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"
44 //#define DEBUG_RTP_TCP
46 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
47 int rtsp_default_protocols
= (1 << RTSP_LOWER_TRANSPORT_UDP
);
50 /* Timeout values for socket select, in ms,
51 * and read_packet(), in seconds */
52 #define SELECT_TIMEOUT_MS 100
53 #define READ_PACKET_TIMEOUT_S 10
54 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
55 #define SDP_MAX_SIZE 16384
57 static void get_word_until_chars(char *buf
, int buf_size
,
58 const char *sep
, const char **pp
)
64 p
+= strspn(p
, SPACE_CHARS
);
66 while (!strchr(sep
, *p
) && *p
!= '\0') {
67 if ((q
- buf
) < buf_size
- 1)
76 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
79 if (**pp
== '/') (*pp
)++;
80 get_word_until_chars(buf
, buf_size
, sep
, pp
);
83 static void get_word(char *buf
, int buf_size
, const char **pp
)
85 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
88 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
89 static int sdp_parse_rtpmap(AVFormatContext
*s
,
90 AVCodecContext
*codec
, RTSPStream
*rtsp_st
,
91 int payload_type
, const char *p
)
98 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
99 * see if we can handle this kind of payload.
100 * The space should normally not be there but some Real streams or
101 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
102 * have a trailing space. */
103 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
104 if (payload_type
>= RTP_PT_PRIVATE
) {
105 RTPDynamicProtocolHandler
*handler
;
106 for (handler
= RTPFirstDynamicPayloadHandler
;
107 handler
; handler
= handler
->next
) {
108 if (!strcasecmp(buf
, handler
->enc_name
) &&
109 codec
->codec_type
== handler
->codec_type
) {
110 codec
->codec_id
= handler
->codec_id
;
111 rtsp_st
->dynamic_handler
= handler
;
113 rtsp_st
->dynamic_protocol_context
= handler
->open();
118 /* We are in a standard case
119 * (from http://www.iana.org/assignments/rtp-parameters). */
120 /* search into AVRtpPayloadTypes[] */
121 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
124 c
= avcodec_find_decoder(codec
->codec_id
);
130 get_word_sep(buf
, sizeof(buf
), "/", &p
);
132 switch (codec
->codec_type
) {
133 case AVMEDIA_TYPE_AUDIO
:
134 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
135 codec
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
136 codec
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
138 codec
->sample_rate
= i
;
139 get_word_sep(buf
, sizeof(buf
), "/", &p
);
143 // TODO: there is a bug here; if it is a mono stream, and
144 // less than 22000Hz, faad upconverts to stereo and twice
145 // the frequency. No problem, but the sample rate is being
146 // set here by the sdp line. Patch on its way. (rdm)
148 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
150 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
153 case AVMEDIA_TYPE_VIDEO
:
154 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
162 /* parse the attribute line from the fmtp a line of an sdp response. This
163 * is broken out as a function because it is used in rtp_h264.c, which is
165 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
166 char *value
, int value_size
)
168 *p
+= strspn(*p
, SPACE_CHARS
);
170 get_word_sep(attr
, attr_size
, "=", p
);
173 get_word_sep(value
, value_size
, ";", p
);
181 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
183 * Used for seeking in the rtp stream.
185 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
189 p
+= strspn(p
, SPACE_CHARS
);
190 if (!av_stristart(p
, "npt=", &p
))
193 *start
= AV_NOPTS_VALUE
;
194 *end
= AV_NOPTS_VALUE
;
196 get_word_sep(buf
, sizeof(buf
), "-", &p
);
197 *start
= parse_date(buf
, 1);
200 get_word_sep(buf
, sizeof(buf
), "-", &p
);
201 *end
= parse_date(buf
, 1);
203 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
204 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
207 typedef struct SDPParseState
{
209 struct in_addr default_ip
;
211 int skip_media
; ///< set if an unknown m= line occurs
214 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
215 int letter
, const char *buf
)
217 RTSPState
*rt
= s
->priv_data
;
218 char buf1
[64], st_type
[64];
220 enum AVMediaType codec_type
;
224 struct in_addr sdp_ip
;
227 dprintf(s
, "sdp: %c='%s'\n", letter
, buf
);
230 if (s1
->skip_media
&& letter
!= 'm')
234 get_word(buf1
, sizeof(buf1
), &p
);
235 if (strcmp(buf1
, "IN") != 0)
237 get_word(buf1
, sizeof(buf1
), &p
);
238 if (strcmp(buf1
, "IP4") != 0)
240 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
241 if (ff_inet_aton(buf1
, &sdp_ip
) == 0)
246 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
249 if (s
->nb_streams
== 0) {
250 s1
->default_ip
= sdp_ip
;
251 s1
->default_ttl
= ttl
;
253 st
= s
->streams
[s
->nb_streams
- 1];
254 rtsp_st
= st
->priv_data
;
255 rtsp_st
->sdp_ip
= sdp_ip
;
256 rtsp_st
->sdp_ttl
= ttl
;
260 av_metadata_set2(&s
->metadata
, "title", p
, 0);
263 if (s
->nb_streams
== 0) {
264 av_metadata_set2(&s
->metadata
, "comment", p
, 0);
271 get_word(st_type
, sizeof(st_type
), &p
);
272 if (!strcmp(st_type
, "audio")) {
273 codec_type
= AVMEDIA_TYPE_AUDIO
;
274 } else if (!strcmp(st_type
, "video")) {
275 codec_type
= AVMEDIA_TYPE_VIDEO
;
276 } else if (!strcmp(st_type
, "application")) {
277 codec_type
= AVMEDIA_TYPE_DATA
;
282 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
285 rtsp_st
->stream_index
= -1;
286 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
288 rtsp_st
->sdp_ip
= s1
->default_ip
;
289 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
291 get_word(buf1
, sizeof(buf1
), &p
); /* port */
292 rtsp_st
->sdp_port
= atoi(buf1
);
294 get_word(buf1
, sizeof(buf1
), &p
); /* protocol (ignored) */
296 /* XXX: handle list of formats */
297 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
298 rtsp_st
->sdp_payload_type
= atoi(buf1
);
300 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
301 /* no corresponding stream */
303 st
= av_new_stream(s
, 0);
306 st
->priv_data
= rtsp_st
;
307 rtsp_st
->stream_index
= st
->index
;
308 st
->codec
->codec_type
= codec_type
;
309 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
310 /* if standard payload type, we can find the codec right now */
311 ff_rtp_get_codec_info(st
->codec
, rtsp_st
->sdp_payload_type
);
314 /* put a default control url */
315 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
316 sizeof(rtsp_st
->control_url
));
319 if (av_strstart(p
, "control:", &p
)) {
320 if (s
->nb_streams
== 0) {
321 if (!strncmp(p
, "rtsp://", 7))
322 av_strlcpy(rt
->control_uri
, p
,
323 sizeof(rt
->control_uri
));
326 /* get the control url */
327 st
= s
->streams
[s
->nb_streams
- 1];
328 rtsp_st
= st
->priv_data
;
330 /* XXX: may need to add full url resolution */
331 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
333 if (proto
[0] == '\0') {
334 /* relative control URL */
335 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
336 av_strlcat(rtsp_st
->control_url
, "/",
337 sizeof(rtsp_st
->control_url
));
338 av_strlcat(rtsp_st
->control_url
, p
,
339 sizeof(rtsp_st
->control_url
));
341 av_strlcpy(rtsp_st
->control_url
, p
,
342 sizeof(rtsp_st
->control_url
));
344 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
345 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
346 get_word(buf1
, sizeof(buf1
), &p
);
347 payload_type
= atoi(buf1
);
348 st
= s
->streams
[s
->nb_streams
- 1];
349 rtsp_st
= st
->priv_data
;
350 sdp_parse_rtpmap(s
, st
->codec
, rtsp_st
, payload_type
, p
);
351 } else if (av_strstart(p
, "fmtp:", &p
) ||
352 av_strstart(p
, "framesize:", &p
)) {
353 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
354 // let dynamic protocol handlers have a stab at the line.
355 get_word(buf1
, sizeof(buf1
), &p
);
356 payload_type
= atoi(buf1
);
357 for (i
= 0; i
< s
->nb_streams
; i
++) {
359 rtsp_st
= st
->priv_data
;
360 if (rtsp_st
->sdp_payload_type
== payload_type
&&
361 rtsp_st
->dynamic_handler
&&
362 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
363 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, i
,
364 rtsp_st
->dynamic_protocol_context
, buf
);
366 } else if (av_strstart(p
, "range:", &p
)) {
369 // this is so that seeking on a streamed file can work.
370 rtsp_parse_range_npt(p
, &start
, &end
);
371 s
->start_time
= start
;
372 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
373 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
374 AV_NOPTS_VALUE
: end
- start
;
375 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
377 rt
->transport
= RTSP_TRANSPORT_RDT
;
379 if (rt
->server_type
== RTSP_SERVER_WMS
)
380 ff_wms_parse_sdp_a_line(s
, p
);
381 if (s
->nb_streams
> 0) {
382 if (rt
->server_type
== RTSP_SERVER_REAL
)
383 ff_real_parse_sdp_a_line(s
, s
->nb_streams
- 1, p
);
385 rtsp_st
= s
->streams
[s
->nb_streams
- 1]->priv_data
;
386 if (rtsp_st
->dynamic_handler
&&
387 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
388 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
390 rtsp_st
->dynamic_protocol_context
, buf
);
397 static int sdp_parse(AVFormatContext
*s
, const char *content
)
401 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
402 * contain long SDP lines containing complete ASF Headers (several
403 * kB) or arrays of MDPR (RM stream descriptor) headers plus
404 * "rulebooks" describing their properties. Therefore, the SDP line
407 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
408 * in rtpdec_xiph.c. */
410 SDPParseState sdp_parse_state
, *s1
= &sdp_parse_state
;
412 memset(s1
, 0, sizeof(SDPParseState
));
415 p
+= strspn(p
, SPACE_CHARS
);
423 /* get the content */
425 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
426 if ((q
- buf
) < sizeof(buf
) - 1)
431 sdp_parse_line(s
, s1
, letter
, buf
);
433 while (*p
!= '\n' && *p
!= '\0')
441 /* close and free RTSP streams */
442 void ff_rtsp_close_streams(AVFormatContext
*s
)
444 RTSPState
*rt
= s
->priv_data
;
448 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
449 rtsp_st
= rt
->rtsp_streams
[i
];
451 if (rtsp_st
->transport_priv
) {
453 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
454 av_write_trailer(rtpctx
);
455 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
457 url_close_dyn_buf(rtpctx
->pb
, &ptr
);
460 url_fclose(rtpctx
->pb
);
462 av_metadata_free(&rtpctx
->streams
[0]->metadata
);
463 av_metadata_free(&rtpctx
->metadata
);
464 av_free(rtpctx
->streams
[0]);
466 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
)
467 ff_rdt_parse_close(rtsp_st
->transport_priv
);
469 rtp_parse_close(rtsp_st
->transport_priv
);
471 if (rtsp_st
->rtp_handle
)
472 url_close(rtsp_st
->rtp_handle
);
473 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
)
474 rtsp_st
->dynamic_handler
->close(
475 rtsp_st
->dynamic_protocol_context
);
478 av_free(rt
->rtsp_streams
);
480 av_close_input_stream (rt
->asf_ctx
);
485 static void *rtsp_rtp_mux_open(AVFormatContext
*s
, AVStream
*st
,
488 RTSPState
*rt
= s
->priv_data
;
489 AVFormatContext
*rtpctx
;
491 AVOutputFormat
*rtp_format
= av_guess_format("rtp", NULL
, NULL
);
496 /* Allocate an AVFormatContext for each output stream */
497 rtpctx
= avformat_alloc_context();
501 rtpctx
->oformat
= rtp_format
;
502 if (!av_new_stream(rtpctx
, 0)) {
506 /* Copy the max delay setting; the rtp muxer reads this. */
507 rtpctx
->max_delay
= s
->max_delay
;
508 /* Copy other stream parameters. */
509 rtpctx
->streams
[0]->sample_aspect_ratio
= st
->sample_aspect_ratio
;
511 /* Set the synchronized start time. */
512 rtpctx
->start_time_realtime
= rt
->start_time
;
514 /* Remove the local codec, link to the original codec
515 * context instead, to give the rtp muxer access to
516 * codec parameters. */
517 av_free(rtpctx
->streams
[0]->codec
);
518 rtpctx
->streams
[0]->codec
= st
->codec
;
521 url_fdopen(&rtpctx
->pb
, handle
);
523 url_open_dyn_packet_buf(&rtpctx
->pb
, RTSP_TCP_MAX_PACKET_SIZE
);
524 ret
= av_write_header(rtpctx
);
528 url_fclose(rtpctx
->pb
);
531 url_close_dyn_buf(rtpctx
->pb
, &ptr
);
534 av_free(rtpctx
->streams
[0]);
539 /* Copy the RTP AVStream timebase back to the original AVStream */
540 st
->time_base
= rtpctx
->streams
[0]->time_base
;
544 static int rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
546 RTSPState
*rt
= s
->priv_data
;
549 /* open the RTP context */
550 if (rtsp_st
->stream_index
>= 0)
551 st
= s
->streams
[rtsp_st
->stream_index
];
553 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
556 rtsp_st
->transport_priv
= rtsp_rtp_mux_open(s
, st
, rtsp_st
->rtp_handle
);
557 /* Ownership of rtp_handle is passed to the rtp mux context */
558 rtsp_st
->rtp_handle
= NULL
;
559 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
)
560 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
561 rtsp_st
->dynamic_protocol_context
,
562 rtsp_st
->dynamic_handler
);
564 rtsp_st
->transport_priv
= rtp_parse_open(s
, st
, rtsp_st
->rtp_handle
,
565 rtsp_st
->sdp_payload_type
);
567 if (!rtsp_st
->transport_priv
) {
568 return AVERROR(ENOMEM
);
569 } else if (rt
->transport
!= RTSP_TRANSPORT_RDT
) {
570 if (rtsp_st
->dynamic_handler
) {
571 rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
572 rtsp_st
->dynamic_protocol_context
,
573 rtsp_st
->dynamic_handler
);
580 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
581 static int rtsp_probe(AVProbeData
*p
)
583 if (av_strstart(p
->filename
, "rtsp:", NULL
))
584 return AVPROBE_SCORE_MAX
;
588 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
594 p
+= strspn(p
, SPACE_CHARS
);
595 v
= strtol(p
, (char **)&p
, 10);
599 v
= strtol(p
, (char **)&p
, 10);
608 /* XXX: only one transport specification is parsed */
609 static void rtsp_parse_transport(RTSPMessageHeader
*reply
, const char *p
)
611 char transport_protocol
[16];
613 char lower_transport
[16];
615 RTSPTransportField
*th
;
618 reply
->nb_transports
= 0;
621 p
+= strspn(p
, SPACE_CHARS
);
625 th
= &reply
->transports
[reply
->nb_transports
];
627 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
629 if (!strcasecmp (transport_protocol
, "rtp")) {
630 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
631 lower_transport
[0] = '\0';
632 /* rtp/avp/<protocol> */
634 get_word_sep(lower_transport
, sizeof(lower_transport
),
637 th
->transport
= RTSP_TRANSPORT_RTP
;
638 } else if (!strcasecmp (transport_protocol
, "x-pn-tng") ||
639 !strcasecmp (transport_protocol
, "x-real-rdt")) {
640 /* x-pn-tng/<protocol> */
641 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
643 th
->transport
= RTSP_TRANSPORT_RDT
;
645 if (!strcasecmp(lower_transport
, "TCP"))
646 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
648 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
652 /* get each parameter */
653 while (*p
!= '\0' && *p
!= ',') {
654 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
655 if (!strcmp(parameter
, "port")) {
658 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
660 } else if (!strcmp(parameter
, "client_port")) {
663 rtsp_parse_range(&th
->client_port_min
,
664 &th
->client_port_max
, &p
);
666 } else if (!strcmp(parameter
, "server_port")) {
669 rtsp_parse_range(&th
->server_port_min
,
670 &th
->server_port_max
, &p
);
672 } else if (!strcmp(parameter
, "interleaved")) {
675 rtsp_parse_range(&th
->interleaved_min
,
676 &th
->interleaved_max
, &p
);
678 } else if (!strcmp(parameter
, "multicast")) {
679 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
680 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
681 } else if (!strcmp(parameter
, "ttl")) {
684 th
->ttl
= strtol(p
, (char **)&p
, 10);
686 } else if (!strcmp(parameter
, "destination")) {
687 struct in_addr ipaddr
;
691 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
692 if (ff_inet_aton(buf
, &ipaddr
))
693 th
->destination
= ntohl(ipaddr
.s_addr
);
696 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
704 reply
->nb_transports
++;
708 void ff_rtsp_parse_line(RTSPMessageHeader
*reply
, const char *buf
,
709 HTTPAuthState
*auth_state
)
713 /* NOTE: we do case independent match for broken servers */
715 if (av_stristart(p
, "Session:", &p
)) {
717 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
718 if (av_stristart(p
, ";timeout=", &p
) &&
719 (t
= strtol(p
, NULL
, 10)) > 0) {
722 } else if (av_stristart(p
, "Content-Length:", &p
)) {
723 reply
->content_length
= strtol(p
, NULL
, 10);
724 } else if (av_stristart(p
, "Transport:", &p
)) {
725 rtsp_parse_transport(reply
, p
);
726 } else if (av_stristart(p
, "CSeq:", &p
)) {
727 reply
->seq
= strtol(p
, NULL
, 10);
728 } else if (av_stristart(p
, "Range:", &p
)) {
729 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
730 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
731 p
+= strspn(p
, SPACE_CHARS
);
732 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
733 } else if (av_stristart(p
, "Server:", &p
)) {
734 p
+= strspn(p
, SPACE_CHARS
);
735 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
736 } else if (av_stristart(p
, "Notice:", &p
) ||
737 av_stristart(p
, "X-Notice:", &p
)) {
738 reply
->notice
= strtol(p
, NULL
, 10);
739 } else if (av_stristart(p
, "Location:", &p
)) {
740 p
+= strspn(p
, SPACE_CHARS
);
741 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
742 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && auth_state
) {
743 p
+= strspn(p
, SPACE_CHARS
);
744 ff_http_auth_handle_header(auth_state
, "WWW-Authenticate", p
);
745 } else if (av_stristart(p
, "Authentication-Info:", &p
) && auth_state
) {
746 p
+= strspn(p
, SPACE_CHARS
);
747 ff_http_auth_handle_header(auth_state
, "Authentication-Info", p
);
751 /* skip a RTP/TCP interleaved packet */
752 void ff_rtsp_skip_packet(AVFormatContext
*s
)
754 RTSPState
*rt
= s
->priv_data
;
758 ret
= url_read_complete(rt
->rtsp_hd
, buf
, 3);
761 len
= AV_RB16(buf
+ 1);
763 dprintf(s
, "skipping RTP packet len=%d\n", len
);
768 if (len1
> sizeof(buf
))
770 ret
= url_read_complete(rt
->rtsp_hd
, buf
, len1
);
777 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
778 unsigned char **content_ptr
,
779 int return_on_interleaved_data
)
781 RTSPState
*rt
= s
->priv_data
;
782 char buf
[4096], buf1
[1024], *q
;
785 int ret
, content_length
, line_count
= 0;
786 unsigned char *content
= NULL
;
788 memset(reply
, 0, sizeof(*reply
));
790 /* parse reply (XXX: use buffers) */
791 rt
->last_reply
[0] = '\0';
795 ret
= url_read_complete(rt
->rtsp_hd
, &ch
, 1);
797 dprintf(s
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
804 /* XXX: only parse it if first char on line ? */
805 if (return_on_interleaved_data
) {
808 ff_rtsp_skip_packet(s
);
809 } else if (ch
!= '\r') {
810 if ((q
- buf
) < sizeof(buf
) - 1)
816 dprintf(s
, "line='%s'\n", buf
);
818 /* test if last line */
822 if (line_count
== 0) {
824 get_word(buf1
, sizeof(buf1
), &p
);
825 get_word(buf1
, sizeof(buf1
), &p
);
826 reply
->status_code
= atoi(buf1
);
827 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
829 ff_rtsp_parse_line(reply
, p
, &rt
->auth_state
);
830 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
831 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
836 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0')
837 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
839 content_length
= reply
->content_length
;
840 if (content_length
> 0) {
841 /* leave some room for a trailing '\0' (useful for simple parsing) */
842 content
= av_malloc(content_length
+ 1);
843 (void)url_read_complete(rt
->rtsp_hd
, content
, content_length
);
844 content
[content_length
] = '\0';
847 *content_ptr
= content
;
851 if (rt
->seq
!= reply
->seq
) {
852 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
853 rt
->seq
, reply
->seq
);
857 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
858 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
859 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
860 rt
->state
= RTSP_STATE_IDLE
;
861 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
862 return AVERROR(EIO
); /* data or server error */
863 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
864 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
865 return AVERROR(EPERM
);
870 int ff_rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
871 const char *method
, const char *url
,
873 const unsigned char *send_content
,
874 int send_content_length
)
876 RTSPState
*rt
= s
->priv_data
;
877 char buf
[4096], *out_buf
;
878 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
880 /* Add in RTSP headers */
883 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
885 av_strlcat(buf
, headers
, sizeof(buf
));
886 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
887 if (rt
->session_id
[0] != '\0' && (!headers
||
888 !strstr(headers
, "\nIf-Match:"))) {
889 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
892 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
893 rt
->auth
, url
, method
);
895 av_strlcat(buf
, str
, sizeof(buf
));
898 if (send_content_length
> 0 && send_content
)
899 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
900 av_strlcat(buf
, "\r\n", sizeof(buf
));
902 /* base64 encode rtsp if tunneling */
903 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
904 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
908 dprintf(s
, "Sending:\n%s--\n", buf
);
910 url_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
911 if (send_content_length
> 0 && send_content
) {
912 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
913 av_log(s
, AV_LOG_ERROR
, "tunneling of RTSP requests "
914 "with content data not supported\n");
915 return AVERROR_PATCHWELCOME
;
917 url_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
919 rt
->last_cmd_time
= av_gettime();
924 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
925 const char *url
, const char *headers
)
927 return ff_rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
930 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
931 const char *headers
, RTSPMessageHeader
*reply
,
932 unsigned char **content_ptr
)
934 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
935 content_ptr
, NULL
, 0);
938 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
939 const char *method
, const char *url
,
941 RTSPMessageHeader
*reply
,
942 unsigned char **content_ptr
,
943 const unsigned char *send_content
,
944 int send_content_length
)
946 RTSPState
*rt
= s
->priv_data
;
947 HTTPAuthType cur_auth_type
;
951 cur_auth_type
= rt
->auth_state
.auth_type
;
952 if ((ret
= ff_rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
954 send_content_length
)))
957 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0) ) < 0)
960 if (reply
->status_code
== 401 && cur_auth_type
== HTTP_AUTH_NONE
&&
961 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
)
964 if (reply
->status_code
> 400){
965 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
969 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
976 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
978 static int make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
979 int lower_transport
, const char *real_challenge
)
981 RTSPState
*rt
= s
->priv_data
;
982 int rtx
, j
, i
, err
, interleave
= 0;
984 RTSPMessageHeader reply1
, *reply
= &reply1
;
986 const char *trans_pref
;
988 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
989 trans_pref
= "x-pn-tng";
991 trans_pref
= "RTP/AVP";
993 /* default timeout: 1 minute */
996 /* for each stream, make the setup request */
997 /* XXX: we assume the same server is used for the control of each
1000 for (j
= RTSP_RTP_PORT_MIN
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1001 char transport
[2048];
1004 * WMS serves all UDP data over a single connection, the RTX, which
1005 * isn't necessarily the first in the SDP but has to be the first
1006 * to be set up, else the second/third SETUP will fail with a 461.
1008 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1009 rt
->server_type
== RTSP_SERVER_WMS
) {
1012 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1013 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1015 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1019 if (rtx
== rt
->nb_rtsp_streams
)
1020 return -1; /* no RTX found */
1021 rtsp_st
= rt
->rtsp_streams
[rtx
];
1023 rtsp_st
= rt
->rtsp_streams
[i
> rtx ? i
: i
- 1];
1025 rtsp_st
= rt
->rtsp_streams
[i
];
1028 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1031 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1032 port
= reply
->transports
[0].client_port_min
;
1036 /* first try in specified port range */
1037 if (RTSP_RTP_PORT_MIN
!= 0) {
1038 while (j
<= RTSP_RTP_PORT_MAX
) {
1039 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1040 "?localport=%d", j
);
1041 /* we will use two ports per rtp stream (rtp and rtcp) */
1043 if (url_open(&rtsp_st
->rtp_handle
, buf
, URL_RDWR
) == 0)
1049 /* then try on any port */
1050 if (url_open(&rtsp_st
->rtp_handle
, "rtp://", URL_RDONLY
) < 0) {
1051 err
= AVERROR_INVALIDDATA
;
1057 port
= rtp_get_local_port(rtsp_st
->rtp_handle
);
1059 snprintf(transport
, sizeof(transport
) - 1,
1060 "%s/UDP;", trans_pref
);
1061 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1062 av_strlcat(transport
, "unicast;", sizeof(transport
));
1063 av_strlcatf(transport
, sizeof(transport
),
1064 "client_port=%d", port
);
1065 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1066 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1067 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1071 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1072 /** For WMS streams, the application streams are only used for
1073 * UDP. When trying to set it up for TCP streams, the server
1074 * will return an error. Therefore, we skip those streams. */
1075 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1076 s
->streams
[rtsp_st
->stream_index
]->codec
->codec_type
==
1079 snprintf(transport
, sizeof(transport
) - 1,
1080 "%s/TCP;", trans_pref
);
1081 if (rt
->server_type
== RTSP_SERVER_WMS
)
1082 av_strlcat(transport
, "unicast;", sizeof(transport
));
1083 av_strlcatf(transport
, sizeof(transport
),
1084 "interleaved=%d-%d",
1085 interleave
, interleave
+ 1);
1089 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1090 snprintf(transport
, sizeof(transport
) - 1,
1091 "%s/UDP;multicast", trans_pref
);
1094 av_strlcat(transport
, ";mode=receive", sizeof(transport
));
1095 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1096 rt
->server_type
== RTSP_SERVER_WMS
)
1097 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1098 snprintf(cmd
, sizeof(cmd
),
1099 "Transport: %s\r\n",
1101 if (i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
) {
1102 char real_res
[41], real_csum
[9];
1103 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1105 av_strlcatf(cmd
, sizeof(cmd
),
1107 "RealChallenge2: %s, sd=%s\r\n",
1108 rt
->session_id
, real_res
, real_csum
);
1110 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1111 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1114 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1115 reply
->nb_transports
!= 1) {
1116 err
= AVERROR_INVALIDDATA
;
1120 /* XXX: same protocol for all streams is required */
1122 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1123 reply
->transports
[0].transport
!= rt
->transport
) {
1124 err
= AVERROR_INVALIDDATA
;
1128 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1129 rt
->transport
= reply
->transports
[0].transport
;
1132 /* close RTP connection if not chosen */
1133 if (reply
->transports
[0].lower_transport
!= RTSP_LOWER_TRANSPORT_UDP
&&
1134 (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)) {
1135 url_close(rtsp_st
->rtp_handle
);
1136 rtsp_st
->rtp_handle
= NULL
;
1139 switch(reply
->transports
[0].lower_transport
) {
1140 case RTSP_LOWER_TRANSPORT_TCP
:
1141 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1142 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1145 case RTSP_LOWER_TRANSPORT_UDP
: {
1148 /* XXX: also use address if specified */
1149 ff_url_join(url
, sizeof(url
), "rtp", NULL
, host
,
1150 reply
->transports
[0].server_port_min
, NULL
);
1151 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1152 rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1153 err
= AVERROR_INVALIDDATA
;
1156 /* Try to initialize the connection state in a
1157 * potential NAT router by sending dummy packets.
1158 * RTP/RTCP dummy packets are used for RDT, too.
1160 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) && s
->iformat
)
1161 rtp_send_punch_packets(rtsp_st
->rtp_handle
);
1164 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1169 if (reply
->transports
[0].destination
) {
1170 in
.s_addr
= htonl(reply
->transports
[0].destination
);
1171 port
= reply
->transports
[0].port_min
;
1172 ttl
= reply
->transports
[0].ttl
;
1174 in
= rtsp_st
->sdp_ip
;
1175 port
= rtsp_st
->sdp_port
;
1176 ttl
= rtsp_st
->sdp_ttl
;
1178 ff_url_join(url
, sizeof(url
), "rtp", NULL
, inet_ntoa(in
),
1179 port
, "?ttl=%d", ttl
);
1180 if (url_open(&rtsp_st
->rtp_handle
, url
, URL_RDWR
) < 0) {
1181 err
= AVERROR_INVALIDDATA
;
1188 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
1192 if (reply
->timeout
> 0)
1193 rt
->timeout
= reply
->timeout
;
1195 if (rt
->server_type
== RTSP_SERVER_REAL
)
1196 rt
->need_subscription
= 1;
1201 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1202 if (rt
->rtsp_streams
[i
]->rtp_handle
) {
1203 url_close(rt
->rtsp_streams
[i
]->rtp_handle
);
1204 rt
->rtsp_streams
[i
]->rtp_handle
= NULL
;
1210 static int rtsp_read_play(AVFormatContext
*s
)
1212 RTSPState
*rt
= s
->priv_data
;
1213 RTSPMessageHeader reply1
, *reply
= &reply1
;
1217 av_log(s
, AV_LOG_DEBUG
, "hello state=%d\n", rt
->state
);
1219 if (!(rt
->server_type
== RTSP_SERVER_REAL
&& rt
->need_subscription
)) {
1220 if (rt
->state
== RTSP_STATE_PAUSED
) {
1223 snprintf(cmd
, sizeof(cmd
),
1224 "Range: npt=%0.3f-\r\n",
1225 (double)rt
->seek_timestamp
/ AV_TIME_BASE
);
1227 ff_rtsp_send_cmd(s
, "PLAY", rt
->control_uri
, cmd
, reply
, NULL
);
1228 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1231 if (reply
->range_start
!= AV_NOPTS_VALUE
&&
1232 rt
->transport
== RTSP_TRANSPORT_RTP
) {
1233 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1234 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
1235 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1236 AVStream
*st
= NULL
;
1239 if (rtsp_st
->stream_index
>= 0)
1240 st
= s
->streams
[rtsp_st
->stream_index
];
1241 rtpctx
->last_rtcp_ntp_time
= AV_NOPTS_VALUE
;
1242 rtpctx
->first_rtcp_ntp_time
= AV_NOPTS_VALUE
;
1244 rtpctx
->range_start_offset
= av_rescale_q(reply
->range_start
,
1250 rt
->state
= RTSP_STATE_STREAMING
;
1254 static int rtsp_setup_input_streams(AVFormatContext
*s
, RTSPMessageHeader
*reply
)
1256 RTSPState
*rt
= s
->priv_data
;
1258 unsigned char *content
= NULL
;
1261 /* describe the stream */
1262 snprintf(cmd
, sizeof(cmd
),
1263 "Accept: application/sdp\r\n");
1264 if (rt
->server_type
== RTSP_SERVER_REAL
) {
1266 * The Require: attribute is needed for proper streaming from
1267 * Realmedia servers.
1270 "Require: com.real.retain-entity-for-setup\r\n",
1273 ff_rtsp_send_cmd(s
, "DESCRIBE", rt
->control_uri
, cmd
, reply
, &content
);
1275 return AVERROR_INVALIDDATA
;
1276 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1278 return AVERROR_INVALIDDATA
;
1281 /* now we got the SDP description, we parse it */
1282 ret
= sdp_parse(s
, (const char *)content
);
1285 return AVERROR_INVALIDDATA
;
1290 static int rtsp_setup_output_streams(AVFormatContext
*s
, const char *addr
)
1292 RTSPState
*rt
= s
->priv_data
;
1293 RTSPMessageHeader reply1
, *reply
= &reply1
;
1296 AVFormatContext sdp_ctx
, *ctx_array
[1];
1298 rt
->start_time
= av_gettime();
1300 /* Announce the stream */
1301 sdp
= av_mallocz(SDP_MAX_SIZE
);
1303 return AVERROR(ENOMEM
);
1304 /* We create the SDP based on the RTSP AVFormatContext where we
1305 * aren't allowed to change the filename field. (We create the SDP
1306 * based on the RTSP context since the contexts for the RTP streams
1307 * don't exist yet.) In order to specify a custom URL with the actual
1308 * peer IP instead of the originally specified hostname, we create
1309 * a temporary copy of the AVFormatContext, where the custom URL is set.
1311 * FIXME: Create the SDP without copying the AVFormatContext.
1312 * This either requires setting up the RTP stream AVFormatContexts
1313 * already here (complicating things immensely) or getting a more
1314 * flexible SDP creation interface.
1317 ff_url_join(sdp_ctx
.filename
, sizeof(sdp_ctx
.filename
),
1318 "rtsp", NULL
, addr
, -1, NULL
);
1319 ctx_array
[0] = &sdp_ctx
;
1320 if (avf_sdp_create(ctx_array
, 1, sdp
, SDP_MAX_SIZE
)) {
1322 return AVERROR_INVALIDDATA
;
1324 av_log(s
, AV_LOG_INFO
, "SDP:\n%s\n", sdp
);
1325 ff_rtsp_send_cmd_with_content(s
, "ANNOUNCE", rt
->control_uri
,
1326 "Content-Type: application/sdp\r\n",
1327 reply
, NULL
, sdp
, strlen(sdp
));
1329 if (reply
->status_code
!= RTSP_STATUS_OK
)
1330 return AVERROR_INVALIDDATA
;
1332 /* Set up the RTSPStreams for each AVStream */
1333 for (i
= 0; i
< s
->nb_streams
; i
++) {
1334 RTSPStream
*rtsp_st
;
1335 AVStream
*st
= s
->streams
[i
];
1337 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
1339 return AVERROR(ENOMEM
);
1340 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
1342 st
->priv_data
= rtsp_st
;
1343 rtsp_st
->stream_index
= i
;
1345 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
, sizeof(rtsp_st
->control_url
));
1346 /* Note, this must match the relative uri set in the sdp content */
1347 av_strlcatf(rtsp_st
->control_url
, sizeof(rtsp_st
->control_url
),
1354 void ff_rtsp_close_connections(AVFormatContext
*s
)
1356 RTSPState
*rt
= s
->priv_data
;
1357 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
) url_close(rt
->rtsp_hd_out
);
1358 url_close(rt
->rtsp_hd
);
1359 rt
->rtsp_hd
= rt
->rtsp_hd_out
= NULL
;
1362 int ff_rtsp_connect(AVFormatContext
*s
)
1364 RTSPState
*rt
= s
->priv_data
;
1365 char host
[1024], path
[1024], tcpname
[1024], cmd
[2048], auth
[128];
1366 char *option_list
, *option
, *filename
;
1367 int port
, err
, tcp_fd
;
1368 RTSPMessageHeader reply1
= {0}, *reply
= &reply1
;
1369 int lower_transport_mask
= 0;
1370 char real_challenge
[64];
1371 struct sockaddr_storage peer
;
1372 socklen_t peer_len
= sizeof(peer
);
1374 if (!ff_network_init())
1375 return AVERROR(EIO
);
1377 rt
->control_transport
= RTSP_MODE_PLAIN
;
1378 /* extract hostname and port */
1379 av_url_split(NULL
, 0, auth
, sizeof(auth
),
1380 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->filename
);
1382 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1385 port
= RTSP_DEFAULT_PORT
;
1387 /* search for options */
1388 option_list
= strrchr(path
, '?');
1390 /* Strip out the RTSP specific options, write out the rest of
1391 * the options back into the same string. */
1392 filename
= option_list
;
1393 while (option_list
) {
1394 /* move the option pointer */
1395 option
= ++option_list
;
1396 option_list
= strchr(option_list
, '&');
1400 /* handle the options */
1401 if (!strcmp(option
, "udp")) {
1402 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP
);
1403 } else if (!strcmp(option
, "multicast")) {
1404 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST
);
1405 } else if (!strcmp(option
, "tcp")) {
1406 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1407 } else if(!strcmp(option
, "http")) {
1408 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1409 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1411 /* Write options back into the buffer, using memmove instead
1412 * of strcpy since the strings may overlap. */
1413 int len
= strlen(option
);
1414 memmove(++filename
, option
, len
);
1416 if (option_list
) *filename
= '&';
1422 if (!lower_transport_mask
)
1423 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1426 /* Only UDP or TCP - UDP multicast isn't supported. */
1427 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1428 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1429 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1430 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1431 "only UDP and TCP are supported for output.\n");
1432 err
= AVERROR(EINVAL
);
1437 /* Construct the URI used in request; this is similar to s->filename,
1438 * but with authentication credentials removed and RTSP specific options
1440 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), "rtsp", NULL
,
1441 host
, port
, "%s", path
);
1443 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1444 /* set up initial handshake for tunneling */
1445 char httpname
[1024];
1446 char sessioncookie
[17];
1449 ff_url_join(httpname
, sizeof(httpname
), "http", auth
, host
, port
, "%s", path
);
1450 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1451 av_get_random_seed(), av_get_random_seed());
1454 if (url_alloc(&rt
->rtsp_hd
, httpname
, URL_RDONLY
) < 0) {
1459 /* generate GET headers */
1460 snprintf(headers
, sizeof(headers
),
1461 "x-sessioncookie: %s\r\n"
1462 "Accept: application/x-rtsp-tunnelled\r\n"
1463 "Pragma: no-cache\r\n"
1464 "Cache-Control: no-cache\r\n",
1466 ff_http_set_headers(rt
->rtsp_hd
, headers
);
1468 /* complete the connection */
1469 if (url_connect(rt
->rtsp_hd
)) {
1475 if (url_alloc(&rt
->rtsp_hd_out
, httpname
, URL_WRONLY
) < 0 ) {
1480 /* generate POST headers */
1481 snprintf(headers
, sizeof(headers
),
1482 "x-sessioncookie: %s\r\n"
1483 "Content-Type: application/x-rtsp-tunnelled\r\n"
1484 "Pragma: no-cache\r\n"
1485 "Cache-Control: no-cache\r\n"
1486 "Content-Length: 32767\r\n"
1487 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1489 ff_http_set_headers(rt
->rtsp_hd_out
, headers
);
1490 ff_http_set_chunked_transfer_encoding(rt
->rtsp_hd_out
, 0);
1492 /* Initialize the authentication state for the POST session. The HTTP
1493 * protocol implementation doesn't properly handle multi-pass
1494 * authentication for POST requests, since it would require one of
1496 * - implementing Expect: 100-continue, which many HTTP servers
1497 * don't support anyway, even less the RTSP servers that do HTTP
1499 * - sending the whole POST data until getting a 401 reply specifying
1500 * what authentication method to use, then resending all that data
1501 * - waiting for potential 401 replies directly after sending the
1502 * POST header (waiting for some unspecified time)
1503 * Therefore, we copy the full auth state, which works for both basic
1504 * and digest. (For digest, we would have to synchronize the nonce
1505 * count variable between the two sessions, if we'd do more requests
1506 * with the original session, though.)
1508 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1510 /* complete the connection */
1511 if (url_connect(rt
->rtsp_hd_out
)) {
1516 /* open the tcp connection */
1517 ff_url_join(tcpname
, sizeof(tcpname
), "tcp", NULL
, host
, port
, NULL
);
1518 if (url_open(&rt
->rtsp_hd
, tcpname
, URL_RDWR
) < 0) {
1522 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1526 tcp_fd
= url_get_file_handle(rt
->rtsp_hd
);
1527 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1528 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1529 NULL
, 0, NI_NUMERICHOST
);
1532 /* request options supported by the server; this also detects server
1534 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1536 if (rt
->server_type
== RTSP_SERVER_REAL
)
1539 * The following entries are required for proper
1540 * streaming from a Realmedia server. They are
1541 * interdependent in some way although we currently
1542 * don't quite understand how. Values were copied
1543 * from mplayer SVN r23589.
1544 * @param CompanyID is a 16-byte ID in base64
1545 * @param ClientChallenge is a 16-byte ID in hex
1547 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1548 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1549 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1550 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1552 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1553 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1554 err
= AVERROR_INVALIDDATA
;
1558 /* detect server type if not standard-compliant RTP */
1559 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1560 rt
->server_type
= RTSP_SERVER_REAL
;
1562 } else if (!strncasecmp(reply
->server
, "WMServer/", 9)) {
1563 rt
->server_type
= RTSP_SERVER_WMS
;
1564 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1565 strcpy(real_challenge
, reply
->real_challenge
);
1570 err
= rtsp_setup_input_streams(s
, reply
);
1572 err
= rtsp_setup_output_streams(s
, host
);
1577 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1578 ~(lower_transport_mask
- 1)];
1580 err
= make_setup_request(s
, host
, port
, lower_transport
,
1581 rt
->server_type
== RTSP_SERVER_REAL ?
1582 real_challenge
: NULL
);
1585 lower_transport_mask
&= ~(1 << lower_transport
);
1586 if (lower_transport_mask
== 0 && err
== 1) {
1587 err
= FF_NETERROR(EPROTONOSUPPORT
);
1592 rt
->state
= RTSP_STATE_IDLE
;
1593 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1596 ff_rtsp_close_streams(s
);
1597 ff_rtsp_close_connections(s
);
1598 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1599 av_strlcpy(s
->filename
, reply
->location
, sizeof(s
->filename
));
1600 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1610 #if CONFIG_RTSP_DEMUXER
1611 static int rtsp_read_header(AVFormatContext
*s
,
1612 AVFormatParameters
*ap
)
1614 RTSPState
*rt
= s
->priv_data
;
1617 ret
= ff_rtsp_connect(s
);
1621 rt
->real_setup_cache
= av_mallocz(2 * s
->nb_streams
* sizeof(*rt
->real_setup_cache
));
1622 if (!rt
->real_setup_cache
)
1623 return AVERROR(ENOMEM
);
1624 rt
->real_setup
= rt
->real_setup_cache
+ s
->nb_streams
* sizeof(*rt
->real_setup
);
1626 if (ap
->initial_pause
) {
1627 /* do not start immediately */
1629 if (rtsp_read_play(s
) < 0) {
1630 ff_rtsp_close_streams(s
);
1631 ff_rtsp_close_connections(s
);
1632 return AVERROR_INVALIDDATA
;
1639 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1640 uint8_t *buf
, int buf_size
)
1642 RTSPState
*rt
= s
->priv_data
;
1643 RTSPStream
*rtsp_st
;
1645 int fd
, fd_max
, n
, i
, ret
, tcp_fd
, timeout_cnt
= 0;
1649 if (url_interrupt_cb())
1650 return AVERROR(EINTR
);
1653 tcp_fd
= fd_max
= url_get_file_handle(rt
->rtsp_hd
);
1654 FD_SET(tcp_fd
, &rfds
);
1659 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1660 rtsp_st
= rt
->rtsp_streams
[i
];
1661 if (rtsp_st
->rtp_handle
) {
1662 /* currently, we cannot probe RTCP handle because of
1663 * blocking restrictions */
1664 fd
= url_get_file_handle(rtsp_st
->rtp_handle
);
1671 tv
.tv_usec
= SELECT_TIMEOUT_MS
* 1000;
1672 n
= select(fd_max
+ 1, &rfds
, NULL
, NULL
, &tv
);
1675 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1676 rtsp_st
= rt
->rtsp_streams
[i
];
1677 if (rtsp_st
->rtp_handle
) {
1678 fd
= url_get_file_handle(rtsp_st
->rtp_handle
);
1679 if (FD_ISSET(fd
, &rfds
)) {
1680 ret
= url_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
1682 *prtsp_st
= rtsp_st
;
1688 #if CONFIG_RTSP_DEMUXER
1689 if (tcp_fd
!= -1 && FD_ISSET(tcp_fd
, &rfds
)) {
1690 RTSPMessageHeader reply
;
1692 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0);
1695 /* XXX: parse message */
1696 if (rt
->state
!= RTSP_STATE_STREAMING
)
1700 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
1701 return FF_NETERROR(ETIMEDOUT
);
1702 } else if (n
< 0 && errno
!= EINTR
)
1703 return AVERROR(errno
);
1707 static int tcp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1708 uint8_t *buf
, int buf_size
)
1710 RTSPState
*rt
= s
->priv_data
;
1711 int id
, len
, i
, ret
;
1712 RTSPStream
*rtsp_st
;
1714 #ifdef DEBUG_RTP_TCP
1715 dprintf(s
, "tcp_read_packet:\n");
1719 RTSPMessageHeader reply
;
1721 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 1);
1724 if (ret
== 1) /* received '$' */
1726 /* XXX: parse message */
1727 if (rt
->state
!= RTSP_STATE_STREAMING
)
1730 ret
= url_read_complete(rt
->rtsp_hd
, buf
, 3);
1734 len
= AV_RB16(buf
+ 1);
1735 #ifdef DEBUG_RTP_TCP
1736 dprintf(s
, "id=%d len=%d\n", id
, len
);
1738 if (len
> buf_size
|| len
< 12)
1741 ret
= url_read_complete(rt
->rtsp_hd
, buf
, len
);
1744 if (rt
->transport
== RTSP_TRANSPORT_RDT
&&
1745 ff_rdt_parse_header(buf
, len
, &id
, NULL
, NULL
, NULL
, NULL
) < 0)
1748 /* find the matching stream */
1749 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1750 rtsp_st
= rt
->rtsp_streams
[i
];
1751 if (id
>= rtsp_st
->interleaved_min
&&
1752 id
<= rtsp_st
->interleaved_max
)
1757 *prtsp_st
= rtsp_st
;
1761 static int rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1763 RTSPState
*rt
= s
->priv_data
;
1765 uint8_t buf
[10 * RTP_MAX_PACKET_LENGTH
];
1766 RTSPStream
*rtsp_st
;
1768 /* get next frames from the same RTP packet */
1769 if (rt
->cur_transport_priv
) {
1770 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1771 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1773 ret
= rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1775 rt
->cur_transport_priv
= NULL
;
1777 } else if (ret
== 1) {
1780 rt
->cur_transport_priv
= NULL
;
1783 /* read next RTP packet */
1785 switch(rt
->lower_transport
) {
1787 #if CONFIG_RTSP_DEMUXER
1788 case RTSP_LOWER_TRANSPORT_TCP
:
1789 len
= tcp_read_packet(s
, &rtsp_st
, buf
, sizeof(buf
));
1792 case RTSP_LOWER_TRANSPORT_UDP
:
1793 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
1794 len
= udp_read_packet(s
, &rtsp_st
, buf
, sizeof(buf
));
1795 if (len
>=0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1796 rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, len
);
1803 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1804 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, buf
, len
);
1806 ret
= rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, buf
, len
);
1808 /* Either bad packet, or a RTCP packet. Check if the
1809 * first_rtcp_ntp_time field was initialized. */
1810 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1811 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
1812 /* first_rtcp_ntp_time has been initialized for this stream,
1813 * copy the same value to all other uninitialized streams,
1814 * in order to map their timestamp origin to the same ntp time
1817 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1818 RTPDemuxContext
*rtpctx2
= rtsp_st
->transport_priv
;
1820 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
)
1821 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
1829 /* more packets may follow, so we save the RTP context */
1830 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
1835 static int rtsp_read_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1837 RTSPState
*rt
= s
->priv_data
;
1839 RTSPMessageHeader reply1
, *reply
= &reply1
;
1842 if (rt
->server_type
== RTSP_SERVER_REAL
) {
1845 for (i
= 0; i
< s
->nb_streams
; i
++)
1846 rt
->real_setup
[i
] = s
->streams
[i
]->discard
;
1848 if (!rt
->need_subscription
) {
1849 if (memcmp (rt
->real_setup
, rt
->real_setup_cache
,
1850 sizeof(enum AVDiscard
) * s
->nb_streams
)) {
1851 snprintf(cmd
, sizeof(cmd
),
1852 "Unsubscribe: %s\r\n",
1853 rt
->last_subscription
);
1854 ff_rtsp_send_cmd(s
, "SET_PARAMETER", rt
->control_uri
,
1856 if (reply
->status_code
!= RTSP_STATUS_OK
)
1857 return AVERROR_INVALIDDATA
;
1858 rt
->need_subscription
= 1;
1862 if (rt
->need_subscription
) {
1863 int r
, rule_nr
, first
= 1;
1865 memcpy(rt
->real_setup_cache
, rt
->real_setup
,
1866 sizeof(enum AVDiscard
) * s
->nb_streams
);
1867 rt
->last_subscription
[0] = 0;
1869 snprintf(cmd
, sizeof(cmd
),
1871 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1873 for (r
= 0; r
< s
->nb_streams
; r
++) {
1874 if (s
->streams
[r
]->priv_data
== rt
->rtsp_streams
[i
]) {
1875 if (s
->streams
[r
]->discard
!= AVDISCARD_ALL
) {
1877 av_strlcat(rt
->last_subscription
, ",",
1878 sizeof(rt
->last_subscription
));
1879 ff_rdt_subscribe_rule(
1880 rt
->last_subscription
,
1881 sizeof(rt
->last_subscription
), i
, rule_nr
);
1888 av_strlcatf(cmd
, sizeof(cmd
), "%s\r\n", rt
->last_subscription
);
1889 ff_rtsp_send_cmd(s
, "SET_PARAMETER", rt
->control_uri
,
1891 if (reply
->status_code
!= RTSP_STATUS_OK
)
1892 return AVERROR_INVALIDDATA
;
1893 rt
->need_subscription
= 0;
1895 if (rt
->state
== RTSP_STATE_STREAMING
)
1900 ret
= rtsp_fetch_packet(s
, pkt
);
1904 /* send dummy request to keep TCP connection alive */
1905 if ((av_gettime() - rt
->last_cmd_time
) / 1000000 >= rt
->timeout
/ 2) {
1906 if (rt
->server_type
== RTSP_SERVER_WMS
) {
1907 ff_rtsp_send_cmd_async(s
, "GET_PARAMETER", rt
->control_uri
, NULL
);
1909 ff_rtsp_send_cmd_async(s
, "OPTIONS", "*", NULL
);
1916 /* pause the stream */
1917 static int rtsp_read_pause(AVFormatContext
*s
)
1919 RTSPState
*rt
= s
->priv_data
;
1920 RTSPMessageHeader reply1
, *reply
= &reply1
;
1922 if (rt
->state
!= RTSP_STATE_STREAMING
)
1924 else if (!(rt
->server_type
== RTSP_SERVER_REAL
&& rt
->need_subscription
)) {
1925 ff_rtsp_send_cmd(s
, "PAUSE", rt
->control_uri
, NULL
, reply
, NULL
);
1926 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1930 rt
->state
= RTSP_STATE_PAUSED
;
1934 static int rtsp_read_seek(AVFormatContext
*s
, int stream_index
,
1935 int64_t timestamp
, int flags
)
1937 RTSPState
*rt
= s
->priv_data
;
1939 rt
->seek_timestamp
= av_rescale_q(timestamp
,
1940 s
->streams
[stream_index
]->time_base
,
1944 case RTSP_STATE_IDLE
:
1946 case RTSP_STATE_STREAMING
:
1947 if (rtsp_read_pause(s
) != 0)
1949 rt
->state
= RTSP_STATE_SEEKING
;
1950 if (rtsp_read_play(s
) != 0)
1953 case RTSP_STATE_PAUSED
:
1954 rt
->state
= RTSP_STATE_IDLE
;
1960 static int rtsp_read_close(AVFormatContext
*s
)
1962 RTSPState
*rt
= s
->priv_data
;
1965 /* NOTE: it is valid to flush the buffer here */
1966 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1967 url_fclose(&rt
->rtsp_gb
);
1970 ff_rtsp_send_cmd_async(s
, "TEARDOWN", rt
->control_uri
, NULL
);
1972 ff_rtsp_close_streams(s
);
1973 ff_rtsp_close_connections(s
);
1975 rt
->real_setup
= NULL
;
1976 av_freep(&rt
->real_setup_cache
);
1980 AVInputFormat rtsp_demuxer
= {
1982 NULL_IF_CONFIG_SMALL("RTSP input format"),
1989 .flags
= AVFMT_NOFILE
,
1990 .read_play
= rtsp_read_play
,
1991 .read_pause
= rtsp_read_pause
,
1995 static int sdp_probe(AVProbeData
*p1
)
1997 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
1999 /* we look for a line beginning "c=IN IP4" */
2000 while (p
< p_end
&& *p
!= '\0') {
2001 if (p
+ sizeof("c=IN IP4") - 1 < p_end
&&
2002 av_strstart(p
, "c=IN IP4", NULL
))
2003 return AVPROBE_SCORE_MAX
/ 2;
2005 while (p
< p_end
- 1 && *p
!= '\n') p
++;
2014 static int sdp_read_header(AVFormatContext
*s
, AVFormatParameters
*ap
)
2016 RTSPState
*rt
= s
->priv_data
;
2017 RTSPStream
*rtsp_st
;
2022 if (!ff_network_init())
2023 return AVERROR(EIO
);
2025 /* read the whole sdp file */
2026 /* XXX: better loading */
2027 content
= av_malloc(SDP_MAX_SIZE
);
2028 size
= get_buffer(s
->pb
, content
, SDP_MAX_SIZE
- 1);
2031 return AVERROR_INVALIDDATA
;
2033 content
[size
] ='\0';
2035 sdp_parse(s
, content
);
2038 /* open each RTP stream */
2039 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
2040 rtsp_st
= rt
->rtsp_streams
[i
];
2042 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
2043 inet_ntoa(rtsp_st
->sdp_ip
), rtsp_st
->sdp_port
,
2044 "?localport=%d&ttl=%d", rtsp_st
->sdp_port
,
2046 if (url_open(&rtsp_st
->rtp_handle
, url
, URL_RDWR
) < 0) {
2047 err
= AVERROR_INVALIDDATA
;
2050 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
2055 ff_rtsp_close_streams(s
);
2060 static int sdp_read_close(AVFormatContext
*s
)
2062 ff_rtsp_close_streams(s
);
2067 AVInputFormat sdp_demuxer
= {
2069 NULL_IF_CONFIG_SMALL("SDP"),