULL;
1070 memset(reply, 0, sizeof(*reply));
1072 /* parse reply (XXX: use buffers) */
1073 rt->last_reply[0] = '\0';
1077 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1078 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1084 /* XXX: only parse it if first char on line ? */
1085 if (return_on_interleaved_data) {
1088 ff_rtsp_skip_packet(s);
1089 } else if (ch != '\r') {
1090 if ((q - buf) < sizeof(buf) - 1)
1096 av_dlog(s, "line='%s'\n", buf);
1098 /* test if last line */
1102 if (line_count == 0) {
1103 /* get reply code */
1104 get_word(buf1, sizeof(buf1), &p);
1105 if (!strncmp(buf1, "RTSP/", 5)) {
1106 get_word(buf1, sizeof(buf1), &p);
1107 reply->status_code = atoi(buf1);
1108 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1110 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1111 get_word(buf1, sizeof(buf1), &p); // object
1115 ff_rtsp_parse_line(reply, p, rt, method);
1116 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1117 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1122 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1123 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1125 content_length = reply->content_length;
1126 if (content_length > 0) {
1127 /* leave some room for a trailing '\0' (useful for simple parsing) */
1128 content = av_malloc(content_length + 1);
1129 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1130 content[content_length] = '\0';
1133 *content_ptr = content;
1139 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1140 const char* ptr = buf;
1142 if (!strcmp(reply->reason, "OPTIONS")) {
1143 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1145 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1146 if (reply->session_id[0])
1147 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1150 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1152 av_strlcat(buf, "\r\n", sizeof(buf));
1154 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1155 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1158 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1160 rt->last_cmd_time = av_gettime();
1161 /* Even if the request from the server had data, it is not the data
1162 * that the caller wants or expects. The memory could also be leaked
1163 * if the actual following reply has content data. */
1165 av_freep(content_ptr);
1166 /* If method is set, this is called from ff_rtsp_send_cmd,
1167 * where a reply to exactly this request is awaited. For
1168 * callers from within packet receiving, we just want to
1169 * return to the caller and go back to receiving packets. */
1175 if (rt->seq != reply->seq) {
1176 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1177 rt->seq, reply->seq);
1181 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1182 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1183 reply->notice == 2306 /* Continuous Feed Terminated */) {
1184 rt->state = RTSP_STATE_IDLE;
1185 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1186 return AVERROR(EIO); /* data or server error */
1187 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1188 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1189 return AVERROR(EPERM);
1195 * Send a command to the RTSP server without waiting for the reply.
1197 * @param s RTSP (de)muxer context
1198 * @param method the method for the request
1199 * @param url the target url for the request
1200 * @param headers extra header lines to include in the request
1201 * @param send_content if non-null, the data to send as request body content
1202 * @param send_content_length the length of the send_content data, or 0 if
1203 * send_content is null
1205 * @return zero if success, nonzero otherwise
1207 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1208 const char *method, const char *url,
1209 const char *headers,
1210 const unsigned char *send_content,
1211 int send_content_length)
1213 RTSPState *rt = s->priv_data;
1214 char buf[4096], *out_buf;
1215 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1217 /* Add in RTSP headers */
1220 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1222 av_strlcat(buf, headers, sizeof(buf));
1223 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1224 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1225 if (rt->session_id[0] != '\0' && (!headers ||
1226 !strstr(headers, "\nIf-Match:"))) {
1227 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1230 char *str = ff_http_auth_create_response(&rt->auth_state,
1231 rt->auth, url, method);
1233 av_strlcat(buf, str, sizeof(buf));
1236 if (send_content_length > 0 && send_content)
1237 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1238 av_strlcat(buf, "\r\n", sizeof(buf));
1240 /* base64 encode rtsp if tunneling */
1241 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1242 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1243 out_buf = base64buf;
1246 av_dlog(s, "Sending:\n%s--\n", buf);
1248 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1249 if (send_content_length > 0 && send_content) {
1250 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1251 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1252 "with content data not supported\n");
1253 return AVERROR_PATCHWELCOME;
1255 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1257 rt->last_cmd_time = av_gettime();
1262 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1263 const char *url, const char *headers)
1265 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1268 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1269 const char *headers, RTSPMessageHeader *reply,
1270 unsigned char **content_ptr)
1272 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1273 content_ptr, NULL, 0);
1276 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1277 const char *method, const char *url,
1279 RTSPMessageHeader *reply,
1280 unsigned char **content_ptr,
1281 const unsigned char *send_content,
1282 int send_content_length)
1284 RTSPState *rt = s->priv_data;
1285 HTTPAuthType cur_auth_type;
1286 int ret, attempts = 0;
1289 cur_auth_type = rt->auth_state.auth_type;
1290 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1292 send_content_length)))
1295 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1299 if (reply->status_code == 401 &&
1300 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1301 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1304 if (reply->status_code > 400){
1305 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1309 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1315 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1316 int lower_transport, const char *real_challenge)
1318 RTSPState *rt = s->priv_data;
1319 int rtx = 0, j, i, err, interleave = 0, port_off;
1320 RTSPStream *rtsp_st;
1321 RTSPMessageHeader reply1, *reply = &reply1;
1323 const char *trans_pref;
1325 if (rt->transport == RTSP_TRANSPORT_RDT)
1326 trans_pref = "x-pn-tng";
1327 else if (rt->transport == RTSP_TRANSPORT_RAW)
1328 trans_pref = "RAW/RAW";
1330 trans_pref = "RTP/AVP";
1332 /* default timeout: 1 minute */
1335 /* Choose a random starting offset within the first half of the
1336 * port range, to allow for a number of ports to try even if the offset
1337 * happens to be at the end of the random range. */
1338 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1339 /* even random offset */
1340 port_off -= port_off & 0x01;
1342 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1343 char transport[2048];
1346 * WMS serves all UDP data over a single connection, the RTX, which
1347 * isn't necessarily the first in the SDP but has to be the first
1348 * to be set up, else the second/third SETUP will fail with a 461.
1350 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1351 rt->server_type == RTSP_SERVER_WMS) {
1354 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1355 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1357 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1361 if (rtx == rt->nb_rtsp_streams)
1362 return -1; /* no RTX found */
1363 rtsp_st = rt->rtsp_streams[rtx];
1365 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1367 rtsp_st = rt->rtsp_streams[i];
1370 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1373 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1374 port = reply->transports[0].client_port_min;
1378 /* first try in specified port range */
1379 while (j <= rt->rtp_port_max) {
1380 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1381 "?localport=%d", j);
1382 /* we will use two ports per rtp stream (rtp and rtcp) */
1384 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1385 &s->interrupt_callback, NULL))
1388 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1393 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1395 snprintf(transport, sizeof(transport) - 1,
1396 "%s/UDP;", trans_pref);
1397 if (rt->server_type != RTSP_SERVER_REAL)
1398 av_strlcat(transport, "unicast;", sizeof(transport));
1399 av_strlcatf(transport, sizeof(transport),
1400 "client_port=%d", port);
1401 if (rt->transport == RTSP_TRANSPORT_RTP &&
1402 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1403 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1407 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1408 /* For WMS streams, the application streams are only used for
1409 * UDP. When trying to set it up for TCP streams, the server
1410 * will return an error. Therefore, we skip those streams. */
1411 if (rt->server_type == RTSP_SERVER_WMS &&
1412 (rtsp_st->stream_index < 0 ||
1413 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1416 snprintf(transport, sizeof(transport) - 1,
1417 "%s/TCP;", trans_pref);
1418 if (rt->transport != RTSP_TRANSPORT_RDT)
1419 av_strlcat(transport, "unicast;", sizeof(transport));
1420 av_strlcatf(transport, sizeof(transport),
1421 "interleaved=%d-%d",
1422 interleave, interleave + 1);
1426 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1427 snprintf(transport, sizeof(transport) - 1,
1428 "%s/UDP;multicast", trans_pref);
1431 av_strlcat(transport, ";mode=record", sizeof(transport));
1432 } else if (rt->server_type == RTSP_SERVER_REAL ||
1433 rt->server_type == RTSP_SERVER_WMS)
1434 av_strlcat(transport, ";mode=play", sizeof(transport));
1435 snprintf(cmd, sizeof(cmd),
1436 "Transport: %s\r\n",
1438 if (rt->accept_dynamic_rate)
1439 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1440 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1441 char real_res[41], real_csum[9];
1442 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1444 av_strlcatf(cmd, sizeof(cmd),
1446 "RealChallenge2: %s, sd=%s\r\n",
1447 rt->session_id, real_res, real_csum);
1449 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1450 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1453 } else if (reply->status_code != RTSP_STATUS_OK ||
1454 reply->nb_transports != 1) {
1455 err = AVERROR_INVALIDDATA;
1459 /* XXX: same protocol for all streams is required */
1461 if (reply->transports[0].lower_transport != rt->lower_transport ||
1462 reply->transports[0].transport != rt->transport) {
1463 err = AVERROR_INVALIDDATA;
1467 rt->lower_transport = reply->transports[0].lower_transport;
1468 rt->transport = reply->transports[0].transport;
1471 /* Fail if the server responded with another lower transport mode
1472 * than what we requested. */
1473 if (reply->transports[0].lower_transport != lower_transport) {
1474 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1475 err = AVERROR_INVALIDDATA;
1479 switch(reply->transports[0].lower_transport) {
1480 case RTSP_LOWER_TRANSPORT_TCP:
1481 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1482 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1485 case RTSP_LOWER_TRANSPORT_UDP: {
1486 char url[1024], options[30] = "";
1487 const char *peer = host;
1489 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1490 av_strlcpy(options, "?connect=1", sizeof(options));
1491 /* Use source address if specified */
1492 if (reply->transports[0].source[0])
1493 peer = reply->transports[0].source;
1494 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1495 reply->transports[0].server_port_min, "%s", options);
1496 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1497 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1498 err = AVERROR_INVALIDDATA;
1501 /* Try to initialize the connection state in a
1502 * potential NAT router by sending dummy packets.
1503 * RTP/RTCP dummy packets are used for RDT, too.
1505 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1507 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1510 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1511 char url[1024], namebuf[50], optbuf[20] = "";
1512 struct sockaddr_storage addr;
1515 if (reply->transports[0].destination.ss_family) {
1516 addr = reply->transports[0].destination;
1517 port = reply->transports[0].port_min;
1518 ttl = reply->transports[0].ttl;
1520 addr = rtsp_st->sdp_ip;
1521 port = rtsp_st->sdp_port;
1522 ttl = rtsp_st->sdp_ttl;
1525 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1526 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1527 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1528 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1529 port, "%s", optbuf);
1530 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1531 &s->interrupt_callback, NULL) < 0) {
1532 err = AVERROR_INVALIDDATA;
1539 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1543 if (rt->nb_rtsp_streams && reply->timeout > 0)
1544 rt->timeout = reply->timeout;
1546 if (rt->server_type == RTSP_SERVER_REAL)
1547 rt->need_subscription = 1;
1552 ff_rtsp_undo_setup(s, 0);
1556 void ff_rtsp_close_connections(AVFormatContext *s)
1558 RTSPState *rt = s->priv_data;
1559 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1560 ffurl_close(rt->rtsp_hd);
1561 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1564 int ff_rtsp_connect(AVFormatContext *s)
1566 RTSPState *rt = s->priv_data;
1567 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1568 int port, err, tcp_fd;
1569 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1570 int lower_transport_mask = 0;
1571 char real_challenge[64] = "";
1572 struct sockaddr_storage peer;
1573 socklen_t peer_len = sizeof(peer);
1575 if (rt->rtp_port_max < rt->rtp_port_min) {
1576 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1577 "than min port %d\n", rt->rtp_port_max,
1579 return AVERROR(EINVAL);
1582 if (!ff_network_init())
1583 return AVERROR(EIO);
1585 if (s->max_delay < 0) /* Not set by the caller */
1586 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1588 rt->control_transport = RTSP_MODE_PLAIN;
1589 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1590 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1591 rt->control_transport = RTSP_MODE_TUNNEL;
1593 /* Only pass through valid flags from here */
1594 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1597 lower_transport_mask = rt->lower_transport_mask;
1598 /* extract hostname and port */
1599 av_url_split(NULL, 0, auth, sizeof(auth),
1600 host, sizeof(host), &port, path, sizeof(path), s->filename);
1602 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1605 port = RTSP_DEFAULT_PORT;
1607 if (!lower_transport_mask)
1608 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1611 /* Only UDP or TCP - UDP multicast isn't supported. */
1612 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1613 (1 << RTSP_LOWER_TRANSPORT_TCP);
1614 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1615 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1616 "only UDP and TCP are supported for output.\n");
1617 err = AVERROR(EINVAL);
1622 /* Construct the URI used in request; this is similar to s->filename,
1623 * but with authentication credentials removed and RTSP specific options
1625 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1626 host, port, "%s", path);
1628 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1629 /* set up initial handshake for tunneling */
1630 char httpname[1024];
1631 char sessioncookie[17];
1634 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1635 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1636 av_get_random_seed(), av_get_random_seed());
1639 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1640 &s->interrupt_callback) < 0) {
1645 /* generate GET headers */
1646 snprintf(headers, sizeof(headers),
1647 "x-sessioncookie: %s\r\n"
1648 "Accept: application/x-rtsp-tunnelled\r\n"
1649 "Pragma: no-cache\r\n"
1650 "Cache-Control: no-cache\r\n",
1652 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1654 /* complete the connection */
1655 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1661 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1662 &s->interrupt_callback) < 0 ) {
1667 /* generate POST headers */
1668 snprintf(headers, sizeof(headers),
1669 "x-sessioncookie: %s\r\n"
1670 "Content-Type: application/x-rtsp-tunnelled\r\n"
1671 "Pragma: no-cache\r\n"
1672 "Cache-Control: no-cache\r\n"
1673 "Content-Length: 32767\r\n"
1674 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1676 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1677 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1679 /* Initialize the authentication state for the POST session. The HTTP
1680 * protocol implementation doesn't properly handle multi-pass
1681 * authentication for POST requests, since it would require one of
1683 * - implementing Expect: 100-continue, which many HTTP servers
1684 * don't support anyway, even less the RTSP servers that do HTTP
1686 * - sending the whole POST data until getting a 401 reply specifying
1687 * what authentication method to use, then resending all that data
1688 * - waiting for potential 401 replies directly after sending the
1689 * POST header (waiting for some unspecified time)
1690 * Therefore, we copy the full auth state, which works for both basic
1691 * and digest. (For digest, we would have to synchronize the nonce
1692 * count variable between the two sessions, if we'd do more requests
1693 * with the original session, though.)
1695 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1697 /* complete the connection */
1698 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1703 /* open the tcp connection */
1704 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1705 "?timeout=%d", rt->stimeout);
1706 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1707 &s->interrupt_callback, NULL) < 0) {
1711 rt->rtsp_hd_out = rt->rtsp_hd;
1715 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1716 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1717 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1718 NULL, 0, NI_NUMERICHOST);
1721 /* request options supported by the server; this also detects server
1723 for (rt->server_type = RTSP_SERVER_RTP;;) {
1725 if (rt->server_type == RTSP_SERVER_REAL)
1728 * The following entries are required for proper
1729 * streaming from a Realmedia server. They are
1730 * interdependent in some way although we currently
1731 * don't quite understand how. Values were copied
1732 * from mplayer SVN r23589.
1733 * ClientChallenge is a 16-byte ID in hex
1734 * CompanyID is a 16-byte ID in base64
1736 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1737 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1738 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1739 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1741 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1742 if (reply->status_code != RTSP_STATUS_OK) {
1743 err = AVERROR_INVALIDDATA;
1747 /* detect server type if not standard-compliant RTP */
1748 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1749 rt->server_type = RTSP_SERVER_REAL;
1751 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1752 rt->server_type = RTSP_SERVER_WMS;
1753 } else if (rt->server_type == RTSP_SERVER_REAL)
1754 strcpy(real_challenge, reply->real_challenge);
1758 if (s->iformat && CONFIG_RTSP_DEMUXER)
1759 err = ff_rtsp_setup_input_streams(s, reply);
1760 else if (CONFIG_RTSP_MUXER)
1761 err = ff_rtsp_setup_output_streams(s, host);
1766 int lower_transport = ff_log2_tab[lower_transport_mask &
1767 ~(lower_transport_mask - 1)];
1769 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1770 rt->server_type == RTSP_SERVER_REAL ?
1771 real_challenge : NULL);
1774 lower_transport_mask &= ~(1 << lower_transport);
1775 if (lower_transport_mask == 0 && err == 1) {
1776 err = AVERROR(EPROTONOSUPPORT);
1781 rt->lower_transport_mask = lower_transport_mask;
1782 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1783 rt->state = RTSP_STATE_IDLE;
1784 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1787 ff_rtsp_close_streams(s);
1788 ff_rtsp_close_connections(s);
1789 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1790 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1791 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1799 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1802 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1803 uint8_t *buf, int buf_size, int64_t wait_end)
1805 RTSPState *rt = s->priv_data;
1806 RTSPStream *rtsp_st;
1807 int n, i, ret, tcp_fd, timeout_cnt = 0;
1809 struct pollfd *p = rt->p;
1810 int *fds = NULL, fdsnum, fdsidx;
1813 if (ff_check_interrupt(&s->interrupt_callback))
1814 return AVERROR_EXIT;
1815 if (wait_end && wait_end - av_gettime() < 0)
1816 return AVERROR(EAGAIN);
1819 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1820 p[max_p].fd = tcp_fd;
1821 p[max_p++].events = POLLIN;
1825 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1826 rtsp_st = rt->rtsp_streams[i];
1827 if (rtsp_st->rtp_handle) {
1828 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1830 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1834 av_log(s, AV_LOG_ERROR,
1835 "Number of fds %d not supported\n", fdsnum);
1836 return AVERROR_INVALIDDATA;
1838 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1839 p[max_p].fd = fds[fdsidx];
1840 p[max_p++].events = POLLIN;
1845 n = poll(p, max_p, POLL_TIMEOUT_MS);
1847 int j = 1 - (tcp_fd == -1);
1849 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1850 rtsp_st = rt->rtsp_streams[i];
1851 if (rtsp_st->rtp_handle) {
1852 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1853 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1855 *prtsp_st = rtsp_st;
1862 #if CONFIG_RTSP_DEMUXER
1863 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1864 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1865 if (rt->state == RTSP_STATE_STREAMING) {
1866 if (!ff_rtsp_parse_streaming_commands(s))
1869 av_log(s, AV_LOG_WARNING,
1870 "Unable to answer to TEARDOWN\n");
1874 RTSPMessageHeader reply;
1875 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1878 /* XXX: parse message */
1879 if (rt->state != RTSP_STATE_STREAMING)
1884 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1885 return AVERROR(ETIMEDOUT);
1886 } else if (n < 0 && errno != EINTR)
1887 return AVERROR(errno);
1891 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1892 const uint8_t *buf, int len)
1894 RTSPState *rt = s->priv_data;
1898 if (rt->nb_rtsp_streams == 1) {
1899 *rtsp_st = rt->rtsp_streams[0];
1902 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1903 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1905 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1906 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1909 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1910 *rtsp_st = rt->rtsp_streams[i];
1917 av_log(s, AV_LOG_WARNING,
1918 "Unable to pick stream for packet - SSRC not known for "
1920 return AVERROR(EAGAIN);
1923 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1924 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1925 *rtsp_st = rt->rtsp_streams[i];
1931 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1932 return AVERROR(EAGAIN);
1935 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1937 RTSPState *rt = s->priv_data;
1939 RTSPStream *rtsp_st, *first_queue_st = NULL;
1940 int64_t wait_end = 0;
1942 if (rt->nb_byes == rt->nb_rtsp_streams)
1945 /* get next frames from the same RTP packet */
1946 if (rt->cur_transport_priv) {
1947 if (rt->transport == RTSP_TRANSPORT_RDT) {
1948 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1949 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1950 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1951 } else if (rt->ts && CONFIG_RTPDEC) {
1952 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1954 rt->recvbuf_pos += ret;
1955 ret = rt->recvbuf_pos < rt->recvbuf_len;
1960 rt->cur_transport_priv = NULL;
1962 } else if (ret == 1) {
1965 rt->cur_transport_priv = NULL;
1969 if (rt->transport == RTSP_TRANSPORT_RTP) {
1971 int64_t first_queue_time = 0;
1972 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1977 queue_time = ff_rtp_queued_packet_time(rtpctx);
1978 if (queue_time && (queue_time - first_queue_time < 0 ||
1979 !first_queue_time)) {
1980 first_queue_time = queue_time;
1981 first_queue_st = rt->rtsp_streams[i];
1984 if (first_queue_time) {
1985 wait_end = first_queue_time + s->max_delay;
1988 first_queue_st = NULL;
1992 /* read next RTP packet */
1994 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1996 return AVERROR(ENOMEM);
1999 switch(rt->lower_transport) {
2001 #if CONFIG_RTSP_DEMUXER
2002 case RTSP_LOWER_TRANSPORT_TCP:
2003 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2006 case RTSP_LOWER_TRANSPORT_UDP:
2007 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2008 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2009 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2010 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2012 case RTSP_LOWER_TRANSPORT_CUSTOM:
2013 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2014 wait_end && wait_end < av_gettime())
2015 len = AVERROR(EAGAIN);
2017 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2018 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2019 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2020 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2023 if (len == AVERROR(EAGAIN) && first_queue_st &&
2024 rt->transport == RTSP_TRANSPORT_RTP) {
2025 rtsp_st = first_queue_st;
2026 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2033 if (rt->transport == RTSP_TRANSPORT_RDT) {
2034 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2035 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2036 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2037 if (rtsp_st->feedback) {
2038 AVIOContext *pb = NULL;
2039 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2041 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2044 /* Either bad packet, or a RTCP packet. Check if the
2045 * first_rtcp_ntp_time field was initialized. */
2046 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2047 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2048 /* first_rtcp_ntp_time has been initialized for this stream,
2049 * copy the same value to all other uninitialized streams,
2050 * in order to map their timestamp origin to the same ntp time
2053 AVStream *st = NULL;
2054 if (rtsp_st->stream_index >= 0)
2055 st = s->streams[rtsp_st->stream_index];
2056 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2057 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2058 AVStream *st2 = NULL;
2059 if (rt->rtsp_streams[i]->stream_index >= 0)
2060 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2061 if (rtpctx2 && st && st2 &&
2062 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2063 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2064 rtpctx2->rtcp_ts_offset = av_rescale_q(
2065 rtpctx->rtcp_ts_offset, st->time_base,
2070 if (ret == -RTCP_BYE) {
2073 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2074 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2076 if (rt->nb_byes == rt->nb_rtsp_streams)
2080 } else if (rt->ts && CONFIG_RTPDEC) {
2081 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2084 rt->recvbuf_len = len;
2085 rt->recvbuf_pos = ret;
2086 rt->cur_transport_priv = rt->ts;
2093 return AVERROR_INVALIDDATA;
2099 /* more packets may follow, so we save the RTP context */
2100 rt->cur_transport_priv = rtsp_st->transport_priv;
2104 #endif /* CONFIG_RTPDEC */
2106 #if CONFIG_SDP_DEMUXER
2107 static int sdp_probe(AVProbeData *p1)
2109 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2111 /* we look for a line beginning "c=IN IP" */
2112 while (p < p_end && *p != '\0') {
2113 if (p + sizeof("c=IN IP") - 1 < p_end &&
2114 av_strstart(p, "c=IN IP", NULL))
2115 return AVPROBE_SCORE_EXTENSION;
2117 while (p < p_end - 1 && *p != '\n') p++;
2126 static void append_source_addrs(char *buf, int size, const char *name,
2127 int count, struct RTSPSource **addrs)
2132 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2133 for (i = 1; i < count; i++)
2134 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2137 static int sdp_read_header(AVFormatContext *s)
2139 RTSPState *rt = s->priv_data;
2140 RTSPStream *rtsp_st;
2145 if (!ff_network_init())
2146 return AVERROR(EIO);
2148 if (s->max_delay < 0) /* Not set by the caller */
2149 s->max_delay = DEFAULT_REORDERING_DELAY;
2150 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2151 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2153 /* read the whole sdp file */
2154 /* XXX: better loading */
2155 content = av_malloc(SDP_MAX_SIZE);
2156 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2159 return AVERROR_INVALIDDATA;
2161 content[size] ='\0';
2163 err = ff_sdp_parse(s, content);
2167 /* open each RTP stream */
2168 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2170 rtsp_st = rt->rtsp_streams[i];
2172 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2173 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2174 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2175 ff_url_join(url, sizeof(url), "rtp", NULL,
2176 namebuf, rtsp_st->sdp_port,
2177 "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
2178 rtsp_st->sdp_port, rtsp_st->sdp_ttl,
2179 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
2180 rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);
2182 append_source_addrs(url, sizeof(url), "sources",
2183 rtsp_st->nb_include_source_addrs,
2184 rtsp_st->include_source_addrs);
2185 append_source_addrs(url, sizeof(url), "block",
2186 rtsp_st->nb_exclude_source_addrs,
2187 rtsp_st->exclude_source_addrs);
2188 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2189 &s->interrupt_callback, NULL) < 0) {
2190 err = AVERROR_INVALIDDATA;
2194 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2199 ff_rtsp_close_streams(s);
2204 static int sdp_read_close(AVFormatContext *s)
2206 ff_rtsp_close_streams(s);
2211 static const AVClass sdp_demuxer_class = {
2212 .class_name = "SDP demuxer",
2213 .item_name = av_default_item_name,
2214 .option = sdp_options,
2215 .version = LIBAVUTIL_VERSION_INT,
2218 AVInputFormat ff_sdp_demuxer = {
2220 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2221 .priv_data_size = sizeof(RTSPState),
2222 .read_probe = sdp_probe,
2223 .read_header = sdp_read_header,
2224 .read_packet = ff_rtsp_fetch_packet,
2225 .read_close = sdp_read_close,
2226 .priv_class = &sdp_demuxer_class,
2228 #endif /* CONFIG_SDP_DEMUXER */
2230 #if CONFIG_RTP_DEMUXER
2231 static int rtp_probe(AVProbeData *p)
2233 if (av_strstart(p->filename, "rtp:", NULL))
2234 return AVPROBE_SCORE_MAX;
2238 static int rtp_read_header(AVFormatContext *s)
2240 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2241 char host[500], sdp[500];
2243 URLContext* in = NULL;
2245 AVCodecContext codec = { 0 };
2246 struct sockaddr_storage addr;
2248 socklen_t addrlen = sizeof(addr);
2249 RTSPState *rt = s->priv_data;
2251 if (!ff_network_init())
2252 return AVERROR(EIO);
2254 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2255 &s->interrupt_callback, NULL);
2260 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2261 if (ret == AVERROR(EAGAIN))
2266 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2270 if ((recvbuf[0] & 0xc0) != 0x80) {
2271 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2276 if (RTP_PT_IS_RTCP(recvbuf[1]))
2279 payload_type = recvbuf[1] & 0x7f;
2282 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2286 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2287 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2288 "without an SDP file describing it\n",
2292 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2293 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2294 "properly you need an SDP file "
2298 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2299 NULL, 0, s->filename);
2301 snprintf(sdp, sizeof(sdp),
2302 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2303 addr.ss_family == AF_INET ? 4 : 6, host,
2304 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2305 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2306 port, payload_type);
2307 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2309 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2312 /* sdp_read_header initializes this again */
2315 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2317 ret = sdp_read_header(s);
2328 static const AVClass rtp_demuxer_class = {
2329 .class_name = "RTP demuxer",
2330 .item_name = av_default_item_name,
2331 .option = rtp_options,
2332 .version = LIBAVUTIL_VERSION_INT,
2335 AVInputFormat ff_rtp_demuxer = {
2337 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2338 .priv_data_size = sizeof(RTSPState),
2339 .read_probe = rtp_probe,
2340 .read_header = rtp_read_header,
2341 .read_packet = ff_rtsp_fetch_packet,
2342 .read_close = sdp_read_close,
2343 .flags = AVFMT_NOFILE,
2344 .priv_class = &rtp_demuxer_class,
2346 #endif /* CONFIG_RTP_DEMUXER */