3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/random_seed.h"
30 #include <sys/select.h>
35 #include "os_support.h"
41 #include "rtpdec_formats.h"
42 #include "rtpenc_chain.h"
45 //#define DEBUG_RTP_TCP
47 /* Timeout values for socket select, in ms,
48 * and read_packet(), in seconds */
49 #define SELECT_TIMEOUT_MS 100
50 #define READ_PACKET_TIMEOUT_S 10
51 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
52 #define SDP_MAX_SIZE 16384
53 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
55 static void get_word_until_chars(char *buf, int buf_size,
56 const char *sep, const char **pp)
62 p += strspn(p, SPACE_CHARS);
64 while (!strchr(sep, *p) && *p != '\0') {
65 if ((q - buf) < buf_size - 1)
74 static void get_word_sep(char *buf, int buf_size, const char *sep,
77 if (**pp == '/') (*pp)++;
78 get_word_until_chars(buf, buf_size, sep, pp);
81 static void get_word(char *buf, int buf_size, const char **pp)
83 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
86 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
88 * Used for seeking in the rtp stream.
90 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
94 p += strspn(p, SPACE_CHARS);
95 if (!av_stristart(p, "npt=", &p))
98 *start = AV_NOPTS_VALUE;
99 *end = AV_NOPTS_VALUE;
101 get_word_sep(buf, sizeof(buf), "-", &p);
102 *start = parse_date(buf, 1);
105 get_word_sep(buf, sizeof(buf), "-", &p);
106 *end = parse_date(buf, 1);
108 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
109 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
112 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
114 struct addrinfo hints, *ai = NULL;
115 memset(&hints, 0, sizeof(hints));
116 hints.ai_flags = AI_NUMERICHOST;
117 if (getaddrinfo(buf, NULL, &hints, &ai))
119 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
125 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
126 RTSPStream *rtsp_st, AVCodecContext *codec)
130 codec->codec_id = handler->codec_id;
131 rtsp_st->dynamic_handler = handler;
133 rtsp_st->dynamic_protocol_context = handler->open();
136 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
137 static int sdp_parse_rtpmap(AVFormatContext *s,
138 AVStream *st, RTSPStream *rtsp_st,
139 int payload_type, const char *p)
141 AVCodecContext *codec = st->codec;
147 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
148 * see if we can handle this kind of payload.
149 * The space should normally not be there but some Real streams or
150 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
151 * have a trailing space. */
152 get_word_sep(buf, sizeof(buf), "/ ", &p);
153 if (payload_type >= RTP_PT_PRIVATE) {
154 RTPDynamicProtocolHandler *handler =
155 ff_rtp_handler_find_by_name(buf, codec->codec_type);
156 init_rtp_handler(handler, rtsp_st, codec);
157 /* If no dynamic handler was found, check with the list of standard
158 * allocated types, if such a stream for some reason happens to
159 * use a private payload type. This isn't handled in rtpdec.c, since
160 * the format name from the rtpmap line never is passed into rtpdec. */
161 if (!rtsp_st->dynamic_handler)
162 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
164 /* We are in a standard case
165 * (from http://www.iana.org/assignments/rtp-parameters). */
166 /* search into AVRtpPayloadTypes[] */
167 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
170 c = avcodec_find_decoder(codec->codec_id);
176 get_word_sep(buf, sizeof(buf), "/", &p);
178 switch (codec->codec_type) {
179 case AVMEDIA_TYPE_AUDIO:
180 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
181 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
182 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
184 codec->sample_rate = i;
185 av_set_pts_info(st, 32, 1, codec->sample_rate);
186 get_word_sep(buf, sizeof(buf), "/", &p);
190 // TODO: there is a bug here; if it is a mono stream, and
191 // less than 22000Hz, faad upconverts to stereo and twice
192 // the frequency. No problem, but the sample rate is being
193 // set here by the sdp line. Patch on its way. (rdm)
195 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
197 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
200 case AVMEDIA_TYPE_VIDEO:
201 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
203 av_set_pts_info(st, 32, 1, i);
211 /* parse the attribute line from the fmtp a line of an sdp response. This
212 * is broken out as a function because it is used in rtp_h264.c, which is
214 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
215 char *value, int value_size)
217 *p += strspn(*p, SPACE_CHARS);
219 get_word_sep(attr, attr_size, "=", p);
222 get_word_sep(value, value_size, ";", p);
230 typedef struct SDPParseState {
232 struct sockaddr_storage default_ip;
234 int skip_media; ///< set if an unknown m= line occurs
237 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
238 int letter, const char *buf)
240 RTSPState *rt = s->priv_data;
241 char buf1[64], st_type[64];
243 enum AVMediaType codec_type;
247 struct sockaddr_storage sdp_ip;
250 dprintf(s, "sdp: %c='%s'\n", letter, buf);
253 if (s1->skip_media && letter != 'm')
257 get_word(buf1, sizeof(buf1), &p);
258 if (strcmp(buf1, "IN") != 0)
260 get_word(buf1, sizeof(buf1), &p);
261 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
263 get_word_sep(buf1, sizeof(buf1), "/", &p);
264 if (get_sockaddr(buf1, &sdp_ip))
269 get_word_sep(buf1, sizeof(buf1), "/", &p);
272 if (s->nb_streams == 0) {
273 s1->default_ip = sdp_ip;
274 s1->default_ttl = ttl;
276 st = s->streams[s->nb_streams - 1];
277 rtsp_st = st->priv_data;
278 rtsp_st->sdp_ip = sdp_ip;
279 rtsp_st->sdp_ttl = ttl;
283 av_metadata_set2(&s->metadata, "title", p, 0);
286 if (s->nb_streams == 0) {
287 av_metadata_set2(&s->metadata, "comment", p, 0);
294 get_word(st_type, sizeof(st_type), &p);
295 if (!strcmp(st_type, "audio")) {
296 codec_type = AVMEDIA_TYPE_AUDIO;
297 } else if (!strcmp(st_type, "video")) {
298 codec_type = AVMEDIA_TYPE_VIDEO;
299 } else if (!strcmp(st_type, "application")) {
300 codec_type = AVMEDIA_TYPE_DATA;
305 rtsp_st = av_mallocz(sizeof(RTSPStream));
308 rtsp_st->stream_index = -1;
309 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
311 rtsp_st->sdp_ip = s1->default_ip;
312 rtsp_st->sdp_ttl = s1->default_ttl;
314 get_word(buf1, sizeof(buf1), &p); /* port */
315 rtsp_st->sdp_port = atoi(buf1);
317 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
319 /* XXX: handle list of formats */
320 get_word(buf1, sizeof(buf1), &p); /* format list */
321 rtsp_st->sdp_payload_type = atoi(buf1);
323 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
324 /* no corresponding stream */
326 st = av_new_stream(s, 0);
329 st->priv_data = rtsp_st;
330 rtsp_st->stream_index = st->index;
331 st->codec->codec_type = codec_type;
332 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
333 RTPDynamicProtocolHandler *handler;
334 /* if standard payload type, we can find the codec right now */
335 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
336 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
337 st->codec->sample_rate > 0)
338 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
339 /* Even static payload types may need a custom depacketizer */
340 handler = ff_rtp_handler_find_by_id(
341 rtsp_st->sdp_payload_type, st->codec->codec_type);
342 init_rtp_handler(handler, rtsp_st, st->codec);
345 /* put a default control url */
346 av_strlcpy(rtsp_st->control_url, rt->control_uri,
347 sizeof(rtsp_st->control_url));
350 if (av_strstart(p, "control:", &p)) {
351 if (s->nb_streams == 0) {
352 if (!strncmp(p, "rtsp://", 7))
353 av_strlcpy(rt->control_uri, p,
354 sizeof(rt->control_uri));
357 /* get the control url */
358 st = s->streams[s->nb_streams - 1];
359 rtsp_st = st->priv_data;
361 /* XXX: may need to add full url resolution */
362 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
364 if (proto[0] == '\0') {
365 /* relative control URL */
366 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
367 av_strlcat(rtsp_st->control_url, "/",
368 sizeof(rtsp_st->control_url));
369 av_strlcat(rtsp_st->control_url, p,
370 sizeof(rtsp_st->control_url));
372 av_strlcpy(rtsp_st->control_url, p,
373 sizeof(rtsp_st->control_url));
375 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
376 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
377 get_word(buf1, sizeof(buf1), &p);
378 payload_type = atoi(buf1);
379 st = s->streams[s->nb_streams - 1];
380 rtsp_st = st->priv_data;
381 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
382 } else if (av_strstart(p, "fmtp:", &p) ||
383 av_strstart(p, "framesize:", &p)) {
384 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
385 // let dynamic protocol handlers have a stab at the line.
386 get_word(buf1, sizeof(buf1), &p);
387 payload_type = atoi(buf1);
388 for (i = 0; i < s->nb_streams; i++) {
390 rtsp_st = st->priv_data;
391 if (rtsp_st->sdp_payload_type == payload_type &&
392 rtsp_st->dynamic_handler &&
393 rtsp_st->dynamic_handler->parse_sdp_a_line)
394 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
395 rtsp_st->dynamic_protocol_context, buf);
397 } else if (av_strstart(p, "range:", &p)) {
400 // this is so that seeking on a streamed file can work.
401 rtsp_parse_range_npt(p, &start, &end);
402 s->start_time = start;
403 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
404 s->duration = (end == AV_NOPTS_VALUE) ?
405 AV_NOPTS_VALUE : end - start;
406 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
408 rt->transport = RTSP_TRANSPORT_RDT;
409 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
411 st = s->streams[s->nb_streams - 1];
412 st->codec->sample_rate = atoi(p);
414 if (rt->server_type == RTSP_SERVER_WMS)
415 ff_wms_parse_sdp_a_line(s, p);
416 if (s->nb_streams > 0) {
417 if (rt->server_type == RTSP_SERVER_REAL)
418 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
420 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
421 if (rtsp_st->dynamic_handler &&
422 rtsp_st->dynamic_handler->parse_sdp_a_line)
423 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
425 rtsp_st->dynamic_protocol_context, buf);
432 int ff_sdp_parse(AVFormatContext *s, const char *content)
436 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
437 * contain long SDP lines containing complete ASF Headers (several
438 * kB) or arrays of MDPR (RM stream descriptor) headers plus
439 * "rulebooks" describing their properties. Therefore, the SDP line
442 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
443 * in rtpdec_xiph.c. */
445 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
447 memset(s1, 0, sizeof(SDPParseState));
450 p += strspn(p, SPACE_CHARS);
458 /* get the content */
460 while (*p != '\n' && *p != '\r' && *p != '\0') {
461 if ((q - buf) < sizeof(buf) - 1)
466 sdp_parse_line(s, s1, letter, buf);
468 while (*p != '\n' && *p != '\0')
475 #endif /* CONFIG_RTPDEC */
477 /* close and free RTSP streams */
478 void ff_rtsp_close_streams(AVFormatContext *s)
480 RTSPState *rt = s->priv_data;
484 for (i = 0; i < rt->nb_rtsp_streams; i++) {
485 rtsp_st = rt->rtsp_streams[i];
487 if (rtsp_st->transport_priv) {
489 AVFormatContext *rtpctx = rtsp_st->transport_priv;
490 av_write_trailer(rtpctx);
491 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
493 url_close_dyn_buf(rtpctx->pb, &ptr);
496 url_fclose(rtpctx->pb);
498 av_metadata_free(&rtpctx->streams[0]->metadata);
499 av_metadata_free(&rtpctx->metadata);
500 av_free(rtpctx->streams[0]);
502 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
503 ff_rdt_parse_close(rtsp_st->transport_priv);
504 else if (CONFIG_RTPDEC)
505 rtp_parse_close(rtsp_st->transport_priv);
507 if (rtsp_st->rtp_handle)
508 url_close(rtsp_st->rtp_handle);
509 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
510 rtsp_st->dynamic_handler->close(
511 rtsp_st->dynamic_protocol_context);
514 av_free(rt->rtsp_streams);
516 av_close_input_stream (rt->asf_ctx);
519 av_free(rt->recvbuf);
522 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
524 RTSPState *rt = s->priv_data;
527 /* open the RTP context */
528 if (rtsp_st->stream_index >= 0)
529 st = s->streams[rtsp_st->stream_index];
531 s->ctx_flags |= AVFMTCTX_NOHEADER;
533 if (s->oformat && CONFIG_RTSP_MUXER) {
534 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
536 RTSP_TCP_MAX_PACKET_SIZE);
537 /* Ownership of rtp_handle is passed to the rtp mux context */
538 rtsp_st->rtp_handle = NULL;
539 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
540 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
541 rtsp_st->dynamic_protocol_context,
542 rtsp_st->dynamic_handler);
543 else if (CONFIG_RTPDEC)
544 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
545 rtsp_st->sdp_payload_type,
546 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
547 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
549 if (!rtsp_st->transport_priv) {
550 return AVERROR(ENOMEM);
551 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
552 if (rtsp_st->dynamic_handler) {
553 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
554 rtsp_st->dynamic_protocol_context,
555 rtsp_st->dynamic_handler);
562 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
563 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
569 p += strspn(p, SPACE_CHARS);
570 v = strtol(p, (char **)&p, 10);
574 v = strtol(p, (char **)&p, 10);
583 /* XXX: only one transport specification is parsed */
584 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
586 char transport_protocol[16];
588 char lower_transport[16];
590 RTSPTransportField *th;
593 reply->nb_transports = 0;
596 p += strspn(p, SPACE_CHARS);
600 th = &reply->transports[reply->nb_transports];
602 get_word_sep(transport_protocol, sizeof(transport_protocol),
604 if (!strcasecmp (transport_protocol, "rtp")) {
605 get_word_sep(profile, sizeof(profile), "/;,", &p);
606 lower_transport[0] = '\0';
607 /* rtp/avp/<protocol> */
609 get_word_sep(lower_transport, sizeof(lower_transport),
612 th->transport = RTSP_TRANSPORT_RTP;
613 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
614 !strcasecmp (transport_protocol, "x-real-rdt")) {
615 /* x-pn-tng/<protocol> */
616 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
618 th->transport = RTSP_TRANSPORT_RDT;
620 if (!strcasecmp(lower_transport, "TCP"))
621 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
623 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
627 /* get each parameter */
628 while (*p != '\0' && *p != ',') {
629 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
630 if (!strcmp(parameter, "port")) {
633 rtsp_parse_range(&th->port_min, &th->port_max, &p);
635 } else if (!strcmp(parameter, "client_port")) {
638 rtsp_parse_range(&th->client_port_min,
639 &th->client_port_max, &p);
641 } else if (!strcmp(parameter, "server_port")) {
644 rtsp_parse_range(&th->server_port_min,
645 &th->server_port_max, &p);
647 } else if (!strcmp(parameter, "interleaved")) {
650 rtsp_parse_range(&th->interleaved_min,
651 &th->interleaved_max, &p);
653 } else if (!strcmp(parameter, "multicast")) {
654 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
655 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
656 } else if (!strcmp(parameter, "ttl")) {
659 th->ttl = strtol(p, (char **)&p, 10);
661 } else if (!strcmp(parameter, "destination")) {
664 get_word_sep(buf, sizeof(buf), ";,", &p);
665 get_sockaddr(buf, &th->destination);
667 } else if (!strcmp(parameter, "source")) {
670 get_word_sep(buf, sizeof(buf), ";,", &p);
671 av_strlcpy(th->source, buf, sizeof(th->source));
675 while (*p != ';' && *p != '\0' && *p != ',')
683 reply->nb_transports++;
687 static void handle_rtp_info(RTSPState *rt, const char *url,
688 uint32_t seq, uint32_t rtptime)
691 if (!rtptime || !url[0])
693 if (rt->transport != RTSP_TRANSPORT_RTP)
695 for (i = 0; i < rt->nb_rtsp_streams; i++) {
696 RTSPStream *rtsp_st = rt->rtsp_streams[i];
697 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
700 if (!strcmp(rtsp_st->control_url, url)) {
701 rtpctx->base_timestamp = rtptime;
707 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
710 char key[20], value[1024], url[1024] = "";
711 uint32_t seq = 0, rtptime = 0;
714 p += strspn(p, SPACE_CHARS);
717 get_word_sep(key, sizeof(key), "=", &p);
721 get_word_sep(value, sizeof(value), ";, ", &p);
723 if (!strcmp(key, "url"))
724 av_strlcpy(url, value, sizeof(url));
725 else if (!strcmp(key, "seq"))
726 seq = strtol(value, NULL, 10);
727 else if (!strcmp(key, "rtptime"))
728 rtptime = strtol(value, NULL, 10);
730 handle_rtp_info(rt, url, seq, rtptime);
739 handle_rtp_info(rt, url, seq, rtptime);
742 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
743 RTSPState *rt, const char *method)
747 /* NOTE: we do case independent match for broken servers */
749 if (av_stristart(p, "Session:", &p)) {
751 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
752 if (av_stristart(p, ";timeout=", &p) &&
753 (t = strtol(p, NULL, 10)) > 0) {
756 } else if (av_stristart(p, "Content-Length:", &p)) {
757 reply->content_length = strtol(p, NULL, 10);
758 } else if (av_stristart(p, "Transport:", &p)) {
759 rtsp_parse_transport(reply, p);
760 } else if (av_stristart(p, "CSeq:", &p)) {
761 reply->seq = strtol(p, NULL, 10);
762 } else if (av_stristart(p, "Range:", &p)) {
763 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
764 } else if (av_stristart(p, "RealChallenge1:", &p)) {
765 p += strspn(p, SPACE_CHARS);
766 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
767 } else if (av_stristart(p, "Server:", &p)) {
768 p += strspn(p, SPACE_CHARS);
769 av_strlcpy(reply->server, p, sizeof(reply->server));
770 } else if (av_stristart(p, "Notice:", &p) ||
771 av_stristart(p, "X-Notice:", &p)) {
772 reply->notice = strtol(p, NULL, 10);
773 } else if (av_stristart(p, "Location:", &p)) {
774 p += strspn(p, SPACE_CHARS);
775 av_strlcpy(reply->location, p , sizeof(reply->location));
776 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
777 p += strspn(p, SPACE_CHARS);
778 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
779 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
780 p += strspn(p, SPACE_CHARS);
781 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
782 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
783 p += strspn(p, SPACE_CHARS);
784 if (method && !strcmp(method, "DESCRIBE"))
785 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
786 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
787 p += strspn(p, SPACE_CHARS);
788 if (method && !strcmp(method, "PLAY"))
789 rtsp_parse_rtp_info(rt, p);
793 /* skip a RTP/TCP interleaved packet */
794 void ff_rtsp_skip_packet(AVFormatContext *s)
796 RTSPState *rt = s->priv_data;
800 ret = url_read_complete(rt->rtsp_hd, buf, 3);
803 len = AV_RB16(buf + 1);
805 dprintf(s, "skipping RTP packet len=%d\n", len);
810 if (len1 > sizeof(buf))
812 ret = url_read_complete(rt->rtsp_hd, buf, len1);
819 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
820 unsigned char **content_ptr,
821 int return_on_interleaved_data, const char *method)
823 RTSPState *rt = s->priv_data;
824 char buf[4096], buf1[1024], *q;
827 int ret, content_length, line_count = 0;
828 unsigned char *content = NULL;
830 memset(reply, 0, sizeof(*reply));
832 /* parse reply (XXX: use buffers) */
833 rt->last_reply[0] = '\0';
837 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
839 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
846 /* XXX: only parse it if first char on line ? */
847 if (return_on_interleaved_data) {
850 ff_rtsp_skip_packet(s);
851 } else if (ch != '\r') {
852 if ((q - buf) < sizeof(buf) - 1)
858 dprintf(s, "line='%s'\n", buf);
860 /* test if last line */
864 if (line_count == 0) {
866 get_word(buf1, sizeof(buf1), &p);
867 get_word(buf1, sizeof(buf1), &p);
868 reply->status_code = atoi(buf1);
869 av_strlcpy(reply->reason, p, sizeof(reply->reason));
871 ff_rtsp_parse_line(reply, p, rt, method);
872 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
873 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
878 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
879 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
881 content_length = reply->content_length;
882 if (content_length > 0) {
883 /* leave some room for a trailing '\0' (useful for simple parsing) */
884 content = av_malloc(content_length + 1);
885 (void)url_read_complete(rt->rtsp_hd, content, content_length);
886 content[content_length] = '\0';
889 *content_ptr = content;
893 if (rt->seq != reply->seq) {
894 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
895 rt->seq, reply->seq);
899 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
900 reply->notice == 2104 /* Start-of-Stream Reached */ ||
901 reply->notice == 2306 /* Continuous Feed Terminated */) {
902 rt->state = RTSP_STATE_IDLE;
903 } else if (reply->notice >= 4400 && reply->notice < 5500) {
904 return AVERROR(EIO); /* data or server error */
905 } else if (reply->notice == 2401 /* Ticket Expired */ ||
906 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
907 return AVERROR(EPERM);
912 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
913 const char *method, const char *url,
915 const unsigned char *send_content,
916 int send_content_length)
918 RTSPState *rt = s->priv_data;
919 char buf[4096], *out_buf;
920 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
922 /* Add in RTSP headers */
925 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
927 av_strlcat(buf, headers, sizeof(buf));
928 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
929 if (rt->session_id[0] != '\0' && (!headers ||
930 !strstr(headers, "\nIf-Match:"))) {
931 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
934 char *str = ff_http_auth_create_response(&rt->auth_state,
935 rt->auth, url, method);
937 av_strlcat(buf, str, sizeof(buf));
940 if (send_content_length > 0 && send_content)
941 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
942 av_strlcat(buf, "\r\n", sizeof(buf));
944 /* base64 encode rtsp if tunneling */
945 if (rt->control_transport == RTSP_MODE_TUNNEL) {
946 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
950 dprintf(s, "Sending:\n%s--\n", buf);
952 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
953 if (send_content_length > 0 && send_content) {
954 if (rt->control_transport == RTSP_MODE_TUNNEL) {
955 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
956 "with content data not supported\n");
957 return AVERROR_PATCHWELCOME;
959 url_write(rt->rtsp_hd_out, send_content, send_content_length);
961 rt->last_cmd_time = av_gettime();
966 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
967 const char *url, const char *headers)
969 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
972 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
973 const char *headers, RTSPMessageHeader *reply,
974 unsigned char **content_ptr)
976 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
977 content_ptr, NULL, 0);
980 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
981 const char *method, const char *url,
983 RTSPMessageHeader *reply,
984 unsigned char **content_ptr,
985 const unsigned char *send_content,
986 int send_content_length)
988 RTSPState *rt = s->priv_data;
989 HTTPAuthType cur_auth_type;
993 cur_auth_type = rt->auth_state.auth_type;
994 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
996 send_content_length)))
999 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1002 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1003 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1006 if (reply->status_code > 400){
1007 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1011 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1018 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1020 static int make_setup_request(AVFormatContext *s, const char *host, int port,
1021 int lower_transport, const char *real_challenge)
1023 RTSPState *rt = s->priv_data;
1024 int rtx, j, i, err, interleave = 0;
1025 RTSPStream *rtsp_st;
1026 RTSPMessageHeader reply1, *reply = &reply1;
1028 const char *trans_pref;
1030 if (rt->transport == RTSP_TRANSPORT_RDT)
1031 trans_pref = "x-pn-tng";
1033 trans_pref = "RTP/AVP";
1035 /* default timeout: 1 minute */
1038 /* for each stream, make the setup request */
1039 /* XXX: we assume the same server is used for the control of each
1042 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1043 char transport[2048];
1046 * WMS serves all UDP data over a single connection, the RTX, which
1047 * isn't necessarily the first in the SDP but has to be the first
1048 * to be set up, else the second/third SETUP will fail with a 461.
1050 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1051 rt->server_type == RTSP_SERVER_WMS) {
1054 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1055 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1057 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1061 if (rtx == rt->nb_rtsp_streams)
1062 return -1; /* no RTX found */
1063 rtsp_st = rt->rtsp_streams[rtx];
1065 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1067 rtsp_st = rt->rtsp_streams[i];
1070 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1073 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1074 port = reply->transports[0].client_port_min;
1078 /* first try in specified port range */
1079 if (RTSP_RTP_PORT_MIN != 0) {
1080 while (j <= RTSP_RTP_PORT_MAX) {
1081 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1082 "?localport=%d", j);
1083 /* we will use two ports per rtp stream (rtp and rtcp) */
1085 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1091 /* then try on any port */
1092 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1093 err = AVERROR_INVALIDDATA;
1097 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1103 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1105 snprintf(transport, sizeof(transport) - 1,
1106 "%s/UDP;", trans_pref);
1107 if (rt->server_type != RTSP_SERVER_REAL)
1108 av_strlcat(transport, "unicast;", sizeof(transport));
1109 av_strlcatf(transport, sizeof(transport),
1110 "client_port=%d", port);
1111 if (rt->transport == RTSP_TRANSPORT_RTP &&
1112 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1113 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1117 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1118 /** For WMS streams, the application streams are only used for
1119 * UDP. When trying to set it up for TCP streams, the server
1120 * will return an error. Therefore, we skip those streams. */
1121 if (rt->server_type == RTSP_SERVER_WMS &&
1122 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1125 snprintf(transport, sizeof(transport) - 1,
1126 "%s/TCP;", trans_pref);
1127 if (rt->server_type == RTSP_SERVER_WMS)
1128 av_strlcat(transport, "unicast;", sizeof(transport));
1129 av_strlcatf(transport, sizeof(transport),
1130 "interleaved=%d-%d",
1131 interleave, interleave + 1);
1135 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1136 snprintf(transport, sizeof(transport) - 1,
1137 "%s/UDP;multicast", trans_pref);
1140 av_strlcat(transport, ";mode=receive", sizeof(transport));
1141 } else if (rt->server_type == RTSP_SERVER_REAL ||
1142 rt->server_type == RTSP_SERVER_WMS)
1143 av_strlcat(transport, ";mode=play", sizeof(transport));
1144 snprintf(cmd, sizeof(cmd),
1145 "Transport: %s\r\n",
1147 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1148 char real_res[41], real_csum[9];
1149 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1151 av_strlcatf(cmd, sizeof(cmd),
1153 "RealChallenge2: %s, sd=%s\r\n",
1154 rt->session_id, real_res, real_csum);
1156 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1157 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1160 } else if (reply->status_code != RTSP_STATUS_OK ||
1161 reply->nb_transports != 1) {
1162 err = AVERROR_INVALIDDATA;
1166 /* XXX: same protocol for all streams is required */
1168 if (reply->transports[0].lower_transport != rt->lower_transport ||
1169 reply->transports[0].transport != rt->transport) {
1170 err = AVERROR_INVALIDDATA;
1174 rt->lower_transport = reply->transports[0].lower_transport;
1175 rt->transport = reply->transports[0].transport;
1178 /* Fail if the server responded with another lower transport mode
1179 * than what we requested. */
1180 if (reply->transports[0].lower_transport != lower_transport) {
1181 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1182 err = AVERROR_INVALIDDATA;
1186 switch(reply->transports[0].lower_transport) {
1187 case RTSP_LOWER_TRANSPORT_TCP:
1188 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1189 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1192 case RTSP_LOWER_TRANSPORT_UDP: {
1193 char url[1024], options[30] = "";
1195 if (rt->filter_source)
1196 av_strlcpy(options, "?connect=1", sizeof(options));
1197 /* Use source address if specified */
1198 if (reply->transports[0].source[0]) {
1199 ff_url_join(url, sizeof(url), "rtp", NULL,
1200 reply->transports[0].source,
1201 reply->transports[0].server_port_min, options);
1203 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1204 reply->transports[0].server_port_min, options);
1206 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1207 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1208 err = AVERROR_INVALIDDATA;
1211 /* Try to initialize the connection state in a
1212 * potential NAT router by sending dummy packets.
1213 * RTP/RTCP dummy packets are used for RDT, too.
1215 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1217 rtp_send_punch_packets(rtsp_st->rtp_handle);
1220 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1221 char url[1024], namebuf[50];
1222 struct sockaddr_storage addr;
1225 if (reply->transports[0].destination.ss_family) {
1226 addr = reply->transports[0].destination;
1227 port = reply->transports[0].port_min;
1228 ttl = reply->transports[0].ttl;
1230 addr = rtsp_st->sdp_ip;
1231 port = rtsp_st->sdp_port;
1232 ttl = rtsp_st->sdp_ttl;
1234 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1235 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1236 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1237 port, "?ttl=%d", ttl);
1238 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1239 err = AVERROR_INVALIDDATA;
1246 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1250 if (reply->timeout > 0)
1251 rt->timeout = reply->timeout;
1253 if (rt->server_type == RTSP_SERVER_REAL)
1254 rt->need_subscription = 1;
1259 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1260 if (rt->rtsp_streams[i]->rtp_handle) {
1261 url_close(rt->rtsp_streams[i]->rtp_handle);
1262 rt->rtsp_streams[i]->rtp_handle = NULL;
1268 void ff_rtsp_close_connections(AVFormatContext *s)
1270 RTSPState *rt = s->priv_data;
1271 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1272 url_close(rt->rtsp_hd);
1273 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1276 int ff_rtsp_connect(AVFormatContext *s)
1278 RTSPState *rt = s->priv_data;
1279 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1280 char *option_list, *option, *filename;
1281 int port, err, tcp_fd;
1282 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1283 int lower_transport_mask = 0;
1284 char real_challenge[64];
1285 struct sockaddr_storage peer;
1286 socklen_t peer_len = sizeof(peer);
1288 if (!ff_network_init())
1289 return AVERROR(EIO);
1291 rt->control_transport = RTSP_MODE_PLAIN;
1292 /* extract hostname and port */
1293 av_url_split(NULL, 0, auth, sizeof(auth),
1294 host, sizeof(host), &port, path, sizeof(path), s->filename);
1296 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1299 port = RTSP_DEFAULT_PORT;
1301 /* search for options */
1302 option_list = strrchr(path, '?');
1304 /* Strip out the RTSP specific options, write out the rest of
1305 * the options back into the same string. */
1306 filename = option_list;
1307 while (option_list) {
1308 /* move the option pointer */
1309 option = ++option_list;
1310 option_list = strchr(option_list, '&');
1314 /* handle the options */
1315 if (!strcmp(option, "udp")) {
1316 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1317 } else if (!strcmp(option, "multicast")) {
1318 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1319 } else if (!strcmp(option, "tcp")) {
1320 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1321 } else if(!strcmp(option, "http")) {
1322 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1323 rt->control_transport = RTSP_MODE_TUNNEL;
1324 } else if (!strcmp(option, "filter_src")) {
1325 rt->filter_source = 1;
1327 /* Write options back into the buffer, using memmove instead
1328 * of strcpy since the strings may overlap. */
1329 int len = strlen(option);
1330 memmove(++filename, option, len);
1332 if (option_list) *filename = '&';
1338 if (!lower_transport_mask)
1339 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1342 /* Only UDP or TCP - UDP multicast isn't supported. */
1343 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1344 (1 << RTSP_LOWER_TRANSPORT_TCP);
1345 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1346 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1347 "only UDP and TCP are supported for output.\n");
1348 err = AVERROR(EINVAL);
1353 /* Construct the URI used in request; this is similar to s->filename,
1354 * but with authentication credentials removed and RTSP specific options
1356 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1357 host, port, "%s", path);
1359 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1360 /* set up initial handshake for tunneling */
1361 char httpname[1024];
1362 char sessioncookie[17];
1365 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1366 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1367 av_get_random_seed(), av_get_random_seed());
1370 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1375 /* generate GET headers */
1376 snprintf(headers, sizeof(headers),
1377 "x-sessioncookie: %s\r\n"
1378 "Accept: application/x-rtsp-tunnelled\r\n"
1379 "Pragma: no-cache\r\n"
1380 "Cache-Control: no-cache\r\n",
1382 ff_http_set_headers(rt->rtsp_hd, headers);
1384 /* complete the connection */
1385 if (url_connect(rt->rtsp_hd)) {
1391 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1396 /* generate POST headers */
1397 snprintf(headers, sizeof(headers),
1398 "x-sessioncookie: %s\r\n"
1399 "Content-Type: application/x-rtsp-tunnelled\r\n"
1400 "Pragma: no-cache\r\n"
1401 "Cache-Control: no-cache\r\n"
1402 "Content-Length: 32767\r\n"
1403 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1405 ff_http_set_headers(rt->rtsp_hd_out, headers);
1406 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1408 /* Initialize the authentication state for the POST session. The HTTP
1409 * protocol implementation doesn't properly handle multi-pass
1410 * authentication for POST requests, since it would require one of
1412 * - implementing Expect: 100-continue, which many HTTP servers
1413 * don't support anyway, even less the RTSP servers that do HTTP
1415 * - sending the whole POST data until getting a 401 reply specifying
1416 * what authentication method to use, then resending all that data
1417 * - waiting for potential 401 replies directly after sending the
1418 * POST header (waiting for some unspecified time)
1419 * Therefore, we copy the full auth state, which works for both basic
1420 * and digest. (For digest, we would have to synchronize the nonce
1421 * count variable between the two sessions, if we'd do more requests
1422 * with the original session, though.)
1424 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1426 /* complete the connection */
1427 if (url_connect(rt->rtsp_hd_out)) {
1432 /* open the tcp connection */
1433 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1434 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1438 rt->rtsp_hd_out = rt->rtsp_hd;
1442 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1443 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1444 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1445 NULL, 0, NI_NUMERICHOST);
1448 /* request options supported by the server; this also detects server
1450 for (rt->server_type = RTSP_SERVER_RTP;;) {
1452 if (rt->server_type == RTSP_SERVER_REAL)
1455 * The following entries are required for proper
1456 * streaming from a Realmedia server. They are
1457 * interdependent in some way although we currently
1458 * don't quite understand how. Values were copied
1459 * from mplayer SVN r23589.
1460 * @param CompanyID is a 16-byte ID in base64
1461 * @param ClientChallenge is a 16-byte ID in hex
1463 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1464 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1465 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1466 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1468 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1469 if (reply->status_code != RTSP_STATUS_OK) {
1470 err = AVERROR_INVALIDDATA;
1474 /* detect server type if not standard-compliant RTP */
1475 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1476 rt->server_type = RTSP_SERVER_REAL;
1478 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1479 rt->server_type = RTSP_SERVER_WMS;
1480 } else if (rt->server_type == RTSP_SERVER_REAL)
1481 strcpy(real_challenge, reply->real_challenge);
1485 if (s->iformat && CONFIG_RTSP_DEMUXER)
1486 err = ff_rtsp_setup_input_streams(s, reply);
1487 else if (CONFIG_RTSP_MUXER)
1488 err = ff_rtsp_setup_output_streams(s, host);
1493 int lower_transport = ff_log2_tab[lower_transport_mask &
1494 ~(lower_transport_mask - 1)];
1496 err = make_setup_request(s, host, port, lower_transport,
1497 rt->server_type == RTSP_SERVER_REAL ?
1498 real_challenge : NULL);
1501 lower_transport_mask &= ~(1 << lower_transport);
1502 if (lower_transport_mask == 0 && err == 1) {
1503 err = FF_NETERROR(EPROTONOSUPPORT);
1508 rt->state = RTSP_STATE_IDLE;
1509 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1512 ff_rtsp_close_streams(s);
1513 ff_rtsp_close_connections(s);
1514 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1515 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1516 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1524 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1527 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1528 uint8_t *buf, int buf_size, int64_t wait_end)
1530 RTSPState *rt = s->priv_data;
1531 RTSPStream *rtsp_st;
1533 int fd, fd_rtcp, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1537 if (url_interrupt_cb())
1538 return AVERROR(EINTR);
1539 if (wait_end && wait_end - av_gettime() < 0)
1540 return AVERROR(EAGAIN);
1543 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1544 FD_SET(tcp_fd, &rfds);
1549 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1550 rtsp_st = rt->rtsp_streams[i];
1551 if (rtsp_st->rtp_handle) {
1552 fd = url_get_file_handle(rtsp_st->rtp_handle);
1553 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1554 if (FFMAX(fd, fd_rtcp) > fd_max)
1555 fd_max = FFMAX(fd, fd_rtcp);
1557 FD_SET(fd_rtcp, &rfds);
1561 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1562 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1565 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1566 rtsp_st = rt->rtsp_streams[i];
1567 if (rtsp_st->rtp_handle) {
1568 fd = url_get_file_handle(rtsp_st->rtp_handle);
1569 fd_rtcp = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1570 if (FD_ISSET(fd_rtcp, &rfds) || FD_ISSET(fd, &rfds)) {
1571 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1573 *prtsp_st = rtsp_st;
1579 #if CONFIG_RTSP_DEMUXER
1580 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1581 RTSPMessageHeader reply;
1583 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1586 /* XXX: parse message */
1587 if (rt->state != RTSP_STATE_STREAMING)
1591 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1592 return FF_NETERROR(ETIMEDOUT);
1593 } else if (n < 0 && errno != EINTR)
1594 return AVERROR(errno);
1598 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1600 RTSPState *rt = s->priv_data;
1602 RTSPStream *rtsp_st, *first_queue_st = NULL;
1603 int64_t wait_end = 0;
1605 if (rt->nb_byes == rt->nb_rtsp_streams)
1608 /* get next frames from the same RTP packet */
1609 if (rt->cur_transport_priv) {
1610 if (rt->transport == RTSP_TRANSPORT_RDT) {
1611 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1613 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1615 rt->cur_transport_priv = NULL;
1617 } else if (ret == 1) {
1620 rt->cur_transport_priv = NULL;
1623 if (rt->transport == RTSP_TRANSPORT_RTP) {
1625 int64_t first_queue_time = 0;
1626 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1627 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1631 queue_time = ff_rtp_queued_packet_time(rtpctx);
1632 if (queue_time && (queue_time - first_queue_time < 0 ||
1633 !first_queue_time)) {
1634 first_queue_time = queue_time;
1635 first_queue_st = rt->rtsp_streams[i];
1638 if (first_queue_time)
1639 wait_end = first_queue_time + s->max_delay;
1642 /* read next RTP packet */
1645 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1647 return AVERROR(ENOMEM);
1650 switch(rt->lower_transport) {
1652 #if CONFIG_RTSP_DEMUXER
1653 case RTSP_LOWER_TRANSPORT_TCP:
1654 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1657 case RTSP_LOWER_TRANSPORT_UDP:
1658 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1659 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1660 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1661 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1664 if (len == AVERROR(EAGAIN) && first_queue_st &&
1665 rt->transport == RTSP_TRANSPORT_RTP) {
1666 rtsp_st = first_queue_st;
1667 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1674 if (rt->transport == RTSP_TRANSPORT_RDT) {
1675 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1677 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1679 /* Either bad packet, or a RTCP packet. Check if the
1680 * first_rtcp_ntp_time field was initialized. */
1681 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1682 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1683 /* first_rtcp_ntp_time has been initialized for this stream,
1684 * copy the same value to all other uninitialized streams,
1685 * in order to map their timestamp origin to the same ntp time
1688 AVStream *st = NULL;
1689 if (rtsp_st->stream_index >= 0)
1690 st = s->streams[rtsp_st->stream_index];
1691 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1692 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1693 AVStream *st2 = NULL;
1694 if (rt->rtsp_streams[i]->stream_index >= 0)
1695 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1696 if (rtpctx2 && st && st2 &&
1697 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1698 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1699 rtpctx2->rtcp_ts_offset = av_rescale_q(
1700 rtpctx->rtcp_ts_offset, st->time_base,
1705 if (ret == -RTCP_BYE) {
1708 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1709 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1711 if (rt->nb_byes == rt->nb_rtsp_streams)
1720 /* more packets may follow, so we save the RTP context */
1721 rt->cur_transport_priv = rtsp_st->transport_priv;
1725 #endif /* CONFIG_RTPDEC */
1727 #if CONFIG_SDP_DEMUXER
1728 static int sdp_probe(AVProbeData *p1)
1730 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1732 /* we look for a line beginning "c=IN IP" */
1733 while (p < p_end && *p != '\0') {
1734 if (p + sizeof("c=IN IP") - 1 < p_end &&
1735 av_strstart(p, "c=IN IP", NULL))
1736 return AVPROBE_SCORE_MAX / 2;
1738 while (p < p_end - 1 && *p != '\n') p++;
1747 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1749 RTSPState *rt = s->priv_data;
1750 RTSPStream *rtsp_st;
1755 if (!ff_network_init())
1756 return AVERROR(EIO);
1758 /* read the whole sdp file */
1759 /* XXX: better loading */
1760 content = av_malloc(SDP_MAX_SIZE);
1761 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1764 return AVERROR_INVALIDDATA;
1766 content[size] ='\0';
1768 ff_sdp_parse(s, content);
1771 /* open each RTP stream */
1772 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1774 rtsp_st = rt->rtsp_streams[i];
1776 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1777 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1778 ff_url_join(url, sizeof(url), "rtp", NULL,
1779 namebuf, rtsp_st->sdp_port,
1780 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1782 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1783 err = AVERROR_INVALIDDATA;
1786 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1791 ff_rtsp_close_streams(s);
1796 static int sdp_read_close(AVFormatContext *s)
1798 ff_rtsp_close_streams(s);
1803 AVInputFormat sdp_demuxer = {
1805 NULL_IF_CONFIG_SMALL("SDP"),
1809 ff_rtsp_fetch_packet,
1812 #endif /* CONFIG_SDP_DEMUXER */
1814 #if CONFIG_RTP_DEMUXER
1815 static int rtp_probe(AVProbeData *p)
1817 if (av_strstart(p->filename, "rtp:", NULL))
1818 return AVPROBE_SCORE_MAX;
1822 static int rtp_read_header(AVFormatContext *s,
1823 AVFormatParameters *ap)
1825 uint8_t recvbuf[1500];
1826 char host[500], sdp[500];
1828 URLContext* in = NULL;
1830 AVCodecContext codec;
1831 struct sockaddr_storage addr;
1833 socklen_t addrlen = sizeof(addr);
1835 if (!ff_network_init())
1836 return AVERROR(EIO);
1838 ret = url_open(&in, s->filename, URL_RDONLY);
1843 ret = url_read(in, recvbuf, sizeof(recvbuf));
1844 if (ret == AVERROR(EAGAIN))
1849 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1853 if ((recvbuf[0] & 0xc0) != 0x80) {
1854 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1859 payload_type = recvbuf[1] & 0x7f;
1862 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1866 memset(&codec, 0, sizeof(codec));
1867 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1868 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1869 "without an SDP file describing it\n",
1873 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1874 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1875 "properly you need an SDP file "
1879 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1880 NULL, 0, s->filename);
1882 snprintf(sdp, sizeof(sdp),
1883 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1884 addr.ss_family == AF_INET ? 4 : 6, host,
1885 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1886 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1887 port, payload_type);
1888 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1890 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1893 /* sdp_read_header initializes this again */
1896 ret = sdp_read_header(s, ap);
1907 AVInputFormat rtp_demuxer = {
1909 NULL_IF_CONFIG_SMALL("RTP input format"),
1913 ff_rtsp_fetch_packet,
1915 .flags = AVFMT_NOFILE,
1917 #endif /* CONFIG_RTP_DEMUXER */