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"
42 #include "rtpenc_chain.h"
45 //#define DEBUG_RTP_TCP
47 /* Timeout values for socket select, in ms,
48 * and read_packet(), in seconds */
49 #define SELECT_TIMEOUT_MS 100
50 #define READ_PACKET_TIMEOUT_S 10
51 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
52 #define SDP_MAX_SIZE 16384
53 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
55 static void get_word_until_chars(char *buf, int buf_size,
56 const char *sep, const char **pp)
62 p += strspn(p, SPACE_CHARS);
64 while (!strchr(sep, *p) && *p != '\0') {
65 if ((q - buf) < buf_size - 1)
74 static void get_word_sep(char *buf, int buf_size, const char *sep,
77 if (**pp == '/') (*pp)++;
78 get_word_until_chars(buf, buf_size, sep, pp);
81 static void get_word(char *buf, int buf_size, const char **pp)
83 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
86 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
88 * Used for seeking in the rtp stream.
90 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
94 p += strspn(p, SPACE_CHARS);
95 if (!av_stristart(p, "npt=", &p))
98 *start = AV_NOPTS_VALUE;
99 *end = AV_NOPTS_VALUE;
101 get_word_sep(buf, sizeof(buf), "-", &p);
102 *start = parse_date(buf, 1);
105 get_word_sep(buf, sizeof(buf), "-", &p);
106 *end = parse_date(buf, 1);
108 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
109 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
112 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
114 struct addrinfo hints, *ai = NULL;
115 memset(&hints, 0, sizeof(hints));
116 hints.ai_flags = AI_NUMERICHOST;
117 if (getaddrinfo(buf, NULL, &hints, &ai))
119 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
125 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
126 RTSPStream *rtsp_st, AVCodecContext *codec)
130 codec->codec_id = handler->codec_id;
131 rtsp_st->dynamic_handler = handler;
133 rtsp_st->dynamic_protocol_context = handler->open();
136 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
137 static int sdp_parse_rtpmap(AVFormatContext *s,
138 AVStream *st, RTSPStream *rtsp_st,
139 int payload_type, const char *p)
141 AVCodecContext *codec = st->codec;
147 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
148 * see if we can handle this kind of payload.
149 * The space should normally not be there but some Real streams or
150 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
151 * have a trailing space. */
152 get_word_sep(buf, sizeof(buf), "/ ", &p);
153 if (payload_type >= RTP_PT_PRIVATE) {
154 RTPDynamicProtocolHandler *handler =
155 ff_rtp_handler_find_by_name(buf, codec->codec_type);
156 init_rtp_handler(handler, rtsp_st, codec);
157 /* If no dynamic handler was found, check with the list of standard
158 * allocated types, if such a stream for some reason happens to
159 * use a private payload type. This isn't handled in rtpdec.c, since
160 * the format name from the rtpmap line never is passed into rtpdec. */
161 if (!rtsp_st->dynamic_handler)
162 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
164 /* We are in a standard case
165 * (from http://www.iana.org/assignments/rtp-parameters). */
166 /* search into AVRtpPayloadTypes[] */
167 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
170 c = avcodec_find_decoder(codec->codec_id);
176 get_word_sep(buf, sizeof(buf), "/", &p);
178 switch (codec->codec_type) {
179 case AVMEDIA_TYPE_AUDIO:
180 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
181 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
182 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
184 codec->sample_rate = i;
185 av_set_pts_info(st, 32, 1, codec->sample_rate);
186 get_word_sep(buf, sizeof(buf), "/", &p);
190 // TODO: there is a bug here; if it is a mono stream, and
191 // less than 22000Hz, faad upconverts to stereo and twice
192 // the frequency. No problem, but the sample rate is being
193 // set here by the sdp line. Patch on its way. (rdm)
195 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
197 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
200 case AVMEDIA_TYPE_VIDEO:
201 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
203 av_set_pts_info(st, 32, 1, i);
211 /* parse the attribute line from the fmtp a line of an sdp response. This
212 * is broken out as a function because it is used in rtp_h264.c, which is
214 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
215 char *value, int value_size)
217 *p += strspn(*p, SPACE_CHARS);
219 get_word_sep(attr, attr_size, "=", p);
222 get_word_sep(value, value_size, ";", p);
230 typedef struct SDPParseState {
232 struct sockaddr_storage default_ip;
234 int skip_media; ///< set if an unknown m= line occurs
237 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
238 int letter, const char *buf)
240 RTSPState *rt = s->priv_data;
241 char buf1[64], st_type[64];
243 enum AVMediaType codec_type;
247 struct sockaddr_storage sdp_ip;
250 dprintf(s, "sdp: %c='%s'\n", letter, buf);
253 if (s1->skip_media && letter != 'm')
257 get_word(buf1, sizeof(buf1), &p);
258 if (strcmp(buf1, "IN") != 0)
260 get_word(buf1, sizeof(buf1), &p);
261 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
263 get_word_sep(buf1, sizeof(buf1), "/", &p);
264 if (get_sockaddr(buf1, &sdp_ip))
269 get_word_sep(buf1, sizeof(buf1), "/", &p);
272 if (s->nb_streams == 0) {
273 s1->default_ip = sdp_ip;
274 s1->default_ttl = ttl;
276 st = s->streams[s->nb_streams - 1];
277 rtsp_st = st->priv_data;
278 rtsp_st->sdp_ip = sdp_ip;
279 rtsp_st->sdp_ttl = ttl;
283 av_metadata_set2(&s->metadata, "title", p, 0);
286 if (s->nb_streams == 0) {
287 av_metadata_set2(&s->metadata, "comment", p, 0);
294 get_word(st_type, sizeof(st_type), &p);
295 if (!strcmp(st_type, "audio")) {
296 codec_type = AVMEDIA_TYPE_AUDIO;
297 } else if (!strcmp(st_type, "video")) {
298 codec_type = AVMEDIA_TYPE_VIDEO;
299 } else if (!strcmp(st_type, "application")) {
300 codec_type = AVMEDIA_TYPE_DATA;
305 rtsp_st = av_mallocz(sizeof(RTSPStream));
308 rtsp_st->stream_index = -1;
309 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
311 rtsp_st->sdp_ip = s1->default_ip;
312 rtsp_st->sdp_ttl = s1->default_ttl;
314 get_word(buf1, sizeof(buf1), &p); /* port */
315 rtsp_st->sdp_port = atoi(buf1);
317 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
319 /* XXX: handle list of formats */
320 get_word(buf1, sizeof(buf1), &p); /* format list */
321 rtsp_st->sdp_payload_type = atoi(buf1);
323 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
324 /* no corresponding stream */
326 st = av_new_stream(s, 0);
329 st->priv_data = rtsp_st;
330 rtsp_st->stream_index = st->index;
331 st->codec->codec_type = codec_type;
332 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
333 RTPDynamicProtocolHandler *handler;
334 /* if standard payload type, we can find the codec right now */
335 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
336 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
337 st->codec->sample_rate > 0)
338 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
339 /* Even static payload types may need a custom depacketizer */
340 handler = ff_rtp_handler_find_by_id(
341 rtsp_st->sdp_payload_type, st->codec->codec_type);
342 init_rtp_handler(handler, rtsp_st, st->codec);
345 /* put a default control url */
346 av_strlcpy(rtsp_st->control_url, rt->control_uri,
347 sizeof(rtsp_st->control_url));
350 if (av_strstart(p, "control:", &p)) {
351 if (s->nb_streams == 0) {
352 if (!strncmp(p, "rtsp://", 7))
353 av_strlcpy(rt->control_uri, p,
354 sizeof(rt->control_uri));
357 /* get the control url */
358 st = s->streams[s->nb_streams - 1];
359 rtsp_st = st->priv_data;
361 /* XXX: may need to add full url resolution */
362 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
364 if (proto[0] == '\0') {
365 /* relative control URL */
366 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
367 av_strlcat(rtsp_st->control_url, "/",
368 sizeof(rtsp_st->control_url));
369 av_strlcat(rtsp_st->control_url, p,
370 sizeof(rtsp_st->control_url));
372 av_strlcpy(rtsp_st->control_url, p,
373 sizeof(rtsp_st->control_url));
375 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
376 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
377 get_word(buf1, sizeof(buf1), &p);
378 payload_type = atoi(buf1);
379 st = s->streams[s->nb_streams - 1];
380 rtsp_st = st->priv_data;
381 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
382 } else if (av_strstart(p, "fmtp:", &p) ||
383 av_strstart(p, "framesize:", &p)) {
384 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
385 // let dynamic protocol handlers have a stab at the line.
386 get_word(buf1, sizeof(buf1), &p);
387 payload_type = atoi(buf1);
388 for (i = 0; i < s->nb_streams; i++) {
390 rtsp_st = st->priv_data;
391 if (rtsp_st->sdp_payload_type == payload_type &&
392 rtsp_st->dynamic_handler &&
393 rtsp_st->dynamic_handler->parse_sdp_a_line)
394 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
395 rtsp_st->dynamic_protocol_context, buf);
397 } else if (av_strstart(p, "range:", &p)) {
400 // this is so that seeking on a streamed file can work.
401 rtsp_parse_range_npt(p, &start, &end);
402 s->start_time = start;
403 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
404 s->duration = (end == AV_NOPTS_VALUE) ?
405 AV_NOPTS_VALUE : end - start;
406 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
408 rt->transport = RTSP_TRANSPORT_RDT;
409 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
411 st = s->streams[s->nb_streams - 1];
412 st->codec->sample_rate = atoi(p);
414 if (rt->server_type == RTSP_SERVER_WMS)
415 ff_wms_parse_sdp_a_line(s, p);
416 if (s->nb_streams > 0) {
417 if (rt->server_type == RTSP_SERVER_REAL)
418 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
420 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
421 if (rtsp_st->dynamic_handler &&
422 rtsp_st->dynamic_handler->parse_sdp_a_line)
423 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
425 rtsp_st->dynamic_protocol_context, buf);
432 int ff_sdp_parse(AVFormatContext *s, const char *content)
436 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
437 * contain long SDP lines containing complete ASF Headers (several
438 * kB) or arrays of MDPR (RM stream descriptor) headers plus
439 * "rulebooks" describing their properties. Therefore, the SDP line
442 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
443 * in rtpdec_xiph.c. */
445 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
447 memset(s1, 0, sizeof(SDPParseState));
450 p += strspn(p, SPACE_CHARS);
458 /* get the content */
460 while (*p != '\n' && *p != '\r' && *p != '\0') {
461 if ((q - buf) < sizeof(buf) - 1)
466 sdp_parse_line(s, s1, letter, buf);
468 while (*p != '\n' && *p != '\0')
475 #endif /* CONFIG_RTPDEC */
477 void ff_rtsp_undo_setup(AVFormatContext *s)
479 RTSPState *rt = s->priv_data;
482 for (i = 0; i < rt->nb_rtsp_streams; i++) {
483 RTSPStream *rtsp_st = rt->rtsp_streams[i];
486 if (rtsp_st->transport_priv) {
488 AVFormatContext *rtpctx = rtsp_st->transport_priv;
489 av_write_trailer(rtpctx);
490 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
492 url_close_dyn_buf(rtpctx->pb, &ptr);
495 url_fclose(rtpctx->pb);
497 av_metadata_free(&rtpctx->streams[0]->metadata);
498 av_metadata_free(&rtpctx->metadata);
499 av_free(rtpctx->streams[0]);
501 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
502 ff_rdt_parse_close(rtsp_st->transport_priv);
503 else if (CONFIG_RTPDEC)
504 rtp_parse_close(rtsp_st->transport_priv);
506 rtsp_st->transport_priv = NULL;
507 if (rtsp_st->rtp_handle)
508 url_close(rtsp_st->rtp_handle);
509 rtsp_st->rtp_handle = NULL;
513 /* close and free RTSP streams */
514 void ff_rtsp_close_streams(AVFormatContext *s)
516 RTSPState *rt = s->priv_data;
520 ff_rtsp_undo_setup(s);
521 for (i = 0; i < rt->nb_rtsp_streams; i++) {
522 rtsp_st = rt->rtsp_streams[i];
524 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
525 rtsp_st->dynamic_handler->close(
526 rtsp_st->dynamic_protocol_context);
529 av_free(rt->rtsp_streams);
531 av_close_input_stream (rt->asf_ctx);
534 av_free(rt->recvbuf);
537 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
539 RTSPState *rt = s->priv_data;
542 /* open the RTP context */
543 if (rtsp_st->stream_index >= 0)
544 st = s->streams[rtsp_st->stream_index];
546 s->ctx_flags |= AVFMTCTX_NOHEADER;
548 if (s->oformat && CONFIG_RTSP_MUXER) {
549 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
551 RTSP_TCP_MAX_PACKET_SIZE);
552 /* Ownership of rtp_handle is passed to the rtp mux context */
553 rtsp_st->rtp_handle = NULL;
554 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
555 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
556 rtsp_st->dynamic_protocol_context,
557 rtsp_st->dynamic_handler);
558 else if (CONFIG_RTPDEC)
559 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
560 rtsp_st->sdp_payload_type,
561 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
562 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
564 if (!rtsp_st->transport_priv) {
565 return AVERROR(ENOMEM);
566 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
567 if (rtsp_st->dynamic_handler) {
568 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
569 rtsp_st->dynamic_protocol_context,
570 rtsp_st->dynamic_handler);
577 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
578 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
584 p += strspn(p, SPACE_CHARS);
585 v = strtol(p, (char **)&p, 10);
589 v = strtol(p, (char **)&p, 10);
598 /* XXX: only one transport specification is parsed */
599 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
601 char transport_protocol[16];
603 char lower_transport[16];
605 RTSPTransportField *th;
608 reply->nb_transports = 0;
611 p += strspn(p, SPACE_CHARS);
615 th = &reply->transports[reply->nb_transports];
617 get_word_sep(transport_protocol, sizeof(transport_protocol),
619 if (!strcasecmp (transport_protocol, "rtp")) {
620 get_word_sep(profile, sizeof(profile), "/;,", &p);
621 lower_transport[0] = '\0';
622 /* rtp/avp/<protocol> */
624 get_word_sep(lower_transport, sizeof(lower_transport),
627 th->transport = RTSP_TRANSPORT_RTP;
628 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
629 !strcasecmp (transport_protocol, "x-real-rdt")) {
630 /* x-pn-tng/<protocol> */
631 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
633 th->transport = RTSP_TRANSPORT_RDT;
635 if (!strcasecmp(lower_transport, "TCP"))
636 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
638 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
642 /* get each parameter */
643 while (*p != '\0' && *p != ',') {
644 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
645 if (!strcmp(parameter, "port")) {
648 rtsp_parse_range(&th->port_min, &th->port_max, &p);
650 } else if (!strcmp(parameter, "client_port")) {
653 rtsp_parse_range(&th->client_port_min,
654 &th->client_port_max, &p);
656 } else if (!strcmp(parameter, "server_port")) {
659 rtsp_parse_range(&th->server_port_min,
660 &th->server_port_max, &p);
662 } else if (!strcmp(parameter, "interleaved")) {
665 rtsp_parse_range(&th->interleaved_min,
666 &th->interleaved_max, &p);
668 } else if (!strcmp(parameter, "multicast")) {
669 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
670 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
671 } else if (!strcmp(parameter, "ttl")) {
674 th->ttl = strtol(p, (char **)&p, 10);
676 } else if (!strcmp(parameter, "destination")) {
679 get_word_sep(buf, sizeof(buf), ";,", &p);
680 get_sockaddr(buf, &th->destination);
682 } else if (!strcmp(parameter, "source")) {
685 get_word_sep(buf, sizeof(buf), ";,", &p);
686 av_strlcpy(th->source, buf, sizeof(th->source));
690 while (*p != ';' && *p != '\0' && *p != ',')
698 reply->nb_transports++;
702 static void handle_rtp_info(RTSPState *rt, const char *url,
703 uint32_t seq, uint32_t rtptime)
706 if (!rtptime || !url[0])
708 if (rt->transport != RTSP_TRANSPORT_RTP)
710 for (i = 0; i < rt->nb_rtsp_streams; i++) {
711 RTSPStream *rtsp_st = rt->rtsp_streams[i];
712 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
715 if (!strcmp(rtsp_st->control_url, url)) {
716 rtpctx->base_timestamp = rtptime;
722 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
725 char key[20], value[1024], url[1024] = "";
726 uint32_t seq = 0, rtptime = 0;
729 p += strspn(p, SPACE_CHARS);
732 get_word_sep(key, sizeof(key), "=", &p);
736 get_word_sep(value, sizeof(value), ";, ", &p);
738 if (!strcmp(key, "url"))
739 av_strlcpy(url, value, sizeof(url));
740 else if (!strcmp(key, "seq"))
741 seq = strtol(value, NULL, 10);
742 else if (!strcmp(key, "rtptime"))
743 rtptime = strtol(value, NULL, 10);
745 handle_rtp_info(rt, url, seq, rtptime);
754 handle_rtp_info(rt, url, seq, rtptime);
757 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
758 RTSPState *rt, const char *method)
762 /* NOTE: we do case independent match for broken servers */
764 if (av_stristart(p, "Session:", &p)) {
766 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
767 if (av_stristart(p, ";timeout=", &p) &&
768 (t = strtol(p, NULL, 10)) > 0) {
771 } else if (av_stristart(p, "Content-Length:", &p)) {
772 reply->content_length = strtol(p, NULL, 10);
773 } else if (av_stristart(p, "Transport:", &p)) {
774 rtsp_parse_transport(reply, p);
775 } else if (av_stristart(p, "CSeq:", &p)) {
776 reply->seq = strtol(p, NULL, 10);
777 } else if (av_stristart(p, "Range:", &p)) {
778 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
779 } else if (av_stristart(p, "RealChallenge1:", &p)) {
780 p += strspn(p, SPACE_CHARS);
781 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
782 } else if (av_stristart(p, "Server:", &p)) {
783 p += strspn(p, SPACE_CHARS);
784 av_strlcpy(reply->server, p, sizeof(reply->server));
785 } else if (av_stristart(p, "Notice:", &p) ||
786 av_stristart(p, "X-Notice:", &p)) {
787 reply->notice = strtol(p, NULL, 10);
788 } else if (av_stristart(p, "Location:", &p)) {
789 p += strspn(p, SPACE_CHARS);
790 av_strlcpy(reply->location, p , sizeof(reply->location));
791 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
792 p += strspn(p, SPACE_CHARS);
793 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
794 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
795 p += strspn(p, SPACE_CHARS);
796 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
797 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
798 p += strspn(p, SPACE_CHARS);
799 if (method && !strcmp(method, "DESCRIBE"))
800 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
801 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
802 p += strspn(p, SPACE_CHARS);
803 if (method && !strcmp(method, "PLAY"))
804 rtsp_parse_rtp_info(rt, p);
808 /* skip a RTP/TCP interleaved packet */
809 void ff_rtsp_skip_packet(AVFormatContext *s)
811 RTSPState *rt = s->priv_data;
815 ret = url_read_complete(rt->rtsp_hd, buf, 3);
818 len = AV_RB16(buf + 1);
820 dprintf(s, "skipping RTP packet len=%d\n", len);
825 if (len1 > sizeof(buf))
827 ret = url_read_complete(rt->rtsp_hd, buf, len1);
834 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
835 unsigned char **content_ptr,
836 int return_on_interleaved_data, const char *method)
838 RTSPState *rt = s->priv_data;
839 char buf[4096], buf1[1024], *q;
842 int ret, content_length, line_count = 0;
843 unsigned char *content = NULL;
845 memset(reply, 0, sizeof(*reply));
847 /* parse reply (XXX: use buffers) */
848 rt->last_reply[0] = '\0';
852 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
854 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
861 /* XXX: only parse it if first char on line ? */
862 if (return_on_interleaved_data) {
865 ff_rtsp_skip_packet(s);
866 } else if (ch != '\r') {
867 if ((q - buf) < sizeof(buf) - 1)
873 dprintf(s, "line='%s'\n", buf);
875 /* test if last line */
879 if (line_count == 0) {
881 get_word(buf1, sizeof(buf1), &p);
882 get_word(buf1, sizeof(buf1), &p);
883 reply->status_code = atoi(buf1);
884 av_strlcpy(reply->reason, p, sizeof(reply->reason));
886 ff_rtsp_parse_line(reply, p, rt, method);
887 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
888 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
893 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
894 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
896 content_length = reply->content_length;
897 if (content_length > 0) {
898 /* leave some room for a trailing '\0' (useful for simple parsing) */
899 content = av_malloc(content_length + 1);
900 (void)url_read_complete(rt->rtsp_hd, content, content_length);
901 content[content_length] = '\0';
904 *content_ptr = content;
908 if (rt->seq != reply->seq) {
909 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
910 rt->seq, reply->seq);
914 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
915 reply->notice == 2104 /* Start-of-Stream Reached */ ||
916 reply->notice == 2306 /* Continuous Feed Terminated */) {
917 rt->state = RTSP_STATE_IDLE;
918 } else if (reply->notice >= 4400 && reply->notice < 5500) {
919 return AVERROR(EIO); /* data or server error */
920 } else if (reply->notice == 2401 /* Ticket Expired */ ||
921 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
922 return AVERROR(EPERM);
927 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
928 const char *method, const char *url,
930 const unsigned char *send_content,
931 int send_content_length)
933 RTSPState *rt = s->priv_data;
934 char buf[4096], *out_buf;
935 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
937 /* Add in RTSP headers */
940 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
942 av_strlcat(buf, headers, sizeof(buf));
943 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
944 if (rt->session_id[0] != '\0' && (!headers ||
945 !strstr(headers, "\nIf-Match:"))) {
946 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
949 char *str = ff_http_auth_create_response(&rt->auth_state,
950 rt->auth, url, method);
952 av_strlcat(buf, str, sizeof(buf));
955 if (send_content_length > 0 && send_content)
956 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
957 av_strlcat(buf, "\r\n", sizeof(buf));
959 /* base64 encode rtsp if tunneling */
960 if (rt->control_transport == RTSP_MODE_TUNNEL) {
961 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
965 dprintf(s, "Sending:\n%s--\n", buf);
967 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
968 if (send_content_length > 0 && send_content) {
969 if (rt->control_transport == RTSP_MODE_TUNNEL) {
970 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
971 "with content data not supported\n");
972 return AVERROR_PATCHWELCOME;
974 url_write(rt->rtsp_hd_out, send_content, send_content_length);
976 rt->last_cmd_time = av_gettime();
981 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
982 const char *url, const char *headers)
984 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
987 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
988 const char *headers, RTSPMessageHeader *reply,
989 unsigned char **content_ptr)
991 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
992 content_ptr, NULL, 0);
995 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
996 const char *method, const char *url,
998 RTSPMessageHeader *reply,
999 unsigned char **content_ptr,
1000 const unsigned char *send_content,
1001 int send_content_length)
1003 RTSPState *rt = s->priv_data;
1004 HTTPAuthType cur_auth_type;
1008 cur_auth_type = rt->auth_state.auth_type;
1009 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1011 send_content_length)))
1014 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1017 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1018 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1021 if (reply->status_code > 400){
1022 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1026 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1033 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1035 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1036 int lower_transport, const char *real_challenge)
1038 RTSPState *rt = s->priv_data;
1039 int rtx, j, i, err, interleave = 0;
1040 RTSPStream *rtsp_st;
1041 RTSPMessageHeader reply1, *reply = &reply1;
1043 const char *trans_pref;
1045 if (rt->transport == RTSP_TRANSPORT_RDT)
1046 trans_pref = "x-pn-tng";
1048 trans_pref = "RTP/AVP";
1050 /* default timeout: 1 minute */
1053 /* for each stream, make the setup request */
1054 /* XXX: we assume the same server is used for the control of each
1057 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1058 char transport[2048];
1061 * WMS serves all UDP data over a single connection, the RTX, which
1062 * isn't necessarily the first in the SDP but has to be the first
1063 * to be set up, else the second/third SETUP will fail with a 461.
1065 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1066 rt->server_type == RTSP_SERVER_WMS) {
1069 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1070 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1072 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1076 if (rtx == rt->nb_rtsp_streams)
1077 return -1; /* no RTX found */
1078 rtsp_st = rt->rtsp_streams[rtx];
1080 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1082 rtsp_st = rt->rtsp_streams[i];
1085 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1088 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1089 port = reply->transports[0].client_port_min;
1093 /* first try in specified port range */
1094 if (RTSP_RTP_PORT_MIN != 0) {
1095 while (j <= RTSP_RTP_PORT_MAX) {
1096 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1097 "?localport=%d", j);
1098 /* we will use two ports per rtp stream (rtp and rtcp) */
1100 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1106 /* then try on any port */
1107 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1108 err = AVERROR_INVALIDDATA;
1112 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1118 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1120 snprintf(transport, sizeof(transport) - 1,
1121 "%s/UDP;", trans_pref);
1122 if (rt->server_type != RTSP_SERVER_REAL)
1123 av_strlcat(transport, "unicast;", sizeof(transport));
1124 av_strlcatf(transport, sizeof(transport),
1125 "client_port=%d", port);
1126 if (rt->transport == RTSP_TRANSPORT_RTP &&
1127 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1128 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1132 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1133 /** For WMS streams, the application streams are only used for
1134 * UDP. When trying to set it up for TCP streams, the server
1135 * will return an error. Therefore, we skip those streams. */
1136 if (rt->server_type == RTSP_SERVER_WMS &&
1137 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1140 snprintf(transport, sizeof(transport) - 1,
1141 "%s/TCP;", trans_pref);
1142 if (rt->server_type == RTSP_SERVER_WMS)
1143 av_strlcat(transport, "unicast;", sizeof(transport));
1144 av_strlcatf(transport, sizeof(transport),
1145 "interleaved=%d-%d",
1146 interleave, interleave + 1);
1150 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1151 snprintf(transport, sizeof(transport) - 1,
1152 "%s/UDP;multicast", trans_pref);
1155 av_strlcat(transport, ";mode=receive", sizeof(transport));
1156 } else if (rt->server_type == RTSP_SERVER_REAL ||
1157 rt->server_type == RTSP_SERVER_WMS)
1158 av_strlcat(transport, ";mode=play", sizeof(transport));
1159 snprintf(cmd, sizeof(cmd),
1160 "Transport: %s\r\n",
1162 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1163 char real_res[41], real_csum[9];
1164 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1166 av_strlcatf(cmd, sizeof(cmd),
1168 "RealChallenge2: %s, sd=%s\r\n",
1169 rt->session_id, real_res, real_csum);
1171 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1172 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1175 } else if (reply->status_code != RTSP_STATUS_OK ||
1176 reply->nb_transports != 1) {
1177 err = AVERROR_INVALIDDATA;
1181 /* XXX: same protocol for all streams is required */
1183 if (reply->transports[0].lower_transport != rt->lower_transport ||
1184 reply->transports[0].transport != rt->transport) {
1185 err = AVERROR_INVALIDDATA;
1189 rt->lower_transport = reply->transports[0].lower_transport;
1190 rt->transport = reply->transports[0].transport;
1193 /* Fail if the server responded with another lower transport mode
1194 * than what we requested. */
1195 if (reply->transports[0].lower_transport != lower_transport) {
1196 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1197 err = AVERROR_INVALIDDATA;
1201 switch(reply->transports[0].lower_transport) {
1202 case RTSP_LOWER_TRANSPORT_TCP:
1203 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1204 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1207 case RTSP_LOWER_TRANSPORT_UDP: {
1208 char url[1024], options[30] = "";
1210 if (rt->filter_source)
1211 av_strlcpy(options, "?connect=1", sizeof(options));
1212 /* Use source address if specified */
1213 if (reply->transports[0].source[0]) {
1214 ff_url_join(url, sizeof(url), "rtp", NULL,
1215 reply->transports[0].source,
1216 reply->transports[0].server_port_min, options);
1218 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1219 reply->transports[0].server_port_min, options);
1221 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1222 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1223 err = AVERROR_INVALIDDATA;
1226 /* Try to initialize the connection state in a
1227 * potential NAT router by sending dummy packets.
1228 * RTP/RTCP dummy packets are used for RDT, too.
1230 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1232 rtp_send_punch_packets(rtsp_st->rtp_handle);
1235 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1236 char url[1024], namebuf[50];
1237 struct sockaddr_storage addr;
1240 if (reply->transports[0].destination.ss_family) {
1241 addr = reply->transports[0].destination;
1242 port = reply->transports[0].port_min;
1243 ttl = reply->transports[0].ttl;
1245 addr = rtsp_st->sdp_ip;
1246 port = rtsp_st->sdp_port;
1247 ttl = rtsp_st->sdp_ttl;
1249 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1250 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1251 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1252 port, "?ttl=%d", ttl);
1253 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1254 err = AVERROR_INVALIDDATA;
1261 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1265 if (reply->timeout > 0)
1266 rt->timeout = reply->timeout;
1268 if (rt->server_type == RTSP_SERVER_REAL)
1269 rt->need_subscription = 1;
1274 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1275 if (rt->rtsp_streams[i]->rtp_handle) {
1276 url_close(rt->rtsp_streams[i]->rtp_handle);
1277 rt->rtsp_streams[i]->rtp_handle = NULL;
1283 void ff_rtsp_close_connections(AVFormatContext *s)
1285 RTSPState *rt = s->priv_data;
1286 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1287 url_close(rt->rtsp_hd);
1288 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1291 int ff_rtsp_connect(AVFormatContext *s)
1293 RTSPState *rt = s->priv_data;
1294 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1295 char *option_list, *option, *filename;
1296 int port, err, tcp_fd;
1297 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1298 int lower_transport_mask = 0;
1299 char real_challenge[64];
1300 struct sockaddr_storage peer;
1301 socklen_t peer_len = sizeof(peer);
1303 if (!ff_network_init())
1304 return AVERROR(EIO);
1306 rt->control_transport = RTSP_MODE_PLAIN;
1307 /* extract hostname and port */
1308 av_url_split(NULL, 0, auth, sizeof(auth),
1309 host, sizeof(host), &port, path, sizeof(path), s->filename);
1311 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1314 port = RTSP_DEFAULT_PORT;
1316 /* search for options */
1317 option_list = strrchr(path, '?');
1319 /* Strip out the RTSP specific options, write out the rest of
1320 * the options back into the same string. */
1321 filename = option_list;
1322 while (option_list) {
1323 /* move the option pointer */
1324 option = ++option_list;
1325 option_list = strchr(option_list, '&');
1329 /* handle the options */
1330 if (!strcmp(option, "udp")) {
1331 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1332 } else if (!strcmp(option, "multicast")) {
1333 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1334 } else if (!strcmp(option, "tcp")) {
1335 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1336 } else if(!strcmp(option, "http")) {
1337 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1338 rt->control_transport = RTSP_MODE_TUNNEL;
1339 } else if (!strcmp(option, "filter_src")) {
1340 rt->filter_source = 1;
1342 /* Write options back into the buffer, using memmove instead
1343 * of strcpy since the strings may overlap. */
1344 int len = strlen(option);
1345 memmove(++filename, option, len);
1347 if (option_list) *filename = '&';
1353 if (!lower_transport_mask)
1354 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1357 /* Only UDP or TCP - UDP multicast isn't supported. */
1358 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1359 (1 << RTSP_LOWER_TRANSPORT_TCP);
1360 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1361 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1362 "only UDP and TCP are supported for output.\n");
1363 err = AVERROR(EINVAL);
1368 /* Construct the URI used in request; this is similar to s->filename,
1369 * but with authentication credentials removed and RTSP specific options
1371 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1372 host, port, "%s", path);
1374 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1375 /* set up initial handshake for tunneling */
1376 char httpname[1024];
1377 char sessioncookie[17];
1380 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1381 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1382 av_get_random_seed(), av_get_random_seed());
1385 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1390 /* generate GET headers */
1391 snprintf(headers, sizeof(headers),
1392 "x-sessioncookie: %s\r\n"
1393 "Accept: application/x-rtsp-tunnelled\r\n"
1394 "Pragma: no-cache\r\n"
1395 "Cache-Control: no-cache\r\n",
1397 ff_http_set_headers(rt->rtsp_hd, headers);
1399 /* complete the connection */
1400 if (url_connect(rt->rtsp_hd)) {
1406 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1411 /* generate POST headers */
1412 snprintf(headers, sizeof(headers),
1413 "x-sessioncookie: %s\r\n"
1414 "Content-Type: application/x-rtsp-tunnelled\r\n"
1415 "Pragma: no-cache\r\n"
1416 "Cache-Control: no-cache\r\n"
1417 "Content-Length: 32767\r\n"
1418 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1420 ff_http_set_headers(rt->rtsp_hd_out, headers);
1421 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1423 /* Initialize the authentication state for the POST session. The HTTP
1424 * protocol implementation doesn't properly handle multi-pass
1425 * authentication for POST requests, since it would require one of
1427 * - implementing Expect: 100-continue, which many HTTP servers
1428 * don't support anyway, even less the RTSP servers that do HTTP
1430 * - sending the whole POST data until getting a 401 reply specifying
1431 * what authentication method to use, then resending all that data
1432 * - waiting for potential 401 replies directly after sending the
1433 * POST header (waiting for some unspecified time)
1434 * Therefore, we copy the full auth state, which works for both basic
1435 * and digest. (For digest, we would have to synchronize the nonce
1436 * count variable between the two sessions, if we'd do more requests
1437 * with the original session, though.)
1439 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1441 /* complete the connection */
1442 if (url_connect(rt->rtsp_hd_out)) {
1447 /* open the tcp connection */
1448 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1449 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1453 rt->rtsp_hd_out = rt->rtsp_hd;
1457 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1458 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1459 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1460 NULL, 0, NI_NUMERICHOST);
1463 /* request options supported by the server; this also detects server
1465 for (rt->server_type = RTSP_SERVER_RTP;;) {
1467 if (rt->server_type == RTSP_SERVER_REAL)
1470 * The following entries are required for proper
1471 * streaming from a Realmedia server. They are
1472 * interdependent in some way although we currently
1473 * don't quite understand how. Values were copied
1474 * from mplayer SVN r23589.
1475 * @param CompanyID is a 16-byte ID in base64
1476 * @param ClientChallenge is a 16-byte ID in hex
1478 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1479 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1480 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1481 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1483 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1484 if (reply->status_code != RTSP_STATUS_OK) {
1485 err = AVERROR_INVALIDDATA;
1489 /* detect server type if not standard-compliant RTP */
1490 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1491 rt->server_type = RTSP_SERVER_REAL;
1493 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1494 rt->server_type = RTSP_SERVER_WMS;
1495 } else if (rt->server_type == RTSP_SERVER_REAL)
1496 strcpy(real_challenge, reply->real_challenge);
1500 if (s->iformat && CONFIG_RTSP_DEMUXER)
1501 err = ff_rtsp_setup_input_streams(s, reply);
1502 else if (CONFIG_RTSP_MUXER)
1503 err = ff_rtsp_setup_output_streams(s, host);
1508 int lower_transport = ff_log2_tab[lower_transport_mask &
1509 ~(lower_transport_mask - 1)];
1511 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1512 rt->server_type == RTSP_SERVER_REAL ?
1513 real_challenge : NULL);
1516 lower_transport_mask &= ~(1 << lower_transport);
1517 if (lower_transport_mask == 0 && err == 1) {
1518 err = FF_NETERROR(EPROTONOSUPPORT);
1523 rt->state = RTSP_STATE_IDLE;
1524 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1527 ff_rtsp_close_streams(s);
1528 ff_rtsp_close_connections(s);
1529 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1530 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1531 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1539 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1542 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1543 uint8_t *buf, int buf_size, int64_t wait_end)
1545 RTSPState *rt = s->priv_data;
1546 RTSPStream *rtsp_st;
1548 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1552 if (url_interrupt_cb())
1553 return AVERROR(EINTR);
1554 if (wait_end && wait_end - av_gettime() < 0)
1555 return AVERROR(EAGAIN);
1558 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1559 FD_SET(tcp_fd, &rfds);
1564 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1565 rtsp_st = rt->rtsp_streams[i];
1566 if (rtsp_st->rtp_handle) {
1567 fd = url_get_file_handle(rtsp_st->rtp_handle);
1568 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1569 if (FFMAX(fd, fd_rtcp) > fd_max)
1570 fd_max = FFMAX(fd, fd_rtcp);
1572 FD_SET(fd_rtcp, &rfds);
1576 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1577 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1580 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1581 rtsp_st = rt->rtsp_streams[i];
1582 if (rtsp_st->rtp_handle) {
1583 fd = url_get_file_handle(rtsp_st->rtp_handle);
1584 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1585 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1586 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1588 *prtsp_st = rtsp_st;
1594 #if CONFIG_RTSP_DEMUXER
1595 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1596 RTSPMessageHeader reply;
1598 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1601 /* XXX: parse message */
1602 if (rt->state != RTSP_STATE_STREAMING)
1606 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1607 return FF_NETERROR(ETIMEDOUT);
1608 } else if (n < 0 && errno != EINTR)
1609 return AVERROR(errno);
1613 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1615 RTSPState *rt = s->priv_data;
1617 RTSPStream *rtsp_st, *first_queue_st = NULL;
1618 int64_t wait_end = 0;
1620 if (rt->nb_byes == rt->nb_rtsp_streams)
1623 /* get next frames from the same RTP packet */
1624 if (rt->cur_transport_priv) {
1625 if (rt->transport == RTSP_TRANSPORT_RDT) {
1626 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1628 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1630 rt->cur_transport_priv = NULL;
1632 } else if (ret == 1) {
1635 rt->cur_transport_priv = NULL;
1638 if (rt->transport == RTSP_TRANSPORT_RTP) {
1640 int64_t first_queue_time = 0;
1641 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1642 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1646 queue_time = ff_rtp_queued_packet_time(rtpctx);
1647 if (queue_time && (queue_time - first_queue_time < 0 ||
1648 !first_queue_time)) {
1649 first_queue_time = queue_time;
1650 first_queue_st = rt->rtsp_streams[i];
1653 if (first_queue_time)
1654 wait_end = first_queue_time + s->max_delay;
1657 /* read next RTP packet */
1660 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1662 return AVERROR(ENOMEM);
1665 switch(rt->lower_transport) {
1667 #if CONFIG_RTSP_DEMUXER
1668 case RTSP_LOWER_TRANSPORT_TCP:
1669 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1672 case RTSP_LOWER_TRANSPORT_UDP:
1673 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1674 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1675 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1676 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1679 if (len == AVERROR(EAGAIN) && first_queue_st &&
1680 rt->transport == RTSP_TRANSPORT_RTP) {
1681 rtsp_st = first_queue_st;
1682 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1689 if (rt->transport == RTSP_TRANSPORT_RDT) {
1690 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1692 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1694 /* Either bad packet, or a RTCP packet. Check if the
1695 * first_rtcp_ntp_time field was initialized. */
1696 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1697 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1698 /* first_rtcp_ntp_time has been initialized for this stream,
1699 * copy the same value to all other uninitialized streams,
1700 * in order to map their timestamp origin to the same ntp time
1703 AVStream *st = NULL;
1704 if (rtsp_st->stream_index >= 0)
1705 st = s->streams[rtsp_st->stream_index];
1706 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1707 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1708 AVStream *st2 = NULL;
1709 if (rt->rtsp_streams[i]->stream_index >= 0)
1710 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1711 if (rtpctx2 && st && st2 &&
1712 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1713 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1714 rtpctx2->rtcp_ts_offset = av_rescale_q(
1715 rtpctx->rtcp_ts_offset, st->time_base,
1720 if (ret == -RTCP_BYE) {
1723 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1724 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1726 if (rt->nb_byes == rt->nb_rtsp_streams)
1735 /* more packets may follow, so we save the RTP context */
1736 rt->cur_transport_priv = rtsp_st->transport_priv;
1740 #endif /* CONFIG_RTPDEC */
1742 #if CONFIG_SDP_DEMUXER
1743 static int sdp_probe(AVProbeData *p1)
1745 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1747 /* we look for a line beginning "c=IN IP" */
1748 while (p < p_end && *p != '\0') {
1749 if (p + sizeof("c=IN IP") - 1 < p_end &&
1750 av_strstart(p, "c=IN IP", NULL))
1751 return AVPROBE_SCORE_MAX / 2;
1753 while (p < p_end - 1 && *p != '\n') p++;
1762 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1764 RTSPState *rt = s->priv_data;
1765 RTSPStream *rtsp_st;
1770 if (!ff_network_init())
1771 return AVERROR(EIO);
1773 /* read the whole sdp file */
1774 /* XXX: better loading */
1775 content = av_malloc(SDP_MAX_SIZE);
1776 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1779 return AVERROR_INVALIDDATA;
1781 content[size] ='\0';
1783 ff_sdp_parse(s, content);
1786 /* open each RTP stream */
1787 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1789 rtsp_st = rt->rtsp_streams[i];
1791 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1792 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1793 ff_url_join(url, sizeof(url), "rtp", NULL,
1794 namebuf, rtsp_st->sdp_port,
1795 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1797 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1798 err = AVERROR_INVALIDDATA;
1801 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1806 ff_rtsp_close_streams(s);
1811 static int sdp_read_close(AVFormatContext *s)
1813 ff_rtsp_close_streams(s);
1818 AVInputFormat sdp_demuxer = {
1820 NULL_IF_CONFIG_SMALL("SDP"),
1824 ff_rtsp_fetch_packet,
1827 #endif /* CONFIG_SDP_DEMUXER */
1829 #if CONFIG_RTP_DEMUXER
1830 static int rtp_probe(AVProbeData *p)
1832 if (av_strstart(p->filename, "rtp:", NULL))
1833 return AVPROBE_SCORE_MAX;
1837 static int rtp_read_header(AVFormatContext *s,
1838 AVFormatParameters *ap)
1840 uint8_t recvbuf[1500];
1841 char host[500], sdp[500];
1843 URLContext* in = NULL;
1845 AVCodecContext codec;
1846 struct sockaddr_storage addr;
1848 socklen_t addrlen = sizeof(addr);
1850 if (!ff_network_init())
1851 return AVERROR(EIO);
1853 ret = url_open(&in, s->filename, URL_RDONLY);
1858 ret = url_read(in, recvbuf, sizeof(recvbuf));
1859 if (ret == AVERROR(EAGAIN))
1864 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1868 if ((recvbuf[0] & 0xc0) != 0x80) {
1869 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1874 payload_type = recvbuf[1] & 0x7f;
1877 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1881 memset(&codec, 0, sizeof(codec));
1882 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1883 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1884 "without an SDP file describing it\n",
1888 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1889 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1890 "properly you need an SDP file "
1894 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1895 NULL, 0, s->filename);
1897 snprintf(sdp, sizeof(sdp),
1898 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1899 addr.ss_family == AF_INET ? 4 : 6, host,
1900 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1901 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1902 port, payload_type);
1903 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1905 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1908 /* sdp_read_header initializes this again */
1911 ret = sdp_read_header(s, ap);
1922 AVInputFormat rtp_demuxer = {
1924 NULL_IF_CONFIG_SMALL("RTP input format"),
1928 ff_rtsp_fetch_packet,
1930 .flags = AVFMT_NOFILE,
1932 #endif /* CONFIG_RTP_DEMUXER */