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 { "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(),
99 static const AVOption sdp_options[] = {
100 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
101 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
102 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
103 RTSP_REORDERING_OPTS(),
107 static const AVOption rtp_options[] = {
108 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
109 RTSP_REORDERING_OPTS(),
113 static void get_word_until_chars(char *buf, int buf_size,
114 const char *sep, const char **pp)
120 p += strspn(p, SPACE_CHARS);
122 while (!strchr(sep, *p) && *p != '\0') {
123 if ((q - buf) < buf_size - 1)
132 static void get_word_sep(char *buf, int buf_size, const char *sep,
135 if (**pp == '/') (*pp)++;
136 get_word_until_chars(buf, buf_size, sep, pp);
139 static void get_word(char *buf, int buf_size, const char **pp)
141 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
144 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
146 * Used for seeking in the rtp stream.
148 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
152 p += strspn(p, SPACE_CHARS);
153 if (!av_stristart(p, "npt=", &p))
156 *start = AV_NOPTS_VALUE;
157 *end = AV_NOPTS_VALUE;
159 get_word_sep(buf, sizeof(buf), "-", &p);
160 av_parse_time(start, buf, 1);
163 get_word_sep(buf, sizeof(buf), "-", &p);
164 av_parse_time(end, buf, 1);
168 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
170 struct addrinfo hints = { 0 }, *ai = NULL;
171 hints.ai_flags = AI_NUMERICHOST;
172 if (getaddrinfo(buf, NULL, &hints, &ai))
174 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
180 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
181 RTSPStream *rtsp_st, AVCodecContext *codec)
186 codec->codec_id = handler->codec_id;
187 rtsp_st->dynamic_handler = handler;
188 if (handler->alloc) {
189 rtsp_st->dynamic_protocol_context = handler->alloc();
190 if (!rtsp_st->dynamic_protocol_context)
191 rtsp_st->dynamic_handler = NULL;
195 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
196 static int sdp_parse_rtpmap(AVFormatContext *s,
197 AVStream *st, RTSPStream *rtsp_st,
198 int payload_type, const char *p)
200 AVCodecContext *codec = st->codec;
206 /* See if we can handle this kind of payload.
207 * The space should normally not be there but some Real streams or
208 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
209 * have a trailing space. */
210 get_word_sep(buf, sizeof(buf), "/ ", &p);
211 if (payload_type < RTP_PT_PRIVATE) {
212 /* We are in a standard case
213 * (from http://www.iana.org/assignments/rtp-parameters). */
214 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
217 if (codec->codec_id == AV_CODEC_ID_NONE) {
218 RTPDynamicProtocolHandler *handler =
219 ff_rtp_handler_find_by_name(buf, codec->codec_type);
220 init_rtp_handler(handler, rtsp_st, codec);
221 /* If no dynamic handler was found, check with the list of standard
222 * allocated types, if such a stream for some reason happens to
223 * use a private payload type. This isn't handled in rtpdec.c, since
224 * the format name from the rtpmap line never is passed into rtpdec. */
225 if (!rtsp_st->dynamic_handler)
226 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
229 c = avcodec_find_decoder(codec->codec_id);
235 get_word_sep(buf, sizeof(buf), "/", &p);
237 switch (codec->codec_type) {
238 case AVMEDIA_TYPE_AUDIO:
239 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
241 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
243 codec->sample_rate = i;
244 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
245 get_word_sep(buf, sizeof(buf), "/", &p);
250 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
252 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
255 case AVMEDIA_TYPE_VIDEO:
256 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
258 avpriv_set_pts_info(st, 32, 1, i);
263 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
264 rtsp_st->dynamic_handler->init(s, st->index,
265 rtsp_st->dynamic_protocol_context);
269 /* parse the attribute line from the fmtp a line of an sdp response. This
270 * is broken out as a function because it is used in rtp_h264.c, which is
272 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
273 char *value, int value_size)
275 *p += strspn(*p, SPACE_CHARS);
277 get_word_sep(attr, attr_size, "=", p);
280 get_word_sep(value, value_size, ";", p);
288 typedef struct SDPParseState {
290 struct sockaddr_storage default_ip;
292 int skip_media; ///< set if an unknown m= line occurs
295 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
296 int letter, const char *buf)
298 RTSPState *rt = s->priv_data;
299 char buf1[64], st_type[64];
301 enum AVMediaType codec_type;
305 struct sockaddr_storage sdp_ip;
308 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
311 if (s1->skip_media && letter != 'm')
315 get_word(buf1, sizeof(buf1), &p);
316 if (strcmp(buf1, "IN") != 0)
318 get_word(buf1, sizeof(buf1), &p);
319 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
321 get_word_sep(buf1, sizeof(buf1), "/", &p);
322 if (get_sockaddr(buf1, &sdp_ip))
327 get_word_sep(buf1, sizeof(buf1), "/", &p);
330 if (s->nb_streams == 0) {
331 s1->default_ip = sdp_ip;
332 s1->default_ttl = ttl;
334 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
335 rtsp_st->sdp_ip = sdp_ip;
336 rtsp_st->sdp_ttl = ttl;
340 av_dict_set(&s->metadata, "title", p, 0);
343 if (s->nb_streams == 0) {
344 av_dict_set(&s->metadata, "comment", p, 0);
351 codec_type = AVMEDIA_TYPE_UNKNOWN;
352 get_word(st_type, sizeof(st_type), &p);
353 if (!strcmp(st_type, "audio")) {
354 codec_type = AVMEDIA_TYPE_AUDIO;
355 } else if (!strcmp(st_type, "video")) {
356 codec_type = AVMEDIA_TYPE_VIDEO;
357 } else if (!strcmp(st_type, "application")) {
358 codec_type = AVMEDIA_TYPE_DATA;
360 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
364 rtsp_st = av_mallocz(sizeof(RTSPStream));
367 rtsp_st->stream_index = -1;
368 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
370 rtsp_st->sdp_ip = s1->default_ip;
371 rtsp_st->sdp_ttl = s1->default_ttl;
373 get_word(buf1, sizeof(buf1), &p); /* port */
374 rtsp_st->sdp_port = atoi(buf1);
376 get_word(buf1, sizeof(buf1), &p); /* protocol */
377 if (!strcmp(buf1, "udp"))
378 rt->transport = RTSP_TRANSPORT_RAW;
379 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
380 rtsp_st->feedback = 1;
382 /* XXX: handle list of formats */
383 get_word(buf1, sizeof(buf1), &p); /* format list */
384 rtsp_st->sdp_payload_type = atoi(buf1);
386 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
387 /* no corresponding stream */
388 if (rt->transport == RTSP_TRANSPORT_RAW) {
389 if (!rt->ts && CONFIG_RTPDEC)
390 rt->ts = ff_mpegts_parse_open(s);
392 RTPDynamicProtocolHandler *handler;
393 handler = ff_rtp_handler_find_by_id(
394 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
395 init_rtp_handler(handler, rtsp_st, NULL);
396 if (handler && handler->init)
397 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
399 } else if (rt->server_type == RTSP_SERVER_WMS &&
400 codec_type == AVMEDIA_TYPE_DATA) {
401 /* RTX stream, a stream that carries all the other actual
402 * audio/video streams. Don't expose this to the callers. */
404 st = avformat_new_stream(s, NULL);
407 st->id = rt->nb_rtsp_streams - 1;
408 rtsp_st->stream_index = st->index;
409 st->codec->codec_type = codec_type;
410 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
411 RTPDynamicProtocolHandler *handler;
412 /* if standard payload type, we can find the codec right now */
413 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
414 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
415 st->codec->sample_rate > 0)
416 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
417 /* Even static payload types may need a custom depacketizer */
418 handler = ff_rtp_handler_find_by_id(
419 rtsp_st->sdp_payload_type, st->codec->codec_type);
420 init_rtp_handler(handler, rtsp_st, st->codec);
421 if (handler && handler->init)
422 handler->init(s, st->index,
423 rtsp_st->dynamic_protocol_context);
426 /* put a default control url */
427 av_strlcpy(rtsp_st->control_url, rt->control_uri,
428 sizeof(rtsp_st->control_url));
431 if (av_strstart(p, "control:", &p)) {
432 if (s->nb_streams == 0) {
433 if (!strncmp(p, "rtsp://", 7))
434 av_strlcpy(rt->control_uri, p,
435 sizeof(rt->control_uri));
438 /* get the control url */
439 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
441 /* XXX: may need to add full url resolution */
442 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
444 if (proto[0] == '\0') {
445 /* relative control URL */
446 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
447 av_strlcat(rtsp_st->control_url, "/",
448 sizeof(rtsp_st->control_url));
449 av_strlcat(rtsp_st->control_url, p,
450 sizeof(rtsp_st->control_url));
452 av_strlcpy(rtsp_st->control_url, p,
453 sizeof(rtsp_st->control_url));
455 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
456 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
457 get_word(buf1, sizeof(buf1), &p);
458 payload_type = atoi(buf1);
459 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
460 if (rtsp_st->stream_index >= 0) {
461 st = s->streams[rtsp_st->stream_index];
462 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
464 } else if (av_strstart(p, "fmtp:", &p) ||
465 av_strstart(p, "framesize:", &p)) {
466 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
467 // let dynamic protocol handlers have a stab at the line.
468 get_word(buf1, sizeof(buf1), &p);
469 payload_type = atoi(buf1);
470 for (i = 0; i < rt->nb_rtsp_streams; i++) {
471 rtsp_st = rt->rtsp_streams[i];
472 if (rtsp_st->sdp_payload_type == payload_type &&
473 rtsp_st->dynamic_handler &&
474 rtsp_st->dynamic_handler->parse_sdp_a_line)
475 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
476 rtsp_st->dynamic_protocol_context, buf);
478 } else if (av_strstart(p, "range:", &p)) {
481 // this is so that seeking on a streamed file can work.
482 rtsp_parse_range_npt(p, &start, &end);
483 s->start_time = start;
484 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
485 s->duration = (end == AV_NOPTS_VALUE) ?
486 AV_NOPTS_VALUE : end - start;
487 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
489 rt->transport = RTSP_TRANSPORT_RDT;
490 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
492 st = s->streams[s->nb_streams - 1];
493 st->codec->sample_rate = atoi(p);
494 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
496 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
497 get_word(buf1, sizeof(buf1), &p); // ignore tag
498 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
499 p += strspn(p, SPACE_CHARS);
500 if (av_strstart(p, "inline:", &p))
501 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
503 if (rt->server_type == RTSP_SERVER_WMS)
504 ff_wms_parse_sdp_a_line(s, p);
505 if (s->nb_streams > 0) {
506 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
508 if (rt->server_type == RTSP_SERVER_REAL)
509 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
511 if (rtsp_st->dynamic_handler &&
512 rtsp_st->dynamic_handler->parse_sdp_a_line)
513 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
514 rtsp_st->stream_index,
515 rtsp_st->dynamic_protocol_context, buf);
522 int ff_sdp_parse(AVFormatContext *s, const char *content)
524 RTSPState *rt = s->priv_data;
527 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
528 * contain long SDP lines containing complete ASF Headers (several
529 * kB) or arrays of MDPR (RM stream descriptor) headers plus
530 * "rulebooks" describing their properties. Therefore, the SDP line
533 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
534 * in rtpdec_xiph.c. */
536 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
540 p += strspn(p, SPACE_CHARS);
548 /* get the content */
550 while (*p != '\n' && *p != '\r' && *p != '\0') {
551 if ((q - buf) < sizeof(buf) - 1)
556 sdp_parse_line(s, s1, letter, buf);
558 while (*p != '\n' && *p != '\0')
563 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
564 if (!rt->p) return AVERROR(ENOMEM);
567 #endif /* CONFIG_RTPDEC */
569 void ff_rtsp_undo_setup(AVFormatContext *s)
571 RTSPState *rt = s->priv_data;
574 for (i = 0; i < rt->nb_rtsp_streams; i++) {
575 RTSPStream *rtsp_st = rt->rtsp_streams[i];
578 if (rtsp_st->transport_priv) {
580 AVFormatContext *rtpctx = rtsp_st->transport_priv;
581 av_write_trailer(rtpctx);
582 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
584 avio_close_dyn_buf(rtpctx->pb, &ptr);
587 avio_close(rtpctx->pb);
589 avformat_free_context(rtpctx);
590 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
591 ff_rdt_parse_close(rtsp_st->transport_priv);
592 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
593 ff_rtp_parse_close(rtsp_st->transport_priv);
595 rtsp_st->transport_priv = NULL;
596 if (rtsp_st->rtp_handle)
597 ffurl_close(rtsp_st->rtp_handle);
598 rtsp_st->rtp_handle = NULL;
602 /* close and free RTSP streams */
603 void ff_rtsp_close_streams(AVFormatContext *s)
605 RTSPState *rt = s->priv_data;
609 ff_rtsp_undo_setup(s);
610 for (i = 0; i < rt->nb_rtsp_streams; i++) {
611 rtsp_st = rt->rtsp_streams[i];
613 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
614 rtsp_st->dynamic_handler->free(
615 rtsp_st->dynamic_protocol_context);
619 av_free(rt->rtsp_streams);
621 avformat_close_input(&rt->asf_ctx);
623 if (rt->ts && CONFIG_RTPDEC)
624 ff_mpegts_parse_close(rt->ts);
626 av_free(rt->recvbuf);
629 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
631 RTSPState *rt = s->priv_data;
633 int reordering_queue_size = rt->reordering_queue_size;
634 if (reordering_queue_size < 0) {
635 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
636 reordering_queue_size = 0;
638 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
641 /* open the RTP context */
642 if (rtsp_st->stream_index >= 0)
643 st = s->streams[rtsp_st->stream_index];
645 s->ctx_flags |= AVFMTCTX_NOHEADER;
647 if (s->oformat && CONFIG_RTSP_MUXER) {
648 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
650 RTSP_TCP_MAX_PACKET_SIZE,
651 rtsp_st->stream_index);
652 /* Ownership of rtp_handle is passed to the rtp mux context */
653 rtsp_st->rtp_handle = NULL;
656 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
657 return 0; // Don't need to open any parser here
658 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
659 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
660 rtsp_st->dynamic_protocol_context,
661 rtsp_st->dynamic_handler);
662 else if (CONFIG_RTPDEC)
663 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
664 rtsp_st->sdp_payload_type,
665 reordering_queue_size);
667 if (!rtsp_st->transport_priv) {
668 return AVERROR(ENOMEM);
669 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
670 if (rtsp_st->dynamic_handler) {
671 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
672 rtsp_st->dynamic_protocol_context,
673 rtsp_st->dynamic_handler);
675 if (rtsp_st->crypto_suite[0])
676 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
677 rtsp_st->crypto_suite,
678 rtsp_st->crypto_params);
684 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
685 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
692 q += strspn(q, SPACE_CHARS);
693 v = strtol(q, &p, 10);
697 v = strtol(p, &p, 10);
706 /* XXX: only one transport specification is parsed */
707 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
709 char transport_protocol[16];
711 char lower_transport[16];
713 RTSPTransportField *th;
716 reply->nb_transports = 0;
719 p += strspn(p, SPACE_CHARS);
723 th = &reply->transports[reply->nb_transports];
725 get_word_sep(transport_protocol, sizeof(transport_protocol),
727 if (!av_strcasecmp (transport_protocol, "rtp")) {
728 get_word_sep(profile, sizeof(profile), "/;,", &p);
729 lower_transport[0] = '\0';
730 /* rtp/avp/<protocol> */
732 get_word_sep(lower_transport, sizeof(lower_transport),
735 th->transport = RTSP_TRANSPORT_RTP;
736 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
737 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
738 /* x-pn-tng/<protocol> */
739 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
741 th->transport = RTSP_TRANSPORT_RDT;
742 } else if (!av_strcasecmp(transport_protocol, "raw")) {
743 get_word_sep(profile, sizeof(profile), "/;,", &p);
744 lower_transport[0] = '\0';
745 /* raw/raw/<protocol> */
747 get_word_sep(lower_transport, sizeof(lower_transport),
750 th->transport = RTSP_TRANSPORT_RAW;
752 if (!av_strcasecmp(lower_transport, "TCP"))
753 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
755 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
759 /* get each parameter */
760 while (*p != '\0' && *p != ',') {
761 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
762 if (!strcmp(parameter, "port")) {
765 rtsp_parse_range(&th->port_min, &th->port_max, &p);
767 } else if (!strcmp(parameter, "client_port")) {
770 rtsp_parse_range(&th->client_port_min,
771 &th->client_port_max, &p);
773 } else if (!strcmp(parameter, "server_port")) {
776 rtsp_parse_range(&th->server_port_min,
777 &th->server_port_max, &p);
779 } else if (!strcmp(parameter, "interleaved")) {
782 rtsp_parse_range(&th->interleaved_min,
783 &th->interleaved_max, &p);
785 } else if (!strcmp(parameter, "multicast")) {
786 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
787 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
788 } else if (!strcmp(parameter, "ttl")) {
792 th->ttl = strtol(p, &end, 10);
795 } else if (!strcmp(parameter, "destination")) {
798 get_word_sep(buf, sizeof(buf), ";,", &p);
799 get_sockaddr(buf, &th->destination);
801 } else if (!strcmp(parameter, "source")) {
804 get_word_sep(buf, sizeof(buf), ";,", &p);
805 av_strlcpy(th->source, buf, sizeof(th->source));
807 } else if (!strcmp(parameter, "mode")) {
810 get_word_sep(buf, sizeof(buf), ";, ", &p);
811 if (!strcmp(buf, "record") ||
812 !strcmp(buf, "receive"))
817 while (*p != ';' && *p != '\0' && *p != ',')
825 reply->nb_transports++;
829 static void handle_rtp_info(RTSPState *rt, const char *url,
830 uint32_t seq, uint32_t rtptime)
833 if (!rtptime || !url[0])
835 if (rt->transport != RTSP_TRANSPORT_RTP)
837 for (i = 0; i < rt->nb_rtsp_streams; i++) {
838 RTSPStream *rtsp_st = rt->rtsp_streams[i];
839 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
842 if (!strcmp(rtsp_st->control_url, url)) {
843 rtpctx->base_timestamp = rtptime;
849 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
852 char key[20], value[1024], url[1024] = "";
853 uint32_t seq = 0, rtptime = 0;
856 p += strspn(p, SPACE_CHARS);
859 get_word_sep(key, sizeof(key), "=", &p);
863 get_word_sep(value, sizeof(value), ";, ", &p);
865 if (!strcmp(key, "url"))
866 av_strlcpy(url, value, sizeof(url));
867 else if (!strcmp(key, "seq"))
868 seq = strtoul(value, NULL, 10);
869 else if (!strcmp(key, "rtptime"))
870 rtptime = strtoul(value, NULL, 10);
872 handle_rtp_info(rt, url, seq, rtptime);
881 handle_rtp_info(rt, url, seq, rtptime);
884 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
885 RTSPState *rt, const char *method)
889 /* NOTE: we do case independent match for broken servers */
891 if (av_stristart(p, "Session:", &p)) {
893 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
894 if (av_stristart(p, ";timeout=", &p) &&
895 (t = strtol(p, NULL, 10)) > 0) {
898 } else if (av_stristart(p, "Content-Length:", &p)) {
899 reply->content_length = strtol(p, NULL, 10);
900 } else if (av_stristart(p, "Transport:", &p)) {
901 rtsp_parse_transport(reply, p);
902 } else if (av_stristart(p, "CSeq:", &p)) {
903 reply->seq = strtol(p, NULL, 10);
904 } else if (av_stristart(p, "Range:", &p)) {
905 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
906 } else if (av_stristart(p, "RealChallenge1:", &p)) {
907 p += strspn(p, SPACE_CHARS);
908 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
909 } else if (av_stristart(p, "Server:", &p)) {
910 p += strspn(p, SPACE_CHARS);
911 av_strlcpy(reply->server, p, sizeof(reply->server));
912 } else if (av_stristart(p, "Notice:", &p) ||
913 av_stristart(p, "X-Notice:", &p)) {
914 reply->notice = strtol(p, NULL, 10);
915 } else if (av_stristart(p, "Location:", &p)) {
916 p += strspn(p, SPACE_CHARS);
917 av_strlcpy(reply->location, p , sizeof(reply->location));
918 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
919 p += strspn(p, SPACE_CHARS);
920 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
921 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
922 p += strspn(p, SPACE_CHARS);
923 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
924 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
925 p += strspn(p, SPACE_CHARS);
926 if (method && !strcmp(method, "DESCRIBE"))
927 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
928 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
929 p += strspn(p, SPACE_CHARS);
930 if (method && !strcmp(method, "PLAY"))
931 rtsp_parse_rtp_info(rt, p);
932 } else if (av_stristart(p, "Public:", &p) && rt) {
933 if (strstr(p, "GET_PARAMETER") &&
934 method && !strcmp(method, "OPTIONS"))
935 rt->get_parameter_supported = 1;
936 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
937 p += strspn(p, SPACE_CHARS);
938 rt->accept_dynamic_rate = atoi(p);
939 } else if (av_stristart(p, "Content-Type:", &p)) {
940 p += strspn(p, SPACE_CHARS);
941 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
945 /* skip a RTP/TCP interleaved packet */
946 void ff_rtsp_skip_packet(AVFormatContext *s)
948 RTSPState *rt = s->priv_data;
952 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
955 len = AV_RB16(buf + 1);
957 av_dlog(s, "skipping RTP packet len=%d\n", len);
962 if (len1 > sizeof(buf))
964 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
971 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
972 unsigned char **content_ptr,
973 int return_on_interleaved_data, const char *method)
975 RTSPState *rt = s->priv_data;
976 char buf[4096], buf1[1024], *q;
979 int ret, content_length, line_count = 0, request = 0;
980 unsigned char *content = NULL;
986 memset(reply, 0, sizeof(*reply));
988 /* parse reply (XXX: use buffers) */
989 rt->last_reply[0] = '\0';
993 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
994 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
1000 /* XXX: only parse it if first char on line ? */
1001 if (return_on_interleaved_data) {
1004 ff_rtsp_skip_packet(s);
1005 } else if (ch != '\r') {
1006 if ((q - buf) < sizeof(buf) - 1)
1012 av_dlog(s, "line='%s'\n", buf);
1014 /* test if last line */
1018 if (line_count == 0) {
1019 /* get reply code */
1020 get_word(buf1, sizeof(buf1), &p);
1021 if (!strncmp(buf1, "RTSP/", 5)) {
1022 get_word(buf1, sizeof(buf1), &p);
1023 reply->status_code = atoi(buf1);
1024 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1026 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1027 get_word(buf1, sizeof(buf1), &p); // object
1031 ff_rtsp_parse_line(reply, p, rt, method);
1032 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1033 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1038 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1039 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1041 content_length = reply->content_length;
1042 if (content_length > 0) {
1043 /* leave some room for a trailing '\0' (useful for simple parsing) */
1044 content = av_malloc(content_length + 1);
1045 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1046 content[content_length] = '\0';
1049 *content_ptr = content;
1055 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1056 const char* ptr = buf;
1058 if (!strcmp(reply->reason, "OPTIONS")) {
1059 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1061 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1062 if (reply->session_id[0])
1063 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1066 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1068 av_strlcat(buf, "\r\n", sizeof(buf));
1070 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1071 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1074 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1076 rt->last_cmd_time = av_gettime();
1077 /* Even if the request from the server had data, it is not the data
1078 * that the caller wants or expects. The memory could also be leaked
1079 * if the actual following reply has content data. */
1081 av_freep(content_ptr);
1082 /* If method is set, this is called from ff_rtsp_send_cmd,
1083 * where a reply to exactly this request is awaited. For
1084 * callers from within packet receiving, we just want to
1085 * return to the caller and go back to receiving packets. */
1091 if (rt->seq != reply->seq) {
1092 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1093 rt->seq, reply->seq);
1097 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1098 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1099 reply->notice == 2306 /* Continuous Feed Terminated */) {
1100 rt->state = RTSP_STATE_IDLE;
1101 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1102 return AVERROR(EIO); /* data or server error */
1103 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1104 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1105 return AVERROR(EPERM);
1111 * Send a command to the RTSP server without waiting for the reply.
1113 * @param s RTSP (de)muxer context
1114 * @param method the method for the request
1115 * @param url the target url for the request
1116 * @param headers extra header lines to include in the request
1117 * @param send_content if non-null, the data to send as request body content
1118 * @param send_content_length the length of the send_content data, or 0 if
1119 * send_content is null
1121 * @return zero if success, nonzero otherwise
1123 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1124 const char *method, const char *url,
1125 const char *headers,
1126 const unsigned char *send_content,
1127 int send_content_length)
1129 RTSPState *rt = s->priv_data;
1130 char buf[4096], *out_buf;
1131 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1133 /* Add in RTSP headers */
1136 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1138 av_strlcat(buf, headers, sizeof(buf));
1139 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1140 if (rt->session_id[0] != '\0' && (!headers ||
1141 !strstr(headers, "\nIf-Match:"))) {
1142 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1145 char *str = ff_http_auth_create_response(&rt->auth_state,
1146 rt->auth, url, method);
1148 av_strlcat(buf, str, sizeof(buf));
1151 if (send_content_length > 0 && send_content)
1152 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1153 av_strlcat(buf, "\r\n", sizeof(buf));
1155 /* base64 encode rtsp if tunneling */
1156 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1157 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1158 out_buf = base64buf;
1161 av_dlog(s, "Sending:\n%s--\n", buf);
1163 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1164 if (send_content_length > 0 && send_content) {
1165 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1166 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1167 "with content data not supported\n");
1168 return AVERROR_PATCHWELCOME;
1170 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1172 rt->last_cmd_time = av_gettime();
1177 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1178 const char *url, const char *headers)
1180 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1183 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1184 const char *headers, RTSPMessageHeader *reply,
1185 unsigned char **content_ptr)
1187 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1188 content_ptr, NULL, 0);
1191 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1192 const char *method, const char *url,
1194 RTSPMessageHeader *reply,
1195 unsigned char **content_ptr,
1196 const unsigned char *send_content,
1197 int send_content_length)
1199 RTSPState *rt = s->priv_data;
1200 HTTPAuthType cur_auth_type;
1201 int ret, attempts = 0;
1204 cur_auth_type = rt->auth_state.auth_type;
1205 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1207 send_content_length)))
1210 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1214 if (reply->status_code == 401 &&
1215 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1216 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1219 if (reply->status_code > 400){
1220 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1224 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1230 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1231 int lower_transport, const char *real_challenge)
1233 RTSPState *rt = s->priv_data;
1234 int rtx = 0, j, i, err, interleave = 0, port_off;
1235 RTSPStream *rtsp_st;
1236 RTSPMessageHeader reply1, *reply = &reply1;
1238 const char *trans_pref;
1240 if (rt->transport == RTSP_TRANSPORT_RDT)
1241 trans_pref = "x-pn-tng";
1242 else if (rt->transport == RTSP_TRANSPORT_RAW)
1243 trans_pref = "RAW/RAW";
1245 trans_pref = "RTP/AVP";
1247 /* default timeout: 1 minute */
1250 /* Choose a random starting offset within the first half of the
1251 * port range, to allow for a number of ports to try even if the offset
1252 * happens to be at the end of the random range. */
1253 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1254 /* even random offset */
1255 port_off -= port_off & 0x01;
1257 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1258 char transport[2048];
1261 * WMS serves all UDP data over a single connection, the RTX, which
1262 * isn't necessarily the first in the SDP but has to be the first
1263 * to be set up, else the second/third SETUP will fail with a 461.
1265 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1266 rt->server_type == RTSP_SERVER_WMS) {
1269 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1270 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1272 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1276 if (rtx == rt->nb_rtsp_streams)
1277 return -1; /* no RTX found */
1278 rtsp_st = rt->rtsp_streams[rtx];
1280 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1282 rtsp_st = rt->rtsp_streams[i];
1285 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1288 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1289 port = reply->transports[0].client_port_min;
1293 /* first try in specified port range */
1294 while (j <= rt->rtp_port_max) {
1295 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1296 "?localport=%d", j);
1297 /* we will use two ports per rtp stream (rtp and rtcp) */
1299 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1300 &s->interrupt_callback, NULL))
1303 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1308 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1310 snprintf(transport, sizeof(transport) - 1,
1311 "%s/UDP;", trans_pref);
1312 if (rt->server_type != RTSP_SERVER_REAL)
1313 av_strlcat(transport, "unicast;", sizeof(transport));
1314 av_strlcatf(transport, sizeof(transport),
1315 "client_port=%d", port);
1316 if (rt->transport == RTSP_TRANSPORT_RTP &&
1317 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1318 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1322 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1323 /* For WMS streams, the application streams are only used for
1324 * UDP. When trying to set it up for TCP streams, the server
1325 * will return an error. Therefore, we skip those streams. */
1326 if (rt->server_type == RTSP_SERVER_WMS &&
1327 (rtsp_st->stream_index < 0 ||
1328 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1331 snprintf(transport, sizeof(transport) - 1,
1332 "%s/TCP;", trans_pref);
1333 if (rt->transport != RTSP_TRANSPORT_RDT)
1334 av_strlcat(transport, "unicast;", sizeof(transport));
1335 av_strlcatf(transport, sizeof(transport),
1336 "interleaved=%d-%d",
1337 interleave, interleave + 1);
1341 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1342 snprintf(transport, sizeof(transport) - 1,
1343 "%s/UDP;multicast", trans_pref);
1346 av_strlcat(transport, ";mode=record", sizeof(transport));
1347 } else if (rt->server_type == RTSP_SERVER_REAL ||
1348 rt->server_type == RTSP_SERVER_WMS)
1349 av_strlcat(transport, ";mode=play", sizeof(transport));
1350 snprintf(cmd, sizeof(cmd),
1351 "Transport: %s\r\n",
1353 if (rt->accept_dynamic_rate)
1354 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1355 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1356 char real_res[41], real_csum[9];
1357 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1359 av_strlcatf(cmd, sizeof(cmd),
1361 "RealChallenge2: %s, sd=%s\r\n",
1362 rt->session_id, real_res, real_csum);
1364 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1365 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1368 } else if (reply->status_code != RTSP_STATUS_OK ||
1369 reply->nb_transports != 1) {
1370 err = AVERROR_INVALIDDATA;
1374 /* XXX: same protocol for all streams is required */
1376 if (reply->transports[0].lower_transport != rt->lower_transport ||
1377 reply->transports[0].transport != rt->transport) {
1378 err = AVERROR_INVALIDDATA;
1382 rt->lower_transport = reply->transports[0].lower_transport;
1383 rt->transport = reply->transports[0].transport;
1386 /* Fail if the server responded with another lower transport mode
1387 * than what we requested. */
1388 if (reply->transports[0].lower_transport != lower_transport) {
1389 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1390 err = AVERROR_INVALIDDATA;
1394 switch(reply->transports[0].lower_transport) {
1395 case RTSP_LOWER_TRANSPORT_TCP:
1396 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1397 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1400 case RTSP_LOWER_TRANSPORT_UDP: {
1401 char url[1024], options[30] = "";
1403 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1404 av_strlcpy(options, "?connect=1", sizeof(options));
1405 /* Use source address if specified */
1406 if (reply->transports[0].source[0]) {
1407 ff_url_join(url, sizeof(url), "rtp", NULL,
1408 reply->transports[0].source,
1409 reply->transports[0].server_port_min, "%s", options);
1411 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1412 reply->transports[0].server_port_min, "%s", options);
1414 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1415 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1416 err = AVERROR_INVALIDDATA;
1419 /* Try to initialize the connection state in a
1420 * potential NAT router by sending dummy packets.
1421 * RTP/RTCP dummy packets are used for RDT, too.
1423 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1425 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1428 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1429 char url[1024], namebuf[50], optbuf[20] = "";
1430 struct sockaddr_storage addr;
1433 if (reply->transports[0].destination.ss_family) {
1434 addr = reply->transports[0].destination;
1435 port = reply->transports[0].port_min;
1436 ttl = reply->transports[0].ttl;
1438 addr = rtsp_st->sdp_ip;
1439 port = rtsp_st->sdp_port;
1440 ttl = rtsp_st->sdp_ttl;
1443 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1444 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1445 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1446 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1447 port, "%s", optbuf);
1448 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1449 &s->interrupt_callback, NULL) < 0) {
1450 err = AVERROR_INVALIDDATA;
1457 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1461 if (rt->nb_rtsp_streams && reply->timeout > 0)
1462 rt->timeout = reply->timeout;
1464 if (rt->server_type == RTSP_SERVER_REAL)
1465 rt->need_subscription = 1;
1470 ff_rtsp_undo_setup(s);
1474 void ff_rtsp_close_connections(AVFormatContext *s)
1476 RTSPState *rt = s->priv_data;
1477 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1478 ffurl_close(rt->rtsp_hd);
1479 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1482 int ff_rtsp_connect(AVFormatContext *s)
1484 RTSPState *rt = s->priv_data;
1485 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1486 int port, err, tcp_fd;
1487 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1488 int lower_transport_mask = 0;
1489 char real_challenge[64] = "";
1490 struct sockaddr_storage peer;
1491 socklen_t peer_len = sizeof(peer);
1493 if (rt->rtp_port_max < rt->rtp_port_min) {
1494 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1495 "than min port %d\n", rt->rtp_port_max,
1497 return AVERROR(EINVAL);
1500 if (!ff_network_init())
1501 return AVERROR(EIO);
1503 if (s->max_delay < 0) /* Not set by the caller */
1504 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1506 rt->control_transport = RTSP_MODE_PLAIN;
1507 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1508 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1509 rt->control_transport = RTSP_MODE_TUNNEL;
1511 /* Only pass through valid flags from here */
1512 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1515 lower_transport_mask = rt->lower_transport_mask;
1516 /* extract hostname and port */
1517 av_url_split(NULL, 0, auth, sizeof(auth),
1518 host, sizeof(host), &port, path, sizeof(path), s->filename);
1520 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1523 port = RTSP_DEFAULT_PORT;
1525 if (!lower_transport_mask)
1526 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1529 /* Only UDP or TCP - UDP multicast isn't supported. */
1530 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1531 (1 << RTSP_LOWER_TRANSPORT_TCP);
1532 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1533 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1534 "only UDP and TCP are supported for output.\n");
1535 err = AVERROR(EINVAL);
1540 /* Construct the URI used in request; this is similar to s->filename,
1541 * but with authentication credentials removed and RTSP specific options
1543 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1544 host, port, "%s", path);
1546 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1547 /* set up initial handshake for tunneling */
1548 char httpname[1024];
1549 char sessioncookie[17];
1552 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1553 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1554 av_get_random_seed(), av_get_random_seed());
1557 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1558 &s->interrupt_callback) < 0) {
1563 /* generate GET headers */
1564 snprintf(headers, sizeof(headers),
1565 "x-sessioncookie: %s\r\n"
1566 "Accept: application/x-rtsp-tunnelled\r\n"
1567 "Pragma: no-cache\r\n"
1568 "Cache-Control: no-cache\r\n",
1570 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1572 /* complete the connection */
1573 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1579 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1580 &s->interrupt_callback) < 0 ) {
1585 /* generate POST headers */
1586 snprintf(headers, sizeof(headers),
1587 "x-sessioncookie: %s\r\n"
1588 "Content-Type: application/x-rtsp-tunnelled\r\n"
1589 "Pragma: no-cache\r\n"
1590 "Cache-Control: no-cache\r\n"
1591 "Content-Length: 32767\r\n"
1592 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1594 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1595 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1597 /* Initialize the authentication state for the POST session. The HTTP
1598 * protocol implementation doesn't properly handle multi-pass
1599 * authentication for POST requests, since it would require one of
1601 * - implementing Expect: 100-continue, which many HTTP servers
1602 * don't support anyway, even less the RTSP servers that do HTTP
1604 * - sending the whole POST data until getting a 401 reply specifying
1605 * what authentication method to use, then resending all that data
1606 * - waiting for potential 401 replies directly after sending the
1607 * POST header (waiting for some unspecified time)
1608 * Therefore, we copy the full auth state, which works for both basic
1609 * and digest. (For digest, we would have to synchronize the nonce
1610 * count variable between the two sessions, if we'd do more requests
1611 * with the original session, though.)
1613 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1615 /* complete the connection */
1616 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1621 /* open the tcp connection */
1622 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1623 "?timeout=%d", rt->stimeout);
1624 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1625 &s->interrupt_callback, NULL) < 0) {
1629 rt->rtsp_hd_out = rt->rtsp_hd;
1633 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1634 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1635 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1636 NULL, 0, NI_NUMERICHOST);
1639 /* request options supported by the server; this also detects server
1641 for (rt->server_type = RTSP_SERVER_RTP;;) {
1643 if (rt->server_type == RTSP_SERVER_REAL)
1646 * The following entries are required for proper
1647 * streaming from a Realmedia server. They are
1648 * interdependent in some way although we currently
1649 * don't quite understand how. Values were copied
1650 * from mplayer SVN r23589.
1651 * ClientChallenge is a 16-byte ID in hex
1652 * CompanyID is a 16-byte ID in base64
1654 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1655 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1656 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1657 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1659 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1660 if (reply->status_code != RTSP_STATUS_OK) {
1661 err = AVERROR_INVALIDDATA;
1665 /* detect server type if not standard-compliant RTP */
1666 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1667 rt->server_type = RTSP_SERVER_REAL;
1669 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1670 rt->server_type = RTSP_SERVER_WMS;
1671 } else if (rt->server_type == RTSP_SERVER_REAL)
1672 strcpy(real_challenge, reply->real_challenge);
1676 if (s->iformat && CONFIG_RTSP_DEMUXER)
1677 err = ff_rtsp_setup_input_streams(s, reply);
1678 else if (CONFIG_RTSP_MUXER)
1679 err = ff_rtsp_setup_output_streams(s, host);
1684 int lower_transport = ff_log2_tab[lower_transport_mask &
1685 ~(lower_transport_mask - 1)];
1687 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1688 rt->server_type == RTSP_SERVER_REAL ?
1689 real_challenge : NULL);
1692 lower_transport_mask &= ~(1 << lower_transport);
1693 if (lower_transport_mask == 0 && err == 1) {
1694 err = AVERROR(EPROTONOSUPPORT);
1699 rt->lower_transport_mask = lower_transport_mask;
1700 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1701 rt->state = RTSP_STATE_IDLE;
1702 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1705 ff_rtsp_close_streams(s);
1706 ff_rtsp_close_connections(s);
1707 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1708 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1709 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1717 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1720 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1721 uint8_t *buf, int buf_size, int64_t wait_end)
1723 RTSPState *rt = s->priv_data;
1724 RTSPStream *rtsp_st;
1725 int n, i, ret, tcp_fd, timeout_cnt = 0;
1727 struct pollfd *p = rt->p;
1728 int *fds = NULL, fdsnum, fdsidx;
1731 if (ff_check_interrupt(&s->interrupt_callback))
1732 return AVERROR_EXIT;
1733 if (wait_end && wait_end - av_gettime() < 0)
1734 return AVERROR(EAGAIN);
1737 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1738 p[max_p].fd = tcp_fd;
1739 p[max_p++].events = POLLIN;
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 (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1748 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1752 av_log(s, AV_LOG_ERROR,
1753 "Number of fds %d not supported\n", fdsnum);
1754 return AVERROR_INVALIDDATA;
1756 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1757 p[max_p].fd = fds[fdsidx];
1758 p[max_p++].events = POLLIN;
1763 n = poll(p, max_p, POLL_TIMEOUT_MS);
1765 int j = 1 - (tcp_fd == -1);
1767 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1768 rtsp_st = rt->rtsp_streams[i];
1769 if (rtsp_st->rtp_handle) {
1770 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1771 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1773 *prtsp_st = rtsp_st;
1780 #if CONFIG_RTSP_DEMUXER
1781 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1782 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1783 if (rt->state == RTSP_STATE_STREAMING) {
1784 if (!ff_rtsp_parse_streaming_commands(s))
1787 av_log(s, AV_LOG_WARNING,
1788 "Unable to answer to TEARDOWN\n");
1792 RTSPMessageHeader reply;
1793 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1796 /* XXX: parse message */
1797 if (rt->state != RTSP_STATE_STREAMING)
1802 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1803 return AVERROR(ETIMEDOUT);
1804 } else if (n < 0 && errno != EINTR)
1805 return AVERROR(errno);
1809 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1810 const uint8_t *buf, int len)
1812 RTSPState *rt = s->priv_data;
1816 if (rt->nb_rtsp_streams == 1) {
1817 *rtsp_st = rt->rtsp_streams[0];
1820 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1821 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1823 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1824 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1827 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1828 *rtsp_st = rt->rtsp_streams[i];
1835 av_log(s, AV_LOG_WARNING,
1836 "Unable to pick stream for packet - SSRC not known for "
1838 return AVERROR(EAGAIN);
1841 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1842 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1843 *rtsp_st = rt->rtsp_streams[i];
1849 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1850 return AVERROR(EAGAIN);
1853 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1855 RTSPState *rt = s->priv_data;
1857 RTSPStream *rtsp_st, *first_queue_st = NULL;
1858 int64_t wait_end = 0;
1860 if (rt->nb_byes == rt->nb_rtsp_streams)
1863 /* get next frames from the same RTP packet */
1864 if (rt->cur_transport_priv) {
1865 if (rt->transport == RTSP_TRANSPORT_RDT) {
1866 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1867 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1868 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1869 } else if (rt->ts && CONFIG_RTPDEC) {
1870 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1872 rt->recvbuf_pos += ret;
1873 ret = rt->recvbuf_pos < rt->recvbuf_len;
1878 rt->cur_transport_priv = NULL;
1880 } else if (ret == 1) {
1883 rt->cur_transport_priv = NULL;
1887 if (rt->transport == RTSP_TRANSPORT_RTP) {
1889 int64_t first_queue_time = 0;
1890 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1891 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1895 queue_time = ff_rtp_queued_packet_time(rtpctx);
1896 if (queue_time && (queue_time - first_queue_time < 0 ||
1897 !first_queue_time)) {
1898 first_queue_time = queue_time;
1899 first_queue_st = rt->rtsp_streams[i];
1902 if (first_queue_time) {
1903 wait_end = first_queue_time + s->max_delay;
1906 first_queue_st = NULL;
1910 /* read next RTP packet */
1912 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1914 return AVERROR(ENOMEM);
1917 switch(rt->lower_transport) {
1919 #if CONFIG_RTSP_DEMUXER
1920 case RTSP_LOWER_TRANSPORT_TCP:
1921 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1924 case RTSP_LOWER_TRANSPORT_UDP:
1925 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1926 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1927 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1928 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1930 case RTSP_LOWER_TRANSPORT_CUSTOM:
1931 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1932 wait_end && wait_end < av_gettime())
1933 len = AVERROR(EAGAIN);
1935 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1936 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1937 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1938 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1941 if (len == AVERROR(EAGAIN) && first_queue_st &&
1942 rt->transport == RTSP_TRANSPORT_RTP) {
1943 rtsp_st = first_queue_st;
1944 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1951 if (rt->transport == RTSP_TRANSPORT_RDT) {
1952 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1953 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1954 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1955 if (rtsp_st->feedback) {
1956 AVIOContext *pb = NULL;
1957 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1959 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1962 /* Either bad packet, or a RTCP packet. Check if the
1963 * first_rtcp_ntp_time field was initialized. */
1964 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1965 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1966 /* first_rtcp_ntp_time has been initialized for this stream,
1967 * copy the same value to all other uninitialized streams,
1968 * in order to map their timestamp origin to the same ntp time
1971 AVStream *st = NULL;
1972 if (rtsp_st->stream_index >= 0)
1973 st = s->streams[rtsp_st->stream_index];
1974 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1975 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1976 AVStream *st2 = NULL;
1977 if (rt->rtsp_streams[i]->stream_index >= 0)
1978 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1979 if (rtpctx2 && st && st2 &&
1980 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1981 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1982 rtpctx2->rtcp_ts_offset = av_rescale_q(
1983 rtpctx->rtcp_ts_offset, st->time_base,
1988 if (ret == -RTCP_BYE) {
1991 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1992 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1994 if (rt->nb_byes == rt->nb_rtsp_streams)
1998 } else if (rt->ts && CONFIG_RTPDEC) {
1999 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2002 rt->recvbuf_len = len;
2003 rt->recvbuf_pos = ret;
2004 rt->cur_transport_priv = rt->ts;
2011 return AVERROR_INVALIDDATA;
2017 /* more packets may follow, so we save the RTP context */
2018 rt->cur_transport_priv = rtsp_st->transport_priv;
2022 #endif /* CONFIG_RTPDEC */
2024 #if CONFIG_SDP_DEMUXER
2025 static int sdp_probe(AVProbeData *p1)
2027 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2029 /* we look for a line beginning "c=IN IP" */
2030 while (p < p_end && *p != '\0') {
2031 if (p + sizeof("c=IN IP") - 1 < p_end &&
2032 av_strstart(p, "c=IN IP", NULL))
2033 return AVPROBE_SCORE_EXTENSION;
2035 while (p < p_end - 1 && *p != '\n') p++;
2044 static int sdp_read_header(AVFormatContext *s)
2046 RTSPState *rt = s->priv_data;
2047 RTSPStream *rtsp_st;
2052 if (!ff_network_init())
2053 return AVERROR(EIO);
2055 if (s->max_delay < 0) /* Not set by the caller */
2056 s->max_delay = DEFAULT_REORDERING_DELAY;
2057 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2058 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2060 /* read the whole sdp file */
2061 /* XXX: better loading */
2062 content = av_malloc(SDP_MAX_SIZE);
2063 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2066 return AVERROR_INVALIDDATA;
2068 content[size] ='\0';
2070 err = ff_sdp_parse(s, content);
2074 /* open each RTP stream */
2075 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2077 rtsp_st = rt->rtsp_streams[i];
2079 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2080 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2081 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2082 ff_url_join(url, sizeof(url), "rtp", NULL,
2083 namebuf, rtsp_st->sdp_port,
2084 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2086 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2087 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2088 &s->interrupt_callback, NULL) < 0) {
2089 err = AVERROR_INVALIDDATA;
2093 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2098 ff_rtsp_close_streams(s);
2103 static int sdp_read_close(AVFormatContext *s)
2105 ff_rtsp_close_streams(s);
2110 static const AVClass sdp_demuxer_class = {
2111 .class_name = "SDP demuxer",
2112 .item_name = av_default_item_name,
2113 .option = sdp_options,
2114 .version = LIBAVUTIL_VERSION_INT,
2117 AVInputFormat ff_sdp_demuxer = {
2119 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2120 .priv_data_size = sizeof(RTSPState),
2121 .read_probe = sdp_probe,
2122 .read_header = sdp_read_header,
2123 .read_packet = ff_rtsp_fetch_packet,
2124 .read_close = sdp_read_close,
2125 .priv_class = &sdp_demuxer_class,
2127 #endif /* CONFIG_SDP_DEMUXER */
2129 #if CONFIG_RTP_DEMUXER
2130 static int rtp_probe(AVProbeData *p)
2132 if (av_strstart(p->filename, "rtp:", NULL))
2133 return AVPROBE_SCORE_MAX;
2137 static int rtp_read_header(AVFormatContext *s)
2139 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2140 char host[500], sdp[500];
2142 URLContext* in = NULL;
2144 AVCodecContext codec = { 0 };
2145 struct sockaddr_storage addr;
2147 socklen_t addrlen = sizeof(addr);
2148 RTSPState *rt = s->priv_data;
2150 if (!ff_network_init())
2151 return AVERROR(EIO);
2153 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2154 &s->interrupt_callback, NULL);
2159 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2160 if (ret == AVERROR(EAGAIN))
2165 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2169 if ((recvbuf[0] & 0xc0) != 0x80) {
2170 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2175 if (RTP_PT_IS_RTCP(recvbuf[1]))
2178 payload_type = recvbuf[1] & 0x7f;
2181 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2185 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2186 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2187 "without an SDP file describing it\n",
2191 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2192 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2193 "properly you need an SDP file "
2197 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2198 NULL, 0, s->filename);
2200 snprintf(sdp, sizeof(sdp),
2201 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2202 addr.ss_family == AF_INET ? 4 : 6, host,
2203 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2204 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2205 port, payload_type);
2206 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2208 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2211 /* sdp_read_header initializes this again */
2214 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2216 ret = sdp_read_header(s);
2227 static const AVClass rtp_demuxer_class = {
2228 .class_name = "RTP demuxer",
2229 .item_name = av_default_item_name,
2230 .option = rtp_options,
2231 .version = LIBAVUTIL_VERSION_INT,
2234 AVInputFormat ff_rtp_demuxer = {
2236 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2237 .priv_data_size = sizeof(RTSPState),
2238 .read_probe = rtp_probe,
2239 .read_header = rtp_read_header,
2240 .read_packet = ff_rtsp_fetch_packet,
2241 .read_close = sdp_read_close,
2242 .flags = AVFMT_NOFILE,
2243 .priv_class = &rtp_demuxer_class,
2245 #endif /* CONFIG_RTP_DEMUXER */