3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
31 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define OFFSET(x) offsetof(RTSPState, x)
61 #define DEC AV_OPT_FLAG_DECODING_PARAM
62 #define ENC AV_OPT_FLAG_ENCODING_PARAM
64 #define RTSP_FLAG_OPTS(name, longname) \
65 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
66 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
68 #define RTSP_MEDIATYPE_OPTS(name, longname) \
69 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
70 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
71 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
74 const AVOption ff_rtsp_options[] = {
75 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
76 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
77 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
78 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
79 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
81 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
82 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
83 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
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, *ai = NULL;
160 memset(&hints, 0, sizeof(hints));
161 hints.ai_flags = AI_NUMERICHOST;
162 if (getaddrinfo(buf, NULL, &hints, &ai))
164 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
170 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
171 RTSPStream *rtsp_st, AVCodecContext *codec)
175 codec->codec_id = handler->codec_id;
176 rtsp_st->dynamic_handler = handler;
177 if (handler->alloc) {
178 rtsp_st->dynamic_protocol_context = handler->alloc();
179 if (!rtsp_st->dynamic_protocol_context)
180 rtsp_st->dynamic_handler = NULL;
184 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
185 static int sdp_parse_rtpmap(AVFormatContext *s,
186 AVStream *st, RTSPStream *rtsp_st,
187 int payload_type, const char *p)
189 AVCodecContext *codec = st->codec;
195 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
196 * see if we can handle this kind of payload.
197 * The space should normally not be there but some Real streams or
198 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
199 * have a trailing space. */
200 get_word_sep(buf, sizeof(buf), "/ ", &p);
201 if (payload_type >= RTP_PT_PRIVATE) {
202 RTPDynamicProtocolHandler *handler =
203 ff_rtp_handler_find_by_name(buf, codec->codec_type);
204 init_rtp_handler(handler, rtsp_st, codec);
205 /* If no dynamic handler was found, check with the list of standard
206 * allocated types, if such a stream for some reason happens to
207 * use a private payload type. This isn't handled in rtpdec.c, since
208 * the format name from the rtpmap line never is passed into rtpdec. */
209 if (!rtsp_st->dynamic_handler)
210 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
212 /* We are in a standard case
213 * (from http://www.iana.org/assignments/rtp-parameters). */
214 /* search into AVRtpPayloadTypes[] */
215 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
218 c = avcodec_find_decoder(codec->codec_id);
224 get_word_sep(buf, sizeof(buf), "/", &p);
226 switch (codec->codec_type) {
227 case AVMEDIA_TYPE_AUDIO:
228 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
229 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
230 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
232 codec->sample_rate = i;
233 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
234 get_word_sep(buf, sizeof(buf), "/", &p);
238 // TODO: there is a bug here; if it is a mono stream, and
239 // less than 22000Hz, faad upconverts to stereo and twice
240 // the frequency. No problem, but the sample rate is being
241 // set here by the sdp line. Patch on its way. (rdm)
243 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
245 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
248 case AVMEDIA_TYPE_VIDEO:
249 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
251 avpriv_set_pts_info(st, 32, 1, i);
256 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
257 rtsp_st->dynamic_handler->init(s, st->index,
258 rtsp_st->dynamic_protocol_context);
262 /* parse the attribute line from the fmtp a line of an sdp response. This
263 * is broken out as a function because it is used in rtp_h264.c, which is
265 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
266 char *value, int value_size)
268 *p += strspn(*p, SPACE_CHARS);
270 get_word_sep(attr, attr_size, "=", p);
273 get_word_sep(value, value_size, ";", p);
281 typedef struct SDPParseState {
283 struct sockaddr_storage default_ip;
285 int skip_media; ///< set if an unknown m= line occurs
288 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
289 int letter, const char *buf)
291 RTSPState *rt = s->priv_data;
292 char buf1[64], st_type[64];
294 enum AVMediaType codec_type;
298 struct sockaddr_storage sdp_ip;
301 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
304 if (s1->skip_media && letter != 'm')
308 get_word(buf1, sizeof(buf1), &p);
309 if (strcmp(buf1, "IN") != 0)
311 get_word(buf1, sizeof(buf1), &p);
312 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
314 get_word_sep(buf1, sizeof(buf1), "/", &p);
315 if (get_sockaddr(buf1, &sdp_ip))
320 get_word_sep(buf1, sizeof(buf1), "/", &p);
323 if (s->nb_streams == 0) {
324 s1->default_ip = sdp_ip;
325 s1->default_ttl = ttl;
327 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
328 rtsp_st->sdp_ip = sdp_ip;
329 rtsp_st->sdp_ttl = ttl;
333 av_dict_set(&s->metadata, "title", p, 0);
336 if (s->nb_streams == 0) {
337 av_dict_set(&s->metadata, "comment", p, 0);
344 codec_type = AVMEDIA_TYPE_UNKNOWN;
345 get_word(st_type, sizeof(st_type), &p);
346 if (!strcmp(st_type, "audio")) {
347 codec_type = AVMEDIA_TYPE_AUDIO;
348 } else if (!strcmp(st_type, "video")) {
349 codec_type = AVMEDIA_TYPE_VIDEO;
350 } else if (!strcmp(st_type, "application")) {
351 codec_type = AVMEDIA_TYPE_DATA;
353 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
357 rtsp_st = av_mallocz(sizeof(RTSPStream));
360 rtsp_st->stream_index = -1;
361 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
363 rtsp_st->sdp_ip = s1->default_ip;
364 rtsp_st->sdp_ttl = s1->default_ttl;
366 get_word(buf1, sizeof(buf1), &p); /* port */
367 rtsp_st->sdp_port = atoi(buf1);
369 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
371 /* XXX: handle list of formats */
372 get_word(buf1, sizeof(buf1), &p); /* format list */
373 rtsp_st->sdp_payload_type = atoi(buf1);
375 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
376 /* no corresponding stream */
378 st = avformat_new_stream(s, NULL);
381 st->id = rt->nb_rtsp_streams - 1;
382 rtsp_st->stream_index = st->index;
383 st->codec->codec_type = codec_type;
384 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
385 RTPDynamicProtocolHandler *handler;
386 /* if standard payload type, we can find the codec right now */
387 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
388 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
389 st->codec->sample_rate > 0)
390 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
391 /* Even static payload types may need a custom depacketizer */
392 handler = ff_rtp_handler_find_by_id(
393 rtsp_st->sdp_payload_type, st->codec->codec_type);
394 init_rtp_handler(handler, rtsp_st, st->codec);
395 if (handler && handler->init)
396 handler->init(s, st->index,
397 rtsp_st->dynamic_protocol_context);
400 /* put a default control url */
401 av_strlcpy(rtsp_st->control_url, rt->control_uri,
402 sizeof(rtsp_st->control_url));
405 if (av_strstart(p, "control:", &p)) {
406 if (s->nb_streams == 0) {
407 if (!strncmp(p, "rtsp://", 7))
408 av_strlcpy(rt->control_uri, p,
409 sizeof(rt->control_uri));
412 /* get the control url */
413 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
415 /* XXX: may need to add full url resolution */
416 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
418 if (proto[0] == '\0') {
419 /* relative control URL */
420 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
421 av_strlcat(rtsp_st->control_url, "/",
422 sizeof(rtsp_st->control_url));
423 av_strlcat(rtsp_st->control_url, p,
424 sizeof(rtsp_st->control_url));
426 av_strlcpy(rtsp_st->control_url, p,
427 sizeof(rtsp_st->control_url));
429 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
430 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
431 get_word(buf1, sizeof(buf1), &p);
432 payload_type = atoi(buf1);
433 st = s->streams[s->nb_streams - 1];
434 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
435 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
436 } else if (av_strstart(p, "fmtp:", &p) ||
437 av_strstart(p, "framesize:", &p)) {
438 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
439 // let dynamic protocol handlers have a stab at the line.
440 get_word(buf1, sizeof(buf1), &p);
441 payload_type = atoi(buf1);
442 for (i = 0; i < rt->nb_rtsp_streams; i++) {
443 rtsp_st = rt->rtsp_streams[i];
444 if (rtsp_st->sdp_payload_type == payload_type &&
445 rtsp_st->dynamic_handler &&
446 rtsp_st->dynamic_handler->parse_sdp_a_line)
447 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
448 rtsp_st->dynamic_protocol_context, buf);
450 } else if (av_strstart(p, "range:", &p)) {
453 // this is so that seeking on a streamed file can work.
454 rtsp_parse_range_npt(p, &start, &end);
455 s->start_time = start;
456 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
457 s->duration = (end == AV_NOPTS_VALUE) ?
458 AV_NOPTS_VALUE : end - start;
459 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
461 rt->transport = RTSP_TRANSPORT_RDT;
462 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
464 st = s->streams[s->nb_streams - 1];
465 st->codec->sample_rate = atoi(p);
467 if (rt->server_type == RTSP_SERVER_WMS)
468 ff_wms_parse_sdp_a_line(s, p);
469 if (s->nb_streams > 0) {
470 if (rt->server_type == RTSP_SERVER_REAL)
471 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
473 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
474 if (rtsp_st->dynamic_handler &&
475 rtsp_st->dynamic_handler->parse_sdp_a_line)
476 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
478 rtsp_st->dynamic_protocol_context, buf);
485 int ff_sdp_parse(AVFormatContext *s, const char *content)
487 RTSPState *rt = s->priv_data;
490 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
491 * contain long SDP lines containing complete ASF Headers (several
492 * kB) or arrays of MDPR (RM stream descriptor) headers plus
493 * "rulebooks" describing their properties. Therefore, the SDP line
496 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
497 * in rtpdec_xiph.c. */
499 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
501 memset(s1, 0, sizeof(SDPParseState));
504 p += strspn(p, SPACE_CHARS);
512 /* get the content */
514 while (*p != '\n' && *p != '\r' && *p != '\0') {
515 if ((q - buf) < sizeof(buf) - 1)
520 sdp_parse_line(s, s1, letter, buf);
522 while (*p != '\n' && *p != '\0')
527 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
528 if (!rt->p) return AVERROR(ENOMEM);
531 #endif /* CONFIG_RTPDEC */
533 void ff_rtsp_undo_setup(AVFormatContext *s)
535 RTSPState *rt = s->priv_data;
538 for (i = 0; i < rt->nb_rtsp_streams; i++) {
539 RTSPStream *rtsp_st = rt->rtsp_streams[i];
542 if (rtsp_st->transport_priv) {
544 AVFormatContext *rtpctx = rtsp_st->transport_priv;
545 av_write_trailer(rtpctx);
546 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
548 avio_close_dyn_buf(rtpctx->pb, &ptr);
551 avio_close(rtpctx->pb);
553 avformat_free_context(rtpctx);
554 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
555 ff_rdt_parse_close(rtsp_st->transport_priv);
556 else if (CONFIG_RTPDEC)
557 ff_rtp_parse_close(rtsp_st->transport_priv);
559 rtsp_st->transport_priv = NULL;
560 if (rtsp_st->rtp_handle)
561 ffurl_close(rtsp_st->rtp_handle);
562 rtsp_st->rtp_handle = NULL;
566 /* close and free RTSP streams */
567 void ff_rtsp_close_streams(AVFormatContext *s)
569 RTSPState *rt = s->priv_data;
573 ff_rtsp_undo_setup(s);
574 for (i = 0; i < rt->nb_rtsp_streams; i++) {
575 rtsp_st = rt->rtsp_streams[i];
577 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
578 rtsp_st->dynamic_handler->free(
579 rtsp_st->dynamic_protocol_context);
583 av_free(rt->rtsp_streams);
585 avformat_close_input(&rt->asf_ctx);
588 av_free(rt->recvbuf);
591 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
593 RTSPState *rt = s->priv_data;
596 /* open the RTP context */
597 if (rtsp_st->stream_index >= 0)
598 st = s->streams[rtsp_st->stream_index];
600 s->ctx_flags |= AVFMTCTX_NOHEADER;
602 if (s->oformat && CONFIG_RTSP_MUXER) {
603 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
605 RTSP_TCP_MAX_PACKET_SIZE);
606 /* Ownership of rtp_handle is passed to the rtp mux context */
607 rtsp_st->rtp_handle = NULL;
608 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
609 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
610 rtsp_st->dynamic_protocol_context,
611 rtsp_st->dynamic_handler);
612 else if (CONFIG_RTPDEC)
613 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
614 rtsp_st->sdp_payload_type,
615 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
616 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
618 if (!rtsp_st->transport_priv) {
619 return AVERROR(ENOMEM);
620 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
621 if (rtsp_st->dynamic_handler) {
622 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
623 rtsp_st->dynamic_protocol_context,
624 rtsp_st->dynamic_handler);
631 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
632 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
638 p += strspn(p, SPACE_CHARS);
639 v = strtol(p, (char **)&p, 10);
643 v = strtol(p, (char **)&p, 10);
652 /* XXX: only one transport specification is parsed */
653 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
655 char transport_protocol[16];
657 char lower_transport[16];
659 RTSPTransportField *th;
662 reply->nb_transports = 0;
665 p += strspn(p, SPACE_CHARS);
669 th = &reply->transports[reply->nb_transports];
671 get_word_sep(transport_protocol, sizeof(transport_protocol),
673 if (!av_strcasecmp (transport_protocol, "rtp")) {
674 get_word_sep(profile, sizeof(profile), "/;,", &p);
675 lower_transport[0] = '\0';
676 /* rtp/avp/<protocol> */
678 get_word_sep(lower_transport, sizeof(lower_transport),
681 th->transport = RTSP_TRANSPORT_RTP;
682 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
683 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
684 /* x-pn-tng/<protocol> */
685 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
687 th->transport = RTSP_TRANSPORT_RDT;
689 if (!av_strcasecmp(lower_transport, "TCP"))
690 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
692 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
696 /* get each parameter */
697 while (*p != '\0' && *p != ',') {
698 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
699 if (!strcmp(parameter, "port")) {
702 rtsp_parse_range(&th->port_min, &th->port_max, &p);
704 } else if (!strcmp(parameter, "client_port")) {
707 rtsp_parse_range(&th->client_port_min,
708 &th->client_port_max, &p);
710 } else if (!strcmp(parameter, "server_port")) {
713 rtsp_parse_range(&th->server_port_min,
714 &th->server_port_max, &p);
716 } else if (!strcmp(parameter, "interleaved")) {
719 rtsp_parse_range(&th->interleaved_min,
720 &th->interleaved_max, &p);
722 } else if (!strcmp(parameter, "multicast")) {
723 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
724 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
725 } else if (!strcmp(parameter, "ttl")) {
728 th->ttl = strtol(p, (char **)&p, 10);
730 } else if (!strcmp(parameter, "destination")) {
733 get_word_sep(buf, sizeof(buf), ";,", &p);
734 get_sockaddr(buf, &th->destination);
736 } else if (!strcmp(parameter, "source")) {
739 get_word_sep(buf, sizeof(buf), ";,", &p);
740 av_strlcpy(th->source, buf, sizeof(th->source));
744 while (*p != ';' && *p != '\0' && *p != ',')
752 reply->nb_transports++;
756 static void handle_rtp_info(RTSPState *rt, const char *url,
757 uint32_t seq, uint32_t rtptime)
760 if (!rtptime || !url[0])
762 if (rt->transport != RTSP_TRANSPORT_RTP)
764 for (i = 0; i < rt->nb_rtsp_streams; i++) {
765 RTSPStream *rtsp_st = rt->rtsp_streams[i];
766 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
769 if (!strcmp(rtsp_st->control_url, url)) {
770 rtpctx->base_timestamp = rtptime;
776 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
779 char key[20], value[1024], url[1024] = "";
780 uint32_t seq = 0, rtptime = 0;
783 p += strspn(p, SPACE_CHARS);
786 get_word_sep(key, sizeof(key), "=", &p);
790 get_word_sep(value, sizeof(value), ";, ", &p);
792 if (!strcmp(key, "url"))
793 av_strlcpy(url, value, sizeof(url));
794 else if (!strcmp(key, "seq"))
795 seq = strtoul(value, NULL, 10);
796 else if (!strcmp(key, "rtptime"))
797 rtptime = strtoul(value, NULL, 10);
799 handle_rtp_info(rt, url, seq, rtptime);
808 handle_rtp_info(rt, url, seq, rtptime);
811 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
812 RTSPState *rt, const char *method)
816 /* NOTE: we do case independent match for broken servers */
818 if (av_stristart(p, "Session:", &p)) {
820 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
821 if (av_stristart(p, ";timeout=", &p) &&
822 (t = strtol(p, NULL, 10)) > 0) {
825 } else if (av_stristart(p, "Content-Length:", &p)) {
826 reply->content_length = strtol(p, NULL, 10);
827 } else if (av_stristart(p, "Transport:", &p)) {
828 rtsp_parse_transport(reply, p);
829 } else if (av_stristart(p, "CSeq:", &p)) {
830 reply->seq = strtol(p, NULL, 10);
831 } else if (av_stristart(p, "Range:", &p)) {
832 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
833 } else if (av_stristart(p, "RealChallenge1:", &p)) {
834 p += strspn(p, SPACE_CHARS);
835 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
836 } else if (av_stristart(p, "Server:", &p)) {
837 p += strspn(p, SPACE_CHARS);
838 av_strlcpy(reply->server, p, sizeof(reply->server));
839 } else if (av_stristart(p, "Notice:", &p) ||
840 av_stristart(p, "X-Notice:", &p)) {
841 reply->notice = strtol(p, NULL, 10);
842 } else if (av_stristart(p, "Location:", &p)) {
843 p += strspn(p, SPACE_CHARS);
844 av_strlcpy(reply->location, p , sizeof(reply->location));
845 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
846 p += strspn(p, SPACE_CHARS);
847 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
848 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
849 p += strspn(p, SPACE_CHARS);
850 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
851 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
852 p += strspn(p, SPACE_CHARS);
853 if (method && !strcmp(method, "DESCRIBE"))
854 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
855 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
856 p += strspn(p, SPACE_CHARS);
857 if (method && !strcmp(method, "PLAY"))
858 rtsp_parse_rtp_info(rt, p);
859 } else if (av_stristart(p, "Public:", &p) && rt) {
860 if (strstr(p, "GET_PARAMETER") &&
861 method && !strcmp(method, "OPTIONS"))
862 rt->get_parameter_supported = 1;
863 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
864 p += strspn(p, SPACE_CHARS);
865 rt->accept_dynamic_rate = atoi(p);
869 /* skip a RTP/TCP interleaved packet */
870 void ff_rtsp_skip_packet(AVFormatContext *s)
872 RTSPState *rt = s->priv_data;
876 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
879 len = AV_RB16(buf + 1);
881 av_dlog(s, "skipping RTP packet len=%d\n", len);
886 if (len1 > sizeof(buf))
888 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
895 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
896 unsigned char **content_ptr,
897 int return_on_interleaved_data, const char *method)
899 RTSPState *rt = s->priv_data;
900 char buf[4096], buf1[1024], *q;
903 int ret, content_length, line_count = 0;
904 unsigned char *content = NULL;
906 memset(reply, 0, sizeof(*reply));
908 /* parse reply (XXX: use buffers) */
909 rt->last_reply[0] = '\0';
913 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
914 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
920 /* XXX: only parse it if first char on line ? */
921 if (return_on_interleaved_data) {
924 ff_rtsp_skip_packet(s);
925 } else if (ch != '\r') {
926 if ((q - buf) < sizeof(buf) - 1)
932 av_dlog(s, "line='%s'\n", buf);
934 /* test if last line */
938 if (line_count == 0) {
940 get_word(buf1, sizeof(buf1), &p);
941 get_word(buf1, sizeof(buf1), &p);
942 reply->status_code = atoi(buf1);
943 av_strlcpy(reply->reason, p, sizeof(reply->reason));
945 ff_rtsp_parse_line(reply, p, rt, method);
946 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
947 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
952 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
953 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
955 content_length = reply->content_length;
956 if (content_length > 0) {
957 /* leave some room for a trailing '\0' (useful for simple parsing) */
958 content = av_malloc(content_length + 1);
959 ffurl_read_complete(rt->rtsp_hd, content, content_length);
960 content[content_length] = '\0';
963 *content_ptr = content;
967 if (rt->seq != reply->seq) {
968 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
969 rt->seq, reply->seq);
973 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
974 reply->notice == 2104 /* Start-of-Stream Reached */ ||
975 reply->notice == 2306 /* Continuous Feed Terminated */) {
976 rt->state = RTSP_STATE_IDLE;
977 } else if (reply->notice >= 4400 && reply->notice < 5500) {
978 return AVERROR(EIO); /* data or server error */
979 } else if (reply->notice == 2401 /* Ticket Expired */ ||
980 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
981 return AVERROR(EPERM);
987 * Send a command to the RTSP server without waiting for the reply.
989 * @param s RTSP (de)muxer context
990 * @param method the method for the request
991 * @param url the target url for the request
992 * @param headers extra header lines to include in the request
993 * @param send_content if non-null, the data to send as request body content
994 * @param send_content_length the length of the send_content data, or 0 if
995 * send_content is null
997 * @return zero if success, nonzero otherwise
999 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1000 const char *method, const char *url,
1001 const char *headers,
1002 const unsigned char *send_content,
1003 int send_content_length)
1005 RTSPState *rt = s->priv_data;
1006 char buf[4096], *out_buf;
1007 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1009 /* Add in RTSP headers */
1012 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1014 av_strlcat(buf, headers, sizeof(buf));
1015 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1016 if (rt->session_id[0] != '\0' && (!headers ||
1017 !strstr(headers, "\nIf-Match:"))) {
1018 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1021 char *str = ff_http_auth_create_response(&rt->auth_state,
1022 rt->auth, url, method);
1024 av_strlcat(buf, str, sizeof(buf));
1027 if (send_content_length > 0 && send_content)
1028 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1029 av_strlcat(buf, "\r\n", sizeof(buf));
1031 /* base64 encode rtsp if tunneling */
1032 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1033 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1034 out_buf = base64buf;
1037 av_dlog(s, "Sending:\n%s--\n", buf);
1039 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1040 if (send_content_length > 0 && send_content) {
1041 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1042 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1043 "with content data not supported\n");
1044 return AVERROR_PATCHWELCOME;
1046 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1048 rt->last_cmd_time = av_gettime();
1053 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1054 const char *url, const char *headers)
1056 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1059 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1060 const char *headers, RTSPMessageHeader *reply,
1061 unsigned char **content_ptr)
1063 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1064 content_ptr, NULL, 0);
1067 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1068 const char *method, const char *url,
1070 RTSPMessageHeader *reply,
1071 unsigned char **content_ptr,
1072 const unsigned char *send_content,
1073 int send_content_length)
1075 RTSPState *rt = s->priv_data;
1076 HTTPAuthType cur_auth_type;
1080 cur_auth_type = rt->auth_state.auth_type;
1081 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1083 send_content_length)))
1086 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1089 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1090 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1093 if (reply->status_code > 400){
1094 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1098 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1104 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1105 int lower_transport, const char *real_challenge)
1107 RTSPState *rt = s->priv_data;
1108 int rtx = 0, j, i, err, interleave = 0, port_off;
1109 RTSPStream *rtsp_st;
1110 RTSPMessageHeader reply1, *reply = &reply1;
1112 const char *trans_pref;
1114 if (rt->transport == RTSP_TRANSPORT_RDT)
1115 trans_pref = "x-pn-tng";
1117 trans_pref = "RTP/AVP";
1119 /* default timeout: 1 minute */
1122 /* Choose a random starting offset within the first half of the
1123 * port range, to allow for a number of ports to try even if the offset
1124 * happens to be at the end of the random range. */
1125 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1126 /* even random offset */
1127 port_off -= port_off & 0x01;
1129 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1130 char transport[2048];
1133 * WMS serves all UDP data over a single connection, the RTX, which
1134 * isn't necessarily the first in the SDP but has to be the first
1135 * to be set up, else the second/third SETUP will fail with a 461.
1137 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1138 rt->server_type == RTSP_SERVER_WMS) {
1141 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1142 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1144 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1148 if (rtx == rt->nb_rtsp_streams)
1149 return -1; /* no RTX found */
1150 rtsp_st = rt->rtsp_streams[rtx];
1152 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1154 rtsp_st = rt->rtsp_streams[i];
1157 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1160 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1161 port = reply->transports[0].client_port_min;
1165 /* first try in specified port range */
1166 while (j <= rt->rtp_port_max) {
1167 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1168 "?localport=%d", j);
1169 /* we will use two ports per rtp stream (rtp and rtcp) */
1171 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1172 &s->interrupt_callback, NULL))
1175 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1180 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1182 snprintf(transport, sizeof(transport) - 1,
1183 "%s/UDP;", trans_pref);
1184 if (rt->server_type != RTSP_SERVER_REAL)
1185 av_strlcat(transport, "unicast;", sizeof(transport));
1186 av_strlcatf(transport, sizeof(transport),
1187 "client_port=%d", port);
1188 if (rt->transport == RTSP_TRANSPORT_RTP &&
1189 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1190 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1194 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1195 /* For WMS streams, the application streams are only used for
1196 * UDP. When trying to set it up for TCP streams, the server
1197 * will return an error. Therefore, we skip those streams. */
1198 if (rt->server_type == RTSP_SERVER_WMS &&
1199 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1202 snprintf(transport, sizeof(transport) - 1,
1203 "%s/TCP;", trans_pref);
1204 if (rt->transport != RTSP_TRANSPORT_RDT)
1205 av_strlcat(transport, "unicast;", sizeof(transport));
1206 av_strlcatf(transport, sizeof(transport),
1207 "interleaved=%d-%d",
1208 interleave, interleave + 1);
1212 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1213 snprintf(transport, sizeof(transport) - 1,
1214 "%s/UDP;multicast", trans_pref);
1217 av_strlcat(transport, ";mode=receive", sizeof(transport));
1218 } else if (rt->server_type == RTSP_SERVER_REAL ||
1219 rt->server_type == RTSP_SERVER_WMS)
1220 av_strlcat(transport, ";mode=play", sizeof(transport));
1221 snprintf(cmd, sizeof(cmd),
1222 "Transport: %s\r\n",
1224 if (rt->accept_dynamic_rate)
1225 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1226 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1227 char real_res[41], real_csum[9];
1228 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1230 av_strlcatf(cmd, sizeof(cmd),
1232 "RealChallenge2: %s, sd=%s\r\n",
1233 rt->session_id, real_res, real_csum);
1235 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1236 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1239 } else if (reply->status_code != RTSP_STATUS_OK ||
1240 reply->nb_transports != 1) {
1241 err = AVERROR_INVALIDDATA;
1245 /* XXX: same protocol for all streams is required */
1247 if (reply->transports[0].lower_transport != rt->lower_transport ||
1248 reply->transports[0].transport != rt->transport) {
1249 err = AVERROR_INVALIDDATA;
1253 rt->lower_transport = reply->transports[0].lower_transport;
1254 rt->transport = reply->transports[0].transport;
1257 /* Fail if the server responded with another lower transport mode
1258 * than what we requested. */
1259 if (reply->transports[0].lower_transport != lower_transport) {
1260 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1261 err = AVERROR_INVALIDDATA;
1265 switch(reply->transports[0].lower_transport) {
1266 case RTSP_LOWER_TRANSPORT_TCP:
1267 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1268 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1271 case RTSP_LOWER_TRANSPORT_UDP: {
1272 char url[1024], options[30] = "";
1274 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1275 av_strlcpy(options, "?connect=1", sizeof(options));
1276 /* Use source address if specified */
1277 if (reply->transports[0].source[0]) {
1278 ff_url_join(url, sizeof(url), "rtp", NULL,
1279 reply->transports[0].source,
1280 reply->transports[0].server_port_min, "%s", options);
1282 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1283 reply->transports[0].server_port_min, "%s", options);
1285 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1286 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1287 err = AVERROR_INVALIDDATA;
1290 /* Try to initialize the connection state in a
1291 * potential NAT router by sending dummy packets.
1292 * RTP/RTCP dummy packets are used for RDT, too.
1294 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1296 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1299 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1300 char url[1024], namebuf[50];
1301 struct sockaddr_storage addr;
1304 if (reply->transports[0].destination.ss_family) {
1305 addr = reply->transports[0].destination;
1306 port = reply->transports[0].port_min;
1307 ttl = reply->transports[0].ttl;
1309 addr = rtsp_st->sdp_ip;
1310 port = rtsp_st->sdp_port;
1311 ttl = rtsp_st->sdp_ttl;
1313 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1314 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1315 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1316 port, "?ttl=%d", ttl);
1317 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1318 &s->interrupt_callback, NULL) < 0) {
1319 err = AVERROR_INVALIDDATA;
1326 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1330 if (reply->timeout > 0)
1331 rt->timeout = reply->timeout;
1333 if (rt->server_type == RTSP_SERVER_REAL)
1334 rt->need_subscription = 1;
1339 ff_rtsp_undo_setup(s);
1343 void ff_rtsp_close_connections(AVFormatContext *s)
1345 RTSPState *rt = s->priv_data;
1346 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1347 ffurl_close(rt->rtsp_hd);
1348 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1351 int ff_rtsp_connect(AVFormatContext *s)
1353 RTSPState *rt = s->priv_data;
1354 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1355 char *option_list, *option, *filename;
1356 int port, err, tcp_fd;
1357 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1358 int lower_transport_mask = 0;
1359 char real_challenge[64] = "";
1360 struct sockaddr_storage peer;
1361 socklen_t peer_len = sizeof(peer);
1363 if (rt->rtp_port_max < rt->rtp_port_min) {
1364 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1365 "than min port %d\n", rt->rtp_port_max,
1367 return AVERROR(EINVAL);
1370 if (!ff_network_init())
1371 return AVERROR(EIO);
1373 rt->control_transport = RTSP_MODE_PLAIN;
1374 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1375 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1376 rt->control_transport = RTSP_MODE_TUNNEL;
1378 /* Only pass through valid flags from here */
1379 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1382 lower_transport_mask = rt->lower_transport_mask;
1383 /* extract hostname and port */
1384 av_url_split(NULL, 0, auth, sizeof(auth),
1385 host, sizeof(host), &port, path, sizeof(path), s->filename);
1387 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1390 port = RTSP_DEFAULT_PORT;
1392 if (!lower_transport_mask)
1393 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1396 /* Only UDP or TCP - UDP multicast isn't supported. */
1397 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1398 (1 << RTSP_LOWER_TRANSPORT_TCP);
1399 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1400 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1401 "only UDP and TCP are supported for output.\n");
1402 err = AVERROR(EINVAL);
1407 /* Construct the URI used in request; this is similar to s->filename,
1408 * but with authentication credentials removed and RTSP specific options
1410 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1411 host, port, "%s", path);
1413 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1414 /* set up initial handshake for tunneling */
1415 char httpname[1024];
1416 char sessioncookie[17];
1419 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1420 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1421 av_get_random_seed(), av_get_random_seed());
1424 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1425 &s->interrupt_callback) < 0) {
1430 /* generate GET headers */
1431 snprintf(headers, sizeof(headers),
1432 "x-sessioncookie: %s\r\n"
1433 "Accept: application/x-rtsp-tunnelled\r\n"
1434 "Pragma: no-cache\r\n"
1435 "Cache-Control: no-cache\r\n",
1437 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1439 /* complete the connection */
1440 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1446 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1447 &s->interrupt_callback) < 0 ) {
1452 /* generate POST headers */
1453 snprintf(headers, sizeof(headers),
1454 "x-sessioncookie: %s\r\n"
1455 "Content-Type: application/x-rtsp-tunnelled\r\n"
1456 "Pragma: no-cache\r\n"
1457 "Cache-Control: no-cache\r\n"
1458 "Content-Length: 32767\r\n"
1459 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1461 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1462 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1464 /* Initialize the authentication state for the POST session. The HTTP
1465 * protocol implementation doesn't properly handle multi-pass
1466 * authentication for POST requests, since it would require one of
1468 * - implementing Expect: 100-continue, which many HTTP servers
1469 * don't support anyway, even less the RTSP servers that do HTTP
1471 * - sending the whole POST data until getting a 401 reply specifying
1472 * what authentication method to use, then resending all that data
1473 * - waiting for potential 401 replies directly after sending the
1474 * POST header (waiting for some unspecified time)
1475 * Therefore, we copy the full auth state, which works for both basic
1476 * and digest. (For digest, we would have to synchronize the nonce
1477 * count variable between the two sessions, if we'd do more requests
1478 * with the original session, though.)
1480 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1482 /* complete the connection */
1483 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1488 /* open the tcp connection */
1489 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1490 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1491 &s->interrupt_callback, NULL) < 0) {
1495 rt->rtsp_hd_out = rt->rtsp_hd;
1499 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1500 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1501 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1502 NULL, 0, NI_NUMERICHOST);
1505 /* request options supported by the server; this also detects server
1507 for (rt->server_type = RTSP_SERVER_RTP;;) {
1509 if (rt->server_type == RTSP_SERVER_REAL)
1512 * The following entries are required for proper
1513 * streaming from a Realmedia server. They are
1514 * interdependent in some way although we currently
1515 * don't quite understand how. Values were copied
1516 * from mplayer SVN r23589.
1517 * ClientChallenge is a 16-byte ID in hex
1518 * CompanyID is a 16-byte ID in base64
1520 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1521 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1522 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1523 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1525 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1526 if (reply->status_code != RTSP_STATUS_OK) {
1527 err = AVERROR_INVALIDDATA;
1531 /* detect server type if not standard-compliant RTP */
1532 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1533 rt->server_type = RTSP_SERVER_REAL;
1535 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1536 rt->server_type = RTSP_SERVER_WMS;
1537 } else if (rt->server_type == RTSP_SERVER_REAL)
1538 strcpy(real_challenge, reply->real_challenge);
1542 if (s->iformat && CONFIG_RTSP_DEMUXER)
1543 err = ff_rtsp_setup_input_streams(s, reply);
1544 else if (CONFIG_RTSP_MUXER)
1545 err = ff_rtsp_setup_output_streams(s, host);
1550 int lower_transport = ff_log2_tab[lower_transport_mask &
1551 ~(lower_transport_mask - 1)];
1553 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1554 rt->server_type == RTSP_SERVER_REAL ?
1555 real_challenge : NULL);
1558 lower_transport_mask &= ~(1 << lower_transport);
1559 if (lower_transport_mask == 0 && err == 1) {
1560 err = AVERROR(EPROTONOSUPPORT);
1565 rt->lower_transport_mask = lower_transport_mask;
1566 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1567 rt->state = RTSP_STATE_IDLE;
1568 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1571 ff_rtsp_close_streams(s);
1572 ff_rtsp_close_connections(s);
1573 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1574 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1575 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1583 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1586 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1587 uint8_t *buf, int buf_size, int64_t wait_end)
1589 RTSPState *rt = s->priv_data;
1590 RTSPStream *rtsp_st;
1591 int n, i, ret, tcp_fd, timeout_cnt = 0;
1593 struct pollfd *p = rt->p;
1596 if (ff_check_interrupt(&s->interrupt_callback))
1597 return AVERROR_EXIT;
1598 if (wait_end && wait_end - av_gettime() < 0)
1599 return AVERROR(EAGAIN);
1602 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1603 p[max_p].fd = tcp_fd;
1604 p[max_p++].events = POLLIN;
1608 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1609 rtsp_st = rt->rtsp_streams[i];
1610 if (rtsp_st->rtp_handle) {
1611 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1612 p[max_p++].events = POLLIN;
1613 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1614 p[max_p++].events = POLLIN;
1617 n = poll(p, max_p, POLL_TIMEOUT_MS);
1619 int j = 1 - (tcp_fd == -1);
1621 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1622 rtsp_st = rt->rtsp_streams[i];
1623 if (rtsp_st->rtp_handle) {
1624 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1625 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1627 *prtsp_st = rtsp_st;
1634 #if CONFIG_RTSP_DEMUXER
1635 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1636 RTSPMessageHeader reply;
1638 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1641 /* XXX: parse message */
1642 if (rt->state != RTSP_STATE_STREAMING)
1646 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1647 return AVERROR(ETIMEDOUT);
1648 } else if (n < 0 && errno != EINTR)
1649 return AVERROR(errno);
1653 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1655 RTSPState *rt = s->priv_data;
1657 RTSPStream *rtsp_st, *first_queue_st = NULL;
1658 int64_t wait_end = 0;
1660 if (rt->nb_byes == rt->nb_rtsp_streams)
1663 /* get next frames from the same RTP packet */
1664 if (rt->cur_transport_priv) {
1665 if (rt->transport == RTSP_TRANSPORT_RDT) {
1666 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1668 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1670 rt->cur_transport_priv = NULL;
1672 } else if (ret == 1) {
1675 rt->cur_transport_priv = NULL;
1678 if (rt->transport == RTSP_TRANSPORT_RTP) {
1680 int64_t first_queue_time = 0;
1681 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1682 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1686 queue_time = ff_rtp_queued_packet_time(rtpctx);
1687 if (queue_time && (queue_time - first_queue_time < 0 ||
1688 !first_queue_time)) {
1689 first_queue_time = queue_time;
1690 first_queue_st = rt->rtsp_streams[i];
1693 if (first_queue_time)
1694 wait_end = first_queue_time + s->max_delay;
1697 /* read next RTP packet */
1700 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1702 return AVERROR(ENOMEM);
1705 switch(rt->lower_transport) {
1707 #if CONFIG_RTSP_DEMUXER
1708 case RTSP_LOWER_TRANSPORT_TCP:
1709 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1712 case RTSP_LOWER_TRANSPORT_UDP:
1713 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1714 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1715 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1716 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1719 if (len == AVERROR(EAGAIN) && first_queue_st &&
1720 rt->transport == RTSP_TRANSPORT_RTP) {
1721 rtsp_st = first_queue_st;
1722 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1729 if (rt->transport == RTSP_TRANSPORT_RDT) {
1730 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1732 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1734 /* Either bad packet, or a RTCP packet. Check if the
1735 * first_rtcp_ntp_time field was initialized. */
1736 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1737 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1738 /* first_rtcp_ntp_time has been initialized for this stream,
1739 * copy the same value to all other uninitialized streams,
1740 * in order to map their timestamp origin to the same ntp time
1743 AVStream *st = NULL;
1744 if (rtsp_st->stream_index >= 0)
1745 st = s->streams[rtsp_st->stream_index];
1746 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1747 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1748 AVStream *st2 = NULL;
1749 if (rt->rtsp_streams[i]->stream_index >= 0)
1750 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1751 if (rtpctx2 && st && st2 &&
1752 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1753 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1754 rtpctx2->rtcp_ts_offset = av_rescale_q(
1755 rtpctx->rtcp_ts_offset, st->time_base,
1760 if (ret == -RTCP_BYE) {
1763 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1764 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1766 if (rt->nb_byes == rt->nb_rtsp_streams)
1775 /* more packets may follow, so we save the RTP context */
1776 rt->cur_transport_priv = rtsp_st->transport_priv;
1780 #endif /* CONFIG_RTPDEC */
1782 #if CONFIG_SDP_DEMUXER
1783 static int sdp_probe(AVProbeData *p1)
1785 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1787 /* we look for a line beginning "c=IN IP" */
1788 while (p < p_end && *p != '\0') {
1789 if (p + sizeof("c=IN IP") - 1 < p_end &&
1790 av_strstart(p, "c=IN IP", NULL))
1791 return AVPROBE_SCORE_MAX / 2;
1793 while (p < p_end - 1 && *p != '\n') p++;
1802 static int sdp_read_header(AVFormatContext *s)
1804 RTSPState *rt = s->priv_data;
1805 RTSPStream *rtsp_st;
1810 if (!ff_network_init())
1811 return AVERROR(EIO);
1813 /* read the whole sdp file */
1814 /* XXX: better loading */
1815 content = av_malloc(SDP_MAX_SIZE);
1816 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1819 return AVERROR_INVALIDDATA;
1821 content[size] ='\0';
1823 err = ff_sdp_parse(s, content);
1827 /* open each RTP stream */
1828 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1830 rtsp_st = rt->rtsp_streams[i];
1832 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1833 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1834 ff_url_join(url, sizeof(url), "rtp", NULL,
1835 namebuf, rtsp_st->sdp_port,
1836 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1838 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1839 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1840 &s->interrupt_callback, NULL) < 0) {
1841 err = AVERROR_INVALIDDATA;
1844 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1849 ff_rtsp_close_streams(s);
1854 static int sdp_read_close(AVFormatContext *s)
1856 ff_rtsp_close_streams(s);
1861 static const AVClass sdp_demuxer_class = {
1862 .class_name = "SDP demuxer",
1863 .item_name = av_default_item_name,
1864 .option = sdp_options,
1865 .version = LIBAVUTIL_VERSION_INT,
1868 AVInputFormat ff_sdp_demuxer = {
1870 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1871 .priv_data_size = sizeof(RTSPState),
1872 .read_probe = sdp_probe,
1873 .read_header = sdp_read_header,
1874 .read_packet = ff_rtsp_fetch_packet,
1875 .read_close = sdp_read_close,
1876 .priv_class = &sdp_demuxer_class
1878 #endif /* CONFIG_SDP_DEMUXER */
1880 #if CONFIG_RTP_DEMUXER
1881 static int rtp_probe(AVProbeData *p)
1883 if (av_strstart(p->filename, "rtp:", NULL))
1884 return AVPROBE_SCORE_MAX;
1888 static int rtp_read_header(AVFormatContext *s)
1890 uint8_t recvbuf[1500];
1891 char host[500], sdp[500];
1893 URLContext* in = NULL;
1895 AVCodecContext codec;
1896 struct sockaddr_storage addr;
1898 socklen_t addrlen = sizeof(addr);
1899 RTSPState *rt = s->priv_data;
1901 if (!ff_network_init())
1902 return AVERROR(EIO);
1904 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1905 &s->interrupt_callback, NULL);
1910 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1911 if (ret == AVERROR(EAGAIN))
1916 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1920 if ((recvbuf[0] & 0xc0) != 0x80) {
1921 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1926 payload_type = recvbuf[1] & 0x7f;
1929 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1933 memset(&codec, 0, sizeof(codec));
1934 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1935 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1936 "without an SDP file describing it\n",
1940 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1941 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1942 "properly you need an SDP file "
1946 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1947 NULL, 0, s->filename);
1949 snprintf(sdp, sizeof(sdp),
1950 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1951 addr.ss_family == AF_INET ? 4 : 6, host,
1952 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1953 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1954 port, payload_type);
1955 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1957 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1960 /* sdp_read_header initializes this again */
1963 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
1965 ret = sdp_read_header(s);
1976 static const AVClass rtp_demuxer_class = {
1977 .class_name = "RTP demuxer",
1978 .item_name = av_default_item_name,
1979 .option = rtp_options,
1980 .version = LIBAVUTIL_VERSION_INT,
1983 AVInputFormat ff_rtp_demuxer = {
1985 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
1986 .priv_data_size = sizeof(RTSPState),
1987 .read_probe = rtp_probe,
1988 .read_header = rtp_read_header,
1989 .read_packet = ff_rtsp_fetch_packet,
1990 .read_close = sdp_read_close,
1991 .flags = AVFMT_NOFILE,
1992 .priv_class = &rtp_demuxer_class
1994 #endif /* CONFIG_RTP_DEMUXER */