3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; 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"
38 #include "os_support.h"
44 #include "rtpdec_formats.h"
45 #include "rtpenc_chain.h"
51 /* Timeout values for socket poll, in ms,
52 * and read_packet(), in seconds */
53 #define POLL_TIMEOUT_MS 100
54 #define READ_PACKET_TIMEOUT_S 10
55 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
56 #define SDP_MAX_SIZE 16384
57 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
58 #define DEFAULT_REORDERING_DELAY 100000
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"),
84 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
85 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
89 static const AVOption sdp_options[] = {
90 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
91 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
95 static const AVOption rtp_options[] = {
96 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
100 static void get_word_until_chars(char *buf, int buf_size,
101 const char *sep, const char **pp)
107 p += strspn(p, SPACE_CHARS);
109 while (!strchr(sep, *p) && *p != '\0') {
110 if ((q - buf) < buf_size - 1)
119 static void get_word_sep(char *buf, int buf_size, const char *sep,
122 if (**pp == '/') (*pp)++;
123 get_word_until_chars(buf, buf_size, sep, pp);
126 static void get_word(char *buf, int buf_size, const char **pp)
128 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
131 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
133 * Used for seeking in the rtp stream.
135 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
139 p += strspn(p, SPACE_CHARS);
140 if (!av_stristart(p, "npt=", &p))
143 *start = AV_NOPTS_VALUE;
144 *end = AV_NOPTS_VALUE;
146 get_word_sep(buf, sizeof(buf), "-", &p);
147 av_parse_time(start, buf, 1);
150 get_word_sep(buf, sizeof(buf), "-", &p);
151 av_parse_time(end, buf, 1);
153 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
154 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
157 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
159 struct addrinfo hints = { 0 }, *ai = NULL;
160 hints.ai_flags = AI_NUMERICHOST;
161 if (getaddrinfo(buf, NULL, &hints, &ai))
163 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
169 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
170 RTSPStream *rtsp_st, AVCodecContext *codec)
174 codec->codec_id = handler->codec_id;
175 rtsp_st->dynamic_handler = handler;
176 if (handler->alloc) {
177 rtsp_st->dynamic_protocol_context = handler->alloc();
178 if (!rtsp_st->dynamic_protocol_context)
179 rtsp_st->dynamic_handler = NULL;
183 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
184 static int sdp_parse_rtpmap(AVFormatContext *s,
185 AVStream *st, RTSPStream *rtsp_st,
186 int payload_type, const char *p)
188 AVCodecContext *codec = st->codec;
194 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
195 * see if we can handle this kind of payload.
196 * The space should normally not be there but some Real streams or
197 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
198 * have a trailing space. */
199 get_word_sep(buf, sizeof(buf), "/ ", &p);
200 if (payload_type < RTP_PT_PRIVATE) {
201 /* We are in a standard case
202 * (from http://www.iana.org/assignments/rtp-parameters). */
203 /* search into AVRtpPayloadTypes[] */
204 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
207 if (codec->codec_id == CODEC_ID_NONE) {
208 RTPDynamicProtocolHandler *handler =
209 ff_rtp_handler_find_by_name(buf, codec->codec_type);
210 init_rtp_handler(handler, rtsp_st, codec);
211 /* If no dynamic handler was found, check with the list of standard
212 * allocated types, if such a stream for some reason happens to
213 * use a private payload type. This isn't handled in rtpdec.c, since
214 * the format name from the rtpmap line never is passed into rtpdec. */
215 if (!rtsp_st->dynamic_handler)
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 */
378 } else if (rt->server_type == RTSP_SERVER_WMS &&
379 codec_type == AVMEDIA_TYPE_DATA) {
380 /* RTX stream, a stream that carries all the other actual
381 * audio/video streams. Don't expose this to the callers. */
383 st = avformat_new_stream(s, NULL);
386 st->id = rt->nb_rtsp_streams - 1;
387 rtsp_st->stream_index = st->index;
388 st->codec->codec_type = codec_type;
389 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
390 RTPDynamicProtocolHandler *handler;
391 /* if standard payload type, we can find the codec right now */
392 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
393 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
394 st->codec->sample_rate > 0)
395 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
396 /* Even static payload types may need a custom depacketizer */
397 handler = ff_rtp_handler_find_by_id(
398 rtsp_st->sdp_payload_type, st->codec->codec_type);
399 init_rtp_handler(handler, rtsp_st, st->codec);
400 if (handler && handler->init)
401 handler->init(s, st->index,
402 rtsp_st->dynamic_protocol_context);
405 /* put a default control url */
406 av_strlcpy(rtsp_st->control_url, rt->control_uri,
407 sizeof(rtsp_st->control_url));
410 if (av_strstart(p, "control:", &p)) {
411 if (s->nb_streams == 0) {
412 if (!strncmp(p, "rtsp://", 7))
413 av_strlcpy(rt->control_uri, p,
414 sizeof(rt->control_uri));
417 /* get the control url */
418 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420 /* XXX: may need to add full url resolution */
421 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
423 if (proto[0] == '\0') {
424 /* relative control URL */
425 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
426 av_strlcat(rtsp_st->control_url, "/",
427 sizeof(rtsp_st->control_url));
428 av_strlcat(rtsp_st->control_url, p,
429 sizeof(rtsp_st->control_url));
431 av_strlcpy(rtsp_st->control_url, p,
432 sizeof(rtsp_st->control_url));
434 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
435 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
436 get_word(buf1, sizeof(buf1), &p);
437 payload_type = atoi(buf1);
438 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
439 if (rtsp_st->stream_index >= 0) {
440 st = s->streams[rtsp_st->stream_index];
441 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
443 } else if (av_strstart(p, "fmtp:", &p) ||
444 av_strstart(p, "framesize:", &p)) {
445 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
446 // let dynamic protocol handlers have a stab at the line.
447 get_word(buf1, sizeof(buf1), &p);
448 payload_type = atoi(buf1);
449 for (i = 0; i < rt->nb_rtsp_streams; i++) {
450 rtsp_st = rt->rtsp_streams[i];
451 if (rtsp_st->sdp_payload_type == payload_type &&
452 rtsp_st->dynamic_handler &&
453 rtsp_st->dynamic_handler->parse_sdp_a_line)
454 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
455 rtsp_st->dynamic_protocol_context, buf);
457 } else if (av_strstart(p, "range:", &p)) {
460 // this is so that seeking on a streamed file can work.
461 rtsp_parse_range_npt(p, &start, &end);
462 s->start_time = start;
463 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
464 s->duration = (end == AV_NOPTS_VALUE) ?
465 AV_NOPTS_VALUE : end - start;
466 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
468 rt->transport = RTSP_TRANSPORT_RDT;
469 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
471 st = s->streams[s->nb_streams - 1];
472 st->codec->sample_rate = atoi(p);
474 if (rt->server_type == RTSP_SERVER_WMS)
475 ff_wms_parse_sdp_a_line(s, p);
476 if (s->nb_streams > 0) {
477 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
479 if (rt->server_type == RTSP_SERVER_REAL)
480 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
482 if (rtsp_st->dynamic_handler &&
483 rtsp_st->dynamic_handler->parse_sdp_a_line)
484 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
485 rtsp_st->stream_index,
486 rtsp_st->dynamic_protocol_context, buf);
493 int ff_sdp_parse(AVFormatContext *s, const char *content)
495 RTSPState *rt = s->priv_data;
498 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
499 * contain long SDP lines containing complete ASF Headers (several
500 * kB) or arrays of MDPR (RM stream descriptor) headers plus
501 * "rulebooks" describing their properties. Therefore, the SDP line
504 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
505 * in rtpdec_xiph.c. */
507 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
511 p += strspn(p, SPACE_CHARS);
519 /* get the content */
521 while (*p != '\n' && *p != '\r' && *p != '\0') {
522 if ((q - buf) < sizeof(buf) - 1)
527 sdp_parse_line(s, s1, letter, buf);
529 while (*p != '\n' && *p != '\0')
534 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
535 if (!rt->p) return AVERROR(ENOMEM);
538 #endif /* CONFIG_RTPDEC */
540 void ff_rtsp_undo_setup(AVFormatContext *s)
542 RTSPState *rt = s->priv_data;
545 for (i = 0; i < rt->nb_rtsp_streams; i++) {
546 RTSPStream *rtsp_st = rt->rtsp_streams[i];
549 if (rtsp_st->transport_priv) {
551 AVFormatContext *rtpctx = rtsp_st->transport_priv;
552 av_write_trailer(rtpctx);
553 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
555 avio_close_dyn_buf(rtpctx->pb, &ptr);
558 avio_close(rtpctx->pb);
560 avformat_free_context(rtpctx);
561 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
562 ff_rdt_parse_close(rtsp_st->transport_priv);
563 else if (CONFIG_RTPDEC)
564 ff_rtp_parse_close(rtsp_st->transport_priv);
566 rtsp_st->transport_priv = NULL;
567 if (rtsp_st->rtp_handle)
568 ffurl_close(rtsp_st->rtp_handle);
569 rtsp_st->rtp_handle = NULL;
573 /* close and free RTSP streams */
574 void ff_rtsp_close_streams(AVFormatContext *s)
576 RTSPState *rt = s->priv_data;
580 ff_rtsp_undo_setup(s);
581 for (i = 0; i < rt->nb_rtsp_streams; i++) {
582 rtsp_st = rt->rtsp_streams[i];
584 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
585 rtsp_st->dynamic_handler->free(
586 rtsp_st->dynamic_protocol_context);
590 av_free(rt->rtsp_streams);
592 avformat_close_input(&rt->asf_ctx);
595 av_free(rt->recvbuf);
598 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
600 RTSPState *rt = s->priv_data;
603 /* open the RTP context */
604 if (rtsp_st->stream_index >= 0)
605 st = s->streams[rtsp_st->stream_index];
607 s->ctx_flags |= AVFMTCTX_NOHEADER;
609 if (s->oformat && CONFIG_RTSP_MUXER) {
610 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
612 RTSP_TCP_MAX_PACKET_SIZE);
613 /* Ownership of rtp_handle is passed to the rtp mux context */
614 rtsp_st->rtp_handle = NULL;
617 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
618 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
619 rtsp_st->dynamic_protocol_context,
620 rtsp_st->dynamic_handler);
621 else if (CONFIG_RTPDEC)
622 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
623 rtsp_st->sdp_payload_type,
624 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
625 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
627 if (!rtsp_st->transport_priv) {
628 return AVERROR(ENOMEM);
629 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
630 if (rtsp_st->dynamic_handler) {
631 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
632 rtsp_st->dynamic_protocol_context,
633 rtsp_st->dynamic_handler);
640 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
641 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
648 q += strspn(q, SPACE_CHARS);
649 v = strtol(q, &p, 10);
653 v = strtol(p, &p, 10);
662 /* XXX: only one transport specification is parsed */
663 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
665 char transport_protocol[16];
667 char lower_transport[16];
669 RTSPTransportField *th;
672 reply->nb_transports = 0;
675 p += strspn(p, SPACE_CHARS);
679 th = &reply->transports[reply->nb_transports];
681 get_word_sep(transport_protocol, sizeof(transport_protocol),
683 if (!av_strcasecmp (transport_protocol, "rtp")) {
684 get_word_sep(profile, sizeof(profile), "/;,", &p);
685 lower_transport[0] = '\0';
686 /* rtp/avp/<protocol> */
688 get_word_sep(lower_transport, sizeof(lower_transport),
691 th->transport = RTSP_TRANSPORT_RTP;
692 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
693 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
694 /* x-pn-tng/<protocol> */
695 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
697 th->transport = RTSP_TRANSPORT_RDT;
699 if (!av_strcasecmp(lower_transport, "TCP"))
700 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
702 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
706 /* get each parameter */
707 while (*p != '\0' && *p != ',') {
708 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
709 if (!strcmp(parameter, "port")) {
712 rtsp_parse_range(&th->port_min, &th->port_max, &p);
714 } else if (!strcmp(parameter, "client_port")) {
717 rtsp_parse_range(&th->client_port_min,
718 &th->client_port_max, &p);
720 } else if (!strcmp(parameter, "server_port")) {
723 rtsp_parse_range(&th->server_port_min,
724 &th->server_port_max, &p);
726 } else if (!strcmp(parameter, "interleaved")) {
729 rtsp_parse_range(&th->interleaved_min,
730 &th->interleaved_max, &p);
732 } else if (!strcmp(parameter, "multicast")) {
733 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
734 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
735 } else if (!strcmp(parameter, "ttl")) {
738 th->ttl = strtol(p, (char **)&p, 10);
740 } else if (!strcmp(parameter, "destination")) {
743 get_word_sep(buf, sizeof(buf), ";,", &p);
744 get_sockaddr(buf, &th->destination);
746 } else if (!strcmp(parameter, "source")) {
749 get_word_sep(buf, sizeof(buf), ";,", &p);
750 av_strlcpy(th->source, buf, sizeof(th->source));
752 } else if (!strcmp(parameter, "mode")) {
755 get_word_sep(buf, sizeof(buf), ";, ", &p);
756 if (!strcmp(buf, "record") ||
757 !strcmp(buf, "receive"))
762 while (*p != ';' && *p != '\0' && *p != ',')
770 reply->nb_transports++;
774 static void handle_rtp_info(RTSPState *rt, const char *url,
775 uint32_t seq, uint32_t rtptime)
778 if (!rtptime || !url[0])
780 if (rt->transport != RTSP_TRANSPORT_RTP)
782 for (i = 0; i < rt->nb_rtsp_streams; i++) {
783 RTSPStream *rtsp_st = rt->rtsp_streams[i];
784 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
787 if (!strcmp(rtsp_st->control_url, url)) {
788 rtpctx->base_timestamp = rtptime;
794 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
797 char key[20], value[1024], url[1024] = "";
798 uint32_t seq = 0, rtptime = 0;
801 p += strspn(p, SPACE_CHARS);
804 get_word_sep(key, sizeof(key), "=", &p);
808 get_word_sep(value, sizeof(value), ";, ", &p);
810 if (!strcmp(key, "url"))
811 av_strlcpy(url, value, sizeof(url));
812 else if (!strcmp(key, "seq"))
813 seq = strtoul(value, NULL, 10);
814 else if (!strcmp(key, "rtptime"))
815 rtptime = strtoul(value, NULL, 10);
817 handle_rtp_info(rt, url, seq, rtptime);
826 handle_rtp_info(rt, url, seq, rtptime);
829 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
830 RTSPState *rt, const char *method)
834 /* NOTE: we do case independent match for broken servers */
836 if (av_stristart(p, "Session:", &p)) {
838 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
839 if (av_stristart(p, ";timeout=", &p) &&
840 (t = strtol(p, NULL, 10)) > 0) {
843 } else if (av_stristart(p, "Content-Length:", &p)) {
844 reply->content_length = strtol(p, NULL, 10);
845 } else if (av_stristart(p, "Transport:", &p)) {
846 rtsp_parse_transport(reply, p);
847 } else if (av_stristart(p, "CSeq:", &p)) {
848 reply->seq = strtol(p, NULL, 10);
849 } else if (av_stristart(p, "Range:", &p)) {
850 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
851 } else if (av_stristart(p, "RealChallenge1:", &p)) {
852 p += strspn(p, SPACE_CHARS);
853 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
854 } else if (av_stristart(p, "Server:", &p)) {
855 p += strspn(p, SPACE_CHARS);
856 av_strlcpy(reply->server, p, sizeof(reply->server));
857 } else if (av_stristart(p, "Notice:", &p) ||
858 av_stristart(p, "X-Notice:", &p)) {
859 reply->notice = strtol(p, NULL, 10);
860 } else if (av_stristart(p, "Location:", &p)) {
861 p += strspn(p, SPACE_CHARS);
862 av_strlcpy(reply->location, p , sizeof(reply->location));
863 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
864 p += strspn(p, SPACE_CHARS);
865 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
866 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
867 p += strspn(p, SPACE_CHARS);
868 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
869 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
870 p += strspn(p, SPACE_CHARS);
871 if (method && !strcmp(method, "DESCRIBE"))
872 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
873 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
874 p += strspn(p, SPACE_CHARS);
875 if (method && !strcmp(method, "PLAY"))
876 rtsp_parse_rtp_info(rt, p);
877 } else if (av_stristart(p, "Public:", &p) && rt) {
878 if (strstr(p, "GET_PARAMETER") &&
879 method && !strcmp(method, "OPTIONS"))
880 rt->get_parameter_supported = 1;
881 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
882 p += strspn(p, SPACE_CHARS);
883 rt->accept_dynamic_rate = atoi(p);
884 } else if (av_stristart(p, "Content-Type:", &p)) {
885 p += strspn(p, SPACE_CHARS);
886 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
890 /* skip a RTP/TCP interleaved packet */
891 void ff_rtsp_skip_packet(AVFormatContext *s)
893 RTSPState *rt = s->priv_data;
897 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
900 len = AV_RB16(buf + 1);
902 av_dlog(s, "skipping RTP packet len=%d\n", len);
907 if (len1 > sizeof(buf))
909 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
916 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
917 unsigned char **content_ptr,
918 int return_on_interleaved_data, const char *method)
920 RTSPState *rt = s->priv_data;
921 char buf[4096], buf1[1024], *q;
924 int ret, content_length, line_count = 0, request = 0;
925 unsigned char *content = NULL;
931 memset(reply, 0, sizeof(*reply));
933 /* parse reply (XXX: use buffers) */
934 rt->last_reply[0] = '\0';
938 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
939 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
945 /* XXX: only parse it if first char on line ? */
946 if (return_on_interleaved_data) {
949 ff_rtsp_skip_packet(s);
950 } else if (ch != '\r') {
951 if ((q - buf) < sizeof(buf) - 1)
957 av_dlog(s, "line='%s'\n", buf);
959 /* test if last line */
963 if (line_count == 0) {
965 get_word(buf1, sizeof(buf1), &p);
966 if (!strncmp(buf1, "RTSP/", 5)) {
967 get_word(buf1, sizeof(buf1), &p);
968 reply->status_code = atoi(buf1);
969 av_strlcpy(reply->reason, p, sizeof(reply->reason));
971 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
972 get_word(buf1, sizeof(buf1), &p); // object
976 ff_rtsp_parse_line(reply, p, rt, method);
977 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
978 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
983 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
984 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
986 content_length = reply->content_length;
987 if (content_length > 0) {
988 /* leave some room for a trailing '\0' (useful for simple parsing) */
989 content = av_malloc(content_length + 1);
990 ffurl_read_complete(rt->rtsp_hd, content, content_length);
991 content[content_length] = '\0';
994 *content_ptr = content;
1000 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1001 const char* ptr = buf;
1003 if (!strcmp(reply->reason, "OPTIONS")) {
1004 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1006 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1007 if (reply->session_id[0])
1008 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1011 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1013 av_strlcat(buf, "\r\n", sizeof(buf));
1015 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1016 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1019 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1021 rt->last_cmd_time = av_gettime();
1022 /* Even if the request from the server had data, it is not the data
1023 * that the caller wants or expects. The memory could also be leaked
1024 * if the actual following reply has content data. */
1026 av_freep(content_ptr);
1027 /* If method is set, this is called from ff_rtsp_send_cmd,
1028 * where a reply to exactly this request is awaited. For
1029 * callers from within packet receiving, we just want to
1030 * return to the caller and go back to receiving packets. */
1036 if (rt->seq != reply->seq) {
1037 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1038 rt->seq, reply->seq);
1042 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1043 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1044 reply->notice == 2306 /* Continuous Feed Terminated */) {
1045 rt->state = RTSP_STATE_IDLE;
1046 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1047 return AVERROR(EIO); /* data or server error */
1048 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1049 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1050 return AVERROR(EPERM);
1056 * Send a command to the RTSP server without waiting for the reply.
1058 * @param s RTSP (de)muxer context
1059 * @param method the method for the request
1060 * @param url the target url for the request
1061 * @param headers extra header lines to include in the request
1062 * @param send_content if non-null, the data to send as request body content
1063 * @param send_content_length the length of the send_content data, or 0 if
1064 * send_content is null
1066 * @return zero if success, nonzero otherwise
1068 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1069 const char *method, const char *url,
1070 const char *headers,
1071 const unsigned char *send_content,
1072 int send_content_length)
1074 RTSPState *rt = s->priv_data;
1075 char buf[4096], *out_buf;
1076 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1078 /* Add in RTSP headers */
1081 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1083 av_strlcat(buf, headers, sizeof(buf));
1084 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1085 if (rt->session_id[0] != '\0' && (!headers ||
1086 !strstr(headers, "\nIf-Match:"))) {
1087 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1090 char *str = ff_http_auth_create_response(&rt->auth_state,
1091 rt->auth, url, method);
1093 av_strlcat(buf, str, sizeof(buf));
1096 if (send_content_length > 0 && send_content)
1097 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1098 av_strlcat(buf, "\r\n", sizeof(buf));
1100 /* base64 encode rtsp if tunneling */
1101 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1102 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1103 out_buf = base64buf;
1106 av_dlog(s, "Sending:\n%s--\n", buf);
1108 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1109 if (send_content_length > 0 && send_content) {
1110 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1111 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1112 "with content data not supported\n");
1113 return AVERROR_PATCHWELCOME;
1115 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1117 rt->last_cmd_time = av_gettime();
1122 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1123 const char *url, const char *headers)
1125 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1128 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1129 const char *headers, RTSPMessageHeader *reply,
1130 unsigned char **content_ptr)
1132 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1133 content_ptr, NULL, 0);
1136 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1137 const char *method, const char *url,
1139 RTSPMessageHeader *reply,
1140 unsigned char **content_ptr,
1141 const unsigned char *send_content,
1142 int send_content_length)
1144 RTSPState *rt = s->priv_data;
1145 HTTPAuthType cur_auth_type;
1146 int ret, attempts = 0;
1149 cur_auth_type = rt->auth_state.auth_type;
1150 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1152 send_content_length)))
1155 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1159 if (reply->status_code == 401 &&
1160 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1161 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1164 if (reply->status_code > 400){
1165 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1169 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1175 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1176 int lower_transport, const char *real_challenge)
1178 RTSPState *rt = s->priv_data;
1179 int rtx = 0, j, i, err, interleave = 0, port_off;
1180 RTSPStream *rtsp_st;
1181 RTSPMessageHeader reply1, *reply = &reply1;
1183 const char *trans_pref;
1185 if (rt->transport == RTSP_TRANSPORT_RDT)
1186 trans_pref = "x-pn-tng";
1188 trans_pref = "RTP/AVP";
1190 /* default timeout: 1 minute */
1193 /* for each stream, make the setup request */
1194 /* XXX: we assume the same server is used for the control of each
1197 /* Choose a random starting offset within the first half of the
1198 * port range, to allow for a number of ports to try even if the offset
1199 * happens to be at the end of the random range. */
1200 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1201 /* even random offset */
1202 port_off -= port_off & 0x01;
1204 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1205 char transport[2048];
1208 * WMS serves all UDP data over a single connection, the RTX, which
1209 * isn't necessarily the first in the SDP but has to be the first
1210 * to be set up, else the second/third SETUP will fail with a 461.
1212 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1213 rt->server_type == RTSP_SERVER_WMS) {
1216 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1217 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1219 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1223 if (rtx == rt->nb_rtsp_streams)
1224 return -1; /* no RTX found */
1225 rtsp_st = rt->rtsp_streams[rtx];
1227 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1229 rtsp_st = rt->rtsp_streams[i];
1232 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1235 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1236 port = reply->transports[0].client_port_min;
1240 /* first try in specified port range */
1241 while (j <= rt->rtp_port_max) {
1242 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1243 "?localport=%d", j);
1244 /* we will use two ports per rtp stream (rtp and rtcp) */
1246 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1247 &s->interrupt_callback, NULL))
1251 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1256 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1258 snprintf(transport, sizeof(transport) - 1,
1259 "%s/UDP;", trans_pref);
1260 if (rt->server_type != RTSP_SERVER_REAL)
1261 av_strlcat(transport, "unicast;", sizeof(transport));
1262 av_strlcatf(transport, sizeof(transport),
1263 "client_port=%d", port);
1264 if (rt->transport == RTSP_TRANSPORT_RTP &&
1265 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1266 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1270 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1271 /* For WMS streams, the application streams are only used for
1272 * UDP. When trying to set it up for TCP streams, the server
1273 * will return an error. Therefore, we skip those streams. */
1274 if (rt->server_type == RTSP_SERVER_WMS &&
1275 (rtsp_st->stream_index < 0 ||
1276 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1279 snprintf(transport, sizeof(transport) - 1,
1280 "%s/TCP;", trans_pref);
1281 if (rt->transport != RTSP_TRANSPORT_RDT)
1282 av_strlcat(transport, "unicast;", sizeof(transport));
1283 av_strlcatf(transport, sizeof(transport),
1284 "interleaved=%d-%d",
1285 interleave, interleave + 1);
1289 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1290 snprintf(transport, sizeof(transport) - 1,
1291 "%s/UDP;multicast", trans_pref);
1294 av_strlcat(transport, ";mode=record", sizeof(transport));
1295 } else if (rt->server_type == RTSP_SERVER_REAL ||
1296 rt->server_type == RTSP_SERVER_WMS)
1297 av_strlcat(transport, ";mode=play", sizeof(transport));
1298 snprintf(cmd, sizeof(cmd),
1299 "Transport: %s\r\n",
1301 if (rt->accept_dynamic_rate)
1302 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1303 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1304 char real_res[41], real_csum[9];
1305 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1307 av_strlcatf(cmd, sizeof(cmd),
1309 "RealChallenge2: %s, sd=%s\r\n",
1310 rt->session_id, real_res, real_csum);
1312 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1313 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1316 } else if (reply->status_code != RTSP_STATUS_OK ||
1317 reply->nb_transports != 1) {
1318 err = AVERROR_INVALIDDATA;
1322 /* XXX: same protocol for all streams is required */
1324 if (reply->transports[0].lower_transport != rt->lower_transport ||
1325 reply->transports[0].transport != rt->transport) {
1326 err = AVERROR_INVALIDDATA;
1330 rt->lower_transport = reply->transports[0].lower_transport;
1331 rt->transport = reply->transports[0].transport;
1334 /* Fail if the server responded with another lower transport mode
1335 * than what we requested. */
1336 if (reply->transports[0].lower_transport != lower_transport) {
1337 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1338 err = AVERROR_INVALIDDATA;
1342 switch(reply->transports[0].lower_transport) {
1343 case RTSP_LOWER_TRANSPORT_TCP:
1344 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1345 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1348 case RTSP_LOWER_TRANSPORT_UDP: {
1349 char url[1024], options[30] = "";
1351 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1352 av_strlcpy(options, "?connect=1", sizeof(options));
1353 /* Use source address if specified */
1354 if (reply->transports[0].source[0]) {
1355 ff_url_join(url, sizeof(url), "rtp", NULL,
1356 reply->transports[0].source,
1357 reply->transports[0].server_port_min, "%s", options);
1359 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1360 reply->transports[0].server_port_min, "%s", options);
1362 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1363 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1364 err = AVERROR_INVALIDDATA;
1367 /* Try to initialize the connection state in a
1368 * potential NAT router by sending dummy packets.
1369 * RTP/RTCP dummy packets are used for RDT, too.
1371 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1373 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1376 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1377 char url[1024], namebuf[50], optbuf[20] = "";
1378 struct sockaddr_storage addr;
1381 if (reply->transports[0].destination.ss_family) {
1382 addr = reply->transports[0].destination;
1383 port = reply->transports[0].port_min;
1384 ttl = reply->transports[0].ttl;
1386 addr = rtsp_st->sdp_ip;
1387 port = rtsp_st->sdp_port;
1388 ttl = rtsp_st->sdp_ttl;
1391 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1392 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1393 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1394 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1395 port, "%s", optbuf);
1396 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1397 &s->interrupt_callback, NULL) < 0) {
1398 err = AVERROR_INVALIDDATA;
1405 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1409 if (rt->nb_rtsp_streams && reply->timeout > 0)
1410 rt->timeout = reply->timeout;
1412 if (rt->server_type == RTSP_SERVER_REAL)
1413 rt->need_subscription = 1;
1418 ff_rtsp_undo_setup(s);
1422 void ff_rtsp_close_connections(AVFormatContext *s)
1424 RTSPState *rt = s->priv_data;
1425 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1426 ffurl_close(rt->rtsp_hd);
1427 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1430 int ff_rtsp_connect(AVFormatContext *s)
1432 RTSPState *rt = s->priv_data;
1433 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1434 int port, err, tcp_fd;
1435 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1436 int lower_transport_mask = 0;
1437 char real_challenge[64] = "";
1438 struct sockaddr_storage peer;
1439 socklen_t peer_len = sizeof(peer);
1441 if (rt->rtp_port_max < rt->rtp_port_min) {
1442 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1443 "than min port %d\n", rt->rtp_port_max,
1445 return AVERROR(EINVAL);
1448 if (!ff_network_init())
1449 return AVERROR(EIO);
1451 if (s->max_delay < 0) /* Not set by the caller */
1452 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1454 rt->control_transport = RTSP_MODE_PLAIN;
1455 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1456 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1457 rt->control_transport = RTSP_MODE_TUNNEL;
1459 /* Only pass through valid flags from here */
1460 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1463 lower_transport_mask = rt->lower_transport_mask;
1464 /* extract hostname and port */
1465 av_url_split(NULL, 0, auth, sizeof(auth),
1466 host, sizeof(host), &port, path, sizeof(path), s->filename);
1468 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1471 port = RTSP_DEFAULT_PORT;
1473 if (!lower_transport_mask)
1474 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1477 /* Only UDP or TCP - UDP multicast isn't supported. */
1478 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1479 (1 << RTSP_LOWER_TRANSPORT_TCP);
1480 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1481 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1482 "only UDP and TCP are supported for output.\n");
1483 err = AVERROR(EINVAL);
1488 /* Construct the URI used in request; this is similar to s->filename,
1489 * but with authentication credentials removed and RTSP specific options
1491 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1492 host, port, "%s", path);
1494 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1495 /* set up initial handshake for tunneling */
1496 char httpname[1024];
1497 char sessioncookie[17];
1500 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1501 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1502 av_get_random_seed(), av_get_random_seed());
1505 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1506 &s->interrupt_callback) < 0) {
1511 /* generate GET headers */
1512 snprintf(headers, sizeof(headers),
1513 "x-sessioncookie: %s\r\n"
1514 "Accept: application/x-rtsp-tunnelled\r\n"
1515 "Pragma: no-cache\r\n"
1516 "Cache-Control: no-cache\r\n",
1518 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1520 /* complete the connection */
1521 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1527 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1528 &s->interrupt_callback) < 0 ) {
1533 /* generate POST headers */
1534 snprintf(headers, sizeof(headers),
1535 "x-sessioncookie: %s\r\n"
1536 "Content-Type: application/x-rtsp-tunnelled\r\n"
1537 "Pragma: no-cache\r\n"
1538 "Cache-Control: no-cache\r\n"
1539 "Content-Length: 32767\r\n"
1540 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1542 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1543 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1545 /* Initialize the authentication state for the POST session. The HTTP
1546 * protocol implementation doesn't properly handle multi-pass
1547 * authentication for POST requests, since it would require one of
1549 * - implementing Expect: 100-continue, which many HTTP servers
1550 * don't support anyway, even less the RTSP servers that do HTTP
1552 * - sending the whole POST data until getting a 401 reply specifying
1553 * what authentication method to use, then resending all that data
1554 * - waiting for potential 401 replies directly after sending the
1555 * POST header (waiting for some unspecified time)
1556 * Therefore, we copy the full auth state, which works for both basic
1557 * and digest. (For digest, we would have to synchronize the nonce
1558 * count variable between the two sessions, if we'd do more requests
1559 * with the original session, though.)
1561 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1563 /* complete the connection */
1564 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1569 /* open the tcp connection */
1570 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1571 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1572 &s->interrupt_callback, NULL) < 0) {
1576 rt->rtsp_hd_out = rt->rtsp_hd;
1580 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1581 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1582 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1583 NULL, 0, NI_NUMERICHOST);
1586 /* request options supported by the server; this also detects server
1588 for (rt->server_type = RTSP_SERVER_RTP;;) {
1590 if (rt->server_type == RTSP_SERVER_REAL)
1593 * The following entries are required for proper
1594 * streaming from a Realmedia server. They are
1595 * interdependent in some way although we currently
1596 * don't quite understand how. Values were copied
1597 * from mplayer SVN r23589.
1598 * ClientChallenge is a 16-byte ID in hex
1599 * CompanyID is a 16-byte ID in base64
1601 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1602 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1603 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1604 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1606 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1607 if (reply->status_code != RTSP_STATUS_OK) {
1608 err = AVERROR_INVALIDDATA;
1612 /* detect server type if not standard-compliant RTP */
1613 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1614 rt->server_type = RTSP_SERVER_REAL;
1616 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1617 rt->server_type = RTSP_SERVER_WMS;
1618 } else if (rt->server_type == RTSP_SERVER_REAL)
1619 strcpy(real_challenge, reply->real_challenge);
1623 if (s->iformat && CONFIG_RTSP_DEMUXER)
1624 err = ff_rtsp_setup_input_streams(s, reply);
1625 else if (CONFIG_RTSP_MUXER)
1626 err = ff_rtsp_setup_output_streams(s, host);
1631 int lower_transport = ff_log2_tab[lower_transport_mask &
1632 ~(lower_transport_mask - 1)];
1634 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1635 rt->server_type == RTSP_SERVER_REAL ?
1636 real_challenge : NULL);
1639 lower_transport_mask &= ~(1 << lower_transport);
1640 if (lower_transport_mask == 0 && err == 1) {
1641 err = AVERROR(EPROTONOSUPPORT);
1646 rt->lower_transport_mask = lower_transport_mask;
1647 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1648 rt->state = RTSP_STATE_IDLE;
1649 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1652 ff_rtsp_close_streams(s);
1653 ff_rtsp_close_connections(s);
1654 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1655 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1656 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1664 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1667 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1668 uint8_t *buf, int buf_size, int64_t wait_end)
1670 RTSPState *rt = s->priv_data;
1671 RTSPStream *rtsp_st;
1672 int n, i, ret, tcp_fd, timeout_cnt = 0;
1674 struct pollfd *p = rt->p;
1677 if (ff_check_interrupt(&s->interrupt_callback))
1678 return AVERROR_EXIT;
1679 if (wait_end && wait_end - av_gettime() < 0)
1680 return AVERROR(EAGAIN);
1683 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1684 p[max_p].fd = tcp_fd;
1685 p[max_p++].events = POLLIN;
1689 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1690 rtsp_st = rt->rtsp_streams[i];
1691 if (rtsp_st->rtp_handle) {
1692 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1693 p[max_p++].events = POLLIN;
1694 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1695 p[max_p++].events = POLLIN;
1698 n = poll(p, max_p, POLL_TIMEOUT_MS);
1700 int j = 1 - (tcp_fd == -1);
1702 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1703 rtsp_st = rt->rtsp_streams[i];
1704 if (rtsp_st->rtp_handle) {
1705 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1706 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1708 *prtsp_st = rtsp_st;
1715 #if CONFIG_RTSP_DEMUXER
1716 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1717 RTSPMessageHeader reply;
1719 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1722 /* XXX: parse message */
1723 if (rt->state != RTSP_STATE_STREAMING)
1727 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1728 return AVERROR(ETIMEDOUT);
1729 } else if (n < 0 && errno != EINTR)
1730 return AVERROR(errno);
1734 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1736 RTSPState *rt = s->priv_data;
1738 RTSPStream *rtsp_st, *first_queue_st = NULL;
1739 int64_t wait_end = 0;
1741 if (rt->nb_byes == rt->nb_rtsp_streams)
1744 /* get next frames from the same RTP packet */
1745 if (rt->cur_transport_priv) {
1746 if (rt->transport == RTSP_TRANSPORT_RDT) {
1747 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1749 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1751 rt->cur_transport_priv = NULL;
1753 } else if (ret == 1) {
1756 rt->cur_transport_priv = NULL;
1759 if (rt->transport == RTSP_TRANSPORT_RTP) {
1761 int64_t first_queue_time = 0;
1762 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1763 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1767 queue_time = ff_rtp_queued_packet_time(rtpctx);
1768 if (queue_time && (queue_time - first_queue_time < 0 ||
1769 !first_queue_time)) {
1770 first_queue_time = queue_time;
1771 first_queue_st = rt->rtsp_streams[i];
1774 if (first_queue_time)
1775 wait_end = first_queue_time + s->max_delay;
1778 /* read next RTP packet */
1781 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1783 return AVERROR(ENOMEM);
1786 switch(rt->lower_transport) {
1788 #if CONFIG_RTSP_DEMUXER
1789 case RTSP_LOWER_TRANSPORT_TCP:
1790 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1793 case RTSP_LOWER_TRANSPORT_UDP:
1794 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1795 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1796 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1797 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1800 if (len == AVERROR(EAGAIN) && first_queue_st &&
1801 rt->transport == RTSP_TRANSPORT_RTP) {
1802 rtsp_st = first_queue_st;
1803 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1810 if (rt->transport == RTSP_TRANSPORT_RDT) {
1811 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1813 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1815 /* Either bad packet, or a RTCP packet. Check if the
1816 * first_rtcp_ntp_time field was initialized. */
1817 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1818 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1819 /* first_rtcp_ntp_time has been initialized for this stream,
1820 * copy the same value to all other uninitialized streams,
1821 * in order to map their timestamp origin to the same ntp time
1824 AVStream *st = NULL;
1825 if (rtsp_st->stream_index >= 0)
1826 st = s->streams[rtsp_st->stream_index];
1827 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1828 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1829 AVStream *st2 = NULL;
1830 if (rt->rtsp_streams[i]->stream_index >= 0)
1831 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1832 if (rtpctx2 && st && st2 &&
1833 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1834 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1835 rtpctx2->rtcp_ts_offset = av_rescale_q(
1836 rtpctx->rtcp_ts_offset, st->time_base,
1841 if (ret == -RTCP_BYE) {
1844 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1845 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1847 if (rt->nb_byes == rt->nb_rtsp_streams)
1856 /* more packets may follow, so we save the RTP context */
1857 rt->cur_transport_priv = rtsp_st->transport_priv;
1861 #endif /* CONFIG_RTPDEC */
1863 #if CONFIG_SDP_DEMUXER
1864 static int sdp_probe(AVProbeData *p1)
1866 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1868 /* we look for a line beginning "c=IN IP" */
1869 while (p < p_end && *p != '\0') {
1870 if (p + sizeof("c=IN IP") - 1 < p_end &&
1871 av_strstart(p, "c=IN IP", NULL))
1872 return AVPROBE_SCORE_MAX / 2;
1874 while (p < p_end - 1 && *p != '\n') p++;
1883 static int sdp_read_header(AVFormatContext *s)
1885 RTSPState *rt = s->priv_data;
1886 RTSPStream *rtsp_st;
1891 if (!ff_network_init())
1892 return AVERROR(EIO);
1894 if (s->max_delay < 0) /* Not set by the caller */
1895 s->max_delay = DEFAULT_REORDERING_DELAY;
1897 /* read the whole sdp file */
1898 /* XXX: better loading */
1899 content = av_malloc(SDP_MAX_SIZE);
1900 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1903 return AVERROR_INVALIDDATA;
1905 content[size] ='\0';
1907 err = ff_sdp_parse(s, content);
1911 /* open each RTP stream */
1912 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1914 rtsp_st = rt->rtsp_streams[i];
1916 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1917 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1918 ff_url_join(url, sizeof(url), "rtp", NULL,
1919 namebuf, rtsp_st->sdp_port,
1920 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1922 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1923 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1924 &s->interrupt_callback, NULL) < 0) {
1925 err = AVERROR_INVALIDDATA;
1928 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1933 ff_rtsp_close_streams(s);
1938 static int sdp_read_close(AVFormatContext *s)
1940 ff_rtsp_close_streams(s);
1945 static const AVClass sdp_demuxer_class = {
1946 .class_name = "SDP demuxer",
1947 .item_name = av_default_item_name,
1948 .option = sdp_options,
1949 .version = LIBAVUTIL_VERSION_INT,
1952 AVInputFormat ff_sdp_demuxer = {
1954 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1955 .priv_data_size = sizeof(RTSPState),
1956 .read_probe = sdp_probe,
1957 .read_header = sdp_read_header,
1958 .read_packet = ff_rtsp_fetch_packet,
1959 .read_close = sdp_read_close,
1960 .priv_class = &sdp_demuxer_class,
1962 #endif /* CONFIG_SDP_DEMUXER */
1964 #if CONFIG_RTP_DEMUXER
1965 static int rtp_probe(AVProbeData *p)
1967 if (av_strstart(p->filename, "rtp:", NULL))
1968 return AVPROBE_SCORE_MAX;
1972 static int rtp_read_header(AVFormatContext *s)
1974 uint8_t recvbuf[1500];
1975 char host[500], sdp[500];
1977 URLContext* in = NULL;
1979 AVCodecContext codec = { 0 };
1980 struct sockaddr_storage addr;
1982 socklen_t addrlen = sizeof(addr);
1983 RTSPState *rt = s->priv_data;
1985 if (!ff_network_init())
1986 return AVERROR(EIO);
1988 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1989 &s->interrupt_callback, NULL);
1994 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1995 if (ret == AVERROR(EAGAIN))
2000 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2004 if ((recvbuf[0] & 0xc0) != 0x80) {
2005 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2010 if (RTP_PT_IS_RTCP(recvbuf[1]))
2013 payload_type = recvbuf[1] & 0x7f;
2016 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2020 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2021 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2022 "without an SDP file describing it\n",
2026 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2027 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2028 "properly you need an SDP file "
2032 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2033 NULL, 0, s->filename);
2035 snprintf(sdp, sizeof(sdp),
2036 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2037 addr.ss_family == AF_INET ? 4 : 6, host,
2038 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2039 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2040 port, payload_type);
2041 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2043 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2046 /* sdp_read_header initializes this again */
2049 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2051 ret = sdp_read_header(s);
2062 static const AVClass rtp_demuxer_class = {
2063 .class_name = "RTP demuxer",
2064 .item_name = av_default_item_name,
2065 .option = rtp_options,
2066 .version = LIBAVUTIL_VERSION_INT,
2069 AVInputFormat ff_rtp_demuxer = {
2071 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2072 .priv_data_size = sizeof(RTSPState),
2073 .read_probe = rtp_probe,
2074 .read_header = rtp_read_header,
2075 .read_packet = ff_rtsp_fetch_packet,
2076 .read_close = sdp_read_close,
2077 .flags = AVFMT_NOFILE,
2078 .priv_class = &rtp_demuxer_class,
2080 #endif /* CONFIG_RTP_DEMUXER */