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
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 const AVOption ff_rtsp_options[] = {
76 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
77 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags)
78 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
79 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
81 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
82 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
83 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
84 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
85 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
86 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
90 static const AVOption sdp_options[] = {
91 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
92 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
96 static const AVOption rtp_options[] = {
97 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
101 static void get_word_until_chars(char *buf, int buf_size,
102 const char *sep, const char **pp)
108 p += strspn(p, SPACE_CHARS);
110 while (!strchr(sep, *p) && *p != '\0') {
111 if ((q - buf) < buf_size - 1)
120 static void get_word_sep(char *buf, int buf_size, const char *sep,
123 if (**pp == '/') (*pp)++;
124 get_word_until_chars(buf, buf_size, sep, pp);
127 static void get_word(char *buf, int buf_size, const char **pp)
129 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
132 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
134 * Used for seeking in the rtp stream.
136 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
140 p += strspn(p, SPACE_CHARS);
141 if (!av_stristart(p, "npt=", &p))
144 *start = AV_NOPTS_VALUE;
145 *end = AV_NOPTS_VALUE;
147 get_word_sep(buf, sizeof(buf), "-", &p);
148 av_parse_time(start, buf, 1);
151 get_word_sep(buf, sizeof(buf), "-", &p);
152 av_parse_time(end, buf, 1);
154 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
155 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
158 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
160 struct addrinfo hints, *ai = NULL;
161 memset(&hints, 0, sizeof(hints));
162 hints.ai_flags = AI_NUMERICHOST;
163 if (getaddrinfo(buf, NULL, &hints, &ai))
165 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
171 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
172 RTSPStream *rtsp_st, AVCodecContext *codec)
176 codec->codec_id = handler->codec_id;
177 rtsp_st->dynamic_handler = handler;
178 if (handler->alloc) {
179 rtsp_st->dynamic_protocol_context = handler->alloc();
180 if (!rtsp_st->dynamic_protocol_context)
181 rtsp_st->dynamic_handler = NULL;
185 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
186 static int sdp_parse_rtpmap(AVFormatContext *s,
187 AVStream *st, RTSPStream *rtsp_st,
188 int payload_type, const char *p)
190 AVCodecContext *codec = st->codec;
196 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
197 * see if we can handle this kind of payload.
198 * The space should normally not be there but some Real streams or
199 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
200 * have a trailing space. */
201 get_word_sep(buf, sizeof(buf), "/ ", &p);
202 if (payload_type >= RTP_PT_PRIVATE) {
203 RTPDynamicProtocolHandler *handler =
204 ff_rtp_handler_find_by_name(buf, codec->codec_type);
205 init_rtp_handler(handler, rtsp_st, codec);
206 /* If no dynamic handler was found, check with the list of standard
207 * allocated types, if such a stream for some reason happens to
208 * use a private payload type. This isn't handled in rtpdec.c, since
209 * the format name from the rtpmap line never is passed into rtpdec. */
210 if (!rtsp_st->dynamic_handler)
211 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
213 /* We are in a standard case
214 * (from http://www.iana.org/assignments/rtp-parameters). */
215 /* search into AVRtpPayloadTypes[] */
216 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
219 c = avcodec_find_decoder(codec->codec_id);
225 get_word_sep(buf, sizeof(buf), "/", &p);
227 switch (codec->codec_type) {
228 case AVMEDIA_TYPE_AUDIO:
229 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
230 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
231 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
233 codec->sample_rate = i;
234 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
235 get_word_sep(buf, sizeof(buf), "/", &p);
239 // TODO: there is a bug here; if it is a mono stream, and
240 // less than 22000Hz, faad upconverts to stereo and twice
241 // the frequency. No problem, but the sample rate is being
242 // set here by the sdp line. Patch on its way. (rdm)
244 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
246 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
249 case AVMEDIA_TYPE_VIDEO:
250 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
252 avpriv_set_pts_info(st, 32, 1, i);
257 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
258 rtsp_st->dynamic_handler->init(s, st->index,
259 rtsp_st->dynamic_protocol_context);
263 /* parse the attribute line from the fmtp a line of an sdp response. This
264 * is broken out as a function because it is used in rtp_h264.c, which is
266 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
267 char *value, int value_size)
269 *p += strspn(*p, SPACE_CHARS);
271 get_word_sep(attr, attr_size, "=", p);
274 get_word_sep(value, value_size, ";", p);
282 typedef struct SDPParseState {
284 struct sockaddr_storage default_ip;
286 int skip_media; ///< set if an unknown m= line occurs
289 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
290 int letter, const char *buf)
292 RTSPState *rt = s->priv_data;
293 char buf1[64], st_type[64];
295 enum AVMediaType codec_type;
299 struct sockaddr_storage sdp_ip;
302 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
305 if (s1->skip_media && letter != 'm')
309 get_word(buf1, sizeof(buf1), &p);
310 if (strcmp(buf1, "IN") != 0)
312 get_word(buf1, sizeof(buf1), &p);
313 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
315 get_word_sep(buf1, sizeof(buf1), "/", &p);
316 if (get_sockaddr(buf1, &sdp_ip))
321 get_word_sep(buf1, sizeof(buf1), "/", &p);
324 if (s->nb_streams == 0) {
325 s1->default_ip = sdp_ip;
326 s1->default_ttl = ttl;
328 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
329 rtsp_st->sdp_ip = sdp_ip;
330 rtsp_st->sdp_ttl = ttl;
334 av_dict_set(&s->metadata, "title", p, 0);
337 if (s->nb_streams == 0) {
338 av_dict_set(&s->metadata, "comment", p, 0);
345 codec_type = AVMEDIA_TYPE_UNKNOWN;
346 get_word(st_type, sizeof(st_type), &p);
347 if (!strcmp(st_type, "audio")) {
348 codec_type = AVMEDIA_TYPE_AUDIO;
349 } else if (!strcmp(st_type, "video")) {
350 codec_type = AVMEDIA_TYPE_VIDEO;
351 } else if (!strcmp(st_type, "application")) {
352 codec_type = AVMEDIA_TYPE_DATA;
354 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
358 rtsp_st = av_mallocz(sizeof(RTSPStream));
361 rtsp_st->stream_index = -1;
362 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
364 rtsp_st->sdp_ip = s1->default_ip;
365 rtsp_st->sdp_ttl = s1->default_ttl;
367 get_word(buf1, sizeof(buf1), &p); /* port */
368 rtsp_st->sdp_port = atoi(buf1);
370 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
372 /* XXX: handle list of formats */
373 get_word(buf1, sizeof(buf1), &p); /* format list */
374 rtsp_st->sdp_payload_type = atoi(buf1);
376 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
377 /* no corresponding stream */
379 st = avformat_new_stream(s, NULL);
382 st->id = rt->nb_rtsp_streams - 1;
383 rtsp_st->stream_index = st->index;
384 st->codec->codec_type = codec_type;
385 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
386 RTPDynamicProtocolHandler *handler;
387 /* if standard payload type, we can find the codec right now */
388 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
389 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
390 st->codec->sample_rate > 0)
391 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
392 /* Even static payload types may need a custom depacketizer */
393 handler = ff_rtp_handler_find_by_id(
394 rtsp_st->sdp_payload_type, st->codec->codec_type);
395 init_rtp_handler(handler, rtsp_st, st->codec);
396 if (handler && handler->init)
397 handler->init(s, st->index,
398 rtsp_st->dynamic_protocol_context);
401 /* put a default control url */
402 av_strlcpy(rtsp_st->control_url, rt->control_uri,
403 sizeof(rtsp_st->control_url));
406 if (av_strstart(p, "control:", &p)) {
407 if (s->nb_streams == 0) {
408 if (!strncmp(p, "rtsp://", 7))
409 av_strlcpy(rt->control_uri, p,
410 sizeof(rt->control_uri));
413 /* get the control url */
414 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
416 /* XXX: may need to add full url resolution */
417 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
419 if (proto[0] == '\0') {
420 /* relative control URL */
421 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
422 av_strlcat(rtsp_st->control_url, "/",
423 sizeof(rtsp_st->control_url));
424 av_strlcat(rtsp_st->control_url, p,
425 sizeof(rtsp_st->control_url));
427 av_strlcpy(rtsp_st->control_url, p,
428 sizeof(rtsp_st->control_url));
430 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
431 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
432 get_word(buf1, sizeof(buf1), &p);
433 payload_type = atoi(buf1);
434 st = s->streams[s->nb_streams - 1];
435 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
436 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
437 } else if (av_strstart(p, "fmtp:", &p) ||
438 av_strstart(p, "framesize:", &p)) {
439 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
440 // let dynamic protocol handlers have a stab at the line.
441 get_word(buf1, sizeof(buf1), &p);
442 payload_type = atoi(buf1);
443 for (i = 0; i < rt->nb_rtsp_streams; i++) {
444 rtsp_st = rt->rtsp_streams[i];
445 if (rtsp_st->sdp_payload_type == payload_type &&
446 rtsp_st->dynamic_handler &&
447 rtsp_st->dynamic_handler->parse_sdp_a_line)
448 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
449 rtsp_st->dynamic_protocol_context, buf);
451 } else if (av_strstart(p, "range:", &p)) {
454 // this is so that seeking on a streamed file can work.
455 rtsp_parse_range_npt(p, &start, &end);
456 s->start_time = start;
457 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
458 s->duration = (end == AV_NOPTS_VALUE) ?
459 AV_NOPTS_VALUE : end - start;
460 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
462 rt->transport = RTSP_TRANSPORT_RDT;
463 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
465 st = s->streams[s->nb_streams - 1];
466 st->codec->sample_rate = atoi(p);
468 if (rt->server_type == RTSP_SERVER_WMS)
469 ff_wms_parse_sdp_a_line(s, p);
470 if (s->nb_streams > 0) {
471 if (rt->server_type == RTSP_SERVER_REAL)
472 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
474 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
475 if (rtsp_st->dynamic_handler &&
476 rtsp_st->dynamic_handler->parse_sdp_a_line)
477 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
479 rtsp_st->dynamic_protocol_context, buf);
486 int ff_sdp_parse(AVFormatContext *s, const char *content)
488 RTSPState *rt = s->priv_data;
491 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
492 * contain long SDP lines containing complete ASF Headers (several
493 * kB) or arrays of MDPR (RM stream descriptor) headers plus
494 * "rulebooks" describing their properties. Therefore, the SDP line
497 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
498 * in rtpdec_xiph.c. */
500 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
502 memset(s1, 0, sizeof(SDPParseState));
505 p += strspn(p, SPACE_CHARS);
513 /* get the content */
515 while (*p != '\n' && *p != '\r' && *p != '\0') {
516 if ((q - buf) < sizeof(buf) - 1)
521 sdp_parse_line(s, s1, letter, buf);
523 while (*p != '\n' && *p != '\0')
528 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
529 if (!rt->p) return AVERROR(ENOMEM);
532 #endif /* CONFIG_RTPDEC */
534 void ff_rtsp_undo_setup(AVFormatContext *s)
536 RTSPState *rt = s->priv_data;
539 for (i = 0; i < rt->nb_rtsp_streams; i++) {
540 RTSPStream *rtsp_st = rt->rtsp_streams[i];
543 if (rtsp_st->transport_priv) {
545 AVFormatContext *rtpctx = rtsp_st->transport_priv;
546 av_write_trailer(rtpctx);
547 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
549 avio_close_dyn_buf(rtpctx->pb, &ptr);
552 avio_close(rtpctx->pb);
554 avformat_free_context(rtpctx);
555 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
556 ff_rdt_parse_close(rtsp_st->transport_priv);
557 else if (CONFIG_RTPDEC)
558 ff_rtp_parse_close(rtsp_st->transport_priv);
560 rtsp_st->transport_priv = NULL;
561 if (rtsp_st->rtp_handle)
562 ffurl_close(rtsp_st->rtp_handle);
563 rtsp_st->rtp_handle = NULL;
567 /* close and free RTSP streams */
568 void ff_rtsp_close_streams(AVFormatContext *s)
570 RTSPState *rt = s->priv_data;
574 ff_rtsp_undo_setup(s);
575 for (i = 0; i < rt->nb_rtsp_streams; i++) {
576 rtsp_st = rt->rtsp_streams[i];
578 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
579 rtsp_st->dynamic_handler->free(
580 rtsp_st->dynamic_protocol_context);
584 av_free(rt->rtsp_streams);
586 avformat_close_input(&rt->asf_ctx);
589 av_free(rt->recvbuf);
592 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
594 RTSPState *rt = s->priv_data;
597 /* open the RTP context */
598 if (rtsp_st->stream_index >= 0)
599 st = s->streams[rtsp_st->stream_index];
601 s->ctx_flags |= AVFMTCTX_NOHEADER;
603 if (s->oformat && CONFIG_RTSP_MUXER) {
604 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
606 RTSP_TCP_MAX_PACKET_SIZE);
607 /* Ownership of rtp_handle is passed to the rtp mux context */
608 rtsp_st->rtp_handle = NULL;
609 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
610 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
611 rtsp_st->dynamic_protocol_context,
612 rtsp_st->dynamic_handler);
613 else if (CONFIG_RTPDEC)
614 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
615 rtsp_st->sdp_payload_type,
616 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
617 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
619 if (!rtsp_st->transport_priv) {
620 return AVERROR(ENOMEM);
621 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
622 if (rtsp_st->dynamic_handler) {
623 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
624 rtsp_st->dynamic_protocol_context,
625 rtsp_st->dynamic_handler);
632 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
633 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
639 p += strspn(p, SPACE_CHARS);
640 v = strtol(p, (char **)&p, 10);
644 v = strtol(p, (char **)&p, 10);
653 /* XXX: only one transport specification is parsed */
654 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
656 char transport_protocol[16];
658 char lower_transport[16];
660 RTSPTransportField *th;
663 reply->nb_transports = 0;
666 p += strspn(p, SPACE_CHARS);
670 th = &reply->transports[reply->nb_transports];
672 get_word_sep(transport_protocol, sizeof(transport_protocol),
674 if (!av_strcasecmp (transport_protocol, "rtp")) {
675 get_word_sep(profile, sizeof(profile), "/;,", &p);
676 lower_transport[0] = '\0';
677 /* rtp/avp/<protocol> */
679 get_word_sep(lower_transport, sizeof(lower_transport),
682 th->transport = RTSP_TRANSPORT_RTP;
683 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
684 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
685 /* x-pn-tng/<protocol> */
686 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
688 th->transport = RTSP_TRANSPORT_RDT;
690 if (!av_strcasecmp(lower_transport, "TCP"))
691 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
693 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
697 /* get each parameter */
698 while (*p != '\0' && *p != ',') {
699 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
700 if (!strcmp(parameter, "port")) {
703 rtsp_parse_range(&th->port_min, &th->port_max, &p);
705 } else if (!strcmp(parameter, "client_port")) {
708 rtsp_parse_range(&th->client_port_min,
709 &th->client_port_max, &p);
711 } else if (!strcmp(parameter, "server_port")) {
714 rtsp_parse_range(&th->server_port_min,
715 &th->server_port_max, &p);
717 } else if (!strcmp(parameter, "interleaved")) {
720 rtsp_parse_range(&th->interleaved_min,
721 &th->interleaved_max, &p);
723 } else if (!strcmp(parameter, "multicast")) {
724 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
725 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
726 } else if (!strcmp(parameter, "ttl")) {
729 th->ttl = strtol(p, (char **)&p, 10);
731 } else if (!strcmp(parameter, "destination")) {
734 get_word_sep(buf, sizeof(buf), ";,", &p);
735 get_sockaddr(buf, &th->destination);
737 } else if (!strcmp(parameter, "source")) {
740 get_word_sep(buf, sizeof(buf), ";,", &p);
741 av_strlcpy(th->source, buf, sizeof(th->source));
745 while (*p != ';' && *p != '\0' && *p != ',')
753 reply->nb_transports++;
757 static void handle_rtp_info(RTSPState *rt, const char *url,
758 uint32_t seq, uint32_t rtptime)
761 if (!rtptime || !url[0])
763 if (rt->transport != RTSP_TRANSPORT_RTP)
765 for (i = 0; i < rt->nb_rtsp_streams; i++) {
766 RTSPStream *rtsp_st = rt->rtsp_streams[i];
767 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
770 if (!strcmp(rtsp_st->control_url, url)) {
771 rtpctx->base_timestamp = rtptime;
777 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
780 char key[20], value[1024], url[1024] = "";
781 uint32_t seq = 0, rtptime = 0;
784 p += strspn(p, SPACE_CHARS);
787 get_word_sep(key, sizeof(key), "=", &p);
791 get_word_sep(value, sizeof(value), ";, ", &p);
793 if (!strcmp(key, "url"))
794 av_strlcpy(url, value, sizeof(url));
795 else if (!strcmp(key, "seq"))
796 seq = strtoul(value, NULL, 10);
797 else if (!strcmp(key, "rtptime"))
798 rtptime = strtoul(value, NULL, 10);
800 handle_rtp_info(rt, url, seq, rtptime);
809 handle_rtp_info(rt, url, seq, rtptime);
812 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
813 RTSPState *rt, const char *method)
817 /* NOTE: we do case independent match for broken servers */
819 if (av_stristart(p, "Session:", &p)) {
821 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
822 if (av_stristart(p, ";timeout=", &p) &&
823 (t = strtol(p, NULL, 10)) > 0) {
826 } else if (av_stristart(p, "Content-Length:", &p)) {
827 reply->content_length = strtol(p, NULL, 10);
828 } else if (av_stristart(p, "Transport:", &p)) {
829 rtsp_parse_transport(reply, p);
830 } else if (av_stristart(p, "CSeq:", &p)) {
831 reply->seq = strtol(p, NULL, 10);
832 } else if (av_stristart(p, "Range:", &p)) {
833 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
834 } else if (av_stristart(p, "RealChallenge1:", &p)) {
835 p += strspn(p, SPACE_CHARS);
836 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
837 } else if (av_stristart(p, "Server:", &p)) {
838 p += strspn(p, SPACE_CHARS);
839 av_strlcpy(reply->server, p, sizeof(reply->server));
840 } else if (av_stristart(p, "Notice:", &p) ||
841 av_stristart(p, "X-Notice:", &p)) {
842 reply->notice = strtol(p, NULL, 10);
843 } else if (av_stristart(p, "Location:", &p)) {
844 p += strspn(p, SPACE_CHARS);
845 av_strlcpy(reply->location, p , sizeof(reply->location));
846 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
847 p += strspn(p, SPACE_CHARS);
848 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
849 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
850 p += strspn(p, SPACE_CHARS);
851 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
852 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
853 p += strspn(p, SPACE_CHARS);
854 if (method && !strcmp(method, "DESCRIBE"))
855 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
856 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
857 p += strspn(p, SPACE_CHARS);
858 if (method && !strcmp(method, "PLAY"))
859 rtsp_parse_rtp_info(rt, p);
860 } else if (av_stristart(p, "Public:", &p) && rt) {
861 if (strstr(p, "GET_PARAMETER") &&
862 method && !strcmp(method, "OPTIONS"))
863 rt->get_parameter_supported = 1;
864 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
865 p += strspn(p, SPACE_CHARS);
866 rt->accept_dynamic_rate = atoi(p);
870 /* skip a RTP/TCP interleaved packet */
871 void ff_rtsp_skip_packet(AVFormatContext *s)
873 RTSPState *rt = s->priv_data;
877 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
880 len = AV_RB16(buf + 1);
882 av_dlog(s, "skipping RTP packet len=%d\n", len);
887 if (len1 > sizeof(buf))
889 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
896 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
897 unsigned char **content_ptr,
898 int return_on_interleaved_data, const char *method)
900 RTSPState *rt = s->priv_data;
901 char buf[4096], buf1[1024], *q;
904 int ret, content_length, line_count = 0, request = 0;
905 unsigned char *content = NULL;
911 memset(reply, 0, sizeof(*reply));
913 /* parse reply (XXX: use buffers) */
914 rt->last_reply[0] = '\0';
918 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
919 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
925 /* XXX: only parse it if first char on line ? */
926 if (return_on_interleaved_data) {
929 ff_rtsp_skip_packet(s);
930 } else if (ch != '\r') {
931 if ((q - buf) < sizeof(buf) - 1)
937 av_dlog(s, "line='%s'\n", buf);
939 /* test if last line */
943 if (line_count == 0) {
945 get_word(buf1, sizeof(buf1), &p);
946 if (!strncmp(buf1, "RTSP/", 5)) {
947 get_word(buf1, sizeof(buf1), &p);
948 reply->status_code = atoi(buf1);
949 av_strlcpy(reply->reason, p, sizeof(reply->reason));
951 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
952 get_word(buf1, sizeof(buf1), &p); // object
956 ff_rtsp_parse_line(reply, p, rt, method);
957 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
958 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
963 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
964 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
966 content_length = reply->content_length;
967 if (content_length > 0) {
968 /* leave some room for a trailing '\0' (useful for simple parsing) */
969 content = av_malloc(content_length + 1);
970 ffurl_read_complete(rt->rtsp_hd, content, content_length);
971 content[content_length] = '\0';
974 *content_ptr = content;
980 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
981 const char* ptr = buf;
983 if (!strcmp(reply->reason, "OPTIONS")) {
984 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
986 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
987 if (reply->session_id[0])
988 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
991 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
993 av_strlcat(buf, "\r\n", sizeof(buf));
995 if (rt->control_transport == RTSP_MODE_TUNNEL) {
996 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
999 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1001 rt->last_cmd_time = av_gettime();
1002 /* Even if the request from the server had data, it is not the data
1003 * that the caller wants or expects. The memory could also be leaked
1004 * if the actual following reply has content data. */
1006 av_freep(content_ptr);
1007 /* If method is set, this is called from ff_rtsp_send_cmd,
1008 * where a reply to exactly this request is awaited. For
1009 * callers from within packet receiving, we just want to
1010 * return to the caller and go back to receiving packets. */
1016 if (rt->seq != reply->seq) {
1017 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1018 rt->seq, reply->seq);
1022 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1023 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1024 reply->notice == 2306 /* Continuous Feed Terminated */) {
1025 rt->state = RTSP_STATE_IDLE;
1026 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1027 return AVERROR(EIO); /* data or server error */
1028 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1029 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1030 return AVERROR(EPERM);
1036 * Send a command to the RTSP server without waiting for the reply.
1038 * @param s RTSP (de)muxer context
1039 * @param method the method for the request
1040 * @param url the target url for the request
1041 * @param headers extra header lines to include in the request
1042 * @param send_content if non-null, the data to send as request body content
1043 * @param send_content_length the length of the send_content data, or 0 if
1044 * send_content is null
1046 * @return zero if success, nonzero otherwise
1048 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1049 const char *method, const char *url,
1050 const char *headers,
1051 const unsigned char *send_content,
1052 int send_content_length)
1054 RTSPState *rt = s->priv_data;
1055 char buf[4096], *out_buf;
1056 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1058 /* Add in RTSP headers */
1061 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1063 av_strlcat(buf, headers, sizeof(buf));
1064 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1065 if (rt->session_id[0] != '\0' && (!headers ||
1066 !strstr(headers, "\nIf-Match:"))) {
1067 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1070 char *str = ff_http_auth_create_response(&rt->auth_state,
1071 rt->auth, url, method);
1073 av_strlcat(buf, str, sizeof(buf));
1076 if (send_content_length > 0 && send_content)
1077 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1078 av_strlcat(buf, "\r\n", sizeof(buf));
1080 /* base64 encode rtsp if tunneling */
1081 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1082 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1083 out_buf = base64buf;
1086 av_dlog(s, "Sending:\n%s--\n", buf);
1088 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1089 if (send_content_length > 0 && send_content) {
1090 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1091 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1092 "with content data not supported\n");
1093 return AVERROR_PATCHWELCOME;
1095 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1097 rt->last_cmd_time = av_gettime();
1102 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1103 const char *url, const char *headers)
1105 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1108 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1109 const char *headers, RTSPMessageHeader *reply,
1110 unsigned char **content_ptr)
1112 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1113 content_ptr, NULL, 0);
1116 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1117 const char *method, const char *url,
1119 RTSPMessageHeader *reply,
1120 unsigned char **content_ptr,
1121 const unsigned char *send_content,
1122 int send_content_length)
1124 RTSPState *rt = s->priv_data;
1125 HTTPAuthType cur_auth_type;
1126 int ret, attempts = 0;
1129 cur_auth_type = rt->auth_state.auth_type;
1130 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1132 send_content_length)))
1135 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1139 if (reply->status_code == 401 &&
1140 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1141 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1144 if (reply->status_code > 400){
1145 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1149 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1155 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1156 int lower_transport, const char *real_challenge)
1158 RTSPState *rt = s->priv_data;
1159 int rtx = 0, j, i, err, interleave = 0, port_off;
1160 RTSPStream *rtsp_st;
1161 RTSPMessageHeader reply1, *reply = &reply1;
1163 const char *trans_pref;
1165 if (rt->transport == RTSP_TRANSPORT_RDT)
1166 trans_pref = "x-pn-tng";
1168 trans_pref = "RTP/AVP";
1170 /* default timeout: 1 minute */
1173 /* Choose a random starting offset within the first half of the
1174 * port range, to allow for a number of ports to try even if the offset
1175 * happens to be at the end of the random range. */
1176 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1177 /* even random offset */
1178 port_off -= port_off & 0x01;
1180 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1181 char transport[2048];
1184 * WMS serves all UDP data over a single connection, the RTX, which
1185 * isn't necessarily the first in the SDP but has to be the first
1186 * to be set up, else the second/third SETUP will fail with a 461.
1188 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1189 rt->server_type == RTSP_SERVER_WMS) {
1192 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1193 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1195 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1199 if (rtx == rt->nb_rtsp_streams)
1200 return -1; /* no RTX found */
1201 rtsp_st = rt->rtsp_streams[rtx];
1203 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1205 rtsp_st = rt->rtsp_streams[i];
1208 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1211 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1212 port = reply->transports[0].client_port_min;
1216 /* first try in specified port range */
1217 while (j <= rt->rtp_port_max) {
1218 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1219 "?localport=%d", j);
1220 /* we will use two ports per rtp stream (rtp and rtcp) */
1222 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1223 &s->interrupt_callback, NULL))
1226 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1231 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1233 snprintf(transport, sizeof(transport) - 1,
1234 "%s/UDP;", trans_pref);
1235 if (rt->server_type != RTSP_SERVER_REAL)
1236 av_strlcat(transport, "unicast;", sizeof(transport));
1237 av_strlcatf(transport, sizeof(transport),
1238 "client_port=%d", port);
1239 if (rt->transport == RTSP_TRANSPORT_RTP &&
1240 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1241 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1245 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1246 /* For WMS streams, the application streams are only used for
1247 * UDP. When trying to set it up for TCP streams, the server
1248 * will return an error. Therefore, we skip those streams. */
1249 if (rt->server_type == RTSP_SERVER_WMS &&
1250 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1253 snprintf(transport, sizeof(transport) - 1,
1254 "%s/TCP;", trans_pref);
1255 if (rt->transport != RTSP_TRANSPORT_RDT)
1256 av_strlcat(transport, "unicast;", sizeof(transport));
1257 av_strlcatf(transport, sizeof(transport),
1258 "interleaved=%d-%d",
1259 interleave, interleave + 1);
1263 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1264 snprintf(transport, sizeof(transport) - 1,
1265 "%s/UDP;multicast", trans_pref);
1268 av_strlcat(transport, ";mode=receive", sizeof(transport));
1269 } else if (rt->server_type == RTSP_SERVER_REAL ||
1270 rt->server_type == RTSP_SERVER_WMS)
1271 av_strlcat(transport, ";mode=play", sizeof(transport));
1272 snprintf(cmd, sizeof(cmd),
1273 "Transport: %s\r\n",
1275 if (rt->accept_dynamic_rate)
1276 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1277 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1278 char real_res[41], real_csum[9];
1279 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1281 av_strlcatf(cmd, sizeof(cmd),
1283 "RealChallenge2: %s, sd=%s\r\n",
1284 rt->session_id, real_res, real_csum);
1286 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1287 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1290 } else if (reply->status_code != RTSP_STATUS_OK ||
1291 reply->nb_transports != 1) {
1292 err = AVERROR_INVALIDDATA;
1296 /* XXX: same protocol for all streams is required */
1298 if (reply->transports[0].lower_transport != rt->lower_transport ||
1299 reply->transports[0].transport != rt->transport) {
1300 err = AVERROR_INVALIDDATA;
1304 rt->lower_transport = reply->transports[0].lower_transport;
1305 rt->transport = reply->transports[0].transport;
1308 /* Fail if the server responded with another lower transport mode
1309 * than what we requested. */
1310 if (reply->transports[0].lower_transport != lower_transport) {
1311 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1312 err = AVERROR_INVALIDDATA;
1316 switch(reply->transports[0].lower_transport) {
1317 case RTSP_LOWER_TRANSPORT_TCP:
1318 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1319 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1322 case RTSP_LOWER_TRANSPORT_UDP: {
1323 char url[1024], options[30] = "";
1325 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1326 av_strlcpy(options, "?connect=1", sizeof(options));
1327 /* Use source address if specified */
1328 if (reply->transports[0].source[0]) {
1329 ff_url_join(url, sizeof(url), "rtp", NULL,
1330 reply->transports[0].source,
1331 reply->transports[0].server_port_min, "%s", options);
1333 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1334 reply->transports[0].server_port_min, "%s", options);
1336 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1337 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1338 err = AVERROR_INVALIDDATA;
1341 /* Try to initialize the connection state in a
1342 * potential NAT router by sending dummy packets.
1343 * RTP/RTCP dummy packets are used for RDT, too.
1345 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1347 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1350 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1351 char url[1024], namebuf[50], optbuf[20] = "";
1352 struct sockaddr_storage addr;
1355 if (reply->transports[0].destination.ss_family) {
1356 addr = reply->transports[0].destination;
1357 port = reply->transports[0].port_min;
1358 ttl = reply->transports[0].ttl;
1360 addr = rtsp_st->sdp_ip;
1361 port = rtsp_st->sdp_port;
1362 ttl = rtsp_st->sdp_ttl;
1365 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1366 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1367 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1368 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1369 port, "%s", optbuf);
1370 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1371 &s->interrupt_callback, NULL) < 0) {
1372 err = AVERROR_INVALIDDATA;
1379 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1383 if (reply->timeout > 0)
1384 rt->timeout = reply->timeout;
1386 if (rt->server_type == RTSP_SERVER_REAL)
1387 rt->need_subscription = 1;
1392 ff_rtsp_undo_setup(s);
1396 void ff_rtsp_close_connections(AVFormatContext *s)
1398 RTSPState *rt = s->priv_data;
1399 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1400 ffurl_close(rt->rtsp_hd);
1401 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1404 int ff_rtsp_connect(AVFormatContext *s)
1406 RTSPState *rt = s->priv_data;
1407 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1408 int port, err, tcp_fd;
1409 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1410 int lower_transport_mask = 0;
1411 char real_challenge[64] = "";
1412 struct sockaddr_storage peer;
1413 socklen_t peer_len = sizeof(peer);
1415 if (rt->rtp_port_max < rt->rtp_port_min) {
1416 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1417 "than min port %d\n", rt->rtp_port_max,
1419 return AVERROR(EINVAL);
1422 if (!ff_network_init())
1423 return AVERROR(EIO);
1425 if (s->max_delay < 0) /* Not set by the caller */
1426 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1428 rt->control_transport = RTSP_MODE_PLAIN;
1429 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1430 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1431 rt->control_transport = RTSP_MODE_TUNNEL;
1433 /* Only pass through valid flags from here */
1434 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1437 lower_transport_mask = rt->lower_transport_mask;
1438 /* extract hostname and port */
1439 av_url_split(NULL, 0, auth, sizeof(auth),
1440 host, sizeof(host), &port, path, sizeof(path), s->filename);
1442 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1445 port = RTSP_DEFAULT_PORT;
1447 if (!lower_transport_mask)
1448 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1451 /* Only UDP or TCP - UDP multicast isn't supported. */
1452 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1453 (1 << RTSP_LOWER_TRANSPORT_TCP);
1454 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1455 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1456 "only UDP and TCP are supported for output.\n");
1457 err = AVERROR(EINVAL);
1462 /* Construct the URI used in request; this is similar to s->filename,
1463 * but with authentication credentials removed and RTSP specific options
1465 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1466 host, port, "%s", path);
1468 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1469 /* set up initial handshake for tunneling */
1470 char httpname[1024];
1471 char sessioncookie[17];
1474 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1475 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1476 av_get_random_seed(), av_get_random_seed());
1479 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1480 &s->interrupt_callback) < 0) {
1485 /* generate GET headers */
1486 snprintf(headers, sizeof(headers),
1487 "x-sessioncookie: %s\r\n"
1488 "Accept: application/x-rtsp-tunnelled\r\n"
1489 "Pragma: no-cache\r\n"
1490 "Cache-Control: no-cache\r\n",
1492 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1494 /* complete the connection */
1495 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1501 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1502 &s->interrupt_callback) < 0 ) {
1507 /* generate POST headers */
1508 snprintf(headers, sizeof(headers),
1509 "x-sessioncookie: %s\r\n"
1510 "Content-Type: application/x-rtsp-tunnelled\r\n"
1511 "Pragma: no-cache\r\n"
1512 "Cache-Control: no-cache\r\n"
1513 "Content-Length: 32767\r\n"
1514 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1516 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1517 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1519 /* Initialize the authentication state for the POST session. The HTTP
1520 * protocol implementation doesn't properly handle multi-pass
1521 * authentication for POST requests, since it would require one of
1523 * - implementing Expect: 100-continue, which many HTTP servers
1524 * don't support anyway, even less the RTSP servers that do HTTP
1526 * - sending the whole POST data until getting a 401 reply specifying
1527 * what authentication method to use, then resending all that data
1528 * - waiting for potential 401 replies directly after sending the
1529 * POST header (waiting for some unspecified time)
1530 * Therefore, we copy the full auth state, which works for both basic
1531 * and digest. (For digest, we would have to synchronize the nonce
1532 * count variable between the two sessions, if we'd do more requests
1533 * with the original session, though.)
1535 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1537 /* complete the connection */
1538 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1543 /* open the tcp connection */
1544 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1545 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1546 &s->interrupt_callback, NULL) < 0) {
1550 rt->rtsp_hd_out = rt->rtsp_hd;
1554 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1555 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1556 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1557 NULL, 0, NI_NUMERICHOST);
1560 /* request options supported by the server; this also detects server
1562 for (rt->server_type = RTSP_SERVER_RTP;;) {
1564 if (rt->server_type == RTSP_SERVER_REAL)
1567 * The following entries are required for proper
1568 * streaming from a Realmedia server. They are
1569 * interdependent in some way although we currently
1570 * don't quite understand how. Values were copied
1571 * from mplayer SVN r23589.
1572 * ClientChallenge is a 16-byte ID in hex
1573 * CompanyID is a 16-byte ID in base64
1575 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1576 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1577 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1578 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1580 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1581 if (reply->status_code != RTSP_STATUS_OK) {
1582 err = AVERROR_INVALIDDATA;
1586 /* detect server type if not standard-compliant RTP */
1587 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1588 rt->server_type = RTSP_SERVER_REAL;
1590 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1591 rt->server_type = RTSP_SERVER_WMS;
1592 } else if (rt->server_type == RTSP_SERVER_REAL)
1593 strcpy(real_challenge, reply->real_challenge);
1597 if (s->iformat && CONFIG_RTSP_DEMUXER)
1598 err = ff_rtsp_setup_input_streams(s, reply);
1599 else if (CONFIG_RTSP_MUXER)
1600 err = ff_rtsp_setup_output_streams(s, host);
1605 int lower_transport = ff_log2_tab[lower_transport_mask &
1606 ~(lower_transport_mask - 1)];
1608 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1609 rt->server_type == RTSP_SERVER_REAL ?
1610 real_challenge : NULL);
1613 lower_transport_mask &= ~(1 << lower_transport);
1614 if (lower_transport_mask == 0 && err == 1) {
1615 err = AVERROR(EPROTONOSUPPORT);
1620 rt->lower_transport_mask = lower_transport_mask;
1621 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1622 rt->state = RTSP_STATE_IDLE;
1623 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1626 ff_rtsp_close_streams(s);
1627 ff_rtsp_close_connections(s);
1628 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1629 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1630 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1638 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1641 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1642 uint8_t *buf, int buf_size, int64_t wait_end)
1644 RTSPState *rt = s->priv_data;
1645 RTSPStream *rtsp_st;
1646 int n, i, ret, tcp_fd, timeout_cnt = 0;
1648 struct pollfd *p = rt->p;
1651 if (ff_check_interrupt(&s->interrupt_callback))
1652 return AVERROR_EXIT;
1653 if (wait_end && wait_end - av_gettime() < 0)
1654 return AVERROR(EAGAIN);
1657 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1658 p[max_p].fd = tcp_fd;
1659 p[max_p++].events = POLLIN;
1663 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1664 rtsp_st = rt->rtsp_streams[i];
1665 if (rtsp_st->rtp_handle) {
1666 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1667 p[max_p++].events = POLLIN;
1668 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1669 p[max_p++].events = POLLIN;
1672 n = poll(p, max_p, POLL_TIMEOUT_MS);
1674 int j = 1 - (tcp_fd == -1);
1676 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1677 rtsp_st = rt->rtsp_streams[i];
1678 if (rtsp_st->rtp_handle) {
1679 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1680 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1682 *prtsp_st = rtsp_st;
1689 #if CONFIG_RTSP_DEMUXER
1690 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1691 RTSPMessageHeader reply;
1693 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1696 /* XXX: parse message */
1697 if (rt->state != RTSP_STATE_STREAMING)
1701 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1702 return AVERROR(ETIMEDOUT);
1703 } else if (n < 0 && errno != EINTR)
1704 return AVERROR(errno);
1708 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1710 RTSPState *rt = s->priv_data;
1712 RTSPStream *rtsp_st, *first_queue_st = NULL;
1713 int64_t wait_end = 0;
1715 if (rt->nb_byes == rt->nb_rtsp_streams)
1718 /* get next frames from the same RTP packet */
1719 if (rt->cur_transport_priv) {
1720 if (rt->transport == RTSP_TRANSPORT_RDT) {
1721 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1723 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1725 rt->cur_transport_priv = NULL;
1727 } else if (ret == 1) {
1730 rt->cur_transport_priv = NULL;
1733 if (rt->transport == RTSP_TRANSPORT_RTP) {
1735 int64_t first_queue_time = 0;
1736 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1737 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1741 queue_time = ff_rtp_queued_packet_time(rtpctx);
1742 if (queue_time && (queue_time - first_queue_time < 0 ||
1743 !first_queue_time)) {
1744 first_queue_time = queue_time;
1745 first_queue_st = rt->rtsp_streams[i];
1748 if (first_queue_time)
1749 wait_end = first_queue_time + s->max_delay;
1752 /* read next RTP packet */
1755 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1757 return AVERROR(ENOMEM);
1760 switch(rt->lower_transport) {
1762 #if CONFIG_RTSP_DEMUXER
1763 case RTSP_LOWER_TRANSPORT_TCP:
1764 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1767 case RTSP_LOWER_TRANSPORT_UDP:
1768 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1769 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1770 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1771 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1774 if (len == AVERROR(EAGAIN) && first_queue_st &&
1775 rt->transport == RTSP_TRANSPORT_RTP) {
1776 rtsp_st = first_queue_st;
1777 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1784 if (rt->transport == RTSP_TRANSPORT_RDT) {
1785 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1787 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1789 /* Either bad packet, or a RTCP packet. Check if the
1790 * first_rtcp_ntp_time field was initialized. */
1791 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1792 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1793 /* first_rtcp_ntp_time has been initialized for this stream,
1794 * copy the same value to all other uninitialized streams,
1795 * in order to map their timestamp origin to the same ntp time
1798 AVStream *st = NULL;
1799 if (rtsp_st->stream_index >= 0)
1800 st = s->streams[rtsp_st->stream_index];
1801 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1802 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1803 AVStream *st2 = NULL;
1804 if (rt->rtsp_streams[i]->stream_index >= 0)
1805 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1806 if (rtpctx2 && st && st2 &&
1807 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1808 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1809 rtpctx2->rtcp_ts_offset = av_rescale_q(
1810 rtpctx->rtcp_ts_offset, st->time_base,
1815 if (ret == -RTCP_BYE) {
1818 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1819 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1821 if (rt->nb_byes == rt->nb_rtsp_streams)
1830 /* more packets may follow, so we save the RTP context */
1831 rt->cur_transport_priv = rtsp_st->transport_priv;
1835 #endif /* CONFIG_RTPDEC */
1837 #if CONFIG_SDP_DEMUXER
1838 static int sdp_probe(AVProbeData *p1)
1840 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1842 /* we look for a line beginning "c=IN IP" */
1843 while (p < p_end && *p != '\0') {
1844 if (p + sizeof("c=IN IP") - 1 < p_end &&
1845 av_strstart(p, "c=IN IP", NULL))
1846 return AVPROBE_SCORE_MAX / 2;
1848 while (p < p_end - 1 && *p != '\n') p++;
1857 static int sdp_read_header(AVFormatContext *s)
1859 RTSPState *rt = s->priv_data;
1860 RTSPStream *rtsp_st;
1865 if (!ff_network_init())
1866 return AVERROR(EIO);
1868 if (s->max_delay < 0) /* Not set by the caller */
1869 s->max_delay = DEFAULT_REORDERING_DELAY;
1871 /* read the whole sdp file */
1872 /* XXX: better loading */
1873 content = av_malloc(SDP_MAX_SIZE);
1874 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1877 return AVERROR_INVALIDDATA;
1879 content[size] ='\0';
1881 err = ff_sdp_parse(s, content);
1885 /* open each RTP stream */
1886 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1888 rtsp_st = rt->rtsp_streams[i];
1890 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1891 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1892 ff_url_join(url, sizeof(url), "rtp", NULL,
1893 namebuf, rtsp_st->sdp_port,
1894 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1896 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1897 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1898 &s->interrupt_callback, NULL) < 0) {
1899 err = AVERROR_INVALIDDATA;
1902 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1907 ff_rtsp_close_streams(s);
1912 static int sdp_read_close(AVFormatContext *s)
1914 ff_rtsp_close_streams(s);
1919 static const AVClass sdp_demuxer_class = {
1920 .class_name = "SDP demuxer",
1921 .item_name = av_default_item_name,
1922 .option = sdp_options,
1923 .version = LIBAVUTIL_VERSION_INT,
1926 AVInputFormat ff_sdp_demuxer = {
1928 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1929 .priv_data_size = sizeof(RTSPState),
1930 .read_probe = sdp_probe,
1931 .read_header = sdp_read_header,
1932 .read_packet = ff_rtsp_fetch_packet,
1933 .read_close = sdp_read_close,
1934 .priv_class = &sdp_demuxer_class,
1936 #endif /* CONFIG_SDP_DEMUXER */
1938 #if CONFIG_RTP_DEMUXER
1939 static int rtp_probe(AVProbeData *p)
1941 if (av_strstart(p->filename, "rtp:", NULL))
1942 return AVPROBE_SCORE_MAX;
1946 static int rtp_read_header(AVFormatContext *s)
1948 uint8_t recvbuf[1500];
1949 char host[500], sdp[500];
1951 URLContext* in = NULL;
1953 AVCodecContext codec;
1954 struct sockaddr_storage addr;
1956 socklen_t addrlen = sizeof(addr);
1957 RTSPState *rt = s->priv_data;
1959 if (!ff_network_init())
1960 return AVERROR(EIO);
1962 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1963 &s->interrupt_callback, NULL);
1968 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1969 if (ret == AVERROR(EAGAIN))
1974 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1978 if ((recvbuf[0] & 0xc0) != 0x80) {
1979 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1984 if (RTP_PT_IS_RTCP(recvbuf[1]))
1987 payload_type = recvbuf[1] & 0x7f;
1990 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1994 memset(&codec, 0, sizeof(codec));
1995 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1996 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1997 "without an SDP file describing it\n",
2001 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2002 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2003 "properly you need an SDP file "
2007 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2008 NULL, 0, s->filename);
2010 snprintf(sdp, sizeof(sdp),
2011 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2012 addr.ss_family == AF_INET ? 4 : 6, host,
2013 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2014 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2015 port, payload_type);
2016 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2018 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2021 /* sdp_read_header initializes this again */
2024 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2026 ret = sdp_read_header(s);
2037 static const AVClass rtp_demuxer_class = {
2038 .class_name = "RTP demuxer",
2039 .item_name = av_default_item_name,
2040 .option = rtp_options,
2041 .version = LIBAVUTIL_VERSION_INT,
2044 AVInputFormat ff_rtp_demuxer = {
2046 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2047 .priv_data_size = sizeof(RTSPState),
2048 .read_probe = rtp_probe,
2049 .read_header = rtp_read_header,
2050 .read_packet = ff_rtsp_fetch_packet,
2051 .read_close = sdp_read_close,
2052 .flags = AVFMT_NOFILE,
2053 .priv_class = &rtp_demuxer_class,
2055 #endif /* CONFIG_RTP_DEMUXER */