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 8192
57 static void get_word_until_chars(char *buf, int buf_size,
58 const char *sep, const char **pp)
64 p += strspn(p, SPACE_CHARS);
66 while (!strchr(sep, *p) && *p != '\0') {
67 if ((q - buf) < buf_size - 1)
76 static void get_word_sep(char *buf, int buf_size, const char *sep,
79 if (**pp == '/') (*pp)++;
80 get_word_until_chars(buf, buf_size, sep, pp);
83 static void get_word(char *buf, int buf_size, const char **pp)
85 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
88 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
89 static int sdp_parse_rtpmap(AVFormatContext *s,
90 AVCodecContext *codec, RTSPStream *rtsp_st,
91 int payload_type, const char *p)
98 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
99 * see if we can handle this kind of payload.
100 * The space should normally not be there but some Real streams or
101 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
102 * have a trailing space. */
103 get_word_sep(buf, sizeof(buf), "/ ", &p);
104 if (payload_type >= RTP_PT_PRIVATE) {
105 RTPDynamicProtocolHandler *handler;
106 for (handler = RTPFirstDynamicPayloadHandler;
107 handler; handler = handler->next) {
108 if (!strcasecmp(buf, handler->enc_name) &&
109 codec->codec_type == handler->codec_type) {
110 codec->codec_id = handler->codec_id;
111 rtsp_st->dynamic_handler = handler;
113 rtsp_st->dynamic_protocol_context = handler->open();
118 /* We are in a standard case
119 * (from http://www.iana.org/assignments/rtp-parameters). */
120 /* search into AVRtpPayloadTypes[] */
121 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
124 c = avcodec_find_decoder(codec->codec_id);
130 get_word_sep(buf, sizeof(buf), "/", &p);
132 switch (codec->codec_type) {
133 case AVMEDIA_TYPE_AUDIO:
134 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
135 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
136 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
138 codec->sample_rate = i;
139 get_word_sep(buf, sizeof(buf), "/", &p);
143 // TODO: there is a bug here; if it is a mono stream, and
144 // less than 22000Hz, faad upconverts to stereo and twice
145 // the frequency. No problem, but the sample rate is being
146 // set here by the sdp line. Patch on its way. (rdm)
148 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
150 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
153 case AVMEDIA_TYPE_VIDEO:
154 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
162 /* parse the attribute line from the fmtp a line of an sdp response. This
163 * is broken out as a function because it is used in rtp_h264.c, which is
165 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
166 char *value, int value_size)
168 *p += strspn(*p, SPACE_CHARS);
170 get_word_sep(attr, attr_size, "=", p);
173 get_word_sep(value, value_size, ";", p);
181 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
183 * Used for seeking in the rtp stream.
185 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
189 p += strspn(p, SPACE_CHARS);
190 if (!av_stristart(p, "npt=", &p))
193 *start = AV_NOPTS_VALUE;
194 *end = AV_NOPTS_VALUE;
196 get_word_sep(buf, sizeof(buf), "-", &p);
197 *start = parse_date(buf, 1);
200 get_word_sep(buf, sizeof(buf), "-", &p);
201 *end = parse_date(buf, 1);
203 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
204 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
207 typedef struct SDPParseState {
209 struct in_addr default_ip;
211 int skip_media; ///< set if an unknown m= line occurs
214 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
215 int letter, const char *buf)
217 RTSPState *rt = s->priv_data;
218 char buf1[64], st_type[64];
220 enum AVMediaType codec_type;
224 struct in_addr sdp_ip;
227 dprintf(s, "sdp: %c='%s'\n", letter, buf);
230 if (s1->skip_media && letter != 'm')
234 get_word(buf1, sizeof(buf1), &p);
235 if (strcmp(buf1, "IN") != 0)
237 get_word(buf1, sizeof(buf1), &p);
238 if (strcmp(buf1, "IP4") != 0)
240 get_word_sep(buf1, sizeof(buf1), "/", &p);
241 if (ff_inet_aton(buf1, &sdp_ip) == 0)
246 get_word_sep(buf1, sizeof(buf1), "/", &p);
249 if (s->nb_streams == 0) {
250 s1->default_ip = sdp_ip;
251 s1->default_ttl = ttl;
253 st = s->streams[s->nb_streams - 1];
254 rtsp_st = st->priv_data;
255 rtsp_st->sdp_ip = sdp_ip;
256 rtsp_st->sdp_ttl = ttl;
260 av_metadata_set2(&s->metadata, "title", p, 0);
263 if (s->nb_streams == 0) {
264 av_metadata_set2(&s->metadata, "comment", p, 0);
271 get_word(st_type, sizeof(st_type), &p);
272 if (!strcmp(st_type, "audio")) {
273 codec_type = AVMEDIA_TYPE_AUDIO;
274 } else if (!strcmp(st_type, "video")) {
275 codec_type = AVMEDIA_TYPE_VIDEO;
276 } else if (!strcmp(st_type, "application")) {
277 codec_type = AVMEDIA_TYPE_DATA;
282 rtsp_st = av_mallocz(sizeof(RTSPStream));
285 rtsp_st->stream_index = -1;
286 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
288 rtsp_st->sdp_ip = s1->default_ip;
289 rtsp_st->sdp_ttl = s1->default_ttl;
291 get_word(buf1, sizeof(buf1), &p); /* port */
292 rtsp_st->sdp_port = atoi(buf1);
294 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
296 /* XXX: handle list of formats */
297 get_word(buf1, sizeof(buf1), &p); /* format list */
298 rtsp_st->sdp_payload_type = atoi(buf1);
300 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
301 /* no corresponding stream */
303 st = av_new_stream(s, 0);
306 st->priv_data = rtsp_st;
307 rtsp_st->stream_index = st->index;
308 st->codec->codec_type = codec_type;
309 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
310 /* if standard payload type, we can find the codec right now */
311 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
314 /* put a default control url */
315 av_strlcpy(rtsp_st->control_url, rt->control_uri,
316 sizeof(rtsp_st->control_url));
319 if (av_strstart(p, "control:", &p)) {
320 if (s->nb_streams == 0) {
321 if (!strncmp(p, "rtsp://", 7))
322 av_strlcpy(rt->control_uri, p,
323 sizeof(rt->control_uri));
326 /* get the control url */
327 st = s->streams[s->nb_streams - 1];
328 rtsp_st = st->priv_data;
330 /* XXX: may need to add full url resolution */
331 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
333 if (proto[0] == '\0') {
334 /* relative control URL */
335 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
336 av_strlcat(rtsp_st->control_url, "/",
337 sizeof(rtsp_st->control_url));
338 av_strlcat(rtsp_st->control_url, p,
339 sizeof(rtsp_st->control_url));
341 av_strlcpy(rtsp_st->control_url, p,
342 sizeof(rtsp_st->control_url));
344 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
345 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
346 get_word(buf1, sizeof(buf1), &p);
347 payload_type = atoi(buf1);
348 st = s->streams[s->nb_streams - 1];
349 rtsp_st = st->priv_data;
350 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
351 } else if (av_strstart(p, "fmtp:", &p) ||
352 av_strstart(p, "framesize:", &p)) {
353 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
354 // let dynamic protocol handlers have a stab at the line.
355 get_word(buf1, sizeof(buf1), &p);
356 payload_type = atoi(buf1);
357 for (i = 0; i < s->nb_streams; i++) {
359 rtsp_st = st->priv_data;
360 if (rtsp_st->sdp_payload_type == payload_type &&
361 rtsp_st->dynamic_handler &&
362 rtsp_st->dynamic_handler->parse_sdp_a_line)
363 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
364 rtsp_st->dynamic_protocol_context, buf);
366 } else if (av_strstart(p, "range:", &p)) {
369 // this is so that seeking on a streamed file can work.
370 rtsp_parse_range_npt(p, &start, &end);
371 s->start_time = start;
372 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
373 s->duration = (end == AV_NOPTS_VALUE) ?
374 AV_NOPTS_VALUE : end - start;
375 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
377 rt->transport = RTSP_TRANSPORT_RDT;
379 if (rt->server_type == RTSP_SERVER_WMS)
380 ff_wms_parse_sdp_a_line(s, p);
381 if (s->nb_streams > 0) {
382 if (rt->server_type == RTSP_SERVER_REAL)
383 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
385 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
386 if (rtsp_st->dynamic_handler &&
387 rtsp_st->dynamic_handler->parse_sdp_a_line)
388 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
390 rtsp_st->dynamic_protocol_context, buf);
397 static int sdp_parse(AVFormatContext *s, const char *content)
401 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
402 * contain long SDP lines containing complete ASF Headers (several
403 * kB) or arrays of MDPR (RM stream descriptor) headers plus
404 * "rulebooks" describing their properties. Therefore, the SDP line
407 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
408 * in rtpdec_xiph.c. */
410 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
412 memset(s1, 0, sizeof(SDPParseState));
415 p += strspn(p, SPACE_CHARS);
423 /* get the content */
425 while (*p != '\n' && *p != '\r' && *p != '\0') {
426 if ((q - buf) < sizeof(buf) - 1)
431 sdp_parse_line(s, s1, letter, buf);
433 while (*p != '\n' && *p != '\0')
441 /* close and free RTSP streams */
442 void ff_rtsp_close_streams(AVFormatContext *s)
444 RTSPState *rt = s->priv_data;
448 for (i = 0; i < rt->nb_rtsp_streams; i++) {
449 rtsp_st = rt->rtsp_streams[i];
451 if (rtsp_st->transport_priv) {
453 AVFormatContext *rtpctx = rtsp_st->transport_priv;
454 av_write_trailer(rtpctx);
455 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
457 url_close_dyn_buf(rtpctx->pb, &ptr);
460 url_fclose(rtpctx->pb);
462 av_metadata_free(&rtpctx->streams[0]->metadata);
463 av_metadata_free(&rtpctx->metadata);
464 av_free(rtpctx->streams[0]);
466 } else if (rt->transport == RTSP_TRANSPORT_RDT)
467 ff_rdt_parse_close(rtsp_st->transport_priv);
469 rtp_parse_close(rtsp_st->transport_priv);
471 if (rtsp_st->rtp_handle)
472 url_close(rtsp_st->rtp_handle);
473 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
474 rtsp_st->dynamic_handler->close(
475 rtsp_st->dynamic_protocol_context);
478 av_free(rt->rtsp_streams);
480 av_close_input_stream (rt->asf_ctx);
485 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
488 RTSPState *rt = s->priv_data;
489 AVFormatContext *rtpctx;
491 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
496 /* Allocate an AVFormatContext for each output stream */
497 rtpctx = avformat_alloc_context();
501 rtpctx->oformat = rtp_format;
502 if (!av_new_stream(rtpctx, 0)) {
506 /* Copy the max delay setting; the rtp muxer reads this. */
507 rtpctx->max_delay = s->max_delay;
508 /* Copy other stream parameters. */
509 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
511 /* Set the synchronized start time. */
512 rtpctx->start_time_realtime = rt->start_time;
514 /* Remove the local codec, link to the original codec
515 * context instead, to give the rtp muxer access to
516 * codec parameters. */
517 av_free(rtpctx->streams[0]->codec);
518 rtpctx->streams[0]->codec = st->codec;
521 url_fdopen(&rtpctx->pb, handle);
523 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
524 ret = av_write_header(rtpctx);
528 url_fclose(rtpctx->pb);
531 url_close_dyn_buf(rtpctx->pb, &ptr);
534 av_free(rtpctx->streams[0]);
539 /* Copy the RTP AVStream timebase back to the original AVStream */
540 st->time_base = rtpctx->streams[0]->time_base;
544 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
546 RTSPState *rt = s->priv_data;
549 /* open the RTP context */
550 if (rtsp_st->stream_index >= 0)
551 st = s->streams[rtsp_st->stream_index];
553 s->ctx_flags |= AVFMTCTX_NOHEADER;
556 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
557 /* Ownership of rtp_handle is passed to the rtp mux context */
558 rtsp_st->rtp_handle = NULL;
559 } else if (rt->transport == RTSP_TRANSPORT_RDT)
560 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
561 rtsp_st->dynamic_protocol_context,
562 rtsp_st->dynamic_handler);
564 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
565 rtsp_st->sdp_payload_type);
567 if (!rtsp_st->transport_priv) {
568 return AVERROR(ENOMEM);
569 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
570 if (rtsp_st->dynamic_handler) {
571 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
572 rtsp_st->dynamic_protocol_context,
573 rtsp_st->dynamic_handler);
580 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
581 static int rtsp_probe(AVProbeData *p)
583 if (av_strstart(p->filename, "rtsp:", NULL))
584 return AVPROBE_SCORE_MAX;
588 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
594 p += strspn(p, SPACE_CHARS);
595 v = strtol(p, (char **)&p, 10);
599 v = strtol(p, (char **)&p, 10);
608 /* XXX: only one transport specification is parsed */
609 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
611 char transport_protocol[16];
613 char lower_transport[16];
615 RTSPTransportField *th;
618 reply->nb_transports = 0;
621 p += strspn(p, SPACE_CHARS);
625 th = &reply->transports[reply->nb_transports];
627 get_word_sep(transport_protocol, sizeof(transport_protocol),
629 if (!strcasecmp (transport_protocol, "rtp")) {
630 get_word_sep(profile, sizeof(profile), "/;,", &p);
631 lower_transport[0] = '\0';
632 /* rtp/avp/<protocol> */
634 get_word_sep(lower_transport, sizeof(lower_transport),
637 th->transport = RTSP_TRANSPORT_RTP;
638 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
639 !strcasecmp (transport_protocol, "x-real-rdt")) {
640 /* x-pn-tng/<protocol> */
641 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
643 th->transport = RTSP_TRANSPORT_RDT;
645 if (!strcasecmp(lower_transport, "TCP"))
646 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
648 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
652 /* get each parameter */
653 while (*p != '\0' && *p != ',') {
654 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
655 if (!strcmp(parameter, "port")) {
658 rtsp_parse_range(&th->port_min, &th->port_max, &p);
660 } else if (!strcmp(parameter, "client_port")) {
663 rtsp_parse_range(&th->client_port_min,
664 &th->client_port_max, &p);
666 } else if (!strcmp(parameter, "server_port")) {
669 rtsp_parse_range(&th->server_port_min,
670 &th->server_port_max, &p);
672 } else if (!strcmp(parameter, "interleaved")) {
675 rtsp_parse_range(&th->interleaved_min,
676 &th->interleaved_max, &p);
678 } else if (!strcmp(parameter, "multicast")) {
679 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
680 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
681 } else if (!strcmp(parameter, "ttl")) {
684 th->ttl = strtol(p, (char **)&p, 10);
686 } else if (!strcmp(parameter, "destination")) {
687 struct in_addr ipaddr;
691 get_word_sep(buf, sizeof(buf), ";,", &p);
692 if (ff_inet_aton(buf, &ipaddr))
693 th->destination = ntohl(ipaddr.s_addr);
696 while (*p != ';' && *p != '\0' && *p != ',')
704 reply->nb_transports++;
708 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
709 HTTPAuthState *auth_state)
713 /* NOTE: we do case independent match for broken servers */
715 if (av_stristart(p, "Session:", &p)) {
717 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
718 if (av_stristart(p, ";timeout=", &p) &&
719 (t = strtol(p, NULL, 10)) > 0) {
722 } else if (av_stristart(p, "Content-Length:", &p)) {
723 reply->content_length = strtol(p, NULL, 10);
724 } else if (av_stristart(p, "Transport:", &p)) {
725 rtsp_parse_transport(reply, p);
726 } else if (av_stristart(p, "CSeq:", &p)) {
727 reply->seq = strtol(p, NULL, 10);
728 } else if (av_stristart(p, "Range:", &p)) {
729 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
730 } else if (av_stristart(p, "RealChallenge1:", &p)) {
731 p += strspn(p, SPACE_CHARS);
732 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
733 } else if (av_stristart(p, "Server:", &p)) {
734 p += strspn(p, SPACE_CHARS);
735 av_strlcpy(reply->server, p, sizeof(reply->server));
736 } else if (av_stristart(p, "Notice:", &p) ||
737 av_stristart(p, "X-Notice:", &p)) {
738 reply->notice = strtol(p, NULL, 10);
739 } else if (av_stristart(p, "Location:", &p)) {
740 p += strspn(p, SPACE_CHARS);
741 av_strlcpy(reply->location, p , sizeof(reply->location));
742 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
743 p += strspn(p, SPACE_CHARS);
744 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
745 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
746 p += strspn(p, SPACE_CHARS);
747 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
751 /* skip a RTP/TCP interleaved packet */
752 void ff_rtsp_skip_packet(AVFormatContext *s)
754 RTSPState *rt = s->priv_data;
758 ret = url_read_complete(rt->rtsp_hd, buf, 3);
761 len = AV_RB16(buf + 1);
763 dprintf(s, "skipping RTP packet len=%d\n", len);
768 if (len1 > sizeof(buf))
770 ret = url_read_complete(rt->rtsp_hd, buf, len1);
777 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
778 unsigned char **content_ptr,
779 int return_on_interleaved_data)
781 RTSPState *rt = s->priv_data;
782 char buf[4096], buf1[1024], *q;
785 int ret, content_length, line_count = 0;
786 unsigned char *content = NULL;
788 memset(reply, 0, sizeof(*reply));
790 /* parse reply (XXX: use buffers) */
791 rt->last_reply[0] = '\0';
795 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
797 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
804 /* XXX: only parse it if first char on line ? */
805 if (return_on_interleaved_data) {
808 ff_rtsp_skip_packet(s);
809 } else if (ch != '\r') {
810 if ((q - buf) < sizeof(buf) - 1)
816 dprintf(s, "line='%s'\n", buf);
818 /* test if last line */
822 if (line_count == 0) {
824 get_word(buf1, sizeof(buf1), &p);
825 get_word(buf1, sizeof(buf1), &p);
826 reply->status_code = atoi(buf1);
827 av_strlcpy(reply->reason, p, sizeof(reply->reason));
829 ff_rtsp_parse_line(reply, p, &rt->auth_state);
830 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
831 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
836 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
837 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
839 content_length = reply->content_length;
840 if (content_length > 0) {
841 /* leave some room for a trailing '\0' (useful for simple parsing) */
842 content = av_malloc(content_length + 1);
843 (void)url_read_complete(rt->rtsp_hd, content, content_length);
844 content[content_length] = '\0';
847 *content_ptr = content;
851 if (rt->seq != reply->seq) {
852 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
853 rt->seq, reply->seq);
857 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
858 reply->notice == 2104 /* Start-of-Stream Reached */ ||
859 reply->notice == 2306 /* Continuous Feed Terminated */) {
860 rt->state = RTSP_STATE_IDLE;
861 } else if (reply->notice >= 4400 && reply->notice < 5500) {
862 return AVERROR(EIO); /* data or server error */
863 } else if (reply->notice == 2401 /* Ticket Expired */ ||
864 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
865 return AVERROR(EPERM);
870 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
871 const char *method, const char *url,
873 const unsigned char *send_content,
874 int send_content_length)
876 RTSPState *rt = s->priv_data;
877 char buf[4096], *out_buf;
878 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
880 /* Add in RTSP headers */
883 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
885 av_strlcat(buf, headers, sizeof(buf));
886 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
887 if (rt->session_id[0] != '\0' && (!headers ||
888 !strstr(headers, "\nIf-Match:"))) {
889 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
892 char *str = ff_http_auth_create_response(&rt->auth_state,
893 rt->auth, url, method);
895 av_strlcat(buf, str, sizeof(buf));
898 if (send_content_length > 0 && send_content)
899 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
900 av_strlcat(buf, "\r\n", sizeof(buf));
902 /* base64 encode rtsp if tunneling */
903 if (rt->control_transport == RTSP_MODE_TUNNEL) {
904 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
908 dprintf(s, "Sending:\n%s--\n", buf);
910 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
911 if (send_content_length > 0 && send_content) {
912 if (rt->control_transport == RTSP_MODE_TUNNEL) {
913 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
914 "with content data not supported\n");
915 return AVERROR_PATCHWELCOME;
917 url_write(rt->rtsp_hd_out, send_content, send_content_length);
919 rt->last_cmd_time = av_gettime();
924 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
925 const char *url, const char *headers)
927 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
930 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
931 const char *headers, RTSPMessageHeader *reply,
932 unsigned char **content_ptr)
934 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
935 content_ptr, NULL, 0);
938 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
939 const char *method, const char *url,
941 RTSPMessageHeader *reply,
942 unsigned char **content_ptr,
943 const unsigned char *send_content,
944 int send_content_length)
946 RTSPState *rt = s->priv_data;
947 HTTPAuthType cur_auth_type;
951 cur_auth_type = rt->auth_state.auth_type;
952 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
954 send_content_length)))
957 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
960 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
961 rt->auth_state.auth_type != HTTP_AUTH_NONE)
964 if (reply->status_code > 400){
965 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
969 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
976 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
978 static int make_setup_request(AVFormatContext *s, const char *host, int port,
979 int lower_transport, const char *real_challenge)
981 RTSPState *rt = s->priv_data;
982 int rtx, j, i, err, interleave = 0;
984 RTSPMessageHeader reply1, *reply = &reply1;
986 const char *trans_pref;
988 if (rt->transport == RTSP_TRANSPORT_RDT)
989 trans_pref = "x-pn-tng";
991 trans_pref = "RTP/AVP";
993 /* default timeout: 1 minute */
996 /* for each stream, make the setup request */
997 /* XXX: we assume the same server is used for the control of each
1000 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1001 char transport[2048];
1004 * WMS serves all UDP data over a single connection, the RTX, which
1005 * isn't necessarily the first in the SDP but has to be the first
1006 * to be set up, else the second/third SETUP will fail with a 461.
1008 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1009 rt->server_type == RTSP_SERVER_WMS) {
1012 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1013 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1015 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1019 if (rtx == rt->nb_rtsp_streams)
1020 return -1; /* no RTX found */
1021 rtsp_st = rt->rtsp_streams[rtx];
1023 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1025 rtsp_st = rt->rtsp_streams[i];
1028 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1031 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1032 port = reply->transports[0].client_port_min;
1036 /* first try in specified port range */
1037 if (RTSP_RTP_PORT_MIN != 0) {
1038 while (j <= RTSP_RTP_PORT_MAX) {
1039 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1040 "?localport=%d", j);
1041 /* we will use two ports per rtp stream (rtp and rtcp) */
1043 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1049 /* then try on any port */
1050 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1051 err = AVERROR_INVALIDDATA;
1057 port = rtp_get_local_port(rtsp_st->rtp_handle);
1059 snprintf(transport, sizeof(transport) - 1,
1060 "%s/UDP;", trans_pref);
1061 if (rt->server_type != RTSP_SERVER_REAL)
1062 av_strlcat(transport, "unicast;", sizeof(transport));
1063 av_strlcatf(transport, sizeof(transport),
1064 "client_port=%d", port);
1065 if (rt->transport == RTSP_TRANSPORT_RTP &&
1066 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1067 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1071 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1072 /** For WMS streams, the application streams are only used for
1073 * UDP. When trying to set it up for TCP streams, the server
1074 * will return an error. Therefore, we skip those streams. */
1075 if (rt->server_type == RTSP_SERVER_WMS &&
1076 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1079 snprintf(transport, sizeof(transport) - 1,
1080 "%s/TCP;", trans_pref);
1081 if (rt->server_type == RTSP_SERVER_WMS)
1082 av_strlcat(transport, "unicast;", sizeof(transport));
1083 av_strlcatf(transport, sizeof(transport),
1084 "interleaved=%d-%d",
1085 interleave, interleave + 1);
1089 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1090 snprintf(transport, sizeof(transport) - 1,
1091 "%s/UDP;multicast", trans_pref);
1094 av_strlcat(transport, ";mode=receive", sizeof(transport));
1095 } else if (rt->server_type == RTSP_SERVER_REAL ||
1096 rt->server_type == RTSP_SERVER_WMS)
1097 av_strlcat(transport, ";mode=play", sizeof(transport));
1098 snprintf(cmd, sizeof(cmd),
1099 "Transport: %s\r\n",
1101 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1102 char real_res[41], real_csum[9];
1103 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1105 av_strlcatf(cmd, sizeof(cmd),
1107 "RealChallenge2: %s, sd=%s\r\n",
1108 rt->session_id, real_res, real_csum);
1110 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1111 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1114 } else if (reply->status_code != RTSP_STATUS_OK ||
1115 reply->nb_transports != 1) {
1116 err = AVERROR_INVALIDDATA;
1120 /* XXX: same protocol for all streams is required */
1122 if (reply->transports[0].lower_transport != rt->lower_transport ||
1123 reply->transports[0].transport != rt->transport) {
1124 err = AVERROR_INVALIDDATA;
1128 rt->lower_transport = reply->transports[0].lower_transport;
1129 rt->transport = reply->transports[0].transport;
1132 /* close RTP connection if not choosen */
1133 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1134 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1135 url_close(rtsp_st->rtp_handle);
1136 rtsp_st->rtp_handle = NULL;
1139 switch(reply->transports[0].lower_transport) {
1140 case RTSP_LOWER_TRANSPORT_TCP:
1141 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1142 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1145 case RTSP_LOWER_TRANSPORT_UDP: {
1148 /* XXX: also use address if specified */
1149 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1150 reply->transports[0].server_port_min, NULL);
1151 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1152 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1153 err = AVERROR_INVALIDDATA;
1156 /* Try to initialize the connection state in a
1157 * potential NAT router by sending dummy packets.
1158 * RTP/RTCP dummy packets are used for RDT, too.
1160 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1161 rtp_send_punch_packets(rtsp_st->rtp_handle);
1164 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1169 if (reply->transports[0].destination) {
1170 in.s_addr = htonl(reply->transports[0].destination);
1171 port = reply->transports[0].port_min;
1172 ttl = reply->transports[0].ttl;
1174 in = rtsp_st->sdp_ip;
1175 port = rtsp_st->sdp_port;
1176 ttl = rtsp_st->sdp_ttl;
1178 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1179 port, "?ttl=%d", ttl);
1180 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1181 err = AVERROR_INVALIDDATA;
1188 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1192 if (reply->timeout > 0)
1193 rt->timeout = reply->timeout;
1195 if (rt->server_type == RTSP_SERVER_REAL)
1196 rt->need_subscription = 1;
1201 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1202 if (rt->rtsp_streams[i]->rtp_handle) {
1203 url_close(rt->rtsp_streams[i]->rtp_handle);
1204 rt->rtsp_streams[i]->rtp_handle = NULL;
1210 static int rtsp_read_play(AVFormatContext *s)
1212 RTSPState *rt = s->priv_data;
1213 RTSPMessageHeader reply1, *reply = &reply1;
1217 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1219 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1220 if (rt->state == RTSP_STATE_PAUSED) {
1223 snprintf(cmd, sizeof(cmd),
1224 "Range: npt=%0.3f-\r\n",
1225 (double)rt->seek_timestamp / AV_TIME_BASE);
1227 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1228 if (reply->status_code != RTSP_STATUS_OK) {
1231 if (reply->range_start != AV_NOPTS_VALUE &&
1232 rt->transport == RTSP_TRANSPORT_RTP) {
1233 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1234 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1235 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1236 AVStream *st = NULL;
1239 if (rtsp_st->stream_index >= 0)
1240 st = s->streams[rtsp_st->stream_index];
1241 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1242 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1244 rtpctx->range_start_offset = av_rescale_q(reply->range_start,
1250 rt->state = RTSP_STATE_STREAMING;
1254 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1256 RTSPState *rt = s->priv_data;
1258 unsigned char *content = NULL;
1261 /* describe the stream */
1262 snprintf(cmd, sizeof(cmd),
1263 "Accept: application/sdp\r\n");
1264 if (rt->server_type == RTSP_SERVER_REAL) {
1266 * The Require: attribute is needed for proper streaming from
1267 * Realmedia servers.
1270 "Require: com.real.retain-entity-for-setup\r\n",
1273 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1275 return AVERROR_INVALIDDATA;
1276 if (reply->status_code != RTSP_STATUS_OK) {
1278 return AVERROR_INVALIDDATA;
1281 /* now we got the SDP description, we parse it */
1282 ret = sdp_parse(s, (const char *)content);
1285 return AVERROR_INVALIDDATA;
1290 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1292 RTSPState *rt = s->priv_data;
1293 RTSPMessageHeader reply1, *reply = &reply1;
1296 AVFormatContext sdp_ctx, *ctx_array[1];
1298 rt->start_time = av_gettime();
1300 /* Announce the stream */
1301 sdp = av_mallocz(SDP_MAX_SIZE);
1303 return AVERROR(ENOMEM);
1304 /* We create the SDP based on the RTSP AVFormatContext where we
1305 * aren't allowed to change the filename field. (We create the SDP
1306 * based on the RTSP context since the contexts for the RTP streams
1307 * don't exist yet.) In order to specify a custom URL with the actual
1308 * peer IP instead of the originally specified hostname, we create
1309 * a temporary copy of the AVFormatContext, where the custom URL is set.
1311 * FIXME: Create the SDP without copying the AVFormatContext.
1312 * This either requires setting up the RTP stream AVFormatContexts
1313 * already here (complicating things immensely) or getting a more
1314 * flexible SDP creation interface.
1317 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1318 "rtsp", NULL, addr, -1, NULL);
1319 ctx_array[0] = &sdp_ctx;
1320 if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
1322 return AVERROR_INVALIDDATA;
1324 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1325 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1326 "Content-Type: application/sdp\r\n",
1327 reply, NULL, sdp, strlen(sdp));
1329 if (reply->status_code != RTSP_STATUS_OK)
1330 return AVERROR_INVALIDDATA;
1332 /* Set up the RTSPStreams for each AVStream */
1333 for (i = 0; i < s->nb_streams; i++) {
1334 RTSPStream *rtsp_st;
1335 AVStream *st = s->streams[i];
1337 rtsp_st = av_mallocz(sizeof(RTSPStream));
1339 return AVERROR(ENOMEM);
1340 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1342 st->priv_data = rtsp_st;
1343 rtsp_st->stream_index = i;
1345 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1346 /* Note, this must match the relative uri set in the sdp content */
1347 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1354 void ff_rtsp_close_connections(AVFormatContext *s)
1356 RTSPState *rt = s->priv_data;
1357 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1358 url_close(rt->rtsp_hd);
1359 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1362 int ff_rtsp_connect(AVFormatContext *s)
1364 RTSPState *rt = s->priv_data;
1365 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1366 char *option_list, *option, *filename;
1367 int port, err, tcp_fd;
1368 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1369 int lower_transport_mask = 0;
1370 char real_challenge[64];
1371 struct sockaddr_storage peer;
1372 socklen_t peer_len = sizeof(peer);
1374 if (!ff_network_init())
1375 return AVERROR(EIO);
1377 rt->control_transport = RTSP_MODE_PLAIN;
1378 /* extract hostname and port */
1379 av_url_split(NULL, 0, auth, sizeof(auth),
1380 host, sizeof(host), &port, path, sizeof(path), s->filename);
1382 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1385 port = RTSP_DEFAULT_PORT;
1387 /* search for options */
1388 option_list = strrchr(path, '?');
1390 /* Strip out the RTSP specific options, write out the rest of
1391 * the options back into the same string. */
1392 filename = option_list;
1393 while (option_list) {
1394 /* move the option pointer */
1395 option = ++option_list;
1396 option_list = strchr(option_list, '&');
1400 /* handle the options */
1401 if (!strcmp(option, "udp")) {
1402 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1403 } else if (!strcmp(option, "multicast")) {
1404 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1405 } else if (!strcmp(option, "tcp")) {
1406 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1407 } else if(!strcmp(option, "http")) {
1408 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1409 rt->control_transport = RTSP_MODE_TUNNEL;
1411 /* Write options back into the buffer, using memmove instead
1412 * of strcpy since the strings may overlap. */
1413 int len = strlen(option);
1414 memmove(++filename, option, len);
1416 if (option_list) *filename = '&';
1422 if (!lower_transport_mask)
1423 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1426 /* Only UDP or TCP - UDP multicast isn't supported. */
1427 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1428 (1 << RTSP_LOWER_TRANSPORT_TCP);
1429 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1430 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1431 "only UDP and TCP are supported for output.\n");
1432 err = AVERROR(EINVAL);
1437 /* Construct the URI used in request; this is similar to s->filename,
1438 * but with authentication credentials removed and RTSP specific options
1440 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1441 host, port, "%s", path);
1443 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1444 /* set up initial handshake for tunneling */
1445 char httpname[1024];
1446 char sessioncookie[17];
1449 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1450 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1451 av_get_random_seed(), av_get_random_seed());
1454 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1459 /* generate GET headers */
1460 snprintf(headers, sizeof(headers),
1461 "x-sessioncookie: %s\r\n"
1462 "Accept: application/x-rtsp-tunnelled\r\n"
1463 "Pragma: no-cache\r\n"
1464 "Cache-Control: no-cache\r\n",
1466 ff_http_set_headers(rt->rtsp_hd, headers);
1468 /* complete the connection */
1469 if (url_connect(rt->rtsp_hd)) {
1475 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1480 /* generate POST headers */
1481 snprintf(headers, sizeof(headers),
1482 "x-sessioncookie: %s\r\n"
1483 "Content-Type: application/x-rtsp-tunnelled\r\n"
1484 "Pragma: no-cache\r\n"
1485 "Cache-Control: no-cache\r\n"
1486 "Content-Length: 32767\r\n"
1487 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1489 ff_http_set_headers(rt->rtsp_hd_out, headers);
1490 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1492 /* Initialize the authentication state for the POST session. The HTTP
1493 * protocol implementation doesn't properly handle multi-pass
1494 * authentication for POST requests, since it would require one of
1496 * - implementing Expect: 100-continue, which many HTTP servers
1497 * don't support anyway, even less the RTSP servers that do HTTP
1499 * - sending the whole POST data until getting a 401 reply specifying
1500 * what authentication method to use, then resending all that data
1501 * - waiting for potential 401 replies directly after sending the
1502 * POST header (waiting for some unspecified time)
1503 * Therefore, we copy the full auth state, which works for both basic
1504 * and digest. (For digest, we would have to synchronize the nonce
1505 * count variable between the two sessions, if we'd do more requests
1506 * with the original session, though.)
1508 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1510 /* complete the connection */
1511 if (url_connect(rt->rtsp_hd_out)) {
1516 /* open the tcp connection */
1517 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1518 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1522 rt->rtsp_hd_out = rt->rtsp_hd;
1526 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1527 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1528 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1529 NULL, 0, NI_NUMERICHOST);
1532 /* request options supported by the server; this also detects server
1534 for (rt->server_type = RTSP_SERVER_RTP;;) {
1536 if (rt->server_type == RTSP_SERVER_REAL)
1539 * The following entries are required for proper
1540 * streaming from a Realmedia server. They are
1541 * interdependent in some way although we currently
1542 * don't quite understand how. Values were copied
1543 * from mplayer SVN r23589.
1544 * @param CompanyID is a 16-byte ID in base64
1545 * @param ClientChallenge is a 16-byte ID in hex
1547 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1548 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1549 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1550 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1552 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1553 if (reply->status_code != RTSP_STATUS_OK) {
1554 err = AVERROR_INVALIDDATA;
1558 /* detect server type if not standard-compliant RTP */
1559 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1560 rt->server_type = RTSP_SERVER_REAL;
1562 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1563 rt->server_type = RTSP_SERVER_WMS;
1564 } else if (rt->server_type == RTSP_SERVER_REAL)
1565 strcpy(real_challenge, reply->real_challenge);
1570 err = rtsp_setup_input_streams(s, reply);
1572 err = rtsp_setup_output_streams(s, host);
1577 int lower_transport = ff_log2_tab[lower_transport_mask &
1578 ~(lower_transport_mask - 1)];
1580 err = make_setup_request(s, host, port, lower_transport,
1581 rt->server_type == RTSP_SERVER_REAL ?
1582 real_challenge : NULL);
1585 lower_transport_mask &= ~(1 << lower_transport);
1586 if (lower_transport_mask == 0 && err == 1) {
1587 err = FF_NETERROR(EPROTONOSUPPORT);
1592 rt->state = RTSP_STATE_IDLE;
1593 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1596 ff_rtsp_close_streams(s);
1597 ff_rtsp_close_connections(s);
1598 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1599 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1600 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1610 #if CONFIG_RTSP_DEMUXER
1611 static int rtsp_read_header(AVFormatContext *s,
1612 AVFormatParameters *ap)
1616 ret = ff_rtsp_connect(s);
1620 if (ap->initial_pause) {
1621 /* do not start immediately */
1623 if (rtsp_read_play(s) < 0) {
1624 ff_rtsp_close_streams(s);
1625 ff_rtsp_close_connections(s);
1626 return AVERROR_INVALIDDATA;
1633 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1634 uint8_t *buf, int buf_size)
1636 RTSPState *rt = s->priv_data;
1637 RTSPStream *rtsp_st;
1639 int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1643 if (url_interrupt_cb())
1644 return AVERROR(EINTR);
1647 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1648 FD_SET(tcp_fd, &rfds);
1653 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1654 rtsp_st = rt->rtsp_streams[i];
1655 if (rtsp_st->rtp_handle) {
1656 /* currently, we cannot probe RTCP handle because of
1657 * blocking restrictions */
1658 fd = url_get_file_handle(rtsp_st->rtp_handle);
1665 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1666 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1669 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1670 rtsp_st = rt->rtsp_streams[i];
1671 if (rtsp_st->rtp_handle) {
1672 fd = url_get_file_handle(rtsp_st->rtp_handle);
1673 if (FD_ISSET(fd, &rfds)) {
1674 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1676 *prtsp_st = rtsp_st;
1682 #if CONFIG_RTSP_DEMUXER
1683 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1684 RTSPMessageHeader reply;
1686 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1689 /* XXX: parse message */
1690 if (rt->state != RTSP_STATE_STREAMING)
1694 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1695 return FF_NETERROR(ETIMEDOUT);
1696 } else if (n < 0 && errno != EINTR)
1697 return AVERROR(errno);
1701 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1702 uint8_t *buf, int buf_size)
1704 RTSPState *rt = s->priv_data;
1705 int id, len, i, ret;
1706 RTSPStream *rtsp_st;
1708 #ifdef DEBUG_RTP_TCP
1709 dprintf(s, "tcp_read_packet:\n");
1713 RTSPMessageHeader reply;
1715 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1718 if (ret == 1) /* received '$' */
1720 /* XXX: parse message */
1721 if (rt->state != RTSP_STATE_STREAMING)
1724 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1728 len = AV_RB16(buf + 1);
1729 #ifdef DEBUG_RTP_TCP
1730 dprintf(s, "id=%d len=%d\n", id, len);
1732 if (len > buf_size || len < 12)
1735 ret = url_read_complete(rt->rtsp_hd, buf, len);
1738 if (rt->transport == RTSP_TRANSPORT_RDT &&
1739 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1742 /* find the matching stream */
1743 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1744 rtsp_st = rt->rtsp_streams[i];
1745 if (id >= rtsp_st->interleaved_min &&
1746 id <= rtsp_st->interleaved_max)
1751 *prtsp_st = rtsp_st;
1755 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1757 RTSPState *rt = s->priv_data;
1759 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1760 RTSPStream *rtsp_st;
1762 /* get next frames from the same RTP packet */
1763 if (rt->cur_transport_priv) {
1764 if (rt->transport == RTSP_TRANSPORT_RDT) {
1765 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1767 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1769 rt->cur_transport_priv = NULL;
1771 } else if (ret == 1) {
1774 rt->cur_transport_priv = NULL;
1777 /* read next RTP packet */
1779 switch(rt->lower_transport) {
1781 #if CONFIG_RTSP_DEMUXER
1782 case RTSP_LOWER_TRANSPORT_TCP:
1783 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1786 case RTSP_LOWER_TRANSPORT_UDP:
1787 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1788 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1789 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1790 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1797 if (rt->transport == RTSP_TRANSPORT_RDT) {
1798 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1800 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1802 /* Either bad packet, or a RTCP packet. Check if the
1803 * first_rtcp_ntp_time field was initialized. */
1804 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1805 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1806 /* first_rtcp_ntp_time has been initialized for this stream,
1807 * copy the same value to all other uninitialized streams,
1808 * in order to map their timestamp origin to the same ntp time
1811 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1812 RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
1814 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1815 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1823 /* more packets may follow, so we save the RTP context */
1824 rt->cur_transport_priv = rtsp_st->transport_priv;
1829 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1831 RTSPState *rt = s->priv_data;
1833 RTSPMessageHeader reply1, *reply = &reply1;
1836 if (rt->server_type == RTSP_SERVER_REAL) {
1838 enum AVDiscard cache[MAX_STREAMS];
1840 for (i = 0; i < s->nb_streams; i++)
1841 cache[i] = s->streams[i]->discard;
1843 if (!rt->need_subscription) {
1844 if (memcmp (cache, rt->real_setup_cache,
1845 sizeof(enum AVDiscard) * s->nb_streams)) {
1846 snprintf(cmd, sizeof(cmd),
1847 "Unsubscribe: %s\r\n",
1848 rt->last_subscription);
1849 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1851 if (reply->status_code != RTSP_STATUS_OK)
1852 return AVERROR_INVALIDDATA;
1853 rt->need_subscription = 1;
1857 if (rt->need_subscription) {
1858 int r, rule_nr, first = 1;
1860 memcpy(rt->real_setup_cache, cache,
1861 sizeof(enum AVDiscard) * s->nb_streams);
1862 rt->last_subscription[0] = 0;
1864 snprintf(cmd, sizeof(cmd),
1866 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1868 for (r = 0; r < s->nb_streams; r++) {
1869 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1870 if (s->streams[r]->discard != AVDISCARD_ALL) {
1872 av_strlcat(rt->last_subscription, ",",
1873 sizeof(rt->last_subscription));
1874 ff_rdt_subscribe_rule(
1875 rt->last_subscription,
1876 sizeof(rt->last_subscription), i, rule_nr);
1883 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1884 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1886 if (reply->status_code != RTSP_STATUS_OK)
1887 return AVERROR_INVALIDDATA;
1888 rt->need_subscription = 0;
1890 if (rt->state == RTSP_STATE_STREAMING)
1895 ret = rtsp_fetch_packet(s, pkt);
1899 /* send dummy request to keep TCP connection alive */
1900 if ((rt->server_type == RTSP_SERVER_WMS ||
1901 rt->server_type == RTSP_SERVER_REAL) &&
1902 (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1903 if (rt->server_type == RTSP_SERVER_WMS) {
1904 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1906 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1913 /* pause the stream */
1914 static int rtsp_read_pause(AVFormatContext *s)
1916 RTSPState *rt = s->priv_data;
1917 RTSPMessageHeader reply1, *reply = &reply1;
1919 if (rt->state != RTSP_STATE_STREAMING)
1921 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1922 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1923 if (reply->status_code != RTSP_STATUS_OK) {
1927 rt->state = RTSP_STATE_PAUSED;
1931 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1932 int64_t timestamp, int flags)
1934 RTSPState *rt = s->priv_data;
1936 rt->seek_timestamp = av_rescale_q(timestamp,
1937 s->streams[stream_index]->time_base,
1941 case RTSP_STATE_IDLE:
1943 case RTSP_STATE_STREAMING:
1944 if (rtsp_read_pause(s) != 0)
1946 rt->state = RTSP_STATE_SEEKING;
1947 if (rtsp_read_play(s) != 0)
1950 case RTSP_STATE_PAUSED:
1951 rt->state = RTSP_STATE_IDLE;
1957 static int rtsp_read_close(AVFormatContext *s)
1959 RTSPState *rt = s->priv_data;
1962 /* NOTE: it is valid to flush the buffer here */
1963 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1964 url_fclose(&rt->rtsp_gb);
1967 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1969 ff_rtsp_close_streams(s);
1970 ff_rtsp_close_connections(s);
1975 AVInputFormat rtsp_demuxer = {
1977 NULL_IF_CONFIG_SMALL("RTSP input format"),
1984 .flags = AVFMT_NOFILE,
1985 .read_play = rtsp_read_play,
1986 .read_pause = rtsp_read_pause,
1990 static int sdp_probe(AVProbeData *p1)
1992 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1994 /* we look for a line beginning "c=IN IP4" */
1995 while (p < p_end && *p != '\0') {
1996 if (p + sizeof("c=IN IP4") - 1 < p_end &&
1997 av_strstart(p, "c=IN IP4", NULL))
1998 return AVPROBE_SCORE_MAX / 2;
2000 while (p < p_end - 1 && *p != '\n') p++;
2009 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2011 RTSPState *rt = s->priv_data;
2012 RTSPStream *rtsp_st;
2017 if (!ff_network_init())
2018 return AVERROR(EIO);
2020 /* read the whole sdp file */
2021 /* XXX: better loading */
2022 content = av_malloc(SDP_MAX_SIZE);
2023 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2026 return AVERROR_INVALIDDATA;
2028 content[size] ='\0';
2030 sdp_parse(s, content);
2033 /* open each RTP stream */
2034 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2035 rtsp_st = rt->rtsp_streams[i];
2037 ff_url_join(url, sizeof(url), "rtp", NULL,
2038 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2039 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2041 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2042 err = AVERROR_INVALIDDATA;
2045 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2050 ff_rtsp_close_streams(s);
2055 static int sdp_read_close(AVFormatContext *s)
2057 ff_rtsp_close_streams(s);
2062 AVInputFormat sdp_demuxer = {
2064 NULL_IF_CONFIG_SMALL("SDP"),