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 #define RTSP_REORDERING_OPTS() \
79 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
81 const AVOption ff_rtsp_options[] = {
82 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "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 RTSP_REORDERING_OPTS(),
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102 RTSP_REORDERING_OPTS(),
106 static const AVOption rtp_options[] = {
107 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108 RTSP_REORDERING_OPTS(),
112 static void get_word_until_chars(char *buf, int buf_size,
113 const char *sep, const char **pp)
119 p += strspn(p, SPACE_CHARS);
121 while (!strchr(sep, *p) && *p != '\0') {
122 if ((q - buf) < buf_size - 1)
131 static void get_word_sep(char *buf, int buf_size, const char *sep,
134 if (**pp == '/') (*pp)++;
135 get_word_until_chars(buf, buf_size, sep, pp);
138 static void get_word(char *buf, int buf_size, const char **pp)
140 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
143 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
145 * Used for seeking in the rtp stream.
147 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
151 p += strspn(p, SPACE_CHARS);
152 if (!av_stristart(p, "npt=", &p))
155 *start = AV_NOPTS_VALUE;
156 *end = AV_NOPTS_VALUE;
158 get_word_sep(buf, sizeof(buf), "-", &p);
159 av_parse_time(start, buf, 1);
162 get_word_sep(buf, sizeof(buf), "-", &p);
163 av_parse_time(end, buf, 1);
167 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
169 struct addrinfo hints = { 0 }, *ai = NULL;
170 hints.ai_flags = AI_NUMERICHOST;
171 if (getaddrinfo(buf, NULL, &hints, &ai))
173 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
179 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
180 RTSPStream *rtsp_st, AVCodecContext *codec)
184 codec->codec_id = handler->codec_id;
185 rtsp_st->dynamic_handler = handler;
186 if (handler->alloc) {
187 rtsp_st->dynamic_protocol_context = handler->alloc();
188 if (!rtsp_st->dynamic_protocol_context)
189 rtsp_st->dynamic_handler = NULL;
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195 AVStream *st, RTSPStream *rtsp_st,
196 int payload_type, const char *p)
198 AVCodecContext *codec = st->codec;
204 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
205 * see if we can handle this kind of payload.
206 * The space should normally not be there but some Real streams or
207 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
208 * have a trailing space. */
209 get_word_sep(buf, sizeof(buf), "/ ", &p);
210 if (payload_type < RTP_PT_PRIVATE) {
211 /* We are in a standard case
212 * (from http://www.iana.org/assignments/rtp-parameters). */
213 /* search into AVRtpPayloadTypes[] */
214 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
217 if (codec->codec_id == AV_CODEC_ID_NONE) {
218 RTPDynamicProtocolHandler *handler =
219 ff_rtp_handler_find_by_name(buf, codec->codec_type);
220 init_rtp_handler(handler, rtsp_st, codec);
221 /* If no dynamic handler was found, check with the list of standard
222 * allocated types, if such a stream for some reason happens to
223 * use a private payload type. This isn't handled in rtpdec.c, since
224 * the format name from the rtpmap line never is passed into rtpdec. */
225 if (!rtsp_st->dynamic_handler)
226 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
229 c = avcodec_find_decoder(codec->codec_id);
235 get_word_sep(buf, sizeof(buf), "/", &p);
237 switch (codec->codec_type) {
238 case AVMEDIA_TYPE_AUDIO:
239 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
240 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
241 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
243 codec->sample_rate = i;
244 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
245 get_word_sep(buf, sizeof(buf), "/", &p);
249 // TODO: there is a bug here; if it is a mono stream, and
250 // less than 22000Hz, faad upconverts to stereo and twice
251 // the frequency. No problem, but the sample rate is being
252 // set here by the sdp line. Patch on its way. (rdm)
254 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
256 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
259 case AVMEDIA_TYPE_VIDEO:
260 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
262 avpriv_set_pts_info(st, 32, 1, i);
267 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
268 rtsp_st->dynamic_handler->init(s, st->index,
269 rtsp_st->dynamic_protocol_context);
273 /* parse the attribute line from the fmtp a line of an sdp response. This
274 * is broken out as a function because it is used in rtp_h264.c, which is
276 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
277 char *value, int value_size)
279 *p += strspn(*p, SPACE_CHARS);
281 get_word_sep(attr, attr_size, "=", p);
284 get_word_sep(value, value_size, ";", p);
292 typedef struct SDPParseState {
294 struct sockaddr_storage default_ip;
296 int skip_media; ///< set if an unknown m= line occurs
299 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
300 int letter, const char *buf)
302 RTSPState *rt = s->priv_data;
303 char buf1[64], st_type[64];
305 enum AVMediaType codec_type;
309 struct sockaddr_storage sdp_ip;
312 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
315 if (s1->skip_media && letter != 'm')
319 get_word(buf1, sizeof(buf1), &p);
320 if (strcmp(buf1, "IN") != 0)
322 get_word(buf1, sizeof(buf1), &p);
323 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
325 get_word_sep(buf1, sizeof(buf1), "/", &p);
326 if (get_sockaddr(buf1, &sdp_ip))
331 get_word_sep(buf1, sizeof(buf1), "/", &p);
334 if (s->nb_streams == 0) {
335 s1->default_ip = sdp_ip;
336 s1->default_ttl = ttl;
338 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
339 rtsp_st->sdp_ip = sdp_ip;
340 rtsp_st->sdp_ttl = ttl;
344 av_dict_set(&s->metadata, "title", p, 0);
347 if (s->nb_streams == 0) {
348 av_dict_set(&s->metadata, "comment", p, 0);
355 codec_type = AVMEDIA_TYPE_UNKNOWN;
356 get_word(st_type, sizeof(st_type), &p);
357 if (!strcmp(st_type, "audio")) {
358 codec_type = AVMEDIA_TYPE_AUDIO;
359 } else if (!strcmp(st_type, "video")) {
360 codec_type = AVMEDIA_TYPE_VIDEO;
361 } else if (!strcmp(st_type, "application")) {
362 codec_type = AVMEDIA_TYPE_DATA;
364 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
368 rtsp_st = av_mallocz(sizeof(RTSPStream));
371 rtsp_st->stream_index = -1;
372 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
374 rtsp_st->sdp_ip = s1->default_ip;
375 rtsp_st->sdp_ttl = s1->default_ttl;
377 get_word(buf1, sizeof(buf1), &p); /* port */
378 rtsp_st->sdp_port = atoi(buf1);
380 get_word(buf1, sizeof(buf1), &p); /* protocol */
381 if (!strcmp(buf1, "udp"))
382 rt->transport = RTSP_TRANSPORT_RAW;
384 /* XXX: handle list of formats */
385 get_word(buf1, sizeof(buf1), &p); /* format list */
386 rtsp_st->sdp_payload_type = atoi(buf1);
388 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
389 /* no corresponding stream */
390 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
391 rt->ts = ff_mpegts_parse_open(s);
392 } else if (rt->server_type == RTSP_SERVER_WMS &&
393 codec_type == AVMEDIA_TYPE_DATA) {
394 /* RTX stream, a stream that carries all the other actual
395 * audio/video streams. Don't expose this to the callers. */
397 st = avformat_new_stream(s, NULL);
400 st->id = rt->nb_rtsp_streams - 1;
401 rtsp_st->stream_index = st->index;
402 st->codec->codec_type = codec_type;
403 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
404 RTPDynamicProtocolHandler *handler;
405 /* if standard payload type, we can find the codec right now */
406 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
407 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
408 st->codec->sample_rate > 0)
409 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
410 /* Even static payload types may need a custom depacketizer */
411 handler = ff_rtp_handler_find_by_id(
412 rtsp_st->sdp_payload_type, st->codec->codec_type);
413 init_rtp_handler(handler, rtsp_st, st->codec);
414 if (handler && handler->init)
415 handler->init(s, st->index,
416 rtsp_st->dynamic_protocol_context);
419 /* put a default control url */
420 av_strlcpy(rtsp_st->control_url, rt->control_uri,
421 sizeof(rtsp_st->control_url));
424 if (av_strstart(p, "control:", &p)) {
425 if (s->nb_streams == 0) {
426 if (!strncmp(p, "rtsp://", 7))
427 av_strlcpy(rt->control_uri, p,
428 sizeof(rt->control_uri));
431 /* get the control url */
432 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
434 /* XXX: may need to add full url resolution */
435 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
437 if (proto[0] == '\0') {
438 /* relative control URL */
439 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
440 av_strlcat(rtsp_st->control_url, "/",
441 sizeof(rtsp_st->control_url));
442 av_strlcat(rtsp_st->control_url, p,
443 sizeof(rtsp_st->control_url));
445 av_strlcpy(rtsp_st->control_url, p,
446 sizeof(rtsp_st->control_url));
448 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
449 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
450 get_word(buf1, sizeof(buf1), &p);
451 payload_type = atoi(buf1);
452 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
453 if (rtsp_st->stream_index >= 0) {
454 st = s->streams[rtsp_st->stream_index];
455 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
457 } else if (av_strstart(p, "fmtp:", &p) ||
458 av_strstart(p, "framesize:", &p)) {
459 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
460 // let dynamic protocol handlers have a stab at the line.
461 get_word(buf1, sizeof(buf1), &p);
462 payload_type = atoi(buf1);
463 for (i = 0; i < rt->nb_rtsp_streams; i++) {
464 rtsp_st = rt->rtsp_streams[i];
465 if (rtsp_st->sdp_payload_type == payload_type &&
466 rtsp_st->dynamic_handler &&
467 rtsp_st->dynamic_handler->parse_sdp_a_line)
468 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
469 rtsp_st->dynamic_protocol_context, buf);
471 } else if (av_strstart(p, "range:", &p)) {
474 // this is so that seeking on a streamed file can work.
475 rtsp_parse_range_npt(p, &start, &end);
476 s->start_time = start;
477 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
478 s->duration = (end == AV_NOPTS_VALUE) ?
479 AV_NOPTS_VALUE : end - start;
480 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
482 rt->transport = RTSP_TRANSPORT_RDT;
483 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
485 st = s->streams[s->nb_streams - 1];
486 st->codec->sample_rate = atoi(p);
488 if (rt->server_type == RTSP_SERVER_WMS)
489 ff_wms_parse_sdp_a_line(s, p);
490 if (s->nb_streams > 0) {
491 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
493 if (rt->server_type == RTSP_SERVER_REAL)
494 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
496 if (rtsp_st->dynamic_handler &&
497 rtsp_st->dynamic_handler->parse_sdp_a_line)
498 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
499 rtsp_st->stream_index,
500 rtsp_st->dynamic_protocol_context, buf);
507 int ff_sdp_parse(AVFormatContext *s, const char *content)
509 RTSPState *rt = s->priv_data;
512 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
513 * contain long SDP lines containing complete ASF Headers (several
514 * kB) or arrays of MDPR (RM stream descriptor) headers plus
515 * "rulebooks" describing their properties. Therefore, the SDP line
518 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
519 * in rtpdec_xiph.c. */
521 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
525 p += strspn(p, SPACE_CHARS);
533 /* get the content */
535 while (*p != '\n' && *p != '\r' && *p != '\0') {
536 if ((q - buf) < sizeof(buf) - 1)
541 sdp_parse_line(s, s1, letter, buf);
543 while (*p != '\n' && *p != '\0')
548 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
549 if (!rt->p) return AVERROR(ENOMEM);
552 #endif /* CONFIG_RTPDEC */
554 void ff_rtsp_undo_setup(AVFormatContext *s)
556 RTSPState *rt = s->priv_data;
559 for (i = 0; i < rt->nb_rtsp_streams; i++) {
560 RTSPStream *rtsp_st = rt->rtsp_streams[i];
563 if (rtsp_st->transport_priv) {
565 AVFormatContext *rtpctx = rtsp_st->transport_priv;
566 av_write_trailer(rtpctx);
567 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
569 avio_close_dyn_buf(rtpctx->pb, &ptr);
572 avio_close(rtpctx->pb);
574 avformat_free_context(rtpctx);
575 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
576 ff_rdt_parse_close(rtsp_st->transport_priv);
577 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
578 ff_rtp_parse_close(rtsp_st->transport_priv);
580 rtsp_st->transport_priv = NULL;
581 if (rtsp_st->rtp_handle)
582 ffurl_close(rtsp_st->rtp_handle);
583 rtsp_st->rtp_handle = NULL;
587 /* close and free RTSP streams */
588 void ff_rtsp_close_streams(AVFormatContext *s)
590 RTSPState *rt = s->priv_data;
594 ff_rtsp_undo_setup(s);
595 for (i = 0; i < rt->nb_rtsp_streams; i++) {
596 rtsp_st = rt->rtsp_streams[i];
598 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
599 rtsp_st->dynamic_handler->free(
600 rtsp_st->dynamic_protocol_context);
604 av_free(rt->rtsp_streams);
606 avformat_close_input(&rt->asf_ctx);
608 if (rt->ts && CONFIG_RTPDEC)
609 ff_mpegts_parse_close(rt->ts);
611 av_free(rt->recvbuf);
614 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
616 RTSPState *rt = s->priv_data;
618 int reordering_queue_size = rt->reordering_queue_size;
619 if (reordering_queue_size < 0) {
620 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
621 reordering_queue_size = 0;
623 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
626 /* open the RTP context */
627 if (rtsp_st->stream_index >= 0)
628 st = s->streams[rtsp_st->stream_index];
630 s->ctx_flags |= AVFMTCTX_NOHEADER;
632 if (s->oformat && CONFIG_RTSP_MUXER) {
633 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
635 RTSP_TCP_MAX_PACKET_SIZE,
636 rtsp_st->stream_index);
637 /* Ownership of rtp_handle is passed to the rtp mux context */
638 rtsp_st->rtp_handle = NULL;
641 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
642 return 0; // Don't need to open any parser here
643 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
644 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
645 rtsp_st->dynamic_protocol_context,
646 rtsp_st->dynamic_handler);
647 else if (CONFIG_RTPDEC)
648 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
649 rtsp_st->sdp_payload_type,
650 reordering_queue_size);
652 if (!rtsp_st->transport_priv) {
653 return AVERROR(ENOMEM);
654 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
655 if (rtsp_st->dynamic_handler) {
656 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
657 rtsp_st->dynamic_protocol_context,
658 rtsp_st->dynamic_handler);
665 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
666 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
673 q += strspn(q, SPACE_CHARS);
674 v = strtol(q, &p, 10);
678 v = strtol(p, &p, 10);
687 /* XXX: only one transport specification is parsed */
688 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
690 char transport_protocol[16];
692 char lower_transport[16];
694 RTSPTransportField *th;
697 reply->nb_transports = 0;
700 p += strspn(p, SPACE_CHARS);
704 th = &reply->transports[reply->nb_transports];
706 get_word_sep(transport_protocol, sizeof(transport_protocol),
708 if (!av_strcasecmp (transport_protocol, "rtp")) {
709 get_word_sep(profile, sizeof(profile), "/;,", &p);
710 lower_transport[0] = '\0';
711 /* rtp/avp/<protocol> */
713 get_word_sep(lower_transport, sizeof(lower_transport),
716 th->transport = RTSP_TRANSPORT_RTP;
717 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
718 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
719 /* x-pn-tng/<protocol> */
720 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
722 th->transport = RTSP_TRANSPORT_RDT;
723 } else if (!av_strcasecmp(transport_protocol, "raw")) {
724 get_word_sep(profile, sizeof(profile), "/;,", &p);
725 lower_transport[0] = '\0';
726 /* raw/raw/<protocol> */
728 get_word_sep(lower_transport, sizeof(lower_transport),
731 th->transport = RTSP_TRANSPORT_RAW;
733 if (!av_strcasecmp(lower_transport, "TCP"))
734 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
736 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
740 /* get each parameter */
741 while (*p != '\0' && *p != ',') {
742 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
743 if (!strcmp(parameter, "port")) {
746 rtsp_parse_range(&th->port_min, &th->port_max, &p);
748 } else if (!strcmp(parameter, "client_port")) {
751 rtsp_parse_range(&th->client_port_min,
752 &th->client_port_max, &p);
754 } else if (!strcmp(parameter, "server_port")) {
757 rtsp_parse_range(&th->server_port_min,
758 &th->server_port_max, &p);
760 } else if (!strcmp(parameter, "interleaved")) {
763 rtsp_parse_range(&th->interleaved_min,
764 &th->interleaved_max, &p);
766 } else if (!strcmp(parameter, "multicast")) {
767 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
768 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
769 } else if (!strcmp(parameter, "ttl")) {
773 th->ttl = strtol(p, &end, 10);
776 } else if (!strcmp(parameter, "destination")) {
779 get_word_sep(buf, sizeof(buf), ";,", &p);
780 get_sockaddr(buf, &th->destination);
782 } else if (!strcmp(parameter, "source")) {
785 get_word_sep(buf, sizeof(buf), ";,", &p);
786 av_strlcpy(th->source, buf, sizeof(th->source));
788 } else if (!strcmp(parameter, "mode")) {
791 get_word_sep(buf, sizeof(buf), ";, ", &p);
792 if (!strcmp(buf, "record") ||
793 !strcmp(buf, "receive"))
798 while (*p != ';' && *p != '\0' && *p != ',')
806 reply->nb_transports++;
810 static void handle_rtp_info(RTSPState *rt, const char *url,
811 uint32_t seq, uint32_t rtptime)
814 if (!rtptime || !url[0])
816 if (rt->transport != RTSP_TRANSPORT_RTP)
818 for (i = 0; i < rt->nb_rtsp_streams; i++) {
819 RTSPStream *rtsp_st = rt->rtsp_streams[i];
820 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
823 if (!strcmp(rtsp_st->control_url, url)) {
824 rtpctx->base_timestamp = rtptime;
830 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
833 char key[20], value[1024], url[1024] = "";
834 uint32_t seq = 0, rtptime = 0;
837 p += strspn(p, SPACE_CHARS);
840 get_word_sep(key, sizeof(key), "=", &p);
844 get_word_sep(value, sizeof(value), ";, ", &p);
846 if (!strcmp(key, "url"))
847 av_strlcpy(url, value, sizeof(url));
848 else if (!strcmp(key, "seq"))
849 seq = strtoul(value, NULL, 10);
850 else if (!strcmp(key, "rtptime"))
851 rtptime = strtoul(value, NULL, 10);
853 handle_rtp_info(rt, url, seq, rtptime);
862 handle_rtp_info(rt, url, seq, rtptime);
865 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
866 RTSPState *rt, const char *method)
870 /* NOTE: we do case independent match for broken servers */
872 if (av_stristart(p, "Session:", &p)) {
874 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
875 if (av_stristart(p, ";timeout=", &p) &&
876 (t = strtol(p, NULL, 10)) > 0) {
879 } else if (av_stristart(p, "Content-Length:", &p)) {
880 reply->content_length = strtol(p, NULL, 10);
881 } else if (av_stristart(p, "Transport:", &p)) {
882 rtsp_parse_transport(reply, p);
883 } else if (av_stristart(p, "CSeq:", &p)) {
884 reply->seq = strtol(p, NULL, 10);
885 } else if (av_stristart(p, "Range:", &p)) {
886 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
887 } else if (av_stristart(p, "RealChallenge1:", &p)) {
888 p += strspn(p, SPACE_CHARS);
889 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
890 } else if (av_stristart(p, "Server:", &p)) {
891 p += strspn(p, SPACE_CHARS);
892 av_strlcpy(reply->server, p, sizeof(reply->server));
893 } else if (av_stristart(p, "Notice:", &p) ||
894 av_stristart(p, "X-Notice:", &p)) {
895 reply->notice = strtol(p, NULL, 10);
896 } else if (av_stristart(p, "Location:", &p)) {
897 p += strspn(p, SPACE_CHARS);
898 av_strlcpy(reply->location, p , sizeof(reply->location));
899 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
900 p += strspn(p, SPACE_CHARS);
901 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
902 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
903 p += strspn(p, SPACE_CHARS);
904 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
905 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
906 p += strspn(p, SPACE_CHARS);
907 if (method && !strcmp(method, "DESCRIBE"))
908 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
909 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
910 p += strspn(p, SPACE_CHARS);
911 if (method && !strcmp(method, "PLAY"))
912 rtsp_parse_rtp_info(rt, p);
913 } else if (av_stristart(p, "Public:", &p) && rt) {
914 if (strstr(p, "GET_PARAMETER") &&
915 method && !strcmp(method, "OPTIONS"))
916 rt->get_parameter_supported = 1;
917 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
918 p += strspn(p, SPACE_CHARS);
919 rt->accept_dynamic_rate = atoi(p);
920 } else if (av_stristart(p, "Content-Type:", &p)) {
921 p += strspn(p, SPACE_CHARS);
922 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
926 /* skip a RTP/TCP interleaved packet */
927 void ff_rtsp_skip_packet(AVFormatContext *s)
929 RTSPState *rt = s->priv_data;
933 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
936 len = AV_RB16(buf + 1);
938 av_dlog(s, "skipping RTP packet len=%d\n", len);
943 if (len1 > sizeof(buf))
945 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
952 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
953 unsigned char **content_ptr,
954 int return_on_interleaved_data, const char *method)
956 RTSPState *rt = s->priv_data;
957 char buf[4096], buf1[1024], *q;
960 int ret, content_length, line_count = 0, request = 0;
961 unsigned char *content = NULL;
967 memset(reply, 0, sizeof(*reply));
969 /* parse reply (XXX: use buffers) */
970 rt->last_reply[0] = '\0';
974 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
975 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
981 /* XXX: only parse it if first char on line ? */
982 if (return_on_interleaved_data) {
985 ff_rtsp_skip_packet(s);
986 } else if (ch != '\r') {
987 if ((q - buf) < sizeof(buf) - 1)
993 av_dlog(s, "line='%s'\n", buf);
995 /* test if last line */
999 if (line_count == 0) {
1000 /* get reply code */
1001 get_word(buf1, sizeof(buf1), &p);
1002 if (!strncmp(buf1, "RTSP/", 5)) {
1003 get_word(buf1, sizeof(buf1), &p);
1004 reply->status_code = atoi(buf1);
1005 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1007 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1008 get_word(buf1, sizeof(buf1), &p); // object
1012 ff_rtsp_parse_line(reply, p, rt, method);
1013 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1014 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1019 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1020 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1022 content_length = reply->content_length;
1023 if (content_length > 0) {
1024 /* leave some room for a trailing '\0' (useful for simple parsing) */
1025 content = av_malloc(content_length + 1);
1026 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1027 content[content_length] = '\0';
1030 *content_ptr = content;
1036 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1037 const char* ptr = buf;
1039 if (!strcmp(reply->reason, "OPTIONS")) {
1040 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1042 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1043 if (reply->session_id[0])
1044 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1047 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1049 av_strlcat(buf, "\r\n", sizeof(buf));
1051 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1052 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1055 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1057 rt->last_cmd_time = av_gettime();
1058 /* Even if the request from the server had data, it is not the data
1059 * that the caller wants or expects. The memory could also be leaked
1060 * if the actual following reply has content data. */
1062 av_freep(content_ptr);
1063 /* If method is set, this is called from ff_rtsp_send_cmd,
1064 * where a reply to exactly this request is awaited. For
1065 * callers from within packet receiving, we just want to
1066 * return to the caller and go back to receiving packets. */
1072 if (rt->seq != reply->seq) {
1073 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1074 rt->seq, reply->seq);
1078 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1079 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1080 reply->notice == 2306 /* Continuous Feed Terminated */) {
1081 rt->state = RTSP_STATE_IDLE;
1082 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1083 return AVERROR(EIO); /* data or server error */
1084 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1085 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1086 return AVERROR(EPERM);
1092 * Send a command to the RTSP server without waiting for the reply.
1094 * @param s RTSP (de)muxer context
1095 * @param method the method for the request
1096 * @param url the target url for the request
1097 * @param headers extra header lines to include in the request
1098 * @param send_content if non-null, the data to send as request body content
1099 * @param send_content_length the length of the send_content data, or 0 if
1100 * send_content is null
1102 * @return zero if success, nonzero otherwise
1104 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1105 const char *method, const char *url,
1106 const char *headers,
1107 const unsigned char *send_content,
1108 int send_content_length)
1110 RTSPState *rt = s->priv_data;
1111 char buf[4096], *out_buf;
1112 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1114 /* Add in RTSP headers */
1117 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1119 av_strlcat(buf, headers, sizeof(buf));
1120 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1121 if (rt->session_id[0] != '\0' && (!headers ||
1122 !strstr(headers, "\nIf-Match:"))) {
1123 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1126 char *str = ff_http_auth_create_response(&rt->auth_state,
1127 rt->auth, url, method);
1129 av_strlcat(buf, str, sizeof(buf));
1132 if (send_content_length > 0 && send_content)
1133 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1134 av_strlcat(buf, "\r\n", sizeof(buf));
1136 /* base64 encode rtsp if tunneling */
1137 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1138 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1139 out_buf = base64buf;
1142 av_dlog(s, "Sending:\n%s--\n", buf);
1144 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1145 if (send_content_length > 0 && send_content) {
1146 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1147 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1148 "with content data not supported\n");
1149 return AVERROR_PATCHWELCOME;
1151 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1153 rt->last_cmd_time = av_gettime();
1158 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1159 const char *url, const char *headers)
1161 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1164 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1165 const char *headers, RTSPMessageHeader *reply,
1166 unsigned char **content_ptr)
1168 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1169 content_ptr, NULL, 0);
1172 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1173 const char *method, const char *url,
1175 RTSPMessageHeader *reply,
1176 unsigned char **content_ptr,
1177 const unsigned char *send_content,
1178 int send_content_length)
1180 RTSPState *rt = s->priv_data;
1181 HTTPAuthType cur_auth_type;
1182 int ret, attempts = 0;
1185 cur_auth_type = rt->auth_state.auth_type;
1186 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1188 send_content_length)))
1191 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1195 if (reply->status_code == 401 &&
1196 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1197 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1200 if (reply->status_code > 400){
1201 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1205 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1211 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1212 int lower_transport, const char *real_challenge)
1214 RTSPState *rt = s->priv_data;
1215 int rtx = 0, j, i, err, interleave = 0, port_off;
1216 RTSPStream *rtsp_st;
1217 RTSPMessageHeader reply1, *reply = &reply1;
1219 const char *trans_pref;
1221 if (rt->transport == RTSP_TRANSPORT_RDT)
1222 trans_pref = "x-pn-tng";
1223 else if (rt->transport == RTSP_TRANSPORT_RAW)
1224 trans_pref = "RAW/RAW";
1226 trans_pref = "RTP/AVP";
1228 /* default timeout: 1 minute */
1231 /* Choose a random starting offset within the first half of the
1232 * port range, to allow for a number of ports to try even if the offset
1233 * happens to be at the end of the random range. */
1234 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1235 /* even random offset */
1236 port_off -= port_off & 0x01;
1238 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1239 char transport[2048];
1242 * WMS serves all UDP data over a single connection, the RTX, which
1243 * isn't necessarily the first in the SDP but has to be the first
1244 * to be set up, else the second/third SETUP will fail with a 461.
1246 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1247 rt->server_type == RTSP_SERVER_WMS) {
1250 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1251 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1253 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1257 if (rtx == rt->nb_rtsp_streams)
1258 return -1; /* no RTX found */
1259 rtsp_st = rt->rtsp_streams[rtx];
1261 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1263 rtsp_st = rt->rtsp_streams[i];
1266 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1269 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1270 port = reply->transports[0].client_port_min;
1274 /* first try in specified port range */
1275 while (j <= rt->rtp_port_max) {
1276 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1277 "?localport=%d", j);
1278 /* we will use two ports per rtp stream (rtp and rtcp) */
1280 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1281 &s->interrupt_callback, NULL))
1284 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1289 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1291 snprintf(transport, sizeof(transport) - 1,
1292 "%s/UDP;", trans_pref);
1293 if (rt->server_type != RTSP_SERVER_REAL)
1294 av_strlcat(transport, "unicast;", sizeof(transport));
1295 av_strlcatf(transport, sizeof(transport),
1296 "client_port=%d", port);
1297 if (rt->transport == RTSP_TRANSPORT_RTP &&
1298 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1299 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1303 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1304 /* For WMS streams, the application streams are only used for
1305 * UDP. When trying to set it up for TCP streams, the server
1306 * will return an error. Therefore, we skip those streams. */
1307 if (rt->server_type == RTSP_SERVER_WMS &&
1308 (rtsp_st->stream_index < 0 ||
1309 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1312 snprintf(transport, sizeof(transport) - 1,
1313 "%s/TCP;", trans_pref);
1314 if (rt->transport != RTSP_TRANSPORT_RDT)
1315 av_strlcat(transport, "unicast;", sizeof(transport));
1316 av_strlcatf(transport, sizeof(transport),
1317 "interleaved=%d-%d",
1318 interleave, interleave + 1);
1322 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1323 snprintf(transport, sizeof(transport) - 1,
1324 "%s/UDP;multicast", trans_pref);
1327 av_strlcat(transport, ";mode=record", sizeof(transport));
1328 } else if (rt->server_type == RTSP_SERVER_REAL ||
1329 rt->server_type == RTSP_SERVER_WMS)
1330 av_strlcat(transport, ";mode=play", sizeof(transport));
1331 snprintf(cmd, sizeof(cmd),
1332 "Transport: %s\r\n",
1334 if (rt->accept_dynamic_rate)
1335 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1336 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1337 char real_res[41], real_csum[9];
1338 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1340 av_strlcatf(cmd, sizeof(cmd),
1342 "RealChallenge2: %s, sd=%s\r\n",
1343 rt->session_id, real_res, real_csum);
1345 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1346 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1349 } else if (reply->status_code != RTSP_STATUS_OK ||
1350 reply->nb_transports != 1) {
1351 err = AVERROR_INVALIDDATA;
1355 /* XXX: same protocol for all streams is required */
1357 if (reply->transports[0].lower_transport != rt->lower_transport ||
1358 reply->transports[0].transport != rt->transport) {
1359 err = AVERROR_INVALIDDATA;
1363 rt->lower_transport = reply->transports[0].lower_transport;
1364 rt->transport = reply->transports[0].transport;
1367 /* Fail if the server responded with another lower transport mode
1368 * than what we requested. */
1369 if (reply->transports[0].lower_transport != lower_transport) {
1370 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1371 err = AVERROR_INVALIDDATA;
1375 switch(reply->transports[0].lower_transport) {
1376 case RTSP_LOWER_TRANSPORT_TCP:
1377 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1378 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1381 case RTSP_LOWER_TRANSPORT_UDP: {
1382 char url[1024], options[30] = "";
1384 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1385 av_strlcpy(options, "?connect=1", sizeof(options));
1386 /* Use source address if specified */
1387 if (reply->transports[0].source[0]) {
1388 ff_url_join(url, sizeof(url), "rtp", NULL,
1389 reply->transports[0].source,
1390 reply->transports[0].server_port_min, "%s", options);
1392 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1393 reply->transports[0].server_port_min, "%s", options);
1395 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1396 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1397 err = AVERROR_INVALIDDATA;
1400 /* Try to initialize the connection state in a
1401 * potential NAT router by sending dummy packets.
1402 * RTP/RTCP dummy packets are used for RDT, too.
1404 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1406 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1409 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1410 char url[1024], namebuf[50], optbuf[20] = "";
1411 struct sockaddr_storage addr;
1414 if (reply->transports[0].destination.ss_family) {
1415 addr = reply->transports[0].destination;
1416 port = reply->transports[0].port_min;
1417 ttl = reply->transports[0].ttl;
1419 addr = rtsp_st->sdp_ip;
1420 port = rtsp_st->sdp_port;
1421 ttl = rtsp_st->sdp_ttl;
1424 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1425 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1426 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1427 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1428 port, "%s", optbuf);
1429 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1430 &s->interrupt_callback, NULL) < 0) {
1431 err = AVERROR_INVALIDDATA;
1438 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1442 if (rt->nb_rtsp_streams && reply->timeout > 0)
1443 rt->timeout = reply->timeout;
1445 if (rt->server_type == RTSP_SERVER_REAL)
1446 rt->need_subscription = 1;
1451 ff_rtsp_undo_setup(s);
1455 void ff_rtsp_close_connections(AVFormatContext *s)
1457 RTSPState *rt = s->priv_data;
1458 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1459 ffurl_close(rt->rtsp_hd);
1460 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1463 int ff_rtsp_connect(AVFormatContext *s)
1465 RTSPState *rt = s->priv_data;
1466 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1467 int port, err, tcp_fd;
1468 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1469 int lower_transport_mask = 0;
1470 char real_challenge[64] = "";
1471 struct sockaddr_storage peer;
1472 socklen_t peer_len = sizeof(peer);
1474 if (rt->rtp_port_max < rt->rtp_port_min) {
1475 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1476 "than min port %d\n", rt->rtp_port_max,
1478 return AVERROR(EINVAL);
1481 if (!ff_network_init())
1482 return AVERROR(EIO);
1484 if (s->max_delay < 0) /* Not set by the caller */
1485 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1487 rt->control_transport = RTSP_MODE_PLAIN;
1488 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1489 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1490 rt->control_transport = RTSP_MODE_TUNNEL;
1492 /* Only pass through valid flags from here */
1493 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1496 lower_transport_mask = rt->lower_transport_mask;
1497 /* extract hostname and port */
1498 av_url_split(NULL, 0, auth, sizeof(auth),
1499 host, sizeof(host), &port, path, sizeof(path), s->filename);
1501 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1504 port = RTSP_DEFAULT_PORT;
1506 if (!lower_transport_mask)
1507 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1510 /* Only UDP or TCP - UDP multicast isn't supported. */
1511 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1512 (1 << RTSP_LOWER_TRANSPORT_TCP);
1513 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1514 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1515 "only UDP and TCP are supported for output.\n");
1516 err = AVERROR(EINVAL);
1521 /* Construct the URI used in request; this is similar to s->filename,
1522 * but with authentication credentials removed and RTSP specific options
1524 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1525 host, port, "%s", path);
1527 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1528 /* set up initial handshake for tunneling */
1529 char httpname[1024];
1530 char sessioncookie[17];
1533 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1534 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1535 av_get_random_seed(), av_get_random_seed());
1538 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1539 &s->interrupt_callback) < 0) {
1544 /* generate GET headers */
1545 snprintf(headers, sizeof(headers),
1546 "x-sessioncookie: %s\r\n"
1547 "Accept: application/x-rtsp-tunnelled\r\n"
1548 "Pragma: no-cache\r\n"
1549 "Cache-Control: no-cache\r\n",
1551 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1553 /* complete the connection */
1554 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1560 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1561 &s->interrupt_callback) < 0 ) {
1566 /* generate POST headers */
1567 snprintf(headers, sizeof(headers),
1568 "x-sessioncookie: %s\r\n"
1569 "Content-Type: application/x-rtsp-tunnelled\r\n"
1570 "Pragma: no-cache\r\n"
1571 "Cache-Control: no-cache\r\n"
1572 "Content-Length: 32767\r\n"
1573 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1575 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1576 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1578 /* Initialize the authentication state for the POST session. The HTTP
1579 * protocol implementation doesn't properly handle multi-pass
1580 * authentication for POST requests, since it would require one of
1582 * - implementing Expect: 100-continue, which many HTTP servers
1583 * don't support anyway, even less the RTSP servers that do HTTP
1585 * - sending the whole POST data until getting a 401 reply specifying
1586 * what authentication method to use, then resending all that data
1587 * - waiting for potential 401 replies directly after sending the
1588 * POST header (waiting for some unspecified time)
1589 * Therefore, we copy the full auth state, which works for both basic
1590 * and digest. (For digest, we would have to synchronize the nonce
1591 * count variable between the two sessions, if we'd do more requests
1592 * with the original session, though.)
1594 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1596 /* complete the connection */
1597 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1602 /* open the tcp connection */
1603 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1604 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1605 &s->interrupt_callback, NULL) < 0) {
1609 rt->rtsp_hd_out = rt->rtsp_hd;
1613 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1614 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1615 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1616 NULL, 0, NI_NUMERICHOST);
1619 /* request options supported by the server; this also detects server
1621 for (rt->server_type = RTSP_SERVER_RTP;;) {
1623 if (rt->server_type == RTSP_SERVER_REAL)
1626 * The following entries are required for proper
1627 * streaming from a Realmedia server. They are
1628 * interdependent in some way although we currently
1629 * don't quite understand how. Values were copied
1630 * from mplayer SVN r23589.
1631 * ClientChallenge is a 16-byte ID in hex
1632 * CompanyID is a 16-byte ID in base64
1634 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1635 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1636 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1637 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1639 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1640 if (reply->status_code != RTSP_STATUS_OK) {
1641 err = AVERROR_INVALIDDATA;
1645 /* detect server type if not standard-compliant RTP */
1646 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1647 rt->server_type = RTSP_SERVER_REAL;
1649 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1650 rt->server_type = RTSP_SERVER_WMS;
1651 } else if (rt->server_type == RTSP_SERVER_REAL)
1652 strcpy(real_challenge, reply->real_challenge);
1656 if (s->iformat && CONFIG_RTSP_DEMUXER)
1657 err = ff_rtsp_setup_input_streams(s, reply);
1658 else if (CONFIG_RTSP_MUXER)
1659 err = ff_rtsp_setup_output_streams(s, host);
1664 int lower_transport = ff_log2_tab[lower_transport_mask &
1665 ~(lower_transport_mask - 1)];
1667 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1668 rt->server_type == RTSP_SERVER_REAL ?
1669 real_challenge : NULL);
1672 lower_transport_mask &= ~(1 << lower_transport);
1673 if (lower_transport_mask == 0 && err == 1) {
1674 err = AVERROR(EPROTONOSUPPORT);
1679 rt->lower_transport_mask = lower_transport_mask;
1680 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1681 rt->state = RTSP_STATE_IDLE;
1682 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1685 ff_rtsp_close_streams(s);
1686 ff_rtsp_close_connections(s);
1687 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1688 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1689 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1697 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1700 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1701 uint8_t *buf, int buf_size, int64_t wait_end)
1703 RTSPState *rt = s->priv_data;
1704 RTSPStream *rtsp_st;
1705 int n, i, ret, tcp_fd, timeout_cnt = 0;
1707 struct pollfd *p = rt->p;
1708 int *fds = NULL, fdsnum, fdsidx;
1711 if (ff_check_interrupt(&s->interrupt_callback))
1712 return AVERROR_EXIT;
1713 if (wait_end && wait_end - av_gettime() < 0)
1714 return AVERROR(EAGAIN);
1717 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1718 p[max_p].fd = tcp_fd;
1719 p[max_p++].events = POLLIN;
1723 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1724 rtsp_st = rt->rtsp_streams[i];
1725 if (rtsp_st->rtp_handle) {
1726 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1728 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1732 av_log(s, AV_LOG_ERROR,
1733 "Number of fds %d not supported\n", fdsnum);
1734 return AVERROR_INVALIDDATA;
1736 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1737 p[max_p].fd = fds[fdsidx];
1738 p[max_p++].events = POLLIN;
1743 n = poll(p, max_p, POLL_TIMEOUT_MS);
1745 int j = 1 - (tcp_fd == -1);
1747 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1748 rtsp_st = rt->rtsp_streams[i];
1749 if (rtsp_st->rtp_handle) {
1750 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1751 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1753 *prtsp_st = rtsp_st;
1760 #if CONFIG_RTSP_DEMUXER
1761 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1762 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1763 if (rt->state == RTSP_STATE_STREAMING) {
1764 if (!ff_rtsp_parse_streaming_commands(s))
1767 av_log(s, AV_LOG_WARNING,
1768 "Unable to answer to TEARDOWN\n");
1772 RTSPMessageHeader reply;
1773 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1776 /* XXX: parse message */
1777 if (rt->state != RTSP_STATE_STREAMING)
1782 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1783 return AVERROR(ETIMEDOUT);
1784 } else if (n < 0 && errno != EINTR)
1785 return AVERROR(errno);
1789 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1790 const uint8_t *buf, int len)
1792 RTSPState *rt = s->priv_data;
1796 if (rt->nb_rtsp_streams == 1) {
1797 *rtsp_st = rt->rtsp_streams[0];
1800 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1801 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1803 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1804 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1807 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1808 *rtsp_st = rt->rtsp_streams[i];
1815 av_log(s, AV_LOG_WARNING,
1816 "Unable to pick stream for packet - SSRC not known for "
1818 return AVERROR(EAGAIN);
1821 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1822 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1823 *rtsp_st = rt->rtsp_streams[i];
1829 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1830 return AVERROR(EAGAIN);
1833 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1835 RTSPState *rt = s->priv_data;
1837 RTSPStream *rtsp_st, *first_queue_st = NULL;
1838 int64_t wait_end = 0;
1840 if (rt->nb_byes == rt->nb_rtsp_streams)
1843 /* get next frames from the same RTP packet */
1844 if (rt->cur_transport_priv) {
1845 if (rt->transport == RTSP_TRANSPORT_RDT) {
1846 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1847 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1848 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1849 } else if (rt->ts && CONFIG_RTPDEC) {
1850 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1852 rt->recvbuf_pos += ret;
1853 ret = rt->recvbuf_pos < rt->recvbuf_len;
1858 rt->cur_transport_priv = NULL;
1860 } else if (ret == 1) {
1863 rt->cur_transport_priv = NULL;
1866 if (rt->transport == RTSP_TRANSPORT_RTP) {
1868 int64_t first_queue_time = 0;
1869 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1870 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1874 queue_time = ff_rtp_queued_packet_time(rtpctx);
1875 if (queue_time && (queue_time - first_queue_time < 0 ||
1876 !first_queue_time)) {
1877 first_queue_time = queue_time;
1878 first_queue_st = rt->rtsp_streams[i];
1881 if (first_queue_time)
1882 wait_end = first_queue_time + s->max_delay;
1885 /* read next RTP packet */
1888 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1890 return AVERROR(ENOMEM);
1893 switch(rt->lower_transport) {
1895 #if CONFIG_RTSP_DEMUXER
1896 case RTSP_LOWER_TRANSPORT_TCP:
1897 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1900 case RTSP_LOWER_TRANSPORT_UDP:
1901 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1902 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1903 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1904 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1906 case RTSP_LOWER_TRANSPORT_CUSTOM:
1907 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1908 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1909 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1910 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1913 if (len == AVERROR(EAGAIN) && first_queue_st &&
1914 rt->transport == RTSP_TRANSPORT_RTP) {
1915 rtsp_st = first_queue_st;
1916 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1923 if (rt->transport == RTSP_TRANSPORT_RDT) {
1924 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1925 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1926 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1928 /* Either bad packet, or a RTCP packet. Check if the
1929 * first_rtcp_ntp_time field was initialized. */
1930 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1931 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1932 /* first_rtcp_ntp_time has been initialized for this stream,
1933 * copy the same value to all other uninitialized streams,
1934 * in order to map their timestamp origin to the same ntp time
1937 AVStream *st = NULL;
1938 if (rtsp_st->stream_index >= 0)
1939 st = s->streams[rtsp_st->stream_index];
1940 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1941 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1942 AVStream *st2 = NULL;
1943 if (rt->rtsp_streams[i]->stream_index >= 0)
1944 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1945 if (rtpctx2 && st && st2 &&
1946 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1947 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1948 rtpctx2->rtcp_ts_offset = av_rescale_q(
1949 rtpctx->rtcp_ts_offset, st->time_base,
1954 if (ret == -RTCP_BYE) {
1957 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1958 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1960 if (rt->nb_byes == rt->nb_rtsp_streams)
1964 } else if (rt->ts && CONFIG_RTPDEC) {
1965 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1968 rt->recvbuf_len = len;
1969 rt->recvbuf_pos = ret;
1970 rt->cur_transport_priv = rt->ts;
1977 return AVERROR_INVALIDDATA;
1983 /* more packets may follow, so we save the RTP context */
1984 rt->cur_transport_priv = rtsp_st->transport_priv;
1988 #endif /* CONFIG_RTPDEC */
1990 #if CONFIG_SDP_DEMUXER
1991 static int sdp_probe(AVProbeData *p1)
1993 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1995 /* we look for a line beginning "c=IN IP" */
1996 while (p < p_end && *p != '\0') {
1997 if (p + sizeof("c=IN IP") - 1 < p_end &&
1998 av_strstart(p, "c=IN IP", NULL))
1999 return AVPROBE_SCORE_MAX / 2;
2001 while (p < p_end - 1 && *p != '\n') p++;
2010 static int sdp_read_header(AVFormatContext *s)
2012 RTSPState *rt = s->priv_data;
2013 RTSPStream *rtsp_st;
2018 if (!ff_network_init())
2019 return AVERROR(EIO);
2021 if (s->max_delay < 0) /* Not set by the caller */
2022 s->max_delay = DEFAULT_REORDERING_DELAY;
2023 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2024 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2026 /* read the whole sdp file */
2027 /* XXX: better loading */
2028 content = av_malloc(SDP_MAX_SIZE);
2029 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2032 return AVERROR_INVALIDDATA;
2034 content[size] ='\0';
2036 err = ff_sdp_parse(s, content);
2040 /* open each RTP stream */
2041 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2043 rtsp_st = rt->rtsp_streams[i];
2045 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2046 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2047 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2048 ff_url_join(url, sizeof(url), "rtp", NULL,
2049 namebuf, rtsp_st->sdp_port,
2050 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2052 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2053 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2054 &s->interrupt_callback, NULL) < 0) {
2055 err = AVERROR_INVALIDDATA;
2059 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2064 ff_rtsp_close_streams(s);
2069 static int sdp_read_close(AVFormatContext *s)
2071 ff_rtsp_close_streams(s);
2076 static const AVClass sdp_demuxer_class = {
2077 .class_name = "SDP demuxer",
2078 .item_name = av_default_item_name,
2079 .option = sdp_options,
2080 .version = LIBAVUTIL_VERSION_INT,
2083 AVInputFormat ff_sdp_demuxer = {
2085 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2086 .priv_data_size = sizeof(RTSPState),
2087 .read_probe = sdp_probe,
2088 .read_header = sdp_read_header,
2089 .read_packet = ff_rtsp_fetch_packet,
2090 .read_close = sdp_read_close,
2091 .priv_class = &sdp_demuxer_class,
2093 #endif /* CONFIG_SDP_DEMUXER */
2095 #if CONFIG_RTP_DEMUXER
2096 static int rtp_probe(AVProbeData *p)
2098 if (av_strstart(p->filename, "rtp:", NULL))
2099 return AVPROBE_SCORE_MAX;
2103 static int rtp_read_header(AVFormatContext *s)
2105 uint8_t recvbuf[1500];
2106 char host[500], sdp[500];
2108 URLContext* in = NULL;
2110 AVCodecContext codec = { 0 };
2111 struct sockaddr_storage addr;
2113 socklen_t addrlen = sizeof(addr);
2114 RTSPState *rt = s->priv_data;
2116 if (!ff_network_init())
2117 return AVERROR(EIO);
2119 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2120 &s->interrupt_callback, NULL);
2125 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2126 if (ret == AVERROR(EAGAIN))
2131 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2135 if ((recvbuf[0] & 0xc0) != 0x80) {
2136 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2141 if (RTP_PT_IS_RTCP(recvbuf[1]))
2144 payload_type = recvbuf[1] & 0x7f;
2147 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2151 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2152 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2153 "without an SDP file describing it\n",
2157 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2158 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2159 "properly you need an SDP file "
2163 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2164 NULL, 0, s->filename);
2166 snprintf(sdp, sizeof(sdp),
2167 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2168 addr.ss_family == AF_INET ? 4 : 6, host,
2169 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2170 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2171 port, payload_type);
2172 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2174 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2177 /* sdp_read_header initializes this again */
2180 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2182 ret = sdp_read_header(s);
2193 static const AVClass rtp_demuxer_class = {
2194 .class_name = "RTP demuxer",
2195 .item_name = av_default_item_name,
2196 .option = rtp_options,
2197 .version = LIBAVUTIL_VERSION_INT,
2200 AVInputFormat ff_rtp_demuxer = {
2202 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2203 .priv_data_size = sizeof(RTSPState),
2204 .read_probe = rtp_probe,
2205 .read_header = rtp_read_header,
2206 .read_packet = ff_rtsp_fetch_packet,
2207 .read_close = sdp_read_close,
2208 .flags = AVFMT_NOFILE,
2209 .priv_class = &rtp_demuxer_class,
2211 #endif /* CONFIG_RTP_DEMUXER */