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/random_seed.h"
30 #include <sys/select.h>
35 #include "os_support.h"
41 #include "rtpdec_formats.h"
44 //#define DEBUG_RTP_TCP
46 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
47 int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
50 /* Timeout values for socket select, in ms,
51 * and read_packet(), in seconds */
52 #define SELECT_TIMEOUT_MS 100
53 #define READ_PACKET_TIMEOUT_S 10
54 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_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 the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
90 static int sdp_parse_rtpmap(AVFormatContext *s,
91 AVCodecContext *codec, RTSPStream *rtsp_st,
92 int payload_type, const char *p)
99 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
100 * see if we can handle this kind of payload.
101 * The space should normally not be there but some Real streams or
102 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
103 * have a trailing space. */
104 get_word_sep(buf, sizeof(buf), "/ ", &p);
105 if (payload_type >= RTP_PT_PRIVATE) {
106 RTPDynamicProtocolHandler *handler;
107 for (handler = RTPFirstDynamicPayloadHandler;
108 handler; handler = handler->next) {
109 if (!strcasecmp(buf, handler->enc_name) &&
110 codec->codec_type == handler->codec_type) {
111 codec->codec_id = handler->codec_id;
112 rtsp_st->dynamic_handler = handler;
114 rtsp_st->dynamic_protocol_context = handler->open();
118 /* If no dynamic handler was found, check with the list of standard
119 * allocated types, if such a stream for some reason happens to
120 * use a private payload type. This isn't handled in rtpdec.c, since
121 * the format name from the rtpmap line never is passed into rtpdec. */
122 if (!rtsp_st->dynamic_handler)
123 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
125 /* We are in a standard case
126 * (from http://www.iana.org/assignments/rtp-parameters). */
127 /* search into AVRtpPayloadTypes[] */
128 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
131 c = avcodec_find_decoder(codec->codec_id);
137 get_word_sep(buf, sizeof(buf), "/", &p);
139 switch (codec->codec_type) {
140 case AVMEDIA_TYPE_AUDIO:
141 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
142 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
143 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
145 codec->sample_rate = i;
146 get_word_sep(buf, sizeof(buf), "/", &p);
150 // TODO: there is a bug here; if it is a mono stream, and
151 // less than 22000Hz, faad upconverts to stereo and twice
152 // the frequency. No problem, but the sample rate is being
153 // set here by the sdp line. Patch on its way. (rdm)
155 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
157 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
160 case AVMEDIA_TYPE_VIDEO:
161 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
169 /* parse the attribute line from the fmtp a line of an sdp response. This
170 * is broken out as a function because it is used in rtp_h264.c, which is
172 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
173 char *value, int value_size)
175 *p += strspn(*p, SPACE_CHARS);
177 get_word_sep(attr, attr_size, "=", p);
180 get_word_sep(value, value_size, ";", p);
188 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
190 * Used for seeking in the rtp stream.
192 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
196 p += strspn(p, SPACE_CHARS);
197 if (!av_stristart(p, "npt=", &p))
200 *start = AV_NOPTS_VALUE;
201 *end = AV_NOPTS_VALUE;
203 get_word_sep(buf, sizeof(buf), "-", &p);
204 *start = parse_date(buf, 1);
207 get_word_sep(buf, sizeof(buf), "-", &p);
208 *end = parse_date(buf, 1);
210 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
211 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
214 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
216 struct addrinfo hints, *ai = NULL;
217 memset(&hints, 0, sizeof(hints));
218 hints.ai_flags = AI_NUMERICHOST;
219 if (getaddrinfo(buf, NULL, &hints, &ai))
221 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
226 typedef struct SDPParseState {
228 struct sockaddr_storage default_ip;
230 int skip_media; ///< set if an unknown m= line occurs
233 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
234 int letter, const char *buf)
236 RTSPState *rt = s->priv_data;
237 char buf1[64], st_type[64];
239 enum AVMediaType codec_type;
243 struct sockaddr_storage sdp_ip;
246 dprintf(s, "sdp: %c='%s'\n", letter, buf);
249 if (s1->skip_media && letter != 'm')
253 get_word(buf1, sizeof(buf1), &p);
254 if (strcmp(buf1, "IN") != 0)
256 get_word(buf1, sizeof(buf1), &p);
257 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
259 get_word_sep(buf1, sizeof(buf1), "/", &p);
260 if (get_sockaddr(buf1, &sdp_ip))
265 get_word_sep(buf1, sizeof(buf1), "/", &p);
268 if (s->nb_streams == 0) {
269 s1->default_ip = sdp_ip;
270 s1->default_ttl = ttl;
272 st = s->streams[s->nb_streams - 1];
273 rtsp_st = st->priv_data;
274 rtsp_st->sdp_ip = sdp_ip;
275 rtsp_st->sdp_ttl = ttl;
279 av_metadata_set2(&s->metadata, "title", p, 0);
282 if (s->nb_streams == 0) {
283 av_metadata_set2(&s->metadata, "comment", p, 0);
290 get_word(st_type, sizeof(st_type), &p);
291 if (!strcmp(st_type, "audio")) {
292 codec_type = AVMEDIA_TYPE_AUDIO;
293 } else if (!strcmp(st_type, "video")) {
294 codec_type = AVMEDIA_TYPE_VIDEO;
295 } else if (!strcmp(st_type, "application")) {
296 codec_type = AVMEDIA_TYPE_DATA;
301 rtsp_st = av_mallocz(sizeof(RTSPStream));
304 rtsp_st->stream_index = -1;
305 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
307 rtsp_st->sdp_ip = s1->default_ip;
308 rtsp_st->sdp_ttl = s1->default_ttl;
310 get_word(buf1, sizeof(buf1), &p); /* port */
311 rtsp_st->sdp_port = atoi(buf1);
313 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
315 /* XXX: handle list of formats */
316 get_word(buf1, sizeof(buf1), &p); /* format list */
317 rtsp_st->sdp_payload_type = atoi(buf1);
319 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
320 /* no corresponding stream */
322 st = av_new_stream(s, 0);
325 st->priv_data = rtsp_st;
326 rtsp_st->stream_index = st->index;
327 st->codec->codec_type = codec_type;
328 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
329 /* if standard payload type, we can find the codec right now */
330 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
333 /* put a default control url */
334 av_strlcpy(rtsp_st->control_url, rt->control_uri,
335 sizeof(rtsp_st->control_url));
338 if (av_strstart(p, "control:", &p)) {
339 if (s->nb_streams == 0) {
340 if (!strncmp(p, "rtsp://", 7))
341 av_strlcpy(rt->control_uri, p,
342 sizeof(rt->control_uri));
345 /* get the control url */
346 st = s->streams[s->nb_streams - 1];
347 rtsp_st = st->priv_data;
349 /* XXX: may need to add full url resolution */
350 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
352 if (proto[0] == '\0') {
353 /* relative control URL */
354 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
355 av_strlcat(rtsp_st->control_url, "/",
356 sizeof(rtsp_st->control_url));
357 av_strlcat(rtsp_st->control_url, p,
358 sizeof(rtsp_st->control_url));
360 av_strlcpy(rtsp_st->control_url, p,
361 sizeof(rtsp_st->control_url));
363 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
364 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
365 get_word(buf1, sizeof(buf1), &p);
366 payload_type = atoi(buf1);
367 st = s->streams[s->nb_streams - 1];
368 rtsp_st = st->priv_data;
369 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
370 } else if (av_strstart(p, "fmtp:", &p) ||
371 av_strstart(p, "framesize:", &p)) {
372 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
373 // let dynamic protocol handlers have a stab at the line.
374 get_word(buf1, sizeof(buf1), &p);
375 payload_type = atoi(buf1);
376 for (i = 0; i < s->nb_streams; i++) {
378 rtsp_st = st->priv_data;
379 if (rtsp_st->sdp_payload_type == payload_type &&
380 rtsp_st->dynamic_handler &&
381 rtsp_st->dynamic_handler->parse_sdp_a_line)
382 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
383 rtsp_st->dynamic_protocol_context, buf);
385 } else if (av_strstart(p, "range:", &p)) {
388 // this is so that seeking on a streamed file can work.
389 rtsp_parse_range_npt(p, &start, &end);
390 s->start_time = start;
391 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
392 s->duration = (end == AV_NOPTS_VALUE) ?
393 AV_NOPTS_VALUE : end - start;
394 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
396 rt->transport = RTSP_TRANSPORT_RDT;
398 if (rt->server_type == RTSP_SERVER_WMS)
399 ff_wms_parse_sdp_a_line(s, p);
400 if (s->nb_streams > 0) {
401 if (rt->server_type == RTSP_SERVER_REAL)
402 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
404 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
405 if (rtsp_st->dynamic_handler &&
406 rtsp_st->dynamic_handler->parse_sdp_a_line)
407 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
409 rtsp_st->dynamic_protocol_context, buf);
416 static int sdp_parse(AVFormatContext *s, const char *content)
420 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
421 * contain long SDP lines containing complete ASF Headers (several
422 * kB) or arrays of MDPR (RM stream descriptor) headers plus
423 * "rulebooks" describing their properties. Therefore, the SDP line
426 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
427 * in rtpdec_xiph.c. */
429 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
431 memset(s1, 0, sizeof(SDPParseState));
434 p += strspn(p, SPACE_CHARS);
442 /* get the content */
444 while (*p != '\n' && *p != '\r' && *p != '\0') {
445 if ((q - buf) < sizeof(buf) - 1)
450 sdp_parse_line(s, s1, letter, buf);
452 while (*p != '\n' && *p != '\0')
460 /* close and free RTSP streams */
461 void ff_rtsp_close_streams(AVFormatContext *s)
463 RTSPState *rt = s->priv_data;
467 for (i = 0; i < rt->nb_rtsp_streams; i++) {
468 rtsp_st = rt->rtsp_streams[i];
470 if (rtsp_st->transport_priv) {
472 AVFormatContext *rtpctx = rtsp_st->transport_priv;
473 av_write_trailer(rtpctx);
474 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
476 url_close_dyn_buf(rtpctx->pb, &ptr);
479 url_fclose(rtpctx->pb);
481 av_metadata_free(&rtpctx->streams[0]->metadata);
482 av_metadata_free(&rtpctx->metadata);
483 av_free(rtpctx->streams[0]);
485 } else if (rt->transport == RTSP_TRANSPORT_RDT)
486 ff_rdt_parse_close(rtsp_st->transport_priv);
488 rtp_parse_close(rtsp_st->transport_priv);
490 if (rtsp_st->rtp_handle)
491 url_close(rtsp_st->rtp_handle);
492 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
493 rtsp_st->dynamic_handler->close(
494 rtsp_st->dynamic_protocol_context);
497 av_free(rt->rtsp_streams);
499 av_close_input_stream (rt->asf_ctx);
502 av_free(rt->recvbuf);
505 static AVFormatContext *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
506 URLContext *handle, int packet_size)
508 AVFormatContext *rtpctx;
510 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
515 /* Allocate an AVFormatContext for each output stream */
516 rtpctx = avformat_alloc_context();
520 rtpctx->oformat = rtp_format;
521 if (!av_new_stream(rtpctx, 0)) {
525 /* Copy the max delay setting; the rtp muxer reads this. */
526 rtpctx->max_delay = s->max_delay;
527 /* Copy other stream parameters. */
528 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
530 /* Set the synchronized start time. */
531 rtpctx->start_time_realtime = s->start_time_realtime;
533 /* Remove the local codec, link to the original codec
534 * context instead, to give the rtp muxer access to
535 * codec parameters. */
536 av_free(rtpctx->streams[0]->codec);
537 rtpctx->streams[0]->codec = st->codec;
540 url_fdopen(&rtpctx->pb, handle);
542 url_open_dyn_packet_buf(&rtpctx->pb, packet_size);
543 ret = av_write_header(rtpctx);
547 url_fclose(rtpctx->pb);
550 url_close_dyn_buf(rtpctx->pb, &ptr);
553 av_free(rtpctx->streams[0]);
558 /* Copy the RTP AVStream timebase back to the original AVStream */
559 st->time_base = rtpctx->streams[0]->time_base;
563 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
565 RTSPState *rt = s->priv_data;
568 /* open the RTP context */
569 if (rtsp_st->stream_index >= 0)
570 st = s->streams[rtsp_st->stream_index];
572 s->ctx_flags |= AVFMTCTX_NOHEADER;
575 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle,
576 RTSP_TCP_MAX_PACKET_SIZE);
577 /* Ownership of rtp_handle is passed to the rtp mux context */
578 rtsp_st->rtp_handle = NULL;
579 } else if (rt->transport == RTSP_TRANSPORT_RDT)
580 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
581 rtsp_st->dynamic_protocol_context,
582 rtsp_st->dynamic_handler);
584 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
585 rtsp_st->sdp_payload_type,
586 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
587 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
589 if (!rtsp_st->transport_priv) {
590 return AVERROR(ENOMEM);
591 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
592 if (rtsp_st->dynamic_handler) {
593 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
594 rtsp_st->dynamic_protocol_context,
595 rtsp_st->dynamic_handler);
602 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
603 static int rtsp_probe(AVProbeData *p)
605 if (av_strstart(p->filename, "rtsp:", NULL))
606 return AVPROBE_SCORE_MAX;
610 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
616 p += strspn(p, SPACE_CHARS);
617 v = strtol(p, (char **)&p, 10);
621 v = strtol(p, (char **)&p, 10);
630 /* XXX: only one transport specification is parsed */
631 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
633 char transport_protocol[16];
635 char lower_transport[16];
637 RTSPTransportField *th;
640 reply->nb_transports = 0;
643 p += strspn(p, SPACE_CHARS);
647 th = &reply->transports[reply->nb_transports];
649 get_word_sep(transport_protocol, sizeof(transport_protocol),
651 if (!strcasecmp (transport_protocol, "rtp")) {
652 get_word_sep(profile, sizeof(profile), "/;,", &p);
653 lower_transport[0] = '\0';
654 /* rtp/avp/<protocol> */
656 get_word_sep(lower_transport, sizeof(lower_transport),
659 th->transport = RTSP_TRANSPORT_RTP;
660 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
661 !strcasecmp (transport_protocol, "x-real-rdt")) {
662 /* x-pn-tng/<protocol> */
663 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
665 th->transport = RTSP_TRANSPORT_RDT;
667 if (!strcasecmp(lower_transport, "TCP"))
668 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
670 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
674 /* get each parameter */
675 while (*p != '\0' && *p != ',') {
676 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
677 if (!strcmp(parameter, "port")) {
680 rtsp_parse_range(&th->port_min, &th->port_max, &p);
682 } else if (!strcmp(parameter, "client_port")) {
685 rtsp_parse_range(&th->client_port_min,
686 &th->client_port_max, &p);
688 } else if (!strcmp(parameter, "server_port")) {
691 rtsp_parse_range(&th->server_port_min,
692 &th->server_port_max, &p);
694 } else if (!strcmp(parameter, "interleaved")) {
697 rtsp_parse_range(&th->interleaved_min,
698 &th->interleaved_max, &p);
700 } else if (!strcmp(parameter, "multicast")) {
701 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
702 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
703 } else if (!strcmp(parameter, "ttl")) {
706 th->ttl = strtol(p, (char **)&p, 10);
708 } else if (!strcmp(parameter, "destination")) {
711 get_word_sep(buf, sizeof(buf), ";,", &p);
712 get_sockaddr(buf, &th->destination);
714 } else if (!strcmp(parameter, "source")) {
717 get_word_sep(buf, sizeof(buf), ";,", &p);
718 av_strlcpy(th->source, buf, sizeof(th->source));
722 while (*p != ';' && *p != '\0' && *p != ',')
730 reply->nb_transports++;
734 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
735 HTTPAuthState *auth_state)
739 /* NOTE: we do case independent match for broken servers */
741 if (av_stristart(p, "Session:", &p)) {
743 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
744 if (av_stristart(p, ";timeout=", &p) &&
745 (t = strtol(p, NULL, 10)) > 0) {
748 } else if (av_stristart(p, "Content-Length:", &p)) {
749 reply->content_length = strtol(p, NULL, 10);
750 } else if (av_stristart(p, "Transport:", &p)) {
751 rtsp_parse_transport(reply, p);
752 } else if (av_stristart(p, "CSeq:", &p)) {
753 reply->seq = strtol(p, NULL, 10);
754 } else if (av_stristart(p, "Range:", &p)) {
755 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
756 } else if (av_stristart(p, "RealChallenge1:", &p)) {
757 p += strspn(p, SPACE_CHARS);
758 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
759 } else if (av_stristart(p, "Server:", &p)) {
760 p += strspn(p, SPACE_CHARS);
761 av_strlcpy(reply->server, p, sizeof(reply->server));
762 } else if (av_stristart(p, "Notice:", &p) ||
763 av_stristart(p, "X-Notice:", &p)) {
764 reply->notice = strtol(p, NULL, 10);
765 } else if (av_stristart(p, "Location:", &p)) {
766 p += strspn(p, SPACE_CHARS);
767 av_strlcpy(reply->location, p , sizeof(reply->location));
768 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
769 p += strspn(p, SPACE_CHARS);
770 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
771 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
772 p += strspn(p, SPACE_CHARS);
773 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
777 /* skip a RTP/TCP interleaved packet */
778 void ff_rtsp_skip_packet(AVFormatContext *s)
780 RTSPState *rt = s->priv_data;
784 ret = url_read_complete(rt->rtsp_hd, buf, 3);
787 len = AV_RB16(buf + 1);
789 dprintf(s, "skipping RTP packet len=%d\n", len);
794 if (len1 > sizeof(buf))
796 ret = url_read_complete(rt->rtsp_hd, buf, len1);
803 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
804 unsigned char **content_ptr,
805 int return_on_interleaved_data)
807 RTSPState *rt = s->priv_data;
808 char buf[4096], buf1[1024], *q;
811 int ret, content_length, line_count = 0;
812 unsigned char *content = NULL;
814 memset(reply, 0, sizeof(*reply));
816 /* parse reply (XXX: use buffers) */
817 rt->last_reply[0] = '\0';
821 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
823 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
830 /* XXX: only parse it if first char on line ? */
831 if (return_on_interleaved_data) {
834 ff_rtsp_skip_packet(s);
835 } else if (ch != '\r') {
836 if ((q - buf) < sizeof(buf) - 1)
842 dprintf(s, "line='%s'\n", buf);
844 /* test if last line */
848 if (line_count == 0) {
850 get_word(buf1, sizeof(buf1), &p);
851 get_word(buf1, sizeof(buf1), &p);
852 reply->status_code = atoi(buf1);
853 av_strlcpy(reply->reason, p, sizeof(reply->reason));
855 ff_rtsp_parse_line(reply, p, &rt->auth_state);
856 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
857 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
862 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
863 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
865 content_length = reply->content_length;
866 if (content_length > 0) {
867 /* leave some room for a trailing '\0' (useful for simple parsing) */
868 content = av_malloc(content_length + 1);
869 (void)url_read_complete(rt->rtsp_hd, content, content_length);
870 content[content_length] = '\0';
873 *content_ptr = content;
877 if (rt->seq != reply->seq) {
878 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
879 rt->seq, reply->seq);
883 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
884 reply->notice == 2104 /* Start-of-Stream Reached */ ||
885 reply->notice == 2306 /* Continuous Feed Terminated */) {
886 rt->state = RTSP_STATE_IDLE;
887 } else if (reply->notice >= 4400 && reply->notice < 5500) {
888 return AVERROR(EIO); /* data or server error */
889 } else if (reply->notice == 2401 /* Ticket Expired */ ||
890 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
891 return AVERROR(EPERM);
896 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
897 const char *method, const char *url,
899 const unsigned char *send_content,
900 int send_content_length)
902 RTSPState *rt = s->priv_data;
903 char buf[4096], *out_buf;
904 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
906 /* Add in RTSP headers */
909 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
911 av_strlcat(buf, headers, sizeof(buf));
912 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
913 if (rt->session_id[0] != '\0' && (!headers ||
914 !strstr(headers, "\nIf-Match:"))) {
915 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
918 char *str = ff_http_auth_create_response(&rt->auth_state,
919 rt->auth, url, method);
921 av_strlcat(buf, str, sizeof(buf));
924 if (send_content_length > 0 && send_content)
925 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
926 av_strlcat(buf, "\r\n", sizeof(buf));
928 /* base64 encode rtsp if tunneling */
929 if (rt->control_transport == RTSP_MODE_TUNNEL) {
930 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
934 dprintf(s, "Sending:\n%s--\n", buf);
936 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
937 if (send_content_length > 0 && send_content) {
938 if (rt->control_transport == RTSP_MODE_TUNNEL) {
939 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
940 "with content data not supported\n");
941 return AVERROR_PATCHWELCOME;
943 url_write(rt->rtsp_hd_out, send_content, send_content_length);
945 rt->last_cmd_time = av_gettime();
950 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
951 const char *url, const char *headers)
953 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
956 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
957 const char *headers, RTSPMessageHeader *reply,
958 unsigned char **content_ptr)
960 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
961 content_ptr, NULL, 0);
964 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
965 const char *method, const char *url,
967 RTSPMessageHeader *reply,
968 unsigned char **content_ptr,
969 const unsigned char *send_content,
970 int send_content_length)
972 RTSPState *rt = s->priv_data;
973 HTTPAuthType cur_auth_type;
977 cur_auth_type = rt->auth_state.auth_type;
978 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
980 send_content_length)))
983 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
986 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
987 rt->auth_state.auth_type != HTTP_AUTH_NONE)
990 if (reply->status_code > 400){
991 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
995 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1002 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1004 static int make_setup_request(AVFormatContext *s, const char *host, int port,
1005 int lower_transport, const char *real_challenge)
1007 RTSPState *rt = s->priv_data;
1008 int rtx, j, i, err, interleave = 0;
1009 RTSPStream *rtsp_st;
1010 RTSPMessageHeader reply1, *reply = &reply1;
1012 const char *trans_pref;
1014 if (rt->transport == RTSP_TRANSPORT_RDT)
1015 trans_pref = "x-pn-tng";
1017 trans_pref = "RTP/AVP";
1019 /* default timeout: 1 minute */
1022 /* for each stream, make the setup request */
1023 /* XXX: we assume the same server is used for the control of each
1026 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1027 char transport[2048];
1030 * WMS serves all UDP data over a single connection, the RTX, which
1031 * isn't necessarily the first in the SDP but has to be the first
1032 * to be set up, else the second/third SETUP will fail with a 461.
1034 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1035 rt->server_type == RTSP_SERVER_WMS) {
1038 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1039 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1041 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1045 if (rtx == rt->nb_rtsp_streams)
1046 return -1; /* no RTX found */
1047 rtsp_st = rt->rtsp_streams[rtx];
1049 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1051 rtsp_st = rt->rtsp_streams[i];
1054 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1057 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1058 port = reply->transports[0].client_port_min;
1062 /* first try in specified port range */
1063 if (RTSP_RTP_PORT_MIN != 0) {
1064 while (j <= RTSP_RTP_PORT_MAX) {
1065 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1066 "?localport=%d", j);
1067 /* we will use two ports per rtp stream (rtp and rtcp) */
1069 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1075 /* then try on any port */
1076 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1077 err = AVERROR_INVALIDDATA;
1083 port = rtp_get_local_port(rtsp_st->rtp_handle);
1085 snprintf(transport, sizeof(transport) - 1,
1086 "%s/UDP;", trans_pref);
1087 if (rt->server_type != RTSP_SERVER_REAL)
1088 av_strlcat(transport, "unicast;", sizeof(transport));
1089 av_strlcatf(transport, sizeof(transport),
1090 "client_port=%d", port);
1091 if (rt->transport == RTSP_TRANSPORT_RTP &&
1092 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1093 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1097 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1098 /** For WMS streams, the application streams are only used for
1099 * UDP. When trying to set it up for TCP streams, the server
1100 * will return an error. Therefore, we skip those streams. */
1101 if (rt->server_type == RTSP_SERVER_WMS &&
1102 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1105 snprintf(transport, sizeof(transport) - 1,
1106 "%s/TCP;", trans_pref);
1107 if (rt->server_type == RTSP_SERVER_WMS)
1108 av_strlcat(transport, "unicast;", sizeof(transport));
1109 av_strlcatf(transport, sizeof(transport),
1110 "interleaved=%d-%d",
1111 interleave, interleave + 1);
1115 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1116 snprintf(transport, sizeof(transport) - 1,
1117 "%s/UDP;multicast", trans_pref);
1120 av_strlcat(transport, ";mode=receive", sizeof(transport));
1121 } else if (rt->server_type == RTSP_SERVER_REAL ||
1122 rt->server_type == RTSP_SERVER_WMS)
1123 av_strlcat(transport, ";mode=play", sizeof(transport));
1124 snprintf(cmd, sizeof(cmd),
1125 "Transport: %s\r\n",
1127 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1128 char real_res[41], real_csum[9];
1129 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1131 av_strlcatf(cmd, sizeof(cmd),
1133 "RealChallenge2: %s, sd=%s\r\n",
1134 rt->session_id, real_res, real_csum);
1136 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1137 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1140 } else if (reply->status_code != RTSP_STATUS_OK ||
1141 reply->nb_transports != 1) {
1142 err = AVERROR_INVALIDDATA;
1146 /* XXX: same protocol for all streams is required */
1148 if (reply->transports[0].lower_transport != rt->lower_transport ||
1149 reply->transports[0].transport != rt->transport) {
1150 err = AVERROR_INVALIDDATA;
1154 rt->lower_transport = reply->transports[0].lower_transport;
1155 rt->transport = reply->transports[0].transport;
1158 /* close RTP connection if not chosen */
1159 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1160 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1161 url_close(rtsp_st->rtp_handle);
1162 rtsp_st->rtp_handle = NULL;
1165 switch(reply->transports[0].lower_transport) {
1166 case RTSP_LOWER_TRANSPORT_TCP:
1167 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1168 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1171 case RTSP_LOWER_TRANSPORT_UDP: {
1174 /* Use source address if specified */
1175 if (reply->transports[0].source[0]) {
1176 ff_url_join(url, sizeof(url), "rtp", NULL,
1177 reply->transports[0].source,
1178 reply->transports[0].server_port_min, NULL);
1180 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1181 reply->transports[0].server_port_min, NULL);
1183 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1184 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1185 err = AVERROR_INVALIDDATA;
1188 /* Try to initialize the connection state in a
1189 * potential NAT router by sending dummy packets.
1190 * RTP/RTCP dummy packets are used for RDT, too.
1192 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1193 rtp_send_punch_packets(rtsp_st->rtp_handle);
1196 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1197 char url[1024], namebuf[50];
1198 struct sockaddr_storage addr;
1201 if (reply->transports[0].destination.ss_family) {
1202 addr = reply->transports[0].destination;
1203 port = reply->transports[0].port_min;
1204 ttl = reply->transports[0].ttl;
1206 addr = rtsp_st->sdp_ip;
1207 port = rtsp_st->sdp_port;
1208 ttl = rtsp_st->sdp_ttl;
1210 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1211 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1212 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1213 port, "?ttl=%d", ttl);
1214 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1215 err = AVERROR_INVALIDDATA;
1222 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1226 if (reply->timeout > 0)
1227 rt->timeout = reply->timeout;
1229 if (rt->server_type == RTSP_SERVER_REAL)
1230 rt->need_subscription = 1;
1235 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1236 if (rt->rtsp_streams[i]->rtp_handle) {
1237 url_close(rt->rtsp_streams[i]->rtp_handle);
1238 rt->rtsp_streams[i]->rtp_handle = NULL;
1244 static int rtsp_read_play(AVFormatContext *s)
1246 RTSPState *rt = s->priv_data;
1247 RTSPMessageHeader reply1, *reply = &reply1;
1251 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1254 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1255 if (rt->state == RTSP_STATE_PAUSED) {
1258 snprintf(cmd, sizeof(cmd),
1259 "Range: npt=%0.3f-\r\n",
1260 (double)rt->seek_timestamp / AV_TIME_BASE);
1262 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1263 if (reply->status_code != RTSP_STATUS_OK) {
1266 if (rt->transport == RTSP_TRANSPORT_RTP) {
1267 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1268 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1269 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1270 AVStream *st = NULL;
1273 if (rtsp_st->stream_index >= 0)
1274 st = s->streams[rtsp_st->stream_index];
1275 ff_rtp_reset_packet_queue(rtpctx);
1276 if (reply->range_start != AV_NOPTS_VALUE) {
1277 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1278 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1280 rtpctx->range_start_offset =
1281 av_rescale_q(reply->range_start, AV_TIME_BASE_Q,
1287 rt->state = RTSP_STATE_STREAMING;
1291 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1293 RTSPState *rt = s->priv_data;
1295 unsigned char *content = NULL;
1298 /* describe the stream */
1299 snprintf(cmd, sizeof(cmd),
1300 "Accept: application/sdp\r\n");
1301 if (rt->server_type == RTSP_SERVER_REAL) {
1303 * The Require: attribute is needed for proper streaming from
1304 * Realmedia servers.
1307 "Require: com.real.retain-entity-for-setup\r\n",
1310 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1312 return AVERROR_INVALIDDATA;
1313 if (reply->status_code != RTSP_STATUS_OK) {
1315 return AVERROR_INVALIDDATA;
1318 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", content);
1319 /* now we got the SDP description, we parse it */
1320 ret = sdp_parse(s, (const char *)content);
1323 return AVERROR_INVALIDDATA;
1328 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1330 RTSPState *rt = s->priv_data;
1331 RTSPMessageHeader reply1, *reply = &reply1;
1334 AVFormatContext sdp_ctx, *ctx_array[1];
1336 s->start_time_realtime = av_gettime();
1338 /* Announce the stream */
1339 sdp = av_mallocz(SDP_MAX_SIZE);
1341 return AVERROR(ENOMEM);
1342 /* We create the SDP based on the RTSP AVFormatContext where we
1343 * aren't allowed to change the filename field. (We create the SDP
1344 * based on the RTSP context since the contexts for the RTP streams
1345 * don't exist yet.) In order to specify a custom URL with the actual
1346 * peer IP instead of the originally specified hostname, we create
1347 * a temporary copy of the AVFormatContext, where the custom URL is set.
1349 * FIXME: Create the SDP without copying the AVFormatContext.
1350 * This either requires setting up the RTP stream AVFormatContexts
1351 * already here (complicating things immensely) or getting a more
1352 * flexible SDP creation interface.
1355 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1356 "rtsp", NULL, addr, -1, NULL);
1357 ctx_array[0] = &sdp_ctx;
1358 if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
1360 return AVERROR_INVALIDDATA;
1362 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1363 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1364 "Content-Type: application/sdp\r\n",
1365 reply, NULL, sdp, strlen(sdp));
1367 if (reply->status_code != RTSP_STATUS_OK)
1368 return AVERROR_INVALIDDATA;
1370 /* Set up the RTSPStreams for each AVStream */
1371 for (i = 0; i < s->nb_streams; i++) {
1372 RTSPStream *rtsp_st;
1373 AVStream *st = s->streams[i];
1375 rtsp_st = av_mallocz(sizeof(RTSPStream));
1377 return AVERROR(ENOMEM);
1378 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1380 st->priv_data = rtsp_st;
1381 rtsp_st->stream_index = i;
1383 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1384 /* Note, this must match the relative uri set in the sdp content */
1385 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1392 void ff_rtsp_close_connections(AVFormatContext *s)
1394 RTSPState *rt = s->priv_data;
1395 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1396 url_close(rt->rtsp_hd);
1397 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1400 int ff_rtsp_connect(AVFormatContext *s)
1402 RTSPState *rt = s->priv_data;
1403 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1404 char *option_list, *option, *filename;
1405 int port, err, tcp_fd;
1406 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1407 int lower_transport_mask = 0;
1408 char real_challenge[64];
1409 struct sockaddr_storage peer;
1410 socklen_t peer_len = sizeof(peer);
1412 if (!ff_network_init())
1413 return AVERROR(EIO);
1415 rt->control_transport = RTSP_MODE_PLAIN;
1416 /* extract hostname and port */
1417 av_url_split(NULL, 0, auth, sizeof(auth),
1418 host, sizeof(host), &port, path, sizeof(path), s->filename);
1420 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1423 port = RTSP_DEFAULT_PORT;
1425 /* search for options */
1426 option_list = strrchr(path, '?');
1428 /* Strip out the RTSP specific options, write out the rest of
1429 * the options back into the same string. */
1430 filename = option_list;
1431 while (option_list) {
1432 /* move the option pointer */
1433 option = ++option_list;
1434 option_list = strchr(option_list, '&');
1438 /* handle the options */
1439 if (!strcmp(option, "udp")) {
1440 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1441 } else if (!strcmp(option, "multicast")) {
1442 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1443 } else if (!strcmp(option, "tcp")) {
1444 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1445 } else if(!strcmp(option, "http")) {
1446 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1447 rt->control_transport = RTSP_MODE_TUNNEL;
1449 /* Write options back into the buffer, using memmove instead
1450 * of strcpy since the strings may overlap. */
1451 int len = strlen(option);
1452 memmove(++filename, option, len);
1454 if (option_list) *filename = '&';
1460 if (!lower_transport_mask)
1461 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1464 /* Only UDP or TCP - UDP multicast isn't supported. */
1465 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1466 (1 << RTSP_LOWER_TRANSPORT_TCP);
1467 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1468 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1469 "only UDP and TCP are supported for output.\n");
1470 err = AVERROR(EINVAL);
1475 /* Construct the URI used in request; this is similar to s->filename,
1476 * but with authentication credentials removed and RTSP specific options
1478 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1479 host, port, "%s", path);
1481 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1482 /* set up initial handshake for tunneling */
1483 char httpname[1024];
1484 char sessioncookie[17];
1487 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1488 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1489 av_get_random_seed(), av_get_random_seed());
1492 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1497 /* generate GET headers */
1498 snprintf(headers, sizeof(headers),
1499 "x-sessioncookie: %s\r\n"
1500 "Accept: application/x-rtsp-tunnelled\r\n"
1501 "Pragma: no-cache\r\n"
1502 "Cache-Control: no-cache\r\n",
1504 ff_http_set_headers(rt->rtsp_hd, headers);
1506 /* complete the connection */
1507 if (url_connect(rt->rtsp_hd)) {
1513 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1518 /* generate POST headers */
1519 snprintf(headers, sizeof(headers),
1520 "x-sessioncookie: %s\r\n"
1521 "Content-Type: application/x-rtsp-tunnelled\r\n"
1522 "Pragma: no-cache\r\n"
1523 "Cache-Control: no-cache\r\n"
1524 "Content-Length: 32767\r\n"
1525 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1527 ff_http_set_headers(rt->rtsp_hd_out, headers);
1528 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1530 /* Initialize the authentication state for the POST session. The HTTP
1531 * protocol implementation doesn't properly handle multi-pass
1532 * authentication for POST requests, since it would require one of
1534 * - implementing Expect: 100-continue, which many HTTP servers
1535 * don't support anyway, even less the RTSP servers that do HTTP
1537 * - sending the whole POST data until getting a 401 reply specifying
1538 * what authentication method to use, then resending all that data
1539 * - waiting for potential 401 replies directly after sending the
1540 * POST header (waiting for some unspecified time)
1541 * Therefore, we copy the full auth state, which works for both basic
1542 * and digest. (For digest, we would have to synchronize the nonce
1543 * count variable between the two sessions, if we'd do more requests
1544 * with the original session, though.)
1546 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1548 /* complete the connection */
1549 if (url_connect(rt->rtsp_hd_out)) {
1554 /* open the tcp connection */
1555 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1556 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1560 rt->rtsp_hd_out = rt->rtsp_hd;
1564 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1565 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1566 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1567 NULL, 0, NI_NUMERICHOST);
1570 /* request options supported by the server; this also detects server
1572 for (rt->server_type = RTSP_SERVER_RTP;;) {
1574 if (rt->server_type == RTSP_SERVER_REAL)
1577 * The following entries are required for proper
1578 * streaming from a Realmedia server. They are
1579 * interdependent in some way although we currently
1580 * don't quite understand how. Values were copied
1581 * from mplayer SVN r23589.
1582 * @param CompanyID is a 16-byte ID in base64
1583 * @param ClientChallenge is a 16-byte ID in hex
1585 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1586 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1587 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1588 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1590 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1591 if (reply->status_code != RTSP_STATUS_OK) {
1592 err = AVERROR_INVALIDDATA;
1596 /* detect server type if not standard-compliant RTP */
1597 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1598 rt->server_type = RTSP_SERVER_REAL;
1600 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1601 rt->server_type = RTSP_SERVER_WMS;
1602 } else if (rt->server_type == RTSP_SERVER_REAL)
1603 strcpy(real_challenge, reply->real_challenge);
1608 err = rtsp_setup_input_streams(s, reply);
1610 err = rtsp_setup_output_streams(s, host);
1615 int lower_transport = ff_log2_tab[lower_transport_mask &
1616 ~(lower_transport_mask - 1)];
1618 err = make_setup_request(s, host, port, lower_transport,
1619 rt->server_type == RTSP_SERVER_REAL ?
1620 real_challenge : NULL);
1623 lower_transport_mask &= ~(1 << lower_transport);
1624 if (lower_transport_mask == 0 && err == 1) {
1625 err = FF_NETERROR(EPROTONOSUPPORT);
1630 rt->state = RTSP_STATE_IDLE;
1631 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1634 ff_rtsp_close_streams(s);
1635 ff_rtsp_close_connections(s);
1636 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1637 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1638 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1646 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1648 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1649 uint8_t *buf, int buf_size, int64_t wait_end)
1651 RTSPState *rt = s->priv_data;
1652 RTSPStream *rtsp_st;
1654 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1658 if (url_interrupt_cb())
1659 return AVERROR(EINTR);
1660 if (wait_end && wait_end - av_gettime() < 0)
1661 return AVERROR(EAGAIN);
1664 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1665 FD_SET(tcp_fd, &rfds);
1670 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1671 rtsp_st = rt->rtsp_streams[i];
1672 if (rtsp_st->rtp_handle) {
1673 fd = url_get_file_handle(rtsp_st->rtp_handle);
1674 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1675 if (FFMAX(fd, fd_rtcp) > fd_max)
1676 fd_max = FFMAX(fd, fd_rtcp);
1678 FD_SET(fd_rtcp, &rfds);
1682 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1683 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1686 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1687 rtsp_st = rt->rtsp_streams[i];
1688 if (rtsp_st->rtp_handle) {
1689 fd = url_get_file_handle(rtsp_st->rtp_handle);
1690 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1691 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1692 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1694 *prtsp_st = rtsp_st;
1700 #if CONFIG_RTSP_DEMUXER
1701 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1702 RTSPMessageHeader reply;
1704 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1707 /* XXX: parse message */
1708 if (rt->state != RTSP_STATE_STREAMING)
1712 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1713 return FF_NETERROR(ETIMEDOUT);
1714 } else if (n < 0 && errno != EINTR)
1715 return AVERROR(errno);
1719 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1720 uint8_t *buf, int buf_size);
1722 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1724 RTSPState *rt = s->priv_data;
1726 RTSPStream *rtsp_st, *first_queue_st = NULL;
1727 int64_t wait_end = 0;
1729 if (rt->nb_byes == rt->nb_rtsp_streams)
1732 /* get next frames from the same RTP packet */
1733 if (rt->cur_transport_priv) {
1734 if (rt->transport == RTSP_TRANSPORT_RDT) {
1735 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1737 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1739 rt->cur_transport_priv = NULL;
1741 } else if (ret == 1) {
1744 rt->cur_transport_priv = NULL;
1747 if (rt->transport == RTSP_TRANSPORT_RTP) {
1749 int64_t first_queue_time = 0;
1750 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1751 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1752 int64_t queue_time = ff_rtp_queued_packet_time(rtpctx);
1753 if (queue_time && (queue_time - first_queue_time < 0 ||
1754 !first_queue_time)) {
1755 first_queue_time = queue_time;
1756 first_queue_st = rt->rtsp_streams[i];
1759 if (first_queue_time)
1760 wait_end = first_queue_time + s->max_delay;
1763 /* read next RTP packet */
1766 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1768 return AVERROR(ENOMEM);
1771 switch(rt->lower_transport) {
1773 #if CONFIG_RTSP_DEMUXER
1774 case RTSP_LOWER_TRANSPORT_TCP:
1775 len = tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1778 case RTSP_LOWER_TRANSPORT_UDP:
1779 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1780 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1781 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1782 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1785 if (len == AVERROR(EAGAIN) && first_queue_st &&
1786 rt->transport == RTSP_TRANSPORT_RTP) {
1787 rtsp_st = first_queue_st;
1788 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1795 if (rt->transport == RTSP_TRANSPORT_RDT) {
1796 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1798 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1800 /* Either bad packet, or a RTCP packet. Check if the
1801 * first_rtcp_ntp_time field was initialized. */
1802 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1803 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1804 /* first_rtcp_ntp_time has been initialized for this stream,
1805 * copy the same value to all other uninitialized streams,
1806 * in order to map their timestamp origin to the same ntp time
1809 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1810 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1812 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1813 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1816 if (ret == -RTCP_BYE) {
1819 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1820 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1822 if (rt->nb_byes == rt->nb_rtsp_streams)
1831 /* more packets may follow, so we save the RTP context */
1832 rt->cur_transport_priv = rtsp_st->transport_priv;
1837 #if CONFIG_RTSP_DEMUXER
1838 static int rtsp_read_header(AVFormatContext *s,
1839 AVFormatParameters *ap)
1841 RTSPState *rt = s->priv_data;
1844 ret = ff_rtsp_connect(s);
1848 rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
1849 if (!rt->real_setup_cache)
1850 return AVERROR(ENOMEM);
1851 rt->real_setup = rt->real_setup_cache + s->nb_streams * sizeof(*rt->real_setup);
1853 if (ap->initial_pause) {
1854 /* do not start immediately */
1856 if (rtsp_read_play(s) < 0) {
1857 ff_rtsp_close_streams(s);
1858 ff_rtsp_close_connections(s);
1859 return AVERROR_INVALIDDATA;
1866 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1867 uint8_t *buf, int buf_size)
1869 RTSPState *rt = s->priv_data;
1870 int id, len, i, ret;
1871 RTSPStream *rtsp_st;
1873 #ifdef DEBUG_RTP_TCP
1874 dprintf(s, "tcp_read_packet:\n");
1878 RTSPMessageHeader reply;
1880 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1883 if (ret == 1) /* received '$' */
1885 /* XXX: parse message */
1886 if (rt->state != RTSP_STATE_STREAMING)
1889 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1893 len = AV_RB16(buf + 1);
1894 #ifdef DEBUG_RTP_TCP
1895 dprintf(s, "id=%d len=%d\n", id, len);
1897 if (len > buf_size || len < 12)
1900 ret = url_read_complete(rt->rtsp_hd, buf, len);
1903 if (rt->transport == RTSP_TRANSPORT_RDT &&
1904 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1907 /* find the matching stream */
1908 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1909 rtsp_st = rt->rtsp_streams[i];
1910 if (id >= rtsp_st->interleaved_min &&
1911 id <= rtsp_st->interleaved_max)
1916 *prtsp_st = rtsp_st;
1919 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1921 RTSPState *rt = s->priv_data;
1923 RTSPMessageHeader reply1, *reply = &reply1;
1926 if (rt->server_type == RTSP_SERVER_REAL) {
1929 for (i = 0; i < s->nb_streams; i++)
1930 rt->real_setup[i] = s->streams[i]->discard;
1932 if (!rt->need_subscription) {
1933 if (memcmp (rt->real_setup, rt->real_setup_cache,
1934 sizeof(enum AVDiscard) * s->nb_streams)) {
1935 snprintf(cmd, sizeof(cmd),
1936 "Unsubscribe: %s\r\n",
1937 rt->last_subscription);
1938 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1940 if (reply->status_code != RTSP_STATUS_OK)
1941 return AVERROR_INVALIDDATA;
1942 rt->need_subscription = 1;
1946 if (rt->need_subscription) {
1947 int r, rule_nr, first = 1;
1949 memcpy(rt->real_setup_cache, rt->real_setup,
1950 sizeof(enum AVDiscard) * s->nb_streams);
1951 rt->last_subscription[0] = 0;
1953 snprintf(cmd, sizeof(cmd),
1955 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1957 for (r = 0; r < s->nb_streams; r++) {
1958 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1959 if (s->streams[r]->discard != AVDISCARD_ALL) {
1961 av_strlcat(rt->last_subscription, ",",
1962 sizeof(rt->last_subscription));
1963 ff_rdt_subscribe_rule(
1964 rt->last_subscription,
1965 sizeof(rt->last_subscription), i, rule_nr);
1972 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1973 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1975 if (reply->status_code != RTSP_STATUS_OK)
1976 return AVERROR_INVALIDDATA;
1977 rt->need_subscription = 0;
1979 if (rt->state == RTSP_STATE_STREAMING)
1984 ret = rtsp_fetch_packet(s, pkt);
1988 /* send dummy request to keep TCP connection alive */
1989 if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1990 if (rt->server_type == RTSP_SERVER_WMS) {
1991 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1993 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
2000 /* pause the stream */
2001 static int rtsp_read_pause(AVFormatContext *s)
2003 RTSPState *rt = s->priv_data;
2004 RTSPMessageHeader reply1, *reply = &reply1;
2006 if (rt->state != RTSP_STATE_STREAMING)
2008 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
2009 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
2010 if (reply->status_code != RTSP_STATUS_OK) {
2014 rt->state = RTSP_STATE_PAUSED;
2018 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
2019 int64_t timestamp, int flags)
2021 RTSPState *rt = s->priv_data;
2023 rt->seek_timestamp = av_rescale_q(timestamp,
2024 s->streams[stream_index]->time_base,
2028 case RTSP_STATE_IDLE:
2030 case RTSP_STATE_STREAMING:
2031 if (rtsp_read_pause(s) != 0)
2033 rt->state = RTSP_STATE_SEEKING;
2034 if (rtsp_read_play(s) != 0)
2037 case RTSP_STATE_PAUSED:
2038 rt->state = RTSP_STATE_IDLE;
2044 static int rtsp_read_close(AVFormatContext *s)
2046 RTSPState *rt = s->priv_data;
2049 /* NOTE: it is valid to flush the buffer here */
2050 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
2051 url_fclose(&rt->rtsp_gb);
2054 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
2056 ff_rtsp_close_streams(s);
2057 ff_rtsp_close_connections(s);
2059 rt->real_setup = NULL;
2060 av_freep(&rt->real_setup_cache);
2064 AVInputFormat rtsp_demuxer = {
2066 NULL_IF_CONFIG_SMALL("RTSP input format"),
2073 .flags = AVFMT_NOFILE,
2074 .read_play = rtsp_read_play,
2075 .read_pause = rtsp_read_pause,
2077 #endif /* CONFIG_RTSP_DEMUXER */
2079 static int sdp_probe(AVProbeData *p1)
2081 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2083 /* we look for a line beginning "c=IN IP" */
2084 while (p < p_end && *p != '\0') {
2085 if (p + sizeof("c=IN IP") - 1 < p_end &&
2086 av_strstart(p, "c=IN IP", NULL))
2087 return AVPROBE_SCORE_MAX / 2;
2089 while (p < p_end - 1 && *p != '\n') p++;
2098 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2100 RTSPState *rt = s->priv_data;
2101 RTSPStream *rtsp_st;
2106 if (!ff_network_init())
2107 return AVERROR(EIO);
2109 /* read the whole sdp file */
2110 /* XXX: better loading */
2111 content = av_malloc(SDP_MAX_SIZE);
2112 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2115 return AVERROR_INVALIDDATA;
2117 content[size] ='\0';
2119 sdp_parse(s, content);
2122 /* open each RTP stream */
2123 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2125 rtsp_st = rt->rtsp_streams[i];
2127 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2128 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2129 ff_url_join(url, sizeof(url), "rtp", NULL,
2130 namebuf, rtsp_st->sdp_port,
2131 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2133 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2134 err = AVERROR_INVALIDDATA;
2137 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2142 ff_rtsp_close_streams(s);
2147 static int sdp_read_close(AVFormatContext *s)
2149 ff_rtsp_close_streams(s);
2154 AVInputFormat sdp_demuxer = {
2156 NULL_IF_CONFIG_SMALL("SDP"),