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/parseutils.h"
26 #include "libavutil/random_seed.h"
36 #include "os_support.h"
42 #include "rtpdec_formats.h"
43 #include "rtpenc_chain.h"
46 //#define DEBUG_RTP_TCP
48 /* Timeout values for socket poll, in ms,
49 * and read_packet(), in seconds */
50 #define POLL_TIMEOUT_MS 100
51 #define READ_PACKET_TIMEOUT_S 10
52 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
53 #define SDP_MAX_SIZE 16384
54 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
56 static void get_word_until_chars(char *buf, int buf_size,
57 const char *sep, const char **pp)
63 p += strspn(p, SPACE_CHARS);
65 while (!strchr(sep, *p) && *p != '\0') {
66 if ((q - buf) < buf_size - 1)
75 static void get_word_sep(char *buf, int buf_size, const char *sep,
78 if (**pp == '/') (*pp)++;
79 get_word_until_chars(buf, buf_size, sep, pp);
82 static void get_word(char *buf, int buf_size, const char **pp)
84 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
87 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
89 * Used for seeking in the rtp stream.
91 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
95 p += strspn(p, SPACE_CHARS);
96 if (!av_stristart(p, "npt=", &p))
99 *start = AV_NOPTS_VALUE;
100 *end = AV_NOPTS_VALUE;
102 get_word_sep(buf, sizeof(buf), "-", &p);
103 av_parse_time(start, buf, 1);
106 get_word_sep(buf, sizeof(buf), "-", &p);
107 av_parse_time(end, buf, 1);
109 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
110 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
113 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
115 struct addrinfo hints, *ai = NULL;
116 memset(&hints, 0, sizeof(hints));
117 hints.ai_flags = AI_NUMERICHOST;
118 if (getaddrinfo(buf, NULL, &hints, &ai))
120 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
126 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
127 RTSPStream *rtsp_st, AVCodecContext *codec)
131 codec->codec_id = handler->codec_id;
132 rtsp_st->dynamic_handler = handler;
134 rtsp_st->dynamic_protocol_context = handler->open();
137 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
138 static int sdp_parse_rtpmap(AVFormatContext *s,
139 AVStream *st, RTSPStream *rtsp_st,
140 int payload_type, const char *p)
142 AVCodecContext *codec = st->codec;
148 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
149 * see if we can handle this kind of payload.
150 * The space should normally not be there but some Real streams or
151 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
152 * have a trailing space. */
153 get_word_sep(buf, sizeof(buf), "/ ", &p);
154 if (payload_type >= RTP_PT_PRIVATE) {
155 RTPDynamicProtocolHandler *handler =
156 ff_rtp_handler_find_by_name(buf, codec->codec_type);
157 init_rtp_handler(handler, rtsp_st, codec);
158 /* If no dynamic handler was found, check with the list of standard
159 * allocated types, if such a stream for some reason happens to
160 * use a private payload type. This isn't handled in rtpdec.c, since
161 * the format name from the rtpmap line never is passed into rtpdec. */
162 if (!rtsp_st->dynamic_handler)
163 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
165 /* We are in a standard case
166 * (from http://www.iana.org/assignments/rtp-parameters). */
167 /* search into AVRtpPayloadTypes[] */
168 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
171 c = avcodec_find_decoder(codec->codec_id);
177 get_word_sep(buf, sizeof(buf), "/", &p);
179 switch (codec->codec_type) {
180 case AVMEDIA_TYPE_AUDIO:
181 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
182 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
183 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
185 codec->sample_rate = i;
186 av_set_pts_info(st, 32, 1, codec->sample_rate);
187 get_word_sep(buf, sizeof(buf), "/", &p);
191 // TODO: there is a bug here; if it is a mono stream, and
192 // less than 22000Hz, faad upconverts to stereo and twice
193 // the frequency. No problem, but the sample rate is being
194 // set here by the sdp line. Patch on its way. (rdm)
196 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
198 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
201 case AVMEDIA_TYPE_VIDEO:
202 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
204 av_set_pts_info(st, 32, 1, i);
212 /* parse the attribute line from the fmtp a line of an sdp response. This
213 * is broken out as a function because it is used in rtp_h264.c, which is
215 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
216 char *value, int value_size)
218 *p += strspn(*p, SPACE_CHARS);
220 get_word_sep(attr, attr_size, "=", p);
223 get_word_sep(value, value_size, ";", p);
231 typedef struct SDPParseState {
233 struct sockaddr_storage default_ip;
235 int skip_media; ///< set if an unknown m= line occurs
238 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
239 int letter, const char *buf)
241 RTSPState *rt = s->priv_data;
242 char buf1[64], st_type[64];
244 enum AVMediaType codec_type;
248 struct sockaddr_storage sdp_ip;
251 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
254 if (s1->skip_media && letter != 'm')
258 get_word(buf1, sizeof(buf1), &p);
259 if (strcmp(buf1, "IN") != 0)
261 get_word(buf1, sizeof(buf1), &p);
262 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
264 get_word_sep(buf1, sizeof(buf1), "/", &p);
265 if (get_sockaddr(buf1, &sdp_ip))
270 get_word_sep(buf1, sizeof(buf1), "/", &p);
273 if (s->nb_streams == 0) {
274 s1->default_ip = sdp_ip;
275 s1->default_ttl = ttl;
277 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
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, rt->nb_rtsp_streams - 1);
329 rtsp_st->stream_index = st->index;
330 st->codec->codec_type = codec_type;
331 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
332 RTPDynamicProtocolHandler *handler;
333 /* if standard payload type, we can find the codec right now */
334 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
335 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
336 st->codec->sample_rate > 0)
337 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
338 /* Even static payload types may need a custom depacketizer */
339 handler = ff_rtp_handler_find_by_id(
340 rtsp_st->sdp_payload_type, st->codec->codec_type);
341 init_rtp_handler(handler, rtsp_st, st->codec);
344 /* put a default control url */
345 av_strlcpy(rtsp_st->control_url, rt->control_uri,
346 sizeof(rtsp_st->control_url));
349 if (av_strstart(p, "control:", &p)) {
350 if (s->nb_streams == 0) {
351 if (!strncmp(p, "rtsp://", 7))
352 av_strlcpy(rt->control_uri, p,
353 sizeof(rt->control_uri));
356 /* get the control url */
357 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
359 /* XXX: may need to add full url resolution */
360 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
362 if (proto[0] == '\0') {
363 /* relative control URL */
364 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
365 av_strlcat(rtsp_st->control_url, "/",
366 sizeof(rtsp_st->control_url));
367 av_strlcat(rtsp_st->control_url, p,
368 sizeof(rtsp_st->control_url));
370 av_strlcpy(rtsp_st->control_url, p,
371 sizeof(rtsp_st->control_url));
373 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
374 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
375 get_word(buf1, sizeof(buf1), &p);
376 payload_type = atoi(buf1);
377 st = s->streams[s->nb_streams - 1];
378 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
379 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
380 } else if (av_strstart(p, "fmtp:", &p) ||
381 av_strstart(p, "framesize:", &p)) {
382 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
383 // let dynamic protocol handlers have a stab at the line.
384 get_word(buf1, sizeof(buf1), &p);
385 payload_type = atoi(buf1);
386 for (i = 0; i < rt->nb_rtsp_streams; i++) {
387 rtsp_st = rt->rtsp_streams[i];
388 if (rtsp_st->sdp_payload_type == payload_type &&
389 rtsp_st->dynamic_handler &&
390 rtsp_st->dynamic_handler->parse_sdp_a_line)
391 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
392 rtsp_st->dynamic_protocol_context, buf);
394 } else if (av_strstart(p, "range:", &p)) {
397 // this is so that seeking on a streamed file can work.
398 rtsp_parse_range_npt(p, &start, &end);
399 s->start_time = start;
400 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
401 s->duration = (end == AV_NOPTS_VALUE) ?
402 AV_NOPTS_VALUE : end - start;
403 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
405 rt->transport = RTSP_TRANSPORT_RDT;
406 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
408 st = s->streams[s->nb_streams - 1];
409 st->codec->sample_rate = atoi(p);
411 if (rt->server_type == RTSP_SERVER_WMS)
412 ff_wms_parse_sdp_a_line(s, p);
413 if (s->nb_streams > 0) {
414 if (rt->server_type == RTSP_SERVER_REAL)
415 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
417 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
418 if (rtsp_st->dynamic_handler &&
419 rtsp_st->dynamic_handler->parse_sdp_a_line)
420 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
422 rtsp_st->dynamic_protocol_context, buf);
430 * Parse the sdp description and allocate the rtp streams and the
431 * pollfd array used for udp ones.
434 int ff_sdp_parse(AVFormatContext *s, const char *content)
436 RTSPState *rt = s->priv_data;
439 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
440 * contain long SDP lines containing complete ASF Headers (several
441 * kB) or arrays of MDPR (RM stream descriptor) headers plus
442 * "rulebooks" describing their properties. Therefore, the SDP line
445 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
446 * in rtpdec_xiph.c. */
448 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
450 memset(s1, 0, sizeof(SDPParseState));
453 p += strspn(p, SPACE_CHARS);
461 /* get the content */
463 while (*p != '\n' && *p != '\r' && *p != '\0') {
464 if ((q - buf) < sizeof(buf) - 1)
469 sdp_parse_line(s, s1, letter, buf);
471 while (*p != '\n' && *p != '\0')
476 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
477 if (!rt->p) return AVERROR(ENOMEM);
480 #endif /* CONFIG_RTPDEC */
482 void ff_rtsp_undo_setup(AVFormatContext *s)
484 RTSPState *rt = s->priv_data;
487 for (i = 0; i < rt->nb_rtsp_streams; i++) {
488 RTSPStream *rtsp_st = rt->rtsp_streams[i];
491 if (rtsp_st->transport_priv) {
493 AVFormatContext *rtpctx = rtsp_st->transport_priv;
494 av_write_trailer(rtpctx);
495 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
497 url_close_dyn_buf(rtpctx->pb, &ptr);
500 url_fclose(rtpctx->pb);
502 avformat_free_context(rtpctx);
503 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
504 ff_rdt_parse_close(rtsp_st->transport_priv);
505 else if (CONFIG_RTPDEC)
506 rtp_parse_close(rtsp_st->transport_priv);
508 rtsp_st->transport_priv = NULL;
509 if (rtsp_st->rtp_handle)
510 url_close(rtsp_st->rtp_handle);
511 rtsp_st->rtp_handle = NULL;
515 /* close and free RTSP streams */
516 void ff_rtsp_close_streams(AVFormatContext *s)
518 RTSPState *rt = s->priv_data;
522 ff_rtsp_undo_setup(s);
523 for (i = 0; i < rt->nb_rtsp_streams; i++) {
524 rtsp_st = rt->rtsp_streams[i];
526 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
527 rtsp_st->dynamic_handler->close(
528 rtsp_st->dynamic_protocol_context);
532 av_free(rt->rtsp_streams);
534 av_close_input_stream (rt->asf_ctx);
538 av_free(rt->recvbuf);
541 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
543 RTSPState *rt = s->priv_data;
546 /* open the RTP context */
547 if (rtsp_st->stream_index >= 0)
548 st = s->streams[rtsp_st->stream_index];
550 s->ctx_flags |= AVFMTCTX_NOHEADER;
552 if (s->oformat && CONFIG_RTSP_MUXER) {
553 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
555 RTSP_TCP_MAX_PACKET_SIZE);
556 /* Ownership of rtp_handle is passed to the rtp mux context */
557 rtsp_st->rtp_handle = NULL;
558 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
559 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
560 rtsp_st->dynamic_protocol_context,
561 rtsp_st->dynamic_handler);
562 else if (CONFIG_RTPDEC)
563 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
564 rtsp_st->sdp_payload_type,
565 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
566 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
568 if (!rtsp_st->transport_priv) {
569 return AVERROR(ENOMEM);
570 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
571 if (rtsp_st->dynamic_handler) {
572 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
573 rtsp_st->dynamic_protocol_context,
574 rtsp_st->dynamic_handler);
581 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
582 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
588 p += strspn(p, SPACE_CHARS);
589 v = strtol(p, (char **)&p, 10);
593 v = strtol(p, (char **)&p, 10);
602 /* XXX: only one transport specification is parsed */
603 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
605 char transport_protocol[16];
607 char lower_transport[16];
609 RTSPTransportField *th;
612 reply->nb_transports = 0;
615 p += strspn(p, SPACE_CHARS);
619 th = &reply->transports[reply->nb_transports];
621 get_word_sep(transport_protocol, sizeof(transport_protocol),
623 if (!strcasecmp (transport_protocol, "rtp")) {
624 get_word_sep(profile, sizeof(profile), "/;,", &p);
625 lower_transport[0] = '\0';
626 /* rtp/avp/<protocol> */
628 get_word_sep(lower_transport, sizeof(lower_transport),
631 th->transport = RTSP_TRANSPORT_RTP;
632 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
633 !strcasecmp (transport_protocol, "x-real-rdt")) {
634 /* x-pn-tng/<protocol> */
635 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
637 th->transport = RTSP_TRANSPORT_RDT;
639 if (!strcasecmp(lower_transport, "TCP"))
640 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
642 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
646 /* get each parameter */
647 while (*p != '\0' && *p != ',') {
648 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
649 if (!strcmp(parameter, "port")) {
652 rtsp_parse_range(&th->port_min, &th->port_max, &p);
654 } else if (!strcmp(parameter, "client_port")) {
657 rtsp_parse_range(&th->client_port_min,
658 &th->client_port_max, &p);
660 } else if (!strcmp(parameter, "server_port")) {
663 rtsp_parse_range(&th->server_port_min,
664 &th->server_port_max, &p);
666 } else if (!strcmp(parameter, "interleaved")) {
669 rtsp_parse_range(&th->interleaved_min,
670 &th->interleaved_max, &p);
672 } else if (!strcmp(parameter, "multicast")) {
673 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
674 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
675 } else if (!strcmp(parameter, "ttl")) {
678 th->ttl = strtol(p, (char **)&p, 10);
680 } else if (!strcmp(parameter, "destination")) {
683 get_word_sep(buf, sizeof(buf), ";,", &p);
684 get_sockaddr(buf, &th->destination);
686 } else if (!strcmp(parameter, "source")) {
689 get_word_sep(buf, sizeof(buf), ";,", &p);
690 av_strlcpy(th->source, buf, sizeof(th->source));
694 while (*p != ';' && *p != '\0' && *p != ',')
702 reply->nb_transports++;
706 static void handle_rtp_info(RTSPState *rt, const char *url,
707 uint32_t seq, uint32_t rtptime)
710 if (!rtptime || !url[0])
712 if (rt->transport != RTSP_TRANSPORT_RTP)
714 for (i = 0; i < rt->nb_rtsp_streams; i++) {
715 RTSPStream *rtsp_st = rt->rtsp_streams[i];
716 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
719 if (!strcmp(rtsp_st->control_url, url)) {
720 rtpctx->base_timestamp = rtptime;
726 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
729 char key[20], value[1024], url[1024] = "";
730 uint32_t seq = 0, rtptime = 0;
733 p += strspn(p, SPACE_CHARS);
736 get_word_sep(key, sizeof(key), "=", &p);
740 get_word_sep(value, sizeof(value), ";, ", &p);
742 if (!strcmp(key, "url"))
743 av_strlcpy(url, value, sizeof(url));
744 else if (!strcmp(key, "seq"))
745 seq = strtol(value, NULL, 10);
746 else if (!strcmp(key, "rtptime"))
747 rtptime = strtol(value, NULL, 10);
749 handle_rtp_info(rt, url, seq, rtptime);
758 handle_rtp_info(rt, url, seq, rtptime);
761 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
762 RTSPState *rt, const char *method)
766 /* NOTE: we do case independent match for broken servers */
768 if (av_stristart(p, "Session:", &p)) {
770 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
771 if (av_stristart(p, ";timeout=", &p) &&
772 (t = strtol(p, NULL, 10)) > 0) {
775 } else if (av_stristart(p, "Content-Length:", &p)) {
776 reply->content_length = strtol(p, NULL, 10);
777 } else if (av_stristart(p, "Transport:", &p)) {
778 rtsp_parse_transport(reply, p);
779 } else if (av_stristart(p, "CSeq:", &p)) {
780 reply->seq = strtol(p, NULL, 10);
781 } else if (av_stristart(p, "Range:", &p)) {
782 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
783 } else if (av_stristart(p, "RealChallenge1:", &p)) {
784 p += strspn(p, SPACE_CHARS);
785 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
786 } else if (av_stristart(p, "Server:", &p)) {
787 p += strspn(p, SPACE_CHARS);
788 av_strlcpy(reply->server, p, sizeof(reply->server));
789 } else if (av_stristart(p, "Notice:", &p) ||
790 av_stristart(p, "X-Notice:", &p)) {
791 reply->notice = strtol(p, NULL, 10);
792 } else if (av_stristart(p, "Location:", &p)) {
793 p += strspn(p, SPACE_CHARS);
794 av_strlcpy(reply->location, p , sizeof(reply->location));
795 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
796 p += strspn(p, SPACE_CHARS);
797 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
798 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
799 p += strspn(p, SPACE_CHARS);
800 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
801 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
802 p += strspn(p, SPACE_CHARS);
803 if (method && !strcmp(method, "DESCRIBE"))
804 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
805 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
806 p += strspn(p, SPACE_CHARS);
807 if (method && !strcmp(method, "PLAY"))
808 rtsp_parse_rtp_info(rt, p);
812 /* skip a RTP/TCP interleaved packet */
813 void ff_rtsp_skip_packet(AVFormatContext *s)
815 RTSPState *rt = s->priv_data;
819 ret = url_read_complete(rt->rtsp_hd, buf, 3);
822 len = AV_RB16(buf + 1);
824 av_dlog(s, "skipping RTP packet len=%d\n", len);
829 if (len1 > sizeof(buf))
831 ret = url_read_complete(rt->rtsp_hd, buf, len1);
838 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
839 unsigned char **content_ptr,
840 int return_on_interleaved_data, const char *method)
842 RTSPState *rt = s->priv_data;
843 char buf[4096], buf1[1024], *q;
846 int ret, content_length, line_count = 0;
847 unsigned char *content = NULL;
849 memset(reply, 0, sizeof(*reply));
851 /* parse reply (XXX: use buffers) */
852 rt->last_reply[0] = '\0';
856 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
858 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
865 /* XXX: only parse it if first char on line ? */
866 if (return_on_interleaved_data) {
869 ff_rtsp_skip_packet(s);
870 } else if (ch != '\r') {
871 if ((q - buf) < sizeof(buf) - 1)
877 av_dlog(s, "line='%s'\n", buf);
879 /* test if last line */
883 if (line_count == 0) {
885 get_word(buf1, sizeof(buf1), &p);
886 get_word(buf1, sizeof(buf1), &p);
887 reply->status_code = atoi(buf1);
888 av_strlcpy(reply->reason, p, sizeof(reply->reason));
890 ff_rtsp_parse_line(reply, p, rt, method);
891 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
892 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
897 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
898 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
900 content_length = reply->content_length;
901 if (content_length > 0) {
902 /* leave some room for a trailing '\0' (useful for simple parsing) */
903 content = av_malloc(content_length + 1);
904 (void)url_read_complete(rt->rtsp_hd, content, content_length);
905 content[content_length] = '\0';
908 *content_ptr = content;
912 if (rt->seq != reply->seq) {
913 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
914 rt->seq, reply->seq);
918 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
919 reply->notice == 2104 /* Start-of-Stream Reached */ ||
920 reply->notice == 2306 /* Continuous Feed Terminated */) {
921 rt->state = RTSP_STATE_IDLE;
922 } else if (reply->notice >= 4400 && reply->notice < 5500) {
923 return AVERROR(EIO); /* data or server error */
924 } else if (reply->notice == 2401 /* Ticket Expired */ ||
925 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
926 return AVERROR(EPERM);
932 * Send a command to the RTSP server without waiting for the reply.
934 * @param s RTSP (de)muxer context
935 * @param method the method for the request
936 * @param url the target url for the request
937 * @param headers extra header lines to include in the request
938 * @param send_content if non-null, the data to send as request body content
939 * @param send_content_length the length of the send_content data, or 0 if
940 * send_content is null
942 * @return zero if success, nonzero otherwise
944 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
945 const char *method, const char *url,
947 const unsigned char *send_content,
948 int send_content_length)
950 RTSPState *rt = s->priv_data;
951 char buf[4096], *out_buf;
952 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
954 /* Add in RTSP headers */
957 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
959 av_strlcat(buf, headers, sizeof(buf));
960 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
961 if (rt->session_id[0] != '\0' && (!headers ||
962 !strstr(headers, "\nIf-Match:"))) {
963 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
966 char *str = ff_http_auth_create_response(&rt->auth_state,
967 rt->auth, url, method);
969 av_strlcat(buf, str, sizeof(buf));
972 if (send_content_length > 0 && send_content)
973 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
974 av_strlcat(buf, "\r\n", sizeof(buf));
976 /* base64 encode rtsp if tunneling */
977 if (rt->control_transport == RTSP_MODE_TUNNEL) {
978 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
982 av_dlog(s, "Sending:\n%s--\n", buf);
984 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
985 if (send_content_length > 0 && send_content) {
986 if (rt->control_transport == RTSP_MODE_TUNNEL) {
987 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
988 "with content data not supported\n");
989 return AVERROR_PATCHWELCOME;
991 url_write(rt->rtsp_hd_out, send_content, send_content_length);
993 rt->last_cmd_time = av_gettime();
998 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
999 const char *url, const char *headers)
1001 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1004 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1005 const char *headers, RTSPMessageHeader *reply,
1006 unsigned char **content_ptr)
1008 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1009 content_ptr, NULL, 0);
1012 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1013 const char *method, const char *url,
1015 RTSPMessageHeader *reply,
1016 unsigned char **content_ptr,
1017 const unsigned char *send_content,
1018 int send_content_length)
1020 RTSPState *rt = s->priv_data;
1021 HTTPAuthType cur_auth_type;
1025 cur_auth_type = rt->auth_state.auth_type;
1026 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1028 send_content_length)))
1031 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1034 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
1035 rt->auth_state.auth_type != HTTP_AUTH_NONE)
1038 if (reply->status_code > 400){
1039 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1043 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1050 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1052 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1053 int lower_transport, const char *real_challenge)
1055 RTSPState *rt = s->priv_data;
1056 int rtx, j, i, err, interleave = 0;
1057 RTSPStream *rtsp_st;
1058 RTSPMessageHeader reply1, *reply = &reply1;
1060 const char *trans_pref;
1062 if (rt->transport == RTSP_TRANSPORT_RDT)
1063 trans_pref = "x-pn-tng";
1065 trans_pref = "RTP/AVP";
1067 /* default timeout: 1 minute */
1070 /* for each stream, make the setup request */
1071 /* XXX: we assume the same server is used for the control of each
1074 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1075 char transport[2048];
1078 * WMS serves all UDP data over a single connection, the RTX, which
1079 * isn't necessarily the first in the SDP but has to be the first
1080 * to be set up, else the second/third SETUP will fail with a 461.
1082 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1083 rt->server_type == RTSP_SERVER_WMS) {
1086 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1087 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1089 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1093 if (rtx == rt->nb_rtsp_streams)
1094 return -1; /* no RTX found */
1095 rtsp_st = rt->rtsp_streams[rtx];
1097 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1099 rtsp_st = rt->rtsp_streams[i];
1102 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1105 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1106 port = reply->transports[0].client_port_min;
1110 /* first try in specified port range */
1111 if (RTSP_RTP_PORT_MIN != 0) {
1112 while (j <= RTSP_RTP_PORT_MAX) {
1113 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1114 "?localport=%d", j);
1115 /* we will use two ports per rtp stream (rtp and rtcp) */
1117 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1123 /* then try on any port */
1124 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1125 err = AVERROR_INVALIDDATA;
1129 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1135 port = rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1137 snprintf(transport, sizeof(transport) - 1,
1138 "%s/UDP;", trans_pref);
1139 if (rt->server_type != RTSP_SERVER_REAL)
1140 av_strlcat(transport, "unicast;", sizeof(transport));
1141 av_strlcatf(transport, sizeof(transport),
1142 "client_port=%d", port);
1143 if (rt->transport == RTSP_TRANSPORT_RTP &&
1144 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1145 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1149 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1150 /** For WMS streams, the application streams are only used for
1151 * UDP. When trying to set it up for TCP streams, the server
1152 * will return an error. Therefore, we skip those streams. */
1153 if (rt->server_type == RTSP_SERVER_WMS &&
1154 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1157 snprintf(transport, sizeof(transport) - 1,
1158 "%s/TCP;", trans_pref);
1159 if (rt->server_type == RTSP_SERVER_WMS)
1160 av_strlcat(transport, "unicast;", sizeof(transport));
1161 av_strlcatf(transport, sizeof(transport),
1162 "interleaved=%d-%d",
1163 interleave, interleave + 1);
1167 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1168 snprintf(transport, sizeof(transport) - 1,
1169 "%s/UDP;multicast", trans_pref);
1172 av_strlcat(transport, ";mode=receive", sizeof(transport));
1173 } else if (rt->server_type == RTSP_SERVER_REAL ||
1174 rt->server_type == RTSP_SERVER_WMS)
1175 av_strlcat(transport, ";mode=play", sizeof(transport));
1176 snprintf(cmd, sizeof(cmd),
1177 "Transport: %s\r\n",
1179 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1180 char real_res[41], real_csum[9];
1181 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1183 av_strlcatf(cmd, sizeof(cmd),
1185 "RealChallenge2: %s, sd=%s\r\n",
1186 rt->session_id, real_res, real_csum);
1188 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1189 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1192 } else if (reply->status_code != RTSP_STATUS_OK ||
1193 reply->nb_transports != 1) {
1194 err = AVERROR_INVALIDDATA;
1198 /* XXX: same protocol for all streams is required */
1200 if (reply->transports[0].lower_transport != rt->lower_transport ||
1201 reply->transports[0].transport != rt->transport) {
1202 err = AVERROR_INVALIDDATA;
1206 rt->lower_transport = reply->transports[0].lower_transport;
1207 rt->transport = reply->transports[0].transport;
1210 /* Fail if the server responded with another lower transport mode
1211 * than what we requested. */
1212 if (reply->transports[0].lower_transport != lower_transport) {
1213 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1214 err = AVERROR_INVALIDDATA;
1218 switch(reply->transports[0].lower_transport) {
1219 case RTSP_LOWER_TRANSPORT_TCP:
1220 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1221 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1224 case RTSP_LOWER_TRANSPORT_UDP: {
1225 char url[1024], options[30] = "";
1227 if (rt->filter_source)
1228 av_strlcpy(options, "?connect=1", sizeof(options));
1229 /* Use source address if specified */
1230 if (reply->transports[0].source[0]) {
1231 ff_url_join(url, sizeof(url), "rtp", NULL,
1232 reply->transports[0].source,
1233 reply->transports[0].server_port_min, options);
1235 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1236 reply->transports[0].server_port_min, options);
1238 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1239 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1240 err = AVERROR_INVALIDDATA;
1243 /* Try to initialize the connection state in a
1244 * potential NAT router by sending dummy packets.
1245 * RTP/RTCP dummy packets are used for RDT, too.
1247 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1249 rtp_send_punch_packets(rtsp_st->rtp_handle);
1252 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1253 char url[1024], namebuf[50];
1254 struct sockaddr_storage addr;
1257 if (reply->transports[0].destination.ss_family) {
1258 addr = reply->transports[0].destination;
1259 port = reply->transports[0].port_min;
1260 ttl = reply->transports[0].ttl;
1262 addr = rtsp_st->sdp_ip;
1263 port = rtsp_st->sdp_port;
1264 ttl = rtsp_st->sdp_ttl;
1266 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1267 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1268 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1269 port, "?ttl=%d", ttl);
1270 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1271 err = AVERROR_INVALIDDATA;
1278 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1282 if (reply->timeout > 0)
1283 rt->timeout = reply->timeout;
1285 if (rt->server_type == RTSP_SERVER_REAL)
1286 rt->need_subscription = 1;
1291 ff_rtsp_undo_setup(s);
1295 void ff_rtsp_close_connections(AVFormatContext *s)
1297 RTSPState *rt = s->priv_data;
1298 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1299 url_close(rt->rtsp_hd);
1300 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1303 int ff_rtsp_connect(AVFormatContext *s)
1305 RTSPState *rt = s->priv_data;
1306 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1307 char *option_list, *option, *filename;
1308 int port, err, tcp_fd;
1309 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1310 int lower_transport_mask = 0;
1311 char real_challenge[64] = "";
1312 struct sockaddr_storage peer;
1313 socklen_t peer_len = sizeof(peer);
1315 if (!ff_network_init())
1316 return AVERROR(EIO);
1318 rt->control_transport = RTSP_MODE_PLAIN;
1319 /* extract hostname and port */
1320 av_url_split(NULL, 0, auth, sizeof(auth),
1321 host, sizeof(host), &port, path, sizeof(path), s->filename);
1323 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1326 port = RTSP_DEFAULT_PORT;
1328 /* search for options */
1329 option_list = strrchr(path, '?');
1331 /* Strip out the RTSP specific options, write out the rest of
1332 * the options back into the same string. */
1333 filename = option_list;
1334 while (option_list) {
1335 /* move the option pointer */
1336 option = ++option_list;
1337 option_list = strchr(option_list, '&');
1341 /* handle the options */
1342 if (!strcmp(option, "udp")) {
1343 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1344 } else if (!strcmp(option, "multicast")) {
1345 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1346 } else if (!strcmp(option, "tcp")) {
1347 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1348 } else if(!strcmp(option, "http")) {
1349 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1350 rt->control_transport = RTSP_MODE_TUNNEL;
1351 } else if (!strcmp(option, "filter_src")) {
1352 rt->filter_source = 1;
1354 /* Write options back into the buffer, using memmove instead
1355 * of strcpy since the strings may overlap. */
1356 int len = strlen(option);
1357 memmove(++filename, option, len);
1359 if (option_list) *filename = '&';
1365 if (!lower_transport_mask)
1366 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1369 /* Only UDP or TCP - UDP multicast isn't supported. */
1370 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1371 (1 << RTSP_LOWER_TRANSPORT_TCP);
1372 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1373 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1374 "only UDP and TCP are supported for output.\n");
1375 err = AVERROR(EINVAL);
1380 /* Construct the URI used in request; this is similar to s->filename,
1381 * but with authentication credentials removed and RTSP specific options
1383 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1384 host, port, "%s", path);
1386 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1387 /* set up initial handshake for tunneling */
1388 char httpname[1024];
1389 char sessioncookie[17];
1392 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1393 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1394 av_get_random_seed(), av_get_random_seed());
1397 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1402 /* generate GET headers */
1403 snprintf(headers, sizeof(headers),
1404 "x-sessioncookie: %s\r\n"
1405 "Accept: application/x-rtsp-tunnelled\r\n"
1406 "Pragma: no-cache\r\n"
1407 "Cache-Control: no-cache\r\n",
1409 ff_http_set_headers(rt->rtsp_hd, headers);
1411 /* complete the connection */
1412 if (url_connect(rt->rtsp_hd)) {
1418 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1423 /* generate POST headers */
1424 snprintf(headers, sizeof(headers),
1425 "x-sessioncookie: %s\r\n"
1426 "Content-Type: application/x-rtsp-tunnelled\r\n"
1427 "Pragma: no-cache\r\n"
1428 "Cache-Control: no-cache\r\n"
1429 "Content-Length: 32767\r\n"
1430 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1432 ff_http_set_headers(rt->rtsp_hd_out, headers);
1433 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1435 /* Initialize the authentication state for the POST session. The HTTP
1436 * protocol implementation doesn't properly handle multi-pass
1437 * authentication for POST requests, since it would require one of
1439 * - implementing Expect: 100-continue, which many HTTP servers
1440 * don't support anyway, even less the RTSP servers that do HTTP
1442 * - sending the whole POST data until getting a 401 reply specifying
1443 * what authentication method to use, then resending all that data
1444 * - waiting for potential 401 replies directly after sending the
1445 * POST header (waiting for some unspecified time)
1446 * Therefore, we copy the full auth state, which works for both basic
1447 * and digest. (For digest, we would have to synchronize the nonce
1448 * count variable between the two sessions, if we'd do more requests
1449 * with the original session, though.)
1451 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1453 /* complete the connection */
1454 if (url_connect(rt->rtsp_hd_out)) {
1459 /* open the tcp connection */
1460 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1461 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1465 rt->rtsp_hd_out = rt->rtsp_hd;
1469 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1470 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1471 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1472 NULL, 0, NI_NUMERICHOST);
1475 /* request options supported by the server; this also detects server
1477 for (rt->server_type = RTSP_SERVER_RTP;;) {
1479 if (rt->server_type == RTSP_SERVER_REAL)
1482 * The following entries are required for proper
1483 * streaming from a Realmedia server. They are
1484 * interdependent in some way although we currently
1485 * don't quite understand how. Values were copied
1486 * from mplayer SVN r23589.
1487 * @param CompanyID is a 16-byte ID in base64
1488 * @param ClientChallenge is a 16-byte ID in hex
1490 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1491 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1492 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1493 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1495 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1496 if (reply->status_code != RTSP_STATUS_OK) {
1497 err = AVERROR_INVALIDDATA;
1501 /* detect server type if not standard-compliant RTP */
1502 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1503 rt->server_type = RTSP_SERVER_REAL;
1505 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1506 rt->server_type = RTSP_SERVER_WMS;
1507 } else if (rt->server_type == RTSP_SERVER_REAL)
1508 strcpy(real_challenge, reply->real_challenge);
1512 if (s->iformat && CONFIG_RTSP_DEMUXER)
1513 err = ff_rtsp_setup_input_streams(s, reply);
1514 else if (CONFIG_RTSP_MUXER)
1515 err = ff_rtsp_setup_output_streams(s, host);
1520 int lower_transport = ff_log2_tab[lower_transport_mask &
1521 ~(lower_transport_mask - 1)];
1523 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1524 rt->server_type == RTSP_SERVER_REAL ?
1525 real_challenge : NULL);
1528 lower_transport_mask &= ~(1 << lower_transport);
1529 if (lower_transport_mask == 0 && err == 1) {
1530 err = FF_NETERROR(EPROTONOSUPPORT);
1535 rt->lower_transport_mask = lower_transport_mask;
1536 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1537 rt->state = RTSP_STATE_IDLE;
1538 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1541 ff_rtsp_close_streams(s);
1542 ff_rtsp_close_connections(s);
1543 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1544 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1545 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1553 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1556 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1557 uint8_t *buf, int buf_size, int64_t wait_end)
1559 RTSPState *rt = s->priv_data;
1560 RTSPStream *rtsp_st;
1561 int n, i, ret, tcp_fd, timeout_cnt = 0;
1563 struct pollfd *p = rt->p;
1566 if (url_interrupt_cb())
1567 return AVERROR(EINTR);
1568 if (wait_end && wait_end - av_gettime() < 0)
1569 return AVERROR(EAGAIN);
1572 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1573 p[max_p].fd = tcp_fd;
1574 p[max_p++].events = POLLIN;
1578 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1579 rtsp_st = rt->rtsp_streams[i];
1580 if (rtsp_st->rtp_handle) {
1581 p[max_p].fd = url_get_file_handle(rtsp_st->rtp_handle);
1582 p[max_p++].events = POLLIN;
1583 p[max_p].fd = rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1584 p[max_p++].events = POLLIN;
1587 n = poll(p, max_p, POLL_TIMEOUT_MS);
1589 int j = 1 - (tcp_fd == -1);
1591 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1592 rtsp_st = rt->rtsp_streams[i];
1593 if (rtsp_st->rtp_handle) {
1594 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1595 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1597 *prtsp_st = rtsp_st;
1604 #if CONFIG_RTSP_DEMUXER
1605 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1606 RTSPMessageHeader reply;
1608 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1611 /* XXX: parse message */
1612 if (rt->state != RTSP_STATE_STREAMING)
1616 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1617 return FF_NETERROR(ETIMEDOUT);
1618 } else if (n < 0 && errno != EINTR)
1619 return AVERROR(errno);
1623 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1625 RTSPState *rt = s->priv_data;
1627 RTSPStream *rtsp_st, *first_queue_st = NULL;
1628 int64_t wait_end = 0;
1630 if (rt->nb_byes == rt->nb_rtsp_streams)
1633 /* get next frames from the same RTP packet */
1634 if (rt->cur_transport_priv) {
1635 if (rt->transport == RTSP_TRANSPORT_RDT) {
1636 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1638 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1640 rt->cur_transport_priv = NULL;
1642 } else if (ret == 1) {
1645 rt->cur_transport_priv = NULL;
1648 if (rt->transport == RTSP_TRANSPORT_RTP) {
1650 int64_t first_queue_time = 0;
1651 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1652 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1656 queue_time = ff_rtp_queued_packet_time(rtpctx);
1657 if (queue_time && (queue_time - first_queue_time < 0 ||
1658 !first_queue_time)) {
1659 first_queue_time = queue_time;
1660 first_queue_st = rt->rtsp_streams[i];
1663 if (first_queue_time)
1664 wait_end = first_queue_time + s->max_delay;
1667 /* read next RTP packet */
1670 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1672 return AVERROR(ENOMEM);
1675 switch(rt->lower_transport) {
1677 #if CONFIG_RTSP_DEMUXER
1678 case RTSP_LOWER_TRANSPORT_TCP:
1679 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1682 case RTSP_LOWER_TRANSPORT_UDP:
1683 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1684 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1685 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1686 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1689 if (len == AVERROR(EAGAIN) && first_queue_st &&
1690 rt->transport == RTSP_TRANSPORT_RTP) {
1691 rtsp_st = first_queue_st;
1692 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1699 if (rt->transport == RTSP_TRANSPORT_RDT) {
1700 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1702 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1704 /* Either bad packet, or a RTCP packet. Check if the
1705 * first_rtcp_ntp_time field was initialized. */
1706 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1707 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1708 /* first_rtcp_ntp_time has been initialized for this stream,
1709 * copy the same value to all other uninitialized streams,
1710 * in order to map their timestamp origin to the same ntp time
1713 AVStream *st = NULL;
1714 if (rtsp_st->stream_index >= 0)
1715 st = s->streams[rtsp_st->stream_index];
1716 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1717 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1718 AVStream *st2 = NULL;
1719 if (rt->rtsp_streams[i]->stream_index >= 0)
1720 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1721 if (rtpctx2 && st && st2 &&
1722 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1723 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1724 rtpctx2->rtcp_ts_offset = av_rescale_q(
1725 rtpctx->rtcp_ts_offset, st->time_base,
1730 if (ret == -RTCP_BYE) {
1733 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1734 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1736 if (rt->nb_byes == rt->nb_rtsp_streams)
1745 /* more packets may follow, so we save the RTP context */
1746 rt->cur_transport_priv = rtsp_st->transport_priv;
1750 #endif /* CONFIG_RTPDEC */
1752 #if CONFIG_SDP_DEMUXER
1753 static int sdp_probe(AVProbeData *p1)
1755 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1757 /* we look for a line beginning "c=IN IP" */
1758 while (p < p_end && *p != '\0') {
1759 if (p + sizeof("c=IN IP") - 1 < p_end &&
1760 av_strstart(p, "c=IN IP", NULL))
1761 return AVPROBE_SCORE_MAX / 2;
1763 while (p < p_end - 1 && *p != '\n') p++;
1772 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1774 RTSPState *rt = s->priv_data;
1775 RTSPStream *rtsp_st;
1780 if (!ff_network_init())
1781 return AVERROR(EIO);
1783 /* read the whole sdp file */
1784 /* XXX: better loading */
1785 content = av_malloc(SDP_MAX_SIZE);
1786 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
1789 return AVERROR_INVALIDDATA;
1791 content[size] ='\0';
1793 err = ff_sdp_parse(s, content);
1797 /* open each RTP stream */
1798 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1800 rtsp_st = rt->rtsp_streams[i];
1802 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1803 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1804 ff_url_join(url, sizeof(url), "rtp", NULL,
1805 namebuf, rtsp_st->sdp_port,
1806 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
1808 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1809 err = AVERROR_INVALIDDATA;
1812 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1817 ff_rtsp_close_streams(s);
1822 static int sdp_read_close(AVFormatContext *s)
1824 ff_rtsp_close_streams(s);
1829 AVInputFormat ff_sdp_demuxer = {
1831 NULL_IF_CONFIG_SMALL("SDP"),
1835 ff_rtsp_fetch_packet,
1838 #endif /* CONFIG_SDP_DEMUXER */
1840 #if CONFIG_RTP_DEMUXER
1841 static int rtp_probe(AVProbeData *p)
1843 if (av_strstart(p->filename, "rtp:", NULL))
1844 return AVPROBE_SCORE_MAX;
1848 static int rtp_read_header(AVFormatContext *s,
1849 AVFormatParameters *ap)
1851 uint8_t recvbuf[1500];
1852 char host[500], sdp[500];
1854 URLContext* in = NULL;
1856 AVCodecContext codec;
1857 struct sockaddr_storage addr;
1859 socklen_t addrlen = sizeof(addr);
1861 if (!ff_network_init())
1862 return AVERROR(EIO);
1864 ret = url_open(&in, s->filename, URL_RDONLY);
1869 ret = url_read(in, recvbuf, sizeof(recvbuf));
1870 if (ret == AVERROR(EAGAIN))
1875 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1879 if ((recvbuf[0] & 0xc0) != 0x80) {
1880 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1885 payload_type = recvbuf[1] & 0x7f;
1888 getsockname(url_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
1892 memset(&codec, 0, sizeof(codec));
1893 if (ff_rtp_get_codec_info(&codec, payload_type)) {
1894 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
1895 "without an SDP file describing it\n",
1899 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
1900 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
1901 "properly you need an SDP file "
1905 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
1906 NULL, 0, s->filename);
1908 snprintf(sdp, sizeof(sdp),
1909 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1910 addr.ss_family == AF_INET ? 4 : 6, host,
1911 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
1912 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
1913 port, payload_type);
1914 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
1916 init_put_byte(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
1919 /* sdp_read_header initializes this again */
1922 ret = sdp_read_header(s, ap);
1933 AVInputFormat ff_rtp_demuxer = {
1935 NULL_IF_CONFIG_SMALL("RTP input format"),
1939 ff_rtsp_fetch_packet,
1941 .flags = AVFMT_NOFILE,
1943 #endif /* CONFIG_RTP_DEMUXER */