3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
30 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
51 /* Timeout values for socket poll, in ms,
52 * and read_packet(), in seconds */
53 #define POLL_TIMEOUT_MS 100
54 #define READ_PACKET_TIMEOUT_S 10
55 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
56 #define SDP_MAX_SIZE 16384
57 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 static void get_word_until_chars(char *buf, int buf_size,
60 const char *sep, const char **pp)
66 p += strspn(p, SPACE_CHARS);
68 while (!strchr(sep, *p) && *p != '\0') {
69 if ((q - buf) < buf_size - 1)
78 static void get_word_sep(char *buf, int buf_size, const char *sep,
81 if (**pp == '/') (*pp)++;
82 get_word_until_chars(buf, buf_size, sep, pp);
85 static void get_word(char *buf, int buf_size, const char **pp)
87 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
90 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
92 * Used for seeking in the rtp stream.
94 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
98 p += strspn(p, SPACE_CHARS);
99 if (!av_stristart(p, "npt=", &p))
102 *start = AV_NOPTS_VALUE;
103 *end = AV_NOPTS_VALUE;
105 get_word_sep(buf, sizeof(buf), "-", &p);
106 av_parse_time(start, buf, 1);
109 get_word_sep(buf, sizeof(buf), "-", &p);
110 av_parse_time(end, buf, 1);
112 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
113 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
116 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
118 struct addrinfo hints, *ai = NULL;
119 memset(&hints, 0, sizeof(hints));
120 hints.ai_flags = AI_NUMERICHOST;
121 if (getaddrinfo(buf, NULL, &hints, &ai))
123 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
129 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
130 RTSPStream *rtsp_st, AVCodecContext *codec)
134 codec->codec_id = handler->codec_id;
135 rtsp_st->dynamic_handler = handler;
137 rtsp_st->dynamic_protocol_context = handler->alloc();
140 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
141 static int sdp_parse_rtpmap(AVFormatContext *s,
142 AVStream *st, RTSPStream *rtsp_st,
143 int payload_type, const char *p)
145 AVCodecContext *codec = st->codec;
151 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
152 * see if we can handle this kind of payload.
153 * The space should normally not be there but some Real streams or
154 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
155 * have a trailing space. */
156 get_word_sep(buf, sizeof(buf), "/ ", &p);
157 if (payload_type >= RTP_PT_PRIVATE) {
158 RTPDynamicProtocolHandler *handler =
159 ff_rtp_handler_find_by_name(buf, codec->codec_type);
160 init_rtp_handler(handler, rtsp_st, codec);
161 /* If no dynamic handler was found, check with the list of standard
162 * allocated types, if such a stream for some reason happens to
163 * use a private payload type. This isn't handled in rtpdec.c, since
164 * the format name from the rtpmap line never is passed into rtpdec. */
165 if (!rtsp_st->dynamic_handler)
166 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
168 /* We are in a standard case
169 * (from http://www.iana.org/assignments/rtp-parameters). */
170 /* search into AVRtpPayloadTypes[] */
171 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
174 c = avcodec_find_decoder(codec->codec_id);
180 get_word_sep(buf, sizeof(buf), "/", &p);
182 switch (codec->codec_type) {
183 case AVMEDIA_TYPE_AUDIO:
184 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
185 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
186 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
188 codec->sample_rate = i;
189 av_set_pts_info(st, 32, 1, codec->sample_rate);
190 get_word_sep(buf, sizeof(buf), "/", &p);
194 // TODO: there is a bug here; if it is a mono stream, and
195 // less than 22000Hz, faad upconverts to stereo and twice
196 // the frequency. No problem, but the sample rate is being
197 // set here by the sdp line. Patch on its way. (rdm)
199 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
201 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
204 case AVMEDIA_TYPE_VIDEO:
205 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
207 av_set_pts_info(st, 32, 1, i);
215 /* parse the attribute line from the fmtp a line of an sdp response. This
216 * is broken out as a function because it is used in rtp_h264.c, which is
218 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
219 char *value, int value_size)
221 *p += strspn(*p, SPACE_CHARS);
223 get_word_sep(attr, attr_size, "=", p);
226 get_word_sep(value, value_size, ";", p);
234 typedef struct SDPParseState {
236 struct sockaddr_storage default_ip;
238 int skip_media; ///< set if an unknown m= line occurs
241 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
242 int letter, const char *buf)
244 RTSPState *rt = s->priv_data;
245 char buf1[64], st_type[64];
247 enum AVMediaType codec_type;
251 struct sockaddr_storage sdp_ip;
254 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
257 if (s1->skip_media && letter != 'm')
261 get_word(buf1, sizeof(buf1), &p);
262 if (strcmp(buf1, "IN") != 0)
264 get_word(buf1, sizeof(buf1), &p);
265 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
267 get_word_sep(buf1, sizeof(buf1), "/", &p);
268 if (get_sockaddr(buf1, &sdp_ip))
273 get_word_sep(buf1, sizeof(buf1), "/", &p);
276 if (s->nb_streams == 0) {
277 s1->default_ip = sdp_ip;
278 s1->default_ttl = ttl;
280 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
281 rtsp_st->sdp_ip = sdp_ip;
282 rtsp_st->sdp_ttl = ttl;
286 av_dict_set(&s->metadata, "title", p, 0);
289 if (s->nb_streams == 0) {
290 av_dict_set(&s->metadata, "comment", p, 0);
297 get_word(st_type, sizeof(st_type), &p);
298 if (!strcmp(st_type, "audio")) {
299 codec_type = AVMEDIA_TYPE_AUDIO;
300 } else if (!strcmp(st_type, "video")) {
301 codec_type = AVMEDIA_TYPE_VIDEO;
302 } else if (!strcmp(st_type, "application")) {
303 codec_type = AVMEDIA_TYPE_DATA;
308 rtsp_st = av_mallocz(sizeof(RTSPStream));
311 rtsp_st->stream_index = -1;
312 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
314 rtsp_st->sdp_ip = s1->default_ip;
315 rtsp_st->sdp_ttl = s1->default_ttl;
317 get_word(buf1, sizeof(buf1), &p); /* port */
318 rtsp_st->sdp_port = atoi(buf1);
320 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
322 /* XXX: handle list of formats */
323 get_word(buf1, sizeof(buf1), &p); /* format list */
324 rtsp_st->sdp_payload_type = atoi(buf1);
326 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
327 /* no corresponding stream */
329 st = av_new_stream(s, rt->nb_rtsp_streams - 1);
332 rtsp_st->stream_index = st->index;
333 st->codec->codec_type = codec_type;
334 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
335 RTPDynamicProtocolHandler *handler;
336 /* if standard payload type, we can find the codec right now */
337 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
338 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
339 st->codec->sample_rate > 0)
340 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
341 /* Even static payload types may need a custom depacketizer */
342 handler = ff_rtp_handler_find_by_id(
343 rtsp_st->sdp_payload_type, st->codec->codec_type);
344 init_rtp_handler(handler, rtsp_st, st->codec);
347 /* put a default control url */
348 av_strlcpy(rtsp_st->control_url, rt->control_uri,
349 sizeof(rtsp_st->control_url));
352 if (av_strstart(p, "control:", &p)) {
353 if (s->nb_streams == 0) {
354 if (!strncmp(p, "rtsp://", 7))
355 av_strlcpy(rt->control_uri, p,
356 sizeof(rt->control_uri));
359 /* get the control url */
360 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
362 /* XXX: may need to add full url resolution */
363 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
365 if (proto[0] == '\0') {
366 /* relative control URL */
367 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
368 av_strlcat(rtsp_st->control_url, "/",
369 sizeof(rtsp_st->control_url));
370 av_strlcat(rtsp_st->control_url, p,
371 sizeof(rtsp_st->control_url));
373 av_strlcpy(rtsp_st->control_url, p,
374 sizeof(rtsp_st->control_url));
376 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
377 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
378 get_word(buf1, sizeof(buf1), &p);
379 payload_type = atoi(buf1);
380 st = s->streams[s->nb_streams - 1];
381 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
382 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
383 } else if (av_strstart(p, "fmtp:", &p) ||
384 av_strstart(p, "framesize:", &p)) {
385 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
386 // let dynamic protocol handlers have a stab at the line.
387 get_word(buf1, sizeof(buf1), &p);
388 payload_type = atoi(buf1);
389 for (i = 0; i < rt->nb_rtsp_streams; i++) {
390 rtsp_st = rt->rtsp_streams[i];
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 = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
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)
434 RTSPState *rt = s->priv_data;
437 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
438 * contain long SDP lines containing complete ASF Headers (several
439 * kB) or arrays of MDPR (RM stream descriptor) headers plus
440 * "rulebooks" describing their properties. Therefore, the SDP line
443 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
444 * in rtpdec_xiph.c. */
446 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
448 memset(s1, 0, sizeof(SDPParseState));
451 p += strspn(p, SPACE_CHARS);
459 /* get the content */
461 while (*p != '\n' && *p != '\r' && *p != '\0') {
462 if ((q - buf) < sizeof(buf) - 1)
467 sdp_parse_line(s, s1, letter, buf);
469 while (*p != '\n' && *p != '\0')
474 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
475 if (!rt->p) return AVERROR(ENOMEM);
478 #endif /* CONFIG_RTPDEC */
480 void ff_rtsp_undo_setup(AVFormatContext *s)
482 RTSPState *rt = s->priv_data;
485 for (i = 0; i < rt->nb_rtsp_streams; i++) {
486 RTSPStream *rtsp_st = rt->rtsp_streams[i];
489 if (rtsp_st->transport_priv) {
491 AVFormatContext *rtpctx = rtsp_st->transport_priv;
492 av_write_trailer(rtpctx);
493 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
495 avio_close_dyn_buf(rtpctx->pb, &ptr);
498 avio_close(rtpctx->pb);
500 avformat_free_context(rtpctx);
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 ffurl_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->free(
526 rtsp_st->dynamic_protocol_context);
530 av_free(rt->rtsp_streams);
532 av_close_input_stream (rt->asf_ctx);
536 av_free(rt->recvbuf);
539 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
541 RTSPState *rt = s->priv_data;
544 /* open the RTP context */
545 if (rtsp_st->stream_index >= 0)
546 st = s->streams[rtsp_st->stream_index];
548 s->ctx_flags |= AVFMTCTX_NOHEADER;
550 if (s->oformat && CONFIG_RTSP_MUXER) {
551 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
553 RTSP_TCP_MAX_PACKET_SIZE);
554 /* Ownership of rtp_handle is passed to the rtp mux context */
555 rtsp_st->rtp_handle = NULL;
556 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
557 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
558 rtsp_st->dynamic_protocol_context,
559 rtsp_st->dynamic_handler);
560 else if (CONFIG_RTPDEC)
561 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
562 rtsp_st->sdp_payload_type,
563 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
564 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
566 if (!rtsp_st->transport_priv) {
567 return AVERROR(ENOMEM);
568 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
569 if (rtsp_st->dynamic_handler) {
570 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
571 rtsp_st->dynamic_protocol_context,
572 rtsp_st->dynamic_handler);
579 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
580 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
586 p += strspn(p, SPACE_CHARS);
587 v = strtol(p, (char **)&p, 10);
591 v = strtol(p, (char **)&p, 10);
600 /* XXX: only one transport specification is parsed */
601 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
603 char transport_protocol[16];
605 char lower_transport[16];
607 RTSPTransportField *th;
610 reply->nb_transports = 0;
613 p += strspn(p, SPACE_CHARS);
617 th = &reply->transports[reply->nb_transports];
619 get_word_sep(transport_protocol, sizeof(transport_protocol),
621 if (!strcasecmp (transport_protocol, "rtp")) {
622 get_word_sep(profile, sizeof(profile), "/;,", &p);
623 lower_transport[0] = '\0';
624 /* rtp/avp/<protocol> */
626 get_word_sep(lower_transport, sizeof(lower_transport),
629 th->transport = RTSP_TRANSPORT_RTP;
630 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
631 !strcasecmp (transport_protocol, "x-real-rdt")) {
632 /* x-pn-tng/<protocol> */
633 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
635 th->transport = RTSP_TRANSPORT_RDT;
637 if (!strcasecmp(lower_transport, "TCP"))
638 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
640 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
644 /* get each parameter */
645 while (*p != '\0' && *p != ',') {
646 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
647 if (!strcmp(parameter, "port")) {
650 rtsp_parse_range(&th->port_min, &th->port_max, &p);
652 } else if (!strcmp(parameter, "client_port")) {
655 rtsp_parse_range(&th->client_port_min,
656 &th->client_port_max, &p);
658 } else if (!strcmp(parameter, "server_port")) {
661 rtsp_parse_range(&th->server_port_min,
662 &th->server_port_max, &p);
664 } else if (!strcmp(parameter, "interleaved")) {
667 rtsp_parse_range(&th->interleaved_min,
668 &th->interleaved_max, &p);
670 } else if (!strcmp(parameter, "multicast")) {
671 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
672 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
673 } else if (!strcmp(parameter, "ttl")) {
676 th->ttl = strtol(p, (char **)&p, 10);
678 } else if (!strcmp(parameter, "destination")) {
681 get_word_sep(buf, sizeof(buf), ";,", &p);
682 get_sockaddr(buf, &th->destination);
684 } else if (!strcmp(parameter, "source")) {
687 get_word_sep(buf, sizeof(buf), ";,", &p);
688 av_strlcpy(th->source, buf, sizeof(th->source));
692 while (*p != ';' && *p != '\0' && *p != ',')
700 reply->nb_transports++;
704 static void handle_rtp_info(RTSPState *rt, const char *url,
705 uint32_t seq, uint32_t rtptime)
708 if (!rtptime || !url[0])
710 if (rt->transport != RTSP_TRANSPORT_RTP)
712 for (i = 0; i < rt->nb_rtsp_streams; i++) {
713 RTSPStream *rtsp_st = rt->rtsp_streams[i];
714 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
717 if (!strcmp(rtsp_st->control_url, url)) {
718 rtpctx->base_timestamp = rtptime;
724 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
727 char key[20], value[1024], url[1024] = "";
728 uint32_t seq = 0, rtptime = 0;
731 p += strspn(p, SPACE_CHARS);
734 get_word_sep(key, sizeof(key), "=", &p);
738 get_word_sep(value, sizeof(value), ";, ", &p);
740 if (!strcmp(key, "url"))
741 av_strlcpy(url, value, sizeof(url));
742 else if (!strcmp(key, "seq"))
743 seq = strtoul(value, NULL, 10);
744 else if (!strcmp(key, "rtptime"))
745 rtptime = strtoul(value, NULL, 10);
747 handle_rtp_info(rt, url, seq, rtptime);
756 handle_rtp_info(rt, url, seq, rtptime);
759 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
760 RTSPState *rt, const char *method)
764 /* NOTE: we do case independent match for broken servers */
766 if (av_stristart(p, "Session:", &p)) {
768 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
769 if (av_stristart(p, ";timeout=", &p) &&
770 (t = strtol(p, NULL, 10)) > 0) {
773 } else if (av_stristart(p, "Content-Length:", &p)) {
774 reply->content_length = strtol(p, NULL, 10);
775 } else if (av_stristart(p, "Transport:", &p)) {
776 rtsp_parse_transport(reply, p);
777 } else if (av_stristart(p, "CSeq:", &p)) {
778 reply->seq = strtol(p, NULL, 10);
779 } else if (av_stristart(p, "Range:", &p)) {
780 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
781 } else if (av_stristart(p, "RealChallenge1:", &p)) {
782 p += strspn(p, SPACE_CHARS);
783 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
784 } else if (av_stristart(p, "Server:", &p)) {
785 p += strspn(p, SPACE_CHARS);
786 av_strlcpy(reply->server, p, sizeof(reply->server));
787 } else if (av_stristart(p, "Notice:", &p) ||
788 av_stristart(p, "X-Notice:", &p)) {
789 reply->notice = strtol(p, NULL, 10);
790 } else if (av_stristart(p, "Location:", &p)) {
791 p += strspn(p, SPACE_CHARS);
792 av_strlcpy(reply->location, p , sizeof(reply->location));
793 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
794 p += strspn(p, SPACE_CHARS);
795 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
796 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
797 p += strspn(p, SPACE_CHARS);
798 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
799 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
800 p += strspn(p, SPACE_CHARS);
801 if (method && !strcmp(method, "DESCRIBE"))
802 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
803 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
804 p += strspn(p, SPACE_CHARS);
805 if (method && !strcmp(method, "PLAY"))
806 rtsp_parse_rtp_info(rt, p);
807 } else if (av_stristart(p, "Public:", &p) && rt) {
808 if (strstr(p, "GET_PARAMETER") &&
809 method && !strcmp(method, "OPTIONS"))
810 rt->get_parameter_supported = 1;
814 /* skip a RTP/TCP interleaved packet */
815 void ff_rtsp_skip_packet(AVFormatContext *s)
817 RTSPState *rt = s->priv_data;
821 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
824 len = AV_RB16(buf + 1);
826 av_dlog(s, "skipping RTP packet len=%d\n", len);
831 if (len1 > sizeof(buf))
833 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
840 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
841 unsigned char **content_ptr,
842 int return_on_interleaved_data, const char *method)
844 RTSPState *rt = s->priv_data;
845 char buf[4096], buf1[1024], *q;
848 int ret, content_length, line_count = 0;
849 unsigned char *content = NULL;
851 memset(reply, 0, sizeof(*reply));
853 /* parse reply (XXX: use buffers) */
854 rt->last_reply[0] = '\0';
858 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
859 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
865 /* XXX: only parse it if first char on line ? */
866 if (return_on_interleaved_data) {
869 ff_rtsp_skip_packet(s);
870 } else if (ch != '\r') {
871 if ((q - buf) < sizeof(buf) - 1)
877 av_dlog(s, "line='%s'\n", buf);
879 /* test if last line */
883 if (line_count == 0) {
885 get_word(buf1, sizeof(buf1), &p);
886 get_word(buf1, sizeof(buf1), &p);
887 reply->status_code = atoi(buf1);
888 av_strlcpy(reply->reason, p, sizeof(reply->reason));
890 ff_rtsp_parse_line(reply, p, rt, method);
891 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
892 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
897 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
898 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
900 content_length = reply->content_length;
901 if (content_length > 0) {
902 /* leave some room for a trailing '\0' (useful for simple parsing) */
903 content = av_malloc(content_length + 1);
904 ffurl_read_complete(rt->rtsp_hd, content, content_length);
905 content[content_length] = '\0';
908 *content_ptr = content;
912 if (rt->seq != reply->seq) {
913 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
914 rt->seq, reply->seq);
918 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
919 reply->notice == 2104 /* Start-of-Stream Reached */ ||
920 reply->notice == 2306 /* Continuous Feed Terminated */) {
921 rt->state = RTSP_STATE_IDLE;
922 } else if (reply->notice >= 4400 && reply->notice < 5500) {
923 return AVERROR(EIO); /* data or server error */
924 } else if (reply->notice == 2401 /* Ticket Expired */ ||
925 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
926 return AVERROR(EPERM);
932 * Send a command to the RTSP server without waiting for the reply.
934 * @param s RTSP (de)muxer context
935 * @param method the method for the request
936 * @param url the target url for the request
937 * @param headers extra header lines to include in the request
938 * @param send_content if non-null, the data to send as request body content
939 * @param send_content_length the length of the send_content data, or 0 if
940 * send_content is null
942 * @return zero if success, nonzero otherwise
944 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
945 const char *method, const char *url,
947 const unsigned char *send_content,
948 int send_content_length)
950 RTSPState *rt = s->priv_data;
951 char buf[4096], *out_buf;
952 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
954 /* Add in RTSP headers */
957 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
959 av_strlcat(buf, headers, sizeof(buf));
960 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
961 if (rt->session_id[0] != '\0' && (!headers ||
962 !strstr(headers, "\nIf-Match:"))) {
963 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
966 char *str = ff_http_auth_create_response(&rt->auth_state,
967 rt->auth, url, method);
969 av_strlcat(buf, str, sizeof(buf));
972 if (send_content_length > 0 && send_content)
973 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
974 av_strlcat(buf, "\r\n", sizeof(buf));
976 /* base64 encode rtsp if tunneling */
977 if (rt->control_transport == RTSP_MODE_TUNNEL) {
978 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
982 av_dlog(s, "Sending:\n%s--\n", buf);
984 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
985 if (send_content_length > 0 && send_content) {
986 if (rt->control_transport == RTSP_MODE_TUNNEL) {
987 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
988 "with content data not supported\n");
989 return AVERROR_PATCHWELCOME;
991 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
993 rt->last_cmd_time = av_gettime();
998 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
999 const char *url, const char *headers)
1001 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1004 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1005 const char *headers, RTSPMessageHeader *reply,
1006 unsigned char **content_ptr)
1008 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1009 content_ptr, NULL, 0);
1012 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1013 const char *method, const char *url,
1015 RTSPMessageHeader *reply,
1016 unsigned char **content_ptr,
1017 const unsigned char *send_content,
1018 int send_content_length)
1020 RTSPState *rt = s->priv_data;
1021 HTTPAuthType cur_auth_type;
1025 cur_auth_type = rt->auth_state.auth_type;
1026 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1028 send_content_length)))
1031 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1034 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1035 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1038 if (reply->status_code > 400){
1039 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1043 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1049 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1050 int lower_transport, const char *real_challenge)
1052 RTSPState *rt = s->priv_data;
1053 int rtx, j, i, err, interleave = 0;
1054 RTSPStream *rtsp_st;
1055 RTSPMessageHeader reply1, *reply = &reply1;
1057 const char *trans_pref;
1059 if (rt->transport == RTSP_TRANSPORT_RDT)
1060 trans_pref = "x-pn-tng";
1062 trans_pref = "RTP/AVP";
1064 /* default timeout: 1 minute */
1067 /* for each stream, make the setup request */
1068 /* XXX: we assume the same server is used for the control of each
1071 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1072 char transport[2048];
1075 * WMS serves all UDP data over a single connection, the RTX, which
1076 * isn't necessarily the first in the SDP but has to be the first
1077 * to be set up, else the second/third SETUP will fail with a 461.
1079 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1080 rt->server_type == RTSP_SERVER_WMS) {
1083 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1084 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1086 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1090 if (rtx == rt->nb_rtsp_streams)
1091 return -1; /* no RTX found */
1092 rtsp_st = rt->rtsp_streams[rtx];
1094 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1096 rtsp_st = rt->rtsp_streams[i];
1099 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1102 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1103 port = reply->transports[0].client_port_min;
1107 /* first try in specified port range */
1108 if (RTSP_RTP_PORT_MIN != 0) {
1109 while (j <= RTSP_RTP_PORT_MAX) {
1110 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1111 "?localport=%d", j);
1112 /* we will use two ports per rtp stream (rtp and rtcp) */
1114 if (ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE) == 0)
1119 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1124 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1126 snprintf(transport, sizeof(transport) - 1,
1127 "%s/UDP;", trans_pref);
1128 if (rt->server_type != RTSP_SERVER_REAL)
1129 av_strlcat(transport, "unicast;", sizeof(transport));
1130 av_strlcatf(transport, sizeof(transport),
1131 "client_port=%d", port);
1132 if (rt->transport == RTSP_TRANSPORT_RTP &&
1133 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1134 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1138 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1139 /* For WMS streams, the application streams are only used for
1140 * UDP. When trying to set it up for TCP streams, the server
1141 * will return an error. Therefore, we skip those streams. */
1142 if (rt->server_type == RTSP_SERVER_WMS &&
1143 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1146 snprintf(transport, sizeof(transport) - 1,
1147 "%s/TCP;", trans_pref);
1148 if (rt->transport != RTSP_TRANSPORT_RDT)
1149 av_strlcat(transport, "unicast;", sizeof(transport));
1150 av_strlcatf(transport, sizeof(transport),
1151 "interleaved=%d-%d",
1152 interleave, interleave + 1);
1156 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1157 snprintf(transport, sizeof(transport) - 1,
1158 "%s/UDP;multicast", trans_pref);
1161 av_strlcat(transport, ";mode=receive", sizeof(transport));
1162 } else if (rt->server_type == RTSP_SERVER_REAL ||
1163 rt->server_type == RTSP_SERVER_WMS)
1164 av_strlcat(transport, ";mode=play", sizeof(transport));
1165 snprintf(cmd, sizeof(cmd),
1166 "Transport: %s\r\n",
1168 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1169 char real_res[41], real_csum[9];
1170 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1172 av_strlcatf(cmd, sizeof(cmd),
1174 "RealChallenge2: %s, sd=%s\r\n",
1175 rt->session_id, real_res, real_csum);
1177 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1178 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1181 } else if (reply->status_code != RTSP_STATUS_OK ||
1182 reply->nb_transports != 1) {
1183 err = AVERROR_INVALIDDATA;
1187 /* XXX: same protocol for all streams is required */
1189 if (reply->transports[0].lower_transport != rt->lower_transport ||
1190 reply->transports[0].transport != rt->transport) {
1191 err = AVERROR_INVALIDDATA;
1195 rt->lower_transport = reply->transports[0].lower_transport;
1196 rt->transport = reply->transports[0].transport;
1199 /* Fail if the server responded with another lower transport mode
1200 * than what we requested. */
1201 if (reply->transports[0].lower_transport != lower_transport) {
1202 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1203 err = AVERROR_INVALIDDATA;
1207 switch(reply->transports[0].lower_transport) {
1208 case RTSP_LOWER_TRANSPORT_TCP:
1209 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1210 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1213 case RTSP_LOWER_TRANSPORT_UDP: {
1214 char url[1024], options[30] = "";
1216 if (rt->filter_source)
1217 av_strlcpy(options, "?connect=1", sizeof(options));
1218 /* Use source address if specified */
1219 if (reply->transports[0].source[0]) {
1220 ff_url_join(url, sizeof(url), "rtp", NULL,
1221 reply->transports[0].source,
1222 reply->transports[0].server_port_min, "%s", options);
1224 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1225 reply->transports[0].server_port_min, "%s", options);
1227 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1228 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1229 err = AVERROR_INVALIDDATA;
1232 /* Try to initialize the connection state in a
1233 * potential NAT router by sending dummy packets.
1234 * RTP/RTCP dummy packets are used for RDT, too.
1236 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1238 rtp_send_punch_packets(rtsp_st->rtp_handle);
1241 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1242 char url[1024], namebuf[50];
1243 struct sockaddr_storage addr;
1246 if (reply->transports[0].destination.ss_family) {
1247 addr = reply->transports[0].destination;
1248 port = reply->transports[0].port_min;
1249 ttl = reply->transports[0].ttl;
1251 addr = rtsp_st->sdp_ip;
1252 port = rtsp_st->sdp_port;
1253 ttl = rtsp_st->sdp_ttl;
1255 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1256 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1257 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1258 port, "?ttl=%d", ttl);
1259 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE) < 0) {
1260 err = AVERROR_INVALIDDATA;
1267 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1271 if (reply->timeout > 0)
1272 rt->timeout = reply->timeout;
1274 if (rt->server_type == RTSP_SERVER_REAL)
1275 rt->need_subscription = 1;
1280 ff_rtsp_undo_setup(s);
1284 void ff_rtsp_close_connections(AVFormatContext *s)
1286 RTSPState *rt = s->priv_data;
1287 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1288 ffurl_close(rt->rtsp_hd);
1289 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1292 int ff_rtsp_connect(AVFormatContext *s)
1294 RTSPState *rt = s->priv_data;
1295 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1296 char *option_list, *option, *filename;
1297 int port, err, tcp_fd;
1298 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1299 int lower_transport_mask = 0;
1300 char real_challenge[64] = "";
1301 struct sockaddr_storage peer;
1302 socklen_t peer_len = sizeof(peer);
1304 if (!ff_network_init())
1305 return AVERROR(EIO);
1307 rt->control_transport = RTSP_MODE_PLAIN;
1308 /* extract hostname and port */
1309 av_url_split(NULL, 0, auth, sizeof(auth),
1310 host, sizeof(host), &port, path, sizeof(path), s->filename);
1312 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1315 port = RTSP_DEFAULT_PORT;
1317 /* search for options */
1318 option_list = strrchr(path, '?');
1320 /* Strip out the RTSP specific options, write out the rest of
1321 * the options back into the same string. */
1322 filename = option_list;
1323 while (option_list) {
1324 /* move the option pointer */
1325 option = ++option_list;
1326 option_list = strchr(option_list, '&');
1330 /* handle the options */
1331 if (!strcmp(option, "udp")) {
1332 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1333 } else if (!strcmp(option, "multicast")) {
1334 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1335 } else if (!strcmp(option, "tcp")) {
1336 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1337 } else if(!strcmp(option, "http")) {
1338 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1339 rt->control_transport = RTSP_MODE_TUNNEL;
1340 } else if (!strcmp(option, "filter_src")) {
1341 rt->filter_source = 1;
1343 /* Write options back into the buffer, using memmove instead
1344 * of strcpy since the strings may overlap. */
1345 int len = strlen(option);
1346 memmove(++filename, option, len);
1348 if (option_list) *filename = '&';
1354 if (!lower_transport_mask)
1355 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1358 /* Only UDP or TCP - UDP multicast isn't supported. */
1359 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1360 (1 << RTSP_LOWER_TRANSPORT_TCP);
1361 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1362 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1363 "only UDP and TCP are supported for output.\n");
1364 err = AVERROR(EINVAL);
1369 /* Construct the URI used in request; this is similar to s->filename,
1370 * but with authentication credentials removed and RTSP specific options
1372 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1373 host, port, "%s", path);
1375 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1376 /* set up initial handshake for tunneling */
1377 char httpname[1024];
1378 char sessioncookie[17];
1381 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1382 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1383 av_get_random_seed(), av_get_random_seed());
1386 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ) < 0) {
1391 /* generate GET headers */
1392 snprintf(headers, sizeof(headers),
1393 "x-sessioncookie: %s\r\n"
1394 "Accept: application/x-rtsp-tunnelled\r\n"
1395 "Pragma: no-cache\r\n"
1396 "Cache-Control: no-cache\r\n",
1398 ff_http_set_headers(rt->rtsp_hd, headers);
1400 /* complete the connection */
1401 if (ffurl_connect(rt->rtsp_hd)) {
1407 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE) < 0 ) {
1412 /* generate POST headers */
1413 snprintf(headers, sizeof(headers),
1414 "x-sessioncookie: %s\r\n"
1415 "Content-Type: application/x-rtsp-tunnelled\r\n"
1416 "Pragma: no-cache\r\n"
1417 "Cache-Control: no-cache\r\n"
1418 "Content-Length: 32767\r\n"
1419 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1421 ff_http_set_headers(rt->rtsp_hd_out, headers);
1422 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1424 /* Initialize the authentication state for the POST session. The HTTP
1425 * protocol implementation doesn't properly handle multi-pass
1426 * authentication for POST requests, since it would require one of
1428 * - implementing Expect: 100-continue, which many HTTP servers
1429 * don't support anyway, even less the RTSP servers that do HTTP
1431 * - sending the whole POST data until getting a 401 reply specifying
1432 * what authentication method to use, then resending all that data
1433 * - waiting for potential 401 replies directly after sending the
1434 * POST header (waiting for some unspecified time)
1435 * Therefore, we copy the full auth state, which works for both basic
1436 * and digest. (For digest, we would have to synchronize the nonce
1437 * count variable between the two sessions, if we'd do more requests
1438 * with the original session, though.)
1440 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1442 /* complete the connection */
1443 if (ffurl_connect(rt->rtsp_hd_out)) {
1448 /* open the tcp connection */
1449 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1450 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE) < 0) {
1454 rt->rtsp_hd_out = rt->rtsp_hd;
1458 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1459 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1460 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1461 NULL, 0, NI_NUMERICHOST);
1464 /* request options supported by the server; this also detects server
1466 for (rt->server_type = RTSP_SERVER_RTP;;) {
1468 if (rt->server_type == RTSP_SERVER_REAL)
1471 * The following entries are required for proper
1472 * streaming from a Realmedia server. They are
1473 * interdependent in some way although we currently
1474 * don't quite understand how. Values were copied
1475 * from mplayer SVN r23589.
1476 * ClientChallenge is a 16-byte ID in hex
1477 * CompanyID is a 16-byte ID in base64
1479 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1480 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1481 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1482 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1484 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1485 if (reply->status_code != RTSP_STATUS_OK) {
1486 err = AVERROR_INVALIDDATA;
1490 /* detect server type if not standard-compliant RTP */
1491 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1492 rt->server_type = RTSP_SERVER_REAL;
1494 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1495 rt->server_type = RTSP_SERVER_WMS;
1496 } else if (rt->server_type == RTSP_SERVER_REAL)
1497 strcpy(real_challenge, reply->real_challenge);
1501 if (s->iformat && CONFIG_RTSP_DEMUXER)
1502 err = ff_rtsp_setup_input_streams(s, reply);
1503 else if (CONFIG_RTSP_MUXER)
1504 err = ff_rtsp_setup_output_streams(s, host);
1509 int lower_transport = ff_log2_tab[lower_transport_mask &
1510 ~(lower_transport_mask - 1)];
1512 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1513 rt->server_type == RTSP_SERVER_REAL ?
1514 real_challenge : NULL);
1517 lower_transport_mask &= ~(1 << lower_transport);
1518 if (lower_transport_mask == 0 && err == 1) {
1519 err = AVERROR(EPROTONOSUPPORT);
1524 rt->lower_transport_mask = lower_transport_mask;
1525 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1526 rt->state = RTSP_STATE_IDLE;
1527 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1530 ff_rtsp_close_streams(s);
1531 ff_rtsp_close_connections(s);
1532 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1533 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1534 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1542 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1545 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1546 uint8_t *buf, int buf_size, int64_t wait_end)
1548 RTSPState *rt = s->priv_data;
1549 RTSPStream *rtsp_st;
1550 int n, i, ret, tcp_fd, timeout_cnt = 0;
1552 struct pollfd *p = rt->p;
1555 if (url_interrupt_cb())
1556 return AVERROR_EXIT;
1557 if (wait_end && wait_end - av_gettime() < 0)
1558 return AVERROR(EAGAIN);
1561 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1562 p[max_p].fd = tcp_fd;
1563 p[max_p++].events = POLLIN;
1567 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1568 rtsp_st = rt->rtsp_streams[i];
1569 if (rtsp_st->rtp_handle) {
1570 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1571 p[max_p++].events = POLLIN;
1572 p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1573 p[max_p++].events = POLLIN;
1576 n = poll(p, max_p, POLL_TIMEOUT_MS);
1578 int j = 1 - (tcp_fd == -1);
1580 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1581 rtsp_st = rt->rtsp_streams[i];
1582 if (rtsp_st->rtp_handle) {
1583 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1584 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1586 *prtsp_st = rtsp_st;
1593 #if CONFIG_RTSP_DEMUXER
1594 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1595 RTSPMessageHeader reply;
1597 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1600 /* XXX: parse message */
1601 if (rt->state != RTSP_STATE_STREAMING)
1605 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1606 return AVERROR(ETIMEDOUT);
1607 } else if (n < 0 && errno != EINTR)
1608 return AVERROR(errno);
1612 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1614 RTSPState *rt = s->priv_data;
1616 RTSPStream *rtsp_st, *first_queue_st = NULL;
1617 int64_t wait_end = 0;
1619 if (rt->nb_byes == rt->nb_rtsp_streams)
1622 /* get next frames from the same RTP packet */
1623 if (rt->cur_transport_priv) {
1624 if (rt->transport == RTSP_TRANSPORT_RDT) {
1625 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1627 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1629 rt->cur_transport_priv = NULL;
1631 } else if (ret == 1) {
1634 rt->cur_transport_priv = NULL;
1637 if (rt->transport == RTSP_TRANSPORT_RTP) {
1639 int64_t first_queue_time = 0;
1640 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1641 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1645 queue_time = ff_rtp_queued_packet_time(rtpctx);
1646 if (queue_time && (queue_time - first_queue_time < 0 ||
1647 !first_queue_time)) {
1648 first_queue_time = queue_time;
1649 first_queue_st = rt->rtsp_streams[i];
1652 if (first_queue_time)
1653 wait_end = first_queue_time + s->max_delay;
1656 /* read next RTP packet */
1659 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1661 return AVERROR(ENOMEM);
1664 switch(rt->lower_transport) {
1666 #if CONFIG_RTSP_DEMUXER
1667 case RTSP_LOWER_TRANSPORT_TCP:
1668 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1671 case RTSP_LOWER_TRANSPORT_UDP:
1672 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1673 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1674 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1675 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1678 if (len == AVERROR(EAGAIN) && first_queue_st &&
1679 rt->transport == RTSP_TRANSPORT_RTP) {
1680 rtsp_st = first_queue_st;
1681 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1688 if (rt->transport == RTSP_TRANSPORT_RDT) {
1689 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1691 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1693 /* Either bad packet, or a RTCP packet. Check if the
1694 * first_rtcp_ntp_time field was initialized. */
1695 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1696 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1697 /* first_rtcp_ntp_time has been initialized for this stream,
1698 * copy the same value to all other uninitialized streams,
1699 * in order to map their timestamp origin to the same ntp time
1702 AVStream *st = NULL;
1703 if (rtsp_st->stream_index >= 0)
1704 st = s->streams[rtsp_st->stream_index];
1705 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1706 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1707 AVStream *st2 = NULL;
1708 if (rt->rtsp_streams[i]->stream_index >= 0)
1709 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1710 if (rtpctx2 && st && st2 &&
1711 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1712 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1713 rtpctx2->rtcp_ts_offset = av_rescale_q(
1714 rtpctx->rtcp_ts_offset, st->time_base,
1719 if (ret == -RTCP_BYE) {
1722 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1723 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1725 if (rt->nb_byes == rt->nb_rtsp_streams)
1734 /* more packets may follow, so we save the RTP context */
1735 rt->cur_transport_priv = rtsp_st->transport_priv;
1739 #endif /* CONFIG_RTPDEC */
1741 #if CONFIG_SDP_DEMUXER
1742 static int sdp_probe(AVProbeData *p1)
1744 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1746 /* we look for a line beginning "c=IN IP" */
1747 while (p < p_end && *p != '\0') {
1748 if (p + sizeof("c=IN IP") - 1 < p_end &&
1749 av_strstart(p, "c=IN IP", NULL))
1750 return AVPROBE_SCORE_MAX / 2;
1752 while (p < p_end - 1 && *p != '\n') p++;
1761 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1763 RTSPState *rt = s->priv_data;
1764 RTSPStream *rtsp_st;
1769 if (!ff_network_init())
1770 return AVERROR(EIO);
1772 /* read the whole sdp file */
1773 /* XXX: better loading */
1774 content = av_malloc(SDP_MAX_SIZE);
1775 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1778 return AVERROR_INVALIDDATA;
1780 content[size] ='\0';
1782 err = 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 (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE) < 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 ff_sdp_demuxer = {
1820 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1821 .priv_data_size = sizeof(RTSPState),
1822 .read_probe = sdp_probe,
1823 .read_header = sdp_read_header,
1824 .read_packet = ff_rtsp_fetch_packet,
1825 .read_close = sdp_read_close,
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 = ffurl_open(&in, s->filename, AVIO_FLAG_READ);
1858 ret = ffurl_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(ffurl_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 ffio_init_context(&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 ff_rtp_demuxer = {
1924 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
1925 .priv_data_size = sizeof(RTSPState),
1926 .read_probe = rtp_probe,
1927 .read_header = rtp_read_header,
1928 .read_packet = ff_rtsp_fetch_packet,
1929 .read_close = sdp_read_close,
1930 .flags = AVFMT_NOFILE,
1932 #endif /* CONFIG_RTP_DEMUXER */