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/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 /* Choose a random starting offset within the first half of the
1217 * port range, to allow for a number of ports to try even if the offset
1218 * happens to be at the end of the random range. */
1219 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1220 /* even random offset */
1221 port_off -= port_off & 0x01;
1223 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1224 char transport[2048];
1227 * WMS serves all UDP data over a single connection, the RTX, which
1228 * isn't necessarily the first in the SDP but has to be the first
1229 * to be set up, else the second/third SETUP will fail with a 461.
1231 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1232 rt->server_type == RTSP_SERVER_WMS) {
1235 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1236 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1238 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1242 if (rtx == rt->nb_rtsp_streams)
1243 return -1; /* no RTX found */
1244 rtsp_st = rt->rtsp_streams[rtx];
1246 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1248 rtsp_st = rt->rtsp_streams[i];
1251 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1254 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1255 port = reply->transports[0].client_port_min;
1259 /* first try in specified port range */
1260 while (j <= rt->rtp_port_max) {
1261 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1262 "?localport=%d", j);
1263 /* we will use two ports per rtp stream (rtp and rtcp) */
1265 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1266 &s->interrupt_callback, NULL))
1269 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1274 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1276 snprintf(transport, sizeof(transport) - 1,
1277 "%s/UDP;", trans_pref);
1278 if (rt->server_type != RTSP_SERVER_REAL)
1279 av_strlcat(transport, "unicast;", sizeof(transport));
1280 av_strlcatf(transport, sizeof(transport),
1281 "client_port=%d", port);
1282 if (rt->transport == RTSP_TRANSPORT_RTP &&
1283 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1284 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1288 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1289 /* For WMS streams, the application streams are only used for
1290 * UDP. When trying to set it up for TCP streams, the server
1291 * will return an error. Therefore, we skip those streams. */
1292 if (rt->server_type == RTSP_SERVER_WMS &&
1293 (rtsp_st->stream_index < 0 ||
1294 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1297 snprintf(transport, sizeof(transport) - 1,
1298 "%s/TCP;", trans_pref);
1299 if (rt->transport != RTSP_TRANSPORT_RDT)
1300 av_strlcat(transport, "unicast;", sizeof(transport));
1301 av_strlcatf(transport, sizeof(transport),
1302 "interleaved=%d-%d",
1303 interleave, interleave + 1);
1307 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1308 snprintf(transport, sizeof(transport) - 1,
1309 "%s/UDP;multicast", trans_pref);
1312 av_strlcat(transport, ";mode=record", sizeof(transport));
1313 } else if (rt->server_type == RTSP_SERVER_REAL ||
1314 rt->server_type == RTSP_SERVER_WMS)
1315 av_strlcat(transport, ";mode=play", sizeof(transport));
1316 snprintf(cmd, sizeof(cmd),
1317 "Transport: %s\r\n",
1319 if (rt->accept_dynamic_rate)
1320 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1321 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1322 char real_res[41], real_csum[9];
1323 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1325 av_strlcatf(cmd, sizeof(cmd),
1327 "RealChallenge2: %s, sd=%s\r\n",
1328 rt->session_id, real_res, real_csum);
1330 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1331 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1334 } else if (reply->status_code != RTSP_STATUS_OK ||
1335 reply->nb_transports != 1) {
1336 err = AVERROR_INVALIDDATA;
1340 /* XXX: same protocol for all streams is required */
1342 if (reply->transports[0].lower_transport != rt->lower_transport ||
1343 reply->transports[0].transport != rt->transport) {
1344 err = AVERROR_INVALIDDATA;
1348 rt->lower_transport = reply->transports[0].lower_transport;
1349 rt->transport = reply->transports[0].transport;
1352 /* Fail if the server responded with another lower transport mode
1353 * than what we requested. */
1354 if (reply->transports[0].lower_transport != lower_transport) {
1355 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1356 err = AVERROR_INVALIDDATA;
1360 switch(reply->transports[0].lower_transport) {
1361 case RTSP_LOWER_TRANSPORT_TCP:
1362 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1363 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1366 case RTSP_LOWER_TRANSPORT_UDP: {
1367 char url[1024], options[30] = "";
1369 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1370 av_strlcpy(options, "?connect=1", sizeof(options));
1371 /* Use source address if specified */
1372 if (reply->transports[0].source[0]) {
1373 ff_url_join(url, sizeof(url), "rtp", NULL,
1374 reply->transports[0].source,
1375 reply->transports[0].server_port_min, "%s", options);
1377 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1378 reply->transports[0].server_port_min, "%s", options);
1380 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1381 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1382 err = AVERROR_INVALIDDATA;
1385 /* Try to initialize the connection state in a
1386 * potential NAT router by sending dummy packets.
1387 * RTP/RTCP dummy packets are used for RDT, too.
1389 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1391 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1394 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1395 char url[1024], namebuf[50], optbuf[20] = "";
1396 struct sockaddr_storage addr;
1399 if (reply->transports[0].destination.ss_family) {
1400 addr = reply->transports[0].destination;
1401 port = reply->transports[0].port_min;
1402 ttl = reply->transports[0].ttl;
1404 addr = rtsp_st->sdp_ip;
1405 port = rtsp_st->sdp_port;
1406 ttl = rtsp_st->sdp_ttl;
1409 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1410 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1411 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1412 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1413 port, "%s", optbuf);
1414 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1415 &s->interrupt_callback, NULL) < 0) {
1416 err = AVERROR_INVALIDDATA;
1423 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1427 if (rt->nb_rtsp_streams && reply->timeout > 0)
1428 rt->timeout = reply->timeout;
1430 if (rt->server_type == RTSP_SERVER_REAL)
1431 rt->need_subscription = 1;
1436 ff_rtsp_undo_setup(s);
1440 void ff_rtsp_close_connections(AVFormatContext *s)
1442 RTSPState *rt = s->priv_data;
1443 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1444 ffurl_close(rt->rtsp_hd);
1445 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1448 int ff_rtsp_connect(AVFormatContext *s)
1450 RTSPState *rt = s->priv_data;
1451 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1452 int port, err, tcp_fd;
1453 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1454 int lower_transport_mask = 0;
1455 char real_challenge[64] = "";
1456 struct sockaddr_storage peer;
1457 socklen_t peer_len = sizeof(peer);
1459 if (rt->rtp_port_max < rt->rtp_port_min) {
1460 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1461 "than min port %d\n", rt->rtp_port_max,
1463 return AVERROR(EINVAL);
1466 if (!ff_network_init())
1467 return AVERROR(EIO);
1469 if (s->max_delay < 0) /* Not set by the caller */
1470 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1472 rt->control_transport = RTSP_MODE_PLAIN;
1473 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1474 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1475 rt->control_transport = RTSP_MODE_TUNNEL;
1477 /* Only pass through valid flags from here */
1478 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1481 lower_transport_mask = rt->lower_transport_mask;
1482 /* extract hostname and port */
1483 av_url_split(NULL, 0, auth, sizeof(auth),
1484 host, sizeof(host), &port, path, sizeof(path), s->filename);
1486 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1489 port = RTSP_DEFAULT_PORT;
1491 if (!lower_transport_mask)
1492 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1495 /* Only UDP or TCP - UDP multicast isn't supported. */
1496 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1497 (1 << RTSP_LOWER_TRANSPORT_TCP);
1498 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1499 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1500 "only UDP and TCP are supported for output.\n");
1501 err = AVERROR(EINVAL);
1506 /* Construct the URI used in request; this is similar to s->filename,
1507 * but with authentication credentials removed and RTSP specific options
1509 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1510 host, port, "%s", path);
1512 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1513 /* set up initial handshake for tunneling */
1514 char httpname[1024];
1515 char sessioncookie[17];
1518 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1519 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1520 av_get_random_seed(), av_get_random_seed());
1523 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1524 &s->interrupt_callback) < 0) {
1529 /* generate GET headers */
1530 snprintf(headers, sizeof(headers),
1531 "x-sessioncookie: %s\r\n"
1532 "Accept: application/x-rtsp-tunnelled\r\n"
1533 "Pragma: no-cache\r\n"
1534 "Cache-Control: no-cache\r\n",
1536 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1538 /* complete the connection */
1539 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1545 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1546 &s->interrupt_callback) < 0 ) {
1551 /* generate POST headers */
1552 snprintf(headers, sizeof(headers),
1553 "x-sessioncookie: %s\r\n"
1554 "Content-Type: application/x-rtsp-tunnelled\r\n"
1555 "Pragma: no-cache\r\n"
1556 "Cache-Control: no-cache\r\n"
1557 "Content-Length: 32767\r\n"
1558 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1560 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1561 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1563 /* Initialize the authentication state for the POST session. The HTTP
1564 * protocol implementation doesn't properly handle multi-pass
1565 * authentication for POST requests, since it would require one of
1567 * - implementing Expect: 100-continue, which many HTTP servers
1568 * don't support anyway, even less the RTSP servers that do HTTP
1570 * - sending the whole POST data until getting a 401 reply specifying
1571 * what authentication method to use, then resending all that data
1572 * - waiting for potential 401 replies directly after sending the
1573 * POST header (waiting for some unspecified time)
1574 * Therefore, we copy the full auth state, which works for both basic
1575 * and digest. (For digest, we would have to synchronize the nonce
1576 * count variable between the two sessions, if we'd do more requests
1577 * with the original session, though.)
1579 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1581 /* complete the connection */
1582 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1587 /* open the tcp connection */
1588 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1589 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1590 &s->interrupt_callback, NULL) < 0) {
1594 rt->rtsp_hd_out = rt->rtsp_hd;
1598 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1599 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1600 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1601 NULL, 0, NI_NUMERICHOST);
1604 /* request options supported by the server; this also detects server
1606 for (rt->server_type = RTSP_SERVER_RTP;;) {
1608 if (rt->server_type == RTSP_SERVER_REAL)
1611 * The following entries are required for proper
1612 * streaming from a Realmedia server. They are
1613 * interdependent in some way although we currently
1614 * don't quite understand how. Values were copied
1615 * from mplayer SVN r23589.
1616 * ClientChallenge is a 16-byte ID in hex
1617 * CompanyID is a 16-byte ID in base64
1619 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1620 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1621 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1622 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1624 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1625 if (reply->status_code != RTSP_STATUS_OK) {
1626 err = AVERROR_INVALIDDATA;
1630 /* detect server type if not standard-compliant RTP */
1631 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1632 rt->server_type = RTSP_SERVER_REAL;
1634 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1635 rt->server_type = RTSP_SERVER_WMS;
1636 } else if (rt->server_type == RTSP_SERVER_REAL)
1637 strcpy(real_challenge, reply->real_challenge);
1641 if (s->iformat && CONFIG_RTSP_DEMUXER)
1642 err = ff_rtsp_setup_input_streams(s, reply);
1643 else if (CONFIG_RTSP_MUXER)
1644 err = ff_rtsp_setup_output_streams(s, host);
1649 int lower_transport = ff_log2_tab[lower_transport_mask &
1650 ~(lower_transport_mask - 1)];
1652 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1653 rt->server_type == RTSP_SERVER_REAL ?
1654 real_challenge : NULL);
1657 lower_transport_mask &= ~(1 << lower_transport);
1658 if (lower_transport_mask == 0 && err == 1) {
1659 err = AVERROR(EPROTONOSUPPORT);
1664 rt->lower_transport_mask = lower_transport_mask;
1665 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1666 rt->state = RTSP_STATE_IDLE;
1667 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1670 ff_rtsp_close_streams(s);
1671 ff_rtsp_close_connections(s);
1672 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1673 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1674 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1682 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1685 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1686 uint8_t *buf, int buf_size, int64_t wait_end)
1688 RTSPState *rt = s->priv_data;
1689 RTSPStream *rtsp_st;
1690 int n, i, ret, tcp_fd, timeout_cnt = 0;
1692 struct pollfd *p = rt->p;
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 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1711 p[max_p++].events = POLLIN;
1712 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1713 p[max_p++].events = POLLIN;
1716 n = poll(p, max_p, POLL_TIMEOUT_MS);
1718 int j = 1 - (tcp_fd == -1);
1720 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1721 rtsp_st = rt->rtsp_streams[i];
1722 if (rtsp_st->rtp_handle) {
1723 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1724 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1726 *prtsp_st = rtsp_st;
1733 #if CONFIG_RTSP_DEMUXER
1734 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1735 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1736 if (rt->state == RTSP_STATE_STREAMING) {
1737 if (!ff_rtsp_parse_streaming_commands(s))
1740 av_log(s, AV_LOG_WARNING,
1741 "Unable to answer to TEARDOWN\n");
1745 RTSPMessageHeader reply;
1746 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1749 /* XXX: parse message */
1750 if (rt->state != RTSP_STATE_STREAMING)
1755 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1756 return AVERROR(ETIMEDOUT);
1757 } else if (n < 0 && errno != EINTR)
1758 return AVERROR(errno);
1762 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1764 RTSPState *rt = s->priv_data;
1766 RTSPStream *rtsp_st, *first_queue_st = NULL;
1767 int64_t wait_end = 0;
1769 if (rt->nb_byes == rt->nb_rtsp_streams)
1772 /* get next frames from the same RTP packet */
1773 if (rt->cur_transport_priv) {
1774 if (rt->transport == RTSP_TRANSPORT_RDT) {
1775 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1776 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1777 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1778 } else if (rt->ts && CONFIG_RTPDEC) {
1779 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1781 rt->recvbuf_pos += ret;
1782 ret = rt->recvbuf_pos < rt->recvbuf_len;
1786 rt->cur_transport_priv = NULL;
1788 } else if (ret == 1) {
1791 rt->cur_transport_priv = NULL;
1794 if (rt->transport == RTSP_TRANSPORT_RTP) {
1796 int64_t first_queue_time = 0;
1797 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1798 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1802 queue_time = ff_rtp_queued_packet_time(rtpctx);
1803 if (queue_time && (queue_time - first_queue_time < 0 ||
1804 !first_queue_time)) {
1805 first_queue_time = queue_time;
1806 first_queue_st = rt->rtsp_streams[i];
1809 if (first_queue_time)
1810 wait_end = first_queue_time + s->max_delay;
1813 /* read next RTP packet */
1816 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1818 return AVERROR(ENOMEM);
1821 switch(rt->lower_transport) {
1823 #if CONFIG_RTSP_DEMUXER
1824 case RTSP_LOWER_TRANSPORT_TCP:
1825 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1828 case RTSP_LOWER_TRANSPORT_UDP:
1829 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1830 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1831 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1832 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1835 if (len == AVERROR(EAGAIN) && first_queue_st &&
1836 rt->transport == RTSP_TRANSPORT_RTP) {
1837 rtsp_st = first_queue_st;
1838 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1845 if (rt->transport == RTSP_TRANSPORT_RDT) {
1846 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1847 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1848 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1850 /* Either bad packet, or a RTCP packet. Check if the
1851 * first_rtcp_ntp_time field was initialized. */
1852 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1853 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1854 /* first_rtcp_ntp_time has been initialized for this stream,
1855 * copy the same value to all other uninitialized streams,
1856 * in order to map their timestamp origin to the same ntp time
1859 AVStream *st = NULL;
1860 if (rtsp_st->stream_index >= 0)
1861 st = s->streams[rtsp_st->stream_index];
1862 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1863 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1864 AVStream *st2 = NULL;
1865 if (rt->rtsp_streams[i]->stream_index >= 0)
1866 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1867 if (rtpctx2 && st && st2 &&
1868 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1869 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1870 rtpctx2->rtcp_ts_offset = av_rescale_q(
1871 rtpctx->rtcp_ts_offset, st->time_base,
1876 if (ret == -RTCP_BYE) {
1879 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1880 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1882 if (rt->nb_byes == rt->nb_rtsp_streams)
1886 } else if (rt->ts && CONFIG_RTPDEC) {
1887 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1890 rt->recvbuf_len = len;
1891 rt->recvbuf_pos = ret;
1892 rt->cur_transport_priv = rt->ts;
1899 return AVERROR_INVALIDDATA;
1905 /* more packets may follow, so we save the RTP context */
1906 rt->cur_transport_priv = rtsp_st->transport_priv;
1910 #endif /* CONFIG_RTPDEC */
1912 #if CONFIG_SDP_DEMUXER
1913 static int sdp_probe(AVProbeData *p1)
1915 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1917 /* we look for a line beginning "c=IN IP" */
1918 while (p < p_end && *p != '\0') {
1919 if (p + sizeof("c=IN IP") - 1 < p_end &&
1920 av_strstart(p, "c=IN IP", NULL))
1921 return AVPROBE_SCORE_MAX / 2;
1923 while (p < p_end - 1 && *p != '\n') p++;
1932 static int sdp_read_header(AVFormatContext *s)
1934 RTSPState *rt = s->priv_data;
1935 RTSPStream *rtsp_st;
1940 if (!ff_network_init())
1941 return AVERROR(EIO);
1943 if (s->max_delay < 0) /* Not set by the caller */
1944 s->max_delay = DEFAULT_REORDERING_DELAY;
1946 /* read the whole sdp file */
1947 /* XXX: better loading */
1948 content = av_malloc(SDP_MAX_SIZE);
1949 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1952 return AVERROR_INVALIDDATA;
1954 content[size] ='\0';
1956 err = ff_sdp_parse(s, content);
1960 /* open each RTP stream */
1961 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1963 rtsp_st = rt->rtsp_streams[i];
1965 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1966 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1967 ff_url_join(url, sizeof(url), "rtp", NULL,
1968 namebuf, rtsp_st->sdp_port,
1969 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1971 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1972 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1973 &s->interrupt_callback, NULL) < 0) {
1974 err = AVERROR_INVALIDDATA;
1977 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1982 ff_rtsp_close_streams(s);
1987 static int sdp_read_close(AVFormatContext *s)
1989 ff_rtsp_close_streams(s);
1994 static const AVClass sdp_demuxer_class = {
1995 .class_name = "SDP demuxer",
1996 .item_name = av_default_item_name,
1997 .option = sdp_options,
1998 .version = LIBAVUTIL_VERSION_INT,
2001 AVInputFormat ff_sdp_demuxer = {
2003 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2004 .priv_data_size = sizeof(RTSPState),
2005 .read_probe = sdp_probe,
2006 .read_header = sdp_read_header,
2007 .read_packet = ff_rtsp_fetch_packet,
2008 .read_close = sdp_read_close,
2009 .priv_class = &sdp_demuxer_class,
2011 #endif /* CONFIG_SDP_DEMUXER */
2013 #if CONFIG_RTP_DEMUXER
2014 static int rtp_probe(AVProbeData *p)
2016 if (av_strstart(p->filename, "rtp:", NULL))
2017 return AVPROBE_SCORE_MAX;
2021 static int rtp_read_header(AVFormatContext *s)
2023 uint8_t recvbuf[1500];
2024 char host[500], sdp[500];
2026 URLContext* in = NULL;
2028 AVCodecContext codec = { 0 };
2029 struct sockaddr_storage addr;
2031 socklen_t addrlen = sizeof(addr);
2032 RTSPState *rt = s->priv_data;
2034 if (!ff_network_init())
2035 return AVERROR(EIO);
2037 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2038 &s->interrupt_callback, NULL);
2043 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2044 if (ret == AVERROR(EAGAIN))
2049 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2053 if ((recvbuf[0] & 0xc0) != 0x80) {
2054 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2059 if (RTP_PT_IS_RTCP(recvbuf[1]))
2062 payload_type = recvbuf[1] & 0x7f;
2065 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2069 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2070 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2071 "without an SDP file describing it\n",
2075 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2076 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2077 "properly you need an SDP file "
2081 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2082 NULL, 0, s->filename);
2084 snprintf(sdp, sizeof(sdp),
2085 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2086 addr.ss_family == AF_INET ? 4 : 6, host,
2087 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2088 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2089 port, payload_type);
2090 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2092 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2095 /* sdp_read_header initializes this again */
2098 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2100 ret = sdp_read_header(s);
2111 static const AVClass rtp_demuxer_class = {
2112 .class_name = "RTP demuxer",
2113 .item_name = av_default_item_name,
2114 .option = rtp_options,
2115 .version = LIBAVUTIL_VERSION_INT,
2118 AVInputFormat ff_rtp_demuxer = {
2120 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2121 .priv_data_size = sizeof(RTSPState),
2122 .read_probe = rtp_probe,
2123 .read_header = rtp_read_header,
2124 .read_packet = ff_rtsp_fetch_packet,
2125 .read_close = sdp_read_close,
2126 .flags = AVFMT_NOFILE,
2127 .priv_class = &rtp_demuxer_class,
2129 #endif /* CONFIG_RTP_DEMUXER */