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" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { 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" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 #define RTSP_REORDERING_OPTS() \
78 { "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 }
80 const AVOption ff_rtsp_options[] = {
81 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
82 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
83 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
84 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
87 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
88 RTSP_FLAG_OPTS("rtsp_flags", "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 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
103 RTSP_REORDERING_OPTS(),
107 static const AVOption rtp_options[] = {
108 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
109 RTSP_REORDERING_OPTS(),
113 static void get_word_until_chars(char *buf, int buf_size,
114 const char *sep, const char **pp)
120 p += strspn(p, SPACE_CHARS);
122 while (!strchr(sep, *p) && *p != '\0') {
123 if ((q - buf) < buf_size - 1)
132 static void get_word_sep(char *buf, int buf_size, const char *sep,
135 if (**pp == '/') (*pp)++;
136 get_word_until_chars(buf, buf_size, sep, pp);
139 static void get_word(char *buf, int buf_size, const char **pp)
141 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
144 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
146 * Used for seeking in the rtp stream.
148 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
152 p += strspn(p, SPACE_CHARS);
153 if (!av_stristart(p, "npt=", &p))
156 *start = AV_NOPTS_VALUE;
157 *end = AV_NOPTS_VALUE;
159 get_word_sep(buf, sizeof(buf), "-", &p);
160 av_parse_time(start, buf, 1);
163 get_word_sep(buf, sizeof(buf), "-", &p);
164 av_parse_time(end, buf, 1);
168 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
170 struct addrinfo hints = { 0 }, *ai = NULL;
171 hints.ai_flags = AI_NUMERICHOST;
172 if (getaddrinfo(buf, NULL, &hints, &ai))
174 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
180 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
181 RTSPStream *rtsp_st, AVCodecContext *codec)
186 codec->codec_id = handler->codec_id;
187 rtsp_st->dynamic_handler = handler;
188 if (handler->alloc) {
189 rtsp_st->dynamic_protocol_context = handler->alloc();
190 if (!rtsp_st->dynamic_protocol_context)
191 rtsp_st->dynamic_handler = NULL;
195 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
196 static int sdp_parse_rtpmap(AVFormatContext *s,
197 AVStream *st, RTSPStream *rtsp_st,
198 int payload_type, const char *p)
200 AVCodecContext *codec = st->codec;
206 /* See if we can handle this kind of payload.
207 * The space should normally not be there but some Real streams or
208 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
209 * have a trailing space. */
210 get_word_sep(buf, sizeof(buf), "/ ", &p);
211 if (payload_type < RTP_PT_PRIVATE) {
212 /* We are in a standard case
213 * (from http://www.iana.org/assignments/rtp-parameters). */
214 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
217 if (codec->codec_id == AV_CODEC_ID_NONE) {
218 RTPDynamicProtocolHandler *handler =
219 ff_rtp_handler_find_by_name(buf, codec->codec_type);
220 init_rtp_handler(handler, rtsp_st, codec);
221 /* If no dynamic handler was found, check with the list of standard
222 * allocated types, if such a stream for some reason happens to
223 * use a private payload type. This isn't handled in rtpdec.c, since
224 * the format name from the rtpmap line never is passed into rtpdec. */
225 if (!rtsp_st->dynamic_handler)
226 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
229 c = avcodec_find_decoder(codec->codec_id);
235 get_word_sep(buf, sizeof(buf), "/", &p);
237 switch (codec->codec_type) {
238 case AVMEDIA_TYPE_AUDIO:
239 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
241 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
243 codec->sample_rate = i;
244 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
245 get_word_sep(buf, sizeof(buf), "/", &p);
250 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
252 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
255 case AVMEDIA_TYPE_VIDEO:
256 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
258 avpriv_set_pts_info(st, 32, 1, i);
263 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
264 rtsp_st->dynamic_handler->init(s, st->index,
265 rtsp_st->dynamic_protocol_context);
269 /* parse the attribute line from the fmtp a line of an sdp response. This
270 * is broken out as a function because it is used in rtp_h264.c, which is
272 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
273 char *value, int value_size)
275 *p += strspn(*p, SPACE_CHARS);
277 get_word_sep(attr, attr_size, "=", p);
280 get_word_sep(value, value_size, ";", p);
288 typedef struct SDPParseState {
290 struct sockaddr_storage default_ip;
292 int skip_media; ///< set if an unknown m= line occurs
293 int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
294 struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
295 int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
296 struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */
299 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
300 struct RTSPSource ***dest, int *dest_count)
302 RTSPSource *rtsp_src, *rtsp_src2;
304 for (i = 0; i < count; i++) {
306 rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
309 memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
310 dynarray_add(dest, dest_count, rtsp_src2);
314 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
315 int letter, const char *buf)
317 RTSPState *rt = s->priv_data;
318 char buf1[64], st_type[64];
320 enum AVMediaType codec_type;
324 RTSPSource *rtsp_src;
325 struct sockaddr_storage sdp_ip;
328 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
331 if (s1->skip_media && letter != 'm')
335 get_word(buf1, sizeof(buf1), &p);
336 if (strcmp(buf1, "IN") != 0)
338 get_word(buf1, sizeof(buf1), &p);
339 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
341 get_word_sep(buf1, sizeof(buf1), "/", &p);
342 if (get_sockaddr(buf1, &sdp_ip))
347 get_word_sep(buf1, sizeof(buf1), "/", &p);
350 if (s->nb_streams == 0) {
351 s1->default_ip = sdp_ip;
352 s1->default_ttl = ttl;
354 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
355 rtsp_st->sdp_ip = sdp_ip;
356 rtsp_st->sdp_ttl = ttl;
360 av_dict_set(&s->metadata, "title", p, 0);
363 if (s->nb_streams == 0) {
364 av_dict_set(&s->metadata, "comment", p, 0);
371 codec_type = AVMEDIA_TYPE_UNKNOWN;
372 get_word(st_type, sizeof(st_type), &p);
373 if (!strcmp(st_type, "audio")) {
374 codec_type = AVMEDIA_TYPE_AUDIO;
375 } else if (!strcmp(st_type, "video")) {
376 codec_type = AVMEDIA_TYPE_VIDEO;
377 } else if (!strcmp(st_type, "application")) {
378 codec_type = AVMEDIA_TYPE_DATA;
380 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
384 rtsp_st = av_mallocz(sizeof(RTSPStream));
387 rtsp_st->stream_index = -1;
388 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
390 rtsp_st->sdp_ip = s1->default_ip;
391 rtsp_st->sdp_ttl = s1->default_ttl;
393 copy_default_source_addrs(s1->default_include_source_addrs,
394 s1->nb_default_include_source_addrs,
395 &rtsp_st->include_source_addrs,
396 &rtsp_st->nb_include_source_addrs);
397 copy_default_source_addrs(s1->default_exclude_source_addrs,
398 s1->nb_default_exclude_source_addrs,
399 &rtsp_st->exclude_source_addrs,
400 &rtsp_st->nb_exclude_source_addrs);
402 get_word(buf1, sizeof(buf1), &p); /* port */
403 rtsp_st->sdp_port = atoi(buf1);
405 get_word(buf1, sizeof(buf1), &p); /* protocol */
406 if (!strcmp(buf1, "udp"))
407 rt->transport = RTSP_TRANSPORT_RAW;
408 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
409 rtsp_st->feedback = 1;
411 /* XXX: handle list of formats */
412 get_word(buf1, sizeof(buf1), &p); /* format list */
413 rtsp_st->sdp_payload_type = atoi(buf1);
415 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
416 /* no corresponding stream */
417 if (rt->transport == RTSP_TRANSPORT_RAW) {
418 if (!rt->ts && CONFIG_RTPDEC)
419 rt->ts = ff_mpegts_parse_open(s);
421 RTPDynamicProtocolHandler *handler;
422 handler = ff_rtp_handler_find_by_id(
423 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
424 init_rtp_handler(handler, rtsp_st, NULL);
425 if (handler && handler->init)
426 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
428 } else if (rt->server_type == RTSP_SERVER_WMS &&
429 codec_type == AVMEDIA_TYPE_DATA) {
430 /* RTX stream, a stream that carries all the other actual
431 * audio/video streams. Don't expose this to the callers. */
433 st = avformat_new_stream(s, NULL);
436 st->id = rt->nb_rtsp_streams - 1;
437 rtsp_st->stream_index = st->index;
438 st->codec->codec_type = codec_type;
439 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
440 RTPDynamicProtocolHandler *handler;
441 /* if standard payload type, we can find the codec right now */
442 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
443 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
444 st->codec->sample_rate > 0)
445 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
446 /* Even static payload types may need a custom depacketizer */
447 handler = ff_rtp_handler_find_by_id(
448 rtsp_st->sdp_payload_type, st->codec->codec_type);
449 init_rtp_handler(handler, rtsp_st, st->codec);
450 if (handler && handler->init)
451 handler->init(s, st->index,
452 rtsp_st->dynamic_protocol_context);
455 /* put a default control url */
456 av_strlcpy(rtsp_st->control_url, rt->control_uri,
457 sizeof(rtsp_st->control_url));
460 if (av_strstart(p, "control:", &p)) {
461 if (s->nb_streams == 0) {
462 if (!strncmp(p, "rtsp://", 7))
463 av_strlcpy(rt->control_uri, p,
464 sizeof(rt->control_uri));
467 /* get the control url */
468 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
470 /* XXX: may need to add full url resolution */
471 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
473 if (proto[0] == '\0') {
474 /* relative control URL */
475 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
476 av_strlcat(rtsp_st->control_url, "/",
477 sizeof(rtsp_st->control_url));
478 av_strlcat(rtsp_st->control_url, p,
479 sizeof(rtsp_st->control_url));
481 av_strlcpy(rtsp_st->control_url, p,
482 sizeof(rtsp_st->control_url));
484 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
485 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
486 get_word(buf1, sizeof(buf1), &p);
487 payload_type = atoi(buf1);
488 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
489 if (rtsp_st->stream_index >= 0) {
490 st = s->streams[rtsp_st->stream_index];
491 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
493 } else if (av_strstart(p, "fmtp:", &p) ||
494 av_strstart(p, "framesize:", &p)) {
495 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
496 // let dynamic protocol handlers have a stab at the line.
497 get_word(buf1, sizeof(buf1), &p);
498 payload_type = atoi(buf1);
499 for (i = 0; i < rt->nb_rtsp_streams; i++) {
500 rtsp_st = rt->rtsp_streams[i];
501 if (rtsp_st->sdp_payload_type == payload_type &&
502 rtsp_st->dynamic_handler &&
503 rtsp_st->dynamic_handler->parse_sdp_a_line)
504 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
505 rtsp_st->dynamic_protocol_context, buf);
507 } else if (av_strstart(p, "range:", &p)) {
510 // this is so that seeking on a streamed file can work.
511 rtsp_parse_range_npt(p, &start, &end);
512 s->start_time = start;
513 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
514 s->duration = (end == AV_NOPTS_VALUE) ?
515 AV_NOPTS_VALUE : end - start;
516 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
518 rt->transport = RTSP_TRANSPORT_RDT;
519 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
521 st = s->streams[s->nb_streams - 1];
522 st->codec->sample_rate = atoi(p);
523 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
525 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
526 get_word(buf1, sizeof(buf1), &p); // ignore tag
527 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
528 p += strspn(p, SPACE_CHARS);
529 if (av_strstart(p, "inline:", &p))
530 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
531 } else if (av_strstart(p, "source-filter:", &p)) {
533 get_word(buf1, sizeof(buf1), &p);
534 if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))
536 exclude = !strcmp(buf1, "excl");
538 get_word(buf1, sizeof(buf1), &p);
539 if (strcmp(buf1, "IN") != 0)
541 get_word(buf1, sizeof(buf1), &p);
542 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))
544 // not checking that the destination address actually matches or is wildcard
545 get_word(buf1, sizeof(buf1), &p);
548 rtsp_src = av_mallocz(sizeof(*rtsp_src));
551 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
553 if (s->nb_streams == 0) {
554 dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
556 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
557 dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
560 if (s->nb_streams == 0) {
561 dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
563 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
564 dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
569 if (rt->server_type == RTSP_SERVER_WMS)
570 ff_wms_parse_sdp_a_line(s, p);
571 if (s->nb_streams > 0) {
572 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
574 if (rt->server_type == RTSP_SERVER_REAL)
575 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
577 if (rtsp_st->dynamic_handler &&
578 rtsp_st->dynamic_handler->parse_sdp_a_line)
579 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
580 rtsp_st->stream_index,
581 rtsp_st->dynamic_protocol_context, buf);
588 int ff_sdp_parse(AVFormatContext *s, const char *content)
590 RTSPState *rt = s->priv_data;
593 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
594 * contain long SDP lines containing complete ASF Headers (several
595 * kB) or arrays of MDPR (RM stream descriptor) headers plus
596 * "rulebooks" describing their properties. Therefore, the SDP line
599 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
600 * in rtpdec_xiph.c. */
602 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
606 p += strspn(p, SPACE_CHARS);
614 /* get the content */
616 while (*p != '\n' && *p != '\r' && *p != '\0') {
617 if ((q - buf) < sizeof(buf) - 1)
622 sdp_parse_line(s, s1, letter, buf);
624 while (*p != '\n' && *p != '\0')
630 for (i = 0; i < s1->nb_default_include_source_addrs; i++)
631 av_free(s1->default_include_source_addrs[i]);
632 av_freep(&s1->default_include_source_addrs);
633 for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
634 av_free(s1->default_exclude_source_addrs[i]);
635 av_freep(&s1->default_exclude_source_addrs);
637 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
638 if (!rt->p) return AVERROR(ENOMEM);
641 #endif /* CONFIG_RTPDEC */
643 void ff_rtsp_undo_setup(AVFormatContext *s)
645 RTSPState *rt = s->priv_data;
648 for (i = 0; i < rt->nb_rtsp_streams; i++) {
649 RTSPStream *rtsp_st = rt->rtsp_streams[i];
652 if (rtsp_st->transport_priv) {
654 AVFormatContext *rtpctx = rtsp_st->transport_priv;
655 av_write_trailer(rtpctx);
656 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
658 avio_close_dyn_buf(rtpctx->pb, &ptr);
661 avio_close(rtpctx->pb);
663 avformat_free_context(rtpctx);
664 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
665 ff_rdt_parse_close(rtsp_st->transport_priv);
666 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
667 ff_rtp_parse_close(rtsp_st->transport_priv);
669 rtsp_st->transport_priv = NULL;
670 if (rtsp_st->rtp_handle)
671 ffurl_close(rtsp_st->rtp_handle);
672 rtsp_st->rtp_handle = NULL;
676 /* close and free RTSP streams */
677 void ff_rtsp_close_streams(AVFormatContext *s)
679 RTSPState *rt = s->priv_data;
683 ff_rtsp_undo_setup(s);
684 for (i = 0; i < rt->nb_rtsp_streams; i++) {
685 rtsp_st = rt->rtsp_streams[i];
687 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
688 rtsp_st->dynamic_handler->free(
689 rtsp_st->dynamic_protocol_context);
690 for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
691 av_free(rtsp_st->include_source_addrs[j]);
692 av_freep(&rtsp_st->include_source_addrs);
693 for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
694 av_free(rtsp_st->exclude_source_addrs[j]);
695 av_freep(&rtsp_st->exclude_source_addrs);
700 av_free(rt->rtsp_streams);
702 avformat_close_input(&rt->asf_ctx);
704 if (rt->ts && CONFIG_RTPDEC)
705 ff_mpegts_parse_close(rt->ts);
707 av_free(rt->recvbuf);
710 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
712 RTSPState *rt = s->priv_data;
714 int reordering_queue_size = rt->reordering_queue_size;
715 if (reordering_queue_size < 0) {
716 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
717 reordering_queue_size = 0;
719 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
722 /* open the RTP context */
723 if (rtsp_st->stream_index >= 0)
724 st = s->streams[rtsp_st->stream_index];
726 s->ctx_flags |= AVFMTCTX_NOHEADER;
728 if (s->oformat && CONFIG_RTSP_MUXER) {
729 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
731 RTSP_TCP_MAX_PACKET_SIZE,
732 rtsp_st->stream_index);
733 /* Ownership of rtp_handle is passed to the rtp mux context */
734 rtsp_st->rtp_handle = NULL;
737 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
738 return 0; // Don't need to open any parser here
739 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
740 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
741 rtsp_st->dynamic_protocol_context,
742 rtsp_st->dynamic_handler);
743 else if (CONFIG_RTPDEC)
744 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
745 rtsp_st->sdp_payload_type,
746 reordering_queue_size);
748 if (!rtsp_st->transport_priv) {
749 return AVERROR(ENOMEM);
750 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
751 if (rtsp_st->dynamic_handler) {
752 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
753 rtsp_st->dynamic_protocol_context,
754 rtsp_st->dynamic_handler);
756 if (rtsp_st->crypto_suite[0])
757 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
758 rtsp_st->crypto_suite,
759 rtsp_st->crypto_params);
765 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
766 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
773 q += strspn(q, SPACE_CHARS);
774 v = strtol(q, &p, 10);
778 v = strtol(p, &p, 10);
787 /* XXX: only one transport specification is parsed */
788 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
790 char transport_protocol[16];
792 char lower_transport[16];
794 RTSPTransportField *th;
797 reply->nb_transports = 0;
800 p += strspn(p, SPACE_CHARS);
804 th = &reply->transports[reply->nb_transports];
806 get_word_sep(transport_protocol, sizeof(transport_protocol),
808 if (!av_strcasecmp (transport_protocol, "rtp")) {
809 get_word_sep(profile, sizeof(profile), "/;,", &p);
810 lower_transport[0] = '\0';
811 /* rtp/avp/<protocol> */
813 get_word_sep(lower_transport, sizeof(lower_transport),
816 th->transport = RTSP_TRANSPORT_RTP;
817 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
818 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
819 /* x-pn-tng/<protocol> */
820 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
822 th->transport = RTSP_TRANSPORT_RDT;
823 } else if (!av_strcasecmp(transport_protocol, "raw")) {
824 get_word_sep(profile, sizeof(profile), "/;,", &p);
825 lower_transport[0] = '\0';
826 /* raw/raw/<protocol> */
828 get_word_sep(lower_transport, sizeof(lower_transport),
831 th->transport = RTSP_TRANSPORT_RAW;
833 if (!av_strcasecmp(lower_transport, "TCP"))
834 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
836 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
840 /* get each parameter */
841 while (*p != '\0' && *p != ',') {
842 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
843 if (!strcmp(parameter, "port")) {
846 rtsp_parse_range(&th->port_min, &th->port_max, &p);
848 } else if (!strcmp(parameter, "client_port")) {
851 rtsp_parse_range(&th->client_port_min,
852 &th->client_port_max, &p);
854 } else if (!strcmp(parameter, "server_port")) {
857 rtsp_parse_range(&th->server_port_min,
858 &th->server_port_max, &p);
860 } else if (!strcmp(parameter, "interleaved")) {
863 rtsp_parse_range(&th->interleaved_min,
864 &th->interleaved_max, &p);
866 } else if (!strcmp(parameter, "multicast")) {
867 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
868 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
869 } else if (!strcmp(parameter, "ttl")) {
873 th->ttl = strtol(p, &end, 10);
876 } else if (!strcmp(parameter, "destination")) {
879 get_word_sep(buf, sizeof(buf), ";,", &p);
880 get_sockaddr(buf, &th->destination);
882 } else if (!strcmp(parameter, "source")) {
885 get_word_sep(buf, sizeof(buf), ";,", &p);
886 av_strlcpy(th->source, buf, sizeof(th->source));
888 } else if (!strcmp(parameter, "mode")) {
891 get_word_sep(buf, sizeof(buf), ";, ", &p);
892 if (!strcmp(buf, "record") ||
893 !strcmp(buf, "receive"))
898 while (*p != ';' && *p != '\0' && *p != ',')
906 reply->nb_transports++;
910 static void handle_rtp_info(RTSPState *rt, const char *url,
911 uint32_t seq, uint32_t rtptime)
914 if (!rtptime || !url[0])
916 if (rt->transport != RTSP_TRANSPORT_RTP)
918 for (i = 0; i < rt->nb_rtsp_streams; i++) {
919 RTSPStream *rtsp_st = rt->rtsp_streams[i];
920 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
923 if (!strcmp(rtsp_st->control_url, url)) {
924 rtpctx->base_timestamp = rtptime;
930 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
933 char key[20], value[1024], url[1024] = "";
934 uint32_t seq = 0, rtptime = 0;
937 p += strspn(p, SPACE_CHARS);
940 get_word_sep(key, sizeof(key), "=", &p);
944 get_word_sep(value, sizeof(value), ";, ", &p);
946 if (!strcmp(key, "url"))
947 av_strlcpy(url, value, sizeof(url));
948 else if (!strcmp(key, "seq"))
949 seq = strtoul(value, NULL, 10);
950 else if (!strcmp(key, "rtptime"))
951 rtptime = strtoul(value, NULL, 10);
953 handle_rtp_info(rt, url, seq, rtptime);
962 handle_rtp_info(rt, url, seq, rtptime);
965 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
966 RTSPState *rt, const char *method)
970 /* NOTE: we do case independent match for broken servers */
972 if (av_stristart(p, "Session:", &p)) {
974 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
975 if (av_stristart(p, ";timeout=", &p) &&
976 (t = strtol(p, NULL, 10)) > 0) {
979 } else if (av_stristart(p, "Content-Length:", &p)) {
980 reply->content_length = strtol(p, NULL, 10);
981 } else if (av_stristart(p, "Transport:", &p)) {
982 rtsp_parse_transport(reply, p);
983 } else if (av_stristart(p, "CSeq:", &p)) {
984 reply->seq = strtol(p, NULL, 10);
985 } else if (av_stristart(p, "Range:", &p)) {
986 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
987 } else if (av_stristart(p, "RealChallenge1:", &p)) {
988 p += strspn(p, SPACE_CHARS);
989 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
990 } else if (av_stristart(p, "Server:", &p)) {
991 p += strspn(p, SPACE_CHARS);
992 av_strlcpy(reply->server, p, sizeof(reply->server));
993 } else if (av_stristart(p, "Notice:", &p) ||
994 av_stristart(p, "X-Notice:", &p)) {
995 reply->notice = strtol(p, NULL, 10);
996 } else if (av_stristart(p, "Location:", &p)) {
997 p += strspn(p, SPACE_CHARS);
998 av_strlcpy(reply->location, p , sizeof(reply->location));
999 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
1000 p += strspn(p, SPACE_CHARS);
1001 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
1002 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
1003 p += strspn(p, SPACE_CHARS);
1004 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
1005 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
1006 p += strspn(p, SPACE_CHARS);
1007 if (method && !strcmp(method, "DESCRIBE"))
1008 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
1009 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
1010 p += strspn(p, SPACE_CHARS);
1011 if (method && !strcmp(method, "PLAY"))
1012 rtsp_parse_rtp_info(rt, p);
1013 } else if (av_stristart(p, "Public:", &p) && rt) {
1014 if (strstr(p, "GET_PARAMETER") &&
1015 method && !strcmp(method, "OPTIONS"))
1016 rt->get_parameter_supported = 1;
1017 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
1018 p += strspn(p, SPACE_CHARS);
1019 rt->accept_dynamic_rate = atoi(p);
1020 } else if (av_stristart(p, "Content-Type:", &p)) {
1021 p += strspn(p, SPACE_CHARS);
1022 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
1026 /* skip a RTP/TCP interleaved packet */
1027 void ff_rtsp_skip_packet(AVFormatContext *s)
1029 RTSPState *rt = s->priv_data;
1033 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
1036 len = AV_RB16(buf + 1);
1038 av_dlog(s, "skipping RTP packet len=%d\n", len);
1043 if (len1 > sizeof(buf))
1045 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
1052 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
1053 unsigned char **content_ptr,
1054 int return_on_interleaved_data, const char *method)
1056 RTSPState *rt = s->priv_data;
1057 char buf[4096], buf1[1024], *q;
1060 int ret, content_length, line_count = 0, request = 0;
1061 unsigned char *content = NULL;
1067 memset(reply, 0, sizeof(*reply));
1069 /* parse reply (XXX: use buffers) */
1070 rt->last_reply[0] = '\0';
1074 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
1075 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1081 /* XXX: only parse it if first char on line ? */
1082 if (return_on_interleaved_data) {
1085 ff_rtsp_skip_packet(s);
1086 } else if (ch != '\r') {
1087 if ((q - buf) < sizeof(buf) - 1)
1093 av_dlog(s, "line='%s'\n", buf);
1095 /* test if last line */
1099 if (line_count == 0) {
1100 /* get reply code */
1101 get_word(buf1, sizeof(buf1), &p);
1102 if (!strncmp(buf1, "RTSP/", 5)) {
1103 get_word(buf1, sizeof(buf1), &p);
1104 reply->status_code = atoi(buf1);
1105 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1107 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1108 get_word(buf1, sizeof(buf1), &p); // object
1112 ff_rtsp_parse_line(reply, p, rt, method);
1113 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1114 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1119 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1120 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1122 content_length = reply->content_length;
1123 if (content_length > 0) {
1124 /* leave some room for a trailing '\0' (useful for simple parsing) */
1125 content = av_malloc(content_length + 1);
1126 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1127 content[content_length] = '\0';
1130 *content_ptr = content;
1136 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1137 const char* ptr = buf;
1139 if (!strcmp(reply->reason, "OPTIONS")) {
1140 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1142 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1143 if (reply->session_id[0])
1144 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1147 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1149 av_strlcat(buf, "\r\n", sizeof(buf));
1151 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1152 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1155 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1157 rt->last_cmd_time = av_gettime();
1158 /* Even if the request from the server had data, it is not the data
1159 * that the caller wants or expects. The memory could also be leaked
1160 * if the actual following reply has content data. */
1162 av_freep(content_ptr);
1163 /* If method is set, this is called from ff_rtsp_send_cmd,
1164 * where a reply to exactly this request is awaited. For
1165 * callers from within packet receiving, we just want to
1166 * return to the caller and go back to receiving packets. */
1172 if (rt->seq != reply->seq) {
1173 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1174 rt->seq, reply->seq);
1178 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1179 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1180 reply->notice == 2306 /* Continuous Feed Terminated */) {
1181 rt->state = RTSP_STATE_IDLE;
1182 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1183 return AVERROR(EIO); /* data or server error */
1184 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1185 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1186 return AVERROR(EPERM);
1192 * Send a command to the RTSP server without waiting for the reply.
1194 * @param s RTSP (de)muxer context
1195 * @param method the method for the request
1196 * @param url the target url for the request
1197 * @param headers extra header lines to include in the request
1198 * @param send_content if non-null, the data to send as request body content
1199 * @param send_content_length the length of the send_content data, or 0 if
1200 * send_content is null
1202 * @return zero if success, nonzero otherwise
1204 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1205 const char *method, const char *url,
1206 const char *headers,
1207 const unsigned char *send_content,
1208 int send_content_length)
1210 RTSPState *rt = s->priv_data;
1211 char buf[4096], *out_buf;
1212 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1214 /* Add in RTSP headers */
1217 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1219 av_strlcat(buf, headers, sizeof(buf));
1220 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1221 av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n", rt->user_agent);
1222 if (rt->session_id[0] != '\0' && (!headers ||
1223 !strstr(headers, "\nIf-Match:"))) {
1224 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1227 char *str = ff_http_auth_create_response(&rt->auth_state,
1228 rt->auth, url, method);
1230 av_strlcat(buf, str, sizeof(buf));
1233 if (send_content_length > 0 && send_content)
1234 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1235 av_strlcat(buf, "\r\n", sizeof(buf));
1237 /* base64 encode rtsp if tunneling */
1238 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1239 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1240 out_buf = base64buf;
1243 av_dlog(s, "Sending:\n%s--\n", buf);
1245 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1246 if (send_content_length > 0 && send_content) {
1247 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1248 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1249 "with content data not supported\n");
1250 return AVERROR_PATCHWELCOME;
1252 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1254 rt->last_cmd_time = av_gettime();
1259 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1260 const char *url, const char *headers)
1262 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1265 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1266 const char *headers, RTSPMessageHeader *reply,
1267 unsigned char **content_ptr)
1269 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1270 content_ptr, NULL, 0);
1273 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1274 const char *method, const char *url,
1276 RTSPMessageHeader *reply,
1277 unsigned char **content_ptr,
1278 const unsigned char *send_content,
1279 int send_content_length)
1281 RTSPState *rt = s->priv_data;
1282 HTTPAuthType cur_auth_type;
1283 int ret, attempts = 0;
1286 cur_auth_type = rt->auth_state.auth_type;
1287 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1289 send_content_length)))
1292 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1296 if (reply->status_code == 401 &&
1297 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1298 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1301 if (reply->status_code > 400){
1302 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1306 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1312 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1313 int lower_transport, const char *real_challenge)
1315 RTSPState *rt = s->priv_data;
1316 int rtx = 0, j, i, err, interleave = 0, port_off;
1317 RTSPStream *rtsp_st;
1318 RTSPMessageHeader reply1, *reply = &reply1;
1320 const char *trans_pref;
1322 if (rt->transport == RTSP_TRANSPORT_RDT)
1323 trans_pref = "x-pn-tng";
1324 else if (rt->transport == RTSP_TRANSPORT_RAW)
1325 trans_pref = "RAW/RAW";
1327 trans_pref = "RTP/AVP";
1329 /* default timeout: 1 minute */
1332 /* Choose a random starting offset within the first half of the
1333 * port range, to allow for a number of ports to try even if the offset
1334 * happens to be at the end of the random range. */
1335 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1336 /* even random offset */
1337 port_off -= port_off & 0x01;
1339 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1340 char transport[2048];
1343 * WMS serves all UDP data over a single connection, the RTX, which
1344 * isn't necessarily the first in the SDP but has to be the first
1345 * to be set up, else the second/third SETUP will fail with a 461.
1347 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1348 rt->server_type == RTSP_SERVER_WMS) {
1351 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1352 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1354 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1358 if (rtx == rt->nb_rtsp_streams)
1359 return -1; /* no RTX found */
1360 rtsp_st = rt->rtsp_streams[rtx];
1362 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1364 rtsp_st = rt->rtsp_streams[i];
1367 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1370 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1371 port = reply->transports[0].client_port_min;
1375 /* first try in specified port range */
1376 while (j <= rt->rtp_port_max) {
1377 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1378 "?localport=%d", j);
1379 /* we will use two ports per rtp stream (rtp and rtcp) */
1381 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1382 &s->interrupt_callback, NULL))
1385 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1390 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1392 snprintf(transport, sizeof(transport) - 1,
1393 "%s/UDP;", trans_pref);
1394 if (rt->server_type != RTSP_SERVER_REAL)
1395 av_strlcat(transport, "unicast;", sizeof(transport));
1396 av_strlcatf(transport, sizeof(transport),
1397 "client_port=%d", port);
1398 if (rt->transport == RTSP_TRANSPORT_RTP &&
1399 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1400 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1404 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1405 /* For WMS streams, the application streams are only used for
1406 * UDP. When trying to set it up for TCP streams, the server
1407 * will return an error. Therefore, we skip those streams. */
1408 if (rt->server_type == RTSP_SERVER_WMS &&
1409 (rtsp_st->stream_index < 0 ||
1410 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1413 snprintf(transport, sizeof(transport) - 1,
1414 "%s/TCP;", trans_pref);
1415 if (rt->transport != RTSP_TRANSPORT_RDT)
1416 av_strlcat(transport, "unicast;", sizeof(transport));
1417 av_strlcatf(transport, sizeof(transport),
1418 "interleaved=%d-%d",
1419 interleave, interleave + 1);
1423 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1424 snprintf(transport, sizeof(transport) - 1,
1425 "%s/UDP;multicast", trans_pref);
1428 av_strlcat(transport, ";mode=record", sizeof(transport));
1429 } else if (rt->server_type == RTSP_SERVER_REAL ||
1430 rt->server_type == RTSP_SERVER_WMS)
1431 av_strlcat(transport, ";mode=play", sizeof(transport));
1432 snprintf(cmd, sizeof(cmd),
1433 "Transport: %s\r\n",
1435 if (rt->accept_dynamic_rate)
1436 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1437 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1438 char real_res[41], real_csum[9];
1439 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1441 av_strlcatf(cmd, sizeof(cmd),
1443 "RealChallenge2: %s, sd=%s\r\n",
1444 rt->session_id, real_res, real_csum);
1446 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1447 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1450 } else if (reply->status_code != RTSP_STATUS_OK ||
1451 reply->nb_transports != 1) {
1452 err = AVERROR_INVALIDDATA;
1456 /* XXX: same protocol for all streams is required */
1458 if (reply->transports[0].lower_transport != rt->lower_transport ||
1459 reply->transports[0].transport != rt->transport) {
1460 err = AVERROR_INVALIDDATA;
1464 rt->lower_transport = reply->transports[0].lower_transport;
1465 rt->transport = reply->transports[0].transport;
1468 /* Fail if the server responded with another lower transport mode
1469 * than what we requested. */
1470 if (reply->transports[0].lower_transport != lower_transport) {
1471 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1472 err = AVERROR_INVALIDDATA;
1476 switch(reply->transports[0].lower_transport) {
1477 case RTSP_LOWER_TRANSPORT_TCP:
1478 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1479 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1482 case RTSP_LOWER_TRANSPORT_UDP: {
1483 char url[1024], options[30] = "";
1484 const char *peer = host;
1486 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1487 av_strlcpy(options, "?connect=1", sizeof(options));
1488 /* Use source address if specified */
1489 if (reply->transports[0].source[0])
1490 peer = reply->transports[0].source;
1491 ff_url_join(url, sizeof(url), "rtp", NULL, peer,
1492 reply->transports[0].server_port_min, "%s", options);
1493 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1494 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1495 err = AVERROR_INVALIDDATA;
1498 /* Try to initialize the connection state in a
1499 * potential NAT router by sending dummy packets.
1500 * RTP/RTCP dummy packets are used for RDT, too.
1502 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1504 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1507 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1508 char url[1024], namebuf[50], optbuf[20] = "";
1509 struct sockaddr_storage addr;
1512 if (reply->transports[0].destination.ss_family) {
1513 addr = reply->transports[0].destination;
1514 port = reply->transports[0].port_min;
1515 ttl = reply->transports[0].ttl;
1517 addr = rtsp_st->sdp_ip;
1518 port = rtsp_st->sdp_port;
1519 ttl = rtsp_st->sdp_ttl;
1522 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1523 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1524 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1525 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1526 port, "%s", optbuf);
1527 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1528 &s->interrupt_callback, NULL) < 0) {
1529 err = AVERROR_INVALIDDATA;
1536 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1540 if (rt->nb_rtsp_streams && reply->timeout > 0)
1541 rt->timeout = reply->timeout;
1543 if (rt->server_type == RTSP_SERVER_REAL)
1544 rt->need_subscription = 1;
1549 ff_rtsp_undo_setup(s);
1553 void ff_rtsp_close_connections(AVFormatContext *s)
1555 RTSPState *rt = s->priv_data;
1556 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1557 ffurl_close(rt->rtsp_hd);
1558 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1561 int ff_rtsp_connect(AVFormatContext *s)
1563 RTSPState *rt = s->priv_data;
1564 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1565 int port, err, tcp_fd;
1566 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1567 int lower_transport_mask = 0;
1568 char real_challenge[64] = "";
1569 struct sockaddr_storage peer;
1570 socklen_t peer_len = sizeof(peer);
1572 if (rt->rtp_port_max < rt->rtp_port_min) {
1573 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1574 "than min port %d\n", rt->rtp_port_max,
1576 return AVERROR(EINVAL);
1579 if (!ff_network_init())
1580 return AVERROR(EIO);
1582 if (s->max_delay < 0) /* Not set by the caller */
1583 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1585 rt->control_transport = RTSP_MODE_PLAIN;
1586 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1587 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1588 rt->control_transport = RTSP_MODE_TUNNEL;
1590 /* Only pass through valid flags from here */
1591 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1594 lower_transport_mask = rt->lower_transport_mask;
1595 /* extract hostname and port */
1596 av_url_split(NULL, 0, auth, sizeof(auth),
1597 host, sizeof(host), &port, path, sizeof(path), s->filename);
1599 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1602 port = RTSP_DEFAULT_PORT;
1604 if (!lower_transport_mask)
1605 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1608 /* Only UDP or TCP - UDP multicast isn't supported. */
1609 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1610 (1 << RTSP_LOWER_TRANSPORT_TCP);
1611 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1612 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1613 "only UDP and TCP are supported for output.\n");
1614 err = AVERROR(EINVAL);
1619 /* Construct the URI used in request; this is similar to s->filename,
1620 * but with authentication credentials removed and RTSP specific options
1622 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1623 host, port, "%s", path);
1625 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1626 /* set up initial handshake for tunneling */
1627 char httpname[1024];
1628 char sessioncookie[17];
1631 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1632 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1633 av_get_random_seed(), av_get_random_seed());
1636 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1637 &s->interrupt_callback) < 0) {
1642 /* generate GET headers */
1643 snprintf(headers, sizeof(headers),
1644 "x-sessioncookie: %s\r\n"
1645 "Accept: application/x-rtsp-tunnelled\r\n"
1646 "Pragma: no-cache\r\n"
1647 "Cache-Control: no-cache\r\n",
1649 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1651 /* complete the connection */
1652 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1658 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1659 &s->interrupt_callback) < 0 ) {
1664 /* generate POST headers */
1665 snprintf(headers, sizeof(headers),
1666 "x-sessioncookie: %s\r\n"
1667 "Content-Type: application/x-rtsp-tunnelled\r\n"
1668 "Pragma: no-cache\r\n"
1669 "Cache-Control: no-cache\r\n"
1670 "Content-Length: 32767\r\n"
1671 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1673 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1674 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1676 /* Initialize the authentication state for the POST session. The HTTP
1677 * protocol implementation doesn't properly handle multi-pass
1678 * authentication for POST requests, since it would require one of
1680 * - implementing Expect: 100-continue, which many HTTP servers
1681 * don't support anyway, even less the RTSP servers that do HTTP
1683 * - sending the whole POST data until getting a 401 reply specifying
1684 * what authentication method to use, then resending all that data
1685 * - waiting for potential 401 replies directly after sending the
1686 * POST header (waiting for some unspecified time)
1687 * Therefore, we copy the full auth state, which works for both basic
1688 * and digest. (For digest, we would have to synchronize the nonce
1689 * count variable between the two sessions, if we'd do more requests
1690 * with the original session, though.)
1692 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1694 /* complete the connection */
1695 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1700 /* open the tcp connection */
1701 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1702 "?timeout=%d", rt->stimeout);
1703 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1704 &s->interrupt_callback, NULL) < 0) {
1708 rt->rtsp_hd_out = rt->rtsp_hd;
1712 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1713 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1714 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1715 NULL, 0, NI_NUMERICHOST);
1718 /* request options supported by the server; this also detects server
1720 for (rt->server_type = RTSP_SERVER_RTP;;) {
1722 if (rt->server_type == RTSP_SERVER_REAL)
1725 * The following entries are required for proper
1726 * streaming from a Realmedia server. They are
1727 * interdependent in some way although we currently
1728 * don't quite understand how. Values were copied
1729 * from mplayer SVN r23589.
1730 * ClientChallenge is a 16-byte ID in hex
1731 * CompanyID is a 16-byte ID in base64
1733 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1734 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1735 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1736 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1738 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1739 if (reply->status_code != RTSP_STATUS_OK) {
1740 err = AVERROR_INVALIDDATA;
1744 /* detect server type if not standard-compliant RTP */
1745 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1746 rt->server_type = RTSP_SERVER_REAL;
1748 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1749 rt->server_type = RTSP_SERVER_WMS;
1750 } else if (rt->server_type == RTSP_SERVER_REAL)
1751 strcpy(real_challenge, reply->real_challenge);
1755 if (s->iformat && CONFIG_RTSP_DEMUXER)
1756 err = ff_rtsp_setup_input_streams(s, reply);
1757 else if (CONFIG_RTSP_MUXER)
1758 err = ff_rtsp_setup_output_streams(s, host);
1763 int lower_transport = ff_log2_tab[lower_transport_mask &
1764 ~(lower_transport_mask - 1)];
1766 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1767 rt->server_type == RTSP_SERVER_REAL ?
1768 real_challenge : NULL);
1771 lower_transport_mask &= ~(1 << lower_transport);
1772 if (lower_transport_mask == 0 && err == 1) {
1773 err = AVERROR(EPROTONOSUPPORT);
1778 rt->lower_transport_mask = lower_transport_mask;
1779 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1780 rt->state = RTSP_STATE_IDLE;
1781 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1784 ff_rtsp_close_streams(s);
1785 ff_rtsp_close_connections(s);
1786 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1787 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1788 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1796 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1799 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1800 uint8_t *buf, int buf_size, int64_t wait_end)
1802 RTSPState *rt = s->priv_data;
1803 RTSPStream *rtsp_st;
1804 int n, i, ret, tcp_fd, timeout_cnt = 0;
1806 struct pollfd *p = rt->p;
1807 int *fds = NULL, fdsnum, fdsidx;
1810 if (ff_check_interrupt(&s->interrupt_callback))
1811 return AVERROR_EXIT;
1812 if (wait_end && wait_end - av_gettime() < 0)
1813 return AVERROR(EAGAIN);
1816 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1817 p[max_p].fd = tcp_fd;
1818 p[max_p++].events = POLLIN;
1822 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1823 rtsp_st = rt->rtsp_streams[i];
1824 if (rtsp_st->rtp_handle) {
1825 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1827 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1831 av_log(s, AV_LOG_ERROR,
1832 "Number of fds %d not supported\n", fdsnum);
1833 return AVERROR_INVALIDDATA;
1835 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1836 p[max_p].fd = fds[fdsidx];
1837 p[max_p++].events = POLLIN;
1842 n = poll(p, max_p, POLL_TIMEOUT_MS);
1844 int j = 1 - (tcp_fd == -1);
1846 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1847 rtsp_st = rt->rtsp_streams[i];
1848 if (rtsp_st->rtp_handle) {
1849 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1850 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1852 *prtsp_st = rtsp_st;
1859 #if CONFIG_RTSP_DEMUXER
1860 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1861 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1862 if (rt->state == RTSP_STATE_STREAMING) {
1863 if (!ff_rtsp_parse_streaming_commands(s))
1866 av_log(s, AV_LOG_WARNING,
1867 "Unable to answer to TEARDOWN\n");
1871 RTSPMessageHeader reply;
1872 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1875 /* XXX: parse message */
1876 if (rt->state != RTSP_STATE_STREAMING)
1881 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1882 return AVERROR(ETIMEDOUT);
1883 } else if (n < 0 && errno != EINTR)
1884 return AVERROR(errno);
1888 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1889 const uint8_t *buf, int len)
1891 RTSPState *rt = s->priv_data;
1895 if (rt->nb_rtsp_streams == 1) {
1896 *rtsp_st = rt->rtsp_streams[0];
1899 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1900 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1902 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1903 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1906 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1907 *rtsp_st = rt->rtsp_streams[i];
1914 av_log(s, AV_LOG_WARNING,
1915 "Unable to pick stream for packet - SSRC not known for "
1917 return AVERROR(EAGAIN);
1920 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1921 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1922 *rtsp_st = rt->rtsp_streams[i];
1928 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1929 return AVERROR(EAGAIN);
1932 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1934 RTSPState *rt = s->priv_data;
1936 RTSPStream *rtsp_st, *first_queue_st = NULL;
1937 int64_t wait_end = 0;
1939 if (rt->nb_byes == rt->nb_rtsp_streams)
1942 /* get next frames from the same RTP packet */
1943 if (rt->cur_transport_priv) {
1944 if (rt->transport == RTSP_TRANSPORT_RDT) {
1945 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1946 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1947 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1948 } else if (rt->ts && CONFIG_RTPDEC) {
1949 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1951 rt->recvbuf_pos += ret;
1952 ret = rt->recvbuf_pos < rt->recvbuf_len;
1957 rt->cur_transport_priv = NULL;
1959 } else if (ret == 1) {
1962 rt->cur_transport_priv = NULL;
1966 if (rt->transport == RTSP_TRANSPORT_RTP) {
1968 int64_t first_queue_time = 0;
1969 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1970 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1974 queue_time = ff_rtp_queued_packet_time(rtpctx);
1975 if (queue_time && (queue_time - first_queue_time < 0 ||
1976 !first_queue_time)) {
1977 first_queue_time = queue_time;
1978 first_queue_st = rt->rtsp_streams[i];
1981 if (first_queue_time) {
1982 wait_end = first_queue_time + s->max_delay;
1985 first_queue_st = NULL;
1989 /* read next RTP packet */
1991 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1993 return AVERROR(ENOMEM);
1996 switch(rt->lower_transport) {
1998 #if CONFIG_RTSP_DEMUXER
1999 case RTSP_LOWER_TRANSPORT_TCP:
2000 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
2003 case RTSP_LOWER_TRANSPORT_UDP:
2004 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
2005 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
2006 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2007 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
2009 case RTSP_LOWER_TRANSPORT_CUSTOM:
2010 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
2011 wait_end && wait_end < av_gettime())
2012 len = AVERROR(EAGAIN);
2014 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
2015 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
2016 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
2017 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
2020 if (len == AVERROR(EAGAIN) && first_queue_st &&
2021 rt->transport == RTSP_TRANSPORT_RTP) {
2022 rtsp_st = first_queue_st;
2023 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
2030 if (rt->transport == RTSP_TRANSPORT_RDT) {
2031 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2032 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
2033 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
2034 if (rtsp_st->feedback) {
2035 AVIOContext *pb = NULL;
2036 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
2038 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
2041 /* Either bad packet, or a RTCP packet. Check if the
2042 * first_rtcp_ntp_time field was initialized. */
2043 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
2044 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
2045 /* first_rtcp_ntp_time has been initialized for this stream,
2046 * copy the same value to all other uninitialized streams,
2047 * in order to map their timestamp origin to the same ntp time
2050 AVStream *st = NULL;
2051 if (rtsp_st->stream_index >= 0)
2052 st = s->streams[rtsp_st->stream_index];
2053 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2054 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
2055 AVStream *st2 = NULL;
2056 if (rt->rtsp_streams[i]->stream_index >= 0)
2057 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
2058 if (rtpctx2 && st && st2 &&
2059 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
2060 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
2061 rtpctx2->rtcp_ts_offset = av_rescale_q(
2062 rtpctx->rtcp_ts_offset, st->time_base,
2067 if (ret == -RTCP_BYE) {
2070 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
2071 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
2073 if (rt->nb_byes == rt->nb_rtsp_streams)
2077 } else if (rt->ts && CONFIG_RTPDEC) {
2078 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2081 rt->recvbuf_len = len;
2082 rt->recvbuf_pos = ret;
2083 rt->cur_transport_priv = rt->ts;
2090 return AVERROR_INVALIDDATA;
2096 /* more packets may follow, so we save the RTP context */
2097 rt->cur_transport_priv = rtsp_st->transport_priv;
2101 #endif /* CONFIG_RTPDEC */
2103 #if CONFIG_SDP_DEMUXER
2104 static int sdp_probe(AVProbeData *p1)
2106 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2108 /* we look for a line beginning "c=IN IP" */
2109 while (p < p_end && *p != '\0') {
2110 if (p + sizeof("c=IN IP") - 1 < p_end &&
2111 av_strstart(p, "c=IN IP", NULL))
2112 return AVPROBE_SCORE_EXTENSION;
2114 while (p < p_end - 1 && *p != '\n') p++;
2123 static void append_source_addrs(char *buf, int size, const char *name,
2124 int count, struct RTSPSource **addrs)
2129 av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
2130 for (i = 1; i < count; i++)
2131 av_strlcatf(buf, size, ",%s", addrs[i]->addr);
2134 static int sdp_read_header(AVFormatContext *s)
2136 RTSPState *rt = s->priv_data;
2137 RTSPStream *rtsp_st;
2142 if (!ff_network_init())
2143 return AVERROR(EIO);
2145 if (s->max_delay < 0) /* Not set by the caller */
2146 s->max_delay = DEFAULT_REORDERING_DELAY;
2147 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2148 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2150 /* read the whole sdp file */
2151 /* XXX: better loading */
2152 content = av_malloc(SDP_MAX_SIZE);
2153 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2156 return AVERROR_INVALIDDATA;
2158 content[size] ='\0';
2160 err = ff_sdp_parse(s, content);
2164 /* open each RTP stream */
2165 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2167 rtsp_st = rt->rtsp_streams[i];
2169 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2170 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2171 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2172 ff_url_join(url, sizeof(url), "rtp", NULL,
2173 namebuf, rtsp_st->sdp_port,
2174 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2176 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2178 append_source_addrs(url, sizeof(url), "sources",
2179 rtsp_st->nb_include_source_addrs,
2180 rtsp_st->include_source_addrs);
2181 append_source_addrs(url, sizeof(url), "block",
2182 rtsp_st->nb_exclude_source_addrs,
2183 rtsp_st->exclude_source_addrs);
2184 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2185 &s->interrupt_callback, NULL) < 0) {
2186 err = AVERROR_INVALIDDATA;
2190 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2195 ff_rtsp_close_streams(s);
2200 static int sdp_read_close(AVFormatContext *s)
2202 ff_rtsp_close_streams(s);
2207 static const AVClass sdp_demuxer_class = {
2208 .class_name = "SDP demuxer",
2209 .item_name = av_default_item_name,
2210 .option = sdp_options,
2211 .version = LIBAVUTIL_VERSION_INT,
2214 AVInputFormat ff_sdp_demuxer = {
2216 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2217 .priv_data_size = sizeof(RTSPState),
2218 .read_probe = sdp_probe,
2219 .read_header = sdp_read_header,
2220 .read_packet = ff_rtsp_fetch_packet,
2221 .read_close = sdp_read_close,
2222 .priv_class = &sdp_demuxer_class,
2224 #endif /* CONFIG_SDP_DEMUXER */
2226 #if CONFIG_RTP_DEMUXER
2227 static int rtp_probe(AVProbeData *p)
2229 if (av_strstart(p->filename, "rtp:", NULL))
2230 return AVPROBE_SCORE_MAX;
2234 static int rtp_read_header(AVFormatContext *s)
2236 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2237 char host[500], sdp[500];
2239 URLContext* in = NULL;
2241 AVCodecContext codec = { 0 };
2242 struct sockaddr_storage addr;
2244 socklen_t addrlen = sizeof(addr);
2245 RTSPState *rt = s->priv_data;
2247 if (!ff_network_init())
2248 return AVERROR(EIO);
2250 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2251 &s->interrupt_callback, NULL);
2256 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2257 if (ret == AVERROR(EAGAIN))
2262 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2266 if ((recvbuf[0] & 0xc0) != 0x80) {
2267 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2272 if (RTP_PT_IS_RTCP(recvbuf[1]))
2275 payload_type = recvbuf[1] & 0x7f;
2278 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2282 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2283 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2284 "without an SDP file describing it\n",
2288 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2289 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2290 "properly you need an SDP file "
2294 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2295 NULL, 0, s->filename);
2297 snprintf(sdp, sizeof(sdp),
2298 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2299 addr.ss_family == AF_INET ? 4 : 6, host,
2300 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2301 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2302 port, payload_type);
2303 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2305 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2308 /* sdp_read_header initializes this again */
2311 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2313 ret = sdp_read_header(s);
2324 static const AVClass rtp_demuxer_class = {
2325 .class_name = "RTP demuxer",
2326 .item_name = av_default_item_name,
2327 .option = rtp_options,
2328 .version = LIBAVUTIL_VERSION_INT,
2331 AVInputFormat ff_rtp_demuxer = {
2333 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2334 .priv_data_size = sizeof(RTSPState),
2335 .read_probe = rtp_probe,
2336 .read_header = rtp_read_header,
2337 .read_packet = ff_rtsp_fetch_packet,
2338 .read_close = sdp_read_close,
2339 .flags = AVFMT_NOFILE,
2340 .priv_class = &rtp_demuxer_class,
2342 #endif /* CONFIG_RTP_DEMUXER */