3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
70 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
74 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
78 const AVOption ff_rtsp_options[] = {
79 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
80 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
81 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
82 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
84 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
85 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
86 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
87 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
88 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
89 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
90 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
94 static const AVOption sdp_options[] = {
95 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
96 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
100 static const AVOption rtp_options[] = {
101 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
105 static void get_word_until_chars(char *buf, int buf_size,
106 const char *sep, const char **pp)
112 p += strspn(p, SPACE_CHARS);
114 while (!strchr(sep, *p) && *p != '\0') {
115 if ((q - buf) < buf_size - 1)
124 static void get_word_sep(char *buf, int buf_size, const char *sep,
127 if (**pp == '/') (*pp)++;
128 get_word_until_chars(buf, buf_size, sep, pp);
131 static void get_word(char *buf, int buf_size, const char **pp)
133 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
136 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
138 * Used for seeking in the rtp stream.
140 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
144 p += strspn(p, SPACE_CHARS);
145 if (!av_stristart(p, "npt=", &p))
148 *start = AV_NOPTS_VALUE;
149 *end = AV_NOPTS_VALUE;
151 get_word_sep(buf, sizeof(buf), "-", &p);
152 av_parse_time(start, buf, 1);
155 get_word_sep(buf, sizeof(buf), "-", &p);
156 av_parse_time(end, buf, 1);
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 == AV_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 */
374 if (!strcmp(buf1, "udp"))
375 rt->transport = RTSP_TRANSPORT_RAW;
377 /* XXX: handle list of formats */
378 get_word(buf1, sizeof(buf1), &p); /* format list */
379 rtsp_st->sdp_payload_type = atoi(buf1);
381 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
382 /* no corresponding stream */
383 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
384 rt->ts = ff_mpegts_parse_open(s);
385 } else if (rt->server_type == RTSP_SERVER_WMS &&
386 codec_type == AVMEDIA_TYPE_DATA) {
387 /* RTX stream, a stream that carries all the other actual
388 * audio/video streams. Don't expose this to the callers. */
390 st = avformat_new_stream(s, NULL);
393 st->id = rt->nb_rtsp_streams - 1;
394 rtsp_st->stream_index = st->index;
395 st->codec->codec_type = codec_type;
396 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
397 RTPDynamicProtocolHandler *handler;
398 /* if standard payload type, we can find the codec right now */
399 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
400 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
401 st->codec->sample_rate > 0)
402 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
403 /* Even static payload types may need a custom depacketizer */
404 handler = ff_rtp_handler_find_by_id(
405 rtsp_st->sdp_payload_type, st->codec->codec_type);
406 init_rtp_handler(handler, rtsp_st, st->codec);
407 if (handler && handler->init)
408 handler->init(s, st->index,
409 rtsp_st->dynamic_protocol_context);
412 /* put a default control url */
413 av_strlcpy(rtsp_st->control_url, rt->control_uri,
414 sizeof(rtsp_st->control_url));
417 if (av_strstart(p, "control:", &p)) {
418 if (s->nb_streams == 0) {
419 if (!strncmp(p, "rtsp://", 7))
420 av_strlcpy(rt->control_uri, p,
421 sizeof(rt->control_uri));
424 /* get the control url */
425 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
427 /* XXX: may need to add full url resolution */
428 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
430 if (proto[0] == '\0') {
431 /* relative control URL */
432 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
433 av_strlcat(rtsp_st->control_url, "/",
434 sizeof(rtsp_st->control_url));
435 av_strlcat(rtsp_st->control_url, p,
436 sizeof(rtsp_st->control_url));
438 av_strlcpy(rtsp_st->control_url, p,
439 sizeof(rtsp_st->control_url));
441 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
442 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
443 get_word(buf1, sizeof(buf1), &p);
444 payload_type = atoi(buf1);
445 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
446 if (rtsp_st->stream_index >= 0) {
447 st = s->streams[rtsp_st->stream_index];
448 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
450 } else if (av_strstart(p, "fmtp:", &p) ||
451 av_strstart(p, "framesize:", &p)) {
452 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
453 // let dynamic protocol handlers have a stab at the line.
454 get_word(buf1, sizeof(buf1), &p);
455 payload_type = atoi(buf1);
456 for (i = 0; i < rt->nb_rtsp_streams; i++) {
457 rtsp_st = rt->rtsp_streams[i];
458 if (rtsp_st->sdp_payload_type == payload_type &&
459 rtsp_st->dynamic_handler &&
460 rtsp_st->dynamic_handler->parse_sdp_a_line)
461 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
462 rtsp_st->dynamic_protocol_context, buf);
464 } else if (av_strstart(p, "range:", &p)) {
467 // this is so that seeking on a streamed file can work.
468 rtsp_parse_range_npt(p, &start, &end);
469 s->start_time = start;
470 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
471 s->duration = (end == AV_NOPTS_VALUE) ?
472 AV_NOPTS_VALUE : end - start;
473 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
475 rt->transport = RTSP_TRANSPORT_RDT;
476 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
478 st = s->streams[s->nb_streams - 1];
479 st->codec->sample_rate = atoi(p);
481 if (rt->server_type == RTSP_SERVER_WMS)
482 ff_wms_parse_sdp_a_line(s, p);
483 if (s->nb_streams > 0) {
484 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
486 if (rt->server_type == RTSP_SERVER_REAL)
487 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
489 if (rtsp_st->dynamic_handler &&
490 rtsp_st->dynamic_handler->parse_sdp_a_line)
491 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
492 rtsp_st->stream_index,
493 rtsp_st->dynamic_protocol_context, buf);
500 int ff_sdp_parse(AVFormatContext *s, const char *content)
502 RTSPState *rt = s->priv_data;
505 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
506 * contain long SDP lines containing complete ASF Headers (several
507 * kB) or arrays of MDPR (RM stream descriptor) headers plus
508 * "rulebooks" describing their properties. Therefore, the SDP line
511 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
512 * in rtpdec_xiph.c. */
514 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
518 p += strspn(p, SPACE_CHARS);
526 /* get the content */
528 while (*p != '\n' && *p != '\r' && *p != '\0') {
529 if ((q - buf) < sizeof(buf) - 1)
534 sdp_parse_line(s, s1, letter, buf);
536 while (*p != '\n' && *p != '\0')
541 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
542 if (!rt->p) return AVERROR(ENOMEM);
545 #endif /* CONFIG_RTPDEC */
547 void ff_rtsp_undo_setup(AVFormatContext *s)
549 RTSPState *rt = s->priv_data;
552 for (i = 0; i < rt->nb_rtsp_streams; i++) {
553 RTSPStream *rtsp_st = rt->rtsp_streams[i];
556 if (rtsp_st->transport_priv) {
558 AVFormatContext *rtpctx = rtsp_st->transport_priv;
559 av_write_trailer(rtpctx);
560 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
562 avio_close_dyn_buf(rtpctx->pb, &ptr);
565 avio_close(rtpctx->pb);
567 avformat_free_context(rtpctx);
568 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
569 ff_rdt_parse_close(rtsp_st->transport_priv);
570 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
571 ff_rtp_parse_close(rtsp_st->transport_priv);
573 rtsp_st->transport_priv = NULL;
574 if (rtsp_st->rtp_handle)
575 ffurl_close(rtsp_st->rtp_handle);
576 rtsp_st->rtp_handle = NULL;
580 /* close and free RTSP streams */
581 void ff_rtsp_close_streams(AVFormatContext *s)
583 RTSPState *rt = s->priv_data;
587 ff_rtsp_undo_setup(s);
588 for (i = 0; i < rt->nb_rtsp_streams; i++) {
589 rtsp_st = rt->rtsp_streams[i];
591 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
592 rtsp_st->dynamic_handler->free(
593 rtsp_st->dynamic_protocol_context);
597 av_free(rt->rtsp_streams);
599 avformat_close_input(&rt->asf_ctx);
601 if (rt->ts && CONFIG_RTPDEC)
602 ff_mpegts_parse_close(rt->ts);
604 av_free(rt->recvbuf);
607 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
609 RTSPState *rt = s->priv_data;
612 /* open the RTP context */
613 if (rtsp_st->stream_index >= 0)
614 st = s->streams[rtsp_st->stream_index];
616 s->ctx_flags |= AVFMTCTX_NOHEADER;
618 if (s->oformat && CONFIG_RTSP_MUXER) {
619 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
621 RTSP_TCP_MAX_PACKET_SIZE);
622 /* Ownership of rtp_handle is passed to the rtp mux context */
623 rtsp_st->rtp_handle = NULL;
626 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
627 return 0; // Don't need to open any parser here
628 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
629 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
630 rtsp_st->dynamic_protocol_context,
631 rtsp_st->dynamic_handler);
632 else if (CONFIG_RTPDEC)
633 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
634 rtsp_st->sdp_payload_type,
635 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
636 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
638 if (!rtsp_st->transport_priv) {
639 return AVERROR(ENOMEM);
640 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
641 if (rtsp_st->dynamic_handler) {
642 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
643 rtsp_st->dynamic_protocol_context,
644 rtsp_st->dynamic_handler);
651 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
652 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
659 q += strspn(q, SPACE_CHARS);
660 v = strtol(q, &p, 10);
664 v = strtol(p, &p, 10);
673 /* XXX: only one transport specification is parsed */
674 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
676 char transport_protocol[16];
678 char lower_transport[16];
680 RTSPTransportField *th;
683 reply->nb_transports = 0;
686 p += strspn(p, SPACE_CHARS);
690 th = &reply->transports[reply->nb_transports];
692 get_word_sep(transport_protocol, sizeof(transport_protocol),
694 if (!av_strcasecmp (transport_protocol, "rtp")) {
695 get_word_sep(profile, sizeof(profile), "/;,", &p);
696 lower_transport[0] = '\0';
697 /* rtp/avp/<protocol> */
699 get_word_sep(lower_transport, sizeof(lower_transport),
702 th->transport = RTSP_TRANSPORT_RTP;
703 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
704 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
705 /* x-pn-tng/<protocol> */
706 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
708 th->transport = RTSP_TRANSPORT_RDT;
709 } else if (!av_strcasecmp(transport_protocol, "raw")) {
710 get_word_sep(profile, sizeof(profile), "/;,", &p);
711 lower_transport[0] = '\0';
712 /* raw/raw/<protocol> */
714 get_word_sep(lower_transport, sizeof(lower_transport),
717 th->transport = RTSP_TRANSPORT_RAW;
719 if (!av_strcasecmp(lower_transport, "TCP"))
720 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
722 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
726 /* get each parameter */
727 while (*p != '\0' && *p != ',') {
728 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
729 if (!strcmp(parameter, "port")) {
732 rtsp_parse_range(&th->port_min, &th->port_max, &p);
734 } else if (!strcmp(parameter, "client_port")) {
737 rtsp_parse_range(&th->client_port_min,
738 &th->client_port_max, &p);
740 } else if (!strcmp(parameter, "server_port")) {
743 rtsp_parse_range(&th->server_port_min,
744 &th->server_port_max, &p);
746 } else if (!strcmp(parameter, "interleaved")) {
749 rtsp_parse_range(&th->interleaved_min,
750 &th->interleaved_max, &p);
752 } else if (!strcmp(parameter, "multicast")) {
753 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
754 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
755 } else if (!strcmp(parameter, "ttl")) {
758 th->ttl = strtol(p, (char **)&p, 10);
760 } else if (!strcmp(parameter, "destination")) {
763 get_word_sep(buf, sizeof(buf), ";,", &p);
764 get_sockaddr(buf, &th->destination);
766 } else if (!strcmp(parameter, "source")) {
769 get_word_sep(buf, sizeof(buf), ";,", &p);
770 av_strlcpy(th->source, buf, sizeof(th->source));
772 } else if (!strcmp(parameter, "mode")) {
775 get_word_sep(buf, sizeof(buf), ";, ", &p);
776 if (!strcmp(buf, "record") ||
777 !strcmp(buf, "receive"))
782 while (*p != ';' && *p != '\0' && *p != ',')
790 reply->nb_transports++;
794 static void handle_rtp_info(RTSPState *rt, const char *url,
795 uint32_t seq, uint32_t rtptime)
798 if (!rtptime || !url[0])
800 if (rt->transport != RTSP_TRANSPORT_RTP)
802 for (i = 0; i < rt->nb_rtsp_streams; i++) {
803 RTSPStream *rtsp_st = rt->rtsp_streams[i];
804 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
807 if (!strcmp(rtsp_st->control_url, url)) {
808 rtpctx->base_timestamp = rtptime;
814 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
817 char key[20], value[1024], url[1024] = "";
818 uint32_t seq = 0, rtptime = 0;
821 p += strspn(p, SPACE_CHARS);
824 get_word_sep(key, sizeof(key), "=", &p);
828 get_word_sep(value, sizeof(value), ";, ", &p);
830 if (!strcmp(key, "url"))
831 av_strlcpy(url, value, sizeof(url));
832 else if (!strcmp(key, "seq"))
833 seq = strtoul(value, NULL, 10);
834 else if (!strcmp(key, "rtptime"))
835 rtptime = strtoul(value, NULL, 10);
837 handle_rtp_info(rt, url, seq, rtptime);
846 handle_rtp_info(rt, url, seq, rtptime);
849 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
850 RTSPState *rt, const char *method)
854 /* NOTE: we do case independent match for broken servers */
856 if (av_stristart(p, "Session:", &p)) {
858 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
859 if (av_stristart(p, ";timeout=", &p) &&
860 (t = strtol(p, NULL, 10)) > 0) {
863 } else if (av_stristart(p, "Content-Length:", &p)) {
864 reply->content_length = strtol(p, NULL, 10);
865 } else if (av_stristart(p, "Transport:", &p)) {
866 rtsp_parse_transport(reply, p);
867 } else if (av_stristart(p, "CSeq:", &p)) {
868 reply->seq = strtol(p, NULL, 10);
869 } else if (av_stristart(p, "Range:", &p)) {
870 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
871 } else if (av_stristart(p, "RealChallenge1:", &p)) {
872 p += strspn(p, SPACE_CHARS);
873 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
874 } else if (av_stristart(p, "Server:", &p)) {
875 p += strspn(p, SPACE_CHARS);
876 av_strlcpy(reply->server, p, sizeof(reply->server));
877 } else if (av_stristart(p, "Notice:", &p) ||
878 av_stristart(p, "X-Notice:", &p)) {
879 reply->notice = strtol(p, NULL, 10);
880 } else if (av_stristart(p, "Location:", &p)) {
881 p += strspn(p, SPACE_CHARS);
882 av_strlcpy(reply->location, p , sizeof(reply->location));
883 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
884 p += strspn(p, SPACE_CHARS);
885 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
886 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
887 p += strspn(p, SPACE_CHARS);
888 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
889 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
890 p += strspn(p, SPACE_CHARS);
891 if (method && !strcmp(method, "DESCRIBE"))
892 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
893 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
894 p += strspn(p, SPACE_CHARS);
895 if (method && !strcmp(method, "PLAY"))
896 rtsp_parse_rtp_info(rt, p);
897 } else if (av_stristart(p, "Public:", &p) && rt) {
898 if (strstr(p, "GET_PARAMETER") &&
899 method && !strcmp(method, "OPTIONS"))
900 rt->get_parameter_supported = 1;
901 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
902 p += strspn(p, SPACE_CHARS);
903 rt->accept_dynamic_rate = atoi(p);
904 } else if (av_stristart(p, "Content-Type:", &p)) {
905 p += strspn(p, SPACE_CHARS);
906 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
910 /* skip a RTP/TCP interleaved packet */
911 void ff_rtsp_skip_packet(AVFormatContext *s)
913 RTSPState *rt = s->priv_data;
917 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
920 len = AV_RB16(buf + 1);
922 av_dlog(s, "skipping RTP packet len=%d\n", len);
927 if (len1 > sizeof(buf))
929 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
936 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
937 unsigned char **content_ptr,
938 int return_on_interleaved_data, const char *method)
940 RTSPState *rt = s->priv_data;
941 char buf[4096], buf1[1024], *q;
944 int ret, content_length, line_count = 0, request = 0;
945 unsigned char *content = NULL;
951 memset(reply, 0, sizeof(*reply));
953 /* parse reply (XXX: use buffers) */
954 rt->last_reply[0] = '\0';
958 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
959 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
965 /* XXX: only parse it if first char on line ? */
966 if (return_on_interleaved_data) {
969 ff_rtsp_skip_packet(s);
970 } else if (ch != '\r') {
971 if ((q - buf) < sizeof(buf) - 1)
977 av_dlog(s, "line='%s'\n", buf);
979 /* test if last line */
983 if (line_count == 0) {
985 get_word(buf1, sizeof(buf1), &p);
986 if (!strncmp(buf1, "RTSP/", 5)) {
987 get_word(buf1, sizeof(buf1), &p);
988 reply->status_code = atoi(buf1);
989 av_strlcpy(reply->reason, p, sizeof(reply->reason));
991 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
992 get_word(buf1, sizeof(buf1), &p); // object
996 ff_rtsp_parse_line(reply, p, rt, method);
997 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
998 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1003 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1004 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1006 content_length = reply->content_length;
1007 if (content_length > 0) {
1008 /* leave some room for a trailing '\0' (useful for simple parsing) */
1009 content = av_malloc(content_length + 1);
1010 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1011 content[content_length] = '\0';
1014 *content_ptr = content;
1020 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1021 const char* ptr = buf;
1023 if (!strcmp(reply->reason, "OPTIONS")) {
1024 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1026 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1027 if (reply->session_id[0])
1028 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1031 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1033 av_strlcat(buf, "\r\n", sizeof(buf));
1035 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1036 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1039 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1041 rt->last_cmd_time = av_gettime();
1042 /* Even if the request from the server had data, it is not the data
1043 * that the caller wants or expects. The memory could also be leaked
1044 * if the actual following reply has content data. */
1046 av_freep(content_ptr);
1047 /* If method is set, this is called from ff_rtsp_send_cmd,
1048 * where a reply to exactly this request is awaited. For
1049 * callers from within packet receiving, we just want to
1050 * return to the caller and go back to receiving packets. */
1056 if (rt->seq != reply->seq) {
1057 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1058 rt->seq, reply->seq);
1062 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1063 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1064 reply->notice == 2306 /* Continuous Feed Terminated */) {
1065 rt->state = RTSP_STATE_IDLE;
1066 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1067 return AVERROR(EIO); /* data or server error */
1068 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1069 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1070 return AVERROR(EPERM);
1076 * Send a command to the RTSP server without waiting for the reply.
1078 * @param s RTSP (de)muxer context
1079 * @param method the method for the request
1080 * @param url the target url for the request
1081 * @param headers extra header lines to include in the request
1082 * @param send_content if non-null, the data to send as request body content
1083 * @param send_content_length the length of the send_content data, or 0 if
1084 * send_content is null
1086 * @return zero if success, nonzero otherwise
1088 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1089 const char *method, const char *url,
1090 const char *headers,
1091 const unsigned char *send_content,
1092 int send_content_length)
1094 RTSPState *rt = s->priv_data;
1095 char buf[4096], *out_buf;
1096 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1098 /* Add in RTSP headers */
1101 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1103 av_strlcat(buf, headers, sizeof(buf));
1104 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1105 if (rt->session_id[0] != '\0' && (!headers ||
1106 !strstr(headers, "\nIf-Match:"))) {
1107 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1110 char *str = ff_http_auth_create_response(&rt->auth_state,
1111 rt->auth, url, method);
1113 av_strlcat(buf, str, sizeof(buf));
1116 if (send_content_length > 0 && send_content)
1117 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1118 av_strlcat(buf, "\r\n", sizeof(buf));
1120 /* base64 encode rtsp if tunneling */
1121 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1122 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1123 out_buf = base64buf;
1126 av_dlog(s, "Sending:\n%s--\n", buf);
1128 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1129 if (send_content_length > 0 && send_content) {
1130 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1131 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1132 "with content data not supported\n");
1133 return AVERROR_PATCHWELCOME;
1135 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1137 rt->last_cmd_time = av_gettime();
1142 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1143 const char *url, const char *headers)
1145 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1148 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1149 const char *headers, RTSPMessageHeader *reply,
1150 unsigned char **content_ptr)
1152 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1153 content_ptr, NULL, 0);
1156 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1157 const char *method, const char *url,
1159 RTSPMessageHeader *reply,
1160 unsigned char **content_ptr,
1161 const unsigned char *send_content,
1162 int send_content_length)
1164 RTSPState *rt = s->priv_data;
1165 HTTPAuthType cur_auth_type;
1166 int ret, attempts = 0;
1169 cur_auth_type = rt->auth_state.auth_type;
1170 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1172 send_content_length)))
1175 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1179 if (reply->status_code == 401 &&
1180 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1181 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1184 if (reply->status_code > 400){
1185 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1189 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1195 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1196 int lower_transport, const char *real_challenge)
1198 RTSPState *rt = s->priv_data;
1199 int rtx = 0, j, i, err, interleave = 0, port_off;
1200 RTSPStream *rtsp_st;
1201 RTSPMessageHeader reply1, *reply = &reply1;
1203 const char *trans_pref;
1205 if (rt->transport == RTSP_TRANSPORT_RDT)
1206 trans_pref = "x-pn-tng";
1207 else if (rt->transport == RTSP_TRANSPORT_RAW)
1208 trans_pref = "RAW/RAW";
1210 trans_pref = "RTP/AVP";
1212 /* default timeout: 1 minute */
1215 /* Choose a random starting offset within the first half of the
1216 * port range, to allow for a number of ports to try even if the offset
1217 * happens to be at the end of the random range. */
1218 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1219 /* even random offset */
1220 port_off -= port_off & 0x01;
1222 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1223 char transport[2048];
1226 * WMS serves all UDP data over a single connection, the RTX, which
1227 * isn't necessarily the first in the SDP but has to be the first
1228 * to be set up, else the second/third SETUP will fail with a 461.
1230 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1231 rt->server_type == RTSP_SERVER_WMS) {
1234 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1235 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1237 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1241 if (rtx == rt->nb_rtsp_streams)
1242 return -1; /* no RTX found */
1243 rtsp_st = rt->rtsp_streams[rtx];
1245 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1247 rtsp_st = rt->rtsp_streams[i];
1250 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1253 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1254 port = reply->transports[0].client_port_min;
1258 /* first try in specified port range */
1259 while (j <= rt->rtp_port_max) {
1260 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1261 "?localport=%d", j);
1262 /* we will use two ports per rtp stream (rtp and rtcp) */
1264 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1265 &s->interrupt_callback, NULL))
1268 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1273 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1275 snprintf(transport, sizeof(transport) - 1,
1276 "%s/UDP;", trans_pref);
1277 if (rt->server_type != RTSP_SERVER_REAL)
1278 av_strlcat(transport, "unicast;", sizeof(transport));
1279 av_strlcatf(transport, sizeof(transport),
1280 "client_port=%d", port);
1281 if (rt->transport == RTSP_TRANSPORT_RTP &&
1282 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1283 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1287 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1288 /* For WMS streams, the application streams are only used for
1289 * UDP. When trying to set it up for TCP streams, the server
1290 * will return an error. Therefore, we skip those streams. */
1291 if (rt->server_type == RTSP_SERVER_WMS &&
1292 (rtsp_st->stream_index < 0 ||
1293 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1296 snprintf(transport, sizeof(transport) - 1,
1297 "%s/TCP;", trans_pref);
1298 if (rt->transport != RTSP_TRANSPORT_RDT)
1299 av_strlcat(transport, "unicast;", sizeof(transport));
1300 av_strlcatf(transport, sizeof(transport),
1301 "interleaved=%d-%d",
1302 interleave, interleave + 1);
1306 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1307 snprintf(transport, sizeof(transport) - 1,
1308 "%s/UDP;multicast", trans_pref);
1311 av_strlcat(transport, ";mode=record", sizeof(transport));
1312 } else if (rt->server_type == RTSP_SERVER_REAL ||
1313 rt->server_type == RTSP_SERVER_WMS)
1314 av_strlcat(transport, ";mode=play", sizeof(transport));
1315 snprintf(cmd, sizeof(cmd),
1316 "Transport: %s\r\n",
1318 if (rt->accept_dynamic_rate)
1319 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1320 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1321 char real_res[41], real_csum[9];
1322 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1324 av_strlcatf(cmd, sizeof(cmd),
1326 "RealChallenge2: %s, sd=%s\r\n",
1327 rt->session_id, real_res, real_csum);
1329 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1330 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1333 } else if (reply->status_code != RTSP_STATUS_OK ||
1334 reply->nb_transports != 1) {
1335 err = AVERROR_INVALIDDATA;
1339 /* XXX: same protocol for all streams is required */
1341 if (reply->transports[0].lower_transport != rt->lower_transport ||
1342 reply->transports[0].transport != rt->transport) {
1343 err = AVERROR_INVALIDDATA;
1347 rt->lower_transport = reply->transports[0].lower_transport;
1348 rt->transport = reply->transports[0].transport;
1351 /* Fail if the server responded with another lower transport mode
1352 * than what we requested. */
1353 if (reply->transports[0].lower_transport != lower_transport) {
1354 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1355 err = AVERROR_INVALIDDATA;
1359 switch(reply->transports[0].lower_transport) {
1360 case RTSP_LOWER_TRANSPORT_TCP:
1361 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1362 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1365 case RTSP_LOWER_TRANSPORT_UDP: {
1366 char url[1024], options[30] = "";
1368 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1369 av_strlcpy(options, "?connect=1", sizeof(options));
1370 /* Use source address if specified */
1371 if (reply->transports[0].source[0]) {
1372 ff_url_join(url, sizeof(url), "rtp", NULL,
1373 reply->transports[0].source,
1374 reply->transports[0].server_port_min, "%s", options);
1376 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1377 reply->transports[0].server_port_min, "%s", options);
1379 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1380 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1381 err = AVERROR_INVALIDDATA;
1384 /* Try to initialize the connection state in a
1385 * potential NAT router by sending dummy packets.
1386 * RTP/RTCP dummy packets are used for RDT, too.
1388 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1390 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1393 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1394 char url[1024], namebuf[50], optbuf[20] = "";
1395 struct sockaddr_storage addr;
1398 if (reply->transports[0].destination.ss_family) {
1399 addr = reply->transports[0].destination;
1400 port = reply->transports[0].port_min;
1401 ttl = reply->transports[0].ttl;
1403 addr = rtsp_st->sdp_ip;
1404 port = rtsp_st->sdp_port;
1405 ttl = rtsp_st->sdp_ttl;
1408 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1409 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1410 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1411 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1412 port, "%s", optbuf);
1413 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1414 &s->interrupt_callback, NULL) < 0) {
1415 err = AVERROR_INVALIDDATA;
1422 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1426 if (rt->nb_rtsp_streams && reply->timeout > 0)
1427 rt->timeout = reply->timeout;
1429 if (rt->server_type == RTSP_SERVER_REAL)
1430 rt->need_subscription = 1;
1435 ff_rtsp_undo_setup(s);
1439 void ff_rtsp_close_connections(AVFormatContext *s)
1441 RTSPState *rt = s->priv_data;
1442 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1443 ffurl_close(rt->rtsp_hd);
1444 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1447 int ff_rtsp_connect(AVFormatContext *s)
1449 RTSPState *rt = s->priv_data;
1450 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1451 int port, err, tcp_fd;
1452 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1453 int lower_transport_mask = 0;
1454 char real_challenge[64] = "";
1455 struct sockaddr_storage peer;
1456 socklen_t peer_len = sizeof(peer);
1458 if (rt->rtp_port_max < rt->rtp_port_min) {
1459 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1460 "than min port %d\n", rt->rtp_port_max,
1462 return AVERROR(EINVAL);
1465 if (!ff_network_init())
1466 return AVERROR(EIO);
1468 if (s->max_delay < 0) /* Not set by the caller */
1469 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1471 rt->control_transport = RTSP_MODE_PLAIN;
1472 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1473 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1474 rt->control_transport = RTSP_MODE_TUNNEL;
1476 /* Only pass through valid flags from here */
1477 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1480 lower_transport_mask = rt->lower_transport_mask;
1481 /* extract hostname and port */
1482 av_url_split(NULL, 0, auth, sizeof(auth),
1483 host, sizeof(host), &port, path, sizeof(path), s->filename);
1485 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1488 port = RTSP_DEFAULT_PORT;
1490 if (!lower_transport_mask)
1491 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1494 /* Only UDP or TCP - UDP multicast isn't supported. */
1495 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1496 (1 << RTSP_LOWER_TRANSPORT_TCP);
1497 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1498 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1499 "only UDP and TCP are supported for output.\n");
1500 err = AVERROR(EINVAL);
1505 /* Construct the URI used in request; this is similar to s->filename,
1506 * but with authentication credentials removed and RTSP specific options
1508 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1509 host, port, "%s", path);
1511 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1512 /* set up initial handshake for tunneling */
1513 char httpname[1024];
1514 char sessioncookie[17];
1517 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1518 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1519 av_get_random_seed(), av_get_random_seed());
1522 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1523 &s->interrupt_callback) < 0) {
1528 /* generate GET headers */
1529 snprintf(headers, sizeof(headers),
1530 "x-sessioncookie: %s\r\n"
1531 "Accept: application/x-rtsp-tunnelled\r\n"
1532 "Pragma: no-cache\r\n"
1533 "Cache-Control: no-cache\r\n",
1535 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1537 /* complete the connection */
1538 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1544 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1545 &s->interrupt_callback) < 0 ) {
1550 /* generate POST headers */
1551 snprintf(headers, sizeof(headers),
1552 "x-sessioncookie: %s\r\n"
1553 "Content-Type: application/x-rtsp-tunnelled\r\n"
1554 "Pragma: no-cache\r\n"
1555 "Cache-Control: no-cache\r\n"
1556 "Content-Length: 32767\r\n"
1557 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1559 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1560 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1562 /* Initialize the authentication state for the POST session. The HTTP
1563 * protocol implementation doesn't properly handle multi-pass
1564 * authentication for POST requests, since it would require one of
1566 * - implementing Expect: 100-continue, which many HTTP servers
1567 * don't support anyway, even less the RTSP servers that do HTTP
1569 * - sending the whole POST data until getting a 401 reply specifying
1570 * what authentication method to use, then resending all that data
1571 * - waiting for potential 401 replies directly after sending the
1572 * POST header (waiting for some unspecified time)
1573 * Therefore, we copy the full auth state, which works for both basic
1574 * and digest. (For digest, we would have to synchronize the nonce
1575 * count variable between the two sessions, if we'd do more requests
1576 * with the original session, though.)
1578 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1580 /* complete the connection */
1581 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1586 /* open the tcp connection */
1587 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1588 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1589 &s->interrupt_callback, NULL) < 0) {
1593 rt->rtsp_hd_out = rt->rtsp_hd;
1597 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1598 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1599 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1600 NULL, 0, NI_NUMERICHOST);
1603 /* request options supported by the server; this also detects server
1605 for (rt->server_type = RTSP_SERVER_RTP;;) {
1607 if (rt->server_type == RTSP_SERVER_REAL)
1610 * The following entries are required for proper
1611 * streaming from a Realmedia server. They are
1612 * interdependent in some way although we currently
1613 * don't quite understand how. Values were copied
1614 * from mplayer SVN r23589.
1615 * ClientChallenge is a 16-byte ID in hex
1616 * CompanyID is a 16-byte ID in base64
1618 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1619 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1620 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1621 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1623 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1624 if (reply->status_code != RTSP_STATUS_OK) {
1625 err = AVERROR_INVALIDDATA;
1629 /* detect server type if not standard-compliant RTP */
1630 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1631 rt->server_type = RTSP_SERVER_REAL;
1633 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1634 rt->server_type = RTSP_SERVER_WMS;
1635 } else if (rt->server_type == RTSP_SERVER_REAL)
1636 strcpy(real_challenge, reply->real_challenge);
1640 if (s->iformat && CONFIG_RTSP_DEMUXER)
1641 err = ff_rtsp_setup_input_streams(s, reply);
1642 else if (CONFIG_RTSP_MUXER)
1643 err = ff_rtsp_setup_output_streams(s, host);
1648 int lower_transport = ff_log2_tab[lower_transport_mask &
1649 ~(lower_transport_mask - 1)];
1651 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1652 rt->server_type == RTSP_SERVER_REAL ?
1653 real_challenge : NULL);
1656 lower_transport_mask &= ~(1 << lower_transport);
1657 if (lower_transport_mask == 0 && err == 1) {
1658 err = AVERROR(EPROTONOSUPPORT);
1663 rt->lower_transport_mask = lower_transport_mask;
1664 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1665 rt->state = RTSP_STATE_IDLE;
1666 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1669 ff_rtsp_close_streams(s);
1670 ff_rtsp_close_connections(s);
1671 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1672 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1673 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1681 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1684 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1685 uint8_t *buf, int buf_size, int64_t wait_end)
1687 RTSPState *rt = s->priv_data;
1688 RTSPStream *rtsp_st;
1689 int n, i, ret, tcp_fd, timeout_cnt = 0;
1691 struct pollfd *p = rt->p;
1692 int *fds = NULL, fdsnum, fdsidx;
1695 if (ff_check_interrupt(&s->interrupt_callback))
1696 return AVERROR_EXIT;
1697 if (wait_end && wait_end - av_gettime() < 0)
1698 return AVERROR(EAGAIN);
1701 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1702 p[max_p].fd = tcp_fd;
1703 p[max_p++].events = POLLIN;
1707 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1708 rtsp_st = rt->rtsp_streams[i];
1709 if (rtsp_st->rtp_handle) {
1710 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1712 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1716 av_log(s, AV_LOG_ERROR,
1717 "Number of fds %d not supported\n", fdsnum);
1718 return AVERROR_INVALIDDATA;
1720 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1721 p[max_p].fd = fds[fdsidx];
1722 p[max_p++].events = POLLIN;
1727 n = poll(p, max_p, POLL_TIMEOUT_MS);
1729 int j = 1 - (tcp_fd == -1);
1731 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1732 rtsp_st = rt->rtsp_streams[i];
1733 if (rtsp_st->rtp_handle) {
1734 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1735 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1737 *prtsp_st = rtsp_st;
1744 #if CONFIG_RTSP_DEMUXER
1745 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1746 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1747 if (rt->state == RTSP_STATE_STREAMING) {
1748 if (!ff_rtsp_parse_streaming_commands(s))
1751 av_log(s, AV_LOG_WARNING,
1752 "Unable to answer to TEARDOWN\n");
1756 RTSPMessageHeader reply;
1757 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1760 /* XXX: parse message */
1761 if (rt->state != RTSP_STATE_STREAMING)
1766 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1767 return AVERROR(ETIMEDOUT);
1768 } else if (n < 0 && errno != EINTR)
1769 return AVERROR(errno);
1773 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1775 RTSPState *rt = s->priv_data;
1777 RTSPStream *rtsp_st, *first_queue_st = NULL;
1778 int64_t wait_end = 0;
1780 if (rt->nb_byes == rt->nb_rtsp_streams)
1783 /* get next frames from the same RTP packet */
1784 if (rt->cur_transport_priv) {
1785 if (rt->transport == RTSP_TRANSPORT_RDT) {
1786 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1787 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1788 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1789 } else if (rt->ts && CONFIG_RTPDEC) {
1790 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1792 rt->recvbuf_pos += ret;
1793 ret = rt->recvbuf_pos < rt->recvbuf_len;
1798 rt->cur_transport_priv = NULL;
1800 } else if (ret == 1) {
1803 rt->cur_transport_priv = NULL;
1806 if (rt->transport == RTSP_TRANSPORT_RTP) {
1808 int64_t first_queue_time = 0;
1809 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1810 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1814 queue_time = ff_rtp_queued_packet_time(rtpctx);
1815 if (queue_time && (queue_time - first_queue_time < 0 ||
1816 !first_queue_time)) {
1817 first_queue_time = queue_time;
1818 first_queue_st = rt->rtsp_streams[i];
1821 if (first_queue_time)
1822 wait_end = first_queue_time + s->max_delay;
1825 /* read next RTP packet */
1828 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1830 return AVERROR(ENOMEM);
1833 switch(rt->lower_transport) {
1835 #if CONFIG_RTSP_DEMUXER
1836 case RTSP_LOWER_TRANSPORT_TCP:
1837 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1840 case RTSP_LOWER_TRANSPORT_UDP:
1841 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1842 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1843 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1844 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1847 if (len == AVERROR(EAGAIN) && first_queue_st &&
1848 rt->transport == RTSP_TRANSPORT_RTP) {
1849 rtsp_st = first_queue_st;
1850 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1857 if (rt->transport == RTSP_TRANSPORT_RDT) {
1858 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1859 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1860 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1862 /* Either bad packet, or a RTCP packet. Check if the
1863 * first_rtcp_ntp_time field was initialized. */
1864 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1865 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1866 /* first_rtcp_ntp_time has been initialized for this stream,
1867 * copy the same value to all other uninitialized streams,
1868 * in order to map their timestamp origin to the same ntp time
1871 AVStream *st = NULL;
1872 if (rtsp_st->stream_index >= 0)
1873 st = s->streams[rtsp_st->stream_index];
1874 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1875 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1876 AVStream *st2 = NULL;
1877 if (rt->rtsp_streams[i]->stream_index >= 0)
1878 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1879 if (rtpctx2 && st && st2 &&
1880 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1881 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1882 rtpctx2->rtcp_ts_offset = av_rescale_q(
1883 rtpctx->rtcp_ts_offset, st->time_base,
1888 if (ret == -RTCP_BYE) {
1891 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1892 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1894 if (rt->nb_byes == rt->nb_rtsp_streams)
1898 } else if (rt->ts && CONFIG_RTPDEC) {
1899 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1902 rt->recvbuf_len = len;
1903 rt->recvbuf_pos = ret;
1904 rt->cur_transport_priv = rt->ts;
1911 return AVERROR_INVALIDDATA;
1917 /* more packets may follow, so we save the RTP context */
1918 rt->cur_transport_priv = rtsp_st->transport_priv;
1922 #endif /* CONFIG_RTPDEC */
1924 #if CONFIG_SDP_DEMUXER
1925 static int sdp_probe(AVProbeData *p1)
1927 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1929 /* we look for a line beginning "c=IN IP" */
1930 while (p < p_end && *p != '\0') {
1931 if (p + sizeof("c=IN IP") - 1 < p_end &&
1932 av_strstart(p, "c=IN IP", NULL))
1933 return AVPROBE_SCORE_MAX / 2;
1935 while (p < p_end - 1 && *p != '\n') p++;
1944 static int sdp_read_header(AVFormatContext *s)
1946 RTSPState *rt = s->priv_data;
1947 RTSPStream *rtsp_st;
1952 if (!ff_network_init())
1953 return AVERROR(EIO);
1955 if (s->max_delay < 0) /* Not set by the caller */
1956 s->max_delay = DEFAULT_REORDERING_DELAY;
1958 /* read the whole sdp file */
1959 /* XXX: better loading */
1960 content = av_malloc(SDP_MAX_SIZE);
1961 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1964 return AVERROR_INVALIDDATA;
1966 content[size] ='\0';
1968 err = ff_sdp_parse(s, content);
1972 /* open each RTP stream */
1973 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1975 rtsp_st = rt->rtsp_streams[i];
1977 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1978 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1979 ff_url_join(url, sizeof(url), "rtp", NULL,
1980 namebuf, rtsp_st->sdp_port,
1981 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1983 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1984 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1985 &s->interrupt_callback, NULL) < 0) {
1986 err = AVERROR_INVALIDDATA;
1989 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1994 ff_rtsp_close_streams(s);
1999 static int sdp_read_close(AVFormatContext *s)
2001 ff_rtsp_close_streams(s);
2006 static const AVClass sdp_demuxer_class = {
2007 .class_name = "SDP demuxer",
2008 .item_name = av_default_item_name,
2009 .option = sdp_options,
2010 .version = LIBAVUTIL_VERSION_INT,
2013 AVInputFormat ff_sdp_demuxer = {
2015 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2016 .priv_data_size = sizeof(RTSPState),
2017 .read_probe = sdp_probe,
2018 .read_header = sdp_read_header,
2019 .read_packet = ff_rtsp_fetch_packet,
2020 .read_close = sdp_read_close,
2021 .priv_class = &sdp_demuxer_class,
2023 #endif /* CONFIG_SDP_DEMUXER */
2025 #if CONFIG_RTP_DEMUXER
2026 static int rtp_probe(AVProbeData *p)
2028 if (av_strstart(p->filename, "rtp:", NULL))
2029 return AVPROBE_SCORE_MAX;
2033 static int rtp_read_header(AVFormatContext *s)
2035 uint8_t recvbuf[1500];
2036 char host[500], sdp[500];
2038 URLContext* in = NULL;
2040 AVCodecContext codec = { 0 };
2041 struct sockaddr_storage addr;
2043 socklen_t addrlen = sizeof(addr);
2044 RTSPState *rt = s->priv_data;
2046 if (!ff_network_init())
2047 return AVERROR(EIO);
2049 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2050 &s->interrupt_callback, NULL);
2055 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2056 if (ret == AVERROR(EAGAIN))
2061 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2065 if ((recvbuf[0] & 0xc0) != 0x80) {
2066 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2071 if (RTP_PT_IS_RTCP(recvbuf[1]))
2074 payload_type = recvbuf[1] & 0x7f;
2077 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2081 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2082 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2083 "without an SDP file describing it\n",
2087 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2088 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2089 "properly you need an SDP file "
2093 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2094 NULL, 0, s->filename);
2096 snprintf(sdp, sizeof(sdp),
2097 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2098 addr.ss_family == AF_INET ? 4 : 6, host,
2099 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2100 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2101 port, payload_type);
2102 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2104 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2107 /* sdp_read_header initializes this again */
2110 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2112 ret = sdp_read_header(s);
2123 static const AVClass rtp_demuxer_class = {
2124 .class_name = "RTP demuxer",
2125 .item_name = av_default_item_name,
2126 .option = rtp_options,
2127 .version = LIBAVUTIL_VERSION_INT,
2130 AVInputFormat ff_rtp_demuxer = {
2132 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2133 .priv_data_size = sizeof(RTSPState),
2134 .read_probe = rtp_probe,
2135 .read_header = rtp_read_header,
2136 .read_packet = ff_rtsp_fetch_packet,
2137 .read_close = sdp_read_close,
2138 .flags = AVFMT_NOFILE,
2139 .priv_class = &rtp_demuxer_class,
2141 #endif /* CONFIG_RTP_DEMUXER */