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"
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 poll, in ms,
48 * and read_packet(), in seconds */
49 #define POLL_TIMEOUT_MS 100
50 #define READ_PACKET_TIMEOUT_S 10
51 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_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 av_dlog(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 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
277 rtsp_st->sdp_ip = sdp_ip;
278 rtsp_st->sdp_ttl = ttl;
282 av_metadata_set2(&s->metadata, "title", p, 0);
285 if (s->nb_streams == 0) {
286 av_metadata_set2(&s->metadata, "comment", p, 0);
293 get_word(st_type, sizeof(st_type), &p);
294 if (!strcmp(st_type, "audio")) {
295 codec_type = AVMEDIA_TYPE_AUDIO;
296 } else if (!strcmp(st_type, "video")) {
297 codec_type = AVMEDIA_TYPE_VIDEO;
298 } else if (!strcmp(st_type, "application")) {
299 codec_type = AVMEDIA_TYPE_DATA;
304 rtsp_st = av_mallocz(sizeof(RTSPStream));
307 rtsp_st->stream_index = -1;
308 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
310 rtsp_st->sdp_ip = s1->default_ip;
311 rtsp_st->sdp_ttl = s1->default_ttl;
313 get_word(buf1, sizeof(buf1), &p); /* port */
314 rtsp_st->sdp_port = atoi(buf1);
316 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
318 /* XXX: handle list of formats */
319 get_word(buf1, sizeof(buf1), &p); /* format list */
320 rtsp_st->sdp_payload_type = atoi(buf1);
322 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
323 /* no corresponding stream */
325 st = av_new_stream(s, 0);
328 rtsp_st->stream_index = st->index;
329 st->codec->codec_type = codec_type;
330 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
331 RTPDynamicProtocolHandler *handler;
332 /* if standard payload type, we can find the codec right now */
333 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
334 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
335 st->codec->sample_rate > 0)
336 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
337 /* Even static payload types may need a custom depacketizer */
338 handler = ff_rtp_handler_find_by_id(
339 rtsp_st->sdp_payload_type, st->codec->codec_type);
340 init_rtp_handler(handler, rtsp_st, st->codec);
343 /* put a default control url */
344 av_strlcpy(rtsp_st->control_url, rt->control_uri,
345 sizeof(rtsp_st->control_url));
348 if (av_strstart(p, "control:", &p)) {
349 if (s->nb_streams == 0) {
350 if (!strncmp(p, "rtsp://", 7))
351 av_strlcpy(rt->control_uri, p,
352 sizeof(rt->control_uri));
355 /* get the control url */
356 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
358 /* XXX: may need to add full url resolution */
359 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
361 if (proto[0] == '\0') {
362 /* relative control URL */
363 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
364 av_strlcat(rtsp_st->control_url, "/",
365 sizeof(rtsp_st->control_url));
366 av_strlcat(rtsp_st->control_url, p,
367 sizeof(rtsp_st->control_url));
369 av_strlcpy(rtsp_st->control_url, p,
370 sizeof(rtsp_st->control_url));
372 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
373 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
374 get_word(buf1, sizeof(buf1), &p);
375 payload_type = atoi(buf1);
376 st = s->streams[s->nb_streams - 1];
377 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
378 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
379 } else if (av_strstart(p, "fmtp:", &p) ||
380 av_strstart(p, "framesize:", &p)) {
381 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
382 // let dynamic protocol handlers have a stab at the line.
383 get_word(buf1, sizeof(buf1), &p);
384 payload_type = atoi(buf1);
385 for (i = 0; i < rt->nb_rtsp_streams; i++) {
386 rtsp_st = rt->rtsp_streams[i];
387 if (rtsp_st->sdp_payload_type == payload_type &&
388 rtsp_st->dynamic_handler &&
389 rtsp_st->dynamic_handler->parse_sdp_a_line)
390 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
391 rtsp_st->dynamic_protocol_context, buf);
393 } else if (av_strstart(p, "range:", &p)) {
396 // this is so that seeking on a streamed file can work.
397 rtsp_parse_range_npt(p, &start, &end);
398 s->start_time = start;
399 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
400 s->duration = (end == AV_NOPTS_VALUE) ?
401 AV_NOPTS_VALUE : end - start;
402 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
404 rt->transport = RTSP_TRANSPORT_RDT;
405 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
407 st = s->streams[s->nb_streams - 1];
408 st->codec->sample_rate = atoi(p);
410 if (rt->server_type == RTSP_SERVER_WMS)
411 ff_wms_parse_sdp_a_line(s, p);
412 if (s->nb_streams > 0) {
413 if (rt->server_type == RTSP_SERVER_REAL)
414 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
416 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
417 if (rtsp_st->dynamic_handler &&
418 rtsp_st->dynamic_handler->parse_sdp_a_line)
419 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
421 rtsp_st->dynamic_protocol_context, buf);
429 * Parse the sdp description and allocate the rtp streams and the
430 * pollfd array used for udp ones.
433 int ff_sdp_parse(AVFormatContext *s, const char *content)
435 RTSPState *rt = s->priv_data;
438 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
439 * contain long SDP lines containing complete ASF Headers (several
440 * kB) or arrays of MDPR (RM stream descriptor) headers plus
441 * "rulebooks" describing their properties. Therefore, the SDP line
444 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
445 * in rtpdec_xiph.c. */
447 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
449 memset(s1, 0, sizeof(SDPParseState));
452 p += strspn(p, SPACE_CHARS);
460 /* get the content */
462 while (*p != '\n' && *p != '\r' && *p != '\0') {
463 if ((q - buf) < sizeof(buf) - 1)
468 sdp_parse_line(s, s1, letter, buf);
470 while (*p != '\n' && *p != '\0')
475 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
476 if (!rt->p) return AVERROR(ENOMEM);
479 #endif /* CONFIG_RTPDEC */
481 void ff_rtsp_undo_setup(AVFormatContext *s)
483 RTSPState *rt = s->priv_data;
486 for (i = 0; i < rt->nb_rtsp_streams; i++) {
487 RTSPStream *rtsp_st = rt->rtsp_streams[i];
490 if (rtsp_st->transport_priv) {
492 AVFormatContext *rtpctx = rtsp_st->transport_priv;
493 av_write_trailer(rtpctx);
494 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
496 url_close_dyn_buf(rtpctx->pb, &ptr);
499 url_fclose(rtpctx->pb);
501 av_metadata_free(&rtpctx->streams[0]->metadata);
502 av_metadata_free(&rtpctx->metadata);
503 av_free(rtpctx->streams[0]->codec->extradata);
504 av_free(rtpctx->streams[0]->codec);
505 av_free(rtpctx->streams[0]->info);
506 av_free(rtpctx->streams[0]);
508 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
509 ff_rdt_parse_close(rtsp_st->transport_priv);
510 else if (CONFIG_RTPDEC)
511 rtp_parse_close(rtsp_st->transport_priv);
513 rtsp_st->transport_priv = NULL;
514 if (rtsp_st->rtp_handle)
515 url_close(rtsp_st->rtp_handle);
516 rtsp_st->rtp_handle = NULL;
520 /* close and free RTSP streams */
521 void ff_rtsp_close_streams(AVFormatContext *s)
523 RTSPState *rt = s->priv_data;
527 ff_rtsp_undo_setup(s);
528 for (i = 0; i < rt->nb_rtsp_streams; i++) {
529 rtsp_st = rt->rtsp_streams[i];
531 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
532 rtsp_st->dynamic_handler->close(
533 rtsp_st->dynamic_protocol_context);
537 av_free(rt->rtsp_streams);
539 av_close_input_stream (rt->asf_ctx);
543 av_free(rt->recvbuf);
546 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
548 RTSPState *rt = s->priv_data;
551 /* open the RTP context */
552 if (rtsp_st->stream_index >= 0)
553 st = s->streams[rtsp_st->stream_index];
555 s->ctx_flags |= AVFMTCTX_NOHEADER;
557 if (s->oformat && CONFIG_RTSP_MUXER) {
558 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
560 RTSP_TCP_MAX_PACKET_SIZE);
561 /* Ownership of rtp_handle is passed to the rtp mux context */
562 rtsp_st->rtp_handle = NULL;
563 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
564 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
565 rtsp_st->dynamic_protocol_context,
566 rtsp_st->dynamic_handler);
567 else if (CONFIG_RTPDEC)
568 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
569 rtsp_st->sdp_payload_type,
570 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
571 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
573 if (!rtsp_st->transport_priv) {
574 return AVERROR(ENOMEM);
575 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
576 if (rtsp_st->dynamic_handler) {
577 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
578 rtsp_st->dynamic_protocol_context,
579 rtsp_st->dynamic_handler);
586 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
587 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
593 p += strspn(p, SPACE_CHARS);
594 v = strtol(p, (char **)&p, 10);
598 v = strtol(p, (char **)&p, 10);
607 /* XXX: only one transport specification is parsed */
608 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
610 char transport_protocol[16];
612 char lower_transport[16];
614 RTSPTransportField *th;
617 reply->nb_transports = 0;
620 p += strspn(p, SPACE_CHARS);
624 th = &reply->transports[reply->nb_transports];
626 get_word_sep(transport_protocol, sizeof(transport_protocol),
628 if (!strcasecmp (transport_protocol, "rtp")) {
629 get_word_sep(profile, sizeof(profile), "/;,", &p);
630 lower_transport[0] = '\0';
631 /* rtp/avp/<protocol> */
633 get_word_sep(lower_transport, sizeof(lower_transport),
636 th->transport = RTSP_TRANSPORT_RTP;
637 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
638 !strcasecmp (transport_protocol, "x-real-rdt")) {
639 /* x-pn-tng/<protocol> */
640 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
642 th->transport = RTSP_TRANSPORT_RDT;
644 if (!strcasecmp(lower_transport, "TCP"))
645 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
647 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
651 /* get each parameter */
652 while (*p != '\0' && *p != ',') {
653 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
654 if (!strcmp(parameter, "port")) {
657 rtsp_parse_range(&th->port_min, &th->port_max, &p);
659 } else if (!strcmp(parameter, "client_port")) {
662 rtsp_parse_range(&th->client_port_min,
663 &th->client_port_max, &p);
665 } else if (!strcmp(parameter, "server_port")) {
668 rtsp_parse_range(&th->server_port_min,
669 &th->server_port_max, &p);
671 } else if (!strcmp(parameter, "interleaved")) {
674 rtsp_parse_range(&th->interleaved_min,
675 &th->interleaved_max, &p);
677 } else if (!strcmp(parameter, "multicast")) {
678 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
679 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
680 } else if (!strcmp(parameter, "ttl")) {
683 th->ttl = strtol(p, (char **)&p, 10);
685 } else if (!strcmp(parameter, "destination")) {
688 get_word_sep(buf, sizeof(buf), ";,", &p);
689 get_sockaddr(buf, &th->destination);
691 } else if (!strcmp(parameter, "source")) {
694 get_word_sep(buf, sizeof(buf), ";,", &p);
695 av_strlcpy(th->source, buf, sizeof(th->source));
699 while (*p != ';' && *p != '\0' && *p != ',')
707 reply->nb_transports++;
711 static void handle_rtp_info(RTSPState *rt, const char *url,
712 uint32_t seq, uint32_t rtptime)
715 if (!rtptime || !url[0])
717 if (rt->transport != RTSP_TRANSPORT_RTP)
719 for (i = 0; i < rt->nb_rtsp_streams; i++) {
720 RTSPStream *rtsp_st = rt->rtsp_streams[i];
721 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
724 if (!strcmp(rtsp_st->control_url, url)) {
725 rtpctx->base_timestamp = rtptime;
731 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
734 char key[20], value[1024], url[1024] = "";
735 uint32_t seq = 0, rtptime = 0;
738 p += strspn(p, SPACE_CHARS);
741 get_word_sep(key, sizeof(key), "=", &p);
745 get_word_sep(value, sizeof(value), ";, ", &p);
747 if (!strcmp(key, "url"))
748 av_strlcpy(url, value, sizeof(url));
749 else if (!strcmp(key, "seq"))
750 seq = strtol(value, NULL, 10);
751 else if (!strcmp(key, "rtptime"))
752 rtptime = strtol(value, NULL, 10);
754 handle_rtp_info(rt, url, seq, rtptime);
763 handle_rtp_info(rt, url, seq, rtptime);
766 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
767 RTSPState *rt, const char *method)
771 /* NOTE: we do case independent match for broken servers */
773 if (av_stristart(p, "Session:", &p)) {
775 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
776 if (av_stristart(p, ";timeout=", &p) &&
777 (t = strtol(p, NULL, 10)) > 0) {
780 } else if (av_stristart(p, "Content-Length:", &p)) {
781 reply->content_length = strtol(p, NULL, 10);
782 } else if (av_stristart(p, "Transport:", &p)) {
783 rtsp_parse_transport(reply, p);
784 } else if (av_stristart(p, "CSeq:", &p)) {
785 reply->seq = strtol(p, NULL, 10);
786 } else if (av_stristart(p, "Range:", &p)) {
787 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
788 } else if (av_stristart(p, "RealChallenge1:", &p)) {
789 p += strspn(p, SPACE_CHARS);
790 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
791 } else if (av_stristart(p, "Server:", &p)) {
792 p += strspn(p, SPACE_CHARS);
793 av_strlcpy(reply->server, p, sizeof(reply->server));
794 } else if (av_stristart(p, "Notice:", &p) ||
795 av_stristart(p, "X-Notice:", &p)) {
796 reply->notice = strtol(p, NULL, 10);
797 } else if (av_stristart(p, "Location:", &p)) {
798 p += strspn(p, SPACE_CHARS);
799 av_strlcpy(reply->location, p , sizeof(reply->location));
800 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
801 p += strspn(p, SPACE_CHARS);
802 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
803 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
804 p += strspn(p, SPACE_CHARS);
805 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
806 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
807 p += strspn(p, SPACE_CHARS);
808 if (method && !strcmp(method, "DESCRIBE"))
809 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
810 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
811 p += strspn(p, SPACE_CHARS);
812 if (method && !strcmp(method, "PLAY"))
813 rtsp_parse_rtp_info(rt, p);
817 /* skip a RTP/TCP interleaved packet */
818 void ff_rtsp_skip_packet(AVFormatContext *s)
820 RTSPState *rt = s->priv_data;
824 ret = url_read_complete(rt->rtsp_hd, buf, 3);
827 len = AV_RB16(buf + 1);
829 av_dlog(s, "skipping RTP packet len=%d\n", len);
834 if (len1 > sizeof(buf))
836 ret = url_read_complete(rt->rtsp_hd, buf, len1);
843 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
844 unsigned char **content_ptr,
845 int return_on_interleaved_data, const char *method)
847 RTSPState *rt = s->priv_data;
848 char buf[4096], buf1[1024], *q;
851 int ret, content_length, line_count = 0;
852 unsigned char *content = NULL;
854 memset(reply, 0, sizeof(*reply));
856 /* parse reply (XXX: use buffers) */
857 rt->last_reply[0] = '\0';
861 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
863 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
870 /* XXX: only parse it if first char on line ? */
871 if (return_on_interleaved_data) {
874 ff_rtsp_skip_packet(s);
875 } else if (ch != '\r') {
876 if ((q - buf) < sizeof(buf) - 1)
882 av_dlog(s, "line='%s'\n", buf);
884 /* test if last line */
888 if (line_count == 0) {
890 get_word(buf1, sizeof(buf1), &p);
891 get_word(buf1, sizeof(buf1), &p);
892 reply->status_code = atoi(buf1);
893 av_strlcpy(reply->reason, p, sizeof(reply->reason));
895 ff_rtsp_parse_line(reply, p, rt, method);
896 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
897 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
902 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
903 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
905 content_length = reply->content_length;
906 if (content_length > 0) {
907 /* leave some room for a trailing '\0' (useful for simple parsing) */
908 content = av_malloc(content_length + 1);
909 (void)url_read_complete(rt->rtsp_hd, content, content_length);
910 content[content_length] = '\0';
913 *content_ptr = content;
917 if (rt->seq != reply->seq) {
918 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
919 rt->seq, reply->seq);
923 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
924 reply->notice == 2104 /* Start-of-Stream Reached */ ||
925 reply->notice == 2306 /* Continuous Feed Terminated */) {
926 rt->state = RTSP_STATE_IDLE;
927 } else if (reply->notice >= 4400 && reply->notice < 5500) {
928 return AVERROR(EIO); /* data or server error */
929 } else if (reply->notice == 2401 /* Ticket Expired */ ||
930 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
931 return AVERROR(EPERM);
937 * Send a command to the RTSP server without waiting for the reply.
939 * @param s RTSP (de)muxer context
940 * @param method the method for the request
941 * @param url the target url for the request
942 * @param headers extra header lines to include in the request
943 * @param send_content if non-null, the data to send as request body content
944 * @param send_content_length the length of the send_content data, or 0 if
945 * send_content is null
947 * @return zero if success, nonzero otherwise
949 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
950 const char *method, const char *url,
952 const unsigned char *send_content,
953 int send_content_length)
955 RTSPState *rt = s->priv_data;
956 char buf[4096], *out_buf;
957 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
959 /* Add in RTSP headers */
962 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
964 av_strlcat(buf, headers, sizeof(buf));
965 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
966 if (rt->session_id[0] != '\0' && (!headers ||
967 !strstr(headers, "\nIf-Match:"))) {
968 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
971 char *str = ff_http_auth_create_response(&rt->auth_state,
972 rt->auth, url, method);
974 av_strlcat(buf, str, sizeof(buf));
977 if (send_content_length > 0 && send_content)
978 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
979 av_strlcat(buf, "\r\n", sizeof(buf));
981 /* base64 encode rtsp if tunneling */
982 if (rt->control_transport == RTSP_MODE_TUNNEL) {
983 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
987 av_dlog(s, "Sending:\n%s--\n", buf);
989 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
990 if (send_content_length > 0 && send_content) {
991 if (rt->control_transport == RTSP_MODE_TUNNEL) {
992 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
993 "with content data not supported\n");
994 return AVERROR_PATCHWELCOME;
996 url_write(rt->rtsp_hd_out, send_content, send_content_length);
998 rt->last_cmd_time = av_gettime();
1003 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1004 const char *url, const char *headers)
1006 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1009 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1010 const char *headers, RTSPMessageHeader *reply,
1011 unsigned char **content_ptr)
1013 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1014 content_ptr, NULL, 0);
1017 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1018 const char *method, const char *url,
1020 RTSPMessageHeader *reply,
1021 unsigned char **content_ptr,
1022 const unsigned char *send_content,
1023 int send_content_length)
1025 RTSPState *rt = s->priv_data;
1026 HTTPAuthType cur_auth_type;
1030 cur_auth_type = rt->auth_state.auth_type;
1031 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1033 send_content_length)))
1036 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1039 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1040 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1043 if (reply->status_code > 400){
1044 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1048 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1055 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1057 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1058 int lower_transport, const char *real_challenge)
1060 RTSPState *rt = s->priv_data;
1061 int rtx, j, i, err, interleave = 0;
1062 RTSPStream *rtsp_st;
1063 RTSPMessageHeader reply1, *reply = &reply1;
1065 const char *trans_pref;
1067 if (rt->transport == RTSP_TRANSPORT_RDT)
1068 trans_pref = "x-pn-tng";
1070 trans_pref = "RTP/AVP";
1072 /* default timeout: 1 minute */
1075 /* for each stream, make the setup request */
1076 /* XXX: we assume the same server is used for the control of each
1079 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1080 char transport[2048];
1083 * WMS serves all UDP data over a single connection, the RTX, which
1084 * isn't necessarily the first in the SDP but has to be the first
1085 * to be set up, else the second/third SETUP will fail with a 461.
1087 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1088 rt->server_type == RTSP_SERVER_WMS) {
1091 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1092 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1094 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1098 if (rtx == rt->nb_rtsp_streams)
1099 return -1; /* no RTX found */
1100 rtsp_st = rt->rtsp_streams[rtx];
1102 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1104 rtsp_st = rt->rtsp_streams[i];
1107 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1110 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1111 port = reply->transports[0].client_port_min;
1115 /* first try in specified port range */
1116 if (RTSP_RTP_PORT_MIN != 0) {
1117 while (j <= RTSP_RTP_PORT_MAX) {
1118 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1119 "?localport=%d", j);
1120 /* we will use two ports per rtp stream (rtp and rtcp) */
1122 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1128 /* then try on any port */
1129 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1130 err = AVERROR_INVALIDDATA;
1134 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1140 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1142 snprintf(transport, sizeof(transport) - 1,
1143 "%s/UDP;", trans_pref);
1144 if (rt->server_type != RTSP_SERVER_REAL)
1145 av_strlcat(transport, "unicast;", sizeof(transport));
1146 av_strlcatf(transport, sizeof(transport),
1147 "client_port=%d", port);
1148 if (rt->transport == RTSP_TRANSPORT_RTP &&
1149 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1150 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1154 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1155 /** For WMS streams, the application streams are only used for
1156 * UDP. When trying to set it up for TCP streams, the server
1157 * will return an error. Therefore, we skip those streams. */
1158 if (rt->server_type == RTSP_SERVER_WMS &&
1159 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1162 snprintf(transport, sizeof(transport) - 1,
1163 "%s/TCP;", trans_pref);
1164 if (rt->server_type == RTSP_SERVER_WMS)
1165 av_strlcat(transport, "unicast;", sizeof(transport));
1166 av_strlcatf(transport, sizeof(transport),
1167 "interleaved=%d-%d",
1168 interleave, interleave + 1);
1172 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1173 snprintf(transport, sizeof(transport) - 1,
1174 "%s/UDP;multicast", trans_pref);
1177 av_strlcat(transport, ";mode=receive", sizeof(transport));
1178 } else if (rt->server_type == RTSP_SERVER_REAL ||
1179 rt->server_type == RTSP_SERVER_WMS)
1180 av_strlcat(transport, ";mode=play", sizeof(transport));
1181 snprintf(cmd, sizeof(cmd),
1182 "Transport: %s\r\n",
1184 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1185 char real_res[41], real_csum[9];
1186 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1188 av_strlcatf(cmd, sizeof(cmd),
1190 "RealChallenge2: %s, sd=%s\r\n",
1191 rt->session_id, real_res, real_csum);
1193 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1194 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1197 } else if (reply->status_code != RTSP_STATUS_OK ||
1198 reply->nb_transports != 1) {
1199 err = AVERROR_INVALIDDATA;
1203 /* XXX: same protocol for all streams is required */
1205 if (reply->transports[0].lower_transport != rt->lower_transport ||
1206 reply->transports[0].transport != rt->transport) {
1207 err = AVERROR_INVALIDDATA;
1211 rt->lower_transport = reply->transports[0].lower_transport;
1212 rt->transport = reply->transports[0].transport;
1215 /* Fail if the server responded with another lower transport mode
1216 * than what we requested. */
1217 if (reply->transports[0].lower_transport != lower_transport) {
1218 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1219 err = AVERROR_INVALIDDATA;
1223 switch(reply->transports[0].lower_transport) {
1224 case RTSP_LOWER_TRANSPORT_TCP:
1225 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1226 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1229 case RTSP_LOWER_TRANSPORT_UDP: {
1230 char url[1024], options[30] = "";
1232 if (rt->filter_source)
1233 av_strlcpy(options, "?connect=1", sizeof(options));
1234 /* Use source address if specified */
1235 if (reply->transports[0].source[0]) {
1236 ff_url_join(url, sizeof(url), "rtp", NULL,
1237 reply->transports[0].source,
1238 reply->transports[0].server_port_min, options);
1240 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1241 reply->transports[0].server_port_min, options);
1243 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1244 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1245 err = AVERROR_INVALIDDATA;
1248 /* Try to initialize the connection state in a
1249 * potential NAT router by sending dummy packets.
1250 * RTP/RTCP dummy packets are used for RDT, too.
1252 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1254 rtp_send_punch_packets(rtsp_st->rtp_handle);
1257 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1258 char url[1024], namebuf[50];
1259 struct sockaddr_storage addr;
1262 if (reply->transports[0].destination.ss_family) {
1263 addr = reply->transports[0].destination;
1264 port = reply->transports[0].port_min;
1265 ttl = reply->transports[0].ttl;
1267 addr = rtsp_st->sdp_ip;
1268 port = rtsp_st->sdp_port;
1269 ttl = rtsp_st->sdp_ttl;
1271 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1272 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1273 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1274 port, "?ttl=%d", ttl);
1275 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1276 err = AVERROR_INVALIDDATA;
1283 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1287 if (reply->timeout > 0)
1288 rt->timeout = reply->timeout;
1290 if (rt->server_type == RTSP_SERVER_REAL)
1291 rt->need_subscription = 1;
1296 ff_rtsp_undo_setup(s);
1300 void ff_rtsp_close_connections(AVFormatContext *s)
1302 RTSPState *rt = s->priv_data;
1303 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1304 url_close(rt->rtsp_hd);
1305 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1308 int ff_rtsp_connect(AVFormatContext *s)
1310 RTSPState *rt = s->priv_data;
1311 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1312 char *option_list, *option, *filename;
1313 int port, err, tcp_fd;
1314 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1315 int lower_transport_mask = 0;
1316 char real_challenge[64] = "";
1317 struct sockaddr_storage peer;
1318 socklen_t peer_len = sizeof(peer);
1320 if (!ff_network_init())
1321 return AVERROR(EIO);
1323 rt->control_transport = RTSP_MODE_PLAIN;
1324 /* extract hostname and port */
1325 av_url_split(NULL, 0, auth, sizeof(auth),
1326 host, sizeof(host), &port, path, sizeof(path), s->filename);
1328 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1331 port = RTSP_DEFAULT_PORT;
1333 /* search for options */
1334 option_list = strrchr(path, '?');
1336 /* Strip out the RTSP specific options, write out the rest of
1337 * the options back into the same string. */
1338 filename = option_list;
1339 while (option_list) {
1340 /* move the option pointer */
1341 option = ++option_list;
1342 option_list = strchr(option_list, '&');
1346 /* handle the options */
1347 if (!strcmp(option, "udp")) {
1348 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1349 } else if (!strcmp(option, "multicast")) {
1350 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1351 } else if (!strcmp(option, "tcp")) {
1352 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1353 } else if(!strcmp(option, "http")) {
1354 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1355 rt->control_transport = RTSP_MODE_TUNNEL;
1356 } else if (!strcmp(option, "filter_src")) {
1357 rt->filter_source = 1;
1359 /* Write options back into the buffer, using memmove instead
1360 * of strcpy since the strings may overlap. */
1361 int len = strlen(option);
1362 memmove(++filename, option, len);
1364 if (option_list) *filename = '&';
1370 if (!lower_transport_mask)
1371 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1374 /* Only UDP or TCP - UDP multicast isn't supported. */
1375 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1376 (1 << RTSP_LOWER_TRANSPORT_TCP);
1377 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1378 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1379 "only UDP and TCP are supported for output.\n");
1380 err = AVERROR(EINVAL);
1385 /* Construct the URI used in request; this is similar to s->filename,
1386 * but with authentication credentials removed and RTSP specific options
1388 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1389 host, port, "%s", path);
1391 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1392 /* set up initial handshake for tunneling */
1393 char httpname[1024];
1394 char sessioncookie[17];
1397 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1398 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1399 av_get_random_seed(), av_get_random_seed());
1402 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1407 /* generate GET headers */
1408 snprintf(headers, sizeof(headers),
1409 "x-sessioncookie: %s\r\n"
1410 "Accept: application/x-rtsp-tunnelled\r\n"
1411 "Pragma: no-cache\r\n"
1412 "Cache-Control: no-cache\r\n",
1414 ff_http_set_headers(rt->rtsp_hd, headers);
1416 /* complete the connection */
1417 if (url_connect(rt->rtsp_hd)) {
1423 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1428 /* generate POST headers */
1429 snprintf(headers, sizeof(headers),
1430 "x-sessioncookie: %s\r\n"
1431 "Content-Type: application/x-rtsp-tunnelled\r\n"
1432 "Pragma: no-cache\r\n"
1433 "Cache-Control: no-cache\r\n"
1434 "Content-Length: 32767\r\n"
1435 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1437 ff_http_set_headers(rt->rtsp_hd_out, headers);
1438 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1440 /* Initialize the authentication state for the POST session. The HTTP
1441 * protocol implementation doesn't properly handle multi-pass
1442 * authentication for POST requests, since it would require one of
1444 * - implementing Expect: 100-continue, which many HTTP servers
1445 * don't support anyway, even less the RTSP servers that do HTTP
1447 * - sending the whole POST data until getting a 401 reply specifying
1448 * what authentication method to use, then resending all that data
1449 * - waiting for potential 401 replies directly after sending the
1450 * POST header (waiting for some unspecified time)
1451 * Therefore, we copy the full auth state, which works for both basic
1452 * and digest. (For digest, we would have to synchronize the nonce
1453 * count variable between the two sessions, if we'd do more requests
1454 * with the original session, though.)
1456 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1458 /* complete the connection */
1459 if (url_connect(rt->rtsp_hd_out)) {
1464 /* open the tcp connection */
1465 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1466 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1470 rt->rtsp_hd_out = rt->rtsp_hd;
1474 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1475 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1476 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1477 NULL, 0, NI_NUMERICHOST);
1480 /* request options supported by the server; this also detects server
1482 for (rt->server_type = RTSP_SERVER_RTP;;) {
1484 if (rt->server_type == RTSP_SERVER_REAL)
1487 * The following entries are required for proper
1488 * streaming from a Realmedia server. They are
1489 * interdependent in some way although we currently
1490 * don't quite understand how. Values were copied
1491 * from mplayer SVN r23589.
1492 * @param CompanyID is a 16-byte ID in base64
1493 * @param ClientChallenge is a 16-byte ID in hex
1495 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1496 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1497 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1498 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1500 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1501 if (reply->status_code != RTSP_STATUS_OK) {
1502 err = AVERROR_INVALIDDATA;
1506 /* detect server type if not standard-compliant RTP */
1507 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1508 rt->server_type = RTSP_SERVER_REAL;
1510 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1511 rt->server_type = RTSP_SERVER_WMS;
1512 } else if (rt->server_type == RTSP_SERVER_REAL)
1513 strcpy(real_challenge, reply->real_challenge);
1517 if (s->iformat && CONFIG_RTSP_DEMUXER)
1518 err = ff_rtsp_setup_input_streams(s, reply);
1519 else if (CONFIG_RTSP_MUXER)
1520 err = ff_rtsp_setup_output_streams(s, host);
1525 int lower_transport = ff_log2_tab[lower_transport_mask &
1526 ~(lower_transport_mask - 1)];
1528 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1529 rt->server_type == RTSP_SERVER_REAL ?
1530 real_challenge : NULL);
1533 lower_transport_mask &= ~(1 << lower_transport);
1534 if (lower_transport_mask == 0 && err == 1) {
1535 err = FF_NETERROR(EPROTONOSUPPORT);
1540 rt->lower_transport_mask = lower_transport_mask;
1541 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1542 rt->state = RTSP_STATE_IDLE;
1543 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1546 ff_rtsp_close_streams(s);
1547 ff_rtsp_close_connections(s);
1548 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1549 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1550 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1558 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1561 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1562 uint8_t *buf, int buf_size, int64_t wait_end)
1564 RTSPState *rt = s->priv_data;
1565 RTSPStream *rtsp_st;
1566 int n, i, ret, tcp_fd, timeout_cnt = 0;
1568 struct pollfd *p = rt->p;
1571 if (url_interrupt_cb())
1572 return AVERROR(EINTR);
1573 if (wait_end && wait_end - av_gettime() < 0)
1574 return AVERROR(EAGAIN);
1577 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1578 p[max_p].fd = tcp_fd;
1579 p[max_p++].events = POLLIN;
1583 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1584 rtsp_st = rt->rtsp_streams[i];
1585 if (rtsp_st->rtp_handle) {
1586 p[max_p].fd = url_get_file_handle(rtsp_st->rtp_handle);
1587 p[max_p++].events = POLLIN;
1588 p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1589 p[max_p++].events = POLLIN;
1592 n = poll(p, max_p, POLL_TIMEOUT_MS);
1594 int j = 1 - (tcp_fd == -1);
1596 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1597 rtsp_st = rt->rtsp_streams[i];
1598 if (rtsp_st->rtp_handle) {
1599 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1600 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1602 *prtsp_st = rtsp_st;
1609 #if CONFIG_RTSP_DEMUXER
1610 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1611 RTSPMessageHeader reply;
1613 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1616 /* XXX: parse message */
1617 if (rt->state != RTSP_STATE_STREAMING)
1621 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1622 return FF_NETERROR(ETIMEDOUT);
1623 } else if (n < 0 && errno != EINTR)
1624 return AVERROR(errno);
1628 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1630 RTSPState *rt = s->priv_data;
1632 RTSPStream *rtsp_st, *first_queue_st = NULL;
1633 int64_t wait_end = 0;
1635 if (rt->nb_byes == rt->nb_rtsp_streams)
1638 /* get next frames from the same RTP packet */
1639 if (rt->cur_transport_priv) {
1640 if (rt->transport == RTSP_TRANSPORT_RDT) {
1641 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1643 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1645 rt->cur_transport_priv = NULL;
1647 } else if (ret == 1) {
1650 rt->cur_transport_priv = NULL;
1653 if (rt->transport == RTSP_TRANSPORT_RTP) {
1655 int64_t first_queue_time = 0;
1656 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1657 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1661 queue_time = ff_rtp_queued_packet_time(rtpctx);
1662 if (queue_time && (queue_time - first_queue_time < 0 ||
1663 !first_queue_time)) {
1664 first_queue_time = queue_time;
1665 first_queue_st = rt->rtsp_streams[i];
1668 if (first_queue_time)
1669 wait_end = first_queue_time + s->max_delay;
1672 /* read next RTP packet */
1675 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1677 return AVERROR(ENOMEM);
1680 switch(rt->lower_transport) {
1682 #if CONFIG_RTSP_DEMUXER
1683 case RTSP_LOWER_TRANSPORT_TCP:
1684 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1687 case RTSP_LOWER_TRANSPORT_UDP:
1688 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1689 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1690 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1691 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1694 if (len == AVERROR(EAGAIN) && first_queue_st &&
1695 rt->transport == RTSP_TRANSPORT_RTP) {
1696 rtsp_st = first_queue_st;
1697 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1704 if (rt->transport == RTSP_TRANSPORT_RDT) {
1705 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1707 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1709 /* Either bad packet, or a RTCP packet. Check if the
1710 * first_rtcp_ntp_time field was initialized. */
1711 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1712 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1713 /* first_rtcp_ntp_time has been initialized for this stream,
1714 * copy the same value to all other uninitialized streams,
1715 * in order to map their timestamp origin to the same ntp time
1718 AVStream *st = NULL;
1719 if (rtsp_st->stream_index >= 0)
1720 st = s->streams[rtsp_st->stream_index];
1721 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1722 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1723 AVStream *st2 = NULL;
1724 if (rt->rtsp_streams[i]->stream_index >= 0)
1725 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1726 if (rtpctx2 && st && st2 &&
1727 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1728 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1729 rtpctx2->rtcp_ts_offset = av_rescale_q(
1730 rtpctx->rtcp_ts_offset, st->time_base,
1735 if (ret == -RTCP_BYE) {
1738 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1739 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1741 if (rt->nb_byes == rt->nb_rtsp_streams)
1750 /* more packets may follow, so we save the RTP context */
1751 rt->cur_transport_priv = rtsp_st->transport_priv;
1755 #endif /* CONFIG_RTPDEC */
1757 #if CONFIG_SDP_DEMUXER
1758 static int sdp_probe(AVProbeData *p1)
1760 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1762 /* we look for a line beginning "c=IN IP" */
1763 while (p < p_end && *p != '\0') {
1764 if (p + sizeof("c=IN IP") - 1 < p_end &&
1765 av_strstart(p, "c=IN IP", NULL))
1766 return AVPROBE_SCORE_MAX / 2;
1768 while (p < p_end - 1 && *p != '\n') p++;
1777 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1779 RTSPState *rt = s->priv_data;
1780 RTSPStream *rtsp_st;
1785 if (!ff_network_init())
1786 return AVERROR(EIO);
1788 /* read the whole sdp file */
1789 /* XXX: better loading */
1790 content = av_malloc(SDP_MAX_SIZE);
1791 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1794 return AVERROR_INVALIDDATA;
1796 content[size] ='\0';
1798 err = ff_sdp_parse(s, content);
1802 /* open each RTP stream */
1803 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1805 rtsp_st = rt->rtsp_streams[i];
1807 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1808 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1809 ff_url_join(url, sizeof(url), "rtp", NULL,
1810 namebuf, rtsp_st->sdp_port,
1811 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1813 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1814 err = AVERROR_INVALIDDATA;
1817 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1822 ff_rtsp_close_streams(s);
1827 static int sdp_read_close(AVFormatContext *s)
1829 ff_rtsp_close_streams(s);
1834 AVInputFormat ff_sdp_demuxer = {
1836 NULL_IF_CONFIG_SMALL("SDP"),
1840 ff_rtsp_fetch_packet,
1843 #endif /* CONFIG_SDP_DEMUXER */
1845 #if CONFIG_RTP_DEMUXER
1846 static int rtp_probe(AVProbeData *p)
1848 if (av_strstart(p->filename, "rtp:", NULL))
1849 return AVPROBE_SCORE_MAX;
1853 static int rtp_read_header(AVFormatContext *s,
1854 AVFormatParameters *ap)
1856 uint8_t recvbuf[1500];
1857 char host[500], sdp[500];
1859 URLContext* in = NULL;
1861 AVCodecContext codec;
1862 struct sockaddr_storage addr;
1864 socklen_t addrlen = sizeof(addr);
1866 if (!ff_network_init())
1867 return AVERROR(EIO);
1869 ret = url_open(&in, s->filename, URL_RDONLY);
1874 ret = url_read(in, recvbuf, sizeof(recvbuf));
1875 if (ret == AVERROR(EAGAIN))
1880 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1884 if ((recvbuf[0] & 0xc0) != 0x80) {
1885 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1890 payload_type = recvbuf[1] & 0x7f;
1893 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1897 memset(&codec, 0, sizeof(codec));
1898 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1899 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1900 "without an SDP file describing it\n",
1904 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1905 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1906 "properly you need an SDP file "
1910 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1911 NULL, 0, s->filename);
1913 snprintf(sdp, sizeof(sdp),
1914 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1915 addr.ss_family == AF_INET ? 4 : 6, host,
1916 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1917 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1918 port, payload_type);
1919 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1921 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1924 /* sdp_read_header initializes this again */
1927 ret = sdp_read_header(s, ap);
1938 AVInputFormat ff_rtp_demuxer = {
1940 NULL_IF_CONFIG_SMALL("RTP input format"),
1944 ff_rtsp_fetch_packet,
1946 .flags = AVFMT_NOFILE,
1948 #endif /* CONFIG_RTP_DEMUXER */