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;
383 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
384 rtsp_st->feedback = 1;
386 /* XXX: handle list of formats */
387 get_word(buf1, sizeof(buf1), &p); /* format list */
388 rtsp_st->sdp_payload_type = atoi(buf1);
390 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
391 /* no corresponding stream */
392 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
393 rt->ts = ff_mpegts_parse_open(s);
394 } else if (rt->server_type == RTSP_SERVER_WMS &&
395 codec_type == AVMEDIA_TYPE_DATA) {
396 /* RTX stream, a stream that carries all the other actual
397 * audio/video streams. Don't expose this to the callers. */
399 st = avformat_new_stream(s, NULL);
402 st->id = rt->nb_rtsp_streams - 1;
403 rtsp_st->stream_index = st->index;
404 st->codec->codec_type = codec_type;
405 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
406 RTPDynamicProtocolHandler *handler;
407 /* if standard payload type, we can find the codec right now */
408 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
409 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
410 st->codec->sample_rate > 0)
411 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
412 /* Even static payload types may need a custom depacketizer */
413 handler = ff_rtp_handler_find_by_id(
414 rtsp_st->sdp_payload_type, st->codec->codec_type);
415 init_rtp_handler(handler, rtsp_st, st->codec);
416 if (handler && handler->init)
417 handler->init(s, st->index,
418 rtsp_st->dynamic_protocol_context);
421 /* put a default control url */
422 av_strlcpy(rtsp_st->control_url, rt->control_uri,
423 sizeof(rtsp_st->control_url));
426 if (av_strstart(p, "control:", &p)) {
427 if (s->nb_streams == 0) {
428 if (!strncmp(p, "rtsp://", 7))
429 av_strlcpy(rt->control_uri, p,
430 sizeof(rt->control_uri));
433 /* get the control url */
434 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
436 /* XXX: may need to add full url resolution */
437 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
439 if (proto[0] == '\0') {
440 /* relative control URL */
441 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
442 av_strlcat(rtsp_st->control_url, "/",
443 sizeof(rtsp_st->control_url));
444 av_strlcat(rtsp_st->control_url, p,
445 sizeof(rtsp_st->control_url));
447 av_strlcpy(rtsp_st->control_url, p,
448 sizeof(rtsp_st->control_url));
450 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
451 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
452 get_word(buf1, sizeof(buf1), &p);
453 payload_type = atoi(buf1);
454 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
455 if (rtsp_st->stream_index >= 0) {
456 st = s->streams[rtsp_st->stream_index];
457 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
459 } else if (av_strstart(p, "fmtp:", &p) ||
460 av_strstart(p, "framesize:", &p)) {
461 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
462 // let dynamic protocol handlers have a stab at the line.
463 get_word(buf1, sizeof(buf1), &p);
464 payload_type = atoi(buf1);
465 for (i = 0; i < rt->nb_rtsp_streams; i++) {
466 rtsp_st = rt->rtsp_streams[i];
467 if (rtsp_st->sdp_payload_type == payload_type &&
468 rtsp_st->dynamic_handler &&
469 rtsp_st->dynamic_handler->parse_sdp_a_line)
470 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
471 rtsp_st->dynamic_protocol_context, buf);
473 } else if (av_strstart(p, "range:", &p)) {
476 // this is so that seeking on a streamed file can work.
477 rtsp_parse_range_npt(p, &start, &end);
478 s->start_time = start;
479 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
480 s->duration = (end == AV_NOPTS_VALUE) ?
481 AV_NOPTS_VALUE : end - start;
482 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
484 rt->transport = RTSP_TRANSPORT_RDT;
485 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
487 st = s->streams[s->nb_streams - 1];
488 st->codec->sample_rate = atoi(p);
490 if (rt->server_type == RTSP_SERVER_WMS)
491 ff_wms_parse_sdp_a_line(s, p);
492 if (s->nb_streams > 0) {
493 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
495 if (rt->server_type == RTSP_SERVER_REAL)
496 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
498 if (rtsp_st->dynamic_handler &&
499 rtsp_st->dynamic_handler->parse_sdp_a_line)
500 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
501 rtsp_st->stream_index,
502 rtsp_st->dynamic_protocol_context, buf);
509 int ff_sdp_parse(AVFormatContext *s, const char *content)
511 RTSPState *rt = s->priv_data;
514 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
515 * contain long SDP lines containing complete ASF Headers (several
516 * kB) or arrays of MDPR (RM stream descriptor) headers plus
517 * "rulebooks" describing their properties. Therefore, the SDP line
520 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
521 * in rtpdec_xiph.c. */
523 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
527 p += strspn(p, SPACE_CHARS);
535 /* get the content */
537 while (*p != '\n' && *p != '\r' && *p != '\0') {
538 if ((q - buf) < sizeof(buf) - 1)
543 sdp_parse_line(s, s1, letter, buf);
545 while (*p != '\n' && *p != '\0')
550 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
551 if (!rt->p) return AVERROR(ENOMEM);
554 #endif /* CONFIG_RTPDEC */
556 void ff_rtsp_undo_setup(AVFormatContext *s)
558 RTSPState *rt = s->priv_data;
561 for (i = 0; i < rt->nb_rtsp_streams; i++) {
562 RTSPStream *rtsp_st = rt->rtsp_streams[i];
565 if (rtsp_st->transport_priv) {
567 AVFormatContext *rtpctx = rtsp_st->transport_priv;
568 av_write_trailer(rtpctx);
569 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
571 avio_close_dyn_buf(rtpctx->pb, &ptr);
574 avio_close(rtpctx->pb);
576 avformat_free_context(rtpctx);
577 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
578 ff_rdt_parse_close(rtsp_st->transport_priv);
579 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
580 ff_rtp_parse_close(rtsp_st->transport_priv);
582 rtsp_st->transport_priv = NULL;
583 if (rtsp_st->rtp_handle)
584 ffurl_close(rtsp_st->rtp_handle);
585 rtsp_st->rtp_handle = NULL;
589 /* close and free RTSP streams */
590 void ff_rtsp_close_streams(AVFormatContext *s)
592 RTSPState *rt = s->priv_data;
596 ff_rtsp_undo_setup(s);
597 for (i = 0; i < rt->nb_rtsp_streams; i++) {
598 rtsp_st = rt->rtsp_streams[i];
600 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
601 rtsp_st->dynamic_handler->free(
602 rtsp_st->dynamic_protocol_context);
606 av_free(rt->rtsp_streams);
608 avformat_close_input(&rt->asf_ctx);
610 if (rt->ts && CONFIG_RTPDEC)
611 ff_mpegts_parse_close(rt->ts);
613 av_free(rt->recvbuf);
616 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
618 RTSPState *rt = s->priv_data;
620 int reordering_queue_size = rt->reordering_queue_size;
621 if (reordering_queue_size < 0) {
622 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
623 reordering_queue_size = 0;
625 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
628 /* open the RTP context */
629 if (rtsp_st->stream_index >= 0)
630 st = s->streams[rtsp_st->stream_index];
632 s->ctx_flags |= AVFMTCTX_NOHEADER;
634 if (s->oformat && CONFIG_RTSP_MUXER) {
635 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
637 RTSP_TCP_MAX_PACKET_SIZE,
638 rtsp_st->stream_index);
639 /* Ownership of rtp_handle is passed to the rtp mux context */
640 rtsp_st->rtp_handle = NULL;
643 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
644 return 0; // Don't need to open any parser here
645 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
646 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
647 rtsp_st->dynamic_protocol_context,
648 rtsp_st->dynamic_handler);
649 else if (CONFIG_RTPDEC)
650 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
651 rtsp_st->sdp_payload_type,
652 reordering_queue_size);
654 if (!rtsp_st->transport_priv) {
655 return AVERROR(ENOMEM);
656 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
657 if (rtsp_st->dynamic_handler) {
658 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
659 rtsp_st->dynamic_protocol_context,
660 rtsp_st->dynamic_handler);
667 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
668 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
675 q += strspn(q, SPACE_CHARS);
676 v = strtol(q, &p, 10);
680 v = strtol(p, &p, 10);
689 /* XXX: only one transport specification is parsed */
690 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
692 char transport_protocol[16];
694 char lower_transport[16];
696 RTSPTransportField *th;
699 reply->nb_transports = 0;
702 p += strspn(p, SPACE_CHARS);
706 th = &reply->transports[reply->nb_transports];
708 get_word_sep(transport_protocol, sizeof(transport_protocol),
710 if (!av_strcasecmp (transport_protocol, "rtp")) {
711 get_word_sep(profile, sizeof(profile), "/;,", &p);
712 lower_transport[0] = '\0';
713 /* rtp/avp/<protocol> */
715 get_word_sep(lower_transport, sizeof(lower_transport),
718 th->transport = RTSP_TRANSPORT_RTP;
719 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
720 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
721 /* x-pn-tng/<protocol> */
722 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
724 th->transport = RTSP_TRANSPORT_RDT;
725 } else if (!av_strcasecmp(transport_protocol, "raw")) {
726 get_word_sep(profile, sizeof(profile), "/;,", &p);
727 lower_transport[0] = '\0';
728 /* raw/raw/<protocol> */
730 get_word_sep(lower_transport, sizeof(lower_transport),
733 th->transport = RTSP_TRANSPORT_RAW;
735 if (!av_strcasecmp(lower_transport, "TCP"))
736 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
738 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
742 /* get each parameter */
743 while (*p != '\0' && *p != ',') {
744 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
745 if (!strcmp(parameter, "port")) {
748 rtsp_parse_range(&th->port_min, &th->port_max, &p);
750 } else if (!strcmp(parameter, "client_port")) {
753 rtsp_parse_range(&th->client_port_min,
754 &th->client_port_max, &p);
756 } else if (!strcmp(parameter, "server_port")) {
759 rtsp_parse_range(&th->server_port_min,
760 &th->server_port_max, &p);
762 } else if (!strcmp(parameter, "interleaved")) {
765 rtsp_parse_range(&th->interleaved_min,
766 &th->interleaved_max, &p);
768 } else if (!strcmp(parameter, "multicast")) {
769 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
770 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
771 } else if (!strcmp(parameter, "ttl")) {
775 th->ttl = strtol(p, &end, 10);
778 } else if (!strcmp(parameter, "destination")) {
781 get_word_sep(buf, sizeof(buf), ";,", &p);
782 get_sockaddr(buf, &th->destination);
784 } else if (!strcmp(parameter, "source")) {
787 get_word_sep(buf, sizeof(buf), ";,", &p);
788 av_strlcpy(th->source, buf, sizeof(th->source));
790 } else if (!strcmp(parameter, "mode")) {
793 get_word_sep(buf, sizeof(buf), ";, ", &p);
794 if (!strcmp(buf, "record") ||
795 !strcmp(buf, "receive"))
800 while (*p != ';' && *p != '\0' && *p != ',')
808 reply->nb_transports++;
812 static void handle_rtp_info(RTSPState *rt, const char *url,
813 uint32_t seq, uint32_t rtptime)
816 if (!rtptime || !url[0])
818 if (rt->transport != RTSP_TRANSPORT_RTP)
820 for (i = 0; i < rt->nb_rtsp_streams; i++) {
821 RTSPStream *rtsp_st = rt->rtsp_streams[i];
822 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
825 if (!strcmp(rtsp_st->control_url, url)) {
826 rtpctx->base_timestamp = rtptime;
832 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
835 char key[20], value[1024], url[1024] = "";
836 uint32_t seq = 0, rtptime = 0;
839 p += strspn(p, SPACE_CHARS);
842 get_word_sep(key, sizeof(key), "=", &p);
846 get_word_sep(value, sizeof(value), ";, ", &p);
848 if (!strcmp(key, "url"))
849 av_strlcpy(url, value, sizeof(url));
850 else if (!strcmp(key, "seq"))
851 seq = strtoul(value, NULL, 10);
852 else if (!strcmp(key, "rtptime"))
853 rtptime = strtoul(value, NULL, 10);
855 handle_rtp_info(rt, url, seq, rtptime);
864 handle_rtp_info(rt, url, seq, rtptime);
867 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
868 RTSPState *rt, const char *method)
872 /* NOTE: we do case independent match for broken servers */
874 if (av_stristart(p, "Session:", &p)) {
876 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
877 if (av_stristart(p, ";timeout=", &p) &&
878 (t = strtol(p, NULL, 10)) > 0) {
881 } else if (av_stristart(p, "Content-Length:", &p)) {
882 reply->content_length = strtol(p, NULL, 10);
883 } else if (av_stristart(p, "Transport:", &p)) {
884 rtsp_parse_transport(reply, p);
885 } else if (av_stristart(p, "CSeq:", &p)) {
886 reply->seq = strtol(p, NULL, 10);
887 } else if (av_stristart(p, "Range:", &p)) {
888 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
889 } else if (av_stristart(p, "RealChallenge1:", &p)) {
890 p += strspn(p, SPACE_CHARS);
891 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
892 } else if (av_stristart(p, "Server:", &p)) {
893 p += strspn(p, SPACE_CHARS);
894 av_strlcpy(reply->server, p, sizeof(reply->server));
895 } else if (av_stristart(p, "Notice:", &p) ||
896 av_stristart(p, "X-Notice:", &p)) {
897 reply->notice = strtol(p, NULL, 10);
898 } else if (av_stristart(p, "Location:", &p)) {
899 p += strspn(p, SPACE_CHARS);
900 av_strlcpy(reply->location, p , sizeof(reply->location));
901 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
902 p += strspn(p, SPACE_CHARS);
903 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
904 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
905 p += strspn(p, SPACE_CHARS);
906 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
907 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
908 p += strspn(p, SPACE_CHARS);
909 if (method && !strcmp(method, "DESCRIBE"))
910 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
911 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
912 p += strspn(p, SPACE_CHARS);
913 if (method && !strcmp(method, "PLAY"))
914 rtsp_parse_rtp_info(rt, p);
915 } else if (av_stristart(p, "Public:", &p) && rt) {
916 if (strstr(p, "GET_PARAMETER") &&
917 method && !strcmp(method, "OPTIONS"))
918 rt->get_parameter_supported = 1;
919 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
920 p += strspn(p, SPACE_CHARS);
921 rt->accept_dynamic_rate = atoi(p);
922 } else if (av_stristart(p, "Content-Type:", &p)) {
923 p += strspn(p, SPACE_CHARS);
924 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
928 /* skip a RTP/TCP interleaved packet */
929 void ff_rtsp_skip_packet(AVFormatContext *s)
931 RTSPState *rt = s->priv_data;
935 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
938 len = AV_RB16(buf + 1);
940 av_dlog(s, "skipping RTP packet len=%d\n", len);
945 if (len1 > sizeof(buf))
947 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
954 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
955 unsigned char **content_ptr,
956 int return_on_interleaved_data, const char *method)
958 RTSPState *rt = s->priv_data;
959 char buf[4096], buf1[1024], *q;
962 int ret, content_length, line_count = 0, request = 0;
963 unsigned char *content = NULL;
969 memset(reply, 0, sizeof(*reply));
971 /* parse reply (XXX: use buffers) */
972 rt->last_reply[0] = '\0';
976 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
977 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
983 /* XXX: only parse it if first char on line ? */
984 if (return_on_interleaved_data) {
987 ff_rtsp_skip_packet(s);
988 } else if (ch != '\r') {
989 if ((q - buf) < sizeof(buf) - 1)
995 av_dlog(s, "line='%s'\n", buf);
997 /* test if last line */
1001 if (line_count == 0) {
1002 /* get reply code */
1003 get_word(buf1, sizeof(buf1), &p);
1004 if (!strncmp(buf1, "RTSP/", 5)) {
1005 get_word(buf1, sizeof(buf1), &p);
1006 reply->status_code = atoi(buf1);
1007 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1009 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1010 get_word(buf1, sizeof(buf1), &p); // object
1014 ff_rtsp_parse_line(reply, p, rt, method);
1015 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1016 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1021 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1022 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1024 content_length = reply->content_length;
1025 if (content_length > 0) {
1026 /* leave some room for a trailing '\0' (useful for simple parsing) */
1027 content = av_malloc(content_length + 1);
1028 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1029 content[content_length] = '\0';
1032 *content_ptr = content;
1038 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1039 const char* ptr = buf;
1041 if (!strcmp(reply->reason, "OPTIONS")) {
1042 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1044 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1045 if (reply->session_id[0])
1046 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1049 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1051 av_strlcat(buf, "\r\n", sizeof(buf));
1053 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1054 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1057 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1059 rt->last_cmd_time = av_gettime();
1060 /* Even if the request from the server had data, it is not the data
1061 * that the caller wants or expects. The memory could also be leaked
1062 * if the actual following reply has content data. */
1064 av_freep(content_ptr);
1065 /* If method is set, this is called from ff_rtsp_send_cmd,
1066 * where a reply to exactly this request is awaited. For
1067 * callers from within packet receiving, we just want to
1068 * return to the caller and go back to receiving packets. */
1074 if (rt->seq != reply->seq) {
1075 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1076 rt->seq, reply->seq);
1080 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1081 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1082 reply->notice == 2306 /* Continuous Feed Terminated */) {
1083 rt->state = RTSP_STATE_IDLE;
1084 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1085 return AVERROR(EIO); /* data or server error */
1086 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1087 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1088 return AVERROR(EPERM);
1094 * Send a command to the RTSP server without waiting for the reply.
1096 * @param s RTSP (de)muxer context
1097 * @param method the method for the request
1098 * @param url the target url for the request
1099 * @param headers extra header lines to include in the request
1100 * @param send_content if non-null, the data to send as request body content
1101 * @param send_content_length the length of the send_content data, or 0 if
1102 * send_content is null
1104 * @return zero if success, nonzero otherwise
1106 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1107 const char *method, const char *url,
1108 const char *headers,
1109 const unsigned char *send_content,
1110 int send_content_length)
1112 RTSPState *rt = s->priv_data;
1113 char buf[4096], *out_buf;
1114 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1116 /* Add in RTSP headers */
1119 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1121 av_strlcat(buf, headers, sizeof(buf));
1122 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1123 if (rt->session_id[0] != '\0' && (!headers ||
1124 !strstr(headers, "\nIf-Match:"))) {
1125 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1128 char *str = ff_http_auth_create_response(&rt->auth_state,
1129 rt->auth, url, method);
1131 av_strlcat(buf, str, sizeof(buf));
1134 if (send_content_length > 0 && send_content)
1135 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1136 av_strlcat(buf, "\r\n", sizeof(buf));
1138 /* base64 encode rtsp if tunneling */
1139 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1140 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1141 out_buf = base64buf;
1144 av_dlog(s, "Sending:\n%s--\n", buf);
1146 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1147 if (send_content_length > 0 && send_content) {
1148 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1149 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1150 "with content data not supported\n");
1151 return AVERROR_PATCHWELCOME;
1153 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1155 rt->last_cmd_time = av_gettime();
1160 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1161 const char *url, const char *headers)
1163 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1166 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1167 const char *headers, RTSPMessageHeader *reply,
1168 unsigned char **content_ptr)
1170 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1171 content_ptr, NULL, 0);
1174 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1175 const char *method, const char *url,
1177 RTSPMessageHeader *reply,
1178 unsigned char **content_ptr,
1179 const unsigned char *send_content,
1180 int send_content_length)
1182 RTSPState *rt = s->priv_data;
1183 HTTPAuthType cur_auth_type;
1184 int ret, attempts = 0;
1187 cur_auth_type = rt->auth_state.auth_type;
1188 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1190 send_content_length)))
1193 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1197 if (reply->status_code == 401 &&
1198 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1199 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1202 if (reply->status_code > 400){
1203 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1207 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1213 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1214 int lower_transport, const char *real_challenge)
1216 RTSPState *rt = s->priv_data;
1217 int rtx = 0, j, i, err, interleave = 0, port_off;
1218 RTSPStream *rtsp_st;
1219 RTSPMessageHeader reply1, *reply = &reply1;
1221 const char *trans_pref;
1223 if (rt->transport == RTSP_TRANSPORT_RDT)
1224 trans_pref = "x-pn-tng";
1225 else if (rt->transport == RTSP_TRANSPORT_RAW)
1226 trans_pref = "RAW/RAW";
1228 trans_pref = "RTP/AVP";
1230 /* default timeout: 1 minute */
1233 /* Choose a random starting offset within the first half of the
1234 * port range, to allow for a number of ports to try even if the offset
1235 * happens to be at the end of the random range. */
1236 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1237 /* even random offset */
1238 port_off -= port_off & 0x01;
1240 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1241 char transport[2048];
1244 * WMS serves all UDP data over a single connection, the RTX, which
1245 * isn't necessarily the first in the SDP but has to be the first
1246 * to be set up, else the second/third SETUP will fail with a 461.
1248 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1249 rt->server_type == RTSP_SERVER_WMS) {
1252 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1253 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1255 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1259 if (rtx == rt->nb_rtsp_streams)
1260 return -1; /* no RTX found */
1261 rtsp_st = rt->rtsp_streams[rtx];
1263 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1265 rtsp_st = rt->rtsp_streams[i];
1268 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1271 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1272 port = reply->transports[0].client_port_min;
1276 /* first try in specified port range */
1277 while (j <= rt->rtp_port_max) {
1278 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1279 "?localport=%d", j);
1280 /* we will use two ports per rtp stream (rtp and rtcp) */
1282 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1283 &s->interrupt_callback, NULL))
1286 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1291 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1293 snprintf(transport, sizeof(transport) - 1,
1294 "%s/UDP;", trans_pref);
1295 if (rt->server_type != RTSP_SERVER_REAL)
1296 av_strlcat(transport, "unicast;", sizeof(transport));
1297 av_strlcatf(transport, sizeof(transport),
1298 "client_port=%d", port);
1299 if (rt->transport == RTSP_TRANSPORT_RTP &&
1300 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1301 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1305 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1306 /* For WMS streams, the application streams are only used for
1307 * UDP. When trying to set it up for TCP streams, the server
1308 * will return an error. Therefore, we skip those streams. */
1309 if (rt->server_type == RTSP_SERVER_WMS &&
1310 (rtsp_st->stream_index < 0 ||
1311 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1314 snprintf(transport, sizeof(transport) - 1,
1315 "%s/TCP;", trans_pref);
1316 if (rt->transport != RTSP_TRANSPORT_RDT)
1317 av_strlcat(transport, "unicast;", sizeof(transport));
1318 av_strlcatf(transport, sizeof(transport),
1319 "interleaved=%d-%d",
1320 interleave, interleave + 1);
1324 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1325 snprintf(transport, sizeof(transport) - 1,
1326 "%s/UDP;multicast", trans_pref);
1329 av_strlcat(transport, ";mode=record", sizeof(transport));
1330 } else if (rt->server_type == RTSP_SERVER_REAL ||
1331 rt->server_type == RTSP_SERVER_WMS)
1332 av_strlcat(transport, ";mode=play", sizeof(transport));
1333 snprintf(cmd, sizeof(cmd),
1334 "Transport: %s\r\n",
1336 if (rt->accept_dynamic_rate)
1337 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1338 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1339 char real_res[41], real_csum[9];
1340 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1342 av_strlcatf(cmd, sizeof(cmd),
1344 "RealChallenge2: %s, sd=%s\r\n",
1345 rt->session_id, real_res, real_csum);
1347 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1348 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1351 } else if (reply->status_code != RTSP_STATUS_OK ||
1352 reply->nb_transports != 1) {
1353 err = AVERROR_INVALIDDATA;
1357 /* XXX: same protocol for all streams is required */
1359 if (reply->transports[0].lower_transport != rt->lower_transport ||
1360 reply->transports[0].transport != rt->transport) {
1361 err = AVERROR_INVALIDDATA;
1365 rt->lower_transport = reply->transports[0].lower_transport;
1366 rt->transport = reply->transports[0].transport;
1369 /* Fail if the server responded with another lower transport mode
1370 * than what we requested. */
1371 if (reply->transports[0].lower_transport != lower_transport) {
1372 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1373 err = AVERROR_INVALIDDATA;
1377 switch(reply->transports[0].lower_transport) {
1378 case RTSP_LOWER_TRANSPORT_TCP:
1379 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1380 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1383 case RTSP_LOWER_TRANSPORT_UDP: {
1384 char url[1024], options[30] = "";
1386 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1387 av_strlcpy(options, "?connect=1", sizeof(options));
1388 /* Use source address if specified */
1389 if (reply->transports[0].source[0]) {
1390 ff_url_join(url, sizeof(url), "rtp", NULL,
1391 reply->transports[0].source,
1392 reply->transports[0].server_port_min, "%s", options);
1394 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1395 reply->transports[0].server_port_min, "%s", options);
1397 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1398 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1399 err = AVERROR_INVALIDDATA;
1402 /* Try to initialize the connection state in a
1403 * potential NAT router by sending dummy packets.
1404 * RTP/RTCP dummy packets are used for RDT, too.
1406 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1408 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1411 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1412 char url[1024], namebuf[50], optbuf[20] = "";
1413 struct sockaddr_storage addr;
1416 if (reply->transports[0].destination.ss_family) {
1417 addr = reply->transports[0].destination;
1418 port = reply->transports[0].port_min;
1419 ttl = reply->transports[0].ttl;
1421 addr = rtsp_st->sdp_ip;
1422 port = rtsp_st->sdp_port;
1423 ttl = rtsp_st->sdp_ttl;
1426 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1427 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1428 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1429 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1430 port, "%s", optbuf);
1431 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1432 &s->interrupt_callback, NULL) < 0) {
1433 err = AVERROR_INVALIDDATA;
1440 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1444 if (rt->nb_rtsp_streams && reply->timeout > 0)
1445 rt->timeout = reply->timeout;
1447 if (rt->server_type == RTSP_SERVER_REAL)
1448 rt->need_subscription = 1;
1453 ff_rtsp_undo_setup(s);
1457 void ff_rtsp_close_connections(AVFormatContext *s)
1459 RTSPState *rt = s->priv_data;
1460 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1461 ffurl_close(rt->rtsp_hd);
1462 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1465 int ff_rtsp_connect(AVFormatContext *s)
1467 RTSPState *rt = s->priv_data;
1468 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1469 int port, err, tcp_fd;
1470 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1471 int lower_transport_mask = 0;
1472 char real_challenge[64] = "";
1473 struct sockaddr_storage peer;
1474 socklen_t peer_len = sizeof(peer);
1476 if (rt->rtp_port_max < rt->rtp_port_min) {
1477 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1478 "than min port %d\n", rt->rtp_port_max,
1480 return AVERROR(EINVAL);
1483 if (!ff_network_init())
1484 return AVERROR(EIO);
1486 if (s->max_delay < 0) /* Not set by the caller */
1487 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1489 rt->control_transport = RTSP_MODE_PLAIN;
1490 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1491 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1492 rt->control_transport = RTSP_MODE_TUNNEL;
1494 /* Only pass through valid flags from here */
1495 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1498 lower_transport_mask = rt->lower_transport_mask;
1499 /* extract hostname and port */
1500 av_url_split(NULL, 0, auth, sizeof(auth),
1501 host, sizeof(host), &port, path, sizeof(path), s->filename);
1503 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1506 port = RTSP_DEFAULT_PORT;
1508 if (!lower_transport_mask)
1509 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1512 /* Only UDP or TCP - UDP multicast isn't supported. */
1513 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1514 (1 << RTSP_LOWER_TRANSPORT_TCP);
1515 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1516 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1517 "only UDP and TCP are supported for output.\n");
1518 err = AVERROR(EINVAL);
1523 /* Construct the URI used in request; this is similar to s->filename,
1524 * but with authentication credentials removed and RTSP specific options
1526 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1527 host, port, "%s", path);
1529 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1530 /* set up initial handshake for tunneling */
1531 char httpname[1024];
1532 char sessioncookie[17];
1535 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1536 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1537 av_get_random_seed(), av_get_random_seed());
1540 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1541 &s->interrupt_callback) < 0) {
1546 /* generate GET headers */
1547 snprintf(headers, sizeof(headers),
1548 "x-sessioncookie: %s\r\n"
1549 "Accept: application/x-rtsp-tunnelled\r\n"
1550 "Pragma: no-cache\r\n"
1551 "Cache-Control: no-cache\r\n",
1553 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1555 /* complete the connection */
1556 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1562 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1563 &s->interrupt_callback) < 0 ) {
1568 /* generate POST headers */
1569 snprintf(headers, sizeof(headers),
1570 "x-sessioncookie: %s\r\n"
1571 "Content-Type: application/x-rtsp-tunnelled\r\n"
1572 "Pragma: no-cache\r\n"
1573 "Cache-Control: no-cache\r\n"
1574 "Content-Length: 32767\r\n"
1575 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1577 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1578 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1580 /* Initialize the authentication state for the POST session. The HTTP
1581 * protocol implementation doesn't properly handle multi-pass
1582 * authentication for POST requests, since it would require one of
1584 * - implementing Expect: 100-continue, which many HTTP servers
1585 * don't support anyway, even less the RTSP servers that do HTTP
1587 * - sending the whole POST data until getting a 401 reply specifying
1588 * what authentication method to use, then resending all that data
1589 * - waiting for potential 401 replies directly after sending the
1590 * POST header (waiting for some unspecified time)
1591 * Therefore, we copy the full auth state, which works for both basic
1592 * and digest. (For digest, we would have to synchronize the nonce
1593 * count variable between the two sessions, if we'd do more requests
1594 * with the original session, though.)
1596 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1598 /* complete the connection */
1599 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1604 /* open the tcp connection */
1605 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1606 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1607 &s->interrupt_callback, NULL) < 0) {
1611 rt->rtsp_hd_out = rt->rtsp_hd;
1615 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1616 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1617 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1618 NULL, 0, NI_NUMERICHOST);
1621 /* request options supported by the server; this also detects server
1623 for (rt->server_type = RTSP_SERVER_RTP;;) {
1625 if (rt->server_type == RTSP_SERVER_REAL)
1628 * The following entries are required for proper
1629 * streaming from a Realmedia server. They are
1630 * interdependent in some way although we currently
1631 * don't quite understand how. Values were copied
1632 * from mplayer SVN r23589.
1633 * ClientChallenge is a 16-byte ID in hex
1634 * CompanyID is a 16-byte ID in base64
1636 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1637 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1638 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1639 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1641 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1642 if (reply->status_code != RTSP_STATUS_OK) {
1643 err = AVERROR_INVALIDDATA;
1647 /* detect server type if not standard-compliant RTP */
1648 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1649 rt->server_type = RTSP_SERVER_REAL;
1651 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1652 rt->server_type = RTSP_SERVER_WMS;
1653 } else if (rt->server_type == RTSP_SERVER_REAL)
1654 strcpy(real_challenge, reply->real_challenge);
1658 if (s->iformat && CONFIG_RTSP_DEMUXER)
1659 err = ff_rtsp_setup_input_streams(s, reply);
1660 else if (CONFIG_RTSP_MUXER)
1661 err = ff_rtsp_setup_output_streams(s, host);
1666 int lower_transport = ff_log2_tab[lower_transport_mask &
1667 ~(lower_transport_mask - 1)];
1669 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1670 rt->server_type == RTSP_SERVER_REAL ?
1671 real_challenge : NULL);
1674 lower_transport_mask &= ~(1 << lower_transport);
1675 if (lower_transport_mask == 0 && err == 1) {
1676 err = AVERROR(EPROTONOSUPPORT);
1681 rt->lower_transport_mask = lower_transport_mask;
1682 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1683 rt->state = RTSP_STATE_IDLE;
1684 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1687 ff_rtsp_close_streams(s);
1688 ff_rtsp_close_connections(s);
1689 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1690 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1691 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1699 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1702 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1703 uint8_t *buf, int buf_size, int64_t wait_end)
1705 RTSPState *rt = s->priv_data;
1706 RTSPStream *rtsp_st;
1707 int n, i, ret, tcp_fd, timeout_cnt = 0;
1709 struct pollfd *p = rt->p;
1710 int *fds = NULL, fdsnum, fdsidx;
1713 if (ff_check_interrupt(&s->interrupt_callback))
1714 return AVERROR_EXIT;
1715 if (wait_end && wait_end - av_gettime() < 0)
1716 return AVERROR(EAGAIN);
1719 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1720 p[max_p].fd = tcp_fd;
1721 p[max_p++].events = POLLIN;
1725 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1726 rtsp_st = rt->rtsp_streams[i];
1727 if (rtsp_st->rtp_handle) {
1728 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1730 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1734 av_log(s, AV_LOG_ERROR,
1735 "Number of fds %d not supported\n", fdsnum);
1736 return AVERROR_INVALIDDATA;
1738 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1739 p[max_p].fd = fds[fdsidx];
1740 p[max_p++].events = POLLIN;
1745 n = poll(p, max_p, POLL_TIMEOUT_MS);
1747 int j = 1 - (tcp_fd == -1);
1749 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1750 rtsp_st = rt->rtsp_streams[i];
1751 if (rtsp_st->rtp_handle) {
1752 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1753 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1755 *prtsp_st = rtsp_st;
1762 #if CONFIG_RTSP_DEMUXER
1763 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1764 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1765 if (rt->state == RTSP_STATE_STREAMING) {
1766 if (!ff_rtsp_parse_streaming_commands(s))
1769 av_log(s, AV_LOG_WARNING,
1770 "Unable to answer to TEARDOWN\n");
1774 RTSPMessageHeader reply;
1775 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1778 /* XXX: parse message */
1779 if (rt->state != RTSP_STATE_STREAMING)
1784 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1785 return AVERROR(ETIMEDOUT);
1786 } else if (n < 0 && errno != EINTR)
1787 return AVERROR(errno);
1791 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1792 const uint8_t *buf, int len)
1794 RTSPState *rt = s->priv_data;
1798 if (rt->nb_rtsp_streams == 1) {
1799 *rtsp_st = rt->rtsp_streams[0];
1802 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1803 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1805 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1806 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1809 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1810 *rtsp_st = rt->rtsp_streams[i];
1817 av_log(s, AV_LOG_WARNING,
1818 "Unable to pick stream for packet - SSRC not known for "
1820 return AVERROR(EAGAIN);
1823 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1824 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1825 *rtsp_st = rt->rtsp_streams[i];
1831 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1832 return AVERROR(EAGAIN);
1835 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1837 RTSPState *rt = s->priv_data;
1839 RTSPStream *rtsp_st, *first_queue_st = NULL;
1840 int64_t wait_end = 0;
1842 if (rt->nb_byes == rt->nb_rtsp_streams)
1845 /* get next frames from the same RTP packet */
1846 if (rt->cur_transport_priv) {
1847 if (rt->transport == RTSP_TRANSPORT_RDT) {
1848 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1849 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1850 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1851 } else if (rt->ts && CONFIG_RTPDEC) {
1852 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1854 rt->recvbuf_pos += ret;
1855 ret = rt->recvbuf_pos < rt->recvbuf_len;
1860 rt->cur_transport_priv = NULL;
1862 } else if (ret == 1) {
1865 rt->cur_transport_priv = NULL;
1869 if (rt->transport == RTSP_TRANSPORT_RTP) {
1871 int64_t first_queue_time = 0;
1872 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1873 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1877 queue_time = ff_rtp_queued_packet_time(rtpctx);
1878 if (queue_time && (queue_time - first_queue_time < 0 ||
1879 !first_queue_time)) {
1880 first_queue_time = queue_time;
1881 first_queue_st = rt->rtsp_streams[i];
1884 if (first_queue_time) {
1885 wait_end = first_queue_time + s->max_delay;
1888 first_queue_st = NULL;
1892 /* read next RTP packet */
1894 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1896 return AVERROR(ENOMEM);
1899 switch(rt->lower_transport) {
1901 #if CONFIG_RTSP_DEMUXER
1902 case RTSP_LOWER_TRANSPORT_TCP:
1903 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1906 case RTSP_LOWER_TRANSPORT_UDP:
1907 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1908 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
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, rtsp_st->rtp_handle, NULL, len);
1912 case RTSP_LOWER_TRANSPORT_CUSTOM:
1913 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1914 wait_end && wait_end < av_gettime())
1915 len = AVERROR(EAGAIN);
1917 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1918 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1919 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1920 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1923 if (len == AVERROR(EAGAIN) && first_queue_st &&
1924 rt->transport == RTSP_TRANSPORT_RTP) {
1925 rtsp_st = first_queue_st;
1926 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1933 if (rt->transport == RTSP_TRANSPORT_RDT) {
1934 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1935 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1936 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1937 if (rtsp_st->feedback) {
1938 AVIOContext *pb = NULL;
1939 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1941 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1944 /* Either bad packet, or a RTCP packet. Check if the
1945 * first_rtcp_ntp_time field was initialized. */
1946 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1947 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1948 /* first_rtcp_ntp_time has been initialized for this stream,
1949 * copy the same value to all other uninitialized streams,
1950 * in order to map their timestamp origin to the same ntp time
1953 AVStream *st = NULL;
1954 if (rtsp_st->stream_index >= 0)
1955 st = s->streams[rtsp_st->stream_index];
1956 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1957 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1958 AVStream *st2 = NULL;
1959 if (rt->rtsp_streams[i]->stream_index >= 0)
1960 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1961 if (rtpctx2 && st && st2 &&
1962 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1963 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1964 rtpctx2->rtcp_ts_offset = av_rescale_q(
1965 rtpctx->rtcp_ts_offset, st->time_base,
1970 if (ret == -RTCP_BYE) {
1973 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1974 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1976 if (rt->nb_byes == rt->nb_rtsp_streams)
1980 } else if (rt->ts && CONFIG_RTPDEC) {
1981 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1984 rt->recvbuf_len = len;
1985 rt->recvbuf_pos = ret;
1986 rt->cur_transport_priv = rt->ts;
1993 return AVERROR_INVALIDDATA;
1999 /* more packets may follow, so we save the RTP context */
2000 rt->cur_transport_priv = rtsp_st->transport_priv;
2004 #endif /* CONFIG_RTPDEC */
2006 #if CONFIG_SDP_DEMUXER
2007 static int sdp_probe(AVProbeData *p1)
2009 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2011 /* we look for a line beginning "c=IN IP" */
2012 while (p < p_end && *p != '\0') {
2013 if (p + sizeof("c=IN IP") - 1 < p_end &&
2014 av_strstart(p, "c=IN IP", NULL))
2015 return AVPROBE_SCORE_MAX / 2;
2017 while (p < p_end - 1 && *p != '\n') p++;
2026 static int sdp_read_header(AVFormatContext *s)
2028 RTSPState *rt = s->priv_data;
2029 RTSPStream *rtsp_st;
2034 if (!ff_network_init())
2035 return AVERROR(EIO);
2037 if (s->max_delay < 0) /* Not set by the caller */
2038 s->max_delay = DEFAULT_REORDERING_DELAY;
2039 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2040 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2042 /* read the whole sdp file */
2043 /* XXX: better loading */
2044 content = av_malloc(SDP_MAX_SIZE);
2045 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2048 return AVERROR_INVALIDDATA;
2050 content[size] ='\0';
2052 err = ff_sdp_parse(s, content);
2056 /* open each RTP stream */
2057 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2059 rtsp_st = rt->rtsp_streams[i];
2061 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2062 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2063 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2064 ff_url_join(url, sizeof(url), "rtp", NULL,
2065 namebuf, rtsp_st->sdp_port,
2066 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2068 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2069 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2070 &s->interrupt_callback, NULL) < 0) {
2071 err = AVERROR_INVALIDDATA;
2075 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2080 ff_rtsp_close_streams(s);
2085 static int sdp_read_close(AVFormatContext *s)
2087 ff_rtsp_close_streams(s);
2092 static const AVClass sdp_demuxer_class = {
2093 .class_name = "SDP demuxer",
2094 .item_name = av_default_item_name,
2095 .option = sdp_options,
2096 .version = LIBAVUTIL_VERSION_INT,
2099 AVInputFormat ff_sdp_demuxer = {
2101 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2102 .priv_data_size = sizeof(RTSPState),
2103 .read_probe = sdp_probe,
2104 .read_header = sdp_read_header,
2105 .read_packet = ff_rtsp_fetch_packet,
2106 .read_close = sdp_read_close,
2107 .priv_class = &sdp_demuxer_class,
2109 #endif /* CONFIG_SDP_DEMUXER */
2111 #if CONFIG_RTP_DEMUXER
2112 static int rtp_probe(AVProbeData *p)
2114 if (av_strstart(p->filename, "rtp:", NULL))
2115 return AVPROBE_SCORE_MAX;
2119 static int rtp_read_header(AVFormatContext *s)
2121 uint8_t recvbuf[1500];
2122 char host[500], sdp[500];
2124 URLContext* in = NULL;
2126 AVCodecContext codec = { 0 };
2127 struct sockaddr_storage addr;
2129 socklen_t addrlen = sizeof(addr);
2130 RTSPState *rt = s->priv_data;
2132 if (!ff_network_init())
2133 return AVERROR(EIO);
2135 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2136 &s->interrupt_callback, NULL);
2141 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2142 if (ret == AVERROR(EAGAIN))
2147 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2151 if ((recvbuf[0] & 0xc0) != 0x80) {
2152 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2157 if (RTP_PT_IS_RTCP(recvbuf[1]))
2160 payload_type = recvbuf[1] & 0x7f;
2163 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2167 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2168 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2169 "without an SDP file describing it\n",
2173 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2174 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2175 "properly you need an SDP file "
2179 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2180 NULL, 0, s->filename);
2182 snprintf(sdp, sizeof(sdp),
2183 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2184 addr.ss_family == AF_INET ? 4 : 6, host,
2185 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2186 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2187 port, payload_type);
2188 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2190 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2193 /* sdp_read_header initializes this again */
2196 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2198 ret = sdp_read_header(s);
2209 static const AVClass rtp_demuxer_class = {
2210 .class_name = "RTP demuxer",
2211 .item_name = av_default_item_name,
2212 .option = rtp_options,
2213 .version = LIBAVUTIL_VERSION_INT,
2216 AVInputFormat ff_rtp_demuxer = {
2218 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2219 .priv_data_size = sizeof(RTSPState),
2220 .read_probe = rtp_probe,
2221 .read_header = rtp_read_header,
2222 .read_packet = ff_rtsp_fetch_packet,
2223 .read_close = sdp_read_close,
2224 .flags = AVFMT_NOFILE,
2225 .priv_class = &rtp_demuxer_class,
2227 #endif /* CONFIG_RTP_DEMUXER */