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 av_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 av_set_pts_info(st, 32, 1, i);
257 /* parse the attribute line from the fmtp a line of an sdp response. This
258 * is broken out as a function because it is used in rtp_h264.c, which is
260 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
261 char *value, int value_size)
263 *p += strspn(*p, SPACE_CHARS);
265 get_word_sep(attr, attr_size, "=", p);
268 get_word_sep(value, value_size, ";", p);
276 typedef struct SDPParseState {
278 struct sockaddr_storage default_ip;
280 int skip_media; ///< set if an unknown m= line occurs
283 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
284 int letter, const char *buf)
286 RTSPState *rt = s->priv_data;
287 char buf1[64], st_type[64];
289 enum AVMediaType codec_type;
293 struct sockaddr_storage sdp_ip;
296 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
299 if (s1->skip_media && letter != 'm')
303 get_word(buf1, sizeof(buf1), &p);
304 if (strcmp(buf1, "IN") != 0)
306 get_word(buf1, sizeof(buf1), &p);
307 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
309 get_word_sep(buf1, sizeof(buf1), "/", &p);
310 if (get_sockaddr(buf1, &sdp_ip))
315 get_word_sep(buf1, sizeof(buf1), "/", &p);
318 if (s->nb_streams == 0) {
319 s1->default_ip = sdp_ip;
320 s1->default_ttl = ttl;
322 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
323 rtsp_st->sdp_ip = sdp_ip;
324 rtsp_st->sdp_ttl = ttl;
328 av_dict_set(&s->metadata, "title", p, 0);
331 if (s->nb_streams == 0) {
332 av_dict_set(&s->metadata, "comment", p, 0);
339 codec_type = AVMEDIA_TYPE_UNKNOWN;
340 get_word(st_type, sizeof(st_type), &p);
341 if (!strcmp(st_type, "audio")) {
342 codec_type = AVMEDIA_TYPE_AUDIO;
343 } else if (!strcmp(st_type, "video")) {
344 codec_type = AVMEDIA_TYPE_VIDEO;
345 } else if (!strcmp(st_type, "application")) {
346 codec_type = AVMEDIA_TYPE_DATA;
348 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
352 rtsp_st = av_mallocz(sizeof(RTSPStream));
355 rtsp_st->stream_index = -1;
356 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
358 rtsp_st->sdp_ip = s1->default_ip;
359 rtsp_st->sdp_ttl = s1->default_ttl;
361 get_word(buf1, sizeof(buf1), &p); /* port */
362 rtsp_st->sdp_port = atoi(buf1);
364 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
366 /* XXX: handle list of formats */
367 get_word(buf1, sizeof(buf1), &p); /* format list */
368 rtsp_st->sdp_payload_type = atoi(buf1);
370 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
371 /* no corresponding stream */
373 st = avformat_new_stream(s, NULL);
376 st->id = rt->nb_rtsp_streams - 1;
377 rtsp_st->stream_index = st->index;
378 st->codec->codec_type = codec_type;
379 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
380 RTPDynamicProtocolHandler *handler;
381 /* if standard payload type, we can find the codec right now */
382 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
383 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
384 st->codec->sample_rate > 0)
385 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
386 /* Even static payload types may need a custom depacketizer */
387 handler = ff_rtp_handler_find_by_id(
388 rtsp_st->sdp_payload_type, st->codec->codec_type);
389 init_rtp_handler(handler, rtsp_st, st->codec);
392 /* put a default control url */
393 av_strlcpy(rtsp_st->control_url, rt->control_uri,
394 sizeof(rtsp_st->control_url));
397 if (av_strstart(p, "control:", &p)) {
398 if (s->nb_streams == 0) {
399 if (!strncmp(p, "rtsp://", 7))
400 av_strlcpy(rt->control_uri, p,
401 sizeof(rt->control_uri));
404 /* get the control url */
405 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
407 /* XXX: may need to add full url resolution */
408 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
410 if (proto[0] == '\0') {
411 /* relative control URL */
412 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
413 av_strlcat(rtsp_st->control_url, "/",
414 sizeof(rtsp_st->control_url));
415 av_strlcat(rtsp_st->control_url, p,
416 sizeof(rtsp_st->control_url));
418 av_strlcpy(rtsp_st->control_url, p,
419 sizeof(rtsp_st->control_url));
421 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
422 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
423 get_word(buf1, sizeof(buf1), &p);
424 payload_type = atoi(buf1);
425 st = s->streams[s->nb_streams - 1];
426 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
427 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
428 } else if (av_strstart(p, "fmtp:", &p) ||
429 av_strstart(p, "framesize:", &p)) {
430 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
431 // let dynamic protocol handlers have a stab at the line.
432 get_word(buf1, sizeof(buf1), &p);
433 payload_type = atoi(buf1);
434 for (i = 0; i < rt->nb_rtsp_streams; i++) {
435 rtsp_st = rt->rtsp_streams[i];
436 if (rtsp_st->sdp_payload_type == payload_type &&
437 rtsp_st->dynamic_handler &&
438 rtsp_st->dynamic_handler->parse_sdp_a_line)
439 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
440 rtsp_st->dynamic_protocol_context, buf);
442 } else if (av_strstart(p, "range:", &p)) {
445 // this is so that seeking on a streamed file can work.
446 rtsp_parse_range_npt(p, &start, &end);
447 s->start_time = start;
448 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
449 s->duration = (end == AV_NOPTS_VALUE) ?
450 AV_NOPTS_VALUE : end - start;
451 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
453 rt->transport = RTSP_TRANSPORT_RDT;
454 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
456 st = s->streams[s->nb_streams - 1];
457 st->codec->sample_rate = atoi(p);
459 if (rt->server_type == RTSP_SERVER_WMS)
460 ff_wms_parse_sdp_a_line(s, p);
461 if (s->nb_streams > 0) {
462 if (rt->server_type == RTSP_SERVER_REAL)
463 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
465 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
466 if (rtsp_st->dynamic_handler &&
467 rtsp_st->dynamic_handler->parse_sdp_a_line)
468 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
470 rtsp_st->dynamic_protocol_context, buf);
477 int ff_sdp_parse(AVFormatContext *s, const char *content)
479 RTSPState *rt = s->priv_data;
482 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
483 * contain long SDP lines containing complete ASF Headers (several
484 * kB) or arrays of MDPR (RM stream descriptor) headers plus
485 * "rulebooks" describing their properties. Therefore, the SDP line
488 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
489 * in rtpdec_xiph.c. */
491 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
493 memset(s1, 0, sizeof(SDPParseState));
496 p += strspn(p, SPACE_CHARS);
504 /* get the content */
506 while (*p != '\n' && *p != '\r' && *p != '\0') {
507 if ((q - buf) < sizeof(buf) - 1)
512 sdp_parse_line(s, s1, letter, buf);
514 while (*p != '\n' && *p != '\0')
519 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
520 if (!rt->p) return AVERROR(ENOMEM);
523 #endif /* CONFIG_RTPDEC */
525 void ff_rtsp_undo_setup(AVFormatContext *s)
527 RTSPState *rt = s->priv_data;
530 for (i = 0; i < rt->nb_rtsp_streams; i++) {
531 RTSPStream *rtsp_st = rt->rtsp_streams[i];
534 if (rtsp_st->transport_priv) {
536 AVFormatContext *rtpctx = rtsp_st->transport_priv;
537 av_write_trailer(rtpctx);
538 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
540 avio_close_dyn_buf(rtpctx->pb, &ptr);
543 avio_close(rtpctx->pb);
545 avformat_free_context(rtpctx);
546 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
547 ff_rdt_parse_close(rtsp_st->transport_priv);
548 else if (CONFIG_RTPDEC)
549 ff_rtp_parse_close(rtsp_st->transport_priv);
551 rtsp_st->transport_priv = NULL;
552 if (rtsp_st->rtp_handle)
553 ffurl_close(rtsp_st->rtp_handle);
554 rtsp_st->rtp_handle = NULL;
558 /* close and free RTSP streams */
559 void ff_rtsp_close_streams(AVFormatContext *s)
561 RTSPState *rt = s->priv_data;
565 ff_rtsp_undo_setup(s);
566 for (i = 0; i < rt->nb_rtsp_streams; i++) {
567 rtsp_st = rt->rtsp_streams[i];
569 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
570 rtsp_st->dynamic_handler->free(
571 rtsp_st->dynamic_protocol_context);
575 av_free(rt->rtsp_streams);
577 av_close_input_stream (rt->asf_ctx);
581 av_free(rt->recvbuf);
584 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
586 RTSPState *rt = s->priv_data;
589 /* open the RTP context */
590 if (rtsp_st->stream_index >= 0)
591 st = s->streams[rtsp_st->stream_index];
593 s->ctx_flags |= AVFMTCTX_NOHEADER;
595 if (s->oformat && CONFIG_RTSP_MUXER) {
596 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
598 RTSP_TCP_MAX_PACKET_SIZE);
599 /* Ownership of rtp_handle is passed to the rtp mux context */
600 rtsp_st->rtp_handle = NULL;
601 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
602 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
603 rtsp_st->dynamic_protocol_context,
604 rtsp_st->dynamic_handler);
605 else if (CONFIG_RTPDEC)
606 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
607 rtsp_st->sdp_payload_type,
608 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
609 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
611 if (!rtsp_st->transport_priv) {
612 return AVERROR(ENOMEM);
613 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
614 if (rtsp_st->dynamic_handler) {
615 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
616 rtsp_st->dynamic_protocol_context,
617 rtsp_st->dynamic_handler);
624 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
625 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
631 p += strspn(p, SPACE_CHARS);
632 v = strtol(p, (char **)&p, 10);
636 v = strtol(p, (char **)&p, 10);
645 /* XXX: only one transport specification is parsed */
646 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
648 char transport_protocol[16];
650 char lower_transport[16];
652 RTSPTransportField *th;
655 reply->nb_transports = 0;
658 p += strspn(p, SPACE_CHARS);
662 th = &reply->transports[reply->nb_transports];
664 get_word_sep(transport_protocol, sizeof(transport_protocol),
666 if (!av_strcasecmp (transport_protocol, "rtp")) {
667 get_word_sep(profile, sizeof(profile), "/;,", &p);
668 lower_transport[0] = '\0';
669 /* rtp/avp/<protocol> */
671 get_word_sep(lower_transport, sizeof(lower_transport),
674 th->transport = RTSP_TRANSPORT_RTP;
675 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
676 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
677 /* x-pn-tng/<protocol> */
678 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
680 th->transport = RTSP_TRANSPORT_RDT;
682 if (!av_strcasecmp(lower_transport, "TCP"))
683 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
685 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
689 /* get each parameter */
690 while (*p != '\0' && *p != ',') {
691 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
692 if (!strcmp(parameter, "port")) {
695 rtsp_parse_range(&th->port_min, &th->port_max, &p);
697 } else if (!strcmp(parameter, "client_port")) {
700 rtsp_parse_range(&th->client_port_min,
701 &th->client_port_max, &p);
703 } else if (!strcmp(parameter, "server_port")) {
706 rtsp_parse_range(&th->server_port_min,
707 &th->server_port_max, &p);
709 } else if (!strcmp(parameter, "interleaved")) {
712 rtsp_parse_range(&th->interleaved_min,
713 &th->interleaved_max, &p);
715 } else if (!strcmp(parameter, "multicast")) {
716 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
717 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
718 } else if (!strcmp(parameter, "ttl")) {
721 th->ttl = strtol(p, (char **)&p, 10);
723 } else if (!strcmp(parameter, "destination")) {
726 get_word_sep(buf, sizeof(buf), ";,", &p);
727 get_sockaddr(buf, &th->destination);
729 } else if (!strcmp(parameter, "source")) {
732 get_word_sep(buf, sizeof(buf), ";,", &p);
733 av_strlcpy(th->source, buf, sizeof(th->source));
737 while (*p != ';' && *p != '\0' && *p != ',')
745 reply->nb_transports++;
749 static void handle_rtp_info(RTSPState *rt, const char *url,
750 uint32_t seq, uint32_t rtptime)
753 if (!rtptime || !url[0])
755 if (rt->transport != RTSP_TRANSPORT_RTP)
757 for (i = 0; i < rt->nb_rtsp_streams; i++) {
758 RTSPStream *rtsp_st = rt->rtsp_streams[i];
759 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
762 if (!strcmp(rtsp_st->control_url, url)) {
763 rtpctx->base_timestamp = rtptime;
769 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
772 char key[20], value[1024], url[1024] = "";
773 uint32_t seq = 0, rtptime = 0;
776 p += strspn(p, SPACE_CHARS);
779 get_word_sep(key, sizeof(key), "=", &p);
783 get_word_sep(value, sizeof(value), ";, ", &p);
785 if (!strcmp(key, "url"))
786 av_strlcpy(url, value, sizeof(url));
787 else if (!strcmp(key, "seq"))
788 seq = strtoul(value, NULL, 10);
789 else if (!strcmp(key, "rtptime"))
790 rtptime = strtoul(value, NULL, 10);
792 handle_rtp_info(rt, url, seq, rtptime);
801 handle_rtp_info(rt, url, seq, rtptime);
804 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
805 RTSPState *rt, const char *method)
809 /* NOTE: we do case independent match for broken servers */
811 if (av_stristart(p, "Session:", &p)) {
813 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
814 if (av_stristart(p, ";timeout=", &p) &&
815 (t = strtol(p, NULL, 10)) > 0) {
818 } else if (av_stristart(p, "Content-Length:", &p)) {
819 reply->content_length = strtol(p, NULL, 10);
820 } else if (av_stristart(p, "Transport:", &p)) {
821 rtsp_parse_transport(reply, p);
822 } else if (av_stristart(p, "CSeq:", &p)) {
823 reply->seq = strtol(p, NULL, 10);
824 } else if (av_stristart(p, "Range:", &p)) {
825 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
826 } else if (av_stristart(p, "RealChallenge1:", &p)) {
827 p += strspn(p, SPACE_CHARS);
828 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
829 } else if (av_stristart(p, "Server:", &p)) {
830 p += strspn(p, SPACE_CHARS);
831 av_strlcpy(reply->server, p, sizeof(reply->server));
832 } else if (av_stristart(p, "Notice:", &p) ||
833 av_stristart(p, "X-Notice:", &p)) {
834 reply->notice = strtol(p, NULL, 10);
835 } else if (av_stristart(p, "Location:", &p)) {
836 p += strspn(p, SPACE_CHARS);
837 av_strlcpy(reply->location, p , sizeof(reply->location));
838 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
839 p += strspn(p, SPACE_CHARS);
840 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
841 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
842 p += strspn(p, SPACE_CHARS);
843 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
844 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
845 p += strspn(p, SPACE_CHARS);
846 if (method && !strcmp(method, "DESCRIBE"))
847 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
848 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
849 p += strspn(p, SPACE_CHARS);
850 if (method && !strcmp(method, "PLAY"))
851 rtsp_parse_rtp_info(rt, p);
852 } else if (av_stristart(p, "Public:", &p) && rt) {
853 if (strstr(p, "GET_PARAMETER") &&
854 method && !strcmp(method, "OPTIONS"))
855 rt->get_parameter_supported = 1;
856 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
857 p += strspn(p, SPACE_CHARS);
858 rt->accept_dynamic_rate = atoi(p);
862 /* skip a RTP/TCP interleaved packet */
863 void ff_rtsp_skip_packet(AVFormatContext *s)
865 RTSPState *rt = s->priv_data;
869 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
872 len = AV_RB16(buf + 1);
874 av_dlog(s, "skipping RTP packet len=%d\n", len);
879 if (len1 > sizeof(buf))
881 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
888 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
889 unsigned char **content_ptr,
890 int return_on_interleaved_data, const char *method)
892 RTSPState *rt = s->priv_data;
893 char buf[4096], buf1[1024], *q;
896 int ret, content_length, line_count = 0;
897 unsigned char *content = NULL;
899 memset(reply, 0, sizeof(*reply));
901 /* parse reply (XXX: use buffers) */
902 rt->last_reply[0] = '\0';
906 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
907 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
913 /* XXX: only parse it if first char on line ? */
914 if (return_on_interleaved_data) {
917 ff_rtsp_skip_packet(s);
918 } else if (ch != '\r') {
919 if ((q - buf) < sizeof(buf) - 1)
925 av_dlog(s, "line='%s'\n", buf);
927 /* test if last line */
931 if (line_count == 0) {
933 get_word(buf1, sizeof(buf1), &p);
934 get_word(buf1, sizeof(buf1), &p);
935 reply->status_code = atoi(buf1);
936 av_strlcpy(reply->reason, p, sizeof(reply->reason));
938 ff_rtsp_parse_line(reply, p, rt, method);
939 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
940 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
945 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
946 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
948 content_length = reply->content_length;
949 if (content_length > 0) {
950 /* leave some room for a trailing '\0' (useful for simple parsing) */
951 content = av_malloc(content_length + 1);
952 ffurl_read_complete(rt->rtsp_hd, content, content_length);
953 content[content_length] = '\0';
956 *content_ptr = content;
960 if (rt->seq != reply->seq) {
961 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
962 rt->seq, reply->seq);
966 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
967 reply->notice == 2104 /* Start-of-Stream Reached */ ||
968 reply->notice == 2306 /* Continuous Feed Terminated */) {
969 rt->state = RTSP_STATE_IDLE;
970 } else if (reply->notice >= 4400 && reply->notice < 5500) {
971 return AVERROR(EIO); /* data or server error */
972 } else if (reply->notice == 2401 /* Ticket Expired */ ||
973 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
974 return AVERROR(EPERM);
980 * Send a command to the RTSP server without waiting for the reply.
982 * @param s RTSP (de)muxer context
983 * @param method the method for the request
984 * @param url the target url for the request
985 * @param headers extra header lines to include in the request
986 * @param send_content if non-null, the data to send as request body content
987 * @param send_content_length the length of the send_content data, or 0 if
988 * send_content is null
990 * @return zero if success, nonzero otherwise
992 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
993 const char *method, const char *url,
995 const unsigned char *send_content,
996 int send_content_length)
998 RTSPState *rt = s->priv_data;
999 char buf[4096], *out_buf;
1000 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1002 /* Add in RTSP headers */
1005 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1007 av_strlcat(buf, headers, sizeof(buf));
1008 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1009 if (rt->session_id[0] != '\0' && (!headers ||
1010 !strstr(headers, "\nIf-Match:"))) {
1011 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1014 char *str = ff_http_auth_create_response(&rt->auth_state,
1015 rt->auth, url, method);
1017 av_strlcat(buf, str, sizeof(buf));
1020 if (send_content_length > 0 && send_content)
1021 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1022 av_strlcat(buf, "\r\n", sizeof(buf));
1024 /* base64 encode rtsp if tunneling */
1025 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1026 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1027 out_buf = base64buf;
1030 av_dlog(s, "Sending:\n%s--\n", buf);
1032 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1033 if (send_content_length > 0 && send_content) {
1034 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1035 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1036 "with content data not supported\n");
1037 return AVERROR_PATCHWELCOME;
1039 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1041 rt->last_cmd_time = av_gettime();
1046 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1047 const char *url, const char *headers)
1049 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1052 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1053 const char *headers, RTSPMessageHeader *reply,
1054 unsigned char **content_ptr)
1056 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1057 content_ptr, NULL, 0);
1060 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1061 const char *method, const char *url,
1063 RTSPMessageHeader *reply,
1064 unsigned char **content_ptr,
1065 const unsigned char *send_content,
1066 int send_content_length)
1068 RTSPState *rt = s->priv_data;
1069 HTTPAuthType cur_auth_type;
1073 cur_auth_type = rt->auth_state.auth_type;
1074 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1076 send_content_length)))
1079 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1082 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1083 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1086 if (reply->status_code > 400){
1087 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1091 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1097 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1098 int lower_transport, const char *real_challenge)
1100 RTSPState *rt = s->priv_data;
1101 int rtx, j, i, err, interleave = 0;
1102 RTSPStream *rtsp_st;
1103 RTSPMessageHeader reply1, *reply = &reply1;
1105 const char *trans_pref;
1107 if (rt->transport == RTSP_TRANSPORT_RDT)
1108 trans_pref = "x-pn-tng";
1110 trans_pref = "RTP/AVP";
1112 /* default timeout: 1 minute */
1115 /* for each stream, make the setup request */
1116 /* XXX: we assume the same server is used for the control of each
1119 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1120 char transport[2048];
1123 * WMS serves all UDP data over a single connection, the RTX, which
1124 * isn't necessarily the first in the SDP but has to be the first
1125 * to be set up, else the second/third SETUP will fail with a 461.
1127 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1128 rt->server_type == RTSP_SERVER_WMS) {
1131 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1132 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1134 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1138 if (rtx == rt->nb_rtsp_streams)
1139 return -1; /* no RTX found */
1140 rtsp_st = rt->rtsp_streams[rtx];
1142 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1144 rtsp_st = rt->rtsp_streams[i];
1147 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1150 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1151 port = reply->transports[0].client_port_min;
1155 /* first try in specified port range */
1156 if (RTSP_RTP_PORT_MIN != 0) {
1157 while (j <= RTSP_RTP_PORT_MAX) {
1158 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1159 "?localport=%d", j);
1160 /* we will use two ports per rtp stream (rtp and rtcp) */
1162 if (ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1163 &s->interrupt_callback, NULL) == 0)
1168 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1173 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1175 snprintf(transport, sizeof(transport) - 1,
1176 "%s/UDP;", trans_pref);
1177 if (rt->server_type != RTSP_SERVER_REAL)
1178 av_strlcat(transport, "unicast;", sizeof(transport));
1179 av_strlcatf(transport, sizeof(transport),
1180 "client_port=%d", port);
1181 if (rt->transport == RTSP_TRANSPORT_RTP &&
1182 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1183 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1187 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1188 /* For WMS streams, the application streams are only used for
1189 * UDP. When trying to set it up for TCP streams, the server
1190 * will return an error. Therefore, we skip those streams. */
1191 if (rt->server_type == RTSP_SERVER_WMS &&
1192 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1195 snprintf(transport, sizeof(transport) - 1,
1196 "%s/TCP;", trans_pref);
1197 if (rt->transport != RTSP_TRANSPORT_RDT)
1198 av_strlcat(transport, "unicast;", sizeof(transport));
1199 av_strlcatf(transport, sizeof(transport),
1200 "interleaved=%d-%d",
1201 interleave, interleave + 1);
1205 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1206 snprintf(transport, sizeof(transport) - 1,
1207 "%s/UDP;multicast", trans_pref);
1210 av_strlcat(transport, ";mode=receive", sizeof(transport));
1211 } else if (rt->server_type == RTSP_SERVER_REAL ||
1212 rt->server_type == RTSP_SERVER_WMS)
1213 av_strlcat(transport, ";mode=play", sizeof(transport));
1214 snprintf(cmd, sizeof(cmd),
1215 "Transport: %s\r\n",
1217 if (rt->accept_dynamic_rate)
1218 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1219 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1220 char real_res[41], real_csum[9];
1221 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1223 av_strlcatf(cmd, sizeof(cmd),
1225 "RealChallenge2: %s, sd=%s\r\n",
1226 rt->session_id, real_res, real_csum);
1228 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1229 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1232 } else if (reply->status_code != RTSP_STATUS_OK ||
1233 reply->nb_transports != 1) {
1234 err = AVERROR_INVALIDDATA;
1238 /* XXX: same protocol for all streams is required */
1240 if (reply->transports[0].lower_transport != rt->lower_transport ||
1241 reply->transports[0].transport != rt->transport) {
1242 err = AVERROR_INVALIDDATA;
1246 rt->lower_transport = reply->transports[0].lower_transport;
1247 rt->transport = reply->transports[0].transport;
1250 /* Fail if the server responded with another lower transport mode
1251 * than what we requested. */
1252 if (reply->transports[0].lower_transport != lower_transport) {
1253 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1254 err = AVERROR_INVALIDDATA;
1258 switch(reply->transports[0].lower_transport) {
1259 case RTSP_LOWER_TRANSPORT_TCP:
1260 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1261 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1264 case RTSP_LOWER_TRANSPORT_UDP: {
1265 char url[1024], options[30] = "";
1267 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1268 av_strlcpy(options, "?connect=1", sizeof(options));
1269 /* Use source address if specified */
1270 if (reply->transports[0].source[0]) {
1271 ff_url_join(url, sizeof(url), "rtp", NULL,
1272 reply->transports[0].source,
1273 reply->transports[0].server_port_min, "%s", options);
1275 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1276 reply->transports[0].server_port_min, "%s", options);
1278 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1279 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1280 err = AVERROR_INVALIDDATA;
1283 /* Try to initialize the connection state in a
1284 * potential NAT router by sending dummy packets.
1285 * RTP/RTCP dummy packets are used for RDT, too.
1287 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1289 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1292 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1293 char url[1024], namebuf[50];
1294 struct sockaddr_storage addr;
1297 if (reply->transports[0].destination.ss_family) {
1298 addr = reply->transports[0].destination;
1299 port = reply->transports[0].port_min;
1300 ttl = reply->transports[0].ttl;
1302 addr = rtsp_st->sdp_ip;
1303 port = rtsp_st->sdp_port;
1304 ttl = rtsp_st->sdp_ttl;
1306 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1307 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1308 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1309 port, "?ttl=%d", ttl);
1310 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1311 &s->interrupt_callback, NULL) < 0) {
1312 err = AVERROR_INVALIDDATA;
1319 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1323 if (reply->timeout > 0)
1324 rt->timeout = reply->timeout;
1326 if (rt->server_type == RTSP_SERVER_REAL)
1327 rt->need_subscription = 1;
1332 ff_rtsp_undo_setup(s);
1336 void ff_rtsp_close_connections(AVFormatContext *s)
1338 RTSPState *rt = s->priv_data;
1339 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1340 ffurl_close(rt->rtsp_hd);
1341 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1344 int ff_rtsp_connect(AVFormatContext *s)
1346 RTSPState *rt = s->priv_data;
1347 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1348 char *option_list, *option, *filename;
1349 int port, err, tcp_fd;
1350 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1351 int lower_transport_mask = 0;
1352 char real_challenge[64] = "";
1353 struct sockaddr_storage peer;
1354 socklen_t peer_len = sizeof(peer);
1356 if (!ff_network_init())
1357 return AVERROR(EIO);
1359 rt->control_transport = RTSP_MODE_PLAIN;
1360 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1361 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1362 rt->control_transport = RTSP_MODE_TUNNEL;
1364 /* Only pass through valid flags from here */
1365 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1368 lower_transport_mask = rt->lower_transport_mask;
1369 /* extract hostname and port */
1370 av_url_split(NULL, 0, auth, sizeof(auth),
1371 host, sizeof(host), &port, path, sizeof(path), s->filename);
1373 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1376 port = RTSP_DEFAULT_PORT;
1378 #if FF_API_RTSP_URL_OPTIONS
1379 /* search for options */
1380 option_list = strrchr(path, '?');
1382 /* Strip out the RTSP specific options, write out the rest of
1383 * the options back into the same string. */
1384 filename = option_list;
1385 while (option_list) {
1387 /* move the option pointer */
1388 option = ++option_list;
1389 option_list = strchr(option_list, '&');
1393 /* handle the options */
1394 if (!strcmp(option, "udp")) {
1395 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1396 } else if (!strcmp(option, "multicast")) {
1397 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1398 } else if (!strcmp(option, "tcp")) {
1399 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1400 } else if(!strcmp(option, "http")) {
1401 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1402 rt->control_transport = RTSP_MODE_TUNNEL;
1403 } else if (!strcmp(option, "filter_src")) {
1404 rt->rtsp_flags |= RTSP_FLAG_FILTER_SRC;
1406 /* Write options back into the buffer, using memmove instead
1407 * of strcpy since the strings may overlap. */
1408 int len = strlen(option);
1409 memmove(++filename, option, len);
1411 if (option_list) *filename = '&';
1415 av_log(s, AV_LOG_WARNING, "Options passed via URL are "
1416 "deprecated, use -rtsp_transport "
1417 "and -rtsp_flags instead.\n");
1423 if (!lower_transport_mask)
1424 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1427 /* Only UDP or TCP - UDP multicast isn't supported. */
1428 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1429 (1 << RTSP_LOWER_TRANSPORT_TCP);
1430 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1431 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1432 "only UDP and TCP are supported for output.\n");
1433 err = AVERROR(EINVAL);
1438 /* Construct the URI used in request; this is similar to s->filename,
1439 * but with authentication credentials removed and RTSP specific options
1441 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1442 host, port, "%s", path);
1444 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1445 /* set up initial handshake for tunneling */
1446 char httpname[1024];
1447 char sessioncookie[17];
1450 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1451 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1452 av_get_random_seed(), av_get_random_seed());
1455 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1456 &s->interrupt_callback) < 0) {
1461 /* generate GET headers */
1462 snprintf(headers, sizeof(headers),
1463 "x-sessioncookie: %s\r\n"
1464 "Accept: application/x-rtsp-tunnelled\r\n"
1465 "Pragma: no-cache\r\n"
1466 "Cache-Control: no-cache\r\n",
1468 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1470 /* complete the connection */
1471 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1477 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1478 &s->interrupt_callback) < 0 ) {
1483 /* generate POST headers */
1484 snprintf(headers, sizeof(headers),
1485 "x-sessioncookie: %s\r\n"
1486 "Content-Type: application/x-rtsp-tunnelled\r\n"
1487 "Pragma: no-cache\r\n"
1488 "Cache-Control: no-cache\r\n"
1489 "Content-Length: 32767\r\n"
1490 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1492 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1493 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1495 /* Initialize the authentication state for the POST session. The HTTP
1496 * protocol implementation doesn't properly handle multi-pass
1497 * authentication for POST requests, since it would require one of
1499 * - implementing Expect: 100-continue, which many HTTP servers
1500 * don't support anyway, even less the RTSP servers that do HTTP
1502 * - sending the whole POST data until getting a 401 reply specifying
1503 * what authentication method to use, then resending all that data
1504 * - waiting for potential 401 replies directly after sending the
1505 * POST header (waiting for some unspecified time)
1506 * Therefore, we copy the full auth state, which works for both basic
1507 * and digest. (For digest, we would have to synchronize the nonce
1508 * count variable between the two sessions, if we'd do more requests
1509 * with the original session, though.)
1511 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1513 /* complete the connection */
1514 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1519 /* open the tcp connection */
1520 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1521 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1522 &s->interrupt_callback, NULL) < 0) {
1526 rt->rtsp_hd_out = rt->rtsp_hd;
1530 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1531 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1532 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1533 NULL, 0, NI_NUMERICHOST);
1536 /* request options supported by the server; this also detects server
1538 for (rt->server_type = RTSP_SERVER_RTP;;) {
1540 if (rt->server_type == RTSP_SERVER_REAL)
1543 * The following entries are required for proper
1544 * streaming from a Realmedia server. They are
1545 * interdependent in some way although we currently
1546 * don't quite understand how. Values were copied
1547 * from mplayer SVN r23589.
1548 * ClientChallenge is a 16-byte ID in hex
1549 * CompanyID is a 16-byte ID in base64
1551 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1552 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1553 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1554 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1556 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1557 if (reply->status_code != RTSP_STATUS_OK) {
1558 err = AVERROR_INVALIDDATA;
1562 /* detect server type if not standard-compliant RTP */
1563 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1564 rt->server_type = RTSP_SERVER_REAL;
1566 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1567 rt->server_type = RTSP_SERVER_WMS;
1568 } else if (rt->server_type == RTSP_SERVER_REAL)
1569 strcpy(real_challenge, reply->real_challenge);
1573 if (s->iformat && CONFIG_RTSP_DEMUXER)
1574 err = ff_rtsp_setup_input_streams(s, reply);
1575 else if (CONFIG_RTSP_MUXER)
1576 err = ff_rtsp_setup_output_streams(s, host);
1581 int lower_transport = ff_log2_tab[lower_transport_mask &
1582 ~(lower_transport_mask - 1)];
1584 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1585 rt->server_type == RTSP_SERVER_REAL ?
1586 real_challenge : NULL);
1589 lower_transport_mask &= ~(1 << lower_transport);
1590 if (lower_transport_mask == 0 && err == 1) {
1591 err = AVERROR(EPROTONOSUPPORT);
1596 rt->lower_transport_mask = lower_transport_mask;
1597 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1598 rt->state = RTSP_STATE_IDLE;
1599 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1602 ff_rtsp_close_streams(s);
1603 ff_rtsp_close_connections(s);
1604 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1605 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1606 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1614 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1617 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1618 uint8_t *buf, int buf_size, int64_t wait_end)
1620 RTSPState *rt = s->priv_data;
1621 RTSPStream *rtsp_st;
1622 int n, i, ret, tcp_fd, timeout_cnt = 0;
1624 struct pollfd *p = rt->p;
1627 if (ff_check_interrupt(&s->interrupt_callback))
1628 return AVERROR_EXIT;
1629 if (wait_end && wait_end - av_gettime() < 0)
1630 return AVERROR(EAGAIN);
1633 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1634 p[max_p].fd = tcp_fd;
1635 p[max_p++].events = POLLIN;
1639 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1640 rtsp_st = rt->rtsp_streams[i];
1641 if (rtsp_st->rtp_handle) {
1642 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1643 p[max_p++].events = POLLIN;
1644 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1645 p[max_p++].events = POLLIN;
1648 n = poll(p, max_p, POLL_TIMEOUT_MS);
1650 int j = 1 - (tcp_fd == -1);
1652 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1653 rtsp_st = rt->rtsp_streams[i];
1654 if (rtsp_st->rtp_handle) {
1655 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1656 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1658 *prtsp_st = rtsp_st;
1665 #if CONFIG_RTSP_DEMUXER
1666 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1667 RTSPMessageHeader reply;
1669 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1672 /* XXX: parse message */
1673 if (rt->state != RTSP_STATE_STREAMING)
1677 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1678 return AVERROR(ETIMEDOUT);
1679 } else if (n < 0 && errno != EINTR)
1680 return AVERROR(errno);
1684 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1686 RTSPState *rt = s->priv_data;
1688 RTSPStream *rtsp_st, *first_queue_st = NULL;
1689 int64_t wait_end = 0;
1691 if (rt->nb_byes == rt->nb_rtsp_streams)
1694 /* get next frames from the same RTP packet */
1695 if (rt->cur_transport_priv) {
1696 if (rt->transport == RTSP_TRANSPORT_RDT) {
1697 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1699 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1701 rt->cur_transport_priv = NULL;
1703 } else if (ret == 1) {
1706 rt->cur_transport_priv = NULL;
1709 if (rt->transport == RTSP_TRANSPORT_RTP) {
1711 int64_t first_queue_time = 0;
1712 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1713 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1717 queue_time = ff_rtp_queued_packet_time(rtpctx);
1718 if (queue_time && (queue_time - first_queue_time < 0 ||
1719 !first_queue_time)) {
1720 first_queue_time = queue_time;
1721 first_queue_st = rt->rtsp_streams[i];
1724 if (first_queue_time)
1725 wait_end = first_queue_time + s->max_delay;
1728 /* read next RTP packet */
1731 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1733 return AVERROR(ENOMEM);
1736 switch(rt->lower_transport) {
1738 #if CONFIG_RTSP_DEMUXER
1739 case RTSP_LOWER_TRANSPORT_TCP:
1740 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1743 case RTSP_LOWER_TRANSPORT_UDP:
1744 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1745 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1746 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1747 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1750 if (len == AVERROR(EAGAIN) && first_queue_st &&
1751 rt->transport == RTSP_TRANSPORT_RTP) {
1752 rtsp_st = first_queue_st;
1753 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1760 if (rt->transport == RTSP_TRANSPORT_RDT) {
1761 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1763 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1765 /* Either bad packet, or a RTCP packet. Check if the
1766 * first_rtcp_ntp_time field was initialized. */
1767 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1768 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1769 /* first_rtcp_ntp_time has been initialized for this stream,
1770 * copy the same value to all other uninitialized streams,
1771 * in order to map their timestamp origin to the same ntp time
1774 AVStream *st = NULL;
1775 if (rtsp_st->stream_index >= 0)
1776 st = s->streams[rtsp_st->stream_index];
1777 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1778 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1779 AVStream *st2 = NULL;
1780 if (rt->rtsp_streams[i]->stream_index >= 0)
1781 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1782 if (rtpctx2 && st && st2 &&
1783 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1784 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1785 rtpctx2->rtcp_ts_offset = av_rescale_q(
1786 rtpctx->rtcp_ts_offset, st->time_base,
1791 if (ret == -RTCP_BYE) {
1794 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1795 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1797 if (rt->nb_byes == rt->nb_rtsp_streams)
1806 /* more packets may follow, so we save the RTP context */
1807 rt->cur_transport_priv = rtsp_st->transport_priv;
1811 #endif /* CONFIG_RTPDEC */
1813 #if CONFIG_SDP_DEMUXER
1814 static int sdp_probe(AVProbeData *p1)
1816 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1818 /* we look for a line beginning "c=IN IP" */
1819 while (p < p_end && *p != '\0') {
1820 if (p + sizeof("c=IN IP") - 1 < p_end &&
1821 av_strstart(p, "c=IN IP", NULL))
1822 return AVPROBE_SCORE_MAX / 2;
1824 while (p < p_end - 1 && *p != '\n') p++;
1833 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1835 RTSPState *rt = s->priv_data;
1836 RTSPStream *rtsp_st;
1841 if (!ff_network_init())
1842 return AVERROR(EIO);
1844 /* read the whole sdp file */
1845 /* XXX: better loading */
1846 content = av_malloc(SDP_MAX_SIZE);
1847 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1850 return AVERROR_INVALIDDATA;
1852 content[size] ='\0';
1854 err = ff_sdp_parse(s, content);
1858 /* open each RTP stream */
1859 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1861 rtsp_st = rt->rtsp_streams[i];
1863 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1864 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1865 ff_url_join(url, sizeof(url), "rtp", NULL,
1866 namebuf, rtsp_st->sdp_port,
1867 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1869 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1870 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1871 &s->interrupt_callback, NULL) < 0) {
1872 err = AVERROR_INVALIDDATA;
1875 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1880 ff_rtsp_close_streams(s);
1885 static int sdp_read_close(AVFormatContext *s)
1887 ff_rtsp_close_streams(s);
1892 static const AVClass sdp_demuxer_class = {
1893 .class_name = "SDP demuxer",
1894 .item_name = av_default_item_name,
1895 .option = sdp_options,
1896 .version = LIBAVUTIL_VERSION_INT,
1899 AVInputFormat ff_sdp_demuxer = {
1901 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1902 .priv_data_size = sizeof(RTSPState),
1903 .read_probe = sdp_probe,
1904 .read_header = sdp_read_header,
1905 .read_packet = ff_rtsp_fetch_packet,
1906 .read_close = sdp_read_close,
1907 .priv_class = &sdp_demuxer_class
1909 #endif /* CONFIG_SDP_DEMUXER */
1911 #if CONFIG_RTP_DEMUXER
1912 static int rtp_probe(AVProbeData *p)
1914 if (av_strstart(p->filename, "rtp:", NULL))
1915 return AVPROBE_SCORE_MAX;
1919 static int rtp_read_header(AVFormatContext *s,
1920 AVFormatParameters *ap)
1922 uint8_t recvbuf[1500];
1923 char host[500], sdp[500];
1925 URLContext* in = NULL;
1927 AVCodecContext codec;
1928 struct sockaddr_storage addr;
1930 socklen_t addrlen = sizeof(addr);
1932 if (!ff_network_init())
1933 return AVERROR(EIO);
1935 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1936 &s->interrupt_callback, NULL);
1941 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1942 if (ret == AVERROR(EAGAIN))
1947 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1951 if ((recvbuf[0] & 0xc0) != 0x80) {
1952 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1957 payload_type = recvbuf[1] & 0x7f;
1960 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1964 memset(&codec, 0, sizeof(codec));
1965 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1966 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1967 "without an SDP file describing it\n",
1971 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1972 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1973 "properly you need an SDP file "
1977 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1978 NULL, 0, s->filename);
1980 snprintf(sdp, sizeof(sdp),
1981 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1982 addr.ss_family == AF_INET ? 4 : 6, host,
1983 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1984 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1985 port, payload_type);
1986 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1988 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1991 /* sdp_read_header initializes this again */
1994 ret = sdp_read_header(s, ap);
2005 static const AVClass rtp_demuxer_class = {
2006 .class_name = "RTP demuxer",
2007 .item_name = av_default_item_name,
2008 .option = rtp_options,
2009 .version = LIBAVUTIL_VERSION_INT,
2012 AVInputFormat ff_rtp_demuxer = {
2014 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2015 .priv_data_size = sizeof(RTSPState),
2016 .read_probe = rtp_probe,
2017 .read_header = rtp_read_header,
2018 .read_packet = ff_rtsp_fetch_packet,
2019 .read_close = sdp_read_close,
2020 .flags = AVFMT_NOFILE,
2021 .priv_class = &rtp_demuxer_class
2023 #endif /* CONFIG_RTP_DEMUXER */