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"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 const AVOption ff_rtsp_options[] = {
78 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
79 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
80 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
81 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
82 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
84 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
85 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
86 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
88 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
89 { "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 },
93 static const AVOption sdp_options[] = {
94 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
95 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
99 static const AVOption rtp_options[] = {
100 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
104 static void get_word_until_chars(char *buf, int buf_size,
105 const char *sep, const char **pp)
111 p += strspn(p, SPACE_CHARS);
113 while (!strchr(sep, *p) && *p != '\0') {
114 if ((q - buf) < buf_size - 1)
123 static void get_word_sep(char *buf, int buf_size, const char *sep,
126 if (**pp == '/') (*pp)++;
127 get_word_until_chars(buf, buf_size, sep, pp);
130 static void get_word(char *buf, int buf_size, const char **pp)
132 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
135 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
137 * Used for seeking in the rtp stream.
139 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
143 p += strspn(p, SPACE_CHARS);
144 if (!av_stristart(p, "npt=", &p))
147 *start = AV_NOPTS_VALUE;
148 *end = AV_NOPTS_VALUE;
150 get_word_sep(buf, sizeof(buf), "-", &p);
151 av_parse_time(start, buf, 1);
154 get_word_sep(buf, sizeof(buf), "-", &p);
155 av_parse_time(end, buf, 1);
157 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
158 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
161 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
163 struct addrinfo hints = { 0 }, *ai = NULL;
164 hints.ai_flags = AI_NUMERICHOST;
165 if (getaddrinfo(buf, NULL, &hints, &ai))
167 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
173 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
174 RTSPStream *rtsp_st, AVCodecContext *codec)
178 codec->codec_id = handler->codec_id;
179 rtsp_st->dynamic_handler = handler;
180 if (handler->alloc) {
181 rtsp_st->dynamic_protocol_context = handler->alloc();
182 if (!rtsp_st->dynamic_protocol_context)
183 rtsp_st->dynamic_handler = NULL;
187 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
188 static int sdp_parse_rtpmap(AVFormatContext *s,
189 AVStream *st, RTSPStream *rtsp_st,
190 int payload_type, const char *p)
192 AVCodecContext *codec = st->codec;
198 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
199 * see if we can handle this kind of payload.
200 * The space should normally not be there but some Real streams or
201 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
202 * have a trailing space. */
203 get_word_sep(buf, sizeof(buf), "/ ", &p);
204 if (payload_type < RTP_PT_PRIVATE) {
205 /* We are in a standard case
206 * (from http://www.iana.org/assignments/rtp-parameters). */
207 /* search into AVRtpPayloadTypes[] */
208 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
211 if (codec->codec_id == AV_CODEC_ID_NONE) {
212 RTPDynamicProtocolHandler *handler =
213 ff_rtp_handler_find_by_name(buf, codec->codec_type);
214 init_rtp_handler(handler, rtsp_st, codec);
215 /* If no dynamic handler was found, check with the list of standard
216 * allocated types, if such a stream for some reason happens to
217 * use a private payload type. This isn't handled in rtpdec.c, since
218 * the format name from the rtpmap line never is passed into rtpdec. */
219 if (!rtsp_st->dynamic_handler)
220 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
223 c = avcodec_find_decoder(codec->codec_id);
229 get_word_sep(buf, sizeof(buf), "/", &p);
231 switch (codec->codec_type) {
232 case AVMEDIA_TYPE_AUDIO:
233 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
234 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
235 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
237 codec->sample_rate = i;
238 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
239 get_word_sep(buf, sizeof(buf), "/", &p);
243 // TODO: there is a bug here; if it is a mono stream, and
244 // less than 22000Hz, faad upconverts to stereo and twice
245 // the frequency. No problem, but the sample rate is being
246 // set here by the sdp line. Patch on its way. (rdm)
248 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
250 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
253 case AVMEDIA_TYPE_VIDEO:
254 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
256 avpriv_set_pts_info(st, 32, 1, i);
261 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262 rtsp_st->dynamic_handler->init(s, st->index,
263 rtsp_st->dynamic_protocol_context);
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271 char *value, int value_size)
273 *p += strspn(*p, SPACE_CHARS);
275 get_word_sep(attr, attr_size, "=", p);
278 get_word_sep(value, value_size, ";", p);
286 typedef struct SDPParseState {
288 struct sockaddr_storage default_ip;
290 int skip_media; ///< set if an unknown m= line occurs
293 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
294 int letter, const char *buf)
296 RTSPState *rt = s->priv_data;
297 char buf1[64], st_type[64];
299 enum AVMediaType codec_type;
303 struct sockaddr_storage sdp_ip;
306 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
309 if (s1->skip_media && letter != 'm')
313 get_word(buf1, sizeof(buf1), &p);
314 if (strcmp(buf1, "IN") != 0)
316 get_word(buf1, sizeof(buf1), &p);
317 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
319 get_word_sep(buf1, sizeof(buf1), "/", &p);
320 if (get_sockaddr(buf1, &sdp_ip))
325 get_word_sep(buf1, sizeof(buf1), "/", &p);
328 if (s->nb_streams == 0) {
329 s1->default_ip = sdp_ip;
330 s1->default_ttl = ttl;
332 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
333 rtsp_st->sdp_ip = sdp_ip;
334 rtsp_st->sdp_ttl = ttl;
338 av_dict_set(&s->metadata, "title", p, 0);
341 if (s->nb_streams == 0) {
342 av_dict_set(&s->metadata, "comment", p, 0);
349 codec_type = AVMEDIA_TYPE_UNKNOWN;
350 get_word(st_type, sizeof(st_type), &p);
351 if (!strcmp(st_type, "audio")) {
352 codec_type = AVMEDIA_TYPE_AUDIO;
353 } else if (!strcmp(st_type, "video")) {
354 codec_type = AVMEDIA_TYPE_VIDEO;
355 } else if (!strcmp(st_type, "application")) {
356 codec_type = AVMEDIA_TYPE_DATA;
358 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
362 rtsp_st = av_mallocz(sizeof(RTSPStream));
365 rtsp_st->stream_index = -1;
366 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
368 rtsp_st->sdp_ip = s1->default_ip;
369 rtsp_st->sdp_ttl = s1->default_ttl;
371 get_word(buf1, sizeof(buf1), &p); /* port */
372 rtsp_st->sdp_port = atoi(buf1);
374 get_word(buf1, sizeof(buf1), &p); /* protocol */
375 if (!strcmp(buf1, "udp"))
376 rt->transport = RTSP_TRANSPORT_RAW;
378 /* XXX: handle list of formats */
379 get_word(buf1, sizeof(buf1), &p); /* format list */
380 rtsp_st->sdp_payload_type = atoi(buf1);
382 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
383 /* no corresponding stream */
384 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
385 rt->ts = ff_mpegts_parse_open(s);
386 } else if (rt->server_type == RTSP_SERVER_WMS &&
387 codec_type == AVMEDIA_TYPE_DATA) {
388 /* RTX stream, a stream that carries all the other actual
389 * audio/video streams. Don't expose this to the callers. */
391 st = avformat_new_stream(s, NULL);
394 st->id = rt->nb_rtsp_streams - 1;
395 rtsp_st->stream_index = st->index;
396 st->codec->codec_type = codec_type;
397 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
398 RTPDynamicProtocolHandler *handler;
399 /* if standard payload type, we can find the codec right now */
400 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
401 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
402 st->codec->sample_rate > 0)
403 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
404 /* Even static payload types may need a custom depacketizer */
405 handler = ff_rtp_handler_find_by_id(
406 rtsp_st->sdp_payload_type, st->codec->codec_type);
407 init_rtp_handler(handler, rtsp_st, st->codec);
408 if (handler && handler->init)
409 handler->init(s, st->index,
410 rtsp_st->dynamic_protocol_context);
413 /* put a default control url */
414 av_strlcpy(rtsp_st->control_url, rt->control_uri,
415 sizeof(rtsp_st->control_url));
418 if (av_strstart(p, "control:", &p)) {
419 if (s->nb_streams == 0) {
420 if (!strncmp(p, "rtsp://", 7))
421 av_strlcpy(rt->control_uri, p,
422 sizeof(rt->control_uri));
425 /* get the control url */
426 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
428 /* XXX: may need to add full url resolution */
429 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
431 if (proto[0] == '\0') {
432 /* relative control URL */
433 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
434 av_strlcat(rtsp_st->control_url, "/",
435 sizeof(rtsp_st->control_url));
436 av_strlcat(rtsp_st->control_url, p,
437 sizeof(rtsp_st->control_url));
439 av_strlcpy(rtsp_st->control_url, p,
440 sizeof(rtsp_st->control_url));
442 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
443 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
444 get_word(buf1, sizeof(buf1), &p);
445 payload_type = atoi(buf1);
446 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
447 if (rtsp_st->stream_index >= 0) {
448 st = s->streams[rtsp_st->stream_index];
449 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
451 } else if (av_strstart(p, "fmtp:", &p) ||
452 av_strstart(p, "framesize:", &p)) {
453 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
454 // let dynamic protocol handlers have a stab at the line.
455 get_word(buf1, sizeof(buf1), &p);
456 payload_type = atoi(buf1);
457 for (i = 0; i < rt->nb_rtsp_streams; i++) {
458 rtsp_st = rt->rtsp_streams[i];
459 if (rtsp_st->sdp_payload_type == payload_type &&
460 rtsp_st->dynamic_handler &&
461 rtsp_st->dynamic_handler->parse_sdp_a_line)
462 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
463 rtsp_st->dynamic_protocol_context, buf);
465 } else if (av_strstart(p, "range:", &p)) {
468 // this is so that seeking on a streamed file can work.
469 rtsp_parse_range_npt(p, &start, &end);
470 s->start_time = start;
471 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
472 s->duration = (end == AV_NOPTS_VALUE) ?
473 AV_NOPTS_VALUE : end - start;
474 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
476 rt->transport = RTSP_TRANSPORT_RDT;
477 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
479 st = s->streams[s->nb_streams - 1];
480 st->codec->sample_rate = atoi(p);
482 if (rt->server_type == RTSP_SERVER_WMS)
483 ff_wms_parse_sdp_a_line(s, p);
484 if (s->nb_streams > 0) {
485 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
487 if (rt->server_type == RTSP_SERVER_REAL)
488 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
490 if (rtsp_st->dynamic_handler &&
491 rtsp_st->dynamic_handler->parse_sdp_a_line)
492 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
493 rtsp_st->stream_index,
494 rtsp_st->dynamic_protocol_context, buf);
501 int ff_sdp_parse(AVFormatContext *s, const char *content)
503 RTSPState *rt = s->priv_data;
506 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
507 * contain long SDP lines containing complete ASF Headers (several
508 * kB) or arrays of MDPR (RM stream descriptor) headers plus
509 * "rulebooks" describing their properties. Therefore, the SDP line
512 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
513 * in rtpdec_xiph.c. */
515 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
519 p += strspn(p, SPACE_CHARS);
527 /* get the content */
529 while (*p != '\n' && *p != '\r' && *p != '\0') {
530 if ((q - buf) < sizeof(buf) - 1)
535 sdp_parse_line(s, s1, letter, buf);
537 while (*p != '\n' && *p != '\0')
542 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
543 if (!rt->p) return AVERROR(ENOMEM);
546 #endif /* CONFIG_RTPDEC */
548 void ff_rtsp_undo_setup(AVFormatContext *s)
550 RTSPState *rt = s->priv_data;
553 for (i = 0; i < rt->nb_rtsp_streams; i++) {
554 RTSPStream *rtsp_st = rt->rtsp_streams[i];
557 if (rtsp_st->transport_priv) {
559 AVFormatContext *rtpctx = rtsp_st->transport_priv;
560 av_write_trailer(rtpctx);
561 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
563 avio_close_dyn_buf(rtpctx->pb, &ptr);
566 avio_close(rtpctx->pb);
568 avformat_free_context(rtpctx);
569 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
570 ff_rdt_parse_close(rtsp_st->transport_priv);
571 else if (rt->transport == RTSP_TRANSPORT_RAW && CONFIG_RTPDEC)
572 ff_rtp_parse_close(rtsp_st->transport_priv);
574 rtsp_st->transport_priv = NULL;
575 if (rtsp_st->rtp_handle)
576 ffurl_close(rtsp_st->rtp_handle);
577 rtsp_st->rtp_handle = NULL;
581 /* close and free RTSP streams */
582 void ff_rtsp_close_streams(AVFormatContext *s)
584 RTSPState *rt = s->priv_data;
588 ff_rtsp_undo_setup(s);
589 for (i = 0; i < rt->nb_rtsp_streams; i++) {
590 rtsp_st = rt->rtsp_streams[i];
592 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
593 rtsp_st->dynamic_handler->free(
594 rtsp_st->dynamic_protocol_context);
598 av_free(rt->rtsp_streams);
600 avformat_close_input(&rt->asf_ctx);
602 if (rt->ts && CONFIG_RTPDEC)
603 ff_mpegts_parse_close(rt->ts);
605 av_free(rt->recvbuf);
608 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
610 RTSPState *rt = s->priv_data;
613 /* open the RTP context */
614 if (rtsp_st->stream_index >= 0)
615 st = s->streams[rtsp_st->stream_index];
617 s->ctx_flags |= AVFMTCTX_NOHEADER;
619 if (s->oformat && CONFIG_RTSP_MUXER) {
620 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
622 RTSP_TCP_MAX_PACKET_SIZE);
623 /* Ownership of rtp_handle is passed to the rtp mux context */
624 rtsp_st->rtp_handle = NULL;
627 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
628 return 0; // Don't need to open any parser here
629 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
630 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
631 rtsp_st->dynamic_protocol_context,
632 rtsp_st->dynamic_handler);
633 else if (CONFIG_RTPDEC)
634 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
635 rtsp_st->sdp_payload_type,
636 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
637 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
639 if (!rtsp_st->transport_priv) {
640 return AVERROR(ENOMEM);
641 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
642 if (rtsp_st->dynamic_handler) {
643 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
644 rtsp_st->dynamic_protocol_context,
645 rtsp_st->dynamic_handler);
652 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
653 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
660 q += strspn(q, SPACE_CHARS);
661 v = strtol(q, &p, 10);
665 v = strtol(p, &p, 10);
674 /* XXX: only one transport specification is parsed */
675 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
677 char transport_protocol[16];
679 char lower_transport[16];
681 RTSPTransportField *th;
684 reply->nb_transports = 0;
687 p += strspn(p, SPACE_CHARS);
691 th = &reply->transports[reply->nb_transports];
693 get_word_sep(transport_protocol, sizeof(transport_protocol),
695 if (!av_strcasecmp (transport_protocol, "rtp")) {
696 get_word_sep(profile, sizeof(profile), "/;,", &p);
697 lower_transport[0] = '\0';
698 /* rtp/avp/<protocol> */
700 get_word_sep(lower_transport, sizeof(lower_transport),
703 th->transport = RTSP_TRANSPORT_RTP;
704 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
705 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
706 /* x-pn-tng/<protocol> */
707 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
709 th->transport = RTSP_TRANSPORT_RDT;
710 } else if (!av_strcasecmp(transport_protocol, "raw")) {
711 get_word_sep(profile, sizeof(profile), "/;,", &p);
712 lower_transport[0] = '\0';
713 /* raw/raw/<protocol> */
715 get_word_sep(lower_transport, sizeof(lower_transport),
718 th->transport = RTSP_TRANSPORT_RAW;
720 if (!av_strcasecmp(lower_transport, "TCP"))
721 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
723 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
727 /* get each parameter */
728 while (*p != '\0' && *p != ',') {
729 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
730 if (!strcmp(parameter, "port")) {
733 rtsp_parse_range(&th->port_min, &th->port_max, &p);
735 } else if (!strcmp(parameter, "client_port")) {
738 rtsp_parse_range(&th->client_port_min,
739 &th->client_port_max, &p);
741 } else if (!strcmp(parameter, "server_port")) {
744 rtsp_parse_range(&th->server_port_min,
745 &th->server_port_max, &p);
747 } else if (!strcmp(parameter, "interleaved")) {
750 rtsp_parse_range(&th->interleaved_min,
751 &th->interleaved_max, &p);
753 } else if (!strcmp(parameter, "multicast")) {
754 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
755 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
756 } else if (!strcmp(parameter, "ttl")) {
759 th->ttl = strtol(p, (char **)&p, 10);
761 } else if (!strcmp(parameter, "destination")) {
764 get_word_sep(buf, sizeof(buf), ";,", &p);
765 get_sockaddr(buf, &th->destination);
767 } else if (!strcmp(parameter, "source")) {
770 get_word_sep(buf, sizeof(buf), ";,", &p);
771 av_strlcpy(th->source, buf, sizeof(th->source));
773 } else if (!strcmp(parameter, "mode")) {
776 get_word_sep(buf, sizeof(buf), ";, ", &p);
777 if (!strcmp(buf, "record") ||
778 !strcmp(buf, "receive"))
783 while (*p != ';' && *p != '\0' && *p != ',')
791 reply->nb_transports++;
795 static void handle_rtp_info(RTSPState *rt, const char *url,
796 uint32_t seq, uint32_t rtptime)
799 if (!rtptime || !url[0])
801 if (rt->transport != RTSP_TRANSPORT_RTP)
803 for (i = 0; i < rt->nb_rtsp_streams; i++) {
804 RTSPStream *rtsp_st = rt->rtsp_streams[i];
805 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
808 if (!strcmp(rtsp_st->control_url, url)) {
809 rtpctx->base_timestamp = rtptime;
815 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
818 char key[20], value[1024], url[1024] = "";
819 uint32_t seq = 0, rtptime = 0;
822 p += strspn(p, SPACE_CHARS);
825 get_word_sep(key, sizeof(key), "=", &p);
829 get_word_sep(value, sizeof(value), ";, ", &p);
831 if (!strcmp(key, "url"))
832 av_strlcpy(url, value, sizeof(url));
833 else if (!strcmp(key, "seq"))
834 seq = strtoul(value, NULL, 10);
835 else if (!strcmp(key, "rtptime"))
836 rtptime = strtoul(value, NULL, 10);
838 handle_rtp_info(rt, url, seq, rtptime);
847 handle_rtp_info(rt, url, seq, rtptime);
850 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
851 RTSPState *rt, const char *method)
855 /* NOTE: we do case independent match for broken servers */
857 if (av_stristart(p, "Session:", &p)) {
859 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
860 if (av_stristart(p, ";timeout=", &p) &&
861 (t = strtol(p, NULL, 10)) > 0) {
864 } else if (av_stristart(p, "Content-Length:", &p)) {
865 reply->content_length = strtol(p, NULL, 10);
866 } else if (av_stristart(p, "Transport:", &p)) {
867 rtsp_parse_transport(reply, p);
868 } else if (av_stristart(p, "CSeq:", &p)) {
869 reply->seq = strtol(p, NULL, 10);
870 } else if (av_stristart(p, "Range:", &p)) {
871 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
872 } else if (av_stristart(p, "RealChallenge1:", &p)) {
873 p += strspn(p, SPACE_CHARS);
874 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
875 } else if (av_stristart(p, "Server:", &p)) {
876 p += strspn(p, SPACE_CHARS);
877 av_strlcpy(reply->server, p, sizeof(reply->server));
878 } else if (av_stristart(p, "Notice:", &p) ||
879 av_stristart(p, "X-Notice:", &p)) {
880 reply->notice = strtol(p, NULL, 10);
881 } else if (av_stristart(p, "Location:", &p)) {
882 p += strspn(p, SPACE_CHARS);
883 av_strlcpy(reply->location, p , sizeof(reply->location));
884 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
885 p += strspn(p, SPACE_CHARS);
886 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
887 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
888 p += strspn(p, SPACE_CHARS);
889 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
890 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
891 p += strspn(p, SPACE_CHARS);
892 if (method && !strcmp(method, "DESCRIBE"))
893 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
894 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
895 p += strspn(p, SPACE_CHARS);
896 if (method && !strcmp(method, "PLAY"))
897 rtsp_parse_rtp_info(rt, p);
898 } else if (av_stristart(p, "Public:", &p) && rt) {
899 if (strstr(p, "GET_PARAMETER") &&
900 method && !strcmp(method, "OPTIONS"))
901 rt->get_parameter_supported = 1;
902 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
903 p += strspn(p, SPACE_CHARS);
904 rt->accept_dynamic_rate = atoi(p);
905 } else if (av_stristart(p, "Content-Type:", &p)) {
906 p += strspn(p, SPACE_CHARS);
907 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
911 /* skip a RTP/TCP interleaved packet */
912 void ff_rtsp_skip_packet(AVFormatContext *s)
914 RTSPState *rt = s->priv_data;
918 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
921 len = AV_RB16(buf + 1);
923 av_dlog(s, "skipping RTP packet len=%d\n", len);
928 if (len1 > sizeof(buf))
930 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
937 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
938 unsigned char **content_ptr,
939 int return_on_interleaved_data, const char *method)
941 RTSPState *rt = s->priv_data;
942 char buf[4096], buf1[1024], *q;
945 int ret, content_length, line_count = 0, request = 0;
946 unsigned char *content = NULL;
952 memset(reply, 0, sizeof(*reply));
954 /* parse reply (XXX: use buffers) */
955 rt->last_reply[0] = '\0';
959 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
960 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
966 /* XXX: only parse it if first char on line ? */
967 if (return_on_interleaved_data) {
970 ff_rtsp_skip_packet(s);
971 } else if (ch != '\r') {
972 if ((q - buf) < sizeof(buf) - 1)
978 av_dlog(s, "line='%s'\n", buf);
980 /* test if last line */
984 if (line_count == 0) {
986 get_word(buf1, sizeof(buf1), &p);
987 if (!strncmp(buf1, "RTSP/", 5)) {
988 get_word(buf1, sizeof(buf1), &p);
989 reply->status_code = atoi(buf1);
990 av_strlcpy(reply->reason, p, sizeof(reply->reason));
992 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
993 get_word(buf1, sizeof(buf1), &p); // object
997 ff_rtsp_parse_line(reply, p, rt, method);
998 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
999 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1004 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1005 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1007 content_length = reply->content_length;
1008 if (content_length > 0) {
1009 /* leave some room for a trailing '\0' (useful for simple parsing) */
1010 content = av_malloc(content_length + 1);
1011 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1012 content[content_length] = '\0';
1015 *content_ptr = content;
1021 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1022 const char* ptr = buf;
1024 if (!strcmp(reply->reason, "OPTIONS")) {
1025 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1027 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1028 if (reply->session_id[0])
1029 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1032 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1034 av_strlcat(buf, "\r\n", sizeof(buf));
1036 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1037 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1040 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1042 rt->last_cmd_time = av_gettime();
1043 /* Even if the request from the server had data, it is not the data
1044 * that the caller wants or expects. The memory could also be leaked
1045 * if the actual following reply has content data. */
1047 av_freep(content_ptr);
1048 /* If method is set, this is called from ff_rtsp_send_cmd,
1049 * where a reply to exactly this request is awaited. For
1050 * callers from within packet receiving, we just want to
1051 * return to the caller and go back to receiving packets. */
1057 if (rt->seq != reply->seq) {
1058 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1059 rt->seq, reply->seq);
1063 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1064 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1065 reply->notice == 2306 /* Continuous Feed Terminated */) {
1066 rt->state = RTSP_STATE_IDLE;
1067 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1068 return AVERROR(EIO); /* data or server error */
1069 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1070 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1071 return AVERROR(EPERM);
1077 * Send a command to the RTSP server without waiting for the reply.
1079 * @param s RTSP (de)muxer context
1080 * @param method the method for the request
1081 * @param url the target url for the request
1082 * @param headers extra header lines to include in the request
1083 * @param send_content if non-null, the data to send as request body content
1084 * @param send_content_length the length of the send_content data, or 0 if
1085 * send_content is null
1087 * @return zero if success, nonzero otherwise
1089 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1090 const char *method, const char *url,
1091 const char *headers,
1092 const unsigned char *send_content,
1093 int send_content_length)
1095 RTSPState *rt = s->priv_data;
1096 char buf[4096], *out_buf;
1097 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1099 /* Add in RTSP headers */
1102 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1104 av_strlcat(buf, headers, sizeof(buf));
1105 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1106 if (rt->session_id[0] != '\0' && (!headers ||
1107 !strstr(headers, "\nIf-Match:"))) {
1108 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1111 char *str = ff_http_auth_create_response(&rt->auth_state,
1112 rt->auth, url, method);
1114 av_strlcat(buf, str, sizeof(buf));
1117 if (send_content_length > 0 && send_content)
1118 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1119 av_strlcat(buf, "\r\n", sizeof(buf));
1121 /* base64 encode rtsp if tunneling */
1122 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1123 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1124 out_buf = base64buf;
1127 av_dlog(s, "Sending:\n%s--\n", buf);
1129 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1130 if (send_content_length > 0 && send_content) {
1131 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1132 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1133 "with content data not supported\n");
1134 return AVERROR_PATCHWELCOME;
1136 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1138 rt->last_cmd_time = av_gettime();
1143 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1144 const char *url, const char *headers)
1146 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1149 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1150 const char *headers, RTSPMessageHeader *reply,
1151 unsigned char **content_ptr)
1153 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1154 content_ptr, NULL, 0);
1157 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1158 const char *method, const char *url,
1160 RTSPMessageHeader *reply,
1161 unsigned char **content_ptr,
1162 const unsigned char *send_content,
1163 int send_content_length)
1165 RTSPState *rt = s->priv_data;
1166 HTTPAuthType cur_auth_type;
1167 int ret, attempts = 0;
1170 cur_auth_type = rt->auth_state.auth_type;
1171 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1173 send_content_length)))
1176 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1180 if (reply->status_code == 401 &&
1181 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1182 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1185 if (reply->status_code > 400){
1186 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1190 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1196 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1197 int lower_transport, const char *real_challenge)
1199 RTSPState *rt = s->priv_data;
1200 int rtx = 0, j, i, err, interleave = 0, port_off;
1201 RTSPStream *rtsp_st;
1202 RTSPMessageHeader reply1, *reply = &reply1;
1204 const char *trans_pref;
1206 if (rt->transport == RTSP_TRANSPORT_RDT)
1207 trans_pref = "x-pn-tng";
1208 else if (rt->transport == RTSP_TRANSPORT_RAW)
1209 trans_pref = "RAW/RAW";
1211 trans_pref = "RTP/AVP";
1213 /* default timeout: 1 minute */
1216 /* for each stream, make the setup request */
1217 /* XXX: we assume the same server is used for the control of each
1220 /* Choose a random starting offset within the first half of the
1221 * port range, to allow for a number of ports to try even if the offset
1222 * happens to be at the end of the random range. */
1223 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1224 /* even random offset */
1225 port_off -= port_off & 0x01;
1227 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1228 char transport[2048];
1231 * WMS serves all UDP data over a single connection, the RTX, which
1232 * isn't necessarily the first in the SDP but has to be the first
1233 * to be set up, else the second/third SETUP will fail with a 461.
1235 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1236 rt->server_type == RTSP_SERVER_WMS) {
1239 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1240 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1242 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1246 if (rtx == rt->nb_rtsp_streams)
1247 return -1; /* no RTX found */
1248 rtsp_st = rt->rtsp_streams[rtx];
1250 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1252 rtsp_st = rt->rtsp_streams[i];
1255 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1258 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1259 port = reply->transports[0].client_port_min;
1263 /* first try in specified port range */
1264 while (j <= rt->rtp_port_max) {
1265 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1266 "?localport=%d", j);
1267 /* we will use two ports per rtp stream (rtp and rtcp) */
1269 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1270 &s->interrupt_callback, NULL))
1274 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1279 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1281 snprintf(transport, sizeof(transport) - 1,
1282 "%s/UDP;", trans_pref);
1283 if (rt->server_type != RTSP_SERVER_REAL)
1284 av_strlcat(transport, "unicast;", sizeof(transport));
1285 av_strlcatf(transport, sizeof(transport),
1286 "client_port=%d", port);
1287 if (rt->transport == RTSP_TRANSPORT_RTP &&
1288 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1289 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1293 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1294 /* For WMS streams, the application streams are only used for
1295 * UDP. When trying to set it up for TCP streams, the server
1296 * will return an error. Therefore, we skip those streams. */
1297 if (rt->server_type == RTSP_SERVER_WMS &&
1298 (rtsp_st->stream_index < 0 ||
1299 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1302 snprintf(transport, sizeof(transport) - 1,
1303 "%s/TCP;", trans_pref);
1304 if (rt->transport != RTSP_TRANSPORT_RDT)
1305 av_strlcat(transport, "unicast;", sizeof(transport));
1306 av_strlcatf(transport, sizeof(transport),
1307 "interleaved=%d-%d",
1308 interleave, interleave + 1);
1312 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1313 snprintf(transport, sizeof(transport) - 1,
1314 "%s/UDP;multicast", trans_pref);
1317 av_strlcat(transport, ";mode=record", sizeof(transport));
1318 } else if (rt->server_type == RTSP_SERVER_REAL ||
1319 rt->server_type == RTSP_SERVER_WMS)
1320 av_strlcat(transport, ";mode=play", sizeof(transport));
1321 snprintf(cmd, sizeof(cmd),
1322 "Transport: %s\r\n",
1324 if (rt->accept_dynamic_rate)
1325 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1326 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1327 char real_res[41], real_csum[9];
1328 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1330 av_strlcatf(cmd, sizeof(cmd),
1332 "RealChallenge2: %s, sd=%s\r\n",
1333 rt->session_id, real_res, real_csum);
1335 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1336 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1339 } else if (reply->status_code != RTSP_STATUS_OK ||
1340 reply->nb_transports != 1) {
1341 err = AVERROR_INVALIDDATA;
1345 /* XXX: same protocol for all streams is required */
1347 if (reply->transports[0].lower_transport != rt->lower_transport ||
1348 reply->transports[0].transport != rt->transport) {
1349 err = AVERROR_INVALIDDATA;
1353 rt->lower_transport = reply->transports[0].lower_transport;
1354 rt->transport = reply->transports[0].transport;
1357 /* Fail if the server responded with another lower transport mode
1358 * than what we requested. */
1359 if (reply->transports[0].lower_transport != lower_transport) {
1360 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1361 err = AVERROR_INVALIDDATA;
1365 switch(reply->transports[0].lower_transport) {
1366 case RTSP_LOWER_TRANSPORT_TCP:
1367 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1368 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1371 case RTSP_LOWER_TRANSPORT_UDP: {
1372 char url[1024], options[30] = "";
1374 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1375 av_strlcpy(options, "?connect=1", sizeof(options));
1376 /* Use source address if specified */
1377 if (reply->transports[0].source[0]) {
1378 ff_url_join(url, sizeof(url), "rtp", NULL,
1379 reply->transports[0].source,
1380 reply->transports[0].server_port_min, "%s", options);
1382 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1383 reply->transports[0].server_port_min, "%s", options);
1385 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1386 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1387 err = AVERROR_INVALIDDATA;
1390 /* Try to initialize the connection state in a
1391 * potential NAT router by sending dummy packets.
1392 * RTP/RTCP dummy packets are used for RDT, too.
1394 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1396 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1399 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1400 char url[1024], namebuf[50], optbuf[20] = "";
1401 struct sockaddr_storage addr;
1404 if (reply->transports[0].destination.ss_family) {
1405 addr = reply->transports[0].destination;
1406 port = reply->transports[0].port_min;
1407 ttl = reply->transports[0].ttl;
1409 addr = rtsp_st->sdp_ip;
1410 port = rtsp_st->sdp_port;
1411 ttl = rtsp_st->sdp_ttl;
1414 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1415 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1416 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1417 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1418 port, "%s", optbuf);
1419 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1420 &s->interrupt_callback, NULL) < 0) {
1421 err = AVERROR_INVALIDDATA;
1428 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1432 if (rt->nb_rtsp_streams && reply->timeout > 0)
1433 rt->timeout = reply->timeout;
1435 if (rt->server_type == RTSP_SERVER_REAL)
1436 rt->need_subscription = 1;
1441 ff_rtsp_undo_setup(s);
1445 void ff_rtsp_close_connections(AVFormatContext *s)
1447 RTSPState *rt = s->priv_data;
1448 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1449 ffurl_close(rt->rtsp_hd);
1450 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1453 int ff_rtsp_connect(AVFormatContext *s)
1455 RTSPState *rt = s->priv_data;
1456 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1457 int port, err, tcp_fd;
1458 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1459 int lower_transport_mask = 0;
1460 char real_challenge[64] = "";
1461 struct sockaddr_storage peer;
1462 socklen_t peer_len = sizeof(peer);
1464 if (rt->rtp_port_max < rt->rtp_port_min) {
1465 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1466 "than min port %d\n", rt->rtp_port_max,
1468 return AVERROR(EINVAL);
1471 if (!ff_network_init())
1472 return AVERROR(EIO);
1474 if (s->max_delay < 0) /* Not set by the caller */
1475 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1477 rt->control_transport = RTSP_MODE_PLAIN;
1478 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1479 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1480 rt->control_transport = RTSP_MODE_TUNNEL;
1482 /* Only pass through valid flags from here */
1483 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1486 lower_transport_mask = rt->lower_transport_mask;
1487 /* extract hostname and port */
1488 av_url_split(NULL, 0, auth, sizeof(auth),
1489 host, sizeof(host), &port, path, sizeof(path), s->filename);
1491 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1494 port = RTSP_DEFAULT_PORT;
1496 if (!lower_transport_mask)
1497 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1500 /* Only UDP or TCP - UDP multicast isn't supported. */
1501 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1502 (1 << RTSP_LOWER_TRANSPORT_TCP);
1503 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1504 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1505 "only UDP and TCP are supported for output.\n");
1506 err = AVERROR(EINVAL);
1511 /* Construct the URI used in request; this is similar to s->filename,
1512 * but with authentication credentials removed and RTSP specific options
1514 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1515 host, port, "%s", path);
1517 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1518 /* set up initial handshake for tunneling */
1519 char httpname[1024];
1520 char sessioncookie[17];
1523 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1524 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1525 av_get_random_seed(), av_get_random_seed());
1528 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1529 &s->interrupt_callback) < 0) {
1534 /* generate GET headers */
1535 snprintf(headers, sizeof(headers),
1536 "x-sessioncookie: %s\r\n"
1537 "Accept: application/x-rtsp-tunnelled\r\n"
1538 "Pragma: no-cache\r\n"
1539 "Cache-Control: no-cache\r\n",
1541 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1543 /* complete the connection */
1544 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1550 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1551 &s->interrupt_callback) < 0 ) {
1556 /* generate POST headers */
1557 snprintf(headers, sizeof(headers),
1558 "x-sessioncookie: %s\r\n"
1559 "Content-Type: application/x-rtsp-tunnelled\r\n"
1560 "Pragma: no-cache\r\n"
1561 "Cache-Control: no-cache\r\n"
1562 "Content-Length: 32767\r\n"
1563 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1565 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1566 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1568 /* Initialize the authentication state for the POST session. The HTTP
1569 * protocol implementation doesn't properly handle multi-pass
1570 * authentication for POST requests, since it would require one of
1572 * - implementing Expect: 100-continue, which many HTTP servers
1573 * don't support anyway, even less the RTSP servers that do HTTP
1575 * - sending the whole POST data until getting a 401 reply specifying
1576 * what authentication method to use, then resending all that data
1577 * - waiting for potential 401 replies directly after sending the
1578 * POST header (waiting for some unspecified time)
1579 * Therefore, we copy the full auth state, which works for both basic
1580 * and digest. (For digest, we would have to synchronize the nonce
1581 * count variable between the two sessions, if we'd do more requests
1582 * with the original session, though.)
1584 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1586 /* complete the connection */
1587 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1592 /* open the tcp connection */
1593 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1594 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1595 &s->interrupt_callback, NULL) < 0) {
1599 rt->rtsp_hd_out = rt->rtsp_hd;
1603 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1604 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1605 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1606 NULL, 0, NI_NUMERICHOST);
1609 /* request options supported by the server; this also detects server
1611 for (rt->server_type = RTSP_SERVER_RTP;;) {
1613 if (rt->server_type == RTSP_SERVER_REAL)
1616 * The following entries are required for proper
1617 * streaming from a Realmedia server. They are
1618 * interdependent in some way although we currently
1619 * don't quite understand how. Values were copied
1620 * from mplayer SVN r23589.
1621 * ClientChallenge is a 16-byte ID in hex
1622 * CompanyID is a 16-byte ID in base64
1624 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1625 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1626 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1627 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1629 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1630 if (reply->status_code != RTSP_STATUS_OK) {
1631 err = AVERROR_INVALIDDATA;
1635 /* detect server type if not standard-compliant RTP */
1636 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1637 rt->server_type = RTSP_SERVER_REAL;
1639 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1640 rt->server_type = RTSP_SERVER_WMS;
1641 } else if (rt->server_type == RTSP_SERVER_REAL)
1642 strcpy(real_challenge, reply->real_challenge);
1646 if (s->iformat && CONFIG_RTSP_DEMUXER)
1647 err = ff_rtsp_setup_input_streams(s, reply);
1648 else if (CONFIG_RTSP_MUXER)
1649 err = ff_rtsp_setup_output_streams(s, host);
1654 int lower_transport = ff_log2_tab[lower_transport_mask &
1655 ~(lower_transport_mask - 1)];
1657 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1658 rt->server_type == RTSP_SERVER_REAL ?
1659 real_challenge : NULL);
1662 lower_transport_mask &= ~(1 << lower_transport);
1663 if (lower_transport_mask == 0 && err == 1) {
1664 err = AVERROR(EPROTONOSUPPORT);
1669 rt->lower_transport_mask = lower_transport_mask;
1670 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1671 rt->state = RTSP_STATE_IDLE;
1672 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1675 ff_rtsp_close_streams(s);
1676 ff_rtsp_close_connections(s);
1677 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1678 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1679 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1687 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1690 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1691 uint8_t *buf, int buf_size, int64_t wait_end)
1693 RTSPState *rt = s->priv_data;
1694 RTSPStream *rtsp_st;
1695 int n, i, ret, tcp_fd, timeout_cnt = 0;
1697 struct pollfd *p = rt->p;
1698 int *fds = NULL, fdsnum, fdsidx;
1701 if (ff_check_interrupt(&s->interrupt_callback))
1702 return AVERROR_EXIT;
1703 if (wait_end && wait_end - av_gettime() < 0)
1704 return AVERROR(EAGAIN);
1707 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1708 p[max_p].fd = tcp_fd;
1709 p[max_p++].events = POLLIN;
1713 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1714 rtsp_st = rt->rtsp_streams[i];
1715 if (rtsp_st->rtp_handle) {
1716 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1718 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1722 av_log(s, AV_LOG_ERROR,
1723 "Number of fds %d not supported\n", fdsnum);
1724 return AVERROR_INVALIDDATA;
1726 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1727 p[max_p].fd = fds[fdsidx];
1728 p[max_p++].events = POLLIN;
1733 n = poll(p, max_p, POLL_TIMEOUT_MS);
1735 int j = 1 - (tcp_fd == -1);
1737 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1738 rtsp_st = rt->rtsp_streams[i];
1739 if (rtsp_st->rtp_handle) {
1740 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1741 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1743 *prtsp_st = rtsp_st;
1750 #if CONFIG_RTSP_DEMUXER
1751 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1752 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1753 if (rt->state == RTSP_STATE_STREAMING) {
1754 if (!ff_rtsp_parse_streaming_commands(s))
1757 av_log(s, AV_LOG_WARNING,
1758 "Unable to answer to TEARDOWN\n");
1762 RTSPMessageHeader reply;
1763 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1766 /* XXX: parse message */
1767 if (rt->state != RTSP_STATE_STREAMING)
1772 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1773 return AVERROR(ETIMEDOUT);
1774 } else if (n < 0 && errno != EINTR)
1775 return AVERROR(errno);
1779 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1781 RTSPState *rt = s->priv_data;
1783 RTSPStream *rtsp_st, *first_queue_st = NULL;
1784 int64_t wait_end = 0;
1786 if (rt->nb_byes == rt->nb_rtsp_streams)
1789 /* get next frames from the same RTP packet */
1790 if (rt->cur_transport_priv) {
1791 if (rt->transport == RTSP_TRANSPORT_RDT) {
1792 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1793 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1794 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1795 } else if (rt->ts && CONFIG_RTPDEC) {
1796 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1798 rt->recvbuf_pos += ret;
1799 ret = rt->recvbuf_pos < rt->recvbuf_len;
1803 rt->cur_transport_priv = NULL;
1805 } else if (ret == 1) {
1808 rt->cur_transport_priv = NULL;
1811 if (rt->transport == RTSP_TRANSPORT_RTP) {
1813 int64_t first_queue_time = 0;
1814 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1815 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1819 queue_time = ff_rtp_queued_packet_time(rtpctx);
1820 if (queue_time && (queue_time - first_queue_time < 0 ||
1821 !first_queue_time)) {
1822 first_queue_time = queue_time;
1823 first_queue_st = rt->rtsp_streams[i];
1826 if (first_queue_time)
1827 wait_end = first_queue_time + s->max_delay;
1830 /* read next RTP packet */
1833 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1835 return AVERROR(ENOMEM);
1838 switch(rt->lower_transport) {
1840 #if CONFIG_RTSP_DEMUXER
1841 case RTSP_LOWER_TRANSPORT_TCP:
1842 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1845 case RTSP_LOWER_TRANSPORT_UDP:
1846 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1847 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1848 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1849 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1852 if (len == AVERROR(EAGAIN) && first_queue_st &&
1853 rt->transport == RTSP_TRANSPORT_RTP) {
1854 rtsp_st = first_queue_st;
1855 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1862 if (rt->transport == RTSP_TRANSPORT_RDT) {
1863 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1864 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1865 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1867 /* Either bad packet, or a RTCP packet. Check if the
1868 * first_rtcp_ntp_time field was initialized. */
1869 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1870 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1871 /* first_rtcp_ntp_time has been initialized for this stream,
1872 * copy the same value to all other uninitialized streams,
1873 * in order to map their timestamp origin to the same ntp time
1876 AVStream *st = NULL;
1877 if (rtsp_st->stream_index >= 0)
1878 st = s->streams[rtsp_st->stream_index];
1879 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1880 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1881 AVStream *st2 = NULL;
1882 if (rt->rtsp_streams[i]->stream_index >= 0)
1883 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1884 if (rtpctx2 && st && st2 &&
1885 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1886 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1887 rtpctx2->rtcp_ts_offset = av_rescale_q(
1888 rtpctx->rtcp_ts_offset, st->time_base,
1893 if (ret == -RTCP_BYE) {
1896 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1897 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1899 if (rt->nb_byes == rt->nb_rtsp_streams)
1903 } else if (rt->ts && CONFIG_RTPDEC) {
1904 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1907 rt->recvbuf_len = len;
1908 rt->recvbuf_pos = ret;
1909 rt->cur_transport_priv = rt->ts;
1916 return AVERROR_INVALIDDATA;
1922 /* more packets may follow, so we save the RTP context */
1923 rt->cur_transport_priv = rtsp_st->transport_priv;
1927 #endif /* CONFIG_RTPDEC */
1929 #if CONFIG_SDP_DEMUXER
1930 static int sdp_probe(AVProbeData *p1)
1932 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1934 /* we look for a line beginning "c=IN IP" */
1935 while (p < p_end && *p != '\0') {
1936 if (p + sizeof("c=IN IP") - 1 < p_end &&
1937 av_strstart(p, "c=IN IP", NULL))
1938 return AVPROBE_SCORE_MAX / 2;
1940 while (p < p_end - 1 && *p != '\n') p++;
1949 static int sdp_read_header(AVFormatContext *s)
1951 RTSPState *rt = s->priv_data;
1952 RTSPStream *rtsp_st;
1957 if (!ff_network_init())
1958 return AVERROR(EIO);
1960 if (s->max_delay < 0) /* Not set by the caller */
1961 s->max_delay = DEFAULT_REORDERING_DELAY;
1963 /* read the whole sdp file */
1964 /* XXX: better loading */
1965 content = av_malloc(SDP_MAX_SIZE);
1966 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1969 return AVERROR_INVALIDDATA;
1971 content[size] ='\0';
1973 err = ff_sdp_parse(s, content);
1977 /* open each RTP stream */
1978 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1980 rtsp_st = rt->rtsp_streams[i];
1982 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1983 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1984 ff_url_join(url, sizeof(url), "rtp", NULL,
1985 namebuf, rtsp_st->sdp_port,
1986 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1988 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1989 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1990 &s->interrupt_callback, NULL) < 0) {
1991 err = AVERROR_INVALIDDATA;
1994 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1999 ff_rtsp_close_streams(s);
2004 static int sdp_read_close(AVFormatContext *s)
2006 ff_rtsp_close_streams(s);
2011 static const AVClass sdp_demuxer_class = {
2012 .class_name = "SDP demuxer",
2013 .item_name = av_default_item_name,
2014 .option = sdp_options,
2015 .version = LIBAVUTIL_VERSION_INT,
2018 AVInputFormat ff_sdp_demuxer = {
2020 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2021 .priv_data_size = sizeof(RTSPState),
2022 .read_probe = sdp_probe,
2023 .read_header = sdp_read_header,
2024 .read_packet = ff_rtsp_fetch_packet,
2025 .read_close = sdp_read_close,
2026 .priv_class = &sdp_demuxer_class,
2028 #endif /* CONFIG_SDP_DEMUXER */
2030 #if CONFIG_RTP_DEMUXER
2031 static int rtp_probe(AVProbeData *p)
2033 if (av_strstart(p->filename, "rtp:", NULL))
2034 return AVPROBE_SCORE_MAX;
2038 static int rtp_read_header(AVFormatContext *s)
2040 uint8_t recvbuf[1500];
2041 char host[500], sdp[500];
2043 URLContext* in = NULL;
2045 AVCodecContext codec = { 0 };
2046 struct sockaddr_storage addr;
2048 socklen_t addrlen = sizeof(addr);
2049 RTSPState *rt = s->priv_data;
2051 if (!ff_network_init())
2052 return AVERROR(EIO);
2054 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2055 &s->interrupt_callback, NULL);
2060 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2061 if (ret == AVERROR(EAGAIN))
2066 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2070 if ((recvbuf[0] & 0xc0) != 0x80) {
2071 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2076 if (RTP_PT_IS_RTCP(recvbuf[1]))
2079 payload_type = recvbuf[1] & 0x7f;
2082 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2086 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2087 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2088 "without an SDP file describing it\n",
2092 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2093 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2094 "properly you need an SDP file "
2098 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2099 NULL, 0, s->filename);
2101 snprintf(sdp, sizeof(sdp),
2102 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2103 addr.ss_family == AF_INET ? 4 : 6, host,
2104 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2105 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2106 port, payload_type);
2107 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2109 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2112 /* sdp_read_header initializes this again */
2115 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2117 ret = sdp_read_header(s);
2128 static const AVClass rtp_demuxer_class = {
2129 .class_name = "RTP demuxer",
2130 .item_name = av_default_item_name,
2131 .option = rtp_options,
2132 .version = LIBAVUTIL_VERSION_INT,
2135 AVInputFormat ff_rtp_demuxer = {
2137 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2138 .priv_data_size = sizeof(RTSPState),
2139 .read_probe = rtp_probe,
2140 .read_header = rtp_read_header,
2141 .read_packet = ff_rtsp_fetch_packet,
2142 .read_close = sdp_read_close,
2143 .flags = AVFMT_NOFILE,
2144 .priv_class = &rtp_demuxer_class,
2146 #endif /* CONFIG_RTP_DEMUXER */