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);
158 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
159 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
162 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
164 struct addrinfo hints = { 0 }, *ai = NULL;
165 hints.ai_flags = AI_NUMERICHOST;
166 if (getaddrinfo(buf, NULL, &hints, &ai))
168 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
174 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
175 RTSPStream *rtsp_st, AVCodecContext *codec)
179 codec->codec_id = handler->codec_id;
180 rtsp_st->dynamic_handler = handler;
181 if (handler->alloc) {
182 rtsp_st->dynamic_protocol_context = handler->alloc();
183 if (!rtsp_st->dynamic_protocol_context)
184 rtsp_st->dynamic_handler = NULL;
188 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
189 static int sdp_parse_rtpmap(AVFormatContext *s,
190 AVStream *st, RTSPStream *rtsp_st,
191 int payload_type, const char *p)
193 AVCodecContext *codec = st->codec;
199 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
200 * see if we can handle this kind of payload.
201 * The space should normally not be there but some Real streams or
202 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
203 * have a trailing space. */
204 get_word_sep(buf, sizeof(buf), "/ ", &p);
205 if (payload_type < RTP_PT_PRIVATE) {
206 /* We are in a standard case
207 * (from http://www.iana.org/assignments/rtp-parameters). */
208 /* search into AVRtpPayloadTypes[] */
209 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
212 if (codec->codec_id == AV_CODEC_ID_NONE) {
213 RTPDynamicProtocolHandler *handler =
214 ff_rtp_handler_find_by_name(buf, codec->codec_type);
215 init_rtp_handler(handler, rtsp_st, codec);
216 /* If no dynamic handler was found, check with the list of standard
217 * allocated types, if such a stream for some reason happens to
218 * use a private payload type. This isn't handled in rtpdec.c, since
219 * the format name from the rtpmap line never is passed into rtpdec. */
220 if (!rtsp_st->dynamic_handler)
221 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
224 c = avcodec_find_decoder(codec->codec_id);
230 get_word_sep(buf, sizeof(buf), "/", &p);
232 switch (codec->codec_type) {
233 case AVMEDIA_TYPE_AUDIO:
234 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
235 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
236 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
238 codec->sample_rate = i;
239 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
240 get_word_sep(buf, sizeof(buf), "/", &p);
244 // TODO: there is a bug here; if it is a mono stream, and
245 // less than 22000Hz, faad upconverts to stereo and twice
246 // the frequency. No problem, but the sample rate is being
247 // set here by the sdp line. Patch on its way. (rdm)
249 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
251 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
254 case AVMEDIA_TYPE_VIDEO:
255 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
257 avpriv_set_pts_info(st, 32, 1, i);
262 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
263 rtsp_st->dynamic_handler->init(s, st->index,
264 rtsp_st->dynamic_protocol_context);
268 /* parse the attribute line from the fmtp a line of an sdp response. This
269 * is broken out as a function because it is used in rtp_h264.c, which is
271 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
272 char *value, int value_size)
274 *p += strspn(*p, SPACE_CHARS);
276 get_word_sep(attr, attr_size, "=", p);
279 get_word_sep(value, value_size, ";", p);
287 typedef struct SDPParseState {
289 struct sockaddr_storage default_ip;
291 int skip_media; ///< set if an unknown m= line occurs
294 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
295 int letter, const char *buf)
297 RTSPState *rt = s->priv_data;
298 char buf1[64], st_type[64];
300 enum AVMediaType codec_type;
304 struct sockaddr_storage sdp_ip;
307 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
310 if (s1->skip_media && letter != 'm')
314 get_word(buf1, sizeof(buf1), &p);
315 if (strcmp(buf1, "IN") != 0)
317 get_word(buf1, sizeof(buf1), &p);
318 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
320 get_word_sep(buf1, sizeof(buf1), "/", &p);
321 if (get_sockaddr(buf1, &sdp_ip))
326 get_word_sep(buf1, sizeof(buf1), "/", &p);
329 if (s->nb_streams == 0) {
330 s1->default_ip = sdp_ip;
331 s1->default_ttl = ttl;
333 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
334 rtsp_st->sdp_ip = sdp_ip;
335 rtsp_st->sdp_ttl = ttl;
339 av_dict_set(&s->metadata, "title", p, 0);
342 if (s->nb_streams == 0) {
343 av_dict_set(&s->metadata, "comment", p, 0);
350 codec_type = AVMEDIA_TYPE_UNKNOWN;
351 get_word(st_type, sizeof(st_type), &p);
352 if (!strcmp(st_type, "audio")) {
353 codec_type = AVMEDIA_TYPE_AUDIO;
354 } else if (!strcmp(st_type, "video")) {
355 codec_type = AVMEDIA_TYPE_VIDEO;
356 } else if (!strcmp(st_type, "application")) {
357 codec_type = AVMEDIA_TYPE_DATA;
359 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
363 rtsp_st = av_mallocz(sizeof(RTSPStream));
366 rtsp_st->stream_index = -1;
367 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
369 rtsp_st->sdp_ip = s1->default_ip;
370 rtsp_st->sdp_ttl = s1->default_ttl;
372 get_word(buf1, sizeof(buf1), &p); /* port */
373 rtsp_st->sdp_port = atoi(buf1);
375 get_word(buf1, sizeof(buf1), &p); /* protocol */
376 if (!strcmp(buf1, "udp"))
377 rt->transport = RTSP_TRANSPORT_RAW;
379 /* XXX: handle list of formats */
380 get_word(buf1, sizeof(buf1), &p); /* format list */
381 rtsp_st->sdp_payload_type = atoi(buf1);
383 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
384 /* no corresponding stream */
385 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
386 rt->ts = ff_mpegts_parse_open(s);
387 } else if (rt->server_type == RTSP_SERVER_WMS &&
388 codec_type == AVMEDIA_TYPE_DATA) {
389 /* RTX stream, a stream that carries all the other actual
390 * audio/video streams. Don't expose this to the callers. */
392 st = avformat_new_stream(s, NULL);
395 st->id = rt->nb_rtsp_streams - 1;
396 rtsp_st->stream_index = st->index;
397 st->codec->codec_type = codec_type;
398 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
399 RTPDynamicProtocolHandler *handler;
400 /* if standard payload type, we can find the codec right now */
401 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
402 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
403 st->codec->sample_rate > 0)
404 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
405 /* Even static payload types may need a custom depacketizer */
406 handler = ff_rtp_handler_find_by_id(
407 rtsp_st->sdp_payload_type, st->codec->codec_type);
408 init_rtp_handler(handler, rtsp_st, st->codec);
409 if (handler && handler->init)
410 handler->init(s, st->index,
411 rtsp_st->dynamic_protocol_context);
414 /* put a default control url */
415 av_strlcpy(rtsp_st->control_url, rt->control_uri,
416 sizeof(rtsp_st->control_url));
419 if (av_strstart(p, "control:", &p)) {
420 if (s->nb_streams == 0) {
421 if (!strncmp(p, "rtsp://", 7))
422 av_strlcpy(rt->control_uri, p,
423 sizeof(rt->control_uri));
426 /* get the control url */
427 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
429 /* XXX: may need to add full url resolution */
430 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
432 if (proto[0] == '\0') {
433 /* relative control URL */
434 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
435 av_strlcat(rtsp_st->control_url, "/",
436 sizeof(rtsp_st->control_url));
437 av_strlcat(rtsp_st->control_url, p,
438 sizeof(rtsp_st->control_url));
440 av_strlcpy(rtsp_st->control_url, p,
441 sizeof(rtsp_st->control_url));
443 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
444 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
445 get_word(buf1, sizeof(buf1), &p);
446 payload_type = atoi(buf1);
447 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
448 if (rtsp_st->stream_index >= 0) {
449 st = s->streams[rtsp_st->stream_index];
450 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
452 } else if (av_strstart(p, "fmtp:", &p) ||
453 av_strstart(p, "framesize:", &p)) {
454 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
455 // let dynamic protocol handlers have a stab at the line.
456 get_word(buf1, sizeof(buf1), &p);
457 payload_type = atoi(buf1);
458 for (i = 0; i < rt->nb_rtsp_streams; i++) {
459 rtsp_st = rt->rtsp_streams[i];
460 if (rtsp_st->sdp_payload_type == payload_type &&
461 rtsp_st->dynamic_handler &&
462 rtsp_st->dynamic_handler->parse_sdp_a_line)
463 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
464 rtsp_st->dynamic_protocol_context, buf);
466 } else if (av_strstart(p, "range:", &p)) {
469 // this is so that seeking on a streamed file can work.
470 rtsp_parse_range_npt(p, &start, &end);
471 s->start_time = start;
472 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
473 s->duration = (end == AV_NOPTS_VALUE) ?
474 AV_NOPTS_VALUE : end - start;
475 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
477 rt->transport = RTSP_TRANSPORT_RDT;
478 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
480 st = s->streams[s->nb_streams - 1];
481 st->codec->sample_rate = atoi(p);
483 if (rt->server_type == RTSP_SERVER_WMS)
484 ff_wms_parse_sdp_a_line(s, p);
485 if (s->nb_streams > 0) {
486 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
488 if (rt->server_type == RTSP_SERVER_REAL)
489 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
491 if (rtsp_st->dynamic_handler &&
492 rtsp_st->dynamic_handler->parse_sdp_a_line)
493 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
494 rtsp_st->stream_index,
495 rtsp_st->dynamic_protocol_context, buf);
502 int ff_sdp_parse(AVFormatContext *s, const char *content)
504 RTSPState *rt = s->priv_data;
507 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
508 * contain long SDP lines containing complete ASF Headers (several
509 * kB) or arrays of MDPR (RM stream descriptor) headers plus
510 * "rulebooks" describing their properties. Therefore, the SDP line
513 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
514 * in rtpdec_xiph.c. */
516 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
520 p += strspn(p, SPACE_CHARS);
528 /* get the content */
530 while (*p != '\n' && *p != '\r' && *p != '\0') {
531 if ((q - buf) < sizeof(buf) - 1)
536 sdp_parse_line(s, s1, letter, buf);
538 while (*p != '\n' && *p != '\0')
543 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
544 if (!rt->p) return AVERROR(ENOMEM);
547 #endif /* CONFIG_RTPDEC */
549 void ff_rtsp_undo_setup(AVFormatContext *s)
551 RTSPState *rt = s->priv_data;
554 for (i = 0; i < rt->nb_rtsp_streams; i++) {
555 RTSPStream *rtsp_st = rt->rtsp_streams[i];
558 if (rtsp_st->transport_priv) {
560 AVFormatContext *rtpctx = rtsp_st->transport_priv;
561 av_write_trailer(rtpctx);
562 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
564 avio_close_dyn_buf(rtpctx->pb, &ptr);
567 avio_close(rtpctx->pb);
569 avformat_free_context(rtpctx);
570 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
571 ff_rdt_parse_close(rtsp_st->transport_priv);
572 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
573 ff_rtp_parse_close(rtsp_st->transport_priv);
575 rtsp_st->transport_priv = NULL;
576 if (rtsp_st->rtp_handle)
577 ffurl_close(rtsp_st->rtp_handle);
578 rtsp_st->rtp_handle = NULL;
582 /* close and free RTSP streams */
583 void ff_rtsp_close_streams(AVFormatContext *s)
585 RTSPState *rt = s->priv_data;
589 ff_rtsp_undo_setup(s);
590 for (i = 0; i < rt->nb_rtsp_streams; i++) {
591 rtsp_st = rt->rtsp_streams[i];
593 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
594 rtsp_st->dynamic_handler->free(
595 rtsp_st->dynamic_protocol_context);
599 av_free(rt->rtsp_streams);
601 avformat_close_input(&rt->asf_ctx);
603 if (rt->ts && CONFIG_RTPDEC)
604 ff_mpegts_parse_close(rt->ts);
606 av_free(rt->recvbuf);
609 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
611 RTSPState *rt = s->priv_data;
614 /* open the RTP context */
615 if (rtsp_st->stream_index >= 0)
616 st = s->streams[rtsp_st->stream_index];
618 s->ctx_flags |= AVFMTCTX_NOHEADER;
620 if (s->oformat && CONFIG_RTSP_MUXER) {
621 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
623 RTSP_TCP_MAX_PACKET_SIZE);
624 /* Ownership of rtp_handle is passed to the rtp mux context */
625 rtsp_st->rtp_handle = NULL;
628 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
629 return 0; // Don't need to open any parser here
630 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
631 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
632 rtsp_st->dynamic_protocol_context,
633 rtsp_st->dynamic_handler);
634 else if (CONFIG_RTPDEC)
635 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
636 rtsp_st->sdp_payload_type,
637 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
638 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
640 if (!rtsp_st->transport_priv) {
641 return AVERROR(ENOMEM);
642 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
643 if (rtsp_st->dynamic_handler) {
644 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
645 rtsp_st->dynamic_protocol_context,
646 rtsp_st->dynamic_handler);
653 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
654 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
661 q += strspn(q, SPACE_CHARS);
662 v = strtol(q, &p, 10);
666 v = strtol(p, &p, 10);
675 /* XXX: only one transport specification is parsed */
676 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
678 char transport_protocol[16];
680 char lower_transport[16];
682 RTSPTransportField *th;
685 reply->nb_transports = 0;
688 p += strspn(p, SPACE_CHARS);
692 th = &reply->transports[reply->nb_transports];
694 get_word_sep(transport_protocol, sizeof(transport_protocol),
696 if (!av_strcasecmp (transport_protocol, "rtp")) {
697 get_word_sep(profile, sizeof(profile), "/;,", &p);
698 lower_transport[0] = '\0';
699 /* rtp/avp/<protocol> */
701 get_word_sep(lower_transport, sizeof(lower_transport),
704 th->transport = RTSP_TRANSPORT_RTP;
705 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
706 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
707 /* x-pn-tng/<protocol> */
708 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
710 th->transport = RTSP_TRANSPORT_RDT;
711 } else if (!av_strcasecmp(transport_protocol, "raw")) {
712 get_word_sep(profile, sizeof(profile), "/;,", &p);
713 lower_transport[0] = '\0';
714 /* raw/raw/<protocol> */
716 get_word_sep(lower_transport, sizeof(lower_transport),
719 th->transport = RTSP_TRANSPORT_RAW;
721 if (!av_strcasecmp(lower_transport, "TCP"))
722 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
724 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
728 /* get each parameter */
729 while (*p != '\0' && *p != ',') {
730 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
731 if (!strcmp(parameter, "port")) {
734 rtsp_parse_range(&th->port_min, &th->port_max, &p);
736 } else if (!strcmp(parameter, "client_port")) {
739 rtsp_parse_range(&th->client_port_min,
740 &th->client_port_max, &p);
742 } else if (!strcmp(parameter, "server_port")) {
745 rtsp_parse_range(&th->server_port_min,
746 &th->server_port_max, &p);
748 } else if (!strcmp(parameter, "interleaved")) {
751 rtsp_parse_range(&th->interleaved_min,
752 &th->interleaved_max, &p);
754 } else if (!strcmp(parameter, "multicast")) {
755 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
756 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
757 } else if (!strcmp(parameter, "ttl")) {
760 th->ttl = strtol(p, (char **)&p, 10);
762 } else if (!strcmp(parameter, "destination")) {
765 get_word_sep(buf, sizeof(buf), ";,", &p);
766 get_sockaddr(buf, &th->destination);
768 } else if (!strcmp(parameter, "source")) {
771 get_word_sep(buf, sizeof(buf), ";,", &p);
772 av_strlcpy(th->source, buf, sizeof(th->source));
774 } else if (!strcmp(parameter, "mode")) {
777 get_word_sep(buf, sizeof(buf), ";, ", &p);
778 if (!strcmp(buf, "record") ||
779 !strcmp(buf, "receive"))
784 while (*p != ';' && *p != '\0' && *p != ',')
792 reply->nb_transports++;
796 static void handle_rtp_info(RTSPState *rt, const char *url,
797 uint32_t seq, uint32_t rtptime)
800 if (!rtptime || !url[0])
802 if (rt->transport != RTSP_TRANSPORT_RTP)
804 for (i = 0; i < rt->nb_rtsp_streams; i++) {
805 RTSPStream *rtsp_st = rt->rtsp_streams[i];
806 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
809 if (!strcmp(rtsp_st->control_url, url)) {
810 rtpctx->base_timestamp = rtptime;
816 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
819 char key[20], value[1024], url[1024] = "";
820 uint32_t seq = 0, rtptime = 0;
823 p += strspn(p, SPACE_CHARS);
826 get_word_sep(key, sizeof(key), "=", &p);
830 get_word_sep(value, sizeof(value), ";, ", &p);
832 if (!strcmp(key, "url"))
833 av_strlcpy(url, value, sizeof(url));
834 else if (!strcmp(key, "seq"))
835 seq = strtoul(value, NULL, 10);
836 else if (!strcmp(key, "rtptime"))
837 rtptime = strtoul(value, NULL, 10);
839 handle_rtp_info(rt, url, seq, rtptime);
848 handle_rtp_info(rt, url, seq, rtptime);
851 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
852 RTSPState *rt, const char *method)
856 /* NOTE: we do case independent match for broken servers */
858 if (av_stristart(p, "Session:", &p)) {
860 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
861 if (av_stristart(p, ";timeout=", &p) &&
862 (t = strtol(p, NULL, 10)) > 0) {
865 } else if (av_stristart(p, "Content-Length:", &p)) {
866 reply->content_length = strtol(p, NULL, 10);
867 } else if (av_stristart(p, "Transport:", &p)) {
868 rtsp_parse_transport(reply, p);
869 } else if (av_stristart(p, "CSeq:", &p)) {
870 reply->seq = strtol(p, NULL, 10);
871 } else if (av_stristart(p, "Range:", &p)) {
872 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
873 } else if (av_stristart(p, "RealChallenge1:", &p)) {
874 p += strspn(p, SPACE_CHARS);
875 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
876 } else if (av_stristart(p, "Server:", &p)) {
877 p += strspn(p, SPACE_CHARS);
878 av_strlcpy(reply->server, p, sizeof(reply->server));
879 } else if (av_stristart(p, "Notice:", &p) ||
880 av_stristart(p, "X-Notice:", &p)) {
881 reply->notice = strtol(p, NULL, 10);
882 } else if (av_stristart(p, "Location:", &p)) {
883 p += strspn(p, SPACE_CHARS);
884 av_strlcpy(reply->location, p , sizeof(reply->location));
885 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
886 p += strspn(p, SPACE_CHARS);
887 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
888 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
889 p += strspn(p, SPACE_CHARS);
890 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
891 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
892 p += strspn(p, SPACE_CHARS);
893 if (method && !strcmp(method, "DESCRIBE"))
894 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
895 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
896 p += strspn(p, SPACE_CHARS);
897 if (method && !strcmp(method, "PLAY"))
898 rtsp_parse_rtp_info(rt, p);
899 } else if (av_stristart(p, "Public:", &p) && rt) {
900 if (strstr(p, "GET_PARAMETER") &&
901 method && !strcmp(method, "OPTIONS"))
902 rt->get_parameter_supported = 1;
903 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
904 p += strspn(p, SPACE_CHARS);
905 rt->accept_dynamic_rate = atoi(p);
906 } else if (av_stristart(p, "Content-Type:", &p)) {
907 p += strspn(p, SPACE_CHARS);
908 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
912 /* skip a RTP/TCP interleaved packet */
913 void ff_rtsp_skip_packet(AVFormatContext *s)
915 RTSPState *rt = s->priv_data;
919 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
922 len = AV_RB16(buf + 1);
924 av_dlog(s, "skipping RTP packet len=%d\n", len);
929 if (len1 > sizeof(buf))
931 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
938 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
939 unsigned char **content_ptr,
940 int return_on_interleaved_data, const char *method)
942 RTSPState *rt = s->priv_data;
943 char buf[4096], buf1[1024], *q;
946 int ret, content_length, line_count = 0, request = 0;
947 unsigned char *content = NULL;
953 memset(reply, 0, sizeof(*reply));
955 /* parse reply (XXX: use buffers) */
956 rt->last_reply[0] = '\0';
960 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
961 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
967 /* XXX: only parse it if first char on line ? */
968 if (return_on_interleaved_data) {
971 ff_rtsp_skip_packet(s);
972 } else if (ch != '\r') {
973 if ((q - buf) < sizeof(buf) - 1)
979 av_dlog(s, "line='%s'\n", buf);
981 /* test if last line */
985 if (line_count == 0) {
987 get_word(buf1, sizeof(buf1), &p);
988 if (!strncmp(buf1, "RTSP/", 5)) {
989 get_word(buf1, sizeof(buf1), &p);
990 reply->status_code = atoi(buf1);
991 av_strlcpy(reply->reason, p, sizeof(reply->reason));
993 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
994 get_word(buf1, sizeof(buf1), &p); // object
998 ff_rtsp_parse_line(reply, p, rt, method);
999 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1000 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1005 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1006 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1008 content_length = reply->content_length;
1009 if (content_length > 0) {
1010 /* leave some room for a trailing '\0' (useful for simple parsing) */
1011 content = av_malloc(content_length + 1);
1012 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1013 content[content_length] = '\0';
1016 *content_ptr = content;
1022 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1023 const char* ptr = buf;
1025 if (!strcmp(reply->reason, "OPTIONS")) {
1026 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1028 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1029 if (reply->session_id[0])
1030 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1033 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1035 av_strlcat(buf, "\r\n", sizeof(buf));
1037 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1038 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1041 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1043 rt->last_cmd_time = av_gettime();
1044 /* Even if the request from the server had data, it is not the data
1045 * that the caller wants or expects. The memory could also be leaked
1046 * if the actual following reply has content data. */
1048 av_freep(content_ptr);
1049 /* If method is set, this is called from ff_rtsp_send_cmd,
1050 * where a reply to exactly this request is awaited. For
1051 * callers from within packet receiving, we just want to
1052 * return to the caller and go back to receiving packets. */
1058 if (rt->seq != reply->seq) {
1059 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1060 rt->seq, reply->seq);
1064 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1065 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1066 reply->notice == 2306 /* Continuous Feed Terminated */) {
1067 rt->state = RTSP_STATE_IDLE;
1068 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1069 return AVERROR(EIO); /* data or server error */
1070 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1071 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1072 return AVERROR(EPERM);
1078 * Send a command to the RTSP server without waiting for the reply.
1080 * @param s RTSP (de)muxer context
1081 * @param method the method for the request
1082 * @param url the target url for the request
1083 * @param headers extra header lines to include in the request
1084 * @param send_content if non-null, the data to send as request body content
1085 * @param send_content_length the length of the send_content data, or 0 if
1086 * send_content is null
1088 * @return zero if success, nonzero otherwise
1090 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1091 const char *method, const char *url,
1092 const char *headers,
1093 const unsigned char *send_content,
1094 int send_content_length)
1096 RTSPState *rt = s->priv_data;
1097 char buf[4096], *out_buf;
1098 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1100 /* Add in RTSP headers */
1103 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1105 av_strlcat(buf, headers, sizeof(buf));
1106 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1107 if (rt->session_id[0] != '\0' && (!headers ||
1108 !strstr(headers, "\nIf-Match:"))) {
1109 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1112 char *str = ff_http_auth_create_response(&rt->auth_state,
1113 rt->auth, url, method);
1115 av_strlcat(buf, str, sizeof(buf));
1118 if (send_content_length > 0 && send_content)
1119 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1120 av_strlcat(buf, "\r\n", sizeof(buf));
1122 /* base64 encode rtsp if tunneling */
1123 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1124 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1125 out_buf = base64buf;
1128 av_dlog(s, "Sending:\n%s--\n", buf);
1130 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1131 if (send_content_length > 0 && send_content) {
1132 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1133 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1134 "with content data not supported\n");
1135 return AVERROR_PATCHWELCOME;
1137 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1139 rt->last_cmd_time = av_gettime();
1144 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1145 const char *url, const char *headers)
1147 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1150 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1151 const char *headers, RTSPMessageHeader *reply,
1152 unsigned char **content_ptr)
1154 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1155 content_ptr, NULL, 0);
1158 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1159 const char *method, const char *url,
1161 RTSPMessageHeader *reply,
1162 unsigned char **content_ptr,
1163 const unsigned char *send_content,
1164 int send_content_length)
1166 RTSPState *rt = s->priv_data;
1167 HTTPAuthType cur_auth_type;
1168 int ret, attempts = 0;
1171 cur_auth_type = rt->auth_state.auth_type;
1172 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1174 send_content_length)))
1177 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1181 if (reply->status_code == 401 &&
1182 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1183 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1186 if (reply->status_code > 400){
1187 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1191 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1197 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1198 int lower_transport, const char *real_challenge)
1200 RTSPState *rt = s->priv_data;
1201 int rtx = 0, j, i, err, interleave = 0, port_off;
1202 RTSPStream *rtsp_st;
1203 RTSPMessageHeader reply1, *reply = &reply1;
1205 const char *trans_pref;
1207 if (rt->transport == RTSP_TRANSPORT_RDT)
1208 trans_pref = "x-pn-tng";
1209 else if (rt->transport == RTSP_TRANSPORT_RAW)
1210 trans_pref = "RAW/RAW";
1212 trans_pref = "RTP/AVP";
1214 /* default timeout: 1 minute */
1217 /* Choose a random starting offset within the first half of the
1218 * port range, to allow for a number of ports to try even if the offset
1219 * happens to be at the end of the random range. */
1220 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1221 /* even random offset */
1222 port_off -= port_off & 0x01;
1224 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1225 char transport[2048];
1228 * WMS serves all UDP data over a single connection, the RTX, which
1229 * isn't necessarily the first in the SDP but has to be the first
1230 * to be set up, else the second/third SETUP will fail with a 461.
1232 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1233 rt->server_type == RTSP_SERVER_WMS) {
1236 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1237 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1239 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1243 if (rtx == rt->nb_rtsp_streams)
1244 return -1; /* no RTX found */
1245 rtsp_st = rt->rtsp_streams[rtx];
1247 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1249 rtsp_st = rt->rtsp_streams[i];
1252 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1255 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1256 port = reply->transports[0].client_port_min;
1260 /* first try in specified port range */
1261 while (j <= rt->rtp_port_max) {
1262 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1263 "?localport=%d", j);
1264 /* we will use two ports per rtp stream (rtp and rtcp) */
1266 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1267 &s->interrupt_callback, NULL))
1270 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1275 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1277 snprintf(transport, sizeof(transport) - 1,
1278 "%s/UDP;", trans_pref);
1279 if (rt->server_type != RTSP_SERVER_REAL)
1280 av_strlcat(transport, "unicast;", sizeof(transport));
1281 av_strlcatf(transport, sizeof(transport),
1282 "client_port=%d", port);
1283 if (rt->transport == RTSP_TRANSPORT_RTP &&
1284 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1285 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1289 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1290 /* For WMS streams, the application streams are only used for
1291 * UDP. When trying to set it up for TCP streams, the server
1292 * will return an error. Therefore, we skip those streams. */
1293 if (rt->server_type == RTSP_SERVER_WMS &&
1294 (rtsp_st->stream_index < 0 ||
1295 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1298 snprintf(transport, sizeof(transport) - 1,
1299 "%s/TCP;", trans_pref);
1300 if (rt->transport != RTSP_TRANSPORT_RDT)
1301 av_strlcat(transport, "unicast;", sizeof(transport));
1302 av_strlcatf(transport, sizeof(transport),
1303 "interleaved=%d-%d",
1304 interleave, interleave + 1);
1308 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1309 snprintf(transport, sizeof(transport) - 1,
1310 "%s/UDP;multicast", trans_pref);
1313 av_strlcat(transport, ";mode=record", sizeof(transport));
1314 } else if (rt->server_type == RTSP_SERVER_REAL ||
1315 rt->server_type == RTSP_SERVER_WMS)
1316 av_strlcat(transport, ";mode=play", sizeof(transport));
1317 snprintf(cmd, sizeof(cmd),
1318 "Transport: %s\r\n",
1320 if (rt->accept_dynamic_rate)
1321 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1322 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1323 char real_res[41], real_csum[9];
1324 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1326 av_strlcatf(cmd, sizeof(cmd),
1328 "RealChallenge2: %s, sd=%s\r\n",
1329 rt->session_id, real_res, real_csum);
1331 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1332 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1335 } else if (reply->status_code != RTSP_STATUS_OK ||
1336 reply->nb_transports != 1) {
1337 err = AVERROR_INVALIDDATA;
1341 /* XXX: same protocol for all streams is required */
1343 if (reply->transports[0].lower_transport != rt->lower_transport ||
1344 reply->transports[0].transport != rt->transport) {
1345 err = AVERROR_INVALIDDATA;
1349 rt->lower_transport = reply->transports[0].lower_transport;
1350 rt->transport = reply->transports[0].transport;
1353 /* Fail if the server responded with another lower transport mode
1354 * than what we requested. */
1355 if (reply->transports[0].lower_transport != lower_transport) {
1356 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1357 err = AVERROR_INVALIDDATA;
1361 switch(reply->transports[0].lower_transport) {
1362 case RTSP_LOWER_TRANSPORT_TCP:
1363 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1364 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1367 case RTSP_LOWER_TRANSPORT_UDP: {
1368 char url[1024], options[30] = "";
1370 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1371 av_strlcpy(options, "?connect=1", sizeof(options));
1372 /* Use source address if specified */
1373 if (reply->transports[0].source[0]) {
1374 ff_url_join(url, sizeof(url), "rtp", NULL,
1375 reply->transports[0].source,
1376 reply->transports[0].server_port_min, "%s", options);
1378 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1379 reply->transports[0].server_port_min, "%s", options);
1381 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1382 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1383 err = AVERROR_INVALIDDATA;
1386 /* Try to initialize the connection state in a
1387 * potential NAT router by sending dummy packets.
1388 * RTP/RTCP dummy packets are used for RDT, too.
1390 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1392 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1395 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1396 char url[1024], namebuf[50], optbuf[20] = "";
1397 struct sockaddr_storage addr;
1400 if (reply->transports[0].destination.ss_family) {
1401 addr = reply->transports[0].destination;
1402 port = reply->transports[0].port_min;
1403 ttl = reply->transports[0].ttl;
1405 addr = rtsp_st->sdp_ip;
1406 port = rtsp_st->sdp_port;
1407 ttl = rtsp_st->sdp_ttl;
1410 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1411 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1412 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1413 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1414 port, "%s", optbuf);
1415 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1416 &s->interrupt_callback, NULL) < 0) {
1417 err = AVERROR_INVALIDDATA;
1424 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1428 if (rt->nb_rtsp_streams && reply->timeout > 0)
1429 rt->timeout = reply->timeout;
1431 if (rt->server_type == RTSP_SERVER_REAL)
1432 rt->need_subscription = 1;
1437 ff_rtsp_undo_setup(s);
1441 void ff_rtsp_close_connections(AVFormatContext *s)
1443 RTSPState *rt = s->priv_data;
1444 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1445 ffurl_close(rt->rtsp_hd);
1446 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1449 int ff_rtsp_connect(AVFormatContext *s)
1451 RTSPState *rt = s->priv_data;
1452 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1453 int port, err, tcp_fd;
1454 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1455 int lower_transport_mask = 0;
1456 char real_challenge[64] = "";
1457 struct sockaddr_storage peer;
1458 socklen_t peer_len = sizeof(peer);
1460 if (rt->rtp_port_max < rt->rtp_port_min) {
1461 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1462 "than min port %d\n", rt->rtp_port_max,
1464 return AVERROR(EINVAL);
1467 if (!ff_network_init())
1468 return AVERROR(EIO);
1470 if (s->max_delay < 0) /* Not set by the caller */
1471 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1473 rt->control_transport = RTSP_MODE_PLAIN;
1474 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1475 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1476 rt->control_transport = RTSP_MODE_TUNNEL;
1478 /* Only pass through valid flags from here */
1479 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1482 lower_transport_mask = rt->lower_transport_mask;
1483 /* extract hostname and port */
1484 av_url_split(NULL, 0, auth, sizeof(auth),
1485 host, sizeof(host), &port, path, sizeof(path), s->filename);
1487 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1490 port = RTSP_DEFAULT_PORT;
1492 if (!lower_transport_mask)
1493 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1496 /* Only UDP or TCP - UDP multicast isn't supported. */
1497 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1498 (1 << RTSP_LOWER_TRANSPORT_TCP);
1499 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1500 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1501 "only UDP and TCP are supported for output.\n");
1502 err = AVERROR(EINVAL);
1507 /* Construct the URI used in request; this is similar to s->filename,
1508 * but with authentication credentials removed and RTSP specific options
1510 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1511 host, port, "%s", path);
1513 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1514 /* set up initial handshake for tunneling */
1515 char httpname[1024];
1516 char sessioncookie[17];
1519 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1520 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1521 av_get_random_seed(), av_get_random_seed());
1524 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1525 &s->interrupt_callback) < 0) {
1530 /* generate GET headers */
1531 snprintf(headers, sizeof(headers),
1532 "x-sessioncookie: %s\r\n"
1533 "Accept: application/x-rtsp-tunnelled\r\n"
1534 "Pragma: no-cache\r\n"
1535 "Cache-Control: no-cache\r\n",
1537 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1539 /* complete the connection */
1540 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1546 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1547 &s->interrupt_callback) < 0 ) {
1552 /* generate POST headers */
1553 snprintf(headers, sizeof(headers),
1554 "x-sessioncookie: %s\r\n"
1555 "Content-Type: application/x-rtsp-tunnelled\r\n"
1556 "Pragma: no-cache\r\n"
1557 "Cache-Control: no-cache\r\n"
1558 "Content-Length: 32767\r\n"
1559 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1561 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1562 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1564 /* Initialize the authentication state for the POST session. The HTTP
1565 * protocol implementation doesn't properly handle multi-pass
1566 * authentication for POST requests, since it would require one of
1568 * - implementing Expect: 100-continue, which many HTTP servers
1569 * don't support anyway, even less the RTSP servers that do HTTP
1571 * - sending the whole POST data until getting a 401 reply specifying
1572 * what authentication method to use, then resending all that data
1573 * - waiting for potential 401 replies directly after sending the
1574 * POST header (waiting for some unspecified time)
1575 * Therefore, we copy the full auth state, which works for both basic
1576 * and digest. (For digest, we would have to synchronize the nonce
1577 * count variable between the two sessions, if we'd do more requests
1578 * with the original session, though.)
1580 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1582 /* complete the connection */
1583 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1588 /* open the tcp connection */
1589 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1590 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1591 &s->interrupt_callback, NULL) < 0) {
1595 rt->rtsp_hd_out = rt->rtsp_hd;
1599 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1600 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1601 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1602 NULL, 0, NI_NUMERICHOST);
1605 /* request options supported by the server; this also detects server
1607 for (rt->server_type = RTSP_SERVER_RTP;;) {
1609 if (rt->server_type == RTSP_SERVER_REAL)
1612 * The following entries are required for proper
1613 * streaming from a Realmedia server. They are
1614 * interdependent in some way although we currently
1615 * don't quite understand how. Values were copied
1616 * from mplayer SVN r23589.
1617 * ClientChallenge is a 16-byte ID in hex
1618 * CompanyID is a 16-byte ID in base64
1620 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1621 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1622 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1623 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1625 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1626 if (reply->status_code != RTSP_STATUS_OK) {
1627 err = AVERROR_INVALIDDATA;
1631 /* detect server type if not standard-compliant RTP */
1632 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1633 rt->server_type = RTSP_SERVER_REAL;
1635 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1636 rt->server_type = RTSP_SERVER_WMS;
1637 } else if (rt->server_type == RTSP_SERVER_REAL)
1638 strcpy(real_challenge, reply->real_challenge);
1642 if (s->iformat && CONFIG_RTSP_DEMUXER)
1643 err = ff_rtsp_setup_input_streams(s, reply);
1644 else if (CONFIG_RTSP_MUXER)
1645 err = ff_rtsp_setup_output_streams(s, host);
1650 int lower_transport = ff_log2_tab[lower_transport_mask &
1651 ~(lower_transport_mask - 1)];
1653 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1654 rt->server_type == RTSP_SERVER_REAL ?
1655 real_challenge : NULL);
1658 lower_transport_mask &= ~(1 << lower_transport);
1659 if (lower_transport_mask == 0 && err == 1) {
1660 err = AVERROR(EPROTONOSUPPORT);
1665 rt->lower_transport_mask = lower_transport_mask;
1666 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1667 rt->state = RTSP_STATE_IDLE;
1668 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1671 ff_rtsp_close_streams(s);
1672 ff_rtsp_close_connections(s);
1673 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1674 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1675 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1683 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1686 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1687 uint8_t *buf, int buf_size, int64_t wait_end)
1689 RTSPState *rt = s->priv_data;
1690 RTSPStream *rtsp_st;
1691 int n, i, ret, tcp_fd, timeout_cnt = 0;
1693 struct pollfd *p = rt->p;
1694 int *fds = NULL, fdsnum, fdsidx;
1697 if (ff_check_interrupt(&s->interrupt_callback))
1698 return AVERROR_EXIT;
1699 if (wait_end && wait_end - av_gettime() < 0)
1700 return AVERROR(EAGAIN);
1703 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1704 p[max_p].fd = tcp_fd;
1705 p[max_p++].events = POLLIN;
1709 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1710 rtsp_st = rt->rtsp_streams[i];
1711 if (rtsp_st->rtp_handle) {
1712 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1714 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1718 av_log(s, AV_LOG_ERROR,
1719 "Number of fds %d not supported\n", fdsnum);
1720 return AVERROR_INVALIDDATA;
1722 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1723 p[max_p].fd = fds[fdsidx];
1724 p[max_p++].events = POLLIN;
1729 n = poll(p, max_p, POLL_TIMEOUT_MS);
1731 int j = 1 - (tcp_fd == -1);
1733 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1734 rtsp_st = rt->rtsp_streams[i];
1735 if (rtsp_st->rtp_handle) {
1736 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1737 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1739 *prtsp_st = rtsp_st;
1746 #if CONFIG_RTSP_DEMUXER
1747 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1748 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1749 if (rt->state == RTSP_STATE_STREAMING) {
1750 if (!ff_rtsp_parse_streaming_commands(s))
1753 av_log(s, AV_LOG_WARNING,
1754 "Unable to answer to TEARDOWN\n");
1758 RTSPMessageHeader reply;
1759 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1762 /* XXX: parse message */
1763 if (rt->state != RTSP_STATE_STREAMING)
1768 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1769 return AVERROR(ETIMEDOUT);
1770 } else if (n < 0 && errno != EINTR)
1771 return AVERROR(errno);
1775 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1777 RTSPState *rt = s->priv_data;
1779 RTSPStream *rtsp_st, *first_queue_st = NULL;
1780 int64_t wait_end = 0;
1782 if (rt->nb_byes == rt->nb_rtsp_streams)
1785 /* get next frames from the same RTP packet */
1786 if (rt->cur_transport_priv) {
1787 if (rt->transport == RTSP_TRANSPORT_RDT) {
1788 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1789 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1790 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1791 } else if (rt->ts && CONFIG_RTPDEC) {
1792 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1794 rt->recvbuf_pos += ret;
1795 ret = rt->recvbuf_pos < rt->recvbuf_len;
1800 rt->cur_transport_priv = NULL;
1802 } else if (ret == 1) {
1805 rt->cur_transport_priv = NULL;
1808 if (rt->transport == RTSP_TRANSPORT_RTP) {
1810 int64_t first_queue_time = 0;
1811 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1812 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1816 queue_time = ff_rtp_queued_packet_time(rtpctx);
1817 if (queue_time && (queue_time - first_queue_time < 0 ||
1818 !first_queue_time)) {
1819 first_queue_time = queue_time;
1820 first_queue_st = rt->rtsp_streams[i];
1823 if (first_queue_time)
1824 wait_end = first_queue_time + s->max_delay;
1827 /* read next RTP packet */
1830 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1832 return AVERROR(ENOMEM);
1835 switch(rt->lower_transport) {
1837 #if CONFIG_RTSP_DEMUXER
1838 case RTSP_LOWER_TRANSPORT_TCP:
1839 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1842 case RTSP_LOWER_TRANSPORT_UDP:
1843 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1844 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1845 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1846 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1849 if (len == AVERROR(EAGAIN) && first_queue_st &&
1850 rt->transport == RTSP_TRANSPORT_RTP) {
1851 rtsp_st = first_queue_st;
1852 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1859 if (rt->transport == RTSP_TRANSPORT_RDT) {
1860 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1861 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1862 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1864 /* Either bad packet, or a RTCP packet. Check if the
1865 * first_rtcp_ntp_time field was initialized. */
1866 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1867 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1868 /* first_rtcp_ntp_time has been initialized for this stream,
1869 * copy the same value to all other uninitialized streams,
1870 * in order to map their timestamp origin to the same ntp time
1873 AVStream *st = NULL;
1874 if (rtsp_st->stream_index >= 0)
1875 st = s->streams[rtsp_st->stream_index];
1876 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1877 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1878 AVStream *st2 = NULL;
1879 if (rt->rtsp_streams[i]->stream_index >= 0)
1880 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1881 if (rtpctx2 && st && st2 &&
1882 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1883 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1884 rtpctx2->rtcp_ts_offset = av_rescale_q(
1885 rtpctx->rtcp_ts_offset, st->time_base,
1890 if (ret == -RTCP_BYE) {
1893 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1894 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1896 if (rt->nb_byes == rt->nb_rtsp_streams)
1900 } else if (rt->ts && CONFIG_RTPDEC) {
1901 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1904 rt->recvbuf_len = len;
1905 rt->recvbuf_pos = ret;
1906 rt->cur_transport_priv = rt->ts;
1913 return AVERROR_INVALIDDATA;
1919 /* more packets may follow, so we save the RTP context */
1920 rt->cur_transport_priv = rtsp_st->transport_priv;
1924 #endif /* CONFIG_RTPDEC */
1926 #if CONFIG_SDP_DEMUXER
1927 static int sdp_probe(AVProbeData *p1)
1929 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1931 /* we look for a line beginning "c=IN IP" */
1932 while (p < p_end && *p != '\0') {
1933 if (p + sizeof("c=IN IP") - 1 < p_end &&
1934 av_strstart(p, "c=IN IP", NULL))
1935 return AVPROBE_SCORE_MAX / 2;
1937 while (p < p_end - 1 && *p != '\n') p++;
1946 static int sdp_read_header(AVFormatContext *s)
1948 RTSPState *rt = s->priv_data;
1949 RTSPStream *rtsp_st;
1954 if (!ff_network_init())
1955 return AVERROR(EIO);
1957 if (s->max_delay < 0) /* Not set by the caller */
1958 s->max_delay = DEFAULT_REORDERING_DELAY;
1960 /* read the whole sdp file */
1961 /* XXX: better loading */
1962 content = av_malloc(SDP_MAX_SIZE);
1963 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1966 return AVERROR_INVALIDDATA;
1968 content[size] ='\0';
1970 err = ff_sdp_parse(s, content);
1974 /* open each RTP stream */
1975 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1977 rtsp_st = rt->rtsp_streams[i];
1979 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1980 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1981 ff_url_join(url, sizeof(url), "rtp", NULL,
1982 namebuf, rtsp_st->sdp_port,
1983 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1985 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1986 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1987 &s->interrupt_callback, NULL) < 0) {
1988 err = AVERROR_INVALIDDATA;
1991 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1996 ff_rtsp_close_streams(s);
2001 static int sdp_read_close(AVFormatContext *s)
2003 ff_rtsp_close_streams(s);
2008 static const AVClass sdp_demuxer_class = {
2009 .class_name = "SDP demuxer",
2010 .item_name = av_default_item_name,
2011 .option = sdp_options,
2012 .version = LIBAVUTIL_VERSION_INT,
2015 AVInputFormat ff_sdp_demuxer = {
2017 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2018 .priv_data_size = sizeof(RTSPState),
2019 .read_probe = sdp_probe,
2020 .read_header = sdp_read_header,
2021 .read_packet = ff_rtsp_fetch_packet,
2022 .read_close = sdp_read_close,
2023 .priv_class = &sdp_demuxer_class,
2025 #endif /* CONFIG_SDP_DEMUXER */
2027 #if CONFIG_RTP_DEMUXER
2028 static int rtp_probe(AVProbeData *p)
2030 if (av_strstart(p->filename, "rtp:", NULL))
2031 return AVPROBE_SCORE_MAX;
2035 static int rtp_read_header(AVFormatContext *s)
2037 uint8_t recvbuf[1500];
2038 char host[500], sdp[500];
2040 URLContext* in = NULL;
2042 AVCodecContext codec = { 0 };
2043 struct sockaddr_storage addr;
2045 socklen_t addrlen = sizeof(addr);
2046 RTSPState *rt = s->priv_data;
2048 if (!ff_network_init())
2049 return AVERROR(EIO);
2051 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2052 &s->interrupt_callback, NULL);
2057 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2058 if (ret == AVERROR(EAGAIN))
2063 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2067 if ((recvbuf[0] & 0xc0) != 0x80) {
2068 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2073 if (RTP_PT_IS_RTCP(recvbuf[1]))
2076 payload_type = recvbuf[1] & 0x7f;
2079 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2083 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2084 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2085 "without an SDP file describing it\n",
2089 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2090 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2091 "properly you need an SDP file "
2095 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2096 NULL, 0, s->filename);
2098 snprintf(sdp, sizeof(sdp),
2099 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2100 addr.ss_family == AF_INET ? 4 : 6, host,
2101 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2102 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2103 port, payload_type);
2104 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2106 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2109 /* sdp_read_header initializes this again */
2112 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2114 ret = sdp_read_header(s);
2125 static const AVClass rtp_demuxer_class = {
2126 .class_name = "RTP demuxer",
2127 .item_name = av_default_item_name,
2128 .option = rtp_options,
2129 .version = LIBAVUTIL_VERSION_INT,
2132 AVInputFormat ff_rtp_demuxer = {
2134 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2135 .priv_data_size = sizeof(RTSPState),
2136 .read_probe = rtp_probe,
2137 .read_header = rtp_read_header,
2138 .read_packet = ff_rtsp_fetch_packet,
2139 .read_close = sdp_read_close,
2140 .flags = AVFMT_NOFILE,
2141 .priv_class = &rtp_demuxer_class,
2143 #endif /* CONFIG_RTP_DEMUXER */