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/parseutils.h"
26 #include "libavutil/random_seed.h"
28 #include "avio_internal.h"
37 #include "os_support.h"
43 #include "rtpdec_formats.h"
44 #include "rtpenc_chain.h"
48 //#define DEBUG_RTP_TCP
50 /* Timeout values for socket poll, in ms,
51 * and read_packet(), in seconds */
52 #define POLL_TIMEOUT_MS 100
53 #define READ_PACKET_TIMEOUT_S 10
54 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
55 #define SDP_MAX_SIZE 16384
56 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
58 static void get_word_until_chars(char *buf, int buf_size,
59 const char *sep, const char **pp)
65 p += strspn(p, SPACE_CHARS);
67 while (!strchr(sep, *p) && *p != '\0') {
68 if ((q - buf) < buf_size - 1)
77 static void get_word_sep(char *buf, int buf_size, const char *sep,
80 if (**pp == '/') (*pp)++;
81 get_word_until_chars(buf, buf_size, sep, pp);
84 static void get_word(char *buf, int buf_size, const char **pp)
86 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
89 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
91 * Used for seeking in the rtp stream.
93 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
97 p += strspn(p, SPACE_CHARS);
98 if (!av_stristart(p, "npt=", &p))
101 *start = AV_NOPTS_VALUE;
102 *end = AV_NOPTS_VALUE;
104 get_word_sep(buf, sizeof(buf), "-", &p);
105 av_parse_time(start, buf, 1);
108 get_word_sep(buf, sizeof(buf), "-", &p);
109 av_parse_time(end, buf, 1);
111 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
112 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
115 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
117 struct addrinfo hints, *ai = NULL;
118 memset(&hints, 0, sizeof(hints));
119 hints.ai_flags = AI_NUMERICHOST;
120 if (getaddrinfo(buf, NULL, &hints, &ai))
122 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
128 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
129 RTSPStream *rtsp_st, AVCodecContext *codec)
133 codec->codec_id = handler->codec_id;
134 rtsp_st->dynamic_handler = handler;
136 rtsp_st->dynamic_protocol_context = handler->open();
139 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
140 static int sdp_parse_rtpmap(AVFormatContext *s,
141 AVStream *st, RTSPStream *rtsp_st,
142 int payload_type, const char *p)
144 AVCodecContext *codec = st->codec;
150 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
151 * see if we can handle this kind of payload.
152 * The space should normally not be there but some Real streams or
153 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
154 * have a trailing space. */
155 get_word_sep(buf, sizeof(buf), "/ ", &p);
156 if (payload_type >= RTP_PT_PRIVATE) {
157 RTPDynamicProtocolHandler *handler =
158 ff_rtp_handler_find_by_name(buf, codec->codec_type);
159 init_rtp_handler(handler, rtsp_st, codec);
160 /* If no dynamic handler was found, check with the list of standard
161 * allocated types, if such a stream for some reason happens to
162 * use a private payload type. This isn't handled in rtpdec.c, since
163 * the format name from the rtpmap line never is passed into rtpdec. */
164 if (!rtsp_st->dynamic_handler)
165 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
167 /* We are in a standard case
168 * (from http://www.iana.org/assignments/rtp-parameters). */
169 /* search into AVRtpPayloadTypes[] */
170 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
173 c = avcodec_find_decoder(codec->codec_id);
179 get_word_sep(buf, sizeof(buf), "/", &p);
181 switch (codec->codec_type) {
182 case AVMEDIA_TYPE_AUDIO:
183 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
184 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
185 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
187 codec->sample_rate = i;
188 av_set_pts_info(st, 32, 1, codec->sample_rate);
189 get_word_sep(buf, sizeof(buf), "/", &p);
193 // TODO: there is a bug here; if it is a mono stream, and
194 // less than 22000Hz, faad upconverts to stereo and twice
195 // the frequency. No problem, but the sample rate is being
196 // set here by the sdp line. Patch on its way. (rdm)
198 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
200 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
203 case AVMEDIA_TYPE_VIDEO:
204 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
206 av_set_pts_info(st, 32, 1, i);
214 /* parse the attribute line from the fmtp a line of an sdp response. This
215 * is broken out as a function because it is used in rtp_h264.c, which is
217 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
218 char *value, int value_size)
220 *p += strspn(*p, SPACE_CHARS);
222 get_word_sep(attr, attr_size, "=", p);
225 get_word_sep(value, value_size, ";", p);
233 typedef struct SDPParseState {
235 struct sockaddr_storage default_ip;
237 int skip_media; ///< set if an unknown m= line occurs
240 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
241 int letter, const char *buf)
243 RTSPState *rt = s->priv_data;
244 char buf1[64], st_type[64];
246 enum AVMediaType codec_type;
250 struct sockaddr_storage sdp_ip;
253 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
256 if (s1->skip_media && letter != 'm')
260 get_word(buf1, sizeof(buf1), &p);
261 if (strcmp(buf1, "IN") != 0)
263 get_word(buf1, sizeof(buf1), &p);
264 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
266 get_word_sep(buf1, sizeof(buf1), "/", &p);
267 if (get_sockaddr(buf1, &sdp_ip))
272 get_word_sep(buf1, sizeof(buf1), "/", &p);
275 if (s->nb_streams == 0) {
276 s1->default_ip = sdp_ip;
277 s1->default_ttl = ttl;
279 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
280 rtsp_st->sdp_ip = sdp_ip;
281 rtsp_st->sdp_ttl = ttl;
285 av_metadata_set2(&s->metadata, "title", p, 0);
288 if (s->nb_streams == 0) {
289 av_metadata_set2(&s->metadata, "comment", p, 0);
296 get_word(st_type, sizeof(st_type), &p);
297 if (!strcmp(st_type, "audio")) {
298 codec_type = AVMEDIA_TYPE_AUDIO;
299 } else if (!strcmp(st_type, "video")) {
300 codec_type = AVMEDIA_TYPE_VIDEO;
301 } else if (!strcmp(st_type, "application")) {
302 codec_type = AVMEDIA_TYPE_DATA;
307 rtsp_st = av_mallocz(sizeof(RTSPStream));
310 rtsp_st->stream_index = -1;
311 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
313 rtsp_st->sdp_ip = s1->default_ip;
314 rtsp_st->sdp_ttl = s1->default_ttl;
316 get_word(buf1, sizeof(buf1), &p); /* port */
317 rtsp_st->sdp_port = atoi(buf1);
319 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
321 /* XXX: handle list of formats */
322 get_word(buf1, sizeof(buf1), &p); /* format list */
323 rtsp_st->sdp_payload_type = atoi(buf1);
325 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
326 /* no corresponding stream */
328 st = av_new_stream(s, rt->nb_rtsp_streams - 1);
331 rtsp_st->stream_index = st->index;
332 st->codec->codec_type = codec_type;
333 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
334 RTPDynamicProtocolHandler *handler;
335 /* if standard payload type, we can find the codec right now */
336 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
337 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
338 st->codec->sample_rate > 0)
339 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
340 /* Even static payload types may need a custom depacketizer */
341 handler = ff_rtp_handler_find_by_id(
342 rtsp_st->sdp_payload_type, st->codec->codec_type);
343 init_rtp_handler(handler, rtsp_st, st->codec);
346 /* put a default control url */
347 av_strlcpy(rtsp_st->control_url, rt->control_uri,
348 sizeof(rtsp_st->control_url));
351 if (av_strstart(p, "control:", &p)) {
352 if (s->nb_streams == 0) {
353 if (!strncmp(p, "rtsp://", 7))
354 av_strlcpy(rt->control_uri, p,
355 sizeof(rt->control_uri));
358 /* get the control url */
359 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
361 /* XXX: may need to add full url resolution */
362 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
364 if (proto[0] == '\0') {
365 /* relative control URL */
366 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
367 av_strlcat(rtsp_st->control_url, "/",
368 sizeof(rtsp_st->control_url));
369 av_strlcat(rtsp_st->control_url, p,
370 sizeof(rtsp_st->control_url));
372 av_strlcpy(rtsp_st->control_url, p,
373 sizeof(rtsp_st->control_url));
375 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
376 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
377 get_word(buf1, sizeof(buf1), &p);
378 payload_type = atoi(buf1);
379 st = s->streams[s->nb_streams - 1];
380 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
381 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
382 } else if (av_strstart(p, "fmtp:", &p) ||
383 av_strstart(p, "framesize:", &p)) {
384 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
385 // let dynamic protocol handlers have a stab at the line.
386 get_word(buf1, sizeof(buf1), &p);
387 payload_type = atoi(buf1);
388 for (i = 0; i < rt->nb_rtsp_streams; i++) {
389 rtsp_st = rt->rtsp_streams[i];
390 if (rtsp_st->sdp_payload_type == payload_type &&
391 rtsp_st->dynamic_handler &&
392 rtsp_st->dynamic_handler->parse_sdp_a_line)
393 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
394 rtsp_st->dynamic_protocol_context, buf);
396 } else if (av_strstart(p, "range:", &p)) {
399 // this is so that seeking on a streamed file can work.
400 rtsp_parse_range_npt(p, &start, &end);
401 s->start_time = start;
402 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
403 s->duration = (end == AV_NOPTS_VALUE) ?
404 AV_NOPTS_VALUE : end - start;
405 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
407 rt->transport = RTSP_TRANSPORT_RDT;
408 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
410 st = s->streams[s->nb_streams - 1];
411 st->codec->sample_rate = atoi(p);
413 if (rt->server_type == RTSP_SERVER_WMS)
414 ff_wms_parse_sdp_a_line(s, p);
415 if (s->nb_streams > 0) {
416 if (rt->server_type == RTSP_SERVER_REAL)
417 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
419 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
420 if (rtsp_st->dynamic_handler &&
421 rtsp_st->dynamic_handler->parse_sdp_a_line)
422 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
424 rtsp_st->dynamic_protocol_context, buf);
432 * Parse the sdp description and allocate the rtp streams and the
433 * pollfd array used for udp ones.
436 int ff_sdp_parse(AVFormatContext *s, const char *content)
438 RTSPState *rt = s->priv_data;
441 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
442 * contain long SDP lines containing complete ASF Headers (several
443 * kB) or arrays of MDPR (RM stream descriptor) headers plus
444 * "rulebooks" describing their properties. Therefore, the SDP line
447 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
448 * in rtpdec_xiph.c. */
450 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
452 memset(s1, 0, sizeof(SDPParseState));
455 p += strspn(p, SPACE_CHARS);
463 /* get the content */
465 while (*p != '\n' && *p != '\r' && *p != '\0') {
466 if ((q - buf) < sizeof(buf) - 1)
471 sdp_parse_line(s, s1, letter, buf);
473 while (*p != '\n' && *p != '\0')
478 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
479 if (!rt->p) return AVERROR(ENOMEM);
482 #endif /* CONFIG_RTPDEC */
484 void ff_rtsp_undo_setup(AVFormatContext *s)
486 RTSPState *rt = s->priv_data;
489 for (i = 0; i < rt->nb_rtsp_streams; i++) {
490 RTSPStream *rtsp_st = rt->rtsp_streams[i];
493 if (rtsp_st->transport_priv) {
495 AVFormatContext *rtpctx = rtsp_st->transport_priv;
496 av_write_trailer(rtpctx);
497 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
499 avio_close_dyn_buf(rtpctx->pb, &ptr);
502 avio_close(rtpctx->pb);
504 avformat_free_context(rtpctx);
505 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
506 ff_rdt_parse_close(rtsp_st->transport_priv);
507 else if (CONFIG_RTPDEC)
508 rtp_parse_close(rtsp_st->transport_priv);
510 rtsp_st->transport_priv = NULL;
511 if (rtsp_st->rtp_handle)
512 url_close(rtsp_st->rtp_handle);
513 rtsp_st->rtp_handle = NULL;
517 /* close and free RTSP streams */
518 void ff_rtsp_close_streams(AVFormatContext *s)
520 RTSPState *rt = s->priv_data;
524 ff_rtsp_undo_setup(s);
525 for (i = 0; i < rt->nb_rtsp_streams; i++) {
526 rtsp_st = rt->rtsp_streams[i];
528 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
529 rtsp_st->dynamic_handler->close(
530 rtsp_st->dynamic_protocol_context);
534 av_free(rt->rtsp_streams);
536 av_close_input_stream (rt->asf_ctx);
540 av_free(rt->recvbuf);
543 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
545 RTSPState *rt = s->priv_data;
548 /* open the RTP context */
549 if (rtsp_st->stream_index >= 0)
550 st = s->streams[rtsp_st->stream_index];
552 s->ctx_flags |= AVFMTCTX_NOHEADER;
554 if (s->oformat && CONFIG_RTSP_MUXER) {
555 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
557 RTSP_TCP_MAX_PACKET_SIZE);
558 /* Ownership of rtp_handle is passed to the rtp mux context */
559 rtsp_st->rtp_handle = NULL;
560 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
561 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
562 rtsp_st->dynamic_protocol_context,
563 rtsp_st->dynamic_handler);
564 else if (CONFIG_RTPDEC)
565 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
566 rtsp_st->sdp_payload_type,
567 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
568 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
570 if (!rtsp_st->transport_priv) {
571 return AVERROR(ENOMEM);
572 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
573 if (rtsp_st->dynamic_handler) {
574 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
575 rtsp_st->dynamic_protocol_context,
576 rtsp_st->dynamic_handler);
583 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
584 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
590 p += strspn(p, SPACE_CHARS);
591 v = strtol(p, (char **)&p, 10);
595 v = strtol(p, (char **)&p, 10);
604 /* XXX: only one transport specification is parsed */
605 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
607 char transport_protocol[16];
609 char lower_transport[16];
611 RTSPTransportField *th;
614 reply->nb_transports = 0;
617 p += strspn(p, SPACE_CHARS);
621 th = &reply->transports[reply->nb_transports];
623 get_word_sep(transport_protocol, sizeof(transport_protocol),
625 if (!strcasecmp (transport_protocol, "rtp")) {
626 get_word_sep(profile, sizeof(profile), "/;,", &p);
627 lower_transport[0] = '\0';
628 /* rtp/avp/<protocol> */
630 get_word_sep(lower_transport, sizeof(lower_transport),
633 th->transport = RTSP_TRANSPORT_RTP;
634 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
635 !strcasecmp (transport_protocol, "x-real-rdt")) {
636 /* x-pn-tng/<protocol> */
637 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
639 th->transport = RTSP_TRANSPORT_RDT;
641 if (!strcasecmp(lower_transport, "TCP"))
642 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
644 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
648 /* get each parameter */
649 while (*p != '\0' && *p != ',') {
650 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
651 if (!strcmp(parameter, "port")) {
654 rtsp_parse_range(&th->port_min, &th->port_max, &p);
656 } else if (!strcmp(parameter, "client_port")) {
659 rtsp_parse_range(&th->client_port_min,
660 &th->client_port_max, &p);
662 } else if (!strcmp(parameter, "server_port")) {
665 rtsp_parse_range(&th->server_port_min,
666 &th->server_port_max, &p);
668 } else if (!strcmp(parameter, "interleaved")) {
671 rtsp_parse_range(&th->interleaved_min,
672 &th->interleaved_max, &p);
674 } else if (!strcmp(parameter, "multicast")) {
675 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
676 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
677 } else if (!strcmp(parameter, "ttl")) {
680 th->ttl = strtol(p, (char **)&p, 10);
682 } else if (!strcmp(parameter, "destination")) {
685 get_word_sep(buf, sizeof(buf), ";,", &p);
686 get_sockaddr(buf, &th->destination);
688 } else if (!strcmp(parameter, "source")) {
691 get_word_sep(buf, sizeof(buf), ";,", &p);
692 av_strlcpy(th->source, buf, sizeof(th->source));
696 while (*p != ';' && *p != '\0' && *p != ',')
704 reply->nb_transports++;
708 static void handle_rtp_info(RTSPState *rt, const char *url,
709 uint32_t seq, uint32_t rtptime)
712 if (!rtptime || !url[0])
714 if (rt->transport != RTSP_TRANSPORT_RTP)
716 for (i = 0; i < rt->nb_rtsp_streams; i++) {
717 RTSPStream *rtsp_st = rt->rtsp_streams[i];
718 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
721 if (!strcmp(rtsp_st->control_url, url)) {
722 rtpctx->base_timestamp = rtptime;
728 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
731 char key[20], value[1024], url[1024] = "";
732 uint32_t seq = 0, rtptime = 0;
735 p += strspn(p, SPACE_CHARS);
738 get_word_sep(key, sizeof(key), "=", &p);
742 get_word_sep(value, sizeof(value), ";, ", &p);
744 if (!strcmp(key, "url"))
745 av_strlcpy(url, value, sizeof(url));
746 else if (!strcmp(key, "seq"))
747 seq = strtol(value, NULL, 10);
748 else if (!strcmp(key, "rtptime"))
749 rtptime = strtol(value, NULL, 10);
751 handle_rtp_info(rt, url, seq, rtptime);
760 handle_rtp_info(rt, url, seq, rtptime);
763 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
764 RTSPState *rt, const char *method)
768 /* NOTE: we do case independent match for broken servers */
770 if (av_stristart(p, "Session:", &p)) {
772 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
773 if (av_stristart(p, ";timeout=", &p) &&
774 (t = strtol(p, NULL, 10)) > 0) {
777 } else if (av_stristart(p, "Content-Length:", &p)) {
778 reply->content_length = strtol(p, NULL, 10);
779 } else if (av_stristart(p, "Transport:", &p)) {
780 rtsp_parse_transport(reply, p);
781 } else if (av_stristart(p, "CSeq:", &p)) {
782 reply->seq = strtol(p, NULL, 10);
783 } else if (av_stristart(p, "Range:", &p)) {
784 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
785 } else if (av_stristart(p, "RealChallenge1:", &p)) {
786 p += strspn(p, SPACE_CHARS);
787 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
788 } else if (av_stristart(p, "Server:", &p)) {
789 p += strspn(p, SPACE_CHARS);
790 av_strlcpy(reply->server, p, sizeof(reply->server));
791 } else if (av_stristart(p, "Notice:", &p) ||
792 av_stristart(p, "X-Notice:", &p)) {
793 reply->notice = strtol(p, NULL, 10);
794 } else if (av_stristart(p, "Location:", &p)) {
795 p += strspn(p, SPACE_CHARS);
796 av_strlcpy(reply->location, p , sizeof(reply->location));
797 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
798 p += strspn(p, SPACE_CHARS);
799 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
800 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
801 p += strspn(p, SPACE_CHARS);
802 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
803 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
804 p += strspn(p, SPACE_CHARS);
805 if (method && !strcmp(method, "DESCRIBE"))
806 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
807 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
808 p += strspn(p, SPACE_CHARS);
809 if (method && !strcmp(method, "PLAY"))
810 rtsp_parse_rtp_info(rt, p);
814 /* skip a RTP/TCP interleaved packet */
815 void ff_rtsp_skip_packet(AVFormatContext *s)
817 RTSPState *rt = s->priv_data;
821 ret = url_read_complete(rt->rtsp_hd, buf, 3);
824 len = AV_RB16(buf + 1);
826 av_dlog(s, "skipping RTP packet len=%d\n", len);
831 if (len1 > sizeof(buf))
833 ret = url_read_complete(rt->rtsp_hd, buf, len1);
840 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
841 unsigned char **content_ptr,
842 int return_on_interleaved_data, const char *method)
844 RTSPState *rt = s->priv_data;
845 char buf[4096], buf1[1024], *q;
848 int ret, content_length, line_count = 0;
849 unsigned char *content = NULL;
851 memset(reply, 0, sizeof(*reply));
853 /* parse reply (XXX: use buffers) */
854 rt->last_reply[0] = '\0';
858 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
860 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
867 /* XXX: only parse it if first char on line ? */
868 if (return_on_interleaved_data) {
871 ff_rtsp_skip_packet(s);
872 } else if (ch != '\r') {
873 if ((q - buf) < sizeof(buf) - 1)
879 av_dlog(s, "line='%s'\n", buf);
881 /* test if last line */
885 if (line_count == 0) {
887 get_word(buf1, sizeof(buf1), &p);
888 get_word(buf1, sizeof(buf1), &p);
889 reply->status_code = atoi(buf1);
890 av_strlcpy(reply->reason, p, sizeof(reply->reason));
892 ff_rtsp_parse_line(reply, p, rt, method);
893 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
894 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
899 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
900 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
902 content_length = reply->content_length;
903 if (content_length > 0) {
904 /* leave some room for a trailing '\0' (useful for simple parsing) */
905 content = av_malloc(content_length + 1);
906 (void)url_read_complete(rt->rtsp_hd, content, content_length);
907 content[content_length] = '\0';
910 *content_ptr = content;
914 if (rt->seq != reply->seq) {
915 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
916 rt->seq, reply->seq);
920 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
921 reply->notice == 2104 /* Start-of-Stream Reached */ ||
922 reply->notice == 2306 /* Continuous Feed Terminated */) {
923 rt->state = RTSP_STATE_IDLE;
924 } else if (reply->notice >= 4400 && reply->notice < 5500) {
925 return AVERROR(EIO); /* data or server error */
926 } else if (reply->notice == 2401 /* Ticket Expired */ ||
927 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
928 return AVERROR(EPERM);
934 * Send a command to the RTSP server without waiting for the reply.
936 * @param s RTSP (de)muxer context
937 * @param method the method for the request
938 * @param url the target url for the request
939 * @param headers extra header lines to include in the request
940 * @param send_content if non-null, the data to send as request body content
941 * @param send_content_length the length of the send_content data, or 0 if
942 * send_content is null
944 * @return zero if success, nonzero otherwise
946 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
947 const char *method, const char *url,
949 const unsigned char *send_content,
950 int send_content_length)
952 RTSPState *rt = s->priv_data;
953 char buf[4096], *out_buf;
954 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
956 /* Add in RTSP headers */
959 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
961 av_strlcat(buf, headers, sizeof(buf));
962 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
963 if (rt->session_id[0] != '\0' && (!headers ||
964 !strstr(headers, "\nIf-Match:"))) {
965 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
968 char *str = ff_http_auth_create_response(&rt->auth_state,
969 rt->auth, url, method);
971 av_strlcat(buf, str, sizeof(buf));
974 if (send_content_length > 0 && send_content)
975 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
976 av_strlcat(buf, "\r\n", sizeof(buf));
978 /* base64 encode rtsp if tunneling */
979 if (rt->control_transport == RTSP_MODE_TUNNEL) {
980 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
984 av_dlog(s, "Sending:\n%s--\n", buf);
986 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
987 if (send_content_length > 0 && send_content) {
988 if (rt->control_transport == RTSP_MODE_TUNNEL) {
989 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
990 "with content data not supported\n");
991 return AVERROR_PATCHWELCOME;
993 url_write(rt->rtsp_hd_out, send_content, send_content_length);
995 rt->last_cmd_time = av_gettime();
1000 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1001 const char *url, const char *headers)
1003 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1006 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1007 const char *headers, RTSPMessageHeader *reply,
1008 unsigned char **content_ptr)
1010 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1011 content_ptr, NULL, 0);
1014 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1015 const char *method, const char *url,
1017 RTSPMessageHeader *reply,
1018 unsigned char **content_ptr,
1019 const unsigned char *send_content,
1020 int send_content_length)
1022 RTSPState *rt = s->priv_data;
1023 HTTPAuthType cur_auth_type;
1027 cur_auth_type = rt->auth_state.auth_type;
1028 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1030 send_content_length)))
1033 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1036 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1037 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1040 if (reply->status_code > 400){
1041 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1045 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1052 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1054 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1055 int lower_transport, const char *real_challenge)
1057 RTSPState *rt = s->priv_data;
1058 int rtx, j, i, err, interleave = 0;
1059 RTSPStream *rtsp_st;
1060 RTSPMessageHeader reply1, *reply = &reply1;
1062 const char *trans_pref;
1064 if (rt->transport == RTSP_TRANSPORT_RDT)
1065 trans_pref = "x-pn-tng";
1067 trans_pref = "RTP/AVP";
1069 /* default timeout: 1 minute */
1072 /* for each stream, make the setup request */
1073 /* XXX: we assume the same server is used for the control of each
1076 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1077 char transport[2048];
1080 * WMS serves all UDP data over a single connection, the RTX, which
1081 * isn't necessarily the first in the SDP but has to be the first
1082 * to be set up, else the second/third SETUP will fail with a 461.
1084 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1085 rt->server_type == RTSP_SERVER_WMS) {
1088 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1089 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1091 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1095 if (rtx == rt->nb_rtsp_streams)
1096 return -1; /* no RTX found */
1097 rtsp_st = rt->rtsp_streams[rtx];
1099 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1101 rtsp_st = rt->rtsp_streams[i];
1104 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1107 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1108 port = reply->transports[0].client_port_min;
1112 /* first try in specified port range */
1113 if (RTSP_RTP_PORT_MIN != 0) {
1114 while (j <= RTSP_RTP_PORT_MAX) {
1115 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1116 "?localport=%d", j);
1117 /* we will use two ports per rtp stream (rtp and rtcp) */
1119 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1125 /* then try on any port */
1126 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1127 err = AVERROR_INVALIDDATA;
1131 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1137 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1139 snprintf(transport, sizeof(transport) - 1,
1140 "%s/UDP;", trans_pref);
1141 if (rt->server_type != RTSP_SERVER_REAL)
1142 av_strlcat(transport, "unicast;", sizeof(transport));
1143 av_strlcatf(transport, sizeof(transport),
1144 "client_port=%d", port);
1145 if (rt->transport == RTSP_TRANSPORT_RTP &&
1146 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1147 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1151 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1152 /** For WMS streams, the application streams are only used for
1153 * UDP. When trying to set it up for TCP streams, the server
1154 * will return an error. Therefore, we skip those streams. */
1155 if (rt->server_type == RTSP_SERVER_WMS &&
1156 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1159 snprintf(transport, sizeof(transport) - 1,
1160 "%s/TCP;", trans_pref);
1161 if (rt->transport != RTSP_TRANSPORT_RDT)
1162 av_strlcat(transport, "unicast;", sizeof(transport));
1163 av_strlcatf(transport, sizeof(transport),
1164 "interleaved=%d-%d",
1165 interleave, interleave + 1);
1169 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1170 snprintf(transport, sizeof(transport) - 1,
1171 "%s/UDP;multicast", trans_pref);
1174 av_strlcat(transport, ";mode=receive", sizeof(transport));
1175 } else if (rt->server_type == RTSP_SERVER_REAL ||
1176 rt->server_type == RTSP_SERVER_WMS)
1177 av_strlcat(transport, ";mode=play", sizeof(transport));
1178 snprintf(cmd, sizeof(cmd),
1179 "Transport: %s\r\n",
1181 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1182 char real_res[41], real_csum[9];
1183 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1185 av_strlcatf(cmd, sizeof(cmd),
1187 "RealChallenge2: %s, sd=%s\r\n",
1188 rt->session_id, real_res, real_csum);
1190 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1191 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1194 } else if (reply->status_code != RTSP_STATUS_OK ||
1195 reply->nb_transports != 1) {
1196 err = AVERROR_INVALIDDATA;
1200 /* XXX: same protocol for all streams is required */
1202 if (reply->transports[0].lower_transport != rt->lower_transport ||
1203 reply->transports[0].transport != rt->transport) {
1204 err = AVERROR_INVALIDDATA;
1208 rt->lower_transport = reply->transports[0].lower_transport;
1209 rt->transport = reply->transports[0].transport;
1212 /* Fail if the server responded with another lower transport mode
1213 * than what we requested. */
1214 if (reply->transports[0].lower_transport != lower_transport) {
1215 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1216 err = AVERROR_INVALIDDATA;
1220 switch(reply->transports[0].lower_transport) {
1221 case RTSP_LOWER_TRANSPORT_TCP:
1222 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1223 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1226 case RTSP_LOWER_TRANSPORT_UDP: {
1227 char url[1024], options[30] = "";
1229 if (rt->filter_source)
1230 av_strlcpy(options, "?connect=1", sizeof(options));
1231 /* Use source address if specified */
1232 if (reply->transports[0].source[0]) {
1233 ff_url_join(url, sizeof(url), "rtp", NULL,
1234 reply->transports[0].source,
1235 reply->transports[0].server_port_min, options);
1237 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1238 reply->transports[0].server_port_min, options);
1240 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1241 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1242 err = AVERROR_INVALIDDATA;
1245 /* Try to initialize the connection state in a
1246 * potential NAT router by sending dummy packets.
1247 * RTP/RTCP dummy packets are used for RDT, too.
1249 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1251 rtp_send_punch_packets(rtsp_st->rtp_handle);
1254 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1255 char url[1024], namebuf[50];
1256 struct sockaddr_storage addr;
1259 if (reply->transports[0].destination.ss_family) {
1260 addr = reply->transports[0].destination;
1261 port = reply->transports[0].port_min;
1262 ttl = reply->transports[0].ttl;
1264 addr = rtsp_st->sdp_ip;
1265 port = rtsp_st->sdp_port;
1266 ttl = rtsp_st->sdp_ttl;
1268 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1269 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1270 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1271 port, "?ttl=%d", ttl);
1272 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1273 err = AVERROR_INVALIDDATA;
1280 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1284 if (reply->timeout > 0)
1285 rt->timeout = reply->timeout;
1287 if (rt->server_type == RTSP_SERVER_REAL)
1288 rt->need_subscription = 1;
1293 ff_rtsp_undo_setup(s);
1297 void ff_rtsp_close_connections(AVFormatContext *s)
1299 RTSPState *rt = s->priv_data;
1300 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1301 url_close(rt->rtsp_hd);
1302 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1305 int ff_rtsp_connect(AVFormatContext *s)
1307 RTSPState *rt = s->priv_data;
1308 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1309 char *option_list, *option, *filename;
1310 int port, err, tcp_fd;
1311 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1312 int lower_transport_mask = 0;
1313 char real_challenge[64] = "";
1314 struct sockaddr_storage peer;
1315 socklen_t peer_len = sizeof(peer);
1317 if (!ff_network_init())
1318 return AVERROR(EIO);
1320 rt->control_transport = RTSP_MODE_PLAIN;
1321 /* extract hostname and port */
1322 av_url_split(NULL, 0, auth, sizeof(auth),
1323 host, sizeof(host), &port, path, sizeof(path), s->filename);
1325 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1328 port = RTSP_DEFAULT_PORT;
1330 /* search for options */
1331 option_list = strrchr(path, '?');
1333 /* Strip out the RTSP specific options, write out the rest of
1334 * the options back into the same string. */
1335 filename = option_list;
1336 while (option_list) {
1337 /* move the option pointer */
1338 option = ++option_list;
1339 option_list = strchr(option_list, '&');
1343 /* handle the options */
1344 if (!strcmp(option, "udp")) {
1345 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1346 } else if (!strcmp(option, "multicast")) {
1347 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1348 } else if (!strcmp(option, "tcp")) {
1349 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1350 } else if(!strcmp(option, "http")) {
1351 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1352 rt->control_transport = RTSP_MODE_TUNNEL;
1353 } else if (!strcmp(option, "filter_src")) {
1354 rt->filter_source = 1;
1356 /* Write options back into the buffer, using memmove instead
1357 * of strcpy since the strings may overlap. */
1358 int len = strlen(option);
1359 memmove(++filename, option, len);
1361 if (option_list) *filename = '&';
1367 if (!lower_transport_mask)
1368 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1371 /* Only UDP or TCP - UDP multicast isn't supported. */
1372 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1373 (1 << RTSP_LOWER_TRANSPORT_TCP);
1374 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1375 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1376 "only UDP and TCP are supported for output.\n");
1377 err = AVERROR(EINVAL);
1382 /* Construct the URI used in request; this is similar to s->filename,
1383 * but with authentication credentials removed and RTSP specific options
1385 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1386 host, port, "%s", path);
1388 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1389 /* set up initial handshake for tunneling */
1390 char httpname[1024];
1391 char sessioncookie[17];
1394 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1395 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1396 av_get_random_seed(), av_get_random_seed());
1399 if (ffurl_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1404 /* generate GET headers */
1405 snprintf(headers, sizeof(headers),
1406 "x-sessioncookie: %s\r\n"
1407 "Accept: application/x-rtsp-tunnelled\r\n"
1408 "Pragma: no-cache\r\n"
1409 "Cache-Control: no-cache\r\n",
1411 ff_http_set_headers(rt->rtsp_hd, headers);
1413 /* complete the connection */
1414 if (ffurl_connect(rt->rtsp_hd)) {
1420 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1425 /* generate POST headers */
1426 snprintf(headers, sizeof(headers),
1427 "x-sessioncookie: %s\r\n"
1428 "Content-Type: application/x-rtsp-tunnelled\r\n"
1429 "Pragma: no-cache\r\n"
1430 "Cache-Control: no-cache\r\n"
1431 "Content-Length: 32767\r\n"
1432 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1434 ff_http_set_headers(rt->rtsp_hd_out, headers);
1435 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1437 /* Initialize the authentication state for the POST session. The HTTP
1438 * protocol implementation doesn't properly handle multi-pass
1439 * authentication for POST requests, since it would require one of
1441 * - implementing Expect: 100-continue, which many HTTP servers
1442 * don't support anyway, even less the RTSP servers that do HTTP
1444 * - sending the whole POST data until getting a 401 reply specifying
1445 * what authentication method to use, then resending all that data
1446 * - waiting for potential 401 replies directly after sending the
1447 * POST header (waiting for some unspecified time)
1448 * Therefore, we copy the full auth state, which works for both basic
1449 * and digest. (For digest, we would have to synchronize the nonce
1450 * count variable between the two sessions, if we'd do more requests
1451 * with the original session, though.)
1453 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1455 /* complete the connection */
1456 if (ffurl_connect(rt->rtsp_hd_out)) {
1461 /* open the tcp connection */
1462 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1463 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1467 rt->rtsp_hd_out = rt->rtsp_hd;
1471 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1472 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1473 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1474 NULL, 0, NI_NUMERICHOST);
1477 /* request options supported by the server; this also detects server
1479 for (rt->server_type = RTSP_SERVER_RTP;;) {
1481 if (rt->server_type == RTSP_SERVER_REAL)
1484 * The following entries are required for proper
1485 * streaming from a Realmedia server. They are
1486 * interdependent in some way although we currently
1487 * don't quite understand how. Values were copied
1488 * from mplayer SVN r23589.
1489 * @param CompanyID is a 16-byte ID in base64
1490 * @param ClientChallenge is a 16-byte ID in hex
1492 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1493 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1494 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1495 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1497 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1498 if (reply->status_code != RTSP_STATUS_OK) {
1499 err = AVERROR_INVALIDDATA;
1503 /* detect server type if not standard-compliant RTP */
1504 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1505 rt->server_type = RTSP_SERVER_REAL;
1507 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1508 rt->server_type = RTSP_SERVER_WMS;
1509 } else if (rt->server_type == RTSP_SERVER_REAL)
1510 strcpy(real_challenge, reply->real_challenge);
1514 if (s->iformat && CONFIG_RTSP_DEMUXER)
1515 err = ff_rtsp_setup_input_streams(s, reply);
1516 else if (CONFIG_RTSP_MUXER)
1517 err = ff_rtsp_setup_output_streams(s, host);
1522 int lower_transport = ff_log2_tab[lower_transport_mask &
1523 ~(lower_transport_mask - 1)];
1525 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1526 rt->server_type == RTSP_SERVER_REAL ?
1527 real_challenge : NULL);
1530 lower_transport_mask &= ~(1 << lower_transport);
1531 if (lower_transport_mask == 0 && err == 1) {
1532 err = AVERROR(EPROTONOSUPPORT);
1537 rt->lower_transport_mask = lower_transport_mask;
1538 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1539 rt->state = RTSP_STATE_IDLE;
1540 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1543 ff_rtsp_close_streams(s);
1544 ff_rtsp_close_connections(s);
1545 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1546 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1547 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1555 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1558 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1559 uint8_t *buf, int buf_size, int64_t wait_end)
1561 RTSPState *rt = s->priv_data;
1562 RTSPStream *rtsp_st;
1563 int n, i, ret, tcp_fd, timeout_cnt = 0;
1565 struct pollfd *p = rt->p;
1568 if (url_interrupt_cb())
1569 return AVERROR_EXIT;
1570 if (wait_end && wait_end - av_gettime() < 0)
1571 return AVERROR(EAGAIN);
1574 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1575 p[max_p].fd = tcp_fd;
1576 p[max_p++].events = POLLIN;
1580 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1581 rtsp_st = rt->rtsp_streams[i];
1582 if (rtsp_st->rtp_handle) {
1583 p[max_p].fd = url_get_file_handle(rtsp_st->rtp_handle);
1584 p[max_p++].events = POLLIN;
1585 p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1586 p[max_p++].events = POLLIN;
1589 n = poll(p, max_p, POLL_TIMEOUT_MS);
1591 int j = 1 - (tcp_fd == -1);
1593 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1594 rtsp_st = rt->rtsp_streams[i];
1595 if (rtsp_st->rtp_handle) {
1596 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1597 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1599 *prtsp_st = rtsp_st;
1606 #if CONFIG_RTSP_DEMUXER
1607 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1608 RTSPMessageHeader reply;
1610 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1613 /* XXX: parse message */
1614 if (rt->state != RTSP_STATE_STREAMING)
1618 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1619 return AVERROR(ETIMEDOUT);
1620 } else if (n < 0 && errno != EINTR)
1621 return AVERROR(errno);
1625 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1627 RTSPState *rt = s->priv_data;
1629 RTSPStream *rtsp_st, *first_queue_st = NULL;
1630 int64_t wait_end = 0;
1632 if (rt->nb_byes == rt->nb_rtsp_streams)
1635 /* get next frames from the same RTP packet */
1636 if (rt->cur_transport_priv) {
1637 if (rt->transport == RTSP_TRANSPORT_RDT) {
1638 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1640 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1642 rt->cur_transport_priv = NULL;
1644 } else if (ret == 1) {
1647 rt->cur_transport_priv = NULL;
1650 if (rt->transport == RTSP_TRANSPORT_RTP) {
1652 int64_t first_queue_time = 0;
1653 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1654 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1658 queue_time = ff_rtp_queued_packet_time(rtpctx);
1659 if (queue_time && (queue_time - first_queue_time < 0 ||
1660 !first_queue_time)) {
1661 first_queue_time = queue_time;
1662 first_queue_st = rt->rtsp_streams[i];
1665 if (first_queue_time)
1666 wait_end = first_queue_time + s->max_delay;
1669 /* read next RTP packet */
1672 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1674 return AVERROR(ENOMEM);
1677 switch(rt->lower_transport) {
1679 #if CONFIG_RTSP_DEMUXER
1680 case RTSP_LOWER_TRANSPORT_TCP:
1681 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1684 case RTSP_LOWER_TRANSPORT_UDP:
1685 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1686 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1687 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1688 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1691 if (len == AVERROR(EAGAIN) && first_queue_st &&
1692 rt->transport == RTSP_TRANSPORT_RTP) {
1693 rtsp_st = first_queue_st;
1694 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1701 if (rt->transport == RTSP_TRANSPORT_RDT) {
1702 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1704 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1706 /* Either bad packet, or a RTCP packet. Check if the
1707 * first_rtcp_ntp_time field was initialized. */
1708 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1709 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1710 /* first_rtcp_ntp_time has been initialized for this stream,
1711 * copy the same value to all other uninitialized streams,
1712 * in order to map their timestamp origin to the same ntp time
1715 AVStream *st = NULL;
1716 if (rtsp_st->stream_index >= 0)
1717 st = s->streams[rtsp_st->stream_index];
1718 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1719 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1720 AVStream *st2 = NULL;
1721 if (rt->rtsp_streams[i]->stream_index >= 0)
1722 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1723 if (rtpctx2 && st && st2 &&
1724 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1725 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1726 rtpctx2->rtcp_ts_offset = av_rescale_q(
1727 rtpctx->rtcp_ts_offset, st->time_base,
1732 if (ret == -RTCP_BYE) {
1735 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1736 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1738 if (rt->nb_byes == rt->nb_rtsp_streams)
1747 /* more packets may follow, so we save the RTP context */
1748 rt->cur_transport_priv = rtsp_st->transport_priv;
1752 #endif /* CONFIG_RTPDEC */
1754 #if CONFIG_SDP_DEMUXER
1755 static int sdp_probe(AVProbeData *p1)
1757 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1759 /* we look for a line beginning "c=IN IP" */
1760 while (p < p_end && *p != '\0') {
1761 if (p + sizeof("c=IN IP") - 1 < p_end &&
1762 av_strstart(p, "c=IN IP", NULL))
1763 return AVPROBE_SCORE_MAX / 2;
1765 while (p < p_end - 1 && *p != '\n') p++;
1774 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1776 RTSPState *rt = s->priv_data;
1777 RTSPStream *rtsp_st;
1782 if (!ff_network_init())
1783 return AVERROR(EIO);
1785 /* read the whole sdp file */
1786 /* XXX: better loading */
1787 content = av_malloc(SDP_MAX_SIZE);
1788 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1791 return AVERROR_INVALIDDATA;
1793 content[size] ='\0';
1795 err = ff_sdp_parse(s, content);
1799 /* open each RTP stream */
1800 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1802 rtsp_st = rt->rtsp_streams[i];
1804 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1805 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1806 ff_url_join(url, sizeof(url), "rtp", NULL,
1807 namebuf, rtsp_st->sdp_port,
1808 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1810 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1811 err = AVERROR_INVALIDDATA;
1814 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1819 ff_rtsp_close_streams(s);
1824 static int sdp_read_close(AVFormatContext *s)
1826 ff_rtsp_close_streams(s);
1831 AVInputFormat ff_sdp_demuxer = {
1833 NULL_IF_CONFIG_SMALL("SDP"),
1837 ff_rtsp_fetch_packet,
1840 #endif /* CONFIG_SDP_DEMUXER */
1842 #if CONFIG_RTP_DEMUXER
1843 static int rtp_probe(AVProbeData *p)
1845 if (av_strstart(p->filename, "rtp:", NULL))
1846 return AVPROBE_SCORE_MAX;
1850 static int rtp_read_header(AVFormatContext *s,
1851 AVFormatParameters *ap)
1853 uint8_t recvbuf[1500];
1854 char host[500], sdp[500];
1856 URLContext* in = NULL;
1858 AVCodecContext codec;
1859 struct sockaddr_storage addr;
1861 socklen_t addrlen = sizeof(addr);
1863 if (!ff_network_init())
1864 return AVERROR(EIO);
1866 ret = url_open(&in, s->filename, URL_RDONLY);
1871 ret = url_read(in, recvbuf, sizeof(recvbuf));
1872 if (ret == AVERROR(EAGAIN))
1877 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1881 if ((recvbuf[0] & 0xc0) != 0x80) {
1882 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1887 payload_type = recvbuf[1] & 0x7f;
1890 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1894 memset(&codec, 0, sizeof(codec));
1895 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1896 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1897 "without an SDP file describing it\n",
1901 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1902 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1903 "properly you need an SDP file "
1907 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1908 NULL, 0, s->filename);
1910 snprintf(sdp, sizeof(sdp),
1911 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1912 addr.ss_family == AF_INET ? 4 : 6, host,
1913 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1914 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1915 port, payload_type);
1916 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1918 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1921 /* sdp_read_header initializes this again */
1924 ret = sdp_read_header(s, ap);
1935 AVInputFormat ff_rtp_demuxer = {
1937 NULL_IF_CONFIG_SMALL("RTP input format"),
1941 ff_rtsp_fetch_packet,
1943 .flags = AVFMT_NOFILE,
1945 #endif /* CONFIG_RTP_DEMUXER */