3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "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" }, \
68 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
70 #define RTSP_MEDIATYPE_OPTS(name, longname) \
71 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 #define RTSP_REORDERING_OPTS() \
77 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
79 const AVOption ff_rtsp_options[] = {
80 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
81 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
82 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
83 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
85 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
86 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
87 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
88 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
89 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
90 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
91 { "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 },
92 { "stimeout", "timeout (in micro seconds) of socket i/o operations.", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },
93 RTSP_REORDERING_OPTS(),
97 static const AVOption sdp_options[] = {
98 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
99 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
100 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
101 RTSP_REORDERING_OPTS(),
105 static const AVOption rtp_options[] = {
106 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
107 RTSP_REORDERING_OPTS(),
111 static void get_word_until_chars(char *buf, int buf_size,
112 const char *sep, const char **pp)
118 p += strspn(p, SPACE_CHARS);
120 while (!strchr(sep, *p) && *p != '\0') {
121 if ((q - buf) < buf_size - 1)
130 static void get_word_sep(char *buf, int buf_size, const char *sep,
133 if (**pp == '/') (*pp)++;
134 get_word_until_chars(buf, buf_size, sep, pp);
137 static void get_word(char *buf, int buf_size, const char **pp)
139 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
142 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
144 * Used for seeking in the rtp stream.
146 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
150 p += strspn(p, SPACE_CHARS);
151 if (!av_stristart(p, "npt=", &p))
154 *start = AV_NOPTS_VALUE;
155 *end = AV_NOPTS_VALUE;
157 get_word_sep(buf, sizeof(buf), "-", &p);
158 av_parse_time(start, buf, 1);
161 get_word_sep(buf, sizeof(buf), "-", &p);
162 av_parse_time(end, buf, 1);
166 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
168 struct addrinfo hints = { 0 }, *ai = NULL;
169 hints.ai_flags = AI_NUMERICHOST;
170 if (getaddrinfo(buf, NULL, &hints, &ai))
172 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
178 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
179 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) {
387 if (!rt->ts && CONFIG_RTPDEC)
388 rt->ts = ff_mpegts_parse_open(s);
390 RTPDynamicProtocolHandler *handler;
391 handler = ff_rtp_handler_find_by_id(
392 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
393 init_rtp_handler(handler, rtsp_st, NULL);
394 if (handler && handler->init)
395 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
397 } else if (rt->server_type == RTSP_SERVER_WMS &&
398 codec_type == AVMEDIA_TYPE_DATA) {
399 /* RTX stream, a stream that carries all the other actual
400 * audio/video streams. Don't expose this to the callers. */
402 st = avformat_new_stream(s, NULL);
405 st->id = rt->nb_rtsp_streams - 1;
406 rtsp_st->stream_index = st->index;
407 st->codec->codec_type = codec_type;
408 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
409 RTPDynamicProtocolHandler *handler;
410 /* if standard payload type, we can find the codec right now */
411 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
412 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
413 st->codec->sample_rate > 0)
414 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
415 /* Even static payload types may need a custom depacketizer */
416 handler = ff_rtp_handler_find_by_id(
417 rtsp_st->sdp_payload_type, st->codec->codec_type);
418 init_rtp_handler(handler, rtsp_st, st->codec);
419 if (handler && handler->init)
420 handler->init(s, st->index,
421 rtsp_st->dynamic_protocol_context);
424 /* put a default control url */
425 av_strlcpy(rtsp_st->control_url, rt->control_uri,
426 sizeof(rtsp_st->control_url));
429 if (av_strstart(p, "control:", &p)) {
430 if (s->nb_streams == 0) {
431 if (!strncmp(p, "rtsp://", 7))
432 av_strlcpy(rt->control_uri, p,
433 sizeof(rt->control_uri));
436 /* get the control url */
437 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
439 /* XXX: may need to add full url resolution */
440 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
442 if (proto[0] == '\0') {
443 /* relative control URL */
444 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
445 av_strlcat(rtsp_st->control_url, "/",
446 sizeof(rtsp_st->control_url));
447 av_strlcat(rtsp_st->control_url, p,
448 sizeof(rtsp_st->control_url));
450 av_strlcpy(rtsp_st->control_url, p,
451 sizeof(rtsp_st->control_url));
453 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
454 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
455 get_word(buf1, sizeof(buf1), &p);
456 payload_type = atoi(buf1);
457 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
458 if (rtsp_st->stream_index >= 0) {
459 st = s->streams[rtsp_st->stream_index];
460 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
462 } else if (av_strstart(p, "fmtp:", &p) ||
463 av_strstart(p, "framesize:", &p)) {
464 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
465 // let dynamic protocol handlers have a stab at the line.
466 get_word(buf1, sizeof(buf1), &p);
467 payload_type = atoi(buf1);
468 for (i = 0; i < rt->nb_rtsp_streams; i++) {
469 rtsp_st = rt->rtsp_streams[i];
470 if (rtsp_st->sdp_payload_type == payload_type &&
471 rtsp_st->dynamic_handler &&
472 rtsp_st->dynamic_handler->parse_sdp_a_line)
473 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
474 rtsp_st->dynamic_protocol_context, buf);
476 } else if (av_strstart(p, "range:", &p)) {
479 // this is so that seeking on a streamed file can work.
480 rtsp_parse_range_npt(p, &start, &end);
481 s->start_time = start;
482 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
483 s->duration = (end == AV_NOPTS_VALUE) ?
484 AV_NOPTS_VALUE : end - start;
485 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
487 rt->transport = RTSP_TRANSPORT_RDT;
488 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
490 st = s->streams[s->nb_streams - 1];
491 st->codec->sample_rate = atoi(p);
492 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
494 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
495 get_word(buf1, sizeof(buf1), &p); // ignore tag
496 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
497 p += strspn(p, SPACE_CHARS);
498 if (av_strstart(p, "inline:", &p))
499 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
501 if (rt->server_type == RTSP_SERVER_WMS)
502 ff_wms_parse_sdp_a_line(s, p);
503 if (s->nb_streams > 0) {
504 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
506 if (rt->server_type == RTSP_SERVER_REAL)
507 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
509 if (rtsp_st->dynamic_handler &&
510 rtsp_st->dynamic_handler->parse_sdp_a_line)
511 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
512 rtsp_st->stream_index,
513 rtsp_st->dynamic_protocol_context, buf);
520 int ff_sdp_parse(AVFormatContext *s, const char *content)
522 RTSPState *rt = s->priv_data;
525 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
526 * contain long SDP lines containing complete ASF Headers (several
527 * kB) or arrays of MDPR (RM stream descriptor) headers plus
528 * "rulebooks" describing their properties. Therefore, the SDP line
531 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
532 * in rtpdec_xiph.c. */
534 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
538 p += strspn(p, SPACE_CHARS);
546 /* get the content */
548 while (*p != '\n' && *p != '\r' && *p != '\0') {
549 if ((q - buf) < sizeof(buf) - 1)
554 sdp_parse_line(s, s1, letter, buf);
556 while (*p != '\n' && *p != '\0')
561 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
562 if (!rt->p) return AVERROR(ENOMEM);
565 #endif /* CONFIG_RTPDEC */
567 void ff_rtsp_undo_setup(AVFormatContext *s)
569 RTSPState *rt = s->priv_data;
572 for (i = 0; i < rt->nb_rtsp_streams; i++) {
573 RTSPStream *rtsp_st = rt->rtsp_streams[i];
576 if (rtsp_st->transport_priv) {
578 AVFormatContext *rtpctx = rtsp_st->transport_priv;
579 av_write_trailer(rtpctx);
580 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
582 avio_close_dyn_buf(rtpctx->pb, &ptr);
585 avio_close(rtpctx->pb);
587 avformat_free_context(rtpctx);
588 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
589 ff_rdt_parse_close(rtsp_st->transport_priv);
590 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
591 ff_rtp_parse_close(rtsp_st->transport_priv);
593 rtsp_st->transport_priv = NULL;
594 if (rtsp_st->rtp_handle)
595 ffurl_close(rtsp_st->rtp_handle);
596 rtsp_st->rtp_handle = NULL;
600 /* close and free RTSP streams */
601 void ff_rtsp_close_streams(AVFormatContext *s)
603 RTSPState *rt = s->priv_data;
607 ff_rtsp_undo_setup(s);
608 for (i = 0; i < rt->nb_rtsp_streams; i++) {
609 rtsp_st = rt->rtsp_streams[i];
611 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
612 rtsp_st->dynamic_handler->free(
613 rtsp_st->dynamic_protocol_context);
617 av_free(rt->rtsp_streams);
619 avformat_close_input(&rt->asf_ctx);
621 if (rt->ts && CONFIG_RTPDEC)
622 ff_mpegts_parse_close(rt->ts);
624 av_free(rt->recvbuf);
627 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
629 RTSPState *rt = s->priv_data;
631 int reordering_queue_size = rt->reordering_queue_size;
632 if (reordering_queue_size < 0) {
633 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
634 reordering_queue_size = 0;
636 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
639 /* open the RTP context */
640 if (rtsp_st->stream_index >= 0)
641 st = s->streams[rtsp_st->stream_index];
643 s->ctx_flags |= AVFMTCTX_NOHEADER;
645 if (s->oformat && CONFIG_RTSP_MUXER) {
646 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
648 RTSP_TCP_MAX_PACKET_SIZE,
649 rtsp_st->stream_index);
650 /* Ownership of rtp_handle is passed to the rtp mux context */
651 rtsp_st->rtp_handle = NULL;
654 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
655 return 0; // Don't need to open any parser here
656 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
657 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
658 rtsp_st->dynamic_protocol_context,
659 rtsp_st->dynamic_handler);
660 else if (CONFIG_RTPDEC)
661 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
662 rtsp_st->sdp_payload_type,
663 reordering_queue_size);
665 if (!rtsp_st->transport_priv) {
666 return AVERROR(ENOMEM);
667 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
668 if (rtsp_st->dynamic_handler) {
669 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
670 rtsp_st->dynamic_protocol_context,
671 rtsp_st->dynamic_handler);
673 if (rtsp_st->crypto_suite[0])
674 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
675 rtsp_st->crypto_suite,
676 rtsp_st->crypto_params);
682 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
683 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
690 q += strspn(q, SPACE_CHARS);
691 v = strtol(q, &p, 10);
695 v = strtol(p, &p, 10);
704 /* XXX: only one transport specification is parsed */
705 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
707 char transport_protocol[16];
709 char lower_transport[16];
711 RTSPTransportField *th;
714 reply->nb_transports = 0;
717 p += strspn(p, SPACE_CHARS);
721 th = &reply->transports[reply->nb_transports];
723 get_word_sep(transport_protocol, sizeof(transport_protocol),
725 if (!av_strcasecmp (transport_protocol, "rtp")) {
726 get_word_sep(profile, sizeof(profile), "/;,", &p);
727 lower_transport[0] = '\0';
728 /* rtp/avp/<protocol> */
730 get_word_sep(lower_transport, sizeof(lower_transport),
733 th->transport = RTSP_TRANSPORT_RTP;
734 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
735 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
736 /* x-pn-tng/<protocol> */
737 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
739 th->transport = RTSP_TRANSPORT_RDT;
740 } else if (!av_strcasecmp(transport_protocol, "raw")) {
741 get_word_sep(profile, sizeof(profile), "/;,", &p);
742 lower_transport[0] = '\0';
743 /* raw/raw/<protocol> */
745 get_word_sep(lower_transport, sizeof(lower_transport),
748 th->transport = RTSP_TRANSPORT_RAW;
750 if (!av_strcasecmp(lower_transport, "TCP"))
751 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
753 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
757 /* get each parameter */
758 while (*p != '\0' && *p != ',') {
759 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
760 if (!strcmp(parameter, "port")) {
763 rtsp_parse_range(&th->port_min, &th->port_max, &p);
765 } else if (!strcmp(parameter, "client_port")) {
768 rtsp_parse_range(&th->client_port_min,
769 &th->client_port_max, &p);
771 } else if (!strcmp(parameter, "server_port")) {
774 rtsp_parse_range(&th->server_port_min,
775 &th->server_port_max, &p);
777 } else if (!strcmp(parameter, "interleaved")) {
780 rtsp_parse_range(&th->interleaved_min,
781 &th->interleaved_max, &p);
783 } else if (!strcmp(parameter, "multicast")) {
784 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
785 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
786 } else if (!strcmp(parameter, "ttl")) {
790 th->ttl = strtol(p, &end, 10);
793 } else if (!strcmp(parameter, "destination")) {
796 get_word_sep(buf, sizeof(buf), ";,", &p);
797 get_sockaddr(buf, &th->destination);
799 } else if (!strcmp(parameter, "source")) {
802 get_word_sep(buf, sizeof(buf), ";,", &p);
803 av_strlcpy(th->source, buf, sizeof(th->source));
805 } else if (!strcmp(parameter, "mode")) {
808 get_word_sep(buf, sizeof(buf), ";, ", &p);
809 if (!strcmp(buf, "record") ||
810 !strcmp(buf, "receive"))
815 while (*p != ';' && *p != '\0' && *p != ',')
823 reply->nb_transports++;
827 static void handle_rtp_info(RTSPState *rt, const char *url,
828 uint32_t seq, uint32_t rtptime)
831 if (!rtptime || !url[0])
833 if (rt->transport != RTSP_TRANSPORT_RTP)
835 for (i = 0; i < rt->nb_rtsp_streams; i++) {
836 RTSPStream *rtsp_st = rt->rtsp_streams[i];
837 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
840 if (!strcmp(rtsp_st->control_url, url)) {
841 rtpctx->base_timestamp = rtptime;
847 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
850 char key[20], value[1024], url[1024] = "";
851 uint32_t seq = 0, rtptime = 0;
854 p += strspn(p, SPACE_CHARS);
857 get_word_sep(key, sizeof(key), "=", &p);
861 get_word_sep(value, sizeof(value), ";, ", &p);
863 if (!strcmp(key, "url"))
864 av_strlcpy(url, value, sizeof(url));
865 else if (!strcmp(key, "seq"))
866 seq = strtoul(value, NULL, 10);
867 else if (!strcmp(key, "rtptime"))
868 rtptime = strtoul(value, NULL, 10);
870 handle_rtp_info(rt, url, seq, rtptime);
879 handle_rtp_info(rt, url, seq, rtptime);
882 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
883 RTSPState *rt, const char *method)
887 /* NOTE: we do case independent match for broken servers */
889 if (av_stristart(p, "Session:", &p)) {
891 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
892 if (av_stristart(p, ";timeout=", &p) &&
893 (t = strtol(p, NULL, 10)) > 0) {
896 } else if (av_stristart(p, "Content-Length:", &p)) {
897 reply->content_length = strtol(p, NULL, 10);
898 } else if (av_stristart(p, "Transport:", &p)) {
899 rtsp_parse_transport(reply, p);
900 } else if (av_stristart(p, "CSeq:", &p)) {
901 reply->seq = strtol(p, NULL, 10);
902 } else if (av_stristart(p, "Range:", &p)) {
903 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
904 } else if (av_stristart(p, "RealChallenge1:", &p)) {
905 p += strspn(p, SPACE_CHARS);
906 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
907 } else if (av_stristart(p, "Server:", &p)) {
908 p += strspn(p, SPACE_CHARS);
909 av_strlcpy(reply->server, p, sizeof(reply->server));
910 } else if (av_stristart(p, "Notice:", &p) ||
911 av_stristart(p, "X-Notice:", &p)) {
912 reply->notice = strtol(p, NULL, 10);
913 } else if (av_stristart(p, "Location:", &p)) {
914 p += strspn(p, SPACE_CHARS);
915 av_strlcpy(reply->location, p , sizeof(reply->location));
916 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
917 p += strspn(p, SPACE_CHARS);
918 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
919 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
920 p += strspn(p, SPACE_CHARS);
921 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
922 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
923 p += strspn(p, SPACE_CHARS);
924 if (method && !strcmp(method, "DESCRIBE"))
925 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
926 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
927 p += strspn(p, SPACE_CHARS);
928 if (method && !strcmp(method, "PLAY"))
929 rtsp_parse_rtp_info(rt, p);
930 } else if (av_stristart(p, "Public:", &p) && rt) {
931 if (strstr(p, "GET_PARAMETER") &&
932 method && !strcmp(method, "OPTIONS"))
933 rt->get_parameter_supported = 1;
934 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
935 p += strspn(p, SPACE_CHARS);
936 rt->accept_dynamic_rate = atoi(p);
937 } else if (av_stristart(p, "Content-Type:", &p)) {
938 p += strspn(p, SPACE_CHARS);
939 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
943 /* skip a RTP/TCP interleaved packet */
944 void ff_rtsp_skip_packet(AVFormatContext *s)
946 RTSPState *rt = s->priv_data;
950 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
953 len = AV_RB16(buf + 1);
955 av_dlog(s, "skipping RTP packet len=%d\n", len);
960 if (len1 > sizeof(buf))
962 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
969 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
970 unsigned char **content_ptr,
971 int return_on_interleaved_data, const char *method)
973 RTSPState *rt = s->priv_data;
974 char buf[4096], buf1[1024], *q;
977 int ret, content_length, line_count = 0, request = 0;
978 unsigned char *content = NULL;
984 memset(reply, 0, sizeof(*reply));
986 /* parse reply (XXX: use buffers) */
987 rt->last_reply[0] = '\0';
991 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
992 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
998 /* XXX: only parse it if first char on line ? */
999 if (return_on_interleaved_data) {
1002 ff_rtsp_skip_packet(s);
1003 } else if (ch != '\r') {
1004 if ((q - buf) < sizeof(buf) - 1)
1010 av_dlog(s, "line='%s'\n", buf);
1012 /* test if last line */
1016 if (line_count == 0) {
1017 /* get reply code */
1018 get_word(buf1, sizeof(buf1), &p);
1019 if (!strncmp(buf1, "RTSP/", 5)) {
1020 get_word(buf1, sizeof(buf1), &p);
1021 reply->status_code = atoi(buf1);
1022 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1024 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1025 get_word(buf1, sizeof(buf1), &p); // object
1029 ff_rtsp_parse_line(reply, p, rt, method);
1030 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1031 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1036 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1037 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1039 content_length = reply->content_length;
1040 if (content_length > 0) {
1041 /* leave some room for a trailing '\0' (useful for simple parsing) */
1042 content = av_malloc(content_length + 1);
1043 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1044 content[content_length] = '\0';
1047 *content_ptr = content;
1053 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1054 const char* ptr = buf;
1056 if (!strcmp(reply->reason, "OPTIONS")) {
1057 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1059 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1060 if (reply->session_id[0])
1061 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1064 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1066 av_strlcat(buf, "\r\n", sizeof(buf));
1068 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1069 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1072 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1074 rt->last_cmd_time = av_gettime();
1075 /* Even if the request from the server had data, it is not the data
1076 * that the caller wants or expects. The memory could also be leaked
1077 * if the actual following reply has content data. */
1079 av_freep(content_ptr);
1080 /* If method is set, this is called from ff_rtsp_send_cmd,
1081 * where a reply to exactly this request is awaited. For
1082 * callers from within packet receiving, we just want to
1083 * return to the caller and go back to receiving packets. */
1089 if (rt->seq != reply->seq) {
1090 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1091 rt->seq, reply->seq);
1095 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1096 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1097 reply->notice == 2306 /* Continuous Feed Terminated */) {
1098 rt->state = RTSP_STATE_IDLE;
1099 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1100 return AVERROR(EIO); /* data or server error */
1101 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1102 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1103 return AVERROR(EPERM);
1109 * Send a command to the RTSP server without waiting for the reply.
1111 * @param s RTSP (de)muxer context
1112 * @param method the method for the request
1113 * @param url the target url for the request
1114 * @param headers extra header lines to include in the request
1115 * @param send_content if non-null, the data to send as request body content
1116 * @param send_content_length the length of the send_content data, or 0 if
1117 * send_content is null
1119 * @return zero if success, nonzero otherwise
1121 static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
1122 const char *method, const char *url,
1123 const char *headers,
1124 const unsigned char *send_content,
1125 int send_content_length)
1127 RTSPState *rt = s->priv_data;
1128 char buf[4096], *out_buf;
1129 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1131 /* Add in RTSP headers */
1134 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1136 av_strlcat(buf, headers, sizeof(buf));
1137 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1138 if (rt->session_id[0] != '\0' && (!headers ||
1139 !strstr(headers, "\nIf-Match:"))) {
1140 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1143 char *str = ff_http_auth_create_response(&rt->auth_state,
1144 rt->auth, url, method);
1146 av_strlcat(buf, str, sizeof(buf));
1149 if (send_content_length > 0 && send_content)
1150 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1151 av_strlcat(buf, "\r\n", sizeof(buf));
1153 /* base64 encode rtsp if tunneling */
1154 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1155 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1156 out_buf = base64buf;
1159 av_dlog(s, "Sending:\n%s--\n", buf);
1161 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1162 if (send_content_length > 0 && send_content) {
1163 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1164 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1165 "with content data not supported\n");
1166 return AVERROR_PATCHWELCOME;
1168 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1170 rt->last_cmd_time = av_gettime();
1175 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1176 const char *url, const char *headers)
1178 return rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1181 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1182 const char *headers, RTSPMessageHeader *reply,
1183 unsigned char **content_ptr)
1185 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1186 content_ptr, NULL, 0);
1189 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1190 const char *method, const char *url,
1192 RTSPMessageHeader *reply,
1193 unsigned char **content_ptr,
1194 const unsigned char *send_content,
1195 int send_content_length)
1197 RTSPState *rt = s->priv_data;
1198 HTTPAuthType cur_auth_type;
1199 int ret, attempts = 0;
1202 cur_auth_type = rt->auth_state.auth_type;
1203 if ((ret = rtsp_send_cmd_with_content_async(s, method, url, header,
1205 send_content_length)))
1208 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1212 if (reply->status_code == 401 &&
1213 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1214 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1217 if (reply->status_code > 400){
1218 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1222 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1228 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1229 int lower_transport, const char *real_challenge)
1231 RTSPState *rt = s->priv_data;
1232 int rtx = 0, j, i, err, interleave = 0, port_off;
1233 RTSPStream *rtsp_st;
1234 RTSPMessageHeader reply1, *reply = &reply1;
1236 const char *trans_pref;
1238 if (rt->transport == RTSP_TRANSPORT_RDT)
1239 trans_pref = "x-pn-tng";
1240 else if (rt->transport == RTSP_TRANSPORT_RAW)
1241 trans_pref = "RAW/RAW";
1243 trans_pref = "RTP/AVP";
1245 /* default timeout: 1 minute */
1248 /* Choose a random starting offset within the first half of the
1249 * port range, to allow for a number of ports to try even if the offset
1250 * happens to be at the end of the random range. */
1251 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1252 /* even random offset */
1253 port_off -= port_off & 0x01;
1255 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1256 char transport[2048];
1259 * WMS serves all UDP data over a single connection, the RTX, which
1260 * isn't necessarily the first in the SDP but has to be the first
1261 * to be set up, else the second/third SETUP will fail with a 461.
1263 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1264 rt->server_type == RTSP_SERVER_WMS) {
1267 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1268 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1270 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1274 if (rtx == rt->nb_rtsp_streams)
1275 return -1; /* no RTX found */
1276 rtsp_st = rt->rtsp_streams[rtx];
1278 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1280 rtsp_st = rt->rtsp_streams[i];
1283 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1286 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1287 port = reply->transports[0].client_port_min;
1291 /* first try in specified port range */
1292 while (j <= rt->rtp_port_max) {
1293 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1294 "?localport=%d", j);
1295 /* we will use two ports per rtp stream (rtp and rtcp) */
1297 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1298 &s->interrupt_callback, NULL))
1301 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1306 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1308 snprintf(transport, sizeof(transport) - 1,
1309 "%s/UDP;", trans_pref);
1310 if (rt->server_type != RTSP_SERVER_REAL)
1311 av_strlcat(transport, "unicast;", sizeof(transport));
1312 av_strlcatf(transport, sizeof(transport),
1313 "client_port=%d", port);
1314 if (rt->transport == RTSP_TRANSPORT_RTP &&
1315 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1316 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1320 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1321 /* For WMS streams, the application streams are only used for
1322 * UDP. When trying to set it up for TCP streams, the server
1323 * will return an error. Therefore, we skip those streams. */
1324 if (rt->server_type == RTSP_SERVER_WMS &&
1325 (rtsp_st->stream_index < 0 ||
1326 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1329 snprintf(transport, sizeof(transport) - 1,
1330 "%s/TCP;", trans_pref);
1331 if (rt->transport != RTSP_TRANSPORT_RDT)
1332 av_strlcat(transport, "unicast;", sizeof(transport));
1333 av_strlcatf(transport, sizeof(transport),
1334 "interleaved=%d-%d",
1335 interleave, interleave + 1);
1339 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1340 snprintf(transport, sizeof(transport) - 1,
1341 "%s/UDP;multicast", trans_pref);
1344 av_strlcat(transport, ";mode=record", sizeof(transport));
1345 } else if (rt->server_type == RTSP_SERVER_REAL ||
1346 rt->server_type == RTSP_SERVER_WMS)
1347 av_strlcat(transport, ";mode=play", sizeof(transport));
1348 snprintf(cmd, sizeof(cmd),
1349 "Transport: %s\r\n",
1351 if (rt->accept_dynamic_rate)
1352 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1353 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1354 char real_res[41], real_csum[9];
1355 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1357 av_strlcatf(cmd, sizeof(cmd),
1359 "RealChallenge2: %s, sd=%s\r\n",
1360 rt->session_id, real_res, real_csum);
1362 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1363 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1366 } else if (reply->status_code != RTSP_STATUS_OK ||
1367 reply->nb_transports != 1) {
1368 err = AVERROR_INVALIDDATA;
1372 /* XXX: same protocol for all streams is required */
1374 if (reply->transports[0].lower_transport != rt->lower_transport ||
1375 reply->transports[0].transport != rt->transport) {
1376 err = AVERROR_INVALIDDATA;
1380 rt->lower_transport = reply->transports[0].lower_transport;
1381 rt->transport = reply->transports[0].transport;
1384 /* Fail if the server responded with another lower transport mode
1385 * than what we requested. */
1386 if (reply->transports[0].lower_transport != lower_transport) {
1387 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1388 err = AVERROR_INVALIDDATA;
1392 switch(reply->transports[0].lower_transport) {
1393 case RTSP_LOWER_TRANSPORT_TCP:
1394 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1395 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1398 case RTSP_LOWER_TRANSPORT_UDP: {
1399 char url[1024], options[30] = "";
1401 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1402 av_strlcpy(options, "?connect=1", sizeof(options));
1403 /* Use source address if specified */
1404 if (reply->transports[0].source[0]) {
1405 ff_url_join(url, sizeof(url), "rtp", NULL,
1406 reply->transports[0].source,
1407 reply->transports[0].server_port_min, "%s", options);
1409 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1410 reply->transports[0].server_port_min, "%s", options);
1412 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1413 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1414 err = AVERROR_INVALIDDATA;
1417 /* Try to initialize the connection state in a
1418 * potential NAT router by sending dummy packets.
1419 * RTP/RTCP dummy packets are used for RDT, too.
1421 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1423 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1426 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1427 char url[1024], namebuf[50], optbuf[20] = "";
1428 struct sockaddr_storage addr;
1431 if (reply->transports[0].destination.ss_family) {
1432 addr = reply->transports[0].destination;
1433 port = reply->transports[0].port_min;
1434 ttl = reply->transports[0].ttl;
1436 addr = rtsp_st->sdp_ip;
1437 port = rtsp_st->sdp_port;
1438 ttl = rtsp_st->sdp_ttl;
1441 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1442 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1443 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1444 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1445 port, "%s", optbuf);
1446 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1447 &s->interrupt_callback, NULL) < 0) {
1448 err = AVERROR_INVALIDDATA;
1455 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1459 if (rt->nb_rtsp_streams && reply->timeout > 0)
1460 rt->timeout = reply->timeout;
1462 if (rt->server_type == RTSP_SERVER_REAL)
1463 rt->need_subscription = 1;
1468 ff_rtsp_undo_setup(s);
1472 void ff_rtsp_close_connections(AVFormatContext *s)
1474 RTSPState *rt = s->priv_data;
1475 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1476 ffurl_close(rt->rtsp_hd);
1477 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1480 int ff_rtsp_connect(AVFormatContext *s)
1482 RTSPState *rt = s->priv_data;
1483 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1484 int port, err, tcp_fd;
1485 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1486 int lower_transport_mask = 0;
1487 char real_challenge[64] = "";
1488 struct sockaddr_storage peer;
1489 socklen_t peer_len = sizeof(peer);
1491 if (rt->rtp_port_max < rt->rtp_port_min) {
1492 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1493 "than min port %d\n", rt->rtp_port_max,
1495 return AVERROR(EINVAL);
1498 if (!ff_network_init())
1499 return AVERROR(EIO);
1501 if (s->max_delay < 0) /* Not set by the caller */
1502 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1504 rt->control_transport = RTSP_MODE_PLAIN;
1505 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1506 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1507 rt->control_transport = RTSP_MODE_TUNNEL;
1509 /* Only pass through valid flags from here */
1510 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1513 lower_transport_mask = rt->lower_transport_mask;
1514 /* extract hostname and port */
1515 av_url_split(NULL, 0, auth, sizeof(auth),
1516 host, sizeof(host), &port, path, sizeof(path), s->filename);
1518 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1521 port = RTSP_DEFAULT_PORT;
1523 if (!lower_transport_mask)
1524 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1527 /* Only UDP or TCP - UDP multicast isn't supported. */
1528 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1529 (1 << RTSP_LOWER_TRANSPORT_TCP);
1530 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1531 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1532 "only UDP and TCP are supported for output.\n");
1533 err = AVERROR(EINVAL);
1538 /* Construct the URI used in request; this is similar to s->filename,
1539 * but with authentication credentials removed and RTSP specific options
1541 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1542 host, port, "%s", path);
1544 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1545 /* set up initial handshake for tunneling */
1546 char httpname[1024];
1547 char sessioncookie[17];
1550 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1551 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1552 av_get_random_seed(), av_get_random_seed());
1555 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1556 &s->interrupt_callback) < 0) {
1561 /* generate GET headers */
1562 snprintf(headers, sizeof(headers),
1563 "x-sessioncookie: %s\r\n"
1564 "Accept: application/x-rtsp-tunnelled\r\n"
1565 "Pragma: no-cache\r\n"
1566 "Cache-Control: no-cache\r\n",
1568 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1570 /* complete the connection */
1571 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1577 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1578 &s->interrupt_callback) < 0 ) {
1583 /* generate POST headers */
1584 snprintf(headers, sizeof(headers),
1585 "x-sessioncookie: %s\r\n"
1586 "Content-Type: application/x-rtsp-tunnelled\r\n"
1587 "Pragma: no-cache\r\n"
1588 "Cache-Control: no-cache\r\n"
1589 "Content-Length: 32767\r\n"
1590 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1592 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1593 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1595 /* Initialize the authentication state for the POST session. The HTTP
1596 * protocol implementation doesn't properly handle multi-pass
1597 * authentication for POST requests, since it would require one of
1599 * - implementing Expect: 100-continue, which many HTTP servers
1600 * don't support anyway, even less the RTSP servers that do HTTP
1602 * - sending the whole POST data until getting a 401 reply specifying
1603 * what authentication method to use, then resending all that data
1604 * - waiting for potential 401 replies directly after sending the
1605 * POST header (waiting for some unspecified time)
1606 * Therefore, we copy the full auth state, which works for both basic
1607 * and digest. (For digest, we would have to synchronize the nonce
1608 * count variable between the two sessions, if we'd do more requests
1609 * with the original session, though.)
1611 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1613 /* complete the connection */
1614 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1619 /* open the tcp connection */
1620 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
1621 "?timeout=%d", rt->stimeout);
1622 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1623 &s->interrupt_callback, NULL) < 0) {
1627 rt->rtsp_hd_out = rt->rtsp_hd;
1631 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1632 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1633 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1634 NULL, 0, NI_NUMERICHOST);
1637 /* request options supported by the server; this also detects server
1639 for (rt->server_type = RTSP_SERVER_RTP;;) {
1641 if (rt->server_type == RTSP_SERVER_REAL)
1644 * The following entries are required for proper
1645 * streaming from a Realmedia server. They are
1646 * interdependent in some way although we currently
1647 * don't quite understand how. Values were copied
1648 * from mplayer SVN r23589.
1649 * ClientChallenge is a 16-byte ID in hex
1650 * CompanyID is a 16-byte ID in base64
1652 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1653 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1654 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1655 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1657 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1658 if (reply->status_code != RTSP_STATUS_OK) {
1659 err = AVERROR_INVALIDDATA;
1663 /* detect server type if not standard-compliant RTP */
1664 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1665 rt->server_type = RTSP_SERVER_REAL;
1667 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1668 rt->server_type = RTSP_SERVER_WMS;
1669 } else if (rt->server_type == RTSP_SERVER_REAL)
1670 strcpy(real_challenge, reply->real_challenge);
1674 if (s->iformat && CONFIG_RTSP_DEMUXER)
1675 err = ff_rtsp_setup_input_streams(s, reply);
1676 else if (CONFIG_RTSP_MUXER)
1677 err = ff_rtsp_setup_output_streams(s, host);
1682 int lower_transport = ff_log2_tab[lower_transport_mask &
1683 ~(lower_transport_mask - 1)];
1685 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1686 rt->server_type == RTSP_SERVER_REAL ?
1687 real_challenge : NULL);
1690 lower_transport_mask &= ~(1 << lower_transport);
1691 if (lower_transport_mask == 0 && err == 1) {
1692 err = AVERROR(EPROTONOSUPPORT);
1697 rt->lower_transport_mask = lower_transport_mask;
1698 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1699 rt->state = RTSP_STATE_IDLE;
1700 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1703 ff_rtsp_close_streams(s);
1704 ff_rtsp_close_connections(s);
1705 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1706 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1707 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1715 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1718 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1719 uint8_t *buf, int buf_size, int64_t wait_end)
1721 RTSPState *rt = s->priv_data;
1722 RTSPStream *rtsp_st;
1723 int n, i, ret, tcp_fd, timeout_cnt = 0;
1725 struct pollfd *p = rt->p;
1726 int *fds = NULL, fdsnum, fdsidx;
1729 if (ff_check_interrupt(&s->interrupt_callback))
1730 return AVERROR_EXIT;
1731 if (wait_end && wait_end - av_gettime() < 0)
1732 return AVERROR(EAGAIN);
1735 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1736 p[max_p].fd = tcp_fd;
1737 p[max_p++].events = POLLIN;
1741 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1742 rtsp_st = rt->rtsp_streams[i];
1743 if (rtsp_st->rtp_handle) {
1744 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1746 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1750 av_log(s, AV_LOG_ERROR,
1751 "Number of fds %d not supported\n", fdsnum);
1752 return AVERROR_INVALIDDATA;
1754 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1755 p[max_p].fd = fds[fdsidx];
1756 p[max_p++].events = POLLIN;
1761 n = poll(p, max_p, POLL_TIMEOUT_MS);
1763 int j = 1 - (tcp_fd == -1);
1765 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1766 rtsp_st = rt->rtsp_streams[i];
1767 if (rtsp_st->rtp_handle) {
1768 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1769 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1771 *prtsp_st = rtsp_st;
1778 #if CONFIG_RTSP_DEMUXER
1779 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1780 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1781 if (rt->state == RTSP_STATE_STREAMING) {
1782 if (!ff_rtsp_parse_streaming_commands(s))
1785 av_log(s, AV_LOG_WARNING,
1786 "Unable to answer to TEARDOWN\n");
1790 RTSPMessageHeader reply;
1791 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1794 /* XXX: parse message */
1795 if (rt->state != RTSP_STATE_STREAMING)
1800 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1801 return AVERROR(ETIMEDOUT);
1802 } else if (n < 0 && errno != EINTR)
1803 return AVERROR(errno);
1807 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1808 const uint8_t *buf, int len)
1810 RTSPState *rt = s->priv_data;
1814 if (rt->nb_rtsp_streams == 1) {
1815 *rtsp_st = rt->rtsp_streams[0];
1818 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1819 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1821 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1822 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1825 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1826 *rtsp_st = rt->rtsp_streams[i];
1833 av_log(s, AV_LOG_WARNING,
1834 "Unable to pick stream for packet - SSRC not known for "
1836 return AVERROR(EAGAIN);
1839 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1840 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1841 *rtsp_st = rt->rtsp_streams[i];
1847 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1848 return AVERROR(EAGAIN);
1851 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1853 RTSPState *rt = s->priv_data;
1855 RTSPStream *rtsp_st, *first_queue_st = NULL;
1856 int64_t wait_end = 0;
1858 if (rt->nb_byes == rt->nb_rtsp_streams)
1861 /* get next frames from the same RTP packet */
1862 if (rt->cur_transport_priv) {
1863 if (rt->transport == RTSP_TRANSPORT_RDT) {
1864 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1865 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1866 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1867 } else if (rt->ts && CONFIG_RTPDEC) {
1868 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1870 rt->recvbuf_pos += ret;
1871 ret = rt->recvbuf_pos < rt->recvbuf_len;
1876 rt->cur_transport_priv = NULL;
1878 } else if (ret == 1) {
1881 rt->cur_transport_priv = NULL;
1885 if (rt->transport == RTSP_TRANSPORT_RTP) {
1887 int64_t first_queue_time = 0;
1888 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1889 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1893 queue_time = ff_rtp_queued_packet_time(rtpctx);
1894 if (queue_time && (queue_time - first_queue_time < 0 ||
1895 !first_queue_time)) {
1896 first_queue_time = queue_time;
1897 first_queue_st = rt->rtsp_streams[i];
1900 if (first_queue_time) {
1901 wait_end = first_queue_time + s->max_delay;
1904 first_queue_st = NULL;
1908 /* read next RTP packet */
1910 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1912 return AVERROR(ENOMEM);
1915 switch(rt->lower_transport) {
1917 #if CONFIG_RTSP_DEMUXER
1918 case RTSP_LOWER_TRANSPORT_TCP:
1919 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1922 case RTSP_LOWER_TRANSPORT_UDP:
1923 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1924 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1925 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1926 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1928 case RTSP_LOWER_TRANSPORT_CUSTOM:
1929 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1930 wait_end && wait_end < av_gettime())
1931 len = AVERROR(EAGAIN);
1933 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1934 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1935 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1936 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1939 if (len == AVERROR(EAGAIN) && first_queue_st &&
1940 rt->transport == RTSP_TRANSPORT_RTP) {
1941 rtsp_st = first_queue_st;
1942 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1949 if (rt->transport == RTSP_TRANSPORT_RDT) {
1950 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1951 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1952 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1953 if (rtsp_st->feedback) {
1954 AVIOContext *pb = NULL;
1955 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1957 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1960 /* Either bad packet, or a RTCP packet. Check if the
1961 * first_rtcp_ntp_time field was initialized. */
1962 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1963 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1964 /* first_rtcp_ntp_time has been initialized for this stream,
1965 * copy the same value to all other uninitialized streams,
1966 * in order to map their timestamp origin to the same ntp time
1969 AVStream *st = NULL;
1970 if (rtsp_st->stream_index >= 0)
1971 st = s->streams[rtsp_st->stream_index];
1972 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1974 AVStream *st2 = NULL;
1975 if (rt->rtsp_streams[i]->stream_index >= 0)
1976 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1977 if (rtpctx2 && st && st2 &&
1978 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1979 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1980 rtpctx2->rtcp_ts_offset = av_rescale_q(
1981 rtpctx->rtcp_ts_offset, st->time_base,
1986 if (ret == -RTCP_BYE) {
1989 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1990 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1992 if (rt->nb_byes == rt->nb_rtsp_streams)
1996 } else if (rt->ts && CONFIG_RTPDEC) {
1997 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2000 rt->recvbuf_len = len;
2001 rt->recvbuf_pos = ret;
2002 rt->cur_transport_priv = rt->ts;
2009 return AVERROR_INVALIDDATA;
2015 /* more packets may follow, so we save the RTP context */
2016 rt->cur_transport_priv = rtsp_st->transport_priv;
2020 #endif /* CONFIG_RTPDEC */
2022 #if CONFIG_SDP_DEMUXER
2023 static int sdp_probe(AVProbeData *p1)
2025 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2027 /* we look for a line beginning "c=IN IP" */
2028 while (p < p_end && *p != '\0') {
2029 if (p + sizeof("c=IN IP") - 1 < p_end &&
2030 av_strstart(p, "c=IN IP", NULL))
2031 return AVPROBE_SCORE_EXTENSION;
2033 while (p < p_end - 1 && *p != '\n') p++;
2042 static int sdp_read_header(AVFormatContext *s)
2044 RTSPState *rt = s->priv_data;
2045 RTSPStream *rtsp_st;
2050 if (!ff_network_init())
2051 return AVERROR(EIO);
2053 if (s->max_delay < 0) /* Not set by the caller */
2054 s->max_delay = DEFAULT_REORDERING_DELAY;
2055 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2056 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2058 /* read the whole sdp file */
2059 /* XXX: better loading */
2060 content = av_malloc(SDP_MAX_SIZE);
2061 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2064 return AVERROR_INVALIDDATA;
2066 content[size] ='\0';
2068 err = ff_sdp_parse(s, content);
2072 /* open each RTP stream */
2073 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2075 rtsp_st = rt->rtsp_streams[i];
2077 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2078 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2079 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2080 ff_url_join(url, sizeof(url), "rtp", NULL,
2081 namebuf, rtsp_st->sdp_port,
2082 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2084 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2085 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2086 &s->interrupt_callback, NULL) < 0) {
2087 err = AVERROR_INVALIDDATA;
2091 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2096 ff_rtsp_close_streams(s);
2101 static int sdp_read_close(AVFormatContext *s)
2103 ff_rtsp_close_streams(s);
2108 static const AVClass sdp_demuxer_class = {
2109 .class_name = "SDP demuxer",
2110 .item_name = av_default_item_name,
2111 .option = sdp_options,
2112 .version = LIBAVUTIL_VERSION_INT,
2115 AVInputFormat ff_sdp_demuxer = {
2117 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2118 .priv_data_size = sizeof(RTSPState),
2119 .read_probe = sdp_probe,
2120 .read_header = sdp_read_header,
2121 .read_packet = ff_rtsp_fetch_packet,
2122 .read_close = sdp_read_close,
2123 .priv_class = &sdp_demuxer_class,
2125 #endif /* CONFIG_SDP_DEMUXER */
2127 #if CONFIG_RTP_DEMUXER
2128 static int rtp_probe(AVProbeData *p)
2130 if (av_strstart(p->filename, "rtp:", NULL))
2131 return AVPROBE_SCORE_MAX;
2135 static int rtp_read_header(AVFormatContext *s)
2137 uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];
2138 char host[500], sdp[500];
2140 URLContext* in = NULL;
2142 AVCodecContext codec = { 0 };
2143 struct sockaddr_storage addr;
2145 socklen_t addrlen = sizeof(addr);
2146 RTSPState *rt = s->priv_data;
2148 if (!ff_network_init())
2149 return AVERROR(EIO);
2151 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2152 &s->interrupt_callback, NULL);
2157 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2158 if (ret == AVERROR(EAGAIN))
2163 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2167 if ((recvbuf[0] & 0xc0) != 0x80) {
2168 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2173 if (RTP_PT_IS_RTCP(recvbuf[1]))
2176 payload_type = recvbuf[1] & 0x7f;
2179 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2183 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2184 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2185 "without an SDP file describing it\n",
2189 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2190 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2191 "properly you need an SDP file "
2195 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2196 NULL, 0, s->filename);
2198 snprintf(sdp, sizeof(sdp),
2199 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2200 addr.ss_family == AF_INET ? 4 : 6, host,
2201 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2202 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2203 port, payload_type);
2204 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2206 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2209 /* sdp_read_header initializes this again */
2212 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2214 ret = sdp_read_header(s);
2225 static const AVClass rtp_demuxer_class = {
2226 .class_name = "RTP demuxer",
2227 .item_name = av_default_item_name,
2228 .option = rtp_options,
2229 .version = LIBAVUTIL_VERSION_INT,
2232 AVInputFormat ff_rtp_demuxer = {
2234 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2235 .priv_data_size = sizeof(RTSPState),
2236 .read_probe = rtp_probe,
2237 .read_header = rtp_read_header,
2238 .read_packet = ff_rtsp_fetch_packet,
2239 .read_close = sdp_read_close,
2240 .flags = AVFMT_NOFILE,
2241 .priv_class = &rtp_demuxer_class,
2243 #endif /* CONFIG_RTP_DEMUXER */