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"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define DEFAULT_USER_AGENT "Lavf/" AV_STRINGIFY(LIBAVFORMAT_VERSION)
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "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 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73 { 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" }, \
74 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
78 #define RTSP_REORDERING_OPTS() \
79 { "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 }
81 const AVOption ff_rtsp_options[] = {
82 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "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 },
94 { "stimeout", "timeout (in micro seconds) of socket i/o operations.", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
95 RTSP_REORDERING_OPTS(),
96 { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = DEFAULT_USER_AGENT}, 0, 0, DEC },
100 static const AVOption sdp_options[] = {
101 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
102 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
103 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
104 RTSP_REORDERING_OPTS(),
108 static const AVOption rtp_options[] = {
109 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
110 RTSP_REORDERING_OPTS(),
114 static void get_word_until_chars(char *buf, int buf_size,
115 const char *sep, const char **pp)
121 p += strspn(p, SPACE_CHARS);
123 while (!strchr(sep, *p) && *p != '\0') {
124 if ((q - buf) < buf_size - 1)
133 static void get_word_sep(char *buf, int buf_size, const char *sep,
136 if (**pp == '/') (*pp)++;
137 get_word_until_chars(buf, buf_size, sep, pp);
140 static void get_word(char *buf, int buf_size, const char **pp)
142 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
145 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
147 * Used for seeking in the rtp stream.
149 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
153 p += strspn(p, SPACE_CHARS);
154 if (!av_stristart(p, "npt=", &p))
157 *start = AV_NOPTS_VALUE;
158 *end = AV_NOPTS_VALUE;
160 get_word_sep(buf, sizeof(buf), "-", &p);
161 av_parse_time(start, buf, 1);
164 get_word_sep(buf, sizeof(buf), "-", &p);
165 av_parse_time(end, buf, 1);
169 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
171 struct addrinfo hints = { 0 }, *ai = NULL;
172 hints.ai_flags = AI_NUMERICHOST;
173 if (getaddrinfo(buf, NULL, &hints, &ai))
175 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
181 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
182 RTSPStream *rtsp_st, AVCodecContext *codec)
187 codec->codec_id = handler->codec_id;
188 rtsp_st->dynamic_handler = handler;
189 if (handler->alloc) {
190 rtsp_st->dynamic_protocol_context = handler->alloc();
191 if (!rtsp_st->dynamic_protocol_context)
192 rtsp_st->dynamic_handler = NULL;
196 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
197 static int sdp_parse_rtpmap(AVFormatContext *s,
198 AVStream *st, RTSPStream *rtsp_st,
199 int payload_type, const char *p)
201 AVCodecContext *codec = st->codec;
207 /* See if we can handle this kind of payload.
208 * The space should normally not be there but some Real streams or
209 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
210 * have a trailing space. */
211 get_word_sep(buf, sizeof(buf), "/ ", &p);
212 if (payload_type < RTP_PT_PRIVATE) {
213 /* We are in a standard case
214 * (from http://www.iana.org/assignments/rtp-parameters). */
215 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
218 if (codec->codec_id == AV_CODEC_ID_NONE) {
219 RTPDynamicProtocolHandler *handler =
220 ff_rtp_handler_find_by_name(buf, codec->codec_type);
221 init_rtp_handler(handler, rtsp_st, codec);
222 /* If no dynamic handler was found, check with the list of standard
223 * allocated types, if such a stream for some reason happens to
224 * use a private payload type. This isn't handled in rtpdec.c, since
225 * the format name from the rtpmap line never is passed into rtpdec. */
226 if (!rtsp_st->dynamic_handler)
227 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
230 c = avcodec_find_decoder(codec->codec_id);
236 get_word_sep(buf, sizeof(buf), "/", &p);
238 switch (codec->codec_type) {
239 case AVMEDIA_TYPE_AUDIO:
240 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
241 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
242 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
244 codec->sample_rate = i;
245 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
246 get_word_sep(buf, sizeof(buf), "/", &p);
251 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
253 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
256 case AVMEDIA_TYPE_VIDEO:
257 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
259 avpriv_set_pts_info(st, 32, 1, i);
264 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
265 rtsp_st->dynamic_handler->init(s, st->index,
266 rtsp_st->dynamic_protocol_context);
270 /* parse the attribute line from the fmtp a line of an sdp response. This
271 * is broken out as a function because it is used in rtp_h264.c, which is
273 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
274 char *value, int value_size)
276 *p += strspn(*p, SPACE_CHARS);
278 get_word_sep(attr, attr_size, "=", p);
281 get_word_sep(value, value_size, ";", p);
289 typedef struct SDPParseState {
291 struct sockaddr_storage default_ip;
293 int skip_media; ///< set if an unknown m= line occurs
296 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
297 int letter, const char *buf)
299 RTSPState *rt = s->priv_data;
300 char buf1[64], st_type[64];
302 enum AVMediaType codec_type;
306 struct sockaddr_storage sdp_ip;
309 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
312 if (s1->skip_media && letter != 'm')
316 get_word(buf1, sizeof(buf1), &p);
317 if (strcmp(buf1, "IN") != 0)
319 get_word(buf1, sizeof(buf1), &p);
320 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
322 get_word_sep(buf1, sizeof(buf1), "/", &p);
323 if (get_sockaddr(buf1, &sdp_ip))
328 get_word_sep(buf1, sizeof(buf1), "/", &p);
331 if (s->nb_streams == 0) {
332 s1->default_ip = sdp_ip;
333 s1->default_ttl = ttl;
335 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
336 rtsp_st->sdp_ip = sdp_ip;
337 rtsp_st->sdp_ttl = ttl;
341 av_dict_set(&s->metadata, "title", p, 0);
344 if (s->nb_streams == 0) {
345 av_dict_set(&s->metadata, "comment", p, 0);
352 codec_type = AVMEDIA_TYPE_UNKNOWN;
353 get_word(st_type, sizeof(st_type), &p);
354 if (!strcmp(st_type, "audio")) {
355 codec_type = AVMEDIA_TYPE_AUDIO;
356 } else if (!strcmp(st_type, "video")) {
357 codec_type = AVMEDIA_TYPE_VIDEO;
358 } else if (!strcmp(st_type, "application")) {
359 codec_type = AVMEDIA_TYPE_DATA;
361 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
365 rtsp_st = av_mallocz(sizeof(RTSPStream));
368 rtsp_st->stream_index = -1;
369 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
371 rtsp_st->sdp_ip = s1->default_ip;
372 rtsp_st->sdp_ttl = s1->default_ttl;
374 get_word(buf1, sizeof(buf1), &p); /* port */
375 rtsp_st->sdp_port = atoi(buf1);
377 get_word(buf1, sizeof(buf1), &p); /* protocol */
378 if (!strcmp(buf1, "udp"))
379 rt->transport = RTSP_TRANSPORT_RAW;
380 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
381 rtsp_st->feedback = 1;
383 /* XXX: handle list of formats */
384 get_word(buf1, sizeof(buf1), &p); /* format list */
385 rtsp_st->sdp_payload_type = atoi(buf1);
387 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
388 /* no corresponding stream */
389 if (rt->transport == RTSP_TRANSPORT_RAW) {
390 if (!rt->ts && CONFIG_RTPDEC)
391 rt->ts = ff_mpegts_parse_open(s);
393 RTPDynamicProtocolHandler *handler;
394 handler = ff_rtp_handler_find_by_id(
395 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
396 init_rtp_handler(handler, rtsp_st, NULL);
397 if (handler && handler->init)
398 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
400 } else if (rt->server_type == RTSP_SERVER_WMS &&
401 codec_type == AVMEDIA_TYPE_DATA) {
402 /* RTX stream, a stream that carries all the other actual
403 * audio/video streams. Don't expose this to the callers. */
405 st = avformat_new_stream(s, NULL);
408 st->id = rt->nb_rtsp_streams - 1;
409 rtsp_st->stream_index = st->index;
410 st->codec->codec_type = codec_type;
411 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
412 RTPDynamicProtocolHandler *handler;
413 /* if standard payload type, we can find the codec right now */
414 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
415 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
416 st->codec->sample_rate > 0)
417 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
418 /* Even static payload types may need a custom depacketizer */
419 handler = ff_rtp_handler_find_by_id(
420 rtsp_st->sdp_payload_type, st->codec->codec_type);
421 init_rtp_handler(handler, rtsp_st, st->codec);
422 if (handler && handler->init)
423 handler->init(s, st->index,
424 rtsp_st->dynamic_protocol_context);
427 /* put a default control url */
428 av_strlcpy(rtsp_st->control_url, rt->control_uri,
429 sizeof(rtsp_st->control_url));
432 if (av_strstart(p, "control:", &p)) {
433 if (s->nb_streams == 0) {
434 if (!strncmp(p, "rtsp://", 7))
435 av_strlcpy(rt->control_uri, p,
436 sizeof(rt->control_uri));
439 /* get the control url */
440 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
442 /* XXX: may need to add full url resolution */
443 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
445 if (proto[0] == '\0') {
446 /* relative control URL */
447 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
448 av_strlcat(rtsp_st->control_url, "/",
449 sizeof(rtsp_st->control_url));
450 av_strlcat(rtsp_st->control_url, p,
451 sizeof(rtsp_st->control_url));
453 av_strlcpy(rtsp_st->control_url, p,
454 sizeof(rtsp_st->control_url));
456 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
457 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
458 get_word(buf1, sizeof(buf1), &p);
459 payload_type = atoi(buf1);
460 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
461 if (rtsp_st->stream_index >= 0) {
462 st = s->streams[rtsp_st->stream_index];
463 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
465 } else if (av_strstart(p, "fmtp:", &p) ||
466 av_strstart(p, "framesize:", &p)) {
467 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
468 // let dynamic protocol handlers have a stab at the line.
469 get_word(buf1, sizeof(buf1), &p);
470 payload_type = atoi(buf1);
471 for (i = 0; i < rt->nb_rtsp_streams; i++) {
472 rtsp_st = rt->rtsp_streams[i];
473 if (rtsp_st->sdp_payload_type == payload_type &&
474 rtsp_st->dynamic_handler &&
475 rtsp_st->dynamic_handler->parse_sdp_a_line)
476 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
477 rtsp_st->dynamic_protocol_context, buf);
479 } else if (av_strstart(p, "range:", &p)) {
482 // this is so that seeking on a streamed file can work.
483 rtsp_parse_range_npt(p, &start, &end);
484 s->start_time = start;
485 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
486 s->duration = (end == AV_NOPTS_VALUE) ?
487 AV_NOPTS_VALUE : end - start;
488 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
490 rt->transport = RTSP_TRANSPORT_RDT;
491 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
493 st = s->streams[s->nb_streams - 1];
494 st->codec->sample_rate = atoi(p);
495 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
497 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
498 get_word(buf1, sizeof(buf1), &p); // ignore tag
499 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
500 p += strspn(p, SPACE_CHARS);
501 if (av_strstart(p, "inline:", &p))
502 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
504 if (rt->server_type == RTSP_SERVER_WMS)
505 ff_wms_parse_sdp_a_line(s, p);
506 if (s->nb_streams > 0) {
507 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
509 if (rt->server_type == RTSP_SERVER_REAL)
510 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
512 if (rtsp_st->dynamic_handler &&
513 rtsp_st->dynamic_handler->parse_sdp_a_line)
514 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
515 rtsp_st->stream_index,
516 rtsp_st->dynamic_protocol_context, buf);
523 int ff_sdp_parse(AVFormatContext *s, const char *content)
525 RTSPState *rt = s->priv_data;
528 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
529 * contain long SDP lines containing complete ASF Headers (several
530 * kB) or arrays of MDPR (RM stream descriptor) headers plus
531 * "rulebooks" describing their properties. Therefore, the SDP line
534 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
535 * in rtpdec_xiph.c. */
537 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
541 p += strspn(p, SPACE_CHARS);
549 /* get the content */
551 while (*p != '\n' && *p != '\r' && *p != '\0') {
552 if ((q - buf) < sizeof(buf) - 1)
557 sdp_parse_line(s, s1, letter, buf);
559 while (*p != '\n' && *p != '\0')
564 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
565 if (!rt->p) return AVERROR(ENOMEM);
568 #endif /* CONFIG_RTPDEC */
570 void ff_rtsp_undo_setup(AVFormatContext *s)
572 RTSPState *rt = s->priv_data;
575 for (i = 0; i < rt->nb_rtsp_streams; i++) {
576 RTSPStream *rtsp_st = rt->rtsp_streams[i];
579 if (rtsp_st->transport_priv) {
581 AVFormatContext *rtpctx = rtsp_st->transport_priv;
582 av_write_trailer(rtpctx);
583 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
585 avio_close_dyn_buf(rtpctx->pb, &ptr);
588 avio_close(rtpctx->pb);
590 avformat_free_context(rtpctx);
591 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
592 ff_rdt_parse_close(rtsp_st->transport_priv);
593 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
594 ff_rtp_parse_close(rtsp_st->transport_priv);
596 rtsp_st->transport_priv = NULL;
597 if (rtsp_st->rtp_handle)
598 ffurl_close(rtsp_st->rtp_handle);
599 rtsp_st->rtp_handle = NULL;
603 /* close and free RTSP streams */
604 void ff_rtsp_close_streams(AVFormatContext *s)
606 RTSPState *rt = s->priv_data;
610 ff_rtsp_undo_setup(s);
611 for (i = 0; i < rt->nb_rtsp_streams; i++) {
612 rtsp_st = rt->rtsp_streams[i];
614 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
615 rtsp_st->dynamic_handler->free(
616 rtsp_st->dynamic_protocol_context);
620 av_free(rt->rtsp_streams);
622 avformat_close_input(&rt->asf_ctx);
624 if (rt->ts && CONFIG_RTPDEC)
625 ff_mpegts_parse_close(rt->ts);
627 av_free(rt->recvbuf);
630 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
632 RTSPState *rt = s->priv_data;
634 int reordering_queue_size = rt->reordering_queue_size;
635 if (reordering_queue_size < 0) {
636 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
637 reordering_queue_size = 0;
639 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
642 /* open the RTP context */
643 if (rtsp_st->stream_index >= 0)
644 st = s->streams[rtsp_st->stream_index];
646 s->ctx_flags |= AVFMTCTX_NOHEADER;
648 if (s->oformat && CONFIG_RTSP_MUXER) {
649 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
651 RTSP_TCP_MAX_PACKET_SIZE,
652 rtsp_st->stream_index);
653 /* Ownership of rtp_handle is passed to the rtp mux context */
654 rtsp_st->rtp_handle = NULL;
657 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
658 return 0; // Don't need to open any parser here
659 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
660 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
661 rtsp_st->dynamic_protocol_context,
662 rtsp_st->dynamic_handler);
663 else if (CONFIG_RTPDEC)
664 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
665 rtsp_st->sdp_payload_type,
666 reordering_queue_size);
668 if (!rtsp_st->transport_priv) {
669 return AVERROR(ENOMEM);
670 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
671 if (rtsp_st->dynamic_handler) {
672 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
673 rtsp_st->dynamic_protocol_context,
674 rtsp_st->dynamic_handler);
676 if (rtsp_st->crypto_suite[0])
677 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
678 rtsp_st->crypto_suite,
679 rtsp_st->crypto_params);
685 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
686 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
693 q += strspn(q, SPACE_CHARS);
694 v = strtol(q, &p, 10);
698 v = strtol(p, &p, 10);
707 /* XXX: only one transport specification is parsed */
708 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
710 char transport_protocol[16];
712 char lower_transport[16];
714 RTSPTransportField *th;
717 reply->nb_transports = 0;
720 p += strspn(p, SPACE_CHARS);
724 th = &reply->transports[reply->nb_transports];
726 get_word_sep(transport_protocol, sizeof(transport_protocol),
728 if (!av_strcasecmp (transport_protocol, "rtp")) {
729 get_word_sep(profile, sizeof(profile), "/;,", &p);
730 lower_transport[0] = '\0';
731 /* rtp/avp/<protocol> */
733 get_word_sep(lower_transport, sizeof(lower_transport),
736 th->transport = RTSP_TRANSPORT_RTP;
737 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
738 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
739 /* x-pn-tng/<protocol> */
740 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
742 th->transport = RTSP_TRANSPORT_RDT;
743 } else if (!av_strcasecmp(transport_protocol, "raw")) {
744 get_word_sep(profile, sizeof(profile), "/;,", &p);
745 lower_transport[0] = '\0';
746 /* raw/raw/<protocol> */
748 get_word_sep(lower_transport, sizeof(lower_transport),
751 th->transport = RTSP_TRANSPORT_RAW;
753 if (!av_strcasecmp(lower_transport, "TCP"))
754 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
756 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
760 /* get each parameter */
761 while (*p != '\0' && *p != ',') {
762 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
763 if (!strcmp(parameter, "port")) {
766 rtsp_parse_range(&th->port_min, &th->port_max, &p);
768 } else if (!strcmp(parameter, "client_port")) {
771 rtsp_parse_range(&th->client_port_min,
772 &th->client_port_max, &p);
774 } else if (!strcmp(parameter, "server_port")) {
777 rtsp_parse_range(&th->server_port_min,
778 &th->server_port_max, &p);
780 } else if (!strcmp(parameter, "interleaved")) {
783 rtsp_parse_range(&th->interleaved_min,
784 &th->interleaved_max, &p);
786 } else if (!strcmp(parameter, "multicast")) {
787 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
788 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
789 } else if (!strcmp(parameter, "ttl")) {
793 th->ttl = strtol(p, &end, 10);
796 } else if (!strcmp(parameter, "destination")) {
799 get_word_sep(buf, sizeof(buf), ";,", &p);
800 get_sockaddr(buf, &th->destination);
802 } else if (!strcmp(parameter, "source")) {
805 get_word_sep(buf, sizeof(buf), ";,", &p);
806 av_strlcpy(th->source, buf, sizeof(th->source));
808 } else if (!strcmp(parameter, "mode")) {
811 get_word_sep(buf, sizeof(buf), ";, ", &p);
812 if (!strcmp(buf, "record") ||
813 !strcmp(buf, "receive"))
818 while (*p != ';' && *p != '\0' && *p != ',')
826 reply->nb_transports++;
830 static void handle_rtp_info(RTSPState *rt, const char *url,
831 uint32_t seq, uint32_t rtptime)
834 if (!rtptime || !url[0])
836 if (rt->transport != RTSP_TRANSPORT_RTP)
838 for (i = 0; i < rt->nb_rtsp_streams; i++) {
839 RTSPStream *rtsp_st = rt->rtsp_streams[i];
840 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
843 if (!strcmp(rtsp_st->control_url, url)) {
844 rtpctx->base_timestamp = rtptime;
850 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
853 char key[20], value[1024], url[1024] = "";
854 uint32_t seq = 0, rtptime = 0;
857 p += strspn(p, SPACE_CHARS);
860 get_word_sep(key, sizeof(key), "=", &p);
864 get_word_sep(value, sizeof(value), ";, ", &p);
866 if (!strcmp(key, "url"))
867 av_strlcpy(url, value, sizeof(url));
868 else if (!strcmp(key, "seq"))
869 seq = strtoul(value, NULL, 10);
870 else if (!strcmp(key, "rtptime"))
871 rtptime = strtoul(value, NULL, 10);
873 handle_rtp_info(rt, url, seq, rtptime);
882 handle_rtp_info(rt, url, seq, rtptime);
885 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
886 RTSPState *rt, const char *method)
890 /* NOTE: we do case independent match for broken servers */
892 if (av_stristart(p, "Session:", &p)) {
894 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
895 if (av_stristart(p, ";timeout=", &p) &&
896 (t = strtol(p, NULL, 10)) > 0) {
899 } else if (av_stristart(p, "Content-Length:", &p)) {
900 reply->content_length = strtol(p, NULL, 10);
901 } else if (av_stristart(p, "Transport:", &p)) {
902 rtsp_parse_transport(reply, p);
903 } else if (av_stristart(p, "CSeq:", &p)) {
904 reply->seq = strtol(p, NULL, 10);
905 } else if (av_stristart(p, "Range:", &p)) {
906 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
907 } else if (av_stristart(p, "RealChallenge1:", &p)) {
908 p += strspn(p, SPACE_CHARS);
909 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
910 } else if (av_stristart(p, "Server:", &p)) {
911 p += strspn(p, SPACE_CHARS);
912 av_strlcpy(reply->server, p, sizeof(reply->server));
913 } else if (av_stristart(p, "Notice:", &p) ||
914 av_stristart(p, "X-Notice:", &p)) {
915 reply->notice = strtol(p, NULL, 10);
916 } else if (av_stristart(p, "Location:", &p)) {
917 p += strspn(p, SPACE_CHARS);
918 av_strlcpy(reply->location, p , sizeof(reply->location));
919 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
920 p += strspn(p, SPACE_CHARS);
921 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
922 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
923 p += strspn(p, SPACE_CHARS);
924 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
925 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
926 p += strspn(p, SPACE_CHARS);
927 if (method && !strcmp(method, "DESCRIBE"))
928 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
929 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
930 p += strspn(p, SPACE_CHARS);
931 if (method && !strcmp(method, "PLAY"))
932 rtsp_parse_rtp_info(rt, p);
933 } else if (av_stristart(p, "Public:", &p) && rt) {
934 if (strstr(p, "GET_PARAMETER") &&
935 method && !strcmp(method, "OPTIONS"))
936 rt->get_parameter_supported = 1;
937 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
938 p += strspn(p, SPACE_CHARS);
939 rt->accept_dynamic_rate = atoi(p);
940 } else if (av_stristart(p, "Content-Type:", &p)) {
941 p += strspn(p, SPACE_CHARS);
942 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
946 /* skip a RTP/TCP interleaved packet */
947 void ff_rtsp_skip_packet(AVFormatContext *s)
949 RTSPState *rt = s->priv_data;
953 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
956 len = AV_RB16(buf + 1);
958 av_dlog(s, "skipping RTP packet len=%d\n", len);
963 if (len1 > sizeof(buf))
965 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
972 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
973 unsigned char **content_ptr,
974 int return_on_interleaved_data, const char *method)
976 RTSPState *rt = s->priv_data;
977 char buf[4096], buf1[1024], *q;
980 int ret, content_length, line_count = 0, request = 0;
981 unsigned char *content = NULL;
987 memset(reply, 0, sizeof(*reply));
989 /* parse reply (XXX: use buffers) */
990 rt->last_reply[0] = '\0';
994 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
995 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1001 /* XXX: only parse it if first char on line ? */
1002 if (return_on_interleaved_data) {
1005 ff_rtsp_skip_packet(s);
1006 } else if (ch != '\r') {
1007 if ((q - buf) < sizeof(buf) - 1)
1013 av_dlog(s, "line='%s'\n", buf);
1015 /* test if last line */
1019 if (line_count == 0) {
1020 /* get reply code */
1021 get_word(buf1, sizeof(buf1), &p);
1022 if (!strncmp(buf1, "RTSP/", 5)) {
1023 get_word(buf1, sizeof(buf1), &p);
1024 reply->status_code = atoi(buf1);
1025 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1027 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1028 get_word(buf1, sizeof(buf1), &p); // object
1032 ff_rtsp_parse_line(reply, p, rt, method);
1033 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1034 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1039 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1040 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1042 content_length = reply->content_length;
1043 if (content_length > 0) {
1044 /* leave some room for a trailing '\0' (useful for simple parsing) */
1045 content = av_malloc(content_length + 1);
1046 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1047 content[content_length] = '\0';
1050 *content_ptr = content;
1056 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1057 const char* ptr = buf;
1059 if (!strcmp(reply->reason, "OPTIONS")) {
1060 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1062 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1063 if (reply->session_id[0])
1064 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1067 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1069 av_strlcat(buf, "\r\n", sizeof(buf));
1071 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1072 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1075 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1077 rt->last_cmd_time = av_gettime();
1078 /* Even if the request from the server had data, it is not the data
1079 * that the caller wants or expects. The memory could also be leaked
1080 * if the actual following reply has content data. */
1082 av_freep(content_ptr);
1083 /* If method is set, this is called from ff_rtsp_send_cmd,
1084 * where a reply to exactly this request is awaited. For
1085 * callers from within packet receiving, we just want to
1086 * return to the caller and go back to receiving packets. */
1092 if (rt->seq != reply->seq) {
1093 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1094 rt->seq, reply->seq);
1098 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1099 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1100 reply->notice == 2306 /* Continuous Feed Terminated */) {
1101 rt->state = RTSP_STATE_IDLE;
1102 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1103 return AVERROR(EIO); /* data or server error */
1104 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1105 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1106 return AVERROR(EPERM);
1112 * Send a command to the RTSP server without waiting for the reply.
1114 * @param s RTSP (de)muxer context
1115 * @param method the method for the request
1116 * @param url the target url for the request
1117 * @param headers extra header lines to include in the request
1118 * @param send_content if non-null, the data to send as request body content
1119 * @param send_content_length the length of the send_content data, or 0 if
1120 * send_content is null
1122 * @return zero if success, nonzero otherwise
1124 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1125 const char *method, const char *url,
1126 const char *headers,
1127 const unsigned char *send_content,
1128 int send_content_length)
1130 RTSPState *rt = s->priv_data;
1131 char buf[4096], *out_buf;
1132 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1134 /* Add in RTSP headers */
1137 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1139 av_strlcat(buf, headers, sizeof(buf));
1140 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1141 if (rt->session_id[0] != '\0' && (!headers ||
1142 !strstr(headers, "\nIf-Match:"))) {
1143 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1146 char *str = ff_http_auth_create_response(&rt->auth_state,
1147 rt->auth, url, method);
1149 av_strlcat(buf, str, sizeof(buf));
1152 if (send_content_length > 0 && send_content)
1153 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1154 av_strlcat(buf, "\r\n", sizeof(buf));
1156 /* base64 encode rtsp if tunneling */
1157 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1158 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1159 out_buf = base64buf;
1162 av_dlog(s, "Sending:\n%s--\n", buf);
1164 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1165 if (send_content_length > 0 && send_content) {
1166 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1167 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1168 "with content data not supported\n");
1169 return AVERROR_PATCHWELCOME;
1171 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1173 rt->last_cmd_time = av_gettime();
1178 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1179 const char *url, const char *headers)
1181 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1184 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1185 const char *headers, RTSPMessageHeader *reply,
1186 unsigned char **content_ptr)
1188 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1189 content_ptr, NULL, 0);
1192 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1193 const char *method, const char *url,
1195 RTSPMessageHeader *reply,
1196 unsigned char **content_ptr,
1197 const unsigned char *send_content,
1198 int send_content_length)
1200 RTSPState *rt = s->priv_data;
1201 HTTPAuthType cur_auth_type;
1202 int ret, attempts = 0;
1205 cur_auth_type = rt->auth_state.auth_type;
1206 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1208 send_content_length)))
1211 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1215 if (reply->status_code == 401 &&
1216 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1217 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1220 if (reply->status_code > 400){
1221 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1225 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1231 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1232 int lower_transport, const char *real_challenge)
1234 RTSPState *rt = s->priv_data;
1235 int rtx = 0, j, i, err, interleave = 0, port_off;
1236 RTSPStream *rtsp_st;
1237 RTSPMessageHeader reply1, *reply = &reply1;
1239 const char *trans_pref;
1241 if (rt->transport == RTSP_TRANSPORT_RDT)
1242 trans_pref = "x-pn-tng";
1243 else if (rt->transport == RTSP_TRANSPORT_RAW)
1244 trans_pref = "RAW/RAW";
1246 trans_pref = "RTP/AVP";
1248 /* default timeout: 1 minute */
1251 /* Choose a random starting offset within the first half of the
1252 * port range, to allow for a number of ports to try even if the offset
1253 * happens to be at the end of the random range. */
1254 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1255 /* even random offset */
1256 port_off -= port_off & 0x01;
1258 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1259 char transport[2048];
1262 * WMS serves all UDP data over a single connection, the RTX, which
1263 * isn't necessarily the first in the SDP but has to be the first
1264 * to be set up, else the second/third SETUP will fail with a 461.
1266 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1267 rt->server_type == RTSP_SERVER_WMS) {
1270 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1271 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1273 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1277 if (rtx == rt->nb_rtsp_streams)
1278 return -1; /* no RTX found */
1279 rtsp_st = rt->rtsp_streams[rtx];
1281 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1283 rtsp_st = rt->rtsp_streams[i];
1286 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1289 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1290 port = reply->transports[0].client_port_min;
1294 /* first try in specified port range */
1295 while (j <= rt->rtp_port_max) {
1296 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1297 "?localport=%d", j);
1298 /* we will use two ports per rtp stream (rtp and rtcp) */
1300 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1301 &s->interrupt_callback, NULL))
1304 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1309 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1311 snprintf(transport, sizeof(transport) - 1,
1312 "%s/UDP;", trans_pref);
1313 if (rt->server_type != RTSP_SERVER_REAL)
1314 av_strlcat(transport, "unicast;", sizeof(transport));
1315 av_strlcatf(transport, sizeof(transport),
1316 "client_port=%d", port);
1317 if (rt->transport == RTSP_TRANSPORT_RTP &&
1318 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1319 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1323 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1324 /* For WMS streams, the application streams are only used for
1325 * UDP. When trying to set it up for TCP streams, the server
1326 * will return an error. Therefore, we skip those streams. */
1327 if (rt->server_type == RTSP_SERVER_WMS &&
1328 (rtsp_st->stream_index < 0 ||
1329 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1332 snprintf(transport, sizeof(transport) - 1,
1333 "%s/TCP;", trans_pref);
1334 if (rt->transport != RTSP_TRANSPORT_RDT)
1335 av_strlcat(transport, "unicast;", sizeof(transport));
1336 av_strlcatf(transport, sizeof(transport),
1337 "interleaved=%d-%d",
1338 interleave, interleave + 1);
1342 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1343 snprintf(transport, sizeof(transport) - 1,
1344 "%s/UDP;multicast", trans_pref);
1347 av_strlcat(transport, ";mode=record", sizeof(transport));
1348 } else if (rt->server_type == RTSP_SERVER_REAL ||
1349 rt->server_type == RTSP_SERVER_WMS)
1350 av_strlcat(transport, ";mode=play", sizeof(transport));
1351 snprintf(cmd, sizeof(cmd),
1352 "Transport: %s\r\n",
1354 if (rt->accept_dynamic_rate)
1355 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1356 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1357 char real_res[41], real_csum[9];
1358 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1360 av_strlcatf(cmd, sizeof(cmd),
1362 "RealChallenge2: %s, sd=%s\r\n",
1363 rt->session_id, real_res, real_csum);
1365 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1366 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1369 } else if (reply->status_code != RTSP_STATUS_OK ||
1370 reply->nb_transports != 1) {
1371 err = AVERROR_INVALIDDATA;
1375 /* XXX: same protocol for all streams is required */
1377 if (reply->transports[0].lower_transport != rt->lower_transport ||
1378 reply->transports[0].transport != rt->transport) {
1379 err = AVERROR_INVALIDDATA;
1383 rt->lower_transport = reply->transports[0].lower_transport;
1384 rt->transport = reply->transports[0].transport;
1387 /* Fail if the server responded with another lower transport mode
1388 * than what we requested. */
1389 if (reply->transports[0].lower_transport != lower_transport) {
1390 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1391 err = AVERROR_INVALIDDATA;
1395 switch(reply->transports[0].lower_transport) {
1396 case RTSP_LOWER_TRANSPORT_TCP:
1397 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1398 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1401 case RTSP_LOWER_TRANSPORT_UDP: {
1402 char url[1024], options[30] = "";
1404 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1405 av_strlcpy(options, "?connect=1", sizeof(options));
1406 /* Use source address if specified */
1407 if (reply->transports[0].source[0]) {
1408 ff_url_join(url, sizeof(url), "rtp", NULL,
1409 reply->transports[0].source,
1410 reply->transports[0].server_port_min, "%s", options);
1412 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1413 reply->transports[0].server_port_min, "%s", options);
1415 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1416 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1417 err = AVERROR_INVALIDDATA;
1420 /* Try to initialize the connection state in a
1421 * potential NAT router by sending dummy packets.
1422 * RTP/RTCP dummy packets are used for RDT, too.
1424 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1426 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1429 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1430 char url[1024], namebuf[50], optbuf[20] = "";
1431 struct sockaddr_storage addr;
1434 if (reply->transports[0].destination.ss_family) {
1435 addr = reply->transports[0].destination;
1436 port = reply->transports[0].port_min;
1437 ttl = reply->transports[0].ttl;
1439 addr = rtsp_st->sdp_ip;
1440 port = rtsp_st->sdp_port;
1441 ttl = rtsp_st->sdp_ttl;
1444 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1445 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1446 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1447 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1448 port, "%s", optbuf);
1449 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1450 &s->interrupt_callback, NULL) < 0) {
1451 err = AVERROR_INVALIDDATA;
1458 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1462 if (rt->nb_rtsp_streams && reply->timeout > 0)
1463 rt->timeout = reply->timeout;
1465 if (rt->server_type == RTSP_SERVER_REAL)
1466 rt->need_subscription = 1;
1471 ff_rtsp_undo_setup(s);
1475 void ff_rtsp_close_connections(AVFormatContext *s)
1477 RTSPState *rt = s->priv_data;
1478 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1479 ffurl_close(rt->rtsp_hd);
1480 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1483 int ff_rtsp_connect(AVFormatContext *s)
1485 RTSPState *rt = s->priv_data;
1486 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1487 int port, err, tcp_fd;
1488 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1489 int lower_transport_mask = 0;
1490 char real_challenge[64] = "";
1491 struct sockaddr_storage peer;
1492 socklen_t peer_len = sizeof(peer);
1494 if (rt->rtp_port_max < rt->rtp_port_min) {
1495 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1496 "than min port %d\n", rt->rtp_port_max,
1498 return AVERROR(EINVAL);
1501 if (!ff_network_init())
1502 return AVERROR(EIO);
1504 if (s->max_delay < 0) /* Not set by the caller */
1505 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1507 rt->control_transport = RTSP_MODE_PLAIN;
1508 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1509 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1510 rt->control_transport = RTSP_MODE_TUNNEL;
1512 /* Only pass through valid flags from here */
1513 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1516 lower_transport_mask = rt->lower_transport_mask;
1517 /* extract hostname and port */
1518 av_url_split(NULL, 0, auth, sizeof(auth),
1519 host, sizeof(host), &port, path, sizeof(path), s->filename);
1521 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1524 port = RTSP_DEFAULT_PORT;
1526 if (!lower_transport_mask)
1527 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1530 /* Only UDP or TCP - UDP multicast isn't supported. */
1531 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1532 (1 << RTSP_LOWER_TRANSPORT_TCP);
1533 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1534 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1535 "only UDP and TCP are supported for output.\n");
1536 err = AVERROR(EINVAL);
1541 /* Construct the URI used in request; this is similar to s->filename,
1542 * but with authentication credentials removed and RTSP specific options
1544 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1545 host, port, "%s", path);
1547 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1548 /* set up initial handshake for tunneling */
1549 char httpname[1024];
1550 char sessioncookie[17];
1553 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1554 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1555 av_get_random_seed(), av_get_random_seed());
1558 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1559 &s->interrupt_callback) < 0) {
1564 /* generate GET headers */
1565 snprintf(headers, sizeof(headers),
1566 "x-sessioncookie: %s\r\n"
1567 "Accept: application/x-rtsp-tunnelled\r\n"
1568 "Pragma: no-cache\r\n"
1569 "Cache-Control: no-cache\r\n",
1571 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1573 /* complete the connection */
1574 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1580 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1581 &s->interrupt_callback) < 0 ) {
1586 /* generate POST headers */
1587 snprintf(headers, sizeof(headers),
1588 "x-sessioncookie: %s\r\n"
1589 "Content-Type: application/x-rtsp-tunnelled\r\n"
1590 "Pragma: no-cache\r\n"
1591 "Cache-Control: no-cache\r\n"
1592 "Content-Length: 32767\r\n"
1593 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1595 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1596 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1598 /* Initialize the authentication state for the POST session. The HTTP
1599 * protocol implementation doesn't properly handle multi-pass
1600 * authentication for POST requests, since it would require one of
1602 * - implementing Expect: 100-continue, which many HTTP servers
1603 * don't support anyway, even less the RTSP servers that do HTTP
1605 * - sending the whole POST data until getting a 401 reply specifying
1606 * what authentication method to use, then resending all that data
1607 * - waiting for potential 401 replies directly after sending the
1608 * POST header (waiting for some unspecified time)
1609 * Therefore, we copy the full auth state, which works for both basic
1610 * and digest. (For digest, we would have to synchronize the nonce
1611 * count variable between the two sessions, if we'd do more requests
1612 * with the original session, though.)
1614 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1616 /* complete the connection */
1617 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1622 /* open the tcp connection */
1623 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1624 "?timeout=%d", rt->stimeout);
1625 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1626 &s->interrupt_callback, NULL) < 0) {
1630 rt->rtsp_hd_out = rt->rtsp_hd;
1634 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1635 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1636 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1637 NULL, 0, NI_NUMERICHOST);
1640 /* request options supported by the server; this also detects server
1642 for (rt->server_type = RTSP_SERVER_RTP;;) {
1644 if (rt->server_type == RTSP_SERVER_REAL)
1647 * The following entries are required for proper
1648 * streaming from a Realmedia server. They are
1649 * interdependent in some way although we currently
1650 * don't quite understand how. Values were copied
1651 * from mplayer SVN r23589.
1652 * ClientChallenge is a 16-byte ID in hex
1653 * CompanyID is a 16-byte ID in base64
1655 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1656 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1657 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1658 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1660 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1661 if (reply->status_code != RTSP_STATUS_OK) {
1662 err = AVERROR_INVALIDDATA;
1666 /* detect server type if not standard-compliant RTP */
1667 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1668 rt->server_type = RTSP_SERVER_REAL;
1670 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1671 rt->server_type = RTSP_SERVER_WMS;
1672 } else if (rt->server_type == RTSP_SERVER_REAL)
1673 strcpy(real_challenge, reply->real_challenge);
1677 if (s->iformat && CONFIG_RTSP_DEMUXER)
1678 err = ff_rtsp_setup_input_streams(s, reply);
1679 else if (CONFIG_RTSP_MUXER)
1680 err = ff_rtsp_setup_output_streams(s, host);
1685 int lower_transport = ff_log2_tab[lower_transport_mask &
1686 ~(lower_transport_mask - 1)];
1688 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1689 rt->server_type == RTSP_SERVER_REAL ?
1690 real_challenge : NULL);
1693 lower_transport_mask &= ~(1 << lower_transport);
1694 if (lower_transport_mask == 0 && err == 1) {
1695 err = AVERROR(EPROTONOSUPPORT);
1700 rt->lower_transport_mask = lower_transport_mask;
1701 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1702 rt->state = RTSP_STATE_IDLE;
1703 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1706 ff_rtsp_close_streams(s);
1707 ff_rtsp_close_connections(s);
1708 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1709 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1710 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1718 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1721 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1722 uint8_t *buf, int buf_size, int64_t wait_end)
1724 RTSPState *rt = s->priv_data;
1725 RTSPStream *rtsp_st;
1726 int n, i, ret, tcp_fd, timeout_cnt = 0;
1728 struct pollfd *p = rt->p;
1729 int *fds = NULL, fdsnum, fdsidx;
1732 if (ff_check_interrupt(&s->interrupt_callback))
1733 return AVERROR_EXIT;
1734 if (wait_end && wait_end - av_gettime() < 0)
1735 return AVERROR(EAGAIN);
1738 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1739 p[max_p].fd = tcp_fd;
1740 p[max_p++].events = POLLIN;
1744 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1745 rtsp_st = rt->rtsp_streams[i];
1746 if (rtsp_st->rtp_handle) {
1747 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1749 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1753 av_log(s, AV_LOG_ERROR,
1754 "Number of fds %d not supported\n", fdsnum);
1755 return AVERROR_INVALIDDATA;
1757 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1758 p[max_p].fd = fds[fdsidx];
1759 p[max_p++].events = POLLIN;
1764 n = poll(p, max_p, POLL_TIMEOUT_MS);
1766 int j = 1 - (tcp_fd == -1);
1768 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1769 rtsp_st = rt->rtsp_streams[i];
1770 if (rtsp_st->rtp_handle) {
1771 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1772 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1774 *prtsp_st = rtsp_st;
1781 #if CONFIG_RTSP_DEMUXER
1782 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1783 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1784 if (rt->state == RTSP_STATE_STREAMING) {
1785 if (!ff_rtsp_parse_streaming_commands(s))
1788 av_log(s, AV_LOG_WARNING,
1789 "Unable to answer to TEARDOWN\n");
1793 RTSPMessageHeader reply;
1794 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1797 /* XXX: parse message */
1798 if (rt->state != RTSP_STATE_STREAMING)
1803 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1804 return AVERROR(ETIMEDOUT);
1805 } else if (n < 0 && errno != EINTR)
1806 return AVERROR(errno);
1810 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1811 const uint8_t *buf, int len)
1813 RTSPState *rt = s->priv_data;
1817 if (rt->nb_rtsp_streams == 1) {
1818 *rtsp_st = rt->rtsp_streams[0];
1821 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1822 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1824 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1825 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1828 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1829 *rtsp_st = rt->rtsp_streams[i];
1836 av_log(s, AV_LOG_WARNING,
1837 "Unable to pick stream for packet - SSRC not known for "
1839 return AVERROR(EAGAIN);
1842 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1843 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1844 *rtsp_st = rt->rtsp_streams[i];
1850 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1851 return AVERROR(EAGAIN);
1854 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1856 RTSPState *rt = s->priv_data;
1858 RTSPStream *rtsp_st, *first_queue_st = NULL;
1859 int64_t wait_end = 0;
1861 if (rt->nb_byes == rt->nb_rtsp_streams)
1864 /* get next frames from the same RTP packet */
1865 if (rt->cur_transport_priv) {
1866 if (rt->transport == RTSP_TRANSPORT_RDT) {
1867 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1868 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1869 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1870 } else if (rt->ts && CONFIG_RTPDEC) {
1871 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1873 rt->recvbuf_pos += ret;
1874 ret = rt->recvbuf_pos < rt->recvbuf_len;
1879 rt->cur_transport_priv = NULL;
1881 } else if (ret == 1) {
1884 rt->cur_transport_priv = NULL;
1888 if (rt->transport == RTSP_TRANSPORT_RTP) {
1890 int64_t first_queue_time = 0;
1891 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1892 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1896 queue_time = ff_rtp_queued_packet_time(rtpctx);
1897 if (queue_time && (queue_time - first_queue_time < 0 ||
1898 !first_queue_time)) {
1899 first_queue_time = queue_time;
1900 first_queue_st = rt->rtsp_streams[i];
1903 if (first_queue_time) {
1904 wait_end = first_queue_time + s->max_delay;
1907 first_queue_st = NULL;
1911 /* read next RTP packet */
1913 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1915 return AVERROR(ENOMEM);
1918 switch(rt->lower_transport) {
1920 #if CONFIG_RTSP_DEMUXER
1921 case RTSP_LOWER_TRANSPORT_TCP:
1922 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1925 case RTSP_LOWER_TRANSPORT_UDP:
1926 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1927 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1928 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1929 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1931 case RTSP_LOWER_TRANSPORT_CUSTOM:
1932 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1933 wait_end && wait_end < av_gettime())
1934 len = AVERROR(EAGAIN);
1936 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1937 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1938 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1939 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1942 if (len == AVERROR(EAGAIN) && first_queue_st &&
1943 rt->transport == RTSP_TRANSPORT_RTP) {
1944 rtsp_st = first_queue_st;
1945 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1952 if (rt->transport == RTSP_TRANSPORT_RDT) {
1953 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1954 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1955 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1956 if (rtsp_st->feedback) {
1957 AVIOContext *pb = NULL;
1958 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1960 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1963 /* Either bad packet, or a RTCP packet. Check if the
1964 * first_rtcp_ntp_time field was initialized. */
1965 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1966 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1967 /* first_rtcp_ntp_time has been initialized for this stream,
1968 * copy the same value to all other uninitialized streams,
1969 * in order to map their timestamp origin to the same ntp time
1972 AVStream *st = NULL;
1973 if (rtsp_st->stream_index >= 0)
1974 st = s->streams[rtsp_st->stream_index];
1975 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1976 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1977 AVStream *st2 = NULL;
1978 if (rt->rtsp_streams[i]->stream_index >= 0)
1979 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1980 if (rtpctx2 && st && st2 &&
1981 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1982 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1983 rtpctx2->rtcp_ts_offset = av_rescale_q(
1984 rtpctx->rtcp_ts_offset, st->time_base,
1989 if (ret == -RTCP_BYE) {
1992 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1993 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1995 if (rt->nb_byes == rt->nb_rtsp_streams)
1999 } else if (rt->ts && CONFIG_RTPDEC) {
2000 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2003 rt->recvbuf_len = len;
2004 rt->recvbuf_pos = ret;
2005 rt->cur_transport_priv = rt->ts;
2012 return AVERROR_INVALIDDATA;
2018 /* more packets may follow, so we save the RTP context */
2019 rt->cur_transport_priv = rtsp_st->transport_priv;
2023 #endif /* CONFIG_RTPDEC */
2025 #if CONFIG_SDP_DEMUXER
2026 static int sdp_probe(AVProbeData *p1)
2028 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2030 /* we look for a line beginning "c=IN IP" */
2031 while (p < p_end && *p != '\0') {
2032 if (p + sizeof("c=IN IP") - 1 < p_end &&
2033 av_strstart(p, "c=IN IP", NULL))
2034 return AVPROBE_SCORE_EXTENSION;
2036 while (p < p_end - 1 && *p != '\n') p++;
2045 static int sdp_read_header(AVFormatContext *s)
2047 RTSPState *rt = s->priv_data;
2048 RTSPStream *rtsp_st;
2053 if (!ff_network_init())
2054 return AVERROR(EIO);
2056 if (s->max_delay < 0) /* Not set by the caller */
2057 s->max_delay = DEFAULT_REORDERING_DELAY;
2058 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2059 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2061 /* read the whole sdp file */
2062 /* XXX: better loading */
2063 content = av_malloc(SDP_MAX_SIZE);
2064 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2067 return AVERROR_INVALIDDATA;
2069 content[size] ='\0';
2071 err = ff_sdp_parse(s, content);
2075 /* open each RTP stream */
2076 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2078 rtsp_st = rt->rtsp_streams[i];
2080 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2081 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2082 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2083 ff_url_join(url, sizeof(url), "rtp", NULL,
2084 namebuf, rtsp_st->sdp_port,
2085 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2087 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2088 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2089 &s->interrupt_callback, NULL) < 0) {
2090 err = AVERROR_INVALIDDATA;
2094 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2099 ff_rtsp_close_streams(s);
2104 static int sdp_read_close(AVFormatContext *s)
2106 ff_rtsp_close_streams(s);
2111 static const AVClass sdp_demuxer_class = {
2112 .class_name = "SDP demuxer",
2113 .item_name = av_default_item_name,
2114 .option = sdp_options,
2115 .version = LIBAVUTIL_VERSION_INT,
2118 AVInputFormat ff_sdp_demuxer = {
2120 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2121 .priv_data_size = sizeof(RTSPState),
2122 .read_probe = sdp_probe,
2123 .read_header = sdp_read_header,
2124 .read_packet = ff_rtsp_fetch_packet,
2125 .read_close = sdp_read_close,
2126 .priv_class = &sdp_demuxer_class,
2128 #endif /* CONFIG_SDP_DEMUXER */
2130 #if CONFIG_RTP_DEMUXER
2131 static int rtp_probe(AVProbeData *p)
2133 if (av_strstart(p->filename, "rtp:", NULL))
2134 return AVPROBE_SCORE_MAX;
2138 static int rtp_read_header(AVFormatContext *s)
2140 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2141 char host[500], sdp[500];
2143 URLContext* in = NULL;
2145 AVCodecContext codec = { 0 };
2146 struct sockaddr_storage addr;
2148 socklen_t addrlen = sizeof(addr);
2149 RTSPState *rt = s->priv_data;
2151 if (!ff_network_init())
2152 return AVERROR(EIO);
2154 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2155 &s->interrupt_callback, NULL);
2160 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2161 if (ret == AVERROR(EAGAIN))
2166 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2170 if ((recvbuf[0] & 0xc0) != 0x80) {
2171 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2176 if (RTP_PT_IS_RTCP(recvbuf[1]))
2179 payload_type = recvbuf[1] & 0x7f;
2182 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2186 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2187 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2188 "without an SDP file describing it\n",
2192 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2193 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2194 "properly you need an SDP file "
2198 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2199 NULL, 0, s->filename);
2201 snprintf(sdp, sizeof(sdp),
2202 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2203 addr.ss_family == AF_INET ? 4 : 6, host,
2204 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2205 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2206 port, payload_type);
2207 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2209 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2212 /* sdp_read_header initializes this again */
2215 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2217 ret = sdp_read_header(s);
2228 static const AVClass rtp_demuxer_class = {
2229 .class_name = "RTP demuxer",
2230 .item_name = av_default_item_name,
2231 .option = rtp_options,
2232 .version = LIBAVUTIL_VERSION_INT,
2235 AVInputFormat ff_rtp_demuxer = {
2237 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2238 .priv_data_size = sizeof(RTSPState),
2239 .read_probe = rtp_probe,
2240 .read_header = rtp_read_header,
2241 .read_packet = ff_rtsp_fetch_packet,
2242 .read_close = sdp_read_close,
2243 .flags = AVFMT_NOFILE,
2244 .priv_class = &rtp_demuxer_class,
2246 #endif /* CONFIG_RTP_DEMUXER */