3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
31 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
68 #define RTSP_MEDIATYPE_OPTS(name, longname) \
69 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
70 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 const AVOption ff_rtsp_options[] = {
75 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
76 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
77 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
78 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
79 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
81 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
82 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
83 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 static const AVOption sdp_options[] = {
88 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
89 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
93 static const AVOption rtp_options[] = {
94 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
98 static void get_word_until_chars(char *buf, int buf_size,
99 const char *sep, const char **pp)
105 p += strspn(p, SPACE_CHARS);
107 while (!strchr(sep, *p) && *p != '\0') {
108 if ((q - buf) < buf_size - 1)
117 static void get_word_sep(char *buf, int buf_size, const char *sep,
120 if (**pp == '/') (*pp)++;
121 get_word_until_chars(buf, buf_size, sep, pp);
124 static void get_word(char *buf, int buf_size, const char **pp)
126 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
129 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
131 * Used for seeking in the rtp stream.
133 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
137 p += strspn(p, SPACE_CHARS);
138 if (!av_stristart(p, "npt=", &p))
141 *start = AV_NOPTS_VALUE;
142 *end = AV_NOPTS_VALUE;
144 get_word_sep(buf, sizeof(buf), "-", &p);
145 av_parse_time(start, buf, 1);
148 get_word_sep(buf, sizeof(buf), "-", &p);
149 av_parse_time(end, buf, 1);
151 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
152 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
155 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
157 struct addrinfo hints, *ai = NULL;
158 memset(&hints, 0, sizeof(hints));
159 hints.ai_flags = AI_NUMERICHOST;
160 if (getaddrinfo(buf, NULL, &hints, &ai))
162 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
168 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
169 RTSPStream *rtsp_st, AVCodecContext *codec)
173 codec->codec_id = handler->codec_id;
174 rtsp_st->dynamic_handler = handler;
175 if (handler->alloc) {
176 rtsp_st->dynamic_protocol_context = handler->alloc();
177 if (!rtsp_st->dynamic_protocol_context)
178 rtsp_st->dynamic_handler = NULL;
182 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
183 static int sdp_parse_rtpmap(AVFormatContext *s,
184 AVStream *st, RTSPStream *rtsp_st,
185 int payload_type, const char *p)
187 AVCodecContext *codec = st->codec;
193 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
194 * see if we can handle this kind of payload.
195 * The space should normally not be there but some Real streams or
196 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
197 * have a trailing space. */
198 get_word_sep(buf, sizeof(buf), "/ ", &p);
199 if (payload_type >= RTP_PT_PRIVATE) {
200 RTPDynamicProtocolHandler *handler =
201 ff_rtp_handler_find_by_name(buf, codec->codec_type);
202 init_rtp_handler(handler, rtsp_st, codec);
203 /* If no dynamic handler was found, check with the list of standard
204 * allocated types, if such a stream for some reason happens to
205 * use a private payload type. This isn't handled in rtpdec.c, since
206 * the format name from the rtpmap line never is passed into rtpdec. */
207 if (!rtsp_st->dynamic_handler)
208 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 /* search into AVRtpPayloadTypes[] */
213 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
216 c = avcodec_find_decoder(codec->codec_id);
222 get_word_sep(buf, sizeof(buf), "/", &p);
224 switch (codec->codec_type) {
225 case AVMEDIA_TYPE_AUDIO:
226 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
227 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
228 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
230 codec->sample_rate = i;
231 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
232 get_word_sep(buf, sizeof(buf), "/", &p);
236 // TODO: there is a bug here; if it is a mono stream, and
237 // less than 22000Hz, faad upconverts to stereo and twice
238 // the frequency. No problem, but the sample rate is being
239 // set here by the sdp line. Patch on its way. (rdm)
241 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
243 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
246 case AVMEDIA_TYPE_VIDEO:
247 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
249 avpriv_set_pts_info(st, 32, 1, i);
254 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
255 rtsp_st->dynamic_handler->init(s, st->index,
256 rtsp_st->dynamic_protocol_context);
260 /* parse the attribute line from the fmtp a line of an sdp response. This
261 * is broken out as a function because it is used in rtp_h264.c, which is
263 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
264 char *value, int value_size)
266 *p += strspn(*p, SPACE_CHARS);
268 get_word_sep(attr, attr_size, "=", p);
271 get_word_sep(value, value_size, ";", p);
279 typedef struct SDPParseState {
281 struct sockaddr_storage default_ip;
283 int skip_media; ///< set if an unknown m= line occurs
286 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
287 int letter, const char *buf)
289 RTSPState *rt = s->priv_data;
290 char buf1[64], st_type[64];
292 enum AVMediaType codec_type;
296 struct sockaddr_storage sdp_ip;
299 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
302 if (s1->skip_media && letter != 'm')
306 get_word(buf1, sizeof(buf1), &p);
307 if (strcmp(buf1, "IN") != 0)
309 get_word(buf1, sizeof(buf1), &p);
310 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
312 get_word_sep(buf1, sizeof(buf1), "/", &p);
313 if (get_sockaddr(buf1, &sdp_ip))
318 get_word_sep(buf1, sizeof(buf1), "/", &p);
321 if (s->nb_streams == 0) {
322 s1->default_ip = sdp_ip;
323 s1->default_ttl = ttl;
325 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
326 rtsp_st->sdp_ip = sdp_ip;
327 rtsp_st->sdp_ttl = ttl;
331 av_dict_set(&s->metadata, "title", p, 0);
334 if (s->nb_streams == 0) {
335 av_dict_set(&s->metadata, "comment", p, 0);
342 codec_type = AVMEDIA_TYPE_UNKNOWN;
343 get_word(st_type, sizeof(st_type), &p);
344 if (!strcmp(st_type, "audio")) {
345 codec_type = AVMEDIA_TYPE_AUDIO;
346 } else if (!strcmp(st_type, "video")) {
347 codec_type = AVMEDIA_TYPE_VIDEO;
348 } else if (!strcmp(st_type, "application")) {
349 codec_type = AVMEDIA_TYPE_DATA;
351 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
355 rtsp_st = av_mallocz(sizeof(RTSPStream));
358 rtsp_st->stream_index = -1;
359 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
361 rtsp_st->sdp_ip = s1->default_ip;
362 rtsp_st->sdp_ttl = s1->default_ttl;
364 get_word(buf1, sizeof(buf1), &p); /* port */
365 rtsp_st->sdp_port = atoi(buf1);
367 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
369 /* XXX: handle list of formats */
370 get_word(buf1, sizeof(buf1), &p); /* format list */
371 rtsp_st->sdp_payload_type = atoi(buf1);
373 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
374 /* no corresponding stream */
376 st = avformat_new_stream(s, NULL);
379 st->id = rt->nb_rtsp_streams - 1;
380 rtsp_st->stream_index = st->index;
381 st->codec->codec_type = codec_type;
382 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
383 RTPDynamicProtocolHandler *handler;
384 /* if standard payload type, we can find the codec right now */
385 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
386 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
387 st->codec->sample_rate > 0)
388 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
389 /* Even static payload types may need a custom depacketizer */
390 handler = ff_rtp_handler_find_by_id(
391 rtsp_st->sdp_payload_type, st->codec->codec_type);
392 init_rtp_handler(handler, rtsp_st, st->codec);
393 if (handler && handler->init)
394 handler->init(s, st->index,
395 rtsp_st->dynamic_protocol_context);
398 /* put a default control url */
399 av_strlcpy(rtsp_st->control_url, rt->control_uri,
400 sizeof(rtsp_st->control_url));
403 if (av_strstart(p, "control:", &p)) {
404 if (s->nb_streams == 0) {
405 if (!strncmp(p, "rtsp://", 7))
406 av_strlcpy(rt->control_uri, p,
407 sizeof(rt->control_uri));
410 /* get the control url */
411 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
413 /* XXX: may need to add full url resolution */
414 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
416 if (proto[0] == '\0') {
417 /* relative control URL */
418 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
419 av_strlcat(rtsp_st->control_url, "/",
420 sizeof(rtsp_st->control_url));
421 av_strlcat(rtsp_st->control_url, p,
422 sizeof(rtsp_st->control_url));
424 av_strlcpy(rtsp_st->control_url, p,
425 sizeof(rtsp_st->control_url));
427 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
428 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
429 get_word(buf1, sizeof(buf1), &p);
430 payload_type = atoi(buf1);
431 st = s->streams[s->nb_streams - 1];
432 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
433 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
434 } else if (av_strstart(p, "fmtp:", &p) ||
435 av_strstart(p, "framesize:", &p)) {
436 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
437 // let dynamic protocol handlers have a stab at the line.
438 get_word(buf1, sizeof(buf1), &p);
439 payload_type = atoi(buf1);
440 for (i = 0; i < rt->nb_rtsp_streams; i++) {
441 rtsp_st = rt->rtsp_streams[i];
442 if (rtsp_st->sdp_payload_type == payload_type &&
443 rtsp_st->dynamic_handler &&
444 rtsp_st->dynamic_handler->parse_sdp_a_line)
445 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
446 rtsp_st->dynamic_protocol_context, buf);
448 } else if (av_strstart(p, "range:", &p)) {
451 // this is so that seeking on a streamed file can work.
452 rtsp_parse_range_npt(p, &start, &end);
453 s->start_time = start;
454 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
455 s->duration = (end == AV_NOPTS_VALUE) ?
456 AV_NOPTS_VALUE : end - start;
457 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
459 rt->transport = RTSP_TRANSPORT_RDT;
460 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
462 st = s->streams[s->nb_streams - 1];
463 st->codec->sample_rate = atoi(p);
465 if (rt->server_type == RTSP_SERVER_WMS)
466 ff_wms_parse_sdp_a_line(s, p);
467 if (s->nb_streams > 0) {
468 if (rt->server_type == RTSP_SERVER_REAL)
469 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
471 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
472 if (rtsp_st->dynamic_handler &&
473 rtsp_st->dynamic_handler->parse_sdp_a_line)
474 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
476 rtsp_st->dynamic_protocol_context, buf);
483 int ff_sdp_parse(AVFormatContext *s, const char *content)
485 RTSPState *rt = s->priv_data;
488 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
489 * contain long SDP lines containing complete ASF Headers (several
490 * kB) or arrays of MDPR (RM stream descriptor) headers plus
491 * "rulebooks" describing their properties. Therefore, the SDP line
494 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
495 * in rtpdec_xiph.c. */
497 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
499 memset(s1, 0, sizeof(SDPParseState));
502 p += strspn(p, SPACE_CHARS);
510 /* get the content */
512 while (*p != '\n' && *p != '\r' && *p != '\0') {
513 if ((q - buf) < sizeof(buf) - 1)
518 sdp_parse_line(s, s1, letter, buf);
520 while (*p != '\n' && *p != '\0')
525 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
526 if (!rt->p) return AVERROR(ENOMEM);
529 #endif /* CONFIG_RTPDEC */
531 void ff_rtsp_undo_setup(AVFormatContext *s)
533 RTSPState *rt = s->priv_data;
536 for (i = 0; i < rt->nb_rtsp_streams; i++) {
537 RTSPStream *rtsp_st = rt->rtsp_streams[i];
540 if (rtsp_st->transport_priv) {
542 AVFormatContext *rtpctx = rtsp_st->transport_priv;
543 av_write_trailer(rtpctx);
544 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
546 avio_close_dyn_buf(rtpctx->pb, &ptr);
549 avio_close(rtpctx->pb);
551 avformat_free_context(rtpctx);
552 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
553 ff_rdt_parse_close(rtsp_st->transport_priv);
554 else if (CONFIG_RTPDEC)
555 ff_rtp_parse_close(rtsp_st->transport_priv);
557 rtsp_st->transport_priv = NULL;
558 if (rtsp_st->rtp_handle)
559 ffurl_close(rtsp_st->rtp_handle);
560 rtsp_st->rtp_handle = NULL;
564 /* close and free RTSP streams */
565 void ff_rtsp_close_streams(AVFormatContext *s)
567 RTSPState *rt = s->priv_data;
571 ff_rtsp_undo_setup(s);
572 for (i = 0; i < rt->nb_rtsp_streams; i++) {
573 rtsp_st = rt->rtsp_streams[i];
575 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
576 rtsp_st->dynamic_handler->free(
577 rtsp_st->dynamic_protocol_context);
581 av_free(rt->rtsp_streams);
583 avformat_close_input(&rt->asf_ctx);
586 av_free(rt->recvbuf);
589 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
591 RTSPState *rt = s->priv_data;
594 /* open the RTP context */
595 if (rtsp_st->stream_index >= 0)
596 st = s->streams[rtsp_st->stream_index];
598 s->ctx_flags |= AVFMTCTX_NOHEADER;
600 if (s->oformat && CONFIG_RTSP_MUXER) {
601 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
603 RTSP_TCP_MAX_PACKET_SIZE);
604 /* Ownership of rtp_handle is passed to the rtp mux context */
605 rtsp_st->rtp_handle = NULL;
606 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
607 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
608 rtsp_st->dynamic_protocol_context,
609 rtsp_st->dynamic_handler);
610 else if (CONFIG_RTPDEC)
611 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
612 rtsp_st->sdp_payload_type,
613 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
614 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
616 if (!rtsp_st->transport_priv) {
617 return AVERROR(ENOMEM);
618 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
619 if (rtsp_st->dynamic_handler) {
620 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
621 rtsp_st->dynamic_protocol_context,
622 rtsp_st->dynamic_handler);
629 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
630 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
636 p += strspn(p, SPACE_CHARS);
637 v = strtol(p, (char **)&p, 10);
641 v = strtol(p, (char **)&p, 10);
650 /* XXX: only one transport specification is parsed */
651 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
653 char transport_protocol[16];
655 char lower_transport[16];
657 RTSPTransportField *th;
660 reply->nb_transports = 0;
663 p += strspn(p, SPACE_CHARS);
667 th = &reply->transports[reply->nb_transports];
669 get_word_sep(transport_protocol, sizeof(transport_protocol),
671 if (!av_strcasecmp (transport_protocol, "rtp")) {
672 get_word_sep(profile, sizeof(profile), "/;,", &p);
673 lower_transport[0] = '\0';
674 /* rtp/avp/<protocol> */
676 get_word_sep(lower_transport, sizeof(lower_transport),
679 th->transport = RTSP_TRANSPORT_RTP;
680 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
681 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
682 /* x-pn-tng/<protocol> */
683 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
685 th->transport = RTSP_TRANSPORT_RDT;
687 if (!av_strcasecmp(lower_transport, "TCP"))
688 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
690 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
694 /* get each parameter */
695 while (*p != '\0' && *p != ',') {
696 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
697 if (!strcmp(parameter, "port")) {
700 rtsp_parse_range(&th->port_min, &th->port_max, &p);
702 } else if (!strcmp(parameter, "client_port")) {
705 rtsp_parse_range(&th->client_port_min,
706 &th->client_port_max, &p);
708 } else if (!strcmp(parameter, "server_port")) {
711 rtsp_parse_range(&th->server_port_min,
712 &th->server_port_max, &p);
714 } else if (!strcmp(parameter, "interleaved")) {
717 rtsp_parse_range(&th->interleaved_min,
718 &th->interleaved_max, &p);
720 } else if (!strcmp(parameter, "multicast")) {
721 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
722 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
723 } else if (!strcmp(parameter, "ttl")) {
726 th->ttl = strtol(p, (char **)&p, 10);
728 } else if (!strcmp(parameter, "destination")) {
731 get_word_sep(buf, sizeof(buf), ";,", &p);
732 get_sockaddr(buf, &th->destination);
734 } else if (!strcmp(parameter, "source")) {
737 get_word_sep(buf, sizeof(buf), ";,", &p);
738 av_strlcpy(th->source, buf, sizeof(th->source));
742 while (*p != ';' && *p != '\0' && *p != ',')
750 reply->nb_transports++;
754 static void handle_rtp_info(RTSPState *rt, const char *url,
755 uint32_t seq, uint32_t rtptime)
758 if (!rtptime || !url[0])
760 if (rt->transport != RTSP_TRANSPORT_RTP)
762 for (i = 0; i < rt->nb_rtsp_streams; i++) {
763 RTSPStream *rtsp_st = rt->rtsp_streams[i];
764 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
767 if (!strcmp(rtsp_st->control_url, url)) {
768 rtpctx->base_timestamp = rtptime;
774 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
777 char key[20], value[1024], url[1024] = "";
778 uint32_t seq = 0, rtptime = 0;
781 p += strspn(p, SPACE_CHARS);
784 get_word_sep(key, sizeof(key), "=", &p);
788 get_word_sep(value, sizeof(value), ";, ", &p);
790 if (!strcmp(key, "url"))
791 av_strlcpy(url, value, sizeof(url));
792 else if (!strcmp(key, "seq"))
793 seq = strtoul(value, NULL, 10);
794 else if (!strcmp(key, "rtptime"))
795 rtptime = strtoul(value, NULL, 10);
797 handle_rtp_info(rt, url, seq, rtptime);
806 handle_rtp_info(rt, url, seq, rtptime);
809 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
810 RTSPState *rt, const char *method)
814 /* NOTE: we do case independent match for broken servers */
816 if (av_stristart(p, "Session:", &p)) {
818 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
819 if (av_stristart(p, ";timeout=", &p) &&
820 (t = strtol(p, NULL, 10)) > 0) {
823 } else if (av_stristart(p, "Content-Length:", &p)) {
824 reply->content_length = strtol(p, NULL, 10);
825 } else if (av_stristart(p, "Transport:", &p)) {
826 rtsp_parse_transport(reply, p);
827 } else if (av_stristart(p, "CSeq:", &p)) {
828 reply->seq = strtol(p, NULL, 10);
829 } else if (av_stristart(p, "Range:", &p)) {
830 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
831 } else if (av_stristart(p, "RealChallenge1:", &p)) {
832 p += strspn(p, SPACE_CHARS);
833 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
834 } else if (av_stristart(p, "Server:", &p)) {
835 p += strspn(p, SPACE_CHARS);
836 av_strlcpy(reply->server, p, sizeof(reply->server));
837 } else if (av_stristart(p, "Notice:", &p) ||
838 av_stristart(p, "X-Notice:", &p)) {
839 reply->notice = strtol(p, NULL, 10);
840 } else if (av_stristart(p, "Location:", &p)) {
841 p += strspn(p, SPACE_CHARS);
842 av_strlcpy(reply->location, p , sizeof(reply->location));
843 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
844 p += strspn(p, SPACE_CHARS);
845 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
846 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
847 p += strspn(p, SPACE_CHARS);
848 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
849 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
850 p += strspn(p, SPACE_CHARS);
851 if (method && !strcmp(method, "DESCRIBE"))
852 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
853 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
854 p += strspn(p, SPACE_CHARS);
855 if (method && !strcmp(method, "PLAY"))
856 rtsp_parse_rtp_info(rt, p);
857 } else if (av_stristart(p, "Public:", &p) && rt) {
858 if (strstr(p, "GET_PARAMETER") &&
859 method && !strcmp(method, "OPTIONS"))
860 rt->get_parameter_supported = 1;
861 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
862 p += strspn(p, SPACE_CHARS);
863 rt->accept_dynamic_rate = atoi(p);
867 /* skip a RTP/TCP interleaved packet */
868 void ff_rtsp_skip_packet(AVFormatContext *s)
870 RTSPState *rt = s->priv_data;
874 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
877 len = AV_RB16(buf + 1);
879 av_dlog(s, "skipping RTP packet len=%d\n", len);
884 if (len1 > sizeof(buf))
886 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
893 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
894 unsigned char **content_ptr,
895 int return_on_interleaved_data, const char *method)
897 RTSPState *rt = s->priv_data;
898 char buf[4096], buf1[1024], *q;
901 int ret, content_length, line_count = 0;
902 unsigned char *content = NULL;
904 memset(reply, 0, sizeof(*reply));
906 /* parse reply (XXX: use buffers) */
907 rt->last_reply[0] = '\0';
911 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
912 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
918 /* XXX: only parse it if first char on line ? */
919 if (return_on_interleaved_data) {
922 ff_rtsp_skip_packet(s);
923 } else if (ch != '\r') {
924 if ((q - buf) < sizeof(buf) - 1)
930 av_dlog(s, "line='%s'\n", buf);
932 /* test if last line */
936 if (line_count == 0) {
938 get_word(buf1, sizeof(buf1), &p);
939 get_word(buf1, sizeof(buf1), &p);
940 reply->status_code = atoi(buf1);
941 av_strlcpy(reply->reason, p, sizeof(reply->reason));
943 ff_rtsp_parse_line(reply, p, rt, method);
944 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
945 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
950 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
951 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
953 content_length = reply->content_length;
954 if (content_length > 0) {
955 /* leave some room for a trailing '\0' (useful for simple parsing) */
956 content = av_malloc(content_length + 1);
957 ffurl_read_complete(rt->rtsp_hd, content, content_length);
958 content[content_length] = '\0';
961 *content_ptr = content;
965 if (rt->seq != reply->seq) {
966 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
967 rt->seq, reply->seq);
971 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
972 reply->notice == 2104 /* Start-of-Stream Reached */ ||
973 reply->notice == 2306 /* Continuous Feed Terminated */) {
974 rt->state = RTSP_STATE_IDLE;
975 } else if (reply->notice >= 4400 && reply->notice < 5500) {
976 return AVERROR(EIO); /* data or server error */
977 } else if (reply->notice == 2401 /* Ticket Expired */ ||
978 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
979 return AVERROR(EPERM);
985 * Send a command to the RTSP server without waiting for the reply.
987 * @param s RTSP (de)muxer context
988 * @param method the method for the request
989 * @param url the target url for the request
990 * @param headers extra header lines to include in the request
991 * @param send_content if non-null, the data to send as request body content
992 * @param send_content_length the length of the send_content data, or 0 if
993 * send_content is null
995 * @return zero if success, nonzero otherwise
997 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
998 const char *method, const char *url,
1000 const unsigned char *send_content,
1001 int send_content_length)
1003 RTSPState *rt = s->priv_data;
1004 char buf[4096], *out_buf;
1005 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1007 /* Add in RTSP headers */
1010 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1012 av_strlcat(buf, headers, sizeof(buf));
1013 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1014 if (rt->session_id[0] != '\0' && (!headers ||
1015 !strstr(headers, "\nIf-Match:"))) {
1016 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1019 char *str = ff_http_auth_create_response(&rt->auth_state,
1020 rt->auth, url, method);
1022 av_strlcat(buf, str, sizeof(buf));
1025 if (send_content_length > 0 && send_content)
1026 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1027 av_strlcat(buf, "\r\n", sizeof(buf));
1029 /* base64 encode rtsp if tunneling */
1030 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1031 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1032 out_buf = base64buf;
1035 av_dlog(s, "Sending:\n%s--\n", buf);
1037 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1038 if (send_content_length > 0 && send_content) {
1039 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1040 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1041 "with content data not supported\n");
1042 return AVERROR_PATCHWELCOME;
1044 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1046 rt->last_cmd_time = av_gettime();
1051 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1052 const char *url, const char *headers)
1054 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1057 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1058 const char *headers, RTSPMessageHeader *reply,
1059 unsigned char **content_ptr)
1061 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1062 content_ptr, NULL, 0);
1065 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1066 const char *method, const char *url,
1068 RTSPMessageHeader *reply,
1069 unsigned char **content_ptr,
1070 const unsigned char *send_content,
1071 int send_content_length)
1073 RTSPState *rt = s->priv_data;
1074 HTTPAuthType cur_auth_type;
1078 cur_auth_type = rt->auth_state.auth_type;
1079 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1081 send_content_length)))
1084 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1087 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1088 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1091 if (reply->status_code > 400){
1092 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1096 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1102 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1103 int lower_transport, const char *real_challenge)
1105 RTSPState *rt = s->priv_data;
1106 int rtx, j, i, err, interleave = 0;
1107 RTSPStream *rtsp_st;
1108 RTSPMessageHeader reply1, *reply = &reply1;
1110 const char *trans_pref;
1112 if (rt->transport == RTSP_TRANSPORT_RDT)
1113 trans_pref = "x-pn-tng";
1115 trans_pref = "RTP/AVP";
1117 /* default timeout: 1 minute */
1120 /* for each stream, make the setup request */
1121 /* XXX: we assume the same server is used for the control of each
1124 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1125 char transport[2048];
1128 * WMS serves all UDP data over a single connection, the RTX, which
1129 * isn't necessarily the first in the SDP but has to be the first
1130 * to be set up, else the second/third SETUP will fail with a 461.
1132 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1133 rt->server_type == RTSP_SERVER_WMS) {
1136 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1137 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1139 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1143 if (rtx == rt->nb_rtsp_streams)
1144 return -1; /* no RTX found */
1145 rtsp_st = rt->rtsp_streams[rtx];
1147 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1149 rtsp_st = rt->rtsp_streams[i];
1152 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1155 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1156 port = reply->transports[0].client_port_min;
1160 /* first try in specified port range */
1161 if (RTSP_RTP_PORT_MIN != 0) {
1162 while (j <= RTSP_RTP_PORT_MAX) {
1163 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1164 "?localport=%d", j);
1165 /* we will use two ports per rtp stream (rtp and rtcp) */
1167 if (ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1168 &s->interrupt_callback, NULL) == 0)
1173 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1178 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1180 snprintf(transport, sizeof(transport) - 1,
1181 "%s/UDP;", trans_pref);
1182 if (rt->server_type != RTSP_SERVER_REAL)
1183 av_strlcat(transport, "unicast;", sizeof(transport));
1184 av_strlcatf(transport, sizeof(transport),
1185 "client_port=%d", port);
1186 if (rt->transport == RTSP_TRANSPORT_RTP &&
1187 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1188 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1192 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1193 /* For WMS streams, the application streams are only used for
1194 * UDP. When trying to set it up for TCP streams, the server
1195 * will return an error. Therefore, we skip those streams. */
1196 if (rt->server_type == RTSP_SERVER_WMS &&
1197 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1200 snprintf(transport, sizeof(transport) - 1,
1201 "%s/TCP;", trans_pref);
1202 if (rt->transport != RTSP_TRANSPORT_RDT)
1203 av_strlcat(transport, "unicast;", sizeof(transport));
1204 av_strlcatf(transport, sizeof(transport),
1205 "interleaved=%d-%d",
1206 interleave, interleave + 1);
1210 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1211 snprintf(transport, sizeof(transport) - 1,
1212 "%s/UDP;multicast", trans_pref);
1215 av_strlcat(transport, ";mode=receive", sizeof(transport));
1216 } else if (rt->server_type == RTSP_SERVER_REAL ||
1217 rt->server_type == RTSP_SERVER_WMS)
1218 av_strlcat(transport, ";mode=play", sizeof(transport));
1219 snprintf(cmd, sizeof(cmd),
1220 "Transport: %s\r\n",
1222 if (rt->accept_dynamic_rate)
1223 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1224 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1225 char real_res[41], real_csum[9];
1226 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1228 av_strlcatf(cmd, sizeof(cmd),
1230 "RealChallenge2: %s, sd=%s\r\n",
1231 rt->session_id, real_res, real_csum);
1233 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1234 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1237 } else if (reply->status_code != RTSP_STATUS_OK ||
1238 reply->nb_transports != 1) {
1239 err = AVERROR_INVALIDDATA;
1243 /* XXX: same protocol for all streams is required */
1245 if (reply->transports[0].lower_transport != rt->lower_transport ||
1246 reply->transports[0].transport != rt->transport) {
1247 err = AVERROR_INVALIDDATA;
1251 rt->lower_transport = reply->transports[0].lower_transport;
1252 rt->transport = reply->transports[0].transport;
1255 /* Fail if the server responded with another lower transport mode
1256 * than what we requested. */
1257 if (reply->transports[0].lower_transport != lower_transport) {
1258 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1259 err = AVERROR_INVALIDDATA;
1263 switch(reply->transports[0].lower_transport) {
1264 case RTSP_LOWER_TRANSPORT_TCP:
1265 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1266 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1269 case RTSP_LOWER_TRANSPORT_UDP: {
1270 char url[1024], options[30] = "";
1272 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1273 av_strlcpy(options, "?connect=1", sizeof(options));
1274 /* Use source address if specified */
1275 if (reply->transports[0].source[0]) {
1276 ff_url_join(url, sizeof(url), "rtp", NULL,
1277 reply->transports[0].source,
1278 reply->transports[0].server_port_min, "%s", options);
1280 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1281 reply->transports[0].server_port_min, "%s", options);
1283 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1284 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1285 err = AVERROR_INVALIDDATA;
1288 /* Try to initialize the connection state in a
1289 * potential NAT router by sending dummy packets.
1290 * RTP/RTCP dummy packets are used for RDT, too.
1292 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1294 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1297 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1298 char url[1024], namebuf[50];
1299 struct sockaddr_storage addr;
1302 if (reply->transports[0].destination.ss_family) {
1303 addr = reply->transports[0].destination;
1304 port = reply->transports[0].port_min;
1305 ttl = reply->transports[0].ttl;
1307 addr = rtsp_st->sdp_ip;
1308 port = rtsp_st->sdp_port;
1309 ttl = rtsp_st->sdp_ttl;
1311 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1312 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1313 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1314 port, "?ttl=%d", ttl);
1315 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1316 &s->interrupt_callback, NULL) < 0) {
1317 err = AVERROR_INVALIDDATA;
1324 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1328 if (reply->timeout > 0)
1329 rt->timeout = reply->timeout;
1331 if (rt->server_type == RTSP_SERVER_REAL)
1332 rt->need_subscription = 1;
1337 ff_rtsp_undo_setup(s);
1341 void ff_rtsp_close_connections(AVFormatContext *s)
1343 RTSPState *rt = s->priv_data;
1344 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1345 ffurl_close(rt->rtsp_hd);
1346 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1349 int ff_rtsp_connect(AVFormatContext *s)
1351 RTSPState *rt = s->priv_data;
1352 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1353 char *option_list, *option, *filename;
1354 int port, err, tcp_fd;
1355 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1356 int lower_transport_mask = 0;
1357 char real_challenge[64] = "";
1358 struct sockaddr_storage peer;
1359 socklen_t peer_len = sizeof(peer);
1361 if (!ff_network_init())
1362 return AVERROR(EIO);
1364 rt->control_transport = RTSP_MODE_PLAIN;
1365 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1366 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1367 rt->control_transport = RTSP_MODE_TUNNEL;
1369 /* Only pass through valid flags from here */
1370 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1373 lower_transport_mask = rt->lower_transport_mask;
1374 /* extract hostname and port */
1375 av_url_split(NULL, 0, auth, sizeof(auth),
1376 host, sizeof(host), &port, path, sizeof(path), s->filename);
1378 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1381 port = RTSP_DEFAULT_PORT;
1383 #if FF_API_RTSP_URL_OPTIONS
1384 /* search for options */
1385 option_list = strrchr(path, '?');
1387 /* Strip out the RTSP specific options, write out the rest of
1388 * the options back into the same string. */
1389 filename = option_list;
1390 while (option_list) {
1392 /* move the option pointer */
1393 option = ++option_list;
1394 option_list = strchr(option_list, '&');
1398 /* handle the options */
1399 if (!strcmp(option, "udp")) {
1400 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1401 } else if (!strcmp(option, "multicast")) {
1402 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1403 } else if (!strcmp(option, "tcp")) {
1404 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1405 } else if(!strcmp(option, "http")) {
1406 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1407 rt->control_transport = RTSP_MODE_TUNNEL;
1408 } else if (!strcmp(option, "filter_src")) {
1409 rt->rtsp_flags |= RTSP_FLAG_FILTER_SRC;
1411 /* Write options back into the buffer, using memmove instead
1412 * of strcpy since the strings may overlap. */
1413 int len = strlen(option);
1414 memmove(++filename, option, len);
1416 if (option_list) *filename = '&';
1420 av_log(s, AV_LOG_WARNING, "Options passed via URL are "
1421 "deprecated, use -rtsp_transport "
1422 "and -rtsp_flags instead.\n");
1428 if (!lower_transport_mask)
1429 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1432 /* Only UDP or TCP - UDP multicast isn't supported. */
1433 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1434 (1 << RTSP_LOWER_TRANSPORT_TCP);
1435 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1436 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1437 "only UDP and TCP are supported for output.\n");
1438 err = AVERROR(EINVAL);
1443 /* Construct the URI used in request; this is similar to s->filename,
1444 * but with authentication credentials removed and RTSP specific options
1446 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1447 host, port, "%s", path);
1449 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1450 /* set up initial handshake for tunneling */
1451 char httpname[1024];
1452 char sessioncookie[17];
1455 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1456 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1457 av_get_random_seed(), av_get_random_seed());
1460 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1461 &s->interrupt_callback) < 0) {
1466 /* generate GET headers */
1467 snprintf(headers, sizeof(headers),
1468 "x-sessioncookie: %s\r\n"
1469 "Accept: application/x-rtsp-tunnelled\r\n"
1470 "Pragma: no-cache\r\n"
1471 "Cache-Control: no-cache\r\n",
1473 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1475 /* complete the connection */
1476 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1482 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1483 &s->interrupt_callback) < 0 ) {
1488 /* generate POST headers */
1489 snprintf(headers, sizeof(headers),
1490 "x-sessioncookie: %s\r\n"
1491 "Content-Type: application/x-rtsp-tunnelled\r\n"
1492 "Pragma: no-cache\r\n"
1493 "Cache-Control: no-cache\r\n"
1494 "Content-Length: 32767\r\n"
1495 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1497 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1498 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1500 /* Initialize the authentication state for the POST session. The HTTP
1501 * protocol implementation doesn't properly handle multi-pass
1502 * authentication for POST requests, since it would require one of
1504 * - implementing Expect: 100-continue, which many HTTP servers
1505 * don't support anyway, even less the RTSP servers that do HTTP
1507 * - sending the whole POST data until getting a 401 reply specifying
1508 * what authentication method to use, then resending all that data
1509 * - waiting for potential 401 replies directly after sending the
1510 * POST header (waiting for some unspecified time)
1511 * Therefore, we copy the full auth state, which works for both basic
1512 * and digest. (For digest, we would have to synchronize the nonce
1513 * count variable between the two sessions, if we'd do more requests
1514 * with the original session, though.)
1516 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1518 /* complete the connection */
1519 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1524 /* open the tcp connection */
1525 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1526 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1527 &s->interrupt_callback, NULL) < 0) {
1531 rt->rtsp_hd_out = rt->rtsp_hd;
1535 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1536 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1537 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1538 NULL, 0, NI_NUMERICHOST);
1541 /* request options supported by the server; this also detects server
1543 for (rt->server_type = RTSP_SERVER_RTP;;) {
1545 if (rt->server_type == RTSP_SERVER_REAL)
1548 * The following entries are required for proper
1549 * streaming from a Realmedia server. They are
1550 * interdependent in some way although we currently
1551 * don't quite understand how. Values were copied
1552 * from mplayer SVN r23589.
1553 * ClientChallenge is a 16-byte ID in hex
1554 * CompanyID is a 16-byte ID in base64
1556 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1557 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1558 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1559 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1561 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1562 if (reply->status_code != RTSP_STATUS_OK) {
1563 err = AVERROR_INVALIDDATA;
1567 /* detect server type if not standard-compliant RTP */
1568 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1569 rt->server_type = RTSP_SERVER_REAL;
1571 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1572 rt->server_type = RTSP_SERVER_WMS;
1573 } else if (rt->server_type == RTSP_SERVER_REAL)
1574 strcpy(real_challenge, reply->real_challenge);
1578 if (s->iformat && CONFIG_RTSP_DEMUXER)
1579 err = ff_rtsp_setup_input_streams(s, reply);
1580 else if (CONFIG_RTSP_MUXER)
1581 err = ff_rtsp_setup_output_streams(s, host);
1586 int lower_transport = ff_log2_tab[lower_transport_mask &
1587 ~(lower_transport_mask - 1)];
1589 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1590 rt->server_type == RTSP_SERVER_REAL ?
1591 real_challenge : NULL);
1594 lower_transport_mask &= ~(1 << lower_transport);
1595 if (lower_transport_mask == 0 && err == 1) {
1596 err = AVERROR(EPROTONOSUPPORT);
1601 rt->lower_transport_mask = lower_transport_mask;
1602 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1603 rt->state = RTSP_STATE_IDLE;
1604 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1607 ff_rtsp_close_streams(s);
1608 ff_rtsp_close_connections(s);
1609 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1610 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1611 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1619 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1622 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1623 uint8_t *buf, int buf_size, int64_t wait_end)
1625 RTSPState *rt = s->priv_data;
1626 RTSPStream *rtsp_st;
1627 int n, i, ret, tcp_fd, timeout_cnt = 0;
1629 struct pollfd *p = rt->p;
1632 if (ff_check_interrupt(&s->interrupt_callback))
1633 return AVERROR_EXIT;
1634 if (wait_end && wait_end - av_gettime() < 0)
1635 return AVERROR(EAGAIN);
1638 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1639 p[max_p].fd = tcp_fd;
1640 p[max_p++].events = POLLIN;
1644 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1645 rtsp_st = rt->rtsp_streams[i];
1646 if (rtsp_st->rtp_handle) {
1647 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1648 p[max_p++].events = POLLIN;
1649 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1650 p[max_p++].events = POLLIN;
1653 n = poll(p, max_p, POLL_TIMEOUT_MS);
1655 int j = 1 - (tcp_fd == -1);
1657 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1658 rtsp_st = rt->rtsp_streams[i];
1659 if (rtsp_st->rtp_handle) {
1660 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1661 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1663 *prtsp_st = rtsp_st;
1670 #if CONFIG_RTSP_DEMUXER
1671 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1672 RTSPMessageHeader reply;
1674 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1677 /* XXX: parse message */
1678 if (rt->state != RTSP_STATE_STREAMING)
1682 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1683 return AVERROR(ETIMEDOUT);
1684 } else if (n < 0 && errno != EINTR)
1685 return AVERROR(errno);
1689 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1691 RTSPState *rt = s->priv_data;
1693 RTSPStream *rtsp_st, *first_queue_st = NULL;
1694 int64_t wait_end = 0;
1696 if (rt->nb_byes == rt->nb_rtsp_streams)
1699 /* get next frames from the same RTP packet */
1700 if (rt->cur_transport_priv) {
1701 if (rt->transport == RTSP_TRANSPORT_RDT) {
1702 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1704 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1706 rt->cur_transport_priv = NULL;
1708 } else if (ret == 1) {
1711 rt->cur_transport_priv = NULL;
1714 if (rt->transport == RTSP_TRANSPORT_RTP) {
1716 int64_t first_queue_time = 0;
1717 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1718 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1722 queue_time = ff_rtp_queued_packet_time(rtpctx);
1723 if (queue_time && (queue_time - first_queue_time < 0 ||
1724 !first_queue_time)) {
1725 first_queue_time = queue_time;
1726 first_queue_st = rt->rtsp_streams[i];
1729 if (first_queue_time)
1730 wait_end = first_queue_time + s->max_delay;
1733 /* read next RTP packet */
1736 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1738 return AVERROR(ENOMEM);
1741 switch(rt->lower_transport) {
1743 #if CONFIG_RTSP_DEMUXER
1744 case RTSP_LOWER_TRANSPORT_TCP:
1745 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1748 case RTSP_LOWER_TRANSPORT_UDP:
1749 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1750 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1751 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1752 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1755 if (len == AVERROR(EAGAIN) && first_queue_st &&
1756 rt->transport == RTSP_TRANSPORT_RTP) {
1757 rtsp_st = first_queue_st;
1758 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1765 if (rt->transport == RTSP_TRANSPORT_RDT) {
1766 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1768 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1770 /* Either bad packet, or a RTCP packet. Check if the
1771 * first_rtcp_ntp_time field was initialized. */
1772 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1773 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1774 /* first_rtcp_ntp_time has been initialized for this stream,
1775 * copy the same value to all other uninitialized streams,
1776 * in order to map their timestamp origin to the same ntp time
1779 AVStream *st = NULL;
1780 if (rtsp_st->stream_index >= 0)
1781 st = s->streams[rtsp_st->stream_index];
1782 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1783 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1784 AVStream *st2 = NULL;
1785 if (rt->rtsp_streams[i]->stream_index >= 0)
1786 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1787 if (rtpctx2 && st && st2 &&
1788 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1789 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1790 rtpctx2->rtcp_ts_offset = av_rescale_q(
1791 rtpctx->rtcp_ts_offset, st->time_base,
1796 if (ret == -RTCP_BYE) {
1799 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1800 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1802 if (rt->nb_byes == rt->nb_rtsp_streams)
1811 /* more packets may follow, so we save the RTP context */
1812 rt->cur_transport_priv = rtsp_st->transport_priv;
1816 #endif /* CONFIG_RTPDEC */
1818 #if CONFIG_SDP_DEMUXER
1819 static int sdp_probe(AVProbeData *p1)
1821 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1823 /* we look for a line beginning "c=IN IP" */
1824 while (p < p_end && *p != '\0') {
1825 if (p + sizeof("c=IN IP") - 1 < p_end &&
1826 av_strstart(p, "c=IN IP", NULL))
1827 return AVPROBE_SCORE_MAX / 2;
1829 while (p < p_end - 1 && *p != '\n') p++;
1838 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1840 RTSPState *rt = s->priv_data;
1841 RTSPStream *rtsp_st;
1846 if (!ff_network_init())
1847 return AVERROR(EIO);
1849 /* read the whole sdp file */
1850 /* XXX: better loading */
1851 content = av_malloc(SDP_MAX_SIZE);
1852 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1855 return AVERROR_INVALIDDATA;
1857 content[size] ='\0';
1859 err = ff_sdp_parse(s, content);
1863 /* open each RTP stream */
1864 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1866 rtsp_st = rt->rtsp_streams[i];
1868 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1869 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1870 ff_url_join(url, sizeof(url), "rtp", NULL,
1871 namebuf, rtsp_st->sdp_port,
1872 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1874 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1875 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1876 &s->interrupt_callback, NULL) < 0) {
1877 err = AVERROR_INVALIDDATA;
1880 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1885 ff_rtsp_close_streams(s);
1890 static int sdp_read_close(AVFormatContext *s)
1892 ff_rtsp_close_streams(s);
1897 static const AVClass sdp_demuxer_class = {
1898 .class_name = "SDP demuxer",
1899 .item_name = av_default_item_name,
1900 .option = sdp_options,
1901 .version = LIBAVUTIL_VERSION_INT,
1904 AVInputFormat ff_sdp_demuxer = {
1906 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1907 .priv_data_size = sizeof(RTSPState),
1908 .read_probe = sdp_probe,
1909 .read_header = sdp_read_header,
1910 .read_packet = ff_rtsp_fetch_packet,
1911 .read_close = sdp_read_close,
1912 .priv_class = &sdp_demuxer_class
1914 #endif /* CONFIG_SDP_DEMUXER */
1916 #if CONFIG_RTP_DEMUXER
1917 static int rtp_probe(AVProbeData *p)
1919 if (av_strstart(p->filename, "rtp:", NULL))
1920 return AVPROBE_SCORE_MAX;
1924 static int rtp_read_header(AVFormatContext *s,
1925 AVFormatParameters *ap)
1927 uint8_t recvbuf[1500];
1928 char host[500], sdp[500];
1930 URLContext* in = NULL;
1932 AVCodecContext codec;
1933 struct sockaddr_storage addr;
1935 socklen_t addrlen = sizeof(addr);
1936 RTSPState *rt = s->priv_data;
1938 if (!ff_network_init())
1939 return AVERROR(EIO);
1941 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1942 &s->interrupt_callback, NULL);
1947 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1948 if (ret == AVERROR(EAGAIN))
1953 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1957 if ((recvbuf[0] & 0xc0) != 0x80) {
1958 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1963 payload_type = recvbuf[1] & 0x7f;
1966 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1970 memset(&codec, 0, sizeof(codec));
1971 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1972 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1973 "without an SDP file describing it\n",
1977 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1978 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1979 "properly you need an SDP file "
1983 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1984 NULL, 0, s->filename);
1986 snprintf(sdp, sizeof(sdp),
1987 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1988 addr.ss_family == AF_INET ? 4 : 6, host,
1989 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1990 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1991 port, payload_type);
1992 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1994 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1997 /* sdp_read_header initializes this again */
2000 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2002 ret = sdp_read_header(s, ap);
2013 static const AVClass rtp_demuxer_class = {
2014 .class_name = "RTP demuxer",
2015 .item_name = av_default_item_name,
2016 .option = rtp_options,
2017 .version = LIBAVUTIL_VERSION_INT,
2020 AVInputFormat ff_rtp_demuxer = {
2022 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2023 .priv_data_size = sizeof(RTSPState),
2024 .read_probe = rtp_probe,
2025 .read_header = rtp_read_header,
2026 .read_packet = ff_rtsp_fetch_packet,
2027 .read_close = sdp_read_close,
2028 .flags = AVFMT_NOFILE,
2029 .priv_class = &rtp_demuxer_class
2031 #endif /* CONFIG_RTP_DEMUXER */