3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
70 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
74 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
78 #define RTSP_REORDERING_OPTS() \
79 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
81 const AVOption ff_rtsp_options[] = {
82 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
94 RTSP_REORDERING_OPTS(),
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 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)
183 codec->codec_id = handler->codec_id;
184 rtsp_st->dynamic_handler = handler;
185 if (handler->alloc) {
186 rtsp_st->dynamic_protocol_context = handler->alloc();
187 if (!rtsp_st->dynamic_protocol_context)
188 rtsp_st->dynamic_handler = NULL;
192 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
193 static int sdp_parse_rtpmap(AVFormatContext *s,
194 AVStream *st, RTSPStream *rtsp_st,
195 int payload_type, const char *p)
197 AVCodecContext *codec = st->codec;
203 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
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 /* search into AVRtpPayloadTypes[] */
213 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
216 if (codec->codec_id == AV_CODEC_ID_NONE) {
217 RTPDynamicProtocolHandler *handler =
218 ff_rtp_handler_find_by_name(buf, codec->codec_type);
219 init_rtp_handler(handler, rtsp_st, codec);
220 /* If no dynamic handler was found, check with the list of standard
221 * allocated types, if such a stream for some reason happens to
222 * use a private payload type. This isn't handled in rtpdec.c, since
223 * the format name from the rtpmap line never is passed into rtpdec. */
224 if (!rtsp_st->dynamic_handler)
225 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
228 c = avcodec_find_decoder(codec->codec_id);
234 get_word_sep(buf, sizeof(buf), "/", &p);
236 switch (codec->codec_type) {
237 case AVMEDIA_TYPE_AUDIO:
238 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
239 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
240 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
242 codec->sample_rate = i;
243 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
244 get_word_sep(buf, sizeof(buf), "/", &p);
248 // TODO: there is a bug here; if it is a mono stream, and
249 // less than 22000Hz, faad upconverts to stereo and twice
250 // the frequency. No problem, but the sample rate is being
251 // set here by the sdp line. Patch on its way. (rdm)
253 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
255 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
258 case AVMEDIA_TYPE_VIDEO:
259 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
261 avpriv_set_pts_info(st, 32, 1, i);
266 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
267 rtsp_st->dynamic_handler->init(s, st->index,
268 rtsp_st->dynamic_protocol_context);
272 /* parse the attribute line from the fmtp a line of an sdp response. This
273 * is broken out as a function because it is used in rtp_h264.c, which is
275 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
276 char *value, int value_size)
278 *p += strspn(*p, SPACE_CHARS);
280 get_word_sep(attr, attr_size, "=", p);
283 get_word_sep(value, value_size, ";", p);
291 typedef struct SDPParseState {
293 struct sockaddr_storage default_ip;
295 int skip_media; ///< set if an unknown m= line occurs
298 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
299 int letter, const char *buf)
301 RTSPState *rt = s->priv_data;
302 char buf1[64], st_type[64];
304 enum AVMediaType codec_type;
308 struct sockaddr_storage sdp_ip;
311 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
314 if (s1->skip_media && letter != 'm')
318 get_word(buf1, sizeof(buf1), &p);
319 if (strcmp(buf1, "IN") != 0)
321 get_word(buf1, sizeof(buf1), &p);
322 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
324 get_word_sep(buf1, sizeof(buf1), "/", &p);
325 if (get_sockaddr(buf1, &sdp_ip))
330 get_word_sep(buf1, sizeof(buf1), "/", &p);
333 if (s->nb_streams == 0) {
334 s1->default_ip = sdp_ip;
335 s1->default_ttl = ttl;
337 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
338 rtsp_st->sdp_ip = sdp_ip;
339 rtsp_st->sdp_ttl = ttl;
343 av_dict_set(&s->metadata, "title", p, 0);
346 if (s->nb_streams == 0) {
347 av_dict_set(&s->metadata, "comment", p, 0);
354 codec_type = AVMEDIA_TYPE_UNKNOWN;
355 get_word(st_type, sizeof(st_type), &p);
356 if (!strcmp(st_type, "audio")) {
357 codec_type = AVMEDIA_TYPE_AUDIO;
358 } else if (!strcmp(st_type, "video")) {
359 codec_type = AVMEDIA_TYPE_VIDEO;
360 } else if (!strcmp(st_type, "application")) {
361 codec_type = AVMEDIA_TYPE_DATA;
363 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
367 rtsp_st = av_mallocz(sizeof(RTSPStream));
370 rtsp_st->stream_index = -1;
371 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
373 rtsp_st->sdp_ip = s1->default_ip;
374 rtsp_st->sdp_ttl = s1->default_ttl;
376 get_word(buf1, sizeof(buf1), &p); /* port */
377 rtsp_st->sdp_port = atoi(buf1);
379 get_word(buf1, sizeof(buf1), &p); /* protocol */
380 if (!strcmp(buf1, "udp"))
381 rt->transport = RTSP_TRANSPORT_RAW;
383 /* XXX: handle list of formats */
384 get_word(buf1, sizeof(buf1), &p); /* format list */
385 rtsp_st->sdp_payload_type = atoi(buf1);
387 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
388 /* no corresponding stream */
389 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
390 rt->ts = ff_mpegts_parse_open(s);
391 } else if (rt->server_type == RTSP_SERVER_WMS &&
392 codec_type == AVMEDIA_TYPE_DATA) {
393 /* RTX stream, a stream that carries all the other actual
394 * audio/video streams. Don't expose this to the callers. */
396 st = avformat_new_stream(s, NULL);
399 st->id = rt->nb_rtsp_streams - 1;
400 rtsp_st->stream_index = st->index;
401 st->codec->codec_type = codec_type;
402 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
403 RTPDynamicProtocolHandler *handler;
404 /* if standard payload type, we can find the codec right now */
405 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
406 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
407 st->codec->sample_rate > 0)
408 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
409 /* Even static payload types may need a custom depacketizer */
410 handler = ff_rtp_handler_find_by_id(
411 rtsp_st->sdp_payload_type, st->codec->codec_type);
412 init_rtp_handler(handler, rtsp_st, st->codec);
413 if (handler && handler->init)
414 handler->init(s, st->index,
415 rtsp_st->dynamic_protocol_context);
418 /* put a default control url */
419 av_strlcpy(rtsp_st->control_url, rt->control_uri,
420 sizeof(rtsp_st->control_url));
423 if (av_strstart(p, "control:", &p)) {
424 if (s->nb_streams == 0) {
425 if (!strncmp(p, "rtsp://", 7))
426 av_strlcpy(rt->control_uri, p,
427 sizeof(rt->control_uri));
430 /* get the control url */
431 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433 /* XXX: may need to add full url resolution */
434 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
436 if (proto[0] == '\0') {
437 /* relative control URL */
438 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
439 av_strlcat(rtsp_st->control_url, "/",
440 sizeof(rtsp_st->control_url));
441 av_strlcat(rtsp_st->control_url, p,
442 sizeof(rtsp_st->control_url));
444 av_strlcpy(rtsp_st->control_url, p,
445 sizeof(rtsp_st->control_url));
447 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
448 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
449 get_word(buf1, sizeof(buf1), &p);
450 payload_type = atoi(buf1);
451 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
452 if (rtsp_st->stream_index >= 0) {
453 st = s->streams[rtsp_st->stream_index];
454 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
456 } else if (av_strstart(p, "fmtp:", &p) ||
457 av_strstart(p, "framesize:", &p)) {
458 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
459 // let dynamic protocol handlers have a stab at the line.
460 get_word(buf1, sizeof(buf1), &p);
461 payload_type = atoi(buf1);
462 for (i = 0; i < rt->nb_rtsp_streams; i++) {
463 rtsp_st = rt->rtsp_streams[i];
464 if (rtsp_st->sdp_payload_type == payload_type &&
465 rtsp_st->dynamic_handler &&
466 rtsp_st->dynamic_handler->parse_sdp_a_line)
467 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
468 rtsp_st->dynamic_protocol_context, buf);
470 } else if (av_strstart(p, "range:", &p)) {
473 // this is so that seeking on a streamed file can work.
474 rtsp_parse_range_npt(p, &start, &end);
475 s->start_time = start;
476 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
477 s->duration = (end == AV_NOPTS_VALUE) ?
478 AV_NOPTS_VALUE : end - start;
479 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
481 rt->transport = RTSP_TRANSPORT_RDT;
482 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
484 st = s->streams[s->nb_streams - 1];
485 st->codec->sample_rate = atoi(p);
487 if (rt->server_type == RTSP_SERVER_WMS)
488 ff_wms_parse_sdp_a_line(s, p);
489 if (s->nb_streams > 0) {
490 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
492 if (rt->server_type == RTSP_SERVER_REAL)
493 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
495 if (rtsp_st->dynamic_handler &&
496 rtsp_st->dynamic_handler->parse_sdp_a_line)
497 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
498 rtsp_st->stream_index,
499 rtsp_st->dynamic_protocol_context, buf);
506 int ff_sdp_parse(AVFormatContext *s, const char *content)
508 RTSPState *rt = s->priv_data;
511 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
512 * contain long SDP lines containing complete ASF Headers (several
513 * kB) or arrays of MDPR (RM stream descriptor) headers plus
514 * "rulebooks" describing their properties. Therefore, the SDP line
517 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
518 * in rtpdec_xiph.c. */
520 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
524 p += strspn(p, SPACE_CHARS);
532 /* get the content */
534 while (*p != '\n' && *p != '\r' && *p != '\0') {
535 if ((q - buf) < sizeof(buf) - 1)
540 sdp_parse_line(s, s1, letter, buf);
542 while (*p != '\n' && *p != '\0')
547 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
548 if (!rt->p) return AVERROR(ENOMEM);
551 #endif /* CONFIG_RTPDEC */
553 void ff_rtsp_undo_setup(AVFormatContext *s)
555 RTSPState *rt = s->priv_data;
558 for (i = 0; i < rt->nb_rtsp_streams; i++) {
559 RTSPStream *rtsp_st = rt->rtsp_streams[i];
562 if (rtsp_st->transport_priv) {
564 AVFormatContext *rtpctx = rtsp_st->transport_priv;
565 av_write_trailer(rtpctx);
566 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
568 avio_close_dyn_buf(rtpctx->pb, &ptr);
571 avio_close(rtpctx->pb);
573 avformat_free_context(rtpctx);
574 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
575 ff_rdt_parse_close(rtsp_st->transport_priv);
576 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
577 ff_rtp_parse_close(rtsp_st->transport_priv);
579 rtsp_st->transport_priv = NULL;
580 if (rtsp_st->rtp_handle)
581 ffurl_close(rtsp_st->rtp_handle);
582 rtsp_st->rtp_handle = NULL;
586 /* close and free RTSP streams */
587 void ff_rtsp_close_streams(AVFormatContext *s)
589 RTSPState *rt = s->priv_data;
593 ff_rtsp_undo_setup(s);
594 for (i = 0; i < rt->nb_rtsp_streams; i++) {
595 rtsp_st = rt->rtsp_streams[i];
597 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
598 rtsp_st->dynamic_handler->free(
599 rtsp_st->dynamic_protocol_context);
603 av_free(rt->rtsp_streams);
605 avformat_close_input(&rt->asf_ctx);
607 if (rt->ts && CONFIG_RTPDEC)
608 ff_mpegts_parse_close(rt->ts);
610 av_free(rt->recvbuf);
613 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
615 RTSPState *rt = s->priv_data;
617 int reordering_queue_size = rt->reordering_queue_size;
618 if (reordering_queue_size < 0) {
619 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
620 reordering_queue_size = 0;
622 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
625 /* open the RTP context */
626 if (rtsp_st->stream_index >= 0)
627 st = s->streams[rtsp_st->stream_index];
629 s->ctx_flags |= AVFMTCTX_NOHEADER;
631 if (s->oformat && CONFIG_RTSP_MUXER) {
632 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
634 RTSP_TCP_MAX_PACKET_SIZE,
635 rtsp_st->stream_index);
636 /* Ownership of rtp_handle is passed to the rtp mux context */
637 rtsp_st->rtp_handle = NULL;
640 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
641 return 0; // Don't need to open any parser here
642 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
643 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
644 rtsp_st->dynamic_protocol_context,
645 rtsp_st->dynamic_handler);
646 else if (CONFIG_RTPDEC)
647 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
648 rtsp_st->sdp_payload_type,
649 reordering_queue_size);
651 if (!rtsp_st->transport_priv) {
652 return AVERROR(ENOMEM);
653 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
654 if (rtsp_st->dynamic_handler) {
655 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
656 rtsp_st->dynamic_protocol_context,
657 rtsp_st->dynamic_handler);
664 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
665 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
672 q += strspn(q, SPACE_CHARS);
673 v = strtol(q, &p, 10);
677 v = strtol(p, &p, 10);
686 /* XXX: only one transport specification is parsed */
687 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
689 char transport_protocol[16];
691 char lower_transport[16];
693 RTSPTransportField *th;
696 reply->nb_transports = 0;
699 p += strspn(p, SPACE_CHARS);
703 th = &reply->transports[reply->nb_transports];
705 get_word_sep(transport_protocol, sizeof(transport_protocol),
707 if (!av_strcasecmp (transport_protocol, "rtp")) {
708 get_word_sep(profile, sizeof(profile), "/;,", &p);
709 lower_transport[0] = '\0';
710 /* rtp/avp/<protocol> */
712 get_word_sep(lower_transport, sizeof(lower_transport),
715 th->transport = RTSP_TRANSPORT_RTP;
716 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
717 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
718 /* x-pn-tng/<protocol> */
719 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
721 th->transport = RTSP_TRANSPORT_RDT;
722 } else if (!av_strcasecmp(transport_protocol, "raw")) {
723 get_word_sep(profile, sizeof(profile), "/;,", &p);
724 lower_transport[0] = '\0';
725 /* raw/raw/<protocol> */
727 get_word_sep(lower_transport, sizeof(lower_transport),
730 th->transport = RTSP_TRANSPORT_RAW;
732 if (!av_strcasecmp(lower_transport, "TCP"))
733 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
735 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
739 /* get each parameter */
740 while (*p != '\0' && *p != ',') {
741 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
742 if (!strcmp(parameter, "port")) {
745 rtsp_parse_range(&th->port_min, &th->port_max, &p);
747 } else if (!strcmp(parameter, "client_port")) {
750 rtsp_parse_range(&th->client_port_min,
751 &th->client_port_max, &p);
753 } else if (!strcmp(parameter, "server_port")) {
756 rtsp_parse_range(&th->server_port_min,
757 &th->server_port_max, &p);
759 } else if (!strcmp(parameter, "interleaved")) {
762 rtsp_parse_range(&th->interleaved_min,
763 &th->interleaved_max, &p);
765 } else if (!strcmp(parameter, "multicast")) {
766 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
767 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
768 } else if (!strcmp(parameter, "ttl")) {
772 th->ttl = strtol(p, &end, 10);
775 } else if (!strcmp(parameter, "destination")) {
778 get_word_sep(buf, sizeof(buf), ";,", &p);
779 get_sockaddr(buf, &th->destination);
781 } else if (!strcmp(parameter, "source")) {
784 get_word_sep(buf, sizeof(buf), ";,", &p);
785 av_strlcpy(th->source, buf, sizeof(th->source));
787 } else if (!strcmp(parameter, "mode")) {
790 get_word_sep(buf, sizeof(buf), ";, ", &p);
791 if (!strcmp(buf, "record") ||
792 !strcmp(buf, "receive"))
797 while (*p != ';' && *p != '\0' && *p != ',')
805 reply->nb_transports++;
809 static void handle_rtp_info(RTSPState *rt, const char *url,
810 uint32_t seq, uint32_t rtptime)
813 if (!rtptime || !url[0])
815 if (rt->transport != RTSP_TRANSPORT_RTP)
817 for (i = 0; i < rt->nb_rtsp_streams; i++) {
818 RTSPStream *rtsp_st = rt->rtsp_streams[i];
819 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
822 if (!strcmp(rtsp_st->control_url, url)) {
823 rtpctx->base_timestamp = rtptime;
829 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
832 char key[20], value[1024], url[1024] = "";
833 uint32_t seq = 0, rtptime = 0;
836 p += strspn(p, SPACE_CHARS);
839 get_word_sep(key, sizeof(key), "=", &p);
843 get_word_sep(value, sizeof(value), ";, ", &p);
845 if (!strcmp(key, "url"))
846 av_strlcpy(url, value, sizeof(url));
847 else if (!strcmp(key, "seq"))
848 seq = strtoul(value, NULL, 10);
849 else if (!strcmp(key, "rtptime"))
850 rtptime = strtoul(value, NULL, 10);
852 handle_rtp_info(rt, url, seq, rtptime);
861 handle_rtp_info(rt, url, seq, rtptime);
864 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
865 RTSPState *rt, const char *method)
869 /* NOTE: we do case independent match for broken servers */
871 if (av_stristart(p, "Session:", &p)) {
873 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
874 if (av_stristart(p, ";timeout=", &p) &&
875 (t = strtol(p, NULL, 10)) > 0) {
878 } else if (av_stristart(p, "Content-Length:", &p)) {
879 reply->content_length = strtol(p, NULL, 10);
880 } else if (av_stristart(p, "Transport:", &p)) {
881 rtsp_parse_transport(reply, p);
882 } else if (av_stristart(p, "CSeq:", &p)) {
883 reply->seq = strtol(p, NULL, 10);
884 } else if (av_stristart(p, "Range:", &p)) {
885 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
886 } else if (av_stristart(p, "RealChallenge1:", &p)) {
887 p += strspn(p, SPACE_CHARS);
888 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
889 } else if (av_stristart(p, "Server:", &p)) {
890 p += strspn(p, SPACE_CHARS);
891 av_strlcpy(reply->server, p, sizeof(reply->server));
892 } else if (av_stristart(p, "Notice:", &p) ||
893 av_stristart(p, "X-Notice:", &p)) {
894 reply->notice = strtol(p, NULL, 10);
895 } else if (av_stristart(p, "Location:", &p)) {
896 p += strspn(p, SPACE_CHARS);
897 av_strlcpy(reply->location, p , sizeof(reply->location));
898 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
899 p += strspn(p, SPACE_CHARS);
900 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
901 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
902 p += strspn(p, SPACE_CHARS);
903 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
904 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
905 p += strspn(p, SPACE_CHARS);
906 if (method && !strcmp(method, "DESCRIBE"))
907 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
908 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
909 p += strspn(p, SPACE_CHARS);
910 if (method && !strcmp(method, "PLAY"))
911 rtsp_parse_rtp_info(rt, p);
912 } else if (av_stristart(p, "Public:", &p) && rt) {
913 if (strstr(p, "GET_PARAMETER") &&
914 method && !strcmp(method, "OPTIONS"))
915 rt->get_parameter_supported = 1;
916 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
917 p += strspn(p, SPACE_CHARS);
918 rt->accept_dynamic_rate = atoi(p);
919 } else if (av_stristart(p, "Content-Type:", &p)) {
920 p += strspn(p, SPACE_CHARS);
921 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
925 /* skip a RTP/TCP interleaved packet */
926 void ff_rtsp_skip_packet(AVFormatContext *s)
928 RTSPState *rt = s->priv_data;
932 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
935 len = AV_RB16(buf + 1);
937 av_dlog(s, "skipping RTP packet len=%d\n", len);
942 if (len1 > sizeof(buf))
944 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
951 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
952 unsigned char **content_ptr,
953 int return_on_interleaved_data, const char *method)
955 RTSPState *rt = s->priv_data;
956 char buf[4096], buf1[1024], *q;
959 int ret, content_length, line_count = 0, request = 0;
960 unsigned char *content = NULL;
966 memset(reply, 0, sizeof(*reply));
968 /* parse reply (XXX: use buffers) */
969 rt->last_reply[0] = '\0';
973 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
974 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
980 /* XXX: only parse it if first char on line ? */
981 if (return_on_interleaved_data) {
984 ff_rtsp_skip_packet(s);
985 } else if (ch != '\r') {
986 if ((q - buf) < sizeof(buf) - 1)
992 av_dlog(s, "line='%s'\n", buf);
994 /* test if last line */
998 if (line_count == 0) {
1000 get_word(buf1, sizeof(buf1), &p);
1001 if (!strncmp(buf1, "RTSP/", 5)) {
1002 get_word(buf1, sizeof(buf1), &p);
1003 reply->status_code = atoi(buf1);
1004 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1006 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1007 get_word(buf1, sizeof(buf1), &p); // object
1011 ff_rtsp_parse_line(reply, p, rt, method);
1012 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1013 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1018 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1019 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1021 content_length = reply->content_length;
1022 if (content_length > 0) {
1023 /* leave some room for a trailing '\0' (useful for simple parsing) */
1024 content = av_malloc(content_length + 1);
1025 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1026 content[content_length] = '\0';
1029 *content_ptr = content;
1035 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1036 const char* ptr = buf;
1038 if (!strcmp(reply->reason, "OPTIONS")) {
1039 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1041 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1042 if (reply->session_id[0])
1043 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1046 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1048 av_strlcat(buf, "\r\n", sizeof(buf));
1050 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1051 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1054 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1056 rt->last_cmd_time = av_gettime();
1057 /* Even if the request from the server had data, it is not the data
1058 * that the caller wants or expects. The memory could also be leaked
1059 * if the actual following reply has content data. */
1061 av_freep(content_ptr);
1062 /* If method is set, this is called from ff_rtsp_send_cmd,
1063 * where a reply to exactly this request is awaited. For
1064 * callers from within packet receiving, we just want to
1065 * return to the caller and go back to receiving packets. */
1071 if (rt->seq != reply->seq) {
1072 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1073 rt->seq, reply->seq);
1077 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1078 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1079 reply->notice == 2306 /* Continuous Feed Terminated */) {
1080 rt->state = RTSP_STATE_IDLE;
1081 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1082 return AVERROR(EIO); /* data or server error */
1083 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1084 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1085 return AVERROR(EPERM);
1091 * Send a command to the RTSP server without waiting for the reply.
1093 * @param s RTSP (de)muxer context
1094 * @param method the method for the request
1095 * @param url the target url for the request
1096 * @param headers extra header lines to include in the request
1097 * @param send_content if non-null, the data to send as request body content
1098 * @param send_content_length the length of the send_content data, or 0 if
1099 * send_content is null
1101 * @return zero if success, nonzero otherwise
1103 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1104 const char *method, const char *url,
1105 const char *headers,
1106 const unsigned char *send_content,
1107 int send_content_length)
1109 RTSPState *rt = s->priv_data;
1110 char buf[4096], *out_buf;
1111 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1113 /* Add in RTSP headers */
1116 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1118 av_strlcat(buf, headers, sizeof(buf));
1119 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1120 if (rt->session_id[0] != '\0' && (!headers ||
1121 !strstr(headers, "\nIf-Match:"))) {
1122 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1125 char *str = ff_http_auth_create_response(&rt->auth_state,
1126 rt->auth, url, method);
1128 av_strlcat(buf, str, sizeof(buf));
1131 if (send_content_length > 0 && send_content)
1132 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1133 av_strlcat(buf, "\r\n", sizeof(buf));
1135 /* base64 encode rtsp if tunneling */
1136 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1137 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1138 out_buf = base64buf;
1141 av_dlog(s, "Sending:\n%s--\n", buf);
1143 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1144 if (send_content_length > 0 && send_content) {
1145 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1146 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1147 "with content data not supported\n");
1148 return AVERROR_PATCHWELCOME;
1150 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1152 rt->last_cmd_time = av_gettime();
1157 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1158 const char *url, const char *headers)
1160 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1163 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1164 const char *headers, RTSPMessageHeader *reply,
1165 unsigned char **content_ptr)
1167 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1168 content_ptr, NULL, 0);
1171 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1172 const char *method, const char *url,
1174 RTSPMessageHeader *reply,
1175 unsigned char **content_ptr,
1176 const unsigned char *send_content,
1177 int send_content_length)
1179 RTSPState *rt = s->priv_data;
1180 HTTPAuthType cur_auth_type;
1181 int ret, attempts = 0;
1184 cur_auth_type = rt->auth_state.auth_type;
1185 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1187 send_content_length)))
1190 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1194 if (reply->status_code == 401 &&
1195 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1196 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1199 if (reply->status_code > 400){
1200 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1204 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1210 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1211 int lower_transport, const char *real_challenge)
1213 RTSPState *rt = s->priv_data;
1214 int rtx = 0, j, i, err, interleave = 0, port_off;
1215 RTSPStream *rtsp_st;
1216 RTSPMessageHeader reply1, *reply = &reply1;
1218 const char *trans_pref;
1220 if (rt->transport == RTSP_TRANSPORT_RDT)
1221 trans_pref = "x-pn-tng";
1222 else if (rt->transport == RTSP_TRANSPORT_RAW)
1223 trans_pref = "RAW/RAW";
1225 trans_pref = "RTP/AVP";
1227 /* default timeout: 1 minute */
1230 /* Choose a random starting offset within the first half of the
1231 * port range, to allow for a number of ports to try even if the offset
1232 * happens to be at the end of the random range. */
1233 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1234 /* even random offset */
1235 port_off -= port_off & 0x01;
1237 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1238 char transport[2048];
1241 * WMS serves all UDP data over a single connection, the RTX, which
1242 * isn't necessarily the first in the SDP but has to be the first
1243 * to be set up, else the second/third SETUP will fail with a 461.
1245 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1246 rt->server_type == RTSP_SERVER_WMS) {
1249 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1250 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1252 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1256 if (rtx == rt->nb_rtsp_streams)
1257 return -1; /* no RTX found */
1258 rtsp_st = rt->rtsp_streams[rtx];
1260 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1262 rtsp_st = rt->rtsp_streams[i];
1265 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1268 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1269 port = reply->transports[0].client_port_min;
1273 /* first try in specified port range */
1274 while (j <= rt->rtp_port_max) {
1275 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1276 "?localport=%d", j);
1277 /* we will use two ports per rtp stream (rtp and rtcp) */
1279 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1280 &s->interrupt_callback, NULL))
1283 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1288 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1290 snprintf(transport, sizeof(transport) - 1,
1291 "%s/UDP;", trans_pref);
1292 if (rt->server_type != RTSP_SERVER_REAL)
1293 av_strlcat(transport, "unicast;", sizeof(transport));
1294 av_strlcatf(transport, sizeof(transport),
1295 "client_port=%d", port);
1296 if (rt->transport == RTSP_TRANSPORT_RTP &&
1297 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1298 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1302 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1303 /* For WMS streams, the application streams are only used for
1304 * UDP. When trying to set it up for TCP streams, the server
1305 * will return an error. Therefore, we skip those streams. */
1306 if (rt->server_type == RTSP_SERVER_WMS &&
1307 (rtsp_st->stream_index < 0 ||
1308 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1311 snprintf(transport, sizeof(transport) - 1,
1312 "%s/TCP;", trans_pref);
1313 if (rt->transport != RTSP_TRANSPORT_RDT)
1314 av_strlcat(transport, "unicast;", sizeof(transport));
1315 av_strlcatf(transport, sizeof(transport),
1316 "interleaved=%d-%d",
1317 interleave, interleave + 1);
1321 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1322 snprintf(transport, sizeof(transport) - 1,
1323 "%s/UDP;multicast", trans_pref);
1326 av_strlcat(transport, ";mode=record", sizeof(transport));
1327 } else if (rt->server_type == RTSP_SERVER_REAL ||
1328 rt->server_type == RTSP_SERVER_WMS)
1329 av_strlcat(transport, ";mode=play", sizeof(transport));
1330 snprintf(cmd, sizeof(cmd),
1331 "Transport: %s\r\n",
1333 if (rt->accept_dynamic_rate)
1334 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1335 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1336 char real_res[41], real_csum[9];
1337 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1339 av_strlcatf(cmd, sizeof(cmd),
1341 "RealChallenge2: %s, sd=%s\r\n",
1342 rt->session_id, real_res, real_csum);
1344 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1345 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1348 } else if (reply->status_code != RTSP_STATUS_OK ||
1349 reply->nb_transports != 1) {
1350 err = AVERROR_INVALIDDATA;
1354 /* XXX: same protocol for all streams is required */
1356 if (reply->transports[0].lower_transport != rt->lower_transport ||
1357 reply->transports[0].transport != rt->transport) {
1358 err = AVERROR_INVALIDDATA;
1362 rt->lower_transport = reply->transports[0].lower_transport;
1363 rt->transport = reply->transports[0].transport;
1366 /* Fail if the server responded with another lower transport mode
1367 * than what we requested. */
1368 if (reply->transports[0].lower_transport != lower_transport) {
1369 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1370 err = AVERROR_INVALIDDATA;
1374 switch(reply->transports[0].lower_transport) {
1375 case RTSP_LOWER_TRANSPORT_TCP:
1376 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1377 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1380 case RTSP_LOWER_TRANSPORT_UDP: {
1381 char url[1024], options[30] = "";
1383 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1384 av_strlcpy(options, "?connect=1", sizeof(options));
1385 /* Use source address if specified */
1386 if (reply->transports[0].source[0]) {
1387 ff_url_join(url, sizeof(url), "rtp", NULL,
1388 reply->transports[0].source,
1389 reply->transports[0].server_port_min, "%s", options);
1391 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1392 reply->transports[0].server_port_min, "%s", options);
1394 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1395 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1396 err = AVERROR_INVALIDDATA;
1399 /* Try to initialize the connection state in a
1400 * potential NAT router by sending dummy packets.
1401 * RTP/RTCP dummy packets are used for RDT, too.
1403 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1405 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1408 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1409 char url[1024], namebuf[50], optbuf[20] = "";
1410 struct sockaddr_storage addr;
1413 if (reply->transports[0].destination.ss_family) {
1414 addr = reply->transports[0].destination;
1415 port = reply->transports[0].port_min;
1416 ttl = reply->transports[0].ttl;
1418 addr = rtsp_st->sdp_ip;
1419 port = rtsp_st->sdp_port;
1420 ttl = rtsp_st->sdp_ttl;
1423 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1424 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1425 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1426 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1427 port, "%s", optbuf);
1428 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1429 &s->interrupt_callback, NULL) < 0) {
1430 err = AVERROR_INVALIDDATA;
1437 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1441 if (rt->nb_rtsp_streams && reply->timeout > 0)
1442 rt->timeout = reply->timeout;
1444 if (rt->server_type == RTSP_SERVER_REAL)
1445 rt->need_subscription = 1;
1450 ff_rtsp_undo_setup(s);
1454 void ff_rtsp_close_connections(AVFormatContext *s)
1456 RTSPState *rt = s->priv_data;
1457 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1458 ffurl_close(rt->rtsp_hd);
1459 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1462 int ff_rtsp_connect(AVFormatContext *s)
1464 RTSPState *rt = s->priv_data;
1465 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1466 int port, err, tcp_fd;
1467 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1468 int lower_transport_mask = 0;
1469 char real_challenge[64] = "";
1470 struct sockaddr_storage peer;
1471 socklen_t peer_len = sizeof(peer);
1473 if (rt->rtp_port_max < rt->rtp_port_min) {
1474 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1475 "than min port %d\n", rt->rtp_port_max,
1477 return AVERROR(EINVAL);
1480 if (!ff_network_init())
1481 return AVERROR(EIO);
1483 if (s->max_delay < 0) /* Not set by the caller */
1484 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1486 rt->control_transport = RTSP_MODE_PLAIN;
1487 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1488 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1489 rt->control_transport = RTSP_MODE_TUNNEL;
1491 /* Only pass through valid flags from here */
1492 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1495 lower_transport_mask = rt->lower_transport_mask;
1496 /* extract hostname and port */
1497 av_url_split(NULL, 0, auth, sizeof(auth),
1498 host, sizeof(host), &port, path, sizeof(path), s->filename);
1500 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1503 port = RTSP_DEFAULT_PORT;
1505 if (!lower_transport_mask)
1506 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1509 /* Only UDP or TCP - UDP multicast isn't supported. */
1510 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1511 (1 << RTSP_LOWER_TRANSPORT_TCP);
1512 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1513 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1514 "only UDP and TCP are supported for output.\n");
1515 err = AVERROR(EINVAL);
1520 /* Construct the URI used in request; this is similar to s->filename,
1521 * but with authentication credentials removed and RTSP specific options
1523 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1524 host, port, "%s", path);
1526 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1527 /* set up initial handshake for tunneling */
1528 char httpname[1024];
1529 char sessioncookie[17];
1532 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1533 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1534 av_get_random_seed(), av_get_random_seed());
1537 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1538 &s->interrupt_callback) < 0) {
1543 /* generate GET headers */
1544 snprintf(headers, sizeof(headers),
1545 "x-sessioncookie: %s\r\n"
1546 "Accept: application/x-rtsp-tunnelled\r\n"
1547 "Pragma: no-cache\r\n"
1548 "Cache-Control: no-cache\r\n",
1550 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1552 /* complete the connection */
1553 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1559 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1560 &s->interrupt_callback) < 0 ) {
1565 /* generate POST headers */
1566 snprintf(headers, sizeof(headers),
1567 "x-sessioncookie: %s\r\n"
1568 "Content-Type: application/x-rtsp-tunnelled\r\n"
1569 "Pragma: no-cache\r\n"
1570 "Cache-Control: no-cache\r\n"
1571 "Content-Length: 32767\r\n"
1572 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1574 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1575 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1577 /* Initialize the authentication state for the POST session. The HTTP
1578 * protocol implementation doesn't properly handle multi-pass
1579 * authentication for POST requests, since it would require one of
1581 * - implementing Expect: 100-continue, which many HTTP servers
1582 * don't support anyway, even less the RTSP servers that do HTTP
1584 * - sending the whole POST data until getting a 401 reply specifying
1585 * what authentication method to use, then resending all that data
1586 * - waiting for potential 401 replies directly after sending the
1587 * POST header (waiting for some unspecified time)
1588 * Therefore, we copy the full auth state, which works for both basic
1589 * and digest. (For digest, we would have to synchronize the nonce
1590 * count variable between the two sessions, if we'd do more requests
1591 * with the original session, though.)
1593 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1595 /* complete the connection */
1596 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1601 /* open the tcp connection */
1602 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1603 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1604 &s->interrupt_callback, NULL) < 0) {
1608 rt->rtsp_hd_out = rt->rtsp_hd;
1612 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1613 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1614 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1615 NULL, 0, NI_NUMERICHOST);
1618 /* request options supported by the server; this also detects server
1620 for (rt->server_type = RTSP_SERVER_RTP;;) {
1622 if (rt->server_type == RTSP_SERVER_REAL)
1625 * The following entries are required for proper
1626 * streaming from a Realmedia server. They are
1627 * interdependent in some way although we currently
1628 * don't quite understand how. Values were copied
1629 * from mplayer SVN r23589.
1630 * ClientChallenge is a 16-byte ID in hex
1631 * CompanyID is a 16-byte ID in base64
1633 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1634 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1635 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1636 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1638 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1639 if (reply->status_code != RTSP_STATUS_OK) {
1640 err = AVERROR_INVALIDDATA;
1644 /* detect server type if not standard-compliant RTP */
1645 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1646 rt->server_type = RTSP_SERVER_REAL;
1648 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1649 rt->server_type = RTSP_SERVER_WMS;
1650 } else if (rt->server_type == RTSP_SERVER_REAL)
1651 strcpy(real_challenge, reply->real_challenge);
1655 if (s->iformat && CONFIG_RTSP_DEMUXER)
1656 err = ff_rtsp_setup_input_streams(s, reply);
1657 else if (CONFIG_RTSP_MUXER)
1658 err = ff_rtsp_setup_output_streams(s, host);
1663 int lower_transport = ff_log2_tab[lower_transport_mask &
1664 ~(lower_transport_mask - 1)];
1666 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1667 rt->server_type == RTSP_SERVER_REAL ?
1668 real_challenge : NULL);
1671 lower_transport_mask &= ~(1 << lower_transport);
1672 if (lower_transport_mask == 0 && err == 1) {
1673 err = AVERROR(EPROTONOSUPPORT);
1678 rt->lower_transport_mask = lower_transport_mask;
1679 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1680 rt->state = RTSP_STATE_IDLE;
1681 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1684 ff_rtsp_close_streams(s);
1685 ff_rtsp_close_connections(s);
1686 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1687 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1688 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1696 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1699 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1700 uint8_t *buf, int buf_size, int64_t wait_end)
1702 RTSPState *rt = s->priv_data;
1703 RTSPStream *rtsp_st;
1704 int n, i, ret, tcp_fd, timeout_cnt = 0;
1706 struct pollfd *p = rt->p;
1707 int *fds = NULL, fdsnum, fdsidx;
1710 if (ff_check_interrupt(&s->interrupt_callback))
1711 return AVERROR_EXIT;
1712 if (wait_end && wait_end - av_gettime() < 0)
1713 return AVERROR(EAGAIN);
1716 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1717 p[max_p].fd = tcp_fd;
1718 p[max_p++].events = POLLIN;
1722 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1723 rtsp_st = rt->rtsp_streams[i];
1724 if (rtsp_st->rtp_handle) {
1725 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1727 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1731 av_log(s, AV_LOG_ERROR,
1732 "Number of fds %d not supported\n", fdsnum);
1733 return AVERROR_INVALIDDATA;
1735 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1736 p[max_p].fd = fds[fdsidx];
1737 p[max_p++].events = POLLIN;
1742 n = poll(p, max_p, POLL_TIMEOUT_MS);
1744 int j = 1 - (tcp_fd == -1);
1746 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1747 rtsp_st = rt->rtsp_streams[i];
1748 if (rtsp_st->rtp_handle) {
1749 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1750 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1752 *prtsp_st = rtsp_st;
1759 #if CONFIG_RTSP_DEMUXER
1760 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1761 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1762 if (rt->state == RTSP_STATE_STREAMING) {
1763 if (!ff_rtsp_parse_streaming_commands(s))
1766 av_log(s, AV_LOG_WARNING,
1767 "Unable to answer to TEARDOWN\n");
1771 RTSPMessageHeader reply;
1772 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1775 /* XXX: parse message */
1776 if (rt->state != RTSP_STATE_STREAMING)
1781 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1782 return AVERROR(ETIMEDOUT);
1783 } else if (n < 0 && errno != EINTR)
1784 return AVERROR(errno);
1788 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1790 RTSPState *rt = s->priv_data;
1792 RTSPStream *rtsp_st, *first_queue_st = NULL;
1793 int64_t wait_end = 0;
1795 if (rt->nb_byes == rt->nb_rtsp_streams)
1798 /* get next frames from the same RTP packet */
1799 if (rt->cur_transport_priv) {
1800 if (rt->transport == RTSP_TRANSPORT_RDT) {
1801 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1802 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1803 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1804 } else if (rt->ts && CONFIG_RTPDEC) {
1805 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1807 rt->recvbuf_pos += ret;
1808 ret = rt->recvbuf_pos < rt->recvbuf_len;
1813 rt->cur_transport_priv = NULL;
1815 } else if (ret == 1) {
1818 rt->cur_transport_priv = NULL;
1821 if (rt->transport == RTSP_TRANSPORT_RTP) {
1823 int64_t first_queue_time = 0;
1824 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1825 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1829 queue_time = ff_rtp_queued_packet_time(rtpctx);
1830 if (queue_time && (queue_time - first_queue_time < 0 ||
1831 !first_queue_time)) {
1832 first_queue_time = queue_time;
1833 first_queue_st = rt->rtsp_streams[i];
1836 if (first_queue_time)
1837 wait_end = first_queue_time + s->max_delay;
1840 /* read next RTP packet */
1843 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1845 return AVERROR(ENOMEM);
1848 switch(rt->lower_transport) {
1850 #if CONFIG_RTSP_DEMUXER
1851 case RTSP_LOWER_TRANSPORT_TCP:
1852 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1855 case RTSP_LOWER_TRANSPORT_UDP:
1856 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1857 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1858 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1859 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1862 if (len == AVERROR(EAGAIN) && first_queue_st &&
1863 rt->transport == RTSP_TRANSPORT_RTP) {
1864 rtsp_st = first_queue_st;
1865 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1872 if (rt->transport == RTSP_TRANSPORT_RDT) {
1873 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1874 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1875 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1877 /* Either bad packet, or a RTCP packet. Check if the
1878 * first_rtcp_ntp_time field was initialized. */
1879 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1880 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1881 /* first_rtcp_ntp_time has been initialized for this stream,
1882 * copy the same value to all other uninitialized streams,
1883 * in order to map their timestamp origin to the same ntp time
1886 AVStream *st = NULL;
1887 if (rtsp_st->stream_index >= 0)
1888 st = s->streams[rtsp_st->stream_index];
1889 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1890 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1891 AVStream *st2 = NULL;
1892 if (rt->rtsp_streams[i]->stream_index >= 0)
1893 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1894 if (rtpctx2 && st && st2 &&
1895 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1896 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1897 rtpctx2->rtcp_ts_offset = av_rescale_q(
1898 rtpctx->rtcp_ts_offset, st->time_base,
1903 if (ret == -RTCP_BYE) {
1906 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1907 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1909 if (rt->nb_byes == rt->nb_rtsp_streams)
1913 } else if (rt->ts && CONFIG_RTPDEC) {
1914 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1917 rt->recvbuf_len = len;
1918 rt->recvbuf_pos = ret;
1919 rt->cur_transport_priv = rt->ts;
1926 return AVERROR_INVALIDDATA;
1932 /* more packets may follow, so we save the RTP context */
1933 rt->cur_transport_priv = rtsp_st->transport_priv;
1937 #endif /* CONFIG_RTPDEC */
1939 #if CONFIG_SDP_DEMUXER
1940 static int sdp_probe(AVProbeData *p1)
1942 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1944 /* we look for a line beginning "c=IN IP" */
1945 while (p < p_end && *p != '\0') {
1946 if (p + sizeof("c=IN IP") - 1 < p_end &&
1947 av_strstart(p, "c=IN IP", NULL))
1948 return AVPROBE_SCORE_MAX / 2;
1950 while (p < p_end - 1 && *p != '\n') p++;
1959 static int sdp_read_header(AVFormatContext *s)
1961 RTSPState *rt = s->priv_data;
1962 RTSPStream *rtsp_st;
1967 if (!ff_network_init())
1968 return AVERROR(EIO);
1970 if (s->max_delay < 0) /* Not set by the caller */
1971 s->max_delay = DEFAULT_REORDERING_DELAY;
1973 /* read the whole sdp file */
1974 /* XXX: better loading */
1975 content = av_malloc(SDP_MAX_SIZE);
1976 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1979 return AVERROR_INVALIDDATA;
1981 content[size] ='\0';
1983 err = ff_sdp_parse(s, content);
1987 /* open each RTP stream */
1988 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1990 rtsp_st = rt->rtsp_streams[i];
1992 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1993 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1994 ff_url_join(url, sizeof(url), "rtp", NULL,
1995 namebuf, rtsp_st->sdp_port,
1996 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1998 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1999 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2000 &s->interrupt_callback, NULL) < 0) {
2001 err = AVERROR_INVALIDDATA;
2004 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2009 ff_rtsp_close_streams(s);
2014 static int sdp_read_close(AVFormatContext *s)
2016 ff_rtsp_close_streams(s);
2021 static const AVClass sdp_demuxer_class = {
2022 .class_name = "SDP demuxer",
2023 .item_name = av_default_item_name,
2024 .option = sdp_options,
2025 .version = LIBAVUTIL_VERSION_INT,
2028 AVInputFormat ff_sdp_demuxer = {
2030 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2031 .priv_data_size = sizeof(RTSPState),
2032 .read_probe = sdp_probe,
2033 .read_header = sdp_read_header,
2034 .read_packet = ff_rtsp_fetch_packet,
2035 .read_close = sdp_read_close,
2036 .priv_class = &sdp_demuxer_class,
2038 #endif /* CONFIG_SDP_DEMUXER */
2040 #if CONFIG_RTP_DEMUXER
2041 static int rtp_probe(AVProbeData *p)
2043 if (av_strstart(p->filename, "rtp:", NULL))
2044 return AVPROBE_SCORE_MAX;
2048 static int rtp_read_header(AVFormatContext *s)
2050 uint8_t recvbuf[1500];
2051 char host[500], sdp[500];
2053 URLContext* in = NULL;
2055 AVCodecContext codec = { 0 };
2056 struct sockaddr_storage addr;
2058 socklen_t addrlen = sizeof(addr);
2059 RTSPState *rt = s->priv_data;
2061 if (!ff_network_init())
2062 return AVERROR(EIO);
2064 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2065 &s->interrupt_callback, NULL);
2070 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2071 if (ret == AVERROR(EAGAIN))
2076 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2080 if ((recvbuf[0] & 0xc0) != 0x80) {
2081 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2086 if (RTP_PT_IS_RTCP(recvbuf[1]))
2089 payload_type = recvbuf[1] & 0x7f;
2092 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2096 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2097 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2098 "without an SDP file describing it\n",
2102 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2103 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2104 "properly you need an SDP file "
2108 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2109 NULL, 0, s->filename);
2111 snprintf(sdp, sizeof(sdp),
2112 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2113 addr.ss_family == AF_INET ? 4 : 6, host,
2114 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2115 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2116 port, payload_type);
2117 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2119 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2122 /* sdp_read_header initializes this again */
2125 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2127 ret = sdp_read_header(s);
2138 static const AVClass rtp_demuxer_class = {
2139 .class_name = "RTP demuxer",
2140 .item_name = av_default_item_name,
2141 .option = rtp_options,
2142 .version = LIBAVUTIL_VERSION_INT,
2145 AVInputFormat ff_rtp_demuxer = {
2147 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2148 .priv_data_size = sizeof(RTSPState),
2149 .read_probe = rtp_probe,
2150 .read_header = rtp_read_header,
2151 .read_packet = ff_rtsp_fetch_packet,
2152 .read_close = sdp_read_close,
2153 .flags = AVFMT_NOFILE,
2154 .priv_class = &rtp_demuxer_class,
2156 #endif /* CONFIG_RTP_DEMUXER */