3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #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, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
68 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {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, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
72 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
76 const AVOption ff_rtsp_options[] = {
77 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
78 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
79 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
80 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
81 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
82 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
83 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
84 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
85 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
86 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
87 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
88 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {-1}, INT_MIN, INT_MAX, DEC },
92 static const AVOption sdp_options[] = {
93 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
94 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
98 static const AVOption rtp_options[] = {
99 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
103 static void get_word_until_chars(char *buf, int buf_size,
104 const char *sep, const char **pp)
110 p += strspn(p, SPACE_CHARS);
112 while (!strchr(sep, *p) && *p != '\0') {
113 if ((q - buf) < buf_size - 1)
122 static void get_word_sep(char *buf, int buf_size, const char *sep,
125 if (**pp == '/') (*pp)++;
126 get_word_until_chars(buf, buf_size, sep, pp);
129 static void get_word(char *buf, int buf_size, const char **pp)
131 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
134 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
136 * Used for seeking in the rtp stream.
138 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
142 p += strspn(p, SPACE_CHARS);
143 if (!av_stristart(p, "npt=", &p))
146 *start = AV_NOPTS_VALUE;
147 *end = AV_NOPTS_VALUE;
149 get_word_sep(buf, sizeof(buf), "-", &p);
150 av_parse_time(start, buf, 1);
153 get_word_sep(buf, sizeof(buf), "-", &p);
154 av_parse_time(end, buf, 1);
156 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
157 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
160 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
162 struct addrinfo hints = { 0 }, *ai = NULL;
163 hints.ai_flags = AI_NUMERICHOST;
164 if (getaddrinfo(buf, NULL, &hints, &ai))
166 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
172 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
173 RTSPStream *rtsp_st, AVCodecContext *codec)
177 codec->codec_id = handler->codec_id;
178 rtsp_st->dynamic_handler = handler;
179 if (handler->alloc) {
180 rtsp_st->dynamic_protocol_context = handler->alloc();
181 if (!rtsp_st->dynamic_protocol_context)
182 rtsp_st->dynamic_handler = NULL;
186 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
187 static int sdp_parse_rtpmap(AVFormatContext *s,
188 AVStream *st, RTSPStream *rtsp_st,
189 int payload_type, const char *p)
191 AVCodecContext *codec = st->codec;
197 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
198 * see if we can handle this kind of payload.
199 * The space should normally not be there but some Real streams or
200 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
201 * have a trailing space. */
202 get_word_sep(buf, sizeof(buf), "/ ", &p);
203 if (payload_type < RTP_PT_PRIVATE) {
204 /* We are in a standard case
205 * (from http://www.iana.org/assignments/rtp-parameters). */
206 /* search into AVRtpPayloadTypes[] */
207 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
210 if (codec->codec_id == CODEC_ID_NONE) {
211 RTPDynamicProtocolHandler *handler =
212 ff_rtp_handler_find_by_name(buf, codec->codec_type);
213 init_rtp_handler(handler, rtsp_st, codec);
214 /* If no dynamic handler was found, check with the list of standard
215 * allocated types, if such a stream for some reason happens to
216 * use a private payload type. This isn't handled in rtpdec.c, since
217 * the format name from the rtpmap line never is passed into rtpdec. */
218 if (!rtsp_st->dynamic_handler)
219 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
222 c = avcodec_find_decoder(codec->codec_id);
228 get_word_sep(buf, sizeof(buf), "/", &p);
230 switch (codec->codec_type) {
231 case AVMEDIA_TYPE_AUDIO:
232 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
233 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
234 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
236 codec->sample_rate = i;
237 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
238 get_word_sep(buf, sizeof(buf), "/", &p);
242 // TODO: there is a bug here; if it is a mono stream, and
243 // less than 22000Hz, faad upconverts to stereo and twice
244 // the frequency. No problem, but the sample rate is being
245 // set here by the sdp line. Patch on its way. (rdm)
247 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
249 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
252 case AVMEDIA_TYPE_VIDEO:
253 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
255 avpriv_set_pts_info(st, 32, 1, i);
260 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
261 rtsp_st->dynamic_handler->init(s, st->index,
262 rtsp_st->dynamic_protocol_context);
266 /* parse the attribute line from the fmtp a line of an sdp response. This
267 * is broken out as a function because it is used in rtp_h264.c, which is
269 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
270 char *value, int value_size)
272 *p += strspn(*p, SPACE_CHARS);
274 get_word_sep(attr, attr_size, "=", p);
277 get_word_sep(value, value_size, ";", p);
285 typedef struct SDPParseState {
287 struct sockaddr_storage default_ip;
289 int skip_media; ///< set if an unknown m= line occurs
292 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
293 int letter, const char *buf)
295 RTSPState *rt = s->priv_data;
296 char buf1[64], st_type[64];
298 enum AVMediaType codec_type;
302 struct sockaddr_storage sdp_ip;
305 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
308 if (s1->skip_media && letter != 'm')
312 get_word(buf1, sizeof(buf1), &p);
313 if (strcmp(buf1, "IN") != 0)
315 get_word(buf1, sizeof(buf1), &p);
316 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
318 get_word_sep(buf1, sizeof(buf1), "/", &p);
319 if (get_sockaddr(buf1, &sdp_ip))
324 get_word_sep(buf1, sizeof(buf1), "/", &p);
327 if (s->nb_streams == 0) {
328 s1->default_ip = sdp_ip;
329 s1->default_ttl = ttl;
331 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
332 rtsp_st->sdp_ip = sdp_ip;
333 rtsp_st->sdp_ttl = ttl;
337 av_dict_set(&s->metadata, "title", p, 0);
340 if (s->nb_streams == 0) {
341 av_dict_set(&s->metadata, "comment", p, 0);
348 codec_type = AVMEDIA_TYPE_UNKNOWN;
349 get_word(st_type, sizeof(st_type), &p);
350 if (!strcmp(st_type, "audio")) {
351 codec_type = AVMEDIA_TYPE_AUDIO;
352 } else if (!strcmp(st_type, "video")) {
353 codec_type = AVMEDIA_TYPE_VIDEO;
354 } else if (!strcmp(st_type, "application")) {
355 codec_type = AVMEDIA_TYPE_DATA;
357 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
361 rtsp_st = av_mallocz(sizeof(RTSPStream));
364 rtsp_st->stream_index = -1;
365 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
367 rtsp_st->sdp_ip = s1->default_ip;
368 rtsp_st->sdp_ttl = s1->default_ttl;
370 get_word(buf1, sizeof(buf1), &p); /* port */
371 rtsp_st->sdp_port = atoi(buf1);
373 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
375 /* XXX: handle list of formats */
376 get_word(buf1, sizeof(buf1), &p); /* format list */
377 rtsp_st->sdp_payload_type = atoi(buf1);
379 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
380 /* no corresponding stream */
381 } else if (rt->server_type == RTSP_SERVER_WMS &&
382 codec_type == AVMEDIA_TYPE_DATA) {
383 /* RTX stream, a stream that carries all the other actual
384 * audio/video streams. Don't expose this to the callers. */
386 st = avformat_new_stream(s, NULL);
389 st->id = rt->nb_rtsp_streams - 1;
390 rtsp_st->stream_index = st->index;
391 st->codec->codec_type = codec_type;
392 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
393 RTPDynamicProtocolHandler *handler;
394 /* if standard payload type, we can find the codec right now */
395 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
396 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
397 st->codec->sample_rate > 0)
398 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
399 /* Even static payload types may need a custom depacketizer */
400 handler = ff_rtp_handler_find_by_id(
401 rtsp_st->sdp_payload_type, st->codec->codec_type);
402 init_rtp_handler(handler, rtsp_st, st->codec);
403 if (handler && handler->init)
404 handler->init(s, st->index,
405 rtsp_st->dynamic_protocol_context);
408 /* put a default control url */
409 av_strlcpy(rtsp_st->control_url, rt->control_uri,
410 sizeof(rtsp_st->control_url));
413 if (av_strstart(p, "control:", &p)) {
414 if (s->nb_streams == 0) {
415 if (!strncmp(p, "rtsp://", 7))
416 av_strlcpy(rt->control_uri, p,
417 sizeof(rt->control_uri));
420 /* get the control url */
421 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
423 /* XXX: may need to add full url resolution */
424 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
426 if (proto[0] == '\0') {
427 /* relative control URL */
428 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
429 av_strlcat(rtsp_st->control_url, "/",
430 sizeof(rtsp_st->control_url));
431 av_strlcat(rtsp_st->control_url, p,
432 sizeof(rtsp_st->control_url));
434 av_strlcpy(rtsp_st->control_url, p,
435 sizeof(rtsp_st->control_url));
437 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
438 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
439 get_word(buf1, sizeof(buf1), &p);
440 payload_type = atoi(buf1);
441 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
442 if (rtsp_st->stream_index >= 0) {
443 st = s->streams[rtsp_st->stream_index];
444 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
446 } else if (av_strstart(p, "fmtp:", &p) ||
447 av_strstart(p, "framesize:", &p)) {
448 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
449 // let dynamic protocol handlers have a stab at the line.
450 get_word(buf1, sizeof(buf1), &p);
451 payload_type = atoi(buf1);
452 for (i = 0; i < rt->nb_rtsp_streams; i++) {
453 rtsp_st = rt->rtsp_streams[i];
454 if (rtsp_st->sdp_payload_type == payload_type &&
455 rtsp_st->dynamic_handler &&
456 rtsp_st->dynamic_handler->parse_sdp_a_line)
457 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
458 rtsp_st->dynamic_protocol_context, buf);
460 } else if (av_strstart(p, "range:", &p)) {
463 // this is so that seeking on a streamed file can work.
464 rtsp_parse_range_npt(p, &start, &end);
465 s->start_time = start;
466 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
467 s->duration = (end == AV_NOPTS_VALUE) ?
468 AV_NOPTS_VALUE : end - start;
469 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
471 rt->transport = RTSP_TRANSPORT_RDT;
472 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
474 st = s->streams[s->nb_streams - 1];
475 st->codec->sample_rate = atoi(p);
477 if (rt->server_type == RTSP_SERVER_WMS)
478 ff_wms_parse_sdp_a_line(s, p);
479 if (s->nb_streams > 0) {
480 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
482 if (rt->server_type == RTSP_SERVER_REAL)
483 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
485 if (rtsp_st->dynamic_handler &&
486 rtsp_st->dynamic_handler->parse_sdp_a_line)
487 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
488 rtsp_st->stream_index,
489 rtsp_st->dynamic_protocol_context, buf);
496 int ff_sdp_parse(AVFormatContext *s, const char *content)
498 RTSPState *rt = s->priv_data;
501 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
502 * contain long SDP lines containing complete ASF Headers (several
503 * kB) or arrays of MDPR (RM stream descriptor) headers plus
504 * "rulebooks" describing their properties. Therefore, the SDP line
507 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
508 * in rtpdec_xiph.c. */
510 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
514 p += strspn(p, SPACE_CHARS);
522 /* get the content */
524 while (*p != '\n' && *p != '\r' && *p != '\0') {
525 if ((q - buf) < sizeof(buf) - 1)
530 sdp_parse_line(s, s1, letter, buf);
532 while (*p != '\n' && *p != '\0')
537 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
538 if (!rt->p) return AVERROR(ENOMEM);
541 #endif /* CONFIG_RTPDEC */
543 void ff_rtsp_undo_setup(AVFormatContext *s)
545 RTSPState *rt = s->priv_data;
548 for (i = 0; i < rt->nb_rtsp_streams; i++) {
549 RTSPStream *rtsp_st = rt->rtsp_streams[i];
552 if (rtsp_st->transport_priv) {
554 AVFormatContext *rtpctx = rtsp_st->transport_priv;
555 av_write_trailer(rtpctx);
556 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
558 avio_close_dyn_buf(rtpctx->pb, &ptr);
561 avio_close(rtpctx->pb);
563 avformat_free_context(rtpctx);
564 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
565 ff_rdt_parse_close(rtsp_st->transport_priv);
566 else if (CONFIG_RTPDEC)
567 ff_rtp_parse_close(rtsp_st->transport_priv);
569 rtsp_st->transport_priv = NULL;
570 if (rtsp_st->rtp_handle)
571 ffurl_close(rtsp_st->rtp_handle);
572 rtsp_st->rtp_handle = NULL;
576 /* close and free RTSP streams */
577 void ff_rtsp_close_streams(AVFormatContext *s)
579 RTSPState *rt = s->priv_data;
583 ff_rtsp_undo_setup(s);
584 for (i = 0; i < rt->nb_rtsp_streams; i++) {
585 rtsp_st = rt->rtsp_streams[i];
587 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
588 rtsp_st->dynamic_handler->free(
589 rtsp_st->dynamic_protocol_context);
593 av_free(rt->rtsp_streams);
595 avformat_close_input(&rt->asf_ctx);
598 av_free(rt->recvbuf);
601 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
603 RTSPState *rt = s->priv_data;
606 /* open the RTP context */
607 if (rtsp_st->stream_index >= 0)
608 st = s->streams[rtsp_st->stream_index];
610 s->ctx_flags |= AVFMTCTX_NOHEADER;
612 if (s->oformat && CONFIG_RTSP_MUXER) {
613 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
615 RTSP_TCP_MAX_PACKET_SIZE);
616 /* Ownership of rtp_handle is passed to the rtp mux context */
617 rtsp_st->rtp_handle = NULL;
620 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
621 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
622 rtsp_st->dynamic_protocol_context,
623 rtsp_st->dynamic_handler);
624 else if (CONFIG_RTPDEC)
625 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
626 rtsp_st->sdp_payload_type,
627 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
628 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
630 if (!rtsp_st->transport_priv) {
631 return AVERROR(ENOMEM);
632 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
633 if (rtsp_st->dynamic_handler) {
634 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
635 rtsp_st->dynamic_protocol_context,
636 rtsp_st->dynamic_handler);
643 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
644 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
651 q += strspn(q, SPACE_CHARS);
652 v = strtol(q, &p, 10);
656 v = strtol(p, &p, 10);
665 /* XXX: only one transport specification is parsed */
666 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
668 char transport_protocol[16];
670 char lower_transport[16];
672 RTSPTransportField *th;
675 reply->nb_transports = 0;
678 p += strspn(p, SPACE_CHARS);
682 th = &reply->transports[reply->nb_transports];
684 get_word_sep(transport_protocol, sizeof(transport_protocol),
686 if (!av_strcasecmp (transport_protocol, "rtp")) {
687 get_word_sep(profile, sizeof(profile), "/;,", &p);
688 lower_transport[0] = '\0';
689 /* rtp/avp/<protocol> */
691 get_word_sep(lower_transport, sizeof(lower_transport),
694 th->transport = RTSP_TRANSPORT_RTP;
695 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
696 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
697 /* x-pn-tng/<protocol> */
698 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
700 th->transport = RTSP_TRANSPORT_RDT;
702 if (!av_strcasecmp(lower_transport, "TCP"))
703 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
705 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
709 /* get each parameter */
710 while (*p != '\0' && *p != ',') {
711 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
712 if (!strcmp(parameter, "port")) {
715 rtsp_parse_range(&th->port_min, &th->port_max, &p);
717 } else if (!strcmp(parameter, "client_port")) {
720 rtsp_parse_range(&th->client_port_min,
721 &th->client_port_max, &p);
723 } else if (!strcmp(parameter, "server_port")) {
726 rtsp_parse_range(&th->server_port_min,
727 &th->server_port_max, &p);
729 } else if (!strcmp(parameter, "interleaved")) {
732 rtsp_parse_range(&th->interleaved_min,
733 &th->interleaved_max, &p);
735 } else if (!strcmp(parameter, "multicast")) {
736 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
737 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
738 } else if (!strcmp(parameter, "ttl")) {
741 th->ttl = strtol(p, (char **)&p, 10);
743 } else if (!strcmp(parameter, "destination")) {
746 get_word_sep(buf, sizeof(buf), ";,", &p);
747 get_sockaddr(buf, &th->destination);
749 } else if (!strcmp(parameter, "source")) {
752 get_word_sep(buf, sizeof(buf), ";,", &p);
753 av_strlcpy(th->source, buf, sizeof(th->source));
755 } else if (!strcmp(parameter, "mode")) {
758 get_word_sep(buf, sizeof(buf), ";, ", &p);
759 if (!strcmp(buf, "record") ||
760 !strcmp(buf, "receive"))
765 while (*p != ';' && *p != '\0' && *p != ',')
773 reply->nb_transports++;
777 static void handle_rtp_info(RTSPState *rt, const char *url,
778 uint32_t seq, uint32_t rtptime)
781 if (!rtptime || !url[0])
783 if (rt->transport != RTSP_TRANSPORT_RTP)
785 for (i = 0; i < rt->nb_rtsp_streams; i++) {
786 RTSPStream *rtsp_st = rt->rtsp_streams[i];
787 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
790 if (!strcmp(rtsp_st->control_url, url)) {
791 rtpctx->base_timestamp = rtptime;
797 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
800 char key[20], value[1024], url[1024] = "";
801 uint32_t seq = 0, rtptime = 0;
804 p += strspn(p, SPACE_CHARS);
807 get_word_sep(key, sizeof(key), "=", &p);
811 get_word_sep(value, sizeof(value), ";, ", &p);
813 if (!strcmp(key, "url"))
814 av_strlcpy(url, value, sizeof(url));
815 else if (!strcmp(key, "seq"))
816 seq = strtoul(value, NULL, 10);
817 else if (!strcmp(key, "rtptime"))
818 rtptime = strtoul(value, NULL, 10);
820 handle_rtp_info(rt, url, seq, rtptime);
829 handle_rtp_info(rt, url, seq, rtptime);
832 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
833 RTSPState *rt, const char *method)
837 /* NOTE: we do case independent match for broken servers */
839 if (av_stristart(p, "Session:", &p)) {
841 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
842 if (av_stristart(p, ";timeout=", &p) &&
843 (t = strtol(p, NULL, 10)) > 0) {
846 } else if (av_stristart(p, "Content-Length:", &p)) {
847 reply->content_length = strtol(p, NULL, 10);
848 } else if (av_stristart(p, "Transport:", &p)) {
849 rtsp_parse_transport(reply, p);
850 } else if (av_stristart(p, "CSeq:", &p)) {
851 reply->seq = strtol(p, NULL, 10);
852 } else if (av_stristart(p, "Range:", &p)) {
853 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
854 } else if (av_stristart(p, "RealChallenge1:", &p)) {
855 p += strspn(p, SPACE_CHARS);
856 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
857 } else if (av_stristart(p, "Server:", &p)) {
858 p += strspn(p, SPACE_CHARS);
859 av_strlcpy(reply->server, p, sizeof(reply->server));
860 } else if (av_stristart(p, "Notice:", &p) ||
861 av_stristart(p, "X-Notice:", &p)) {
862 reply->notice = strtol(p, NULL, 10);
863 } else if (av_stristart(p, "Location:", &p)) {
864 p += strspn(p, SPACE_CHARS);
865 av_strlcpy(reply->location, p , sizeof(reply->location));
866 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
867 p += strspn(p, SPACE_CHARS);
868 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
869 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
870 p += strspn(p, SPACE_CHARS);
871 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
872 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
873 p += strspn(p, SPACE_CHARS);
874 if (method && !strcmp(method, "DESCRIBE"))
875 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
876 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
877 p += strspn(p, SPACE_CHARS);
878 if (method && !strcmp(method, "PLAY"))
879 rtsp_parse_rtp_info(rt, p);
880 } else if (av_stristart(p, "Public:", &p) && rt) {
881 if (strstr(p, "GET_PARAMETER") &&
882 method && !strcmp(method, "OPTIONS"))
883 rt->get_parameter_supported = 1;
884 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
885 p += strspn(p, SPACE_CHARS);
886 rt->accept_dynamic_rate = atoi(p);
887 } else if (av_stristart(p, "Content-Type:", &p)) {
888 p += strspn(p, SPACE_CHARS);
889 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
893 /* skip a RTP/TCP interleaved packet */
894 void ff_rtsp_skip_packet(AVFormatContext *s)
896 RTSPState *rt = s->priv_data;
900 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
903 len = AV_RB16(buf + 1);
905 av_dlog(s, "skipping RTP packet len=%d\n", len);
910 if (len1 > sizeof(buf))
912 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
919 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
920 unsigned char **content_ptr,
921 int return_on_interleaved_data, const char *method)
923 RTSPState *rt = s->priv_data;
924 char buf[4096], buf1[1024], *q;
927 int ret, content_length, line_count = 0, request = 0;
928 unsigned char *content = NULL;
934 memset(reply, 0, sizeof(*reply));
936 /* parse reply (XXX: use buffers) */
937 rt->last_reply[0] = '\0';
941 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
942 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
948 /* XXX: only parse it if first char on line ? */
949 if (return_on_interleaved_data) {
952 ff_rtsp_skip_packet(s);
953 } else if (ch != '\r') {
954 if ((q - buf) < sizeof(buf) - 1)
960 av_dlog(s, "line='%s'\n", buf);
962 /* test if last line */
966 if (line_count == 0) {
968 get_word(buf1, sizeof(buf1), &p);
969 if (!strncmp(buf1, "RTSP/", 5)) {
970 get_word(buf1, sizeof(buf1), &p);
971 reply->status_code = atoi(buf1);
972 av_strlcpy(reply->reason, p, sizeof(reply->reason));
974 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
975 get_word(buf1, sizeof(buf1), &p); // object
979 ff_rtsp_parse_line(reply, p, rt, method);
980 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
981 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
986 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
987 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
989 content_length = reply->content_length;
990 if (content_length > 0) {
991 /* leave some room for a trailing '\0' (useful for simple parsing) */
992 content = av_malloc(content_length + 1);
993 ffurl_read_complete(rt->rtsp_hd, content, content_length);
994 content[content_length] = '\0';
997 *content_ptr = content;
1003 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1004 const char* ptr = buf;
1006 if (!strcmp(reply->reason, "OPTIONS")) {
1007 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1009 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1010 if (reply->session_id[0])
1011 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1014 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1016 av_strlcat(buf, "\r\n", sizeof(buf));
1018 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1019 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1022 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1024 rt->last_cmd_time = av_gettime();
1025 /* Even if the request from the server had data, it is not the data
1026 * that the caller wants or expects. The memory could also be leaked
1027 * if the actual following reply has content data. */
1029 av_freep(content_ptr);
1030 /* If method is set, this is called from ff_rtsp_send_cmd,
1031 * where a reply to exactly this request is awaited. For
1032 * callers from within packet receiving, we just want to
1033 * return to the caller and go back to receiving packets. */
1039 if (rt->seq != reply->seq) {
1040 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1041 rt->seq, reply->seq);
1045 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1046 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1047 reply->notice == 2306 /* Continuous Feed Terminated */) {
1048 rt->state = RTSP_STATE_IDLE;
1049 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1050 return AVERROR(EIO); /* data or server error */
1051 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1052 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1053 return AVERROR(EPERM);
1059 * Send a command to the RTSP server without waiting for the reply.
1061 * @param s RTSP (de)muxer context
1062 * @param method the method for the request
1063 * @param url the target url for the request
1064 * @param headers extra header lines to include in the request
1065 * @param send_content if non-null, the data to send as request body content
1066 * @param send_content_length the length of the send_content data, or 0 if
1067 * send_content is null
1069 * @return zero if success, nonzero otherwise
1071 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1072 const char *method, const char *url,
1073 const char *headers,
1074 const unsigned char *send_content,
1075 int send_content_length)
1077 RTSPState *rt = s->priv_data;
1078 char buf[4096], *out_buf;
1079 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1081 /* Add in RTSP headers */
1084 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1086 av_strlcat(buf, headers, sizeof(buf));
1087 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1088 if (rt->session_id[0] != '\0' && (!headers ||
1089 !strstr(headers, "\nIf-Match:"))) {
1090 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1093 char *str = ff_http_auth_create_response(&rt->auth_state,
1094 rt->auth, url, method);
1096 av_strlcat(buf, str, sizeof(buf));
1099 if (send_content_length > 0 && send_content)
1100 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1101 av_strlcat(buf, "\r\n", sizeof(buf));
1103 /* base64 encode rtsp if tunneling */
1104 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1105 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1106 out_buf = base64buf;
1109 av_dlog(s, "Sending:\n%s--\n", buf);
1111 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1112 if (send_content_length > 0 && send_content) {
1113 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1114 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1115 "with content data not supported\n");
1116 return AVERROR_PATCHWELCOME;
1118 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1120 rt->last_cmd_time = av_gettime();
1125 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1126 const char *url, const char *headers)
1128 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1131 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1132 const char *headers, RTSPMessageHeader *reply,
1133 unsigned char **content_ptr)
1135 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1136 content_ptr, NULL, 0);
1139 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1140 const char *method, const char *url,
1142 RTSPMessageHeader *reply,
1143 unsigned char **content_ptr,
1144 const unsigned char *send_content,
1145 int send_content_length)
1147 RTSPState *rt = s->priv_data;
1148 HTTPAuthType cur_auth_type;
1149 int ret, attempts = 0;
1152 cur_auth_type = rt->auth_state.auth_type;
1153 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1155 send_content_length)))
1158 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1162 if (reply->status_code == 401 &&
1163 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1164 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1167 if (reply->status_code > 400){
1168 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1172 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1178 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1179 int lower_transport, const char *real_challenge)
1181 RTSPState *rt = s->priv_data;
1182 int rtx = 0, j, i, err, interleave = 0, port_off;
1183 RTSPStream *rtsp_st;
1184 RTSPMessageHeader reply1, *reply = &reply1;
1186 const char *trans_pref;
1188 if (rt->transport == RTSP_TRANSPORT_RDT)
1189 trans_pref = "x-pn-tng";
1191 trans_pref = "RTP/AVP";
1193 /* default timeout: 1 minute */
1196 /* for each stream, make the setup request */
1197 /* XXX: we assume the same server is used for the control of each
1200 /* Choose a random starting offset within the first half of the
1201 * port range, to allow for a number of ports to try even if the offset
1202 * happens to be at the end of the random range. */
1203 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1204 /* even random offset */
1205 port_off -= port_off & 0x01;
1207 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1208 char transport[2048];
1211 * WMS serves all UDP data over a single connection, the RTX, which
1212 * isn't necessarily the first in the SDP but has to be the first
1213 * to be set up, else the second/third SETUP will fail with a 461.
1215 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1216 rt->server_type == RTSP_SERVER_WMS) {
1219 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1220 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1222 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1226 if (rtx == rt->nb_rtsp_streams)
1227 return -1; /* no RTX found */
1228 rtsp_st = rt->rtsp_streams[rtx];
1230 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1232 rtsp_st = rt->rtsp_streams[i];
1235 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1238 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1239 port = reply->transports[0].client_port_min;
1243 /* first try in specified port range */
1244 while (j <= rt->rtp_port_max) {
1245 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1246 "?localport=%d", j);
1247 /* we will use two ports per rtp stream (rtp and rtcp) */
1249 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1250 &s->interrupt_callback, NULL))
1254 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1259 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1261 snprintf(transport, sizeof(transport) - 1,
1262 "%s/UDP;", trans_pref);
1263 if (rt->server_type != RTSP_SERVER_REAL)
1264 av_strlcat(transport, "unicast;", sizeof(transport));
1265 av_strlcatf(transport, sizeof(transport),
1266 "client_port=%d", port);
1267 if (rt->transport == RTSP_TRANSPORT_RTP &&
1268 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1269 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1273 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1274 /* For WMS streams, the application streams are only used for
1275 * UDP. When trying to set it up for TCP streams, the server
1276 * will return an error. Therefore, we skip those streams. */
1277 if (rt->server_type == RTSP_SERVER_WMS &&
1278 (rtsp_st->stream_index < 0 ||
1279 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1282 snprintf(transport, sizeof(transport) - 1,
1283 "%s/TCP;", trans_pref);
1284 if (rt->transport != RTSP_TRANSPORT_RDT)
1285 av_strlcat(transport, "unicast;", sizeof(transport));
1286 av_strlcatf(transport, sizeof(transport),
1287 "interleaved=%d-%d",
1288 interleave, interleave + 1);
1292 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1293 snprintf(transport, sizeof(transport) - 1,
1294 "%s/UDP;multicast", trans_pref);
1297 av_strlcat(transport, ";mode=record", sizeof(transport));
1298 } else if (rt->server_type == RTSP_SERVER_REAL ||
1299 rt->server_type == RTSP_SERVER_WMS)
1300 av_strlcat(transport, ";mode=play", sizeof(transport));
1301 snprintf(cmd, sizeof(cmd),
1302 "Transport: %s\r\n",
1304 if (rt->accept_dynamic_rate)
1305 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1306 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1307 char real_res[41], real_csum[9];
1308 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1310 av_strlcatf(cmd, sizeof(cmd),
1312 "RealChallenge2: %s, sd=%s\r\n",
1313 rt->session_id, real_res, real_csum);
1315 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1316 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1319 } else if (reply->status_code != RTSP_STATUS_OK ||
1320 reply->nb_transports != 1) {
1321 err = AVERROR_INVALIDDATA;
1325 /* XXX: same protocol for all streams is required */
1327 if (reply->transports[0].lower_transport != rt->lower_transport ||
1328 reply->transports[0].transport != rt->transport) {
1329 err = AVERROR_INVALIDDATA;
1333 rt->lower_transport = reply->transports[0].lower_transport;
1334 rt->transport = reply->transports[0].transport;
1337 /* Fail if the server responded with another lower transport mode
1338 * than what we requested. */
1339 if (reply->transports[0].lower_transport != lower_transport) {
1340 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1341 err = AVERROR_INVALIDDATA;
1345 switch(reply->transports[0].lower_transport) {
1346 case RTSP_LOWER_TRANSPORT_TCP:
1347 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1348 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1351 case RTSP_LOWER_TRANSPORT_UDP: {
1352 char url[1024], options[30] = "";
1354 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1355 av_strlcpy(options, "?connect=1", sizeof(options));
1356 /* Use source address if specified */
1357 if (reply->transports[0].source[0]) {
1358 ff_url_join(url, sizeof(url), "rtp", NULL,
1359 reply->transports[0].source,
1360 reply->transports[0].server_port_min, "%s", options);
1362 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1363 reply->transports[0].server_port_min, "%s", options);
1365 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1366 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1367 err = AVERROR_INVALIDDATA;
1370 /* Try to initialize the connection state in a
1371 * potential NAT router by sending dummy packets.
1372 * RTP/RTCP dummy packets are used for RDT, too.
1374 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1376 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1379 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1380 char url[1024], namebuf[50], optbuf[20] = "";
1381 struct sockaddr_storage addr;
1384 if (reply->transports[0].destination.ss_family) {
1385 addr = reply->transports[0].destination;
1386 port = reply->transports[0].port_min;
1387 ttl = reply->transports[0].ttl;
1389 addr = rtsp_st->sdp_ip;
1390 port = rtsp_st->sdp_port;
1391 ttl = rtsp_st->sdp_ttl;
1394 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1395 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1396 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1397 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1398 port, "%s", optbuf);
1399 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1400 &s->interrupt_callback, NULL) < 0) {
1401 err = AVERROR_INVALIDDATA;
1408 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1412 if (rt->nb_rtsp_streams && reply->timeout > 0)
1413 rt->timeout = reply->timeout;
1415 if (rt->server_type == RTSP_SERVER_REAL)
1416 rt->need_subscription = 1;
1421 ff_rtsp_undo_setup(s);
1425 void ff_rtsp_close_connections(AVFormatContext *s)
1427 RTSPState *rt = s->priv_data;
1428 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1429 ffurl_close(rt->rtsp_hd);
1430 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1433 int ff_rtsp_connect(AVFormatContext *s)
1435 RTSPState *rt = s->priv_data;
1436 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1437 int port, err, tcp_fd;
1438 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1439 int lower_transport_mask = 0;
1440 char real_challenge[64] = "";
1441 struct sockaddr_storage peer;
1442 socklen_t peer_len = sizeof(peer);
1444 if (rt->rtp_port_max < rt->rtp_port_min) {
1445 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1446 "than min port %d\n", rt->rtp_port_max,
1448 return AVERROR(EINVAL);
1451 if (!ff_network_init())
1452 return AVERROR(EIO);
1454 if (s->max_delay < 0) /* Not set by the caller */
1455 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1457 rt->control_transport = RTSP_MODE_PLAIN;
1458 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1459 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1460 rt->control_transport = RTSP_MODE_TUNNEL;
1462 /* Only pass through valid flags from here */
1463 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1466 lower_transport_mask = rt->lower_transport_mask;
1467 /* extract hostname and port */
1468 av_url_split(NULL, 0, auth, sizeof(auth),
1469 host, sizeof(host), &port, path, sizeof(path), s->filename);
1471 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1474 port = RTSP_DEFAULT_PORT;
1476 if (!lower_transport_mask)
1477 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1480 /* Only UDP or TCP - UDP multicast isn't supported. */
1481 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1482 (1 << RTSP_LOWER_TRANSPORT_TCP);
1483 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1484 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1485 "only UDP and TCP are supported for output.\n");
1486 err = AVERROR(EINVAL);
1491 /* Construct the URI used in request; this is similar to s->filename,
1492 * but with authentication credentials removed and RTSP specific options
1494 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1495 host, port, "%s", path);
1497 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1498 /* set up initial handshake for tunneling */
1499 char httpname[1024];
1500 char sessioncookie[17];
1503 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1504 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1505 av_get_random_seed(), av_get_random_seed());
1508 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1509 &s->interrupt_callback) < 0) {
1514 /* generate GET headers */
1515 snprintf(headers, sizeof(headers),
1516 "x-sessioncookie: %s\r\n"
1517 "Accept: application/x-rtsp-tunnelled\r\n"
1518 "Pragma: no-cache\r\n"
1519 "Cache-Control: no-cache\r\n",
1521 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1523 /* complete the connection */
1524 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1530 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1531 &s->interrupt_callback) < 0 ) {
1536 /* generate POST headers */
1537 snprintf(headers, sizeof(headers),
1538 "x-sessioncookie: %s\r\n"
1539 "Content-Type: application/x-rtsp-tunnelled\r\n"
1540 "Pragma: no-cache\r\n"
1541 "Cache-Control: no-cache\r\n"
1542 "Content-Length: 32767\r\n"
1543 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1545 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1546 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1548 /* Initialize the authentication state for the POST session. The HTTP
1549 * protocol implementation doesn't properly handle multi-pass
1550 * authentication for POST requests, since it would require one of
1552 * - implementing Expect: 100-continue, which many HTTP servers
1553 * don't support anyway, even less the RTSP servers that do HTTP
1555 * - sending the whole POST data until getting a 401 reply specifying
1556 * what authentication method to use, then resending all that data
1557 * - waiting for potential 401 replies directly after sending the
1558 * POST header (waiting for some unspecified time)
1559 * Therefore, we copy the full auth state, which works for both basic
1560 * and digest. (For digest, we would have to synchronize the nonce
1561 * count variable between the two sessions, if we'd do more requests
1562 * with the original session, though.)
1564 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1566 /* complete the connection */
1567 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1572 /* open the tcp connection */
1573 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1574 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1575 &s->interrupt_callback, NULL) < 0) {
1579 rt->rtsp_hd_out = rt->rtsp_hd;
1583 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1584 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1585 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1586 NULL, 0, NI_NUMERICHOST);
1589 /* request options supported by the server; this also detects server
1591 for (rt->server_type = RTSP_SERVER_RTP;;) {
1593 if (rt->server_type == RTSP_SERVER_REAL)
1596 * The following entries are required for proper
1597 * streaming from a Realmedia server. They are
1598 * interdependent in some way although we currently
1599 * don't quite understand how. Values were copied
1600 * from mplayer SVN r23589.
1601 * ClientChallenge is a 16-byte ID in hex
1602 * CompanyID is a 16-byte ID in base64
1604 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1605 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1606 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1607 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1609 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1610 if (reply->status_code != RTSP_STATUS_OK) {
1611 err = AVERROR_INVALIDDATA;
1615 /* detect server type if not standard-compliant RTP */
1616 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1617 rt->server_type = RTSP_SERVER_REAL;
1619 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1620 rt->server_type = RTSP_SERVER_WMS;
1621 } else if (rt->server_type == RTSP_SERVER_REAL)
1622 strcpy(real_challenge, reply->real_challenge);
1626 if (s->iformat && CONFIG_RTSP_DEMUXER)
1627 err = ff_rtsp_setup_input_streams(s, reply);
1628 else if (CONFIG_RTSP_MUXER)
1629 err = ff_rtsp_setup_output_streams(s, host);
1634 int lower_transport = ff_log2_tab[lower_transport_mask &
1635 ~(lower_transport_mask - 1)];
1637 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1638 rt->server_type == RTSP_SERVER_REAL ?
1639 real_challenge : NULL);
1642 lower_transport_mask &= ~(1 << lower_transport);
1643 if (lower_transport_mask == 0 && err == 1) {
1644 err = AVERROR(EPROTONOSUPPORT);
1649 rt->lower_transport_mask = lower_transport_mask;
1650 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1651 rt->state = RTSP_STATE_IDLE;
1652 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1655 ff_rtsp_close_streams(s);
1656 ff_rtsp_close_connections(s);
1657 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1658 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1659 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1667 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1670 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1671 uint8_t *buf, int buf_size, int64_t wait_end)
1673 RTSPState *rt = s->priv_data;
1674 RTSPStream *rtsp_st;
1675 int n, i, ret, tcp_fd, timeout_cnt = 0;
1677 struct pollfd *p = rt->p;
1680 if (ff_check_interrupt(&s->interrupt_callback))
1681 return AVERROR_EXIT;
1682 if (wait_end && wait_end - av_gettime() < 0)
1683 return AVERROR(EAGAIN);
1686 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1687 p[max_p].fd = tcp_fd;
1688 p[max_p++].events = POLLIN;
1692 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1693 rtsp_st = rt->rtsp_streams[i];
1694 if (rtsp_st->rtp_handle) {
1695 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1696 p[max_p++].events = POLLIN;
1697 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1698 p[max_p++].events = POLLIN;
1701 n = poll(p, max_p, POLL_TIMEOUT_MS);
1703 int j = 1 - (tcp_fd == -1);
1705 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1706 rtsp_st = rt->rtsp_streams[i];
1707 if (rtsp_st->rtp_handle) {
1708 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1709 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1711 *prtsp_st = rtsp_st;
1718 #if CONFIG_RTSP_DEMUXER
1719 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1720 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1721 if (rt->state == RTSP_STATE_STREAMING) {
1722 if (!ff_rtsp_parse_streaming_commands(s))
1725 av_log(s, AV_LOG_WARNING,
1726 "Unable to answer to TEARDOWN\n");
1730 RTSPMessageHeader reply;
1731 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1734 /* XXX: parse message */
1735 if (rt->state != RTSP_STATE_STREAMING)
1740 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1741 return AVERROR(ETIMEDOUT);
1742 } else if (n < 0 && errno != EINTR)
1743 return AVERROR(errno);
1747 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1749 RTSPState *rt = s->priv_data;
1751 RTSPStream *rtsp_st, *first_queue_st = NULL;
1752 int64_t wait_end = 0;
1754 if (rt->nb_byes == rt->nb_rtsp_streams)
1757 /* get next frames from the same RTP packet */
1758 if (rt->cur_transport_priv) {
1759 if (rt->transport == RTSP_TRANSPORT_RDT) {
1760 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1762 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1764 rt->cur_transport_priv = NULL;
1766 } else if (ret == 1) {
1769 rt->cur_transport_priv = NULL;
1772 if (rt->transport == RTSP_TRANSPORT_RTP) {
1774 int64_t first_queue_time = 0;
1775 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1776 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1780 queue_time = ff_rtp_queued_packet_time(rtpctx);
1781 if (queue_time && (queue_time - first_queue_time < 0 ||
1782 !first_queue_time)) {
1783 first_queue_time = queue_time;
1784 first_queue_st = rt->rtsp_streams[i];
1787 if (first_queue_time)
1788 wait_end = first_queue_time + s->max_delay;
1791 /* read next RTP packet */
1794 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1796 return AVERROR(ENOMEM);
1799 switch(rt->lower_transport) {
1801 #if CONFIG_RTSP_DEMUXER
1802 case RTSP_LOWER_TRANSPORT_TCP:
1803 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1806 case RTSP_LOWER_TRANSPORT_UDP:
1807 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1808 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1809 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1810 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1813 if (len == AVERROR(EAGAIN) && first_queue_st &&
1814 rt->transport == RTSP_TRANSPORT_RTP) {
1815 rtsp_st = first_queue_st;
1816 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1823 if (rt->transport == RTSP_TRANSPORT_RDT) {
1824 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1826 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1828 /* Either bad packet, or a RTCP packet. Check if the
1829 * first_rtcp_ntp_time field was initialized. */
1830 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1831 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1832 /* first_rtcp_ntp_time has been initialized for this stream,
1833 * copy the same value to all other uninitialized streams,
1834 * in order to map their timestamp origin to the same ntp time
1837 AVStream *st = NULL;
1838 if (rtsp_st->stream_index >= 0)
1839 st = s->streams[rtsp_st->stream_index];
1840 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1841 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1842 AVStream *st2 = NULL;
1843 if (rt->rtsp_streams[i]->stream_index >= 0)
1844 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1845 if (rtpctx2 && st && st2 &&
1846 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1847 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1848 rtpctx2->rtcp_ts_offset = av_rescale_q(
1849 rtpctx->rtcp_ts_offset, st->time_base,
1854 if (ret == -RTCP_BYE) {
1857 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1858 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1860 if (rt->nb_byes == rt->nb_rtsp_streams)
1869 /* more packets may follow, so we save the RTP context */
1870 rt->cur_transport_priv = rtsp_st->transport_priv;
1874 #endif /* CONFIG_RTPDEC */
1876 #if CONFIG_SDP_DEMUXER
1877 static int sdp_probe(AVProbeData *p1)
1879 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1881 /* we look for a line beginning "c=IN IP" */
1882 while (p < p_end && *p != '\0') {
1883 if (p + sizeof("c=IN IP") - 1 < p_end &&
1884 av_strstart(p, "c=IN IP", NULL))
1885 return AVPROBE_SCORE_MAX / 2;
1887 while (p < p_end - 1 && *p != '\n') p++;
1896 static int sdp_read_header(AVFormatContext *s)
1898 RTSPState *rt = s->priv_data;
1899 RTSPStream *rtsp_st;
1904 if (!ff_network_init())
1905 return AVERROR(EIO);
1907 if (s->max_delay < 0) /* Not set by the caller */
1908 s->max_delay = DEFAULT_REORDERING_DELAY;
1910 /* read the whole sdp file */
1911 /* XXX: better loading */
1912 content = av_malloc(SDP_MAX_SIZE);
1913 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1916 return AVERROR_INVALIDDATA;
1918 content[size] ='\0';
1920 err = ff_sdp_parse(s, content);
1924 /* open each RTP stream */
1925 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1927 rtsp_st = rt->rtsp_streams[i];
1929 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1930 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1931 ff_url_join(url, sizeof(url), "rtp", NULL,
1932 namebuf, rtsp_st->sdp_port,
1933 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1935 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1936 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1937 &s->interrupt_callback, NULL) < 0) {
1938 err = AVERROR_INVALIDDATA;
1941 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1946 ff_rtsp_close_streams(s);
1951 static int sdp_read_close(AVFormatContext *s)
1953 ff_rtsp_close_streams(s);
1958 static const AVClass sdp_demuxer_class = {
1959 .class_name = "SDP demuxer",
1960 .item_name = av_default_item_name,
1961 .option = sdp_options,
1962 .version = LIBAVUTIL_VERSION_INT,
1965 AVInputFormat ff_sdp_demuxer = {
1967 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1968 .priv_data_size = sizeof(RTSPState),
1969 .read_probe = sdp_probe,
1970 .read_header = sdp_read_header,
1971 .read_packet = ff_rtsp_fetch_packet,
1972 .read_close = sdp_read_close,
1973 .priv_class = &sdp_demuxer_class,
1975 #endif /* CONFIG_SDP_DEMUXER */
1977 #if CONFIG_RTP_DEMUXER
1978 static int rtp_probe(AVProbeData *p)
1980 if (av_strstart(p->filename, "rtp:", NULL))
1981 return AVPROBE_SCORE_MAX;
1985 static int rtp_read_header(AVFormatContext *s)
1987 uint8_t recvbuf[1500];
1988 char host[500], sdp[500];
1990 URLContext* in = NULL;
1992 AVCodecContext codec = { 0 };
1993 struct sockaddr_storage addr;
1995 socklen_t addrlen = sizeof(addr);
1996 RTSPState *rt = s->priv_data;
1998 if (!ff_network_init())
1999 return AVERROR(EIO);
2001 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2002 &s->interrupt_callback, NULL);
2007 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2008 if (ret == AVERROR(EAGAIN))
2013 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2017 if ((recvbuf[0] & 0xc0) != 0x80) {
2018 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2023 if (RTP_PT_IS_RTCP(recvbuf[1]))
2026 payload_type = recvbuf[1] & 0x7f;
2029 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2033 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2034 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2035 "without an SDP file describing it\n",
2039 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2040 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2041 "properly you need an SDP file "
2045 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2046 NULL, 0, s->filename);
2048 snprintf(sdp, sizeof(sdp),
2049 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2050 addr.ss_family == AF_INET ? 4 : 6, host,
2051 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2052 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2053 port, payload_type);
2054 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2056 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2059 /* sdp_read_header initializes this again */
2062 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2064 ret = sdp_read_header(s);
2075 static const AVClass rtp_demuxer_class = {
2076 .class_name = "RTP demuxer",
2077 .item_name = av_default_item_name,
2078 .option = rtp_options,
2079 .version = LIBAVUTIL_VERSION_INT,
2082 AVInputFormat ff_rtp_demuxer = {
2084 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2085 .priv_data_size = sizeof(RTSPState),
2086 .read_probe = rtp_probe,
2087 .read_header = rtp_read_header,
2088 .read_packet = ff_rtsp_fetch_packet,
2089 .read_close = sdp_read_close,
2090 .flags = AVFMT_NOFILE,
2091 .priv_class = &rtp_demuxer_class,
2093 #endif /* CONFIG_RTP_DEMUXER */