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"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
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 RTSP_REORDERING_OPTS(),
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102 RTSP_REORDERING_OPTS(),
106 static const AVOption rtp_options[] = {
107 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108 RTSP_REORDERING_OPTS(),
112 static void get_word_until_chars(char *buf, int buf_size,
113 const char *sep, const char **pp)
119 p += strspn(p, SPACE_CHARS);
121 while (!strchr(sep, *p) && *p != '\0') {
122 if ((q - buf) < buf_size - 1)
131 static void get_word_sep(char *buf, int buf_size, const char *sep,
134 if (**pp == '/') (*pp)++;
135 get_word_until_chars(buf, buf_size, sep, pp);
138 static void get_word(char *buf, int buf_size, const char **pp)
140 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
143 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
145 * Used for seeking in the rtp stream.
147 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
151 p += strspn(p, SPACE_CHARS);
152 if (!av_stristart(p, "npt=", &p))
155 *start = AV_NOPTS_VALUE;
156 *end = AV_NOPTS_VALUE;
158 get_word_sep(buf, sizeof(buf), "-", &p);
159 av_parse_time(start, buf, 1);
162 get_word_sep(buf, sizeof(buf), "-", &p);
163 av_parse_time(end, buf, 1);
167 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
169 struct addrinfo hints = { 0 }, *ai = NULL;
170 hints.ai_flags = AI_NUMERICHOST;
171 if (getaddrinfo(buf, NULL, &hints, &ai))
173 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
179 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
180 RTSPStream *rtsp_st, AVCodecContext *codec)
184 codec->codec_id = handler->codec_id;
185 rtsp_st->dynamic_handler = handler;
186 if (handler->alloc) {
187 rtsp_st->dynamic_protocol_context = handler->alloc();
188 if (!rtsp_st->dynamic_protocol_context)
189 rtsp_st->dynamic_handler = NULL;
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195 AVStream *st, RTSPStream *rtsp_st,
196 int payload_type, const char *p)
198 AVCodecContext *codec = st->codec;
204 /* See if we can handle this kind of payload.
205 * The space should normally not be there but some Real streams or
206 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207 * have a trailing space. */
208 get_word_sep(buf, sizeof(buf), "/ ", &p);
209 if (payload_type < RTP_PT_PRIVATE) {
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
215 if (codec->codec_id == AV_CODEC_ID_NONE) {
216 RTPDynamicProtocolHandler *handler =
217 ff_rtp_handler_find_by_name(buf, codec->codec_type);
218 init_rtp_handler(handler, rtsp_st, codec);
219 /* If no dynamic handler was found, check with the list of standard
220 * allocated types, if such a stream for some reason happens to
221 * use a private payload type. This isn't handled in rtpdec.c, since
222 * the format name from the rtpmap line never is passed into rtpdec. */
223 if (!rtsp_st->dynamic_handler)
224 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
227 c = avcodec_find_decoder(codec->codec_id);
233 get_word_sep(buf, sizeof(buf), "/", &p);
235 switch (codec->codec_type) {
236 case AVMEDIA_TYPE_AUDIO:
237 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
238 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
239 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
241 codec->sample_rate = i;
242 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243 get_word_sep(buf, sizeof(buf), "/", &p);
248 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
250 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
253 case AVMEDIA_TYPE_VIDEO:
254 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
256 avpriv_set_pts_info(st, 32, 1, i);
261 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262 rtsp_st->dynamic_handler->init(s, st->index,
263 rtsp_st->dynamic_protocol_context);
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271 char *value, int value_size)
273 *p += strspn(*p, SPACE_CHARS);
275 get_word_sep(attr, attr_size, "=", p);
278 get_word_sep(value, value_size, ";", p);
286 typedef struct SDPParseState {
288 struct sockaddr_storage default_ip;
290 int skip_media; ///< set if an unknown m= line occurs
293 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
294 int letter, const char *buf)
296 RTSPState *rt = s->priv_data;
297 char buf1[64], st_type[64];
299 enum AVMediaType codec_type;
303 struct sockaddr_storage sdp_ip;
306 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
309 if (s1->skip_media && letter != 'm')
313 get_word(buf1, sizeof(buf1), &p);
314 if (strcmp(buf1, "IN") != 0)
316 get_word(buf1, sizeof(buf1), &p);
317 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
319 get_word_sep(buf1, sizeof(buf1), "/", &p);
320 if (get_sockaddr(buf1, &sdp_ip))
325 get_word_sep(buf1, sizeof(buf1), "/", &p);
328 if (s->nb_streams == 0) {
329 s1->default_ip = sdp_ip;
330 s1->default_ttl = ttl;
332 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
333 rtsp_st->sdp_ip = sdp_ip;
334 rtsp_st->sdp_ttl = ttl;
338 av_dict_set(&s->metadata, "title", p, 0);
341 if (s->nb_streams == 0) {
342 av_dict_set(&s->metadata, "comment", p, 0);
349 codec_type = AVMEDIA_TYPE_UNKNOWN;
350 get_word(st_type, sizeof(st_type), &p);
351 if (!strcmp(st_type, "audio")) {
352 codec_type = AVMEDIA_TYPE_AUDIO;
353 } else if (!strcmp(st_type, "video")) {
354 codec_type = AVMEDIA_TYPE_VIDEO;
355 } else if (!strcmp(st_type, "application")) {
356 codec_type = AVMEDIA_TYPE_DATA;
358 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
362 rtsp_st = av_mallocz(sizeof(RTSPStream));
365 rtsp_st->stream_index = -1;
366 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
368 rtsp_st->sdp_ip = s1->default_ip;
369 rtsp_st->sdp_ttl = s1->default_ttl;
371 get_word(buf1, sizeof(buf1), &p); /* port */
372 rtsp_st->sdp_port = atoi(buf1);
374 get_word(buf1, sizeof(buf1), &p); /* protocol */
375 if (!strcmp(buf1, "udp"))
376 rt->transport = RTSP_TRANSPORT_RAW;
377 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
378 rtsp_st->feedback = 1;
380 /* XXX: handle list of formats */
381 get_word(buf1, sizeof(buf1), &p); /* format list */
382 rtsp_st->sdp_payload_type = atoi(buf1);
384 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
385 /* no corresponding stream */
386 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
387 rt->ts = ff_mpegts_parse_open(s);
388 } else if (rt->server_type == RTSP_SERVER_WMS &&
389 codec_type == AVMEDIA_TYPE_DATA) {
390 /* RTX stream, a stream that carries all the other actual
391 * audio/video streams. Don't expose this to the callers. */
393 st = avformat_new_stream(s, NULL);
396 st->id = rt->nb_rtsp_streams - 1;
397 rtsp_st->stream_index = st->index;
398 st->codec->codec_type = codec_type;
399 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
400 RTPDynamicProtocolHandler *handler;
401 /* if standard payload type, we can find the codec right now */
402 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
403 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
404 st->codec->sample_rate > 0)
405 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
406 /* Even static payload types may need a custom depacketizer */
407 handler = ff_rtp_handler_find_by_id(
408 rtsp_st->sdp_payload_type, st->codec->codec_type);
409 init_rtp_handler(handler, rtsp_st, st->codec);
410 if (handler && handler->init)
411 handler->init(s, st->index,
412 rtsp_st->dynamic_protocol_context);
415 /* put a default control url */
416 av_strlcpy(rtsp_st->control_url, rt->control_uri,
417 sizeof(rtsp_st->control_url));
420 if (av_strstart(p, "control:", &p)) {
421 if (s->nb_streams == 0) {
422 if (!strncmp(p, "rtsp://", 7))
423 av_strlcpy(rt->control_uri, p,
424 sizeof(rt->control_uri));
427 /* get the control url */
428 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
430 /* XXX: may need to add full url resolution */
431 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
433 if (proto[0] == '\0') {
434 /* relative control URL */
435 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
436 av_strlcat(rtsp_st->control_url, "/",
437 sizeof(rtsp_st->control_url));
438 av_strlcat(rtsp_st->control_url, p,
439 sizeof(rtsp_st->control_url));
441 av_strlcpy(rtsp_st->control_url, p,
442 sizeof(rtsp_st->control_url));
444 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
445 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
446 get_word(buf1, sizeof(buf1), &p);
447 payload_type = atoi(buf1);
448 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
449 if (rtsp_st->stream_index >= 0) {
450 st = s->streams[rtsp_st->stream_index];
451 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
453 } else if (av_strstart(p, "fmtp:", &p) ||
454 av_strstart(p, "framesize:", &p)) {
455 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
456 // let dynamic protocol handlers have a stab at the line.
457 get_word(buf1, sizeof(buf1), &p);
458 payload_type = atoi(buf1);
459 for (i = 0; i < rt->nb_rtsp_streams; i++) {
460 rtsp_st = rt->rtsp_streams[i];
461 if (rtsp_st->sdp_payload_type == payload_type &&
462 rtsp_st->dynamic_handler &&
463 rtsp_st->dynamic_handler->parse_sdp_a_line)
464 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
465 rtsp_st->dynamic_protocol_context, buf);
467 } else if (av_strstart(p, "range:", &p)) {
470 // this is so that seeking on a streamed file can work.
471 rtsp_parse_range_npt(p, &start, &end);
472 s->start_time = start;
473 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
474 s->duration = (end == AV_NOPTS_VALUE) ?
475 AV_NOPTS_VALUE : end - start;
476 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
478 rt->transport = RTSP_TRANSPORT_RDT;
479 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
481 st = s->streams[s->nb_streams - 1];
482 st->codec->sample_rate = atoi(p);
484 if (rt->server_type == RTSP_SERVER_WMS)
485 ff_wms_parse_sdp_a_line(s, p);
486 if (s->nb_streams > 0) {
487 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
489 if (rt->server_type == RTSP_SERVER_REAL)
490 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
492 if (rtsp_st->dynamic_handler &&
493 rtsp_st->dynamic_handler->parse_sdp_a_line)
494 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
495 rtsp_st->stream_index,
496 rtsp_st->dynamic_protocol_context, buf);
503 int ff_sdp_parse(AVFormatContext *s, const char *content)
505 RTSPState *rt = s->priv_data;
508 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
509 * contain long SDP lines containing complete ASF Headers (several
510 * kB) or arrays of MDPR (RM stream descriptor) headers plus
511 * "rulebooks" describing their properties. Therefore, the SDP line
514 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
515 * in rtpdec_xiph.c. */
517 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
521 p += strspn(p, SPACE_CHARS);
529 /* get the content */
531 while (*p != '\n' && *p != '\r' && *p != '\0') {
532 if ((q - buf) < sizeof(buf) - 1)
537 sdp_parse_line(s, s1, letter, buf);
539 while (*p != '\n' && *p != '\0')
544 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
545 if (!rt->p) return AVERROR(ENOMEM);
548 #endif /* CONFIG_RTPDEC */
550 void ff_rtsp_undo_setup(AVFormatContext *s)
552 RTSPState *rt = s->priv_data;
555 for (i = 0; i < rt->nb_rtsp_streams; i++) {
556 RTSPStream *rtsp_st = rt->rtsp_streams[i];
559 if (rtsp_st->transport_priv) {
561 AVFormatContext *rtpctx = rtsp_st->transport_priv;
562 av_write_trailer(rtpctx);
563 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
565 avio_close_dyn_buf(rtpctx->pb, &ptr);
568 avio_close(rtpctx->pb);
570 avformat_free_context(rtpctx);
571 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
572 ff_rdt_parse_close(rtsp_st->transport_priv);
573 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
574 ff_rtp_parse_close(rtsp_st->transport_priv);
576 rtsp_st->transport_priv = NULL;
577 if (rtsp_st->rtp_handle)
578 ffurl_close(rtsp_st->rtp_handle);
579 rtsp_st->rtp_handle = NULL;
583 /* close and free RTSP streams */
584 void ff_rtsp_close_streams(AVFormatContext *s)
586 RTSPState *rt = s->priv_data;
590 ff_rtsp_undo_setup(s);
591 for (i = 0; i < rt->nb_rtsp_streams; i++) {
592 rtsp_st = rt->rtsp_streams[i];
594 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
595 rtsp_st->dynamic_handler->free(
596 rtsp_st->dynamic_protocol_context);
600 av_free(rt->rtsp_streams);
602 avformat_close_input(&rt->asf_ctx);
604 if (rt->ts && CONFIG_RTPDEC)
605 ff_mpegts_parse_close(rt->ts);
607 av_free(rt->recvbuf);
610 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
612 RTSPState *rt = s->priv_data;
614 int reordering_queue_size = rt->reordering_queue_size;
615 if (reordering_queue_size < 0) {
616 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
617 reordering_queue_size = 0;
619 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
622 /* open the RTP context */
623 if (rtsp_st->stream_index >= 0)
624 st = s->streams[rtsp_st->stream_index];
626 s->ctx_flags |= AVFMTCTX_NOHEADER;
628 if (s->oformat && CONFIG_RTSP_MUXER) {
629 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
631 RTSP_TCP_MAX_PACKET_SIZE,
632 rtsp_st->stream_index);
633 /* Ownership of rtp_handle is passed to the rtp mux context */
634 rtsp_st->rtp_handle = NULL;
637 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
638 return 0; // Don't need to open any parser here
639 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
640 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
641 rtsp_st->dynamic_protocol_context,
642 rtsp_st->dynamic_handler);
643 else if (CONFIG_RTPDEC)
644 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
645 rtsp_st->sdp_payload_type,
646 reordering_queue_size);
648 if (!rtsp_st->transport_priv) {
649 return AVERROR(ENOMEM);
650 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
651 if (rtsp_st->dynamic_handler) {
652 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
653 rtsp_st->dynamic_protocol_context,
654 rtsp_st->dynamic_handler);
661 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
662 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
669 q += strspn(q, SPACE_CHARS);
670 v = strtol(q, &p, 10);
674 v = strtol(p, &p, 10);
683 /* XXX: only one transport specification is parsed */
684 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
686 char transport_protocol[16];
688 char lower_transport[16];
690 RTSPTransportField *th;
693 reply->nb_transports = 0;
696 p += strspn(p, SPACE_CHARS);
700 th = &reply->transports[reply->nb_transports];
702 get_word_sep(transport_protocol, sizeof(transport_protocol),
704 if (!av_strcasecmp (transport_protocol, "rtp")) {
705 get_word_sep(profile, sizeof(profile), "/;,", &p);
706 lower_transport[0] = '\0';
707 /* rtp/avp/<protocol> */
709 get_word_sep(lower_transport, sizeof(lower_transport),
712 th->transport = RTSP_TRANSPORT_RTP;
713 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
714 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
715 /* x-pn-tng/<protocol> */
716 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
718 th->transport = RTSP_TRANSPORT_RDT;
719 } else if (!av_strcasecmp(transport_protocol, "raw")) {
720 get_word_sep(profile, sizeof(profile), "/;,", &p);
721 lower_transport[0] = '\0';
722 /* raw/raw/<protocol> */
724 get_word_sep(lower_transport, sizeof(lower_transport),
727 th->transport = RTSP_TRANSPORT_RAW;
729 if (!av_strcasecmp(lower_transport, "TCP"))
730 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
732 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
736 /* get each parameter */
737 while (*p != '\0' && *p != ',') {
738 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
739 if (!strcmp(parameter, "port")) {
742 rtsp_parse_range(&th->port_min, &th->port_max, &p);
744 } else if (!strcmp(parameter, "client_port")) {
747 rtsp_parse_range(&th->client_port_min,
748 &th->client_port_max, &p);
750 } else if (!strcmp(parameter, "server_port")) {
753 rtsp_parse_range(&th->server_port_min,
754 &th->server_port_max, &p);
756 } else if (!strcmp(parameter, "interleaved")) {
759 rtsp_parse_range(&th->interleaved_min,
760 &th->interleaved_max, &p);
762 } else if (!strcmp(parameter, "multicast")) {
763 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
764 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
765 } else if (!strcmp(parameter, "ttl")) {
769 th->ttl = strtol(p, &end, 10);
772 } else if (!strcmp(parameter, "destination")) {
775 get_word_sep(buf, sizeof(buf), ";,", &p);
776 get_sockaddr(buf, &th->destination);
778 } else if (!strcmp(parameter, "source")) {
781 get_word_sep(buf, sizeof(buf), ";,", &p);
782 av_strlcpy(th->source, buf, sizeof(th->source));
784 } else if (!strcmp(parameter, "mode")) {
787 get_word_sep(buf, sizeof(buf), ";, ", &p);
788 if (!strcmp(buf, "record") ||
789 !strcmp(buf, "receive"))
794 while (*p != ';' && *p != '\0' && *p != ',')
802 reply->nb_transports++;
806 static void handle_rtp_info(RTSPState *rt, const char *url,
807 uint32_t seq, uint32_t rtptime)
810 if (!rtptime || !url[0])
812 if (rt->transport != RTSP_TRANSPORT_RTP)
814 for (i = 0; i < rt->nb_rtsp_streams; i++) {
815 RTSPStream *rtsp_st = rt->rtsp_streams[i];
816 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
819 if (!strcmp(rtsp_st->control_url, url)) {
820 rtpctx->base_timestamp = rtptime;
826 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
829 char key[20], value[1024], url[1024] = "";
830 uint32_t seq = 0, rtptime = 0;
833 p += strspn(p, SPACE_CHARS);
836 get_word_sep(key, sizeof(key), "=", &p);
840 get_word_sep(value, sizeof(value), ";, ", &p);
842 if (!strcmp(key, "url"))
843 av_strlcpy(url, value, sizeof(url));
844 else if (!strcmp(key, "seq"))
845 seq = strtoul(value, NULL, 10);
846 else if (!strcmp(key, "rtptime"))
847 rtptime = strtoul(value, NULL, 10);
849 handle_rtp_info(rt, url, seq, rtptime);
858 handle_rtp_info(rt, url, seq, rtptime);
861 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
862 RTSPState *rt, const char *method)
866 /* NOTE: we do case independent match for broken servers */
868 if (av_stristart(p, "Session:", &p)) {
870 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
871 if (av_stristart(p, ";timeout=", &p) &&
872 (t = strtol(p, NULL, 10)) > 0) {
875 } else if (av_stristart(p, "Content-Length:", &p)) {
876 reply->content_length = strtol(p, NULL, 10);
877 } else if (av_stristart(p, "Transport:", &p)) {
878 rtsp_parse_transport(reply, p);
879 } else if (av_stristart(p, "CSeq:", &p)) {
880 reply->seq = strtol(p, NULL, 10);
881 } else if (av_stristart(p, "Range:", &p)) {
882 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
883 } else if (av_stristart(p, "RealChallenge1:", &p)) {
884 p += strspn(p, SPACE_CHARS);
885 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
886 } else if (av_stristart(p, "Server:", &p)) {
887 p += strspn(p, SPACE_CHARS);
888 av_strlcpy(reply->server, p, sizeof(reply->server));
889 } else if (av_stristart(p, "Notice:", &p) ||
890 av_stristart(p, "X-Notice:", &p)) {
891 reply->notice = strtol(p, NULL, 10);
892 } else if (av_stristart(p, "Location:", &p)) {
893 p += strspn(p, SPACE_CHARS);
894 av_strlcpy(reply->location, p , sizeof(reply->location));
895 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
896 p += strspn(p, SPACE_CHARS);
897 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
898 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
899 p += strspn(p, SPACE_CHARS);
900 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
901 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
902 p += strspn(p, SPACE_CHARS);
903 if (method && !strcmp(method, "DESCRIBE"))
904 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
905 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
906 p += strspn(p, SPACE_CHARS);
907 if (method && !strcmp(method, "PLAY"))
908 rtsp_parse_rtp_info(rt, p);
909 } else if (av_stristart(p, "Public:", &p) && rt) {
910 if (strstr(p, "GET_PARAMETER") &&
911 method && !strcmp(method, "OPTIONS"))
912 rt->get_parameter_supported = 1;
913 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
914 p += strspn(p, SPACE_CHARS);
915 rt->accept_dynamic_rate = atoi(p);
916 } else if (av_stristart(p, "Content-Type:", &p)) {
917 p += strspn(p, SPACE_CHARS);
918 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
922 /* skip a RTP/TCP interleaved packet */
923 void ff_rtsp_skip_packet(AVFormatContext *s)
925 RTSPState *rt = s->priv_data;
929 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
932 len = AV_RB16(buf + 1);
934 av_dlog(s, "skipping RTP packet len=%d\n", len);
939 if (len1 > sizeof(buf))
941 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
948 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
949 unsigned char **content_ptr,
950 int return_on_interleaved_data, const char *method)
952 RTSPState *rt = s->priv_data;
953 char buf[4096], buf1[1024], *q;
956 int ret, content_length, line_count = 0, request = 0;
957 unsigned char *content = NULL;
963 memset(reply, 0, sizeof(*reply));
965 /* parse reply (XXX: use buffers) */
966 rt->last_reply[0] = '\0';
970 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
971 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
977 /* XXX: only parse it if first char on line ? */
978 if (return_on_interleaved_data) {
981 ff_rtsp_skip_packet(s);
982 } else if (ch != '\r') {
983 if ((q - buf) < sizeof(buf) - 1)
989 av_dlog(s, "line='%s'\n", buf);
991 /* test if last line */
995 if (line_count == 0) {
997 get_word(buf1, sizeof(buf1), &p);
998 if (!strncmp(buf1, "RTSP/", 5)) {
999 get_word(buf1, sizeof(buf1), &p);
1000 reply->status_code = atoi(buf1);
1001 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1003 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1004 get_word(buf1, sizeof(buf1), &p); // object
1008 ff_rtsp_parse_line(reply, p, rt, method);
1009 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1010 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1015 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1016 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1018 content_length = reply->content_length;
1019 if (content_length > 0) {
1020 /* leave some room for a trailing '\0' (useful for simple parsing) */
1021 content = av_malloc(content_length + 1);
1022 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1023 content[content_length] = '\0';
1026 *content_ptr = content;
1032 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1033 const char* ptr = buf;
1035 if (!strcmp(reply->reason, "OPTIONS")) {
1036 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1038 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1039 if (reply->session_id[0])
1040 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1043 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1045 av_strlcat(buf, "\r\n", sizeof(buf));
1047 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1048 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1051 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1053 rt->last_cmd_time = av_gettime();
1054 /* Even if the request from the server had data, it is not the data
1055 * that the caller wants or expects. The memory could also be leaked
1056 * if the actual following reply has content data. */
1058 av_freep(content_ptr);
1059 /* If method is set, this is called from ff_rtsp_send_cmd,
1060 * where a reply to exactly this request is awaited. For
1061 * callers from within packet receiving, we just want to
1062 * return to the caller and go back to receiving packets. */
1068 if (rt->seq != reply->seq) {
1069 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1070 rt->seq, reply->seq);
1074 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1075 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1076 reply->notice == 2306 /* Continuous Feed Terminated */) {
1077 rt->state = RTSP_STATE_IDLE;
1078 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1079 return AVERROR(EIO); /* data or server error */
1080 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1081 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1082 return AVERROR(EPERM);
1088 * Send a command to the RTSP server without waiting for the reply.
1090 * @param s RTSP (de)muxer context
1091 * @param method the method for the request
1092 * @param url the target url for the request
1093 * @param headers extra header lines to include in the request
1094 * @param send_content if non-null, the data to send as request body content
1095 * @param send_content_length the length of the send_content data, or 0 if
1096 * send_content is null
1098 * @return zero if success, nonzero otherwise
1100 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1101 const char *method, const char *url,
1102 const char *headers,
1103 const unsigned char *send_content,
1104 int send_content_length)
1106 RTSPState *rt = s->priv_data;
1107 char buf[4096], *out_buf;
1108 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1110 /* Add in RTSP headers */
1113 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1115 av_strlcat(buf, headers, sizeof(buf));
1116 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1117 if (rt->session_id[0] != '\0' && (!headers ||
1118 !strstr(headers, "\nIf-Match:"))) {
1119 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1122 char *str = ff_http_auth_create_response(&rt->auth_state,
1123 rt->auth, url, method);
1125 av_strlcat(buf, str, sizeof(buf));
1128 if (send_content_length > 0 && send_content)
1129 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1130 av_strlcat(buf, "\r\n", sizeof(buf));
1132 /* base64 encode rtsp if tunneling */
1133 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1134 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1135 out_buf = base64buf;
1138 av_dlog(s, "Sending:\n%s--\n", buf);
1140 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1141 if (send_content_length > 0 && send_content) {
1142 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1143 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1144 "with content data not supported\n");
1145 return AVERROR_PATCHWELCOME;
1147 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1149 rt->last_cmd_time = av_gettime();
1154 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1155 const char *url, const char *headers)
1157 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1160 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1161 const char *headers, RTSPMessageHeader *reply,
1162 unsigned char **content_ptr)
1164 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1165 content_ptr, NULL, 0);
1168 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1169 const char *method, const char *url,
1171 RTSPMessageHeader *reply,
1172 unsigned char **content_ptr,
1173 const unsigned char *send_content,
1174 int send_content_length)
1176 RTSPState *rt = s->priv_data;
1177 HTTPAuthType cur_auth_type;
1178 int ret, attempts = 0;
1181 cur_auth_type = rt->auth_state.auth_type;
1182 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1184 send_content_length)))
1187 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1191 if (reply->status_code == 401 &&
1192 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1193 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1196 if (reply->status_code > 400){
1197 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1201 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1207 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1208 int lower_transport, const char *real_challenge)
1210 RTSPState *rt = s->priv_data;
1211 int rtx = 0, j, i, err, interleave = 0, port_off;
1212 RTSPStream *rtsp_st;
1213 RTSPMessageHeader reply1, *reply = &reply1;
1215 const char *trans_pref;
1217 if (rt->transport == RTSP_TRANSPORT_RDT)
1218 trans_pref = "x-pn-tng";
1219 else if (rt->transport == RTSP_TRANSPORT_RAW)
1220 trans_pref = "RAW/RAW";
1222 trans_pref = "RTP/AVP";
1224 /* default timeout: 1 minute */
1227 /* Choose a random starting offset within the first half of the
1228 * port range, to allow for a number of ports to try even if the offset
1229 * happens to be at the end of the random range. */
1230 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1231 /* even random offset */
1232 port_off -= port_off & 0x01;
1234 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1235 char transport[2048];
1238 * WMS serves all UDP data over a single connection, the RTX, which
1239 * isn't necessarily the first in the SDP but has to be the first
1240 * to be set up, else the second/third SETUP will fail with a 461.
1242 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1243 rt->server_type == RTSP_SERVER_WMS) {
1246 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1247 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1249 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1253 if (rtx == rt->nb_rtsp_streams)
1254 return -1; /* no RTX found */
1255 rtsp_st = rt->rtsp_streams[rtx];
1257 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1259 rtsp_st = rt->rtsp_streams[i];
1262 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1265 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1266 port = reply->transports[0].client_port_min;
1270 /* first try in specified port range */
1271 while (j <= rt->rtp_port_max) {
1272 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1273 "?localport=%d", j);
1274 /* we will use two ports per rtp stream (rtp and rtcp) */
1276 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1277 &s->interrupt_callback, NULL))
1280 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1285 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1287 snprintf(transport, sizeof(transport) - 1,
1288 "%s/UDP;", trans_pref);
1289 if (rt->server_type != RTSP_SERVER_REAL)
1290 av_strlcat(transport, "unicast;", sizeof(transport));
1291 av_strlcatf(transport, sizeof(transport),
1292 "client_port=%d", port);
1293 if (rt->transport == RTSP_TRANSPORT_RTP &&
1294 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1295 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1299 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1300 /* For WMS streams, the application streams are only used for
1301 * UDP. When trying to set it up for TCP streams, the server
1302 * will return an error. Therefore, we skip those streams. */
1303 if (rt->server_type == RTSP_SERVER_WMS &&
1304 (rtsp_st->stream_index < 0 ||
1305 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1308 snprintf(transport, sizeof(transport) - 1,
1309 "%s/TCP;", trans_pref);
1310 if (rt->transport != RTSP_TRANSPORT_RDT)
1311 av_strlcat(transport, "unicast;", sizeof(transport));
1312 av_strlcatf(transport, sizeof(transport),
1313 "interleaved=%d-%d",
1314 interleave, interleave + 1);
1318 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1319 snprintf(transport, sizeof(transport) - 1,
1320 "%s/UDP;multicast", trans_pref);
1323 av_strlcat(transport, ";mode=record", sizeof(transport));
1324 } else if (rt->server_type == RTSP_SERVER_REAL ||
1325 rt->server_type == RTSP_SERVER_WMS)
1326 av_strlcat(transport, ";mode=play", sizeof(transport));
1327 snprintf(cmd, sizeof(cmd),
1328 "Transport: %s\r\n",
1330 if (rt->accept_dynamic_rate)
1331 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1332 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1333 char real_res[41], real_csum[9];
1334 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1336 av_strlcatf(cmd, sizeof(cmd),
1338 "RealChallenge2: %s, sd=%s\r\n",
1339 rt->session_id, real_res, real_csum);
1341 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1342 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1345 } else if (reply->status_code != RTSP_STATUS_OK ||
1346 reply->nb_transports != 1) {
1347 err = AVERROR_INVALIDDATA;
1351 /* XXX: same protocol for all streams is required */
1353 if (reply->transports[0].lower_transport != rt->lower_transport ||
1354 reply->transports[0].transport != rt->transport) {
1355 err = AVERROR_INVALIDDATA;
1359 rt->lower_transport = reply->transports[0].lower_transport;
1360 rt->transport = reply->transports[0].transport;
1363 /* Fail if the server responded with another lower transport mode
1364 * than what we requested. */
1365 if (reply->transports[0].lower_transport != lower_transport) {
1366 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1367 err = AVERROR_INVALIDDATA;
1371 switch(reply->transports[0].lower_transport) {
1372 case RTSP_LOWER_TRANSPORT_TCP:
1373 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1374 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1377 case RTSP_LOWER_TRANSPORT_UDP: {
1378 char url[1024], options[30] = "";
1380 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1381 av_strlcpy(options, "?connect=1", sizeof(options));
1382 /* Use source address if specified */
1383 if (reply->transports[0].source[0]) {
1384 ff_url_join(url, sizeof(url), "rtp", NULL,
1385 reply->transports[0].source,
1386 reply->transports[0].server_port_min, "%s", options);
1388 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1389 reply->transports[0].server_port_min, "%s", options);
1391 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1392 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1393 err = AVERROR_INVALIDDATA;
1396 /* Try to initialize the connection state in a
1397 * potential NAT router by sending dummy packets.
1398 * RTP/RTCP dummy packets are used for RDT, too.
1400 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1402 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1405 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1406 char url[1024], namebuf[50], optbuf[20] = "";
1407 struct sockaddr_storage addr;
1410 if (reply->transports[0].destination.ss_family) {
1411 addr = reply->transports[0].destination;
1412 port = reply->transports[0].port_min;
1413 ttl = reply->transports[0].ttl;
1415 addr = rtsp_st->sdp_ip;
1416 port = rtsp_st->sdp_port;
1417 ttl = rtsp_st->sdp_ttl;
1420 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1421 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1422 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1423 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1424 port, "%s", optbuf);
1425 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1426 &s->interrupt_callback, NULL) < 0) {
1427 err = AVERROR_INVALIDDATA;
1434 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1438 if (rt->nb_rtsp_streams && reply->timeout > 0)
1439 rt->timeout = reply->timeout;
1441 if (rt->server_type == RTSP_SERVER_REAL)
1442 rt->need_subscription = 1;
1447 ff_rtsp_undo_setup(s);
1451 void ff_rtsp_close_connections(AVFormatContext *s)
1453 RTSPState *rt = s->priv_data;
1454 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1455 ffurl_close(rt->rtsp_hd);
1456 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1459 int ff_rtsp_connect(AVFormatContext *s)
1461 RTSPState *rt = s->priv_data;
1462 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1463 int port, err, tcp_fd;
1464 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1465 int lower_transport_mask = 0;
1466 char real_challenge[64] = "";
1467 struct sockaddr_storage peer;
1468 socklen_t peer_len = sizeof(peer);
1470 if (rt->rtp_port_max < rt->rtp_port_min) {
1471 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1472 "than min port %d\n", rt->rtp_port_max,
1474 return AVERROR(EINVAL);
1477 if (!ff_network_init())
1478 return AVERROR(EIO);
1480 if (s->max_delay < 0) /* Not set by the caller */
1481 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1483 rt->control_transport = RTSP_MODE_PLAIN;
1484 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1485 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1486 rt->control_transport = RTSP_MODE_TUNNEL;
1488 /* Only pass through valid flags from here */
1489 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1492 lower_transport_mask = rt->lower_transport_mask;
1493 /* extract hostname and port */
1494 av_url_split(NULL, 0, auth, sizeof(auth),
1495 host, sizeof(host), &port, path, sizeof(path), s->filename);
1497 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1500 port = RTSP_DEFAULT_PORT;
1502 if (!lower_transport_mask)
1503 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1506 /* Only UDP or TCP - UDP multicast isn't supported. */
1507 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1508 (1 << RTSP_LOWER_TRANSPORT_TCP);
1509 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1510 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1511 "only UDP and TCP are supported for output.\n");
1512 err = AVERROR(EINVAL);
1517 /* Construct the URI used in request; this is similar to s->filename,
1518 * but with authentication credentials removed and RTSP specific options
1520 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1521 host, port, "%s", path);
1523 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1524 /* set up initial handshake for tunneling */
1525 char httpname[1024];
1526 char sessioncookie[17];
1529 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1530 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1531 av_get_random_seed(), av_get_random_seed());
1534 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1535 &s->interrupt_callback) < 0) {
1540 /* generate GET headers */
1541 snprintf(headers, sizeof(headers),
1542 "x-sessioncookie: %s\r\n"
1543 "Accept: application/x-rtsp-tunnelled\r\n"
1544 "Pragma: no-cache\r\n"
1545 "Cache-Control: no-cache\r\n",
1547 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1549 /* complete the connection */
1550 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1556 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1557 &s->interrupt_callback) < 0 ) {
1562 /* generate POST headers */
1563 snprintf(headers, sizeof(headers),
1564 "x-sessioncookie: %s\r\n"
1565 "Content-Type: application/x-rtsp-tunnelled\r\n"
1566 "Pragma: no-cache\r\n"
1567 "Cache-Control: no-cache\r\n"
1568 "Content-Length: 32767\r\n"
1569 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1571 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1572 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1574 /* Initialize the authentication state for the POST session. The HTTP
1575 * protocol implementation doesn't properly handle multi-pass
1576 * authentication for POST requests, since it would require one of
1578 * - implementing Expect: 100-continue, which many HTTP servers
1579 * don't support anyway, even less the RTSP servers that do HTTP
1581 * - sending the whole POST data until getting a 401 reply specifying
1582 * what authentication method to use, then resending all that data
1583 * - waiting for potential 401 replies directly after sending the
1584 * POST header (waiting for some unspecified time)
1585 * Therefore, we copy the full auth state, which works for both basic
1586 * and digest. (For digest, we would have to synchronize the nonce
1587 * count variable between the two sessions, if we'd do more requests
1588 * with the original session, though.)
1590 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1592 /* complete the connection */
1593 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1598 /* open the tcp connection */
1599 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1600 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1601 &s->interrupt_callback, NULL) < 0) {
1605 rt->rtsp_hd_out = rt->rtsp_hd;
1609 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1610 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1611 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1612 NULL, 0, NI_NUMERICHOST);
1615 /* request options supported by the server; this also detects server
1617 for (rt->server_type = RTSP_SERVER_RTP;;) {
1619 if (rt->server_type == RTSP_SERVER_REAL)
1622 * The following entries are required for proper
1623 * streaming from a Realmedia server. They are
1624 * interdependent in some way although we currently
1625 * don't quite understand how. Values were copied
1626 * from mplayer SVN r23589.
1627 * ClientChallenge is a 16-byte ID in hex
1628 * CompanyID is a 16-byte ID in base64
1630 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1631 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1632 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1633 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1635 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1636 if (reply->status_code != RTSP_STATUS_OK) {
1637 err = AVERROR_INVALIDDATA;
1641 /* detect server type if not standard-compliant RTP */
1642 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1643 rt->server_type = RTSP_SERVER_REAL;
1645 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1646 rt->server_type = RTSP_SERVER_WMS;
1647 } else if (rt->server_type == RTSP_SERVER_REAL)
1648 strcpy(real_challenge, reply->real_challenge);
1652 if (s->iformat && CONFIG_RTSP_DEMUXER)
1653 err = ff_rtsp_setup_input_streams(s, reply);
1654 else if (CONFIG_RTSP_MUXER)
1655 err = ff_rtsp_setup_output_streams(s, host);
1660 int lower_transport = ff_log2_tab[lower_transport_mask &
1661 ~(lower_transport_mask - 1)];
1663 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1664 rt->server_type == RTSP_SERVER_REAL ?
1665 real_challenge : NULL);
1668 lower_transport_mask &= ~(1 << lower_transport);
1669 if (lower_transport_mask == 0 && err == 1) {
1670 err = AVERROR(EPROTONOSUPPORT);
1675 rt->lower_transport_mask = lower_transport_mask;
1676 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1677 rt->state = RTSP_STATE_IDLE;
1678 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1681 ff_rtsp_close_streams(s);
1682 ff_rtsp_close_connections(s);
1683 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1684 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1685 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1693 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1696 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1697 uint8_t *buf, int buf_size, int64_t wait_end)
1699 RTSPState *rt = s->priv_data;
1700 RTSPStream *rtsp_st;
1701 int n, i, ret, tcp_fd, timeout_cnt = 0;
1703 struct pollfd *p = rt->p;
1704 int *fds = NULL, fdsnum, fdsidx;
1707 if (ff_check_interrupt(&s->interrupt_callback))
1708 return AVERROR_EXIT;
1709 if (wait_end && wait_end - av_gettime() < 0)
1710 return AVERROR(EAGAIN);
1713 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1714 p[max_p].fd = tcp_fd;
1715 p[max_p++].events = POLLIN;
1719 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1720 rtsp_st = rt->rtsp_streams[i];
1721 if (rtsp_st->rtp_handle) {
1722 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1724 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1728 av_log(s, AV_LOG_ERROR,
1729 "Number of fds %d not supported\n", fdsnum);
1730 return AVERROR_INVALIDDATA;
1732 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1733 p[max_p].fd = fds[fdsidx];
1734 p[max_p++].events = POLLIN;
1739 n = poll(p, max_p, POLL_TIMEOUT_MS);
1741 int j = 1 - (tcp_fd == -1);
1743 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1744 rtsp_st = rt->rtsp_streams[i];
1745 if (rtsp_st->rtp_handle) {
1746 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1747 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1749 *prtsp_st = rtsp_st;
1756 #if CONFIG_RTSP_DEMUXER
1757 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1758 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1759 if (rt->state == RTSP_STATE_STREAMING) {
1760 if (!ff_rtsp_parse_streaming_commands(s))
1763 av_log(s, AV_LOG_WARNING,
1764 "Unable to answer to TEARDOWN\n");
1768 RTSPMessageHeader reply;
1769 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1772 /* XXX: parse message */
1773 if (rt->state != RTSP_STATE_STREAMING)
1778 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1779 return AVERROR(ETIMEDOUT);
1780 } else if (n < 0 && errno != EINTR)
1781 return AVERROR(errno);
1785 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1786 const uint8_t *buf, int len)
1788 RTSPState *rt = s->priv_data;
1792 if (rt->nb_rtsp_streams == 1) {
1793 *rtsp_st = rt->rtsp_streams[0];
1796 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1797 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1799 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1800 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1803 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1804 *rtsp_st = rt->rtsp_streams[i];
1811 av_log(s, AV_LOG_WARNING,
1812 "Unable to pick stream for packet - SSRC not known for "
1814 return AVERROR(EAGAIN);
1817 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1818 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1819 *rtsp_st = rt->rtsp_streams[i];
1825 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1826 return AVERROR(EAGAIN);
1829 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1831 RTSPState *rt = s->priv_data;
1833 RTSPStream *rtsp_st, *first_queue_st = NULL;
1834 int64_t wait_end = 0;
1836 if (rt->nb_byes == rt->nb_rtsp_streams)
1839 /* get next frames from the same RTP packet */
1840 if (rt->cur_transport_priv) {
1841 if (rt->transport == RTSP_TRANSPORT_RDT) {
1842 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1843 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1844 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1845 } else if (rt->ts && CONFIG_RTPDEC) {
1846 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1848 rt->recvbuf_pos += ret;
1849 ret = rt->recvbuf_pos < rt->recvbuf_len;
1854 rt->cur_transport_priv = NULL;
1856 } else if (ret == 1) {
1859 rt->cur_transport_priv = NULL;
1863 if (rt->transport == RTSP_TRANSPORT_RTP) {
1865 int64_t first_queue_time = 0;
1866 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1867 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1871 queue_time = ff_rtp_queued_packet_time(rtpctx);
1872 if (queue_time && (queue_time - first_queue_time < 0 ||
1873 !first_queue_time)) {
1874 first_queue_time = queue_time;
1875 first_queue_st = rt->rtsp_streams[i];
1878 if (first_queue_time) {
1879 wait_end = first_queue_time + s->max_delay;
1882 first_queue_st = NULL;
1886 /* read next RTP packet */
1888 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1890 return AVERROR(ENOMEM);
1893 switch(rt->lower_transport) {
1895 #if CONFIG_RTSP_DEMUXER
1896 case RTSP_LOWER_TRANSPORT_TCP:
1897 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1900 case RTSP_LOWER_TRANSPORT_UDP:
1901 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1902 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1903 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1904 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1906 case RTSP_LOWER_TRANSPORT_CUSTOM:
1907 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1908 wait_end && wait_end < av_gettime())
1909 len = AVERROR(EAGAIN);
1911 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1912 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1913 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1914 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1917 if (len == AVERROR(EAGAIN) && first_queue_st &&
1918 rt->transport == RTSP_TRANSPORT_RTP) {
1919 rtsp_st = first_queue_st;
1920 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1927 if (rt->transport == RTSP_TRANSPORT_RDT) {
1928 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1929 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1930 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1931 if (rtsp_st->feedback) {
1932 AVIOContext *pb = NULL;
1933 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1935 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1938 /* Either bad packet, or a RTCP packet. Check if the
1939 * first_rtcp_ntp_time field was initialized. */
1940 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1941 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1942 /* first_rtcp_ntp_time has been initialized for this stream,
1943 * copy the same value to all other uninitialized streams,
1944 * in order to map their timestamp origin to the same ntp time
1947 AVStream *st = NULL;
1948 if (rtsp_st->stream_index >= 0)
1949 st = s->streams[rtsp_st->stream_index];
1950 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1951 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1952 AVStream *st2 = NULL;
1953 if (rt->rtsp_streams[i]->stream_index >= 0)
1954 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1955 if (rtpctx2 && st && st2 &&
1956 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1957 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1958 rtpctx2->rtcp_ts_offset = av_rescale_q(
1959 rtpctx->rtcp_ts_offset, st->time_base,
1964 if (ret == -RTCP_BYE) {
1967 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1968 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1970 if (rt->nb_byes == rt->nb_rtsp_streams)
1974 } else if (rt->ts && CONFIG_RTPDEC) {
1975 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1978 rt->recvbuf_len = len;
1979 rt->recvbuf_pos = ret;
1980 rt->cur_transport_priv = rt->ts;
1987 return AVERROR_INVALIDDATA;
1993 /* more packets may follow, so we save the RTP context */
1994 rt->cur_transport_priv = rtsp_st->transport_priv;
1998 #endif /* CONFIG_RTPDEC */
2000 #if CONFIG_SDP_DEMUXER
2001 static int sdp_probe(AVProbeData *p1)
2003 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2005 /* we look for a line beginning "c=IN IP" */
2006 while (p < p_end && *p != '\0') {
2007 if (p + sizeof("c=IN IP") - 1 < p_end &&
2008 av_strstart(p, "c=IN IP", NULL))
2009 return AVPROBE_SCORE_MAX / 2;
2011 while (p < p_end - 1 && *p != '\n') p++;
2020 static int sdp_read_header(AVFormatContext *s)
2022 RTSPState *rt = s->priv_data;
2023 RTSPStream *rtsp_st;
2028 if (!ff_network_init())
2029 return AVERROR(EIO);
2031 if (s->max_delay < 0) /* Not set by the caller */
2032 s->max_delay = DEFAULT_REORDERING_DELAY;
2033 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2034 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2036 /* read the whole sdp file */
2037 /* XXX: better loading */
2038 content = av_malloc(SDP_MAX_SIZE);
2039 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2042 return AVERROR_INVALIDDATA;
2044 content[size] ='\0';
2046 err = ff_sdp_parse(s, content);
2050 /* open each RTP stream */
2051 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2053 rtsp_st = rt->rtsp_streams[i];
2055 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2056 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2057 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2058 ff_url_join(url, sizeof(url), "rtp", NULL,
2059 namebuf, rtsp_st->sdp_port,
2060 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2062 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2063 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2064 &s->interrupt_callback, NULL) < 0) {
2065 err = AVERROR_INVALIDDATA;
2069 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2074 ff_rtsp_close_streams(s);
2079 static int sdp_read_close(AVFormatContext *s)
2081 ff_rtsp_close_streams(s);
2086 static const AVClass sdp_demuxer_class = {
2087 .class_name = "SDP demuxer",
2088 .item_name = av_default_item_name,
2089 .option = sdp_options,
2090 .version = LIBAVUTIL_VERSION_INT,
2093 AVInputFormat ff_sdp_demuxer = {
2095 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2096 .priv_data_size = sizeof(RTSPState),
2097 .read_probe = sdp_probe,
2098 .read_header = sdp_read_header,
2099 .read_packet = ff_rtsp_fetch_packet,
2100 .read_close = sdp_read_close,
2101 .priv_class = &sdp_demuxer_class,
2103 #endif /* CONFIG_SDP_DEMUXER */
2105 #if CONFIG_RTP_DEMUXER
2106 static int rtp_probe(AVProbeData *p)
2108 if (av_strstart(p->filename, "rtp:", NULL))
2109 return AVPROBE_SCORE_MAX;
2113 static int rtp_read_header(AVFormatContext *s)
2115 uint8_t recvbuf[1500];
2116 char host[500], sdp[500];
2118 URLContext* in = NULL;
2120 AVCodecContext codec = { 0 };
2121 struct sockaddr_storage addr;
2123 socklen_t addrlen = sizeof(addr);
2124 RTSPState *rt = s->priv_data;
2126 if (!ff_network_init())
2127 return AVERROR(EIO);
2129 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2130 &s->interrupt_callback, NULL);
2135 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2136 if (ret == AVERROR(EAGAIN))
2141 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2145 if ((recvbuf[0] & 0xc0) != 0x80) {
2146 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2151 if (RTP_PT_IS_RTCP(recvbuf[1]))
2154 payload_type = recvbuf[1] & 0x7f;
2157 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2161 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2162 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2163 "without an SDP file describing it\n",
2167 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2168 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2169 "properly you need an SDP file "
2173 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2174 NULL, 0, s->filename);
2176 snprintf(sdp, sizeof(sdp),
2177 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2178 addr.ss_family == AF_INET ? 4 : 6, host,
2179 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2180 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2181 port, payload_type);
2182 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2184 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2187 /* sdp_read_header initializes this again */
2190 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2192 ret = sdp_read_header(s);
2203 static const AVClass rtp_demuxer_class = {
2204 .class_name = "RTP demuxer",
2205 .item_name = av_default_item_name,
2206 .option = rtp_options,
2207 .version = LIBAVUTIL_VERSION_INT,
2210 AVInputFormat ff_rtp_demuxer = {
2212 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2213 .priv_data_size = sizeof(RTSPState),
2214 .read_probe = rtp_probe,
2215 .read_header = rtp_read_header,
2216 .read_packet = ff_rtsp_fetch_packet,
2217 .read_close = sdp_read_close,
2218 .flags = AVFMT_NOFILE,
2219 .priv_class = &rtp_demuxer_class,
2221 #endif /* CONFIG_RTP_DEMUXER */