3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/random_seed.h"
30 #include <sys/select.h>
35 #include "os_support.h"
41 #include "rtpdec_formats.h"
42 #include "rtpenc_chain.h"
45 //#define DEBUG_RTP_TCP
47 /* Timeout values for socket select, in ms,
48 * and read_packet(), in seconds */
49 #define SELECT_TIMEOUT_MS 100
50 #define READ_PACKET_TIMEOUT_S 10
51 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
52 #define SDP_MAX_SIZE 16384
53 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
55 static void get_word_until_chars(char *buf
, int buf_size
,
56 const char *sep
, const char **pp
)
62 p
+= strspn(p
, SPACE_CHARS
);
64 while (!strchr(sep
, *p
) && *p
!= '\0') {
65 if ((q
- buf
) < buf_size
- 1)
74 static void get_word_sep(char *buf
, int buf_size
, const char *sep
,
77 if (**pp
== '/') (*pp
)++;
78 get_word_until_chars(buf
, buf_size
, sep
, pp
);
81 static void get_word(char *buf
, int buf_size
, const char **pp
)
83 get_word_until_chars(buf
, buf_size
, SPACE_CHARS
, pp
);
86 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
88 * Used for seeking in the rtp stream.
90 static void rtsp_parse_range_npt(const char *p
, int64_t *start
, int64_t *end
)
94 p
+= strspn(p
, SPACE_CHARS
);
95 if (!av_stristart(p
, "npt=", &p
))
98 *start
= AV_NOPTS_VALUE
;
99 *end
= AV_NOPTS_VALUE
;
101 get_word_sep(buf
, sizeof(buf
), "-", &p
);
102 *start
= parse_date(buf
, 1);
105 get_word_sep(buf
, sizeof(buf
), "-", &p
);
106 *end
= parse_date(buf
, 1);
108 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
109 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
112 static int get_sockaddr(const char *buf
, struct sockaddr_storage
*sock
)
114 struct addrinfo hints
, *ai
= NULL
;
115 memset(&hints
, 0, sizeof(hints
));
116 hints
.ai_flags
= AI_NUMERICHOST
;
117 if (getaddrinfo(buf
, NULL
, &hints
, &ai
))
119 memcpy(sock
, ai
->ai_addr
, FFMIN(sizeof(*sock
), ai
->ai_addrlen
));
125 static void init_rtp_handler(RTPDynamicProtocolHandler
*handler
,
126 RTSPStream
*rtsp_st
, AVCodecContext
*codec
)
130 codec
->codec_id
= handler
->codec_id
;
131 rtsp_st
->dynamic_handler
= handler
;
133 rtsp_st
->dynamic_protocol_context
= handler
->open();
136 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
137 static int sdp_parse_rtpmap(AVFormatContext
*s
,
138 AVStream
*st
, RTSPStream
*rtsp_st
,
139 int payload_type
, const char *p
)
141 AVCodecContext
*codec
= st
->codec
;
147 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
148 * see if we can handle this kind of payload.
149 * The space should normally not be there but some Real streams or
150 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
151 * have a trailing space. */
152 get_word_sep(buf
, sizeof(buf
), "/ ", &p
);
153 if (payload_type
>= RTP_PT_PRIVATE
) {
154 RTPDynamicProtocolHandler
*handler
=
155 ff_rtp_handler_find_by_name(buf
, codec
->codec_type
);
156 init_rtp_handler(handler
, rtsp_st
, codec
);
157 /* If no dynamic handler was found, check with the list of standard
158 * allocated types, if such a stream for some reason happens to
159 * use a private payload type. This isn't handled in rtpdec.c, since
160 * the format name from the rtpmap line never is passed into rtpdec. */
161 if (!rtsp_st
->dynamic_handler
)
162 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
164 /* We are in a standard case
165 * (from http://www.iana.org/assignments/rtp-parameters). */
166 /* search into AVRtpPayloadTypes[] */
167 codec
->codec_id
= ff_rtp_codec_id(buf
, codec
->codec_type
);
170 c
= avcodec_find_decoder(codec
->codec_id
);
176 get_word_sep(buf
, sizeof(buf
), "/", &p
);
178 switch (codec
->codec_type
) {
179 case AVMEDIA_TYPE_AUDIO
:
180 av_log(s
, AV_LOG_DEBUG
, "audio codec set to: %s\n", c_name
);
181 codec
->sample_rate
= RTSP_DEFAULT_AUDIO_SAMPLERATE
;
182 codec
->channels
= RTSP_DEFAULT_NB_AUDIO_CHANNELS
;
184 codec
->sample_rate
= i
;
185 av_set_pts_info(st
, 32, 1, codec
->sample_rate
);
186 get_word_sep(buf
, sizeof(buf
), "/", &p
);
190 // TODO: there is a bug here; if it is a mono stream, and
191 // less than 22000Hz, faad upconverts to stereo and twice
192 // the frequency. No problem, but the sample rate is being
193 // set here by the sdp line. Patch on its way. (rdm)
195 av_log(s
, AV_LOG_DEBUG
, "audio samplerate set to: %i\n",
197 av_log(s
, AV_LOG_DEBUG
, "audio channels set to: %i\n",
200 case AVMEDIA_TYPE_VIDEO
:
201 av_log(s
, AV_LOG_DEBUG
, "video codec set to: %s\n", c_name
);
203 av_set_pts_info(st
, 32, 1, i
);
211 /* parse the attribute line from the fmtp a line of an sdp response. This
212 * is broken out as a function because it is used in rtp_h264.c, which is
214 int ff_rtsp_next_attr_and_value(const char **p
, char *attr
, int attr_size
,
215 char *value
, int value_size
)
217 *p
+= strspn(*p
, SPACE_CHARS
);
219 get_word_sep(attr
, attr_size
, "=", p
);
222 get_word_sep(value
, value_size
, ";", p
);
230 typedef struct SDPParseState
{
232 struct sockaddr_storage default_ip
;
234 int skip_media
; ///< set if an unknown m= line occurs
237 static void sdp_parse_line(AVFormatContext
*s
, SDPParseState
*s1
,
238 int letter
, const char *buf
)
240 RTSPState
*rt
= s
->priv_data
;
241 char buf1
[64], st_type
[64];
243 enum AVMediaType codec_type
;
247 struct sockaddr_storage sdp_ip
;
250 dprintf(s
, "sdp: %c='%s'\n", letter
, buf
);
253 if (s1
->skip_media
&& letter
!= 'm')
257 get_word(buf1
, sizeof(buf1
), &p
);
258 if (strcmp(buf1
, "IN") != 0)
260 get_word(buf1
, sizeof(buf1
), &p
);
261 if (strcmp(buf1
, "IP4") && strcmp(buf1
, "IP6"))
263 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
264 if (get_sockaddr(buf1
, &sdp_ip
))
269 get_word_sep(buf1
, sizeof(buf1
), "/", &p
);
272 if (s
->nb_streams
== 0) {
273 s1
->default_ip
= sdp_ip
;
274 s1
->default_ttl
= ttl
;
276 st
= s
->streams
[s
->nb_streams
- 1];
277 rtsp_st
= st
->priv_data
;
278 rtsp_st
->sdp_ip
= sdp_ip
;
279 rtsp_st
->sdp_ttl
= ttl
;
283 av_metadata_set2(&s
->metadata
, "title", p
, 0);
286 if (s
->nb_streams
== 0) {
287 av_metadata_set2(&s
->metadata
, "comment", p
, 0);
294 get_word(st_type
, sizeof(st_type
), &p
);
295 if (!strcmp(st_type
, "audio")) {
296 codec_type
= AVMEDIA_TYPE_AUDIO
;
297 } else if (!strcmp(st_type
, "video")) {
298 codec_type
= AVMEDIA_TYPE_VIDEO
;
299 } else if (!strcmp(st_type
, "application")) {
300 codec_type
= AVMEDIA_TYPE_DATA
;
305 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
308 rtsp_st
->stream_index
= -1;
309 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
311 rtsp_st
->sdp_ip
= s1
->default_ip
;
312 rtsp_st
->sdp_ttl
= s1
->default_ttl
;
314 get_word(buf1
, sizeof(buf1
), &p
); /* port */
315 rtsp_st
->sdp_port
= atoi(buf1
);
317 get_word(buf1
, sizeof(buf1
), &p
); /* protocol (ignored) */
319 /* XXX: handle list of formats */
320 get_word(buf1
, sizeof(buf1
), &p
); /* format list */
321 rtsp_st
->sdp_payload_type
= atoi(buf1
);
323 if (!strcmp(ff_rtp_enc_name(rtsp_st
->sdp_payload_type
), "MP2T")) {
324 /* no corresponding stream */
326 st
= av_new_stream(s
, 0);
329 st
->priv_data
= rtsp_st
;
330 rtsp_st
->stream_index
= st
->index
;
331 st
->codec
->codec_type
= codec_type
;
332 if (rtsp_st
->sdp_payload_type
< RTP_PT_PRIVATE
) {
333 RTPDynamicProtocolHandler
*handler
;
334 /* if standard payload type, we can find the codec right now */
335 ff_rtp_get_codec_info(st
->codec
, rtsp_st
->sdp_payload_type
);
336 if (st
->codec
->codec_type
== AVMEDIA_TYPE_AUDIO
&&
337 st
->codec
->sample_rate
> 0)
338 av_set_pts_info(st
, 32, 1, st
->codec
->sample_rate
);
339 /* Even static payload types may need a custom depacketizer */
340 handler
= ff_rtp_handler_find_by_id(
341 rtsp_st
->sdp_payload_type
, st
->codec
->codec_type
);
342 init_rtp_handler(handler
, rtsp_st
, st
->codec
);
345 /* put a default control url */
346 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
,
347 sizeof(rtsp_st
->control_url
));
350 if (av_strstart(p
, "control:", &p
)) {
351 if (s
->nb_streams
== 0) {
352 if (!strncmp(p
, "rtsp://", 7))
353 av_strlcpy(rt
->control_uri
, p
,
354 sizeof(rt
->control_uri
));
357 /* get the control url */
358 st
= s
->streams
[s
->nb_streams
- 1];
359 rtsp_st
= st
->priv_data
;
361 /* XXX: may need to add full url resolution */
362 av_url_split(proto
, sizeof(proto
), NULL
, 0, NULL
, 0,
364 if (proto
[0] == '\0') {
365 /* relative control URL */
366 if (rtsp_st
->control_url
[strlen(rtsp_st
->control_url
)-1]!='/')
367 av_strlcat(rtsp_st
->control_url
, "/",
368 sizeof(rtsp_st
->control_url
));
369 av_strlcat(rtsp_st
->control_url
, p
,
370 sizeof(rtsp_st
->control_url
));
372 av_strlcpy(rtsp_st
->control_url
, p
,
373 sizeof(rtsp_st
->control_url
));
375 } else if (av_strstart(p
, "rtpmap:", &p
) && s
->nb_streams
> 0) {
376 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
377 get_word(buf1
, sizeof(buf1
), &p
);
378 payload_type
= atoi(buf1
);
379 st
= s
->streams
[s
->nb_streams
- 1];
380 rtsp_st
= st
->priv_data
;
381 sdp_parse_rtpmap(s
, st
, rtsp_st
, payload_type
, p
);
382 } else if (av_strstart(p
, "fmtp:", &p
) ||
383 av_strstart(p
, "framesize:", &p
)) {
384 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
385 // let dynamic protocol handlers have a stab at the line.
386 get_word(buf1
, sizeof(buf1
), &p
);
387 payload_type
= atoi(buf1
);
388 for (i
= 0; i
< s
->nb_streams
; i
++) {
390 rtsp_st
= st
->priv_data
;
391 if (rtsp_st
->sdp_payload_type
== payload_type
&&
392 rtsp_st
->dynamic_handler
&&
393 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
394 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
, i
,
395 rtsp_st
->dynamic_protocol_context
, buf
);
397 } else if (av_strstart(p
, "range:", &p
)) {
400 // this is so that seeking on a streamed file can work.
401 rtsp_parse_range_npt(p
, &start
, &end
);
402 s
->start_time
= start
;
403 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
404 s
->duration
= (end
== AV_NOPTS_VALUE
) ?
405 AV_NOPTS_VALUE
: end
- start
;
406 } else if (av_strstart(p
, "IsRealDataType:integer;",&p
)) {
408 rt
->transport
= RTSP_TRANSPORT_RDT
;
409 } else if (av_strstart(p
, "SampleRate:integer;", &p
) &&
411 st
= s
->streams
[s
->nb_streams
- 1];
412 st
->codec
->sample_rate
= atoi(p
);
414 if (rt
->server_type
== RTSP_SERVER_WMS
)
415 ff_wms_parse_sdp_a_line(s
, p
);
416 if (s
->nb_streams
> 0) {
417 if (rt
->server_type
== RTSP_SERVER_REAL
)
418 ff_real_parse_sdp_a_line(s
, s
->nb_streams
- 1, p
);
420 rtsp_st
= s
->streams
[s
->nb_streams
- 1]->priv_data
;
421 if (rtsp_st
->dynamic_handler
&&
422 rtsp_st
->dynamic_handler
->parse_sdp_a_line
)
423 rtsp_st
->dynamic_handler
->parse_sdp_a_line(s
,
425 rtsp_st
->dynamic_protocol_context
, buf
);
432 int ff_sdp_parse(AVFormatContext
*s
, const char *content
)
436 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
437 * contain long SDP lines containing complete ASF Headers (several
438 * kB) or arrays of MDPR (RM stream descriptor) headers plus
439 * "rulebooks" describing their properties. Therefore, the SDP line
442 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
443 * in rtpdec_xiph.c. */
445 SDPParseState sdp_parse_state
, *s1
= &sdp_parse_state
;
447 memset(s1
, 0, sizeof(SDPParseState
));
450 p
+= strspn(p
, SPACE_CHARS
);
458 /* get the content */
460 while (*p
!= '\n' && *p
!= '\r' && *p
!= '\0') {
461 if ((q
- buf
) < sizeof(buf
) - 1)
466 sdp_parse_line(s
, s1
, letter
, buf
);
468 while (*p
!= '\n' && *p
!= '\0')
475 #endif /* CONFIG_RTPDEC */
477 void ff_rtsp_undo_setup(AVFormatContext
*s
)
479 RTSPState
*rt
= s
->priv_data
;
482 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
483 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
486 if (rtsp_st
->transport_priv
) {
488 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
489 av_write_trailer(rtpctx
);
490 if (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
492 url_close_dyn_buf(rtpctx
->pb
, &ptr
);
495 url_fclose(rtpctx
->pb
);
497 av_metadata_free(&rtpctx
->streams
[0]->metadata
);
498 av_metadata_free(&rtpctx
->metadata
);
499 av_free(rtpctx
->streams
[0]);
501 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
502 ff_rdt_parse_close(rtsp_st
->transport_priv
);
503 else if (CONFIG_RTPDEC
)
504 rtp_parse_close(rtsp_st
->transport_priv
);
506 rtsp_st
->transport_priv
= NULL
;
507 if (rtsp_st
->rtp_handle
)
508 url_close(rtsp_st
->rtp_handle
);
509 rtsp_st
->rtp_handle
= NULL
;
513 /* close and free RTSP streams */
514 void ff_rtsp_close_streams(AVFormatContext
*s
)
516 RTSPState
*rt
= s
->priv_data
;
520 ff_rtsp_undo_setup(s
);
521 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
522 rtsp_st
= rt
->rtsp_streams
[i
];
524 if (rtsp_st
->dynamic_handler
&& rtsp_st
->dynamic_protocol_context
)
525 rtsp_st
->dynamic_handler
->close(
526 rtsp_st
->dynamic_protocol_context
);
529 av_free(rt
->rtsp_streams
);
531 av_close_input_stream (rt
->asf_ctx
);
534 av_free(rt
->recvbuf
);
537 static int rtsp_open_transport_ctx(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
539 RTSPState
*rt
= s
->priv_data
;
542 /* open the RTP context */
543 if (rtsp_st
->stream_index
>= 0)
544 st
= s
->streams
[rtsp_st
->stream_index
];
546 s
->ctx_flags
|= AVFMTCTX_NOHEADER
;
548 if (s
->oformat
&& CONFIG_RTSP_MUXER
) {
549 rtsp_st
->transport_priv
= ff_rtp_chain_mux_open(s
, st
,
551 RTSP_TCP_MAX_PACKET_SIZE
);
552 /* Ownership of rtp_handle is passed to the rtp mux context */
553 rtsp_st
->rtp_handle
= NULL
;
554 } else if (rt
->transport
== RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
)
555 rtsp_st
->transport_priv
= ff_rdt_parse_open(s
, st
->index
,
556 rtsp_st
->dynamic_protocol_context
,
557 rtsp_st
->dynamic_handler
);
558 else if (CONFIG_RTPDEC
)
559 rtsp_st
->transport_priv
= rtp_parse_open(s
, st
, rtsp_st
->rtp_handle
,
560 rtsp_st
->sdp_payload_type
,
561 (rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
|| !s
->max_delay
)
562 ?
0 : RTP_REORDER_QUEUE_DEFAULT_SIZE
);
564 if (!rtsp_st
->transport_priv
) {
565 return AVERROR(ENOMEM
);
566 } else if (rt
->transport
!= RTSP_TRANSPORT_RDT
&& CONFIG_RTPDEC
) {
567 if (rtsp_st
->dynamic_handler
) {
568 rtp_parse_set_dynamic_protocol(rtsp_st
->transport_priv
,
569 rtsp_st
->dynamic_protocol_context
,
570 rtsp_st
->dynamic_handler
);
577 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
578 static void rtsp_parse_range(int *min_ptr
, int *max_ptr
, const char **pp
)
584 p
+= strspn(p
, SPACE_CHARS
);
585 v
= strtol(p
, (char **)&p
, 10);
589 v
= strtol(p
, (char **)&p
, 10);
598 /* XXX: only one transport specification is parsed */
599 static void rtsp_parse_transport(RTSPMessageHeader
*reply
, const char *p
)
601 char transport_protocol
[16];
603 char lower_transport
[16];
605 RTSPTransportField
*th
;
608 reply
->nb_transports
= 0;
611 p
+= strspn(p
, SPACE_CHARS
);
615 th
= &reply
->transports
[reply
->nb_transports
];
617 get_word_sep(transport_protocol
, sizeof(transport_protocol
),
619 if (!strcasecmp (transport_protocol
, "rtp")) {
620 get_word_sep(profile
, sizeof(profile
), "/;,", &p
);
621 lower_transport
[0] = '\0';
622 /* rtp/avp/<protocol> */
624 get_word_sep(lower_transport
, sizeof(lower_transport
),
627 th
->transport
= RTSP_TRANSPORT_RTP
;
628 } else if (!strcasecmp (transport_protocol
, "x-pn-tng") ||
629 !strcasecmp (transport_protocol
, "x-real-rdt")) {
630 /* x-pn-tng/<protocol> */
631 get_word_sep(lower_transport
, sizeof(lower_transport
), "/;,", &p
);
633 th
->transport
= RTSP_TRANSPORT_RDT
;
635 if (!strcasecmp(lower_transport
, "TCP"))
636 th
->lower_transport
= RTSP_LOWER_TRANSPORT_TCP
;
638 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP
;
642 /* get each parameter */
643 while (*p
!= '\0' && *p
!= ',') {
644 get_word_sep(parameter
, sizeof(parameter
), "=;,", &p
);
645 if (!strcmp(parameter
, "port")) {
648 rtsp_parse_range(&th
->port_min
, &th
->port_max
, &p
);
650 } else if (!strcmp(parameter
, "client_port")) {
653 rtsp_parse_range(&th
->client_port_min
,
654 &th
->client_port_max
, &p
);
656 } else if (!strcmp(parameter
, "server_port")) {
659 rtsp_parse_range(&th
->server_port_min
,
660 &th
->server_port_max
, &p
);
662 } else if (!strcmp(parameter
, "interleaved")) {
665 rtsp_parse_range(&th
->interleaved_min
,
666 &th
->interleaved_max
, &p
);
668 } else if (!strcmp(parameter
, "multicast")) {
669 if (th
->lower_transport
== RTSP_LOWER_TRANSPORT_UDP
)
670 th
->lower_transport
= RTSP_LOWER_TRANSPORT_UDP_MULTICAST
;
671 } else if (!strcmp(parameter
, "ttl")) {
674 th
->ttl
= strtol(p
, (char **)&p
, 10);
676 } else if (!strcmp(parameter
, "destination")) {
679 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
680 get_sockaddr(buf
, &th
->destination
);
682 } else if (!strcmp(parameter
, "source")) {
685 get_word_sep(buf
, sizeof(buf
), ";,", &p
);
686 av_strlcpy(th
->source
, buf
, sizeof(th
->source
));
690 while (*p
!= ';' && *p
!= '\0' && *p
!= ',')
698 reply
->nb_transports
++;
702 static void handle_rtp_info(RTSPState
*rt
, const char *url
,
703 uint32_t seq
, uint32_t rtptime
)
706 if (!rtptime
|| !url
[0])
708 if (rt
->transport
!= RTSP_TRANSPORT_RTP
)
710 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
711 RTSPStream
*rtsp_st
= rt
->rtsp_streams
[i
];
712 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
715 if (!strcmp(rtsp_st
->control_url
, url
)) {
716 rtpctx
->base_timestamp
= rtptime
;
722 static void rtsp_parse_rtp_info(RTSPState
*rt
, const char *p
)
725 char key
[20], value
[1024], url
[1024] = "";
726 uint32_t seq
= 0, rtptime
= 0;
729 p
+= strspn(p
, SPACE_CHARS
);
732 get_word_sep(key
, sizeof(key
), "=", &p
);
736 get_word_sep(value
, sizeof(value
), ";, ", &p
);
738 if (!strcmp(key
, "url"))
739 av_strlcpy(url
, value
, sizeof(url
));
740 else if (!strcmp(key
, "seq"))
741 seq
= strtol(value
, NULL
, 10);
742 else if (!strcmp(key
, "rtptime"))
743 rtptime
= strtol(value
, NULL
, 10);
745 handle_rtp_info(rt
, url
, seq
, rtptime
);
754 handle_rtp_info(rt
, url
, seq
, rtptime
);
757 void ff_rtsp_parse_line(RTSPMessageHeader
*reply
, const char *buf
,
758 RTSPState
*rt
, const char *method
)
762 /* NOTE: we do case independent match for broken servers */
764 if (av_stristart(p
, "Session:", &p
)) {
766 get_word_sep(reply
->session_id
, sizeof(reply
->session_id
), ";", &p
);
767 if (av_stristart(p
, ";timeout=", &p
) &&
768 (t
= strtol(p
, NULL
, 10)) > 0) {
771 } else if (av_stristart(p
, "Content-Length:", &p
)) {
772 reply
->content_length
= strtol(p
, NULL
, 10);
773 } else if (av_stristart(p
, "Transport:", &p
)) {
774 rtsp_parse_transport(reply
, p
);
775 } else if (av_stristart(p
, "CSeq:", &p
)) {
776 reply
->seq
= strtol(p
, NULL
, 10);
777 } else if (av_stristart(p
, "Range:", &p
)) {
778 rtsp_parse_range_npt(p
, &reply
->range_start
, &reply
->range_end
);
779 } else if (av_stristart(p
, "RealChallenge1:", &p
)) {
780 p
+= strspn(p
, SPACE_CHARS
);
781 av_strlcpy(reply
->real_challenge
, p
, sizeof(reply
->real_challenge
));
782 } else if (av_stristart(p
, "Server:", &p
)) {
783 p
+= strspn(p
, SPACE_CHARS
);
784 av_strlcpy(reply
->server
, p
, sizeof(reply
->server
));
785 } else if (av_stristart(p
, "Notice:", &p
) ||
786 av_stristart(p
, "X-Notice:", &p
)) {
787 reply
->notice
= strtol(p
, NULL
, 10);
788 } else if (av_stristart(p
, "Location:", &p
)) {
789 p
+= strspn(p
, SPACE_CHARS
);
790 av_strlcpy(reply
->location
, p
, sizeof(reply
->location
));
791 } else if (av_stristart(p
, "WWW-Authenticate:", &p
) && rt
) {
792 p
+= strspn(p
, SPACE_CHARS
);
793 ff_http_auth_handle_header(&rt
->auth_state
, "WWW-Authenticate", p
);
794 } else if (av_stristart(p
, "Authentication-Info:", &p
) && rt
) {
795 p
+= strspn(p
, SPACE_CHARS
);
796 ff_http_auth_handle_header(&rt
->auth_state
, "Authentication-Info", p
);
797 } else if (av_stristart(p
, "Content-Base:", &p
) && rt
) {
798 p
+= strspn(p
, SPACE_CHARS
);
799 if (method
&& !strcmp(method
, "DESCRIBE"))
800 av_strlcpy(rt
->control_uri
, p
, sizeof(rt
->control_uri
));
801 } else if (av_stristart(p
, "RTP-Info:", &p
) && rt
) {
802 p
+= strspn(p
, SPACE_CHARS
);
803 if (method
&& !strcmp(method
, "PLAY"))
804 rtsp_parse_rtp_info(rt
, p
);
808 /* skip a RTP/TCP interleaved packet */
809 void ff_rtsp_skip_packet(AVFormatContext
*s
)
811 RTSPState
*rt
= s
->priv_data
;
815 ret
= url_read_complete(rt
->rtsp_hd
, buf
, 3);
818 len
= AV_RB16(buf
+ 1);
820 dprintf(s
, "skipping RTP packet len=%d\n", len
);
825 if (len1
> sizeof(buf
))
827 ret
= url_read_complete(rt
->rtsp_hd
, buf
, len1
);
834 int ff_rtsp_read_reply(AVFormatContext
*s
, RTSPMessageHeader
*reply
,
835 unsigned char **content_ptr
,
836 int return_on_interleaved_data
, const char *method
)
838 RTSPState
*rt
= s
->priv_data
;
839 char buf
[4096], buf1
[1024], *q
;
842 int ret
, content_length
, line_count
= 0;
843 unsigned char *content
= NULL
;
845 memset(reply
, 0, sizeof(*reply
));
847 /* parse reply (XXX: use buffers) */
848 rt
->last_reply
[0] = '\0';
852 ret
= url_read_complete(rt
->rtsp_hd
, &ch
, 1);
854 dprintf(s
, "ret=%d c=%02x [%c]\n", ret
, ch
, ch
);
861 /* XXX: only parse it if first char on line ? */
862 if (return_on_interleaved_data
) {
865 ff_rtsp_skip_packet(s
);
866 } else if (ch
!= '\r') {
867 if ((q
- buf
) < sizeof(buf
) - 1)
873 dprintf(s
, "line='%s'\n", buf
);
875 /* test if last line */
879 if (line_count
== 0) {
881 get_word(buf1
, sizeof(buf1
), &p
);
882 get_word(buf1
, sizeof(buf1
), &p
);
883 reply
->status_code
= atoi(buf1
);
884 av_strlcpy(reply
->reason
, p
, sizeof(reply
->reason
));
886 ff_rtsp_parse_line(reply
, p
, rt
, method
);
887 av_strlcat(rt
->last_reply
, p
, sizeof(rt
->last_reply
));
888 av_strlcat(rt
->last_reply
, "\n", sizeof(rt
->last_reply
));
893 if (rt
->session_id
[0] == '\0' && reply
->session_id
[0] != '\0')
894 av_strlcpy(rt
->session_id
, reply
->session_id
, sizeof(rt
->session_id
));
896 content_length
= reply
->content_length
;
897 if (content_length
> 0) {
898 /* leave some room for a trailing '\0' (useful for simple parsing) */
899 content
= av_malloc(content_length
+ 1);
900 (void)url_read_complete(rt
->rtsp_hd
, content
, content_length
);
901 content
[content_length
] = '\0';
904 *content_ptr
= content
;
908 if (rt
->seq
!= reply
->seq
) {
909 av_log(s
, AV_LOG_WARNING
, "CSeq %d expected, %d received.\n",
910 rt
->seq
, reply
->seq
);
914 if (reply
->notice
== 2101 /* End-of-Stream Reached */ ||
915 reply
->notice
== 2104 /* Start-of-Stream Reached */ ||
916 reply
->notice
== 2306 /* Continuous Feed Terminated */) {
917 rt
->state
= RTSP_STATE_IDLE
;
918 } else if (reply
->notice
>= 4400 && reply
->notice
< 5500) {
919 return AVERROR(EIO
); /* data or server error */
920 } else if (reply
->notice
== 2401 /* Ticket Expired */ ||
921 (reply
->notice
>= 5500 && reply
->notice
< 5600) /* end of term */ )
922 return AVERROR(EPERM
);
927 int ff_rtsp_send_cmd_with_content_async(AVFormatContext
*s
,
928 const char *method
, const char *url
,
930 const unsigned char *send_content
,
931 int send_content_length
)
933 RTSPState
*rt
= s
->priv_data
;
934 char buf
[4096], *out_buf
;
935 char base64buf
[AV_BASE64_SIZE(sizeof(buf
))];
937 /* Add in RTSP headers */
940 snprintf(buf
, sizeof(buf
), "%s %s RTSP/1.0\r\n", method
, url
);
942 av_strlcat(buf
, headers
, sizeof(buf
));
943 av_strlcatf(buf
, sizeof(buf
), "CSeq: %d\r\n", rt
->seq
);
944 if (rt
->session_id
[0] != '\0' && (!headers
||
945 !strstr(headers
, "\nIf-Match:"))) {
946 av_strlcatf(buf
, sizeof(buf
), "Session: %s\r\n", rt
->session_id
);
949 char *str
= ff_http_auth_create_response(&rt
->auth_state
,
950 rt
->auth
, url
, method
);
952 av_strlcat(buf
, str
, sizeof(buf
));
955 if (send_content_length
> 0 && send_content
)
956 av_strlcatf(buf
, sizeof(buf
), "Content-Length: %d\r\n", send_content_length
);
957 av_strlcat(buf
, "\r\n", sizeof(buf
));
959 /* base64 encode rtsp if tunneling */
960 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
961 av_base64_encode(base64buf
, sizeof(base64buf
), buf
, strlen(buf
));
965 dprintf(s
, "Sending:\n%s--\n", buf
);
967 url_write(rt
->rtsp_hd_out
, out_buf
, strlen(out_buf
));
968 if (send_content_length
> 0 && send_content
) {
969 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
970 av_log(s
, AV_LOG_ERROR
, "tunneling of RTSP requests "
971 "with content data not supported\n");
972 return AVERROR_PATCHWELCOME
;
974 url_write(rt
->rtsp_hd_out
, send_content
, send_content_length
);
976 rt
->last_cmd_time
= av_gettime();
981 int ff_rtsp_send_cmd_async(AVFormatContext
*s
, const char *method
,
982 const char *url
, const char *headers
)
984 return ff_rtsp_send_cmd_with_content_async(s
, method
, url
, headers
, NULL
, 0);
987 int ff_rtsp_send_cmd(AVFormatContext
*s
, const char *method
, const char *url
,
988 const char *headers
, RTSPMessageHeader
*reply
,
989 unsigned char **content_ptr
)
991 return ff_rtsp_send_cmd_with_content(s
, method
, url
, headers
, reply
,
992 content_ptr
, NULL
, 0);
995 int ff_rtsp_send_cmd_with_content(AVFormatContext
*s
,
996 const char *method
, const char *url
,
998 RTSPMessageHeader
*reply
,
999 unsigned char **content_ptr
,
1000 const unsigned char *send_content
,
1001 int send_content_length
)
1003 RTSPState
*rt
= s
->priv_data
;
1004 HTTPAuthType cur_auth_type
;
1008 cur_auth_type
= rt
->auth_state
.auth_type
;
1009 if ((ret
= ff_rtsp_send_cmd_with_content_async(s
, method
, url
, header
,
1011 send_content_length
)))
1014 if ((ret
= ff_rtsp_read_reply(s
, reply
, content_ptr
, 0, method
) ) < 0)
1017 if (reply
->status_code
== 401 && cur_auth_type
== HTTP_AUTH_NONE
&&
1018 rt
->auth_state
.auth_type
!= HTTP_AUTH_NONE
)
1021 if (reply
->status_code
> 400){
1022 av_log(s
, AV_LOG_ERROR
, "method %s failed: %d%s\n",
1026 av_log(s
, AV_LOG_DEBUG
, "%s\n", rt
->last_reply
);
1033 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
1035 int ff_rtsp_make_setup_request(AVFormatContext
*s
, const char *host
, int port
,
1036 int lower_transport
, const char *real_challenge
)
1038 RTSPState
*rt
= s
->priv_data
;
1039 int rtx
, j
, i
, err
, interleave
= 0;
1040 RTSPStream
*rtsp_st
;
1041 RTSPMessageHeader reply1
, *reply
= &reply1
;
1043 const char *trans_pref
;
1045 if (rt
->transport
== RTSP_TRANSPORT_RDT
)
1046 trans_pref
= "x-pn-tng";
1048 trans_pref
= "RTP/AVP";
1050 /* default timeout: 1 minute */
1053 /* for each stream, make the setup request */
1054 /* XXX: we assume the same server is used for the control of each
1057 for (j
= RTSP_RTP_PORT_MIN
, i
= 0; i
< rt
->nb_rtsp_streams
; ++i
) {
1058 char transport
[2048];
1061 * WMS serves all UDP data over a single connection, the RTX, which
1062 * isn't necessarily the first in the SDP but has to be the first
1063 * to be set up, else the second/third SETUP will fail with a 461.
1065 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
&&
1066 rt
->server_type
== RTSP_SERVER_WMS
) {
1069 for (rtx
= 0; rtx
< rt
->nb_rtsp_streams
; rtx
++) {
1070 int len
= strlen(rt
->rtsp_streams
[rtx
]->control_url
);
1072 !strcmp(rt
->rtsp_streams
[rtx
]->control_url
+ len
- 4,
1076 if (rtx
== rt
->nb_rtsp_streams
)
1077 return -1; /* no RTX found */
1078 rtsp_st
= rt
->rtsp_streams
[rtx
];
1080 rtsp_st
= rt
->rtsp_streams
[i
> rtx ? i
: i
- 1];
1082 rtsp_st
= rt
->rtsp_streams
[i
];
1085 if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP
) {
1088 if (rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) {
1089 port
= reply
->transports
[0].client_port_min
;
1093 /* first try in specified port range */
1094 if (RTSP_RTP_PORT_MIN
!= 0) {
1095 while (j
<= RTSP_RTP_PORT_MAX
) {
1096 ff_url_join(buf
, sizeof(buf
), "rtp", NULL
, host
, -1,
1097 "?localport=%d", j
);
1098 /* we will use two ports per rtp stream (rtp and rtcp) */
1100 if (url_open(&rtsp_st
->rtp_handle
, buf
, URL_RDWR
) == 0)
1106 /* then try on any port */
1107 if (url_open(&rtsp_st
->rtp_handle
, "rtp://", URL_RDONLY
) < 0) {
1108 err
= AVERROR_INVALIDDATA
;
1112 av_log(s
, AV_LOG_ERROR
, "Unable to open an input RTP port\n");
1118 port
= rtp_get_local_rtp_port(rtsp_st
->rtp_handle
);
1120 snprintf(transport
, sizeof(transport
) - 1,
1121 "%s/UDP;", trans_pref
);
1122 if (rt
->server_type
!= RTSP_SERVER_REAL
)
1123 av_strlcat(transport
, "unicast;", sizeof(transport
));
1124 av_strlcatf(transport
, sizeof(transport
),
1125 "client_port=%d", port
);
1126 if (rt
->transport
== RTSP_TRANSPORT_RTP
&&
1127 !(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 0))
1128 av_strlcatf(transport
, sizeof(transport
), "-%d", port
+ 1);
1132 else if (lower_transport
== RTSP_LOWER_TRANSPORT_TCP
) {
1133 /** For WMS streams, the application streams are only used for
1134 * UDP. When trying to set it up for TCP streams, the server
1135 * will return an error. Therefore, we skip those streams. */
1136 if (rt
->server_type
== RTSP_SERVER_WMS
&&
1137 s
->streams
[rtsp_st
->stream_index
]->codec
->codec_type
==
1140 snprintf(transport
, sizeof(transport
) - 1,
1141 "%s/TCP;", trans_pref
);
1142 if (rt
->server_type
== RTSP_SERVER_WMS
)
1143 av_strlcat(transport
, "unicast;", sizeof(transport
));
1144 av_strlcatf(transport
, sizeof(transport
),
1145 "interleaved=%d-%d",
1146 interleave
, interleave
+ 1);
1150 else if (lower_transport
== RTSP_LOWER_TRANSPORT_UDP_MULTICAST
) {
1151 snprintf(transport
, sizeof(transport
) - 1,
1152 "%s/UDP;multicast", trans_pref
);
1155 av_strlcat(transport
, ";mode=receive", sizeof(transport
));
1156 } else if (rt
->server_type
== RTSP_SERVER_REAL
||
1157 rt
->server_type
== RTSP_SERVER_WMS
)
1158 av_strlcat(transport
, ";mode=play", sizeof(transport
));
1159 snprintf(cmd
, sizeof(cmd
),
1160 "Transport: %s\r\n",
1162 if (i
== 0 && rt
->server_type
== RTSP_SERVER_REAL
&& CONFIG_RTPDEC
) {
1163 char real_res
[41], real_csum
[9];
1164 ff_rdt_calc_response_and_checksum(real_res
, real_csum
,
1166 av_strlcatf(cmd
, sizeof(cmd
),
1168 "RealChallenge2: %s, sd=%s\r\n",
1169 rt
->session_id
, real_res
, real_csum
);
1171 ff_rtsp_send_cmd(s
, "SETUP", rtsp_st
->control_url
, cmd
, reply
, NULL
);
1172 if (reply
->status_code
== 461 /* Unsupported protocol */ && i
== 0) {
1175 } else if (reply
->status_code
!= RTSP_STATUS_OK
||
1176 reply
->nb_transports
!= 1) {
1177 err
= AVERROR_INVALIDDATA
;
1181 /* XXX: same protocol for all streams is required */
1183 if (reply
->transports
[0].lower_transport
!= rt
->lower_transport
||
1184 reply
->transports
[0].transport
!= rt
->transport
) {
1185 err
= AVERROR_INVALIDDATA
;
1189 rt
->lower_transport
= reply
->transports
[0].lower_transport
;
1190 rt
->transport
= reply
->transports
[0].transport
;
1193 /* Fail if the server responded with another lower transport mode
1194 * than what we requested. */
1195 if (reply
->transports
[0].lower_transport
!= lower_transport
) {
1196 av_log(s
, AV_LOG_ERROR
, "Nonmatching transport in server reply\n");
1197 err
= AVERROR_INVALIDDATA
;
1201 switch(reply
->transports
[0].lower_transport
) {
1202 case RTSP_LOWER_TRANSPORT_TCP
:
1203 rtsp_st
->interleaved_min
= reply
->transports
[0].interleaved_min
;
1204 rtsp_st
->interleaved_max
= reply
->transports
[0].interleaved_max
;
1207 case RTSP_LOWER_TRANSPORT_UDP
: {
1208 char url
[1024], options
[30] = "";
1210 if (rt
->filter_source
)
1211 av_strlcpy(options
, "?connect=1", sizeof(options
));
1212 /* Use source address if specified */
1213 if (reply
->transports
[0].source
[0]) {
1214 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1215 reply
->transports
[0].source
,
1216 reply
->transports
[0].server_port_min
, options
);
1218 ff_url_join(url
, sizeof(url
), "rtp", NULL
, host
,
1219 reply
->transports
[0].server_port_min
, options
);
1221 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) &&
1222 rtp_set_remote_url(rtsp_st
->rtp_handle
, url
) < 0) {
1223 err
= AVERROR_INVALIDDATA
;
1226 /* Try to initialize the connection state in a
1227 * potential NAT router by sending dummy packets.
1228 * RTP/RTCP dummy packets are used for RDT, too.
1230 if (!(rt
->server_type
== RTSP_SERVER_WMS
&& i
> 1) && s
->iformat
&&
1232 rtp_send_punch_packets(rtsp_st
->rtp_handle
);
1235 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
: {
1236 char url
[1024], namebuf
[50];
1237 struct sockaddr_storage addr
;
1240 if (reply
->transports
[0].destination
.ss_family
) {
1241 addr
= reply
->transports
[0].destination
;
1242 port
= reply
->transports
[0].port_min
;
1243 ttl
= reply
->transports
[0].ttl
;
1245 addr
= rtsp_st
->sdp_ip
;
1246 port
= rtsp_st
->sdp_port
;
1247 ttl
= rtsp_st
->sdp_ttl
;
1249 getnameinfo((struct sockaddr
*) &addr
, sizeof(addr
),
1250 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1251 ff_url_join(url
, sizeof(url
), "rtp", NULL
, namebuf
,
1252 port
, "?ttl=%d", ttl
);
1253 if (url_open(&rtsp_st
->rtp_handle
, url
, URL_RDWR
) < 0) {
1254 err
= AVERROR_INVALIDDATA
;
1261 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
1265 if (reply
->timeout
> 0)
1266 rt
->timeout
= reply
->timeout
;
1268 if (rt
->server_type
== RTSP_SERVER_REAL
)
1269 rt
->need_subscription
= 1;
1274 ff_rtsp_undo_setup(s
);
1278 void ff_rtsp_close_connections(AVFormatContext
*s
)
1280 RTSPState
*rt
= s
->priv_data
;
1281 if (rt
->rtsp_hd_out
!= rt
->rtsp_hd
) url_close(rt
->rtsp_hd_out
);
1282 url_close(rt
->rtsp_hd
);
1283 rt
->rtsp_hd
= rt
->rtsp_hd_out
= NULL
;
1286 int ff_rtsp_connect(AVFormatContext
*s
)
1288 RTSPState
*rt
= s
->priv_data
;
1289 char host
[1024], path
[1024], tcpname
[1024], cmd
[2048], auth
[128];
1290 char *option_list
, *option
, *filename
;
1291 int port
, err
, tcp_fd
;
1292 RTSPMessageHeader reply1
= {0}, *reply
= &reply1
;
1293 int lower_transport_mask
= 0;
1294 char real_challenge
[64] = "";
1295 struct sockaddr_storage peer
;
1296 socklen_t peer_len
= sizeof(peer
);
1298 if (!ff_network_init())
1299 return AVERROR(EIO
);
1301 rt
->control_transport
= RTSP_MODE_PLAIN
;
1302 /* extract hostname and port */
1303 av_url_split(NULL
, 0, auth
, sizeof(auth
),
1304 host
, sizeof(host
), &port
, path
, sizeof(path
), s
->filename
);
1306 av_strlcpy(rt
->auth
, auth
, sizeof(rt
->auth
));
1309 port
= RTSP_DEFAULT_PORT
;
1311 /* search for options */
1312 option_list
= strrchr(path
, '?');
1314 /* Strip out the RTSP specific options, write out the rest of
1315 * the options back into the same string. */
1316 filename
= option_list
;
1317 while (option_list
) {
1318 /* move the option pointer */
1319 option
= ++option_list
;
1320 option_list
= strchr(option_list
, '&');
1324 /* handle the options */
1325 if (!strcmp(option
, "udp")) {
1326 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP
);
1327 } else if (!strcmp(option
, "multicast")) {
1328 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST
);
1329 } else if (!strcmp(option
, "tcp")) {
1330 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1331 } else if(!strcmp(option
, "http")) {
1332 lower_transport_mask
|= (1<< RTSP_LOWER_TRANSPORT_TCP
);
1333 rt
->control_transport
= RTSP_MODE_TUNNEL
;
1334 } else if (!strcmp(option
, "filter_src")) {
1335 rt
->filter_source
= 1;
1337 /* Write options back into the buffer, using memmove instead
1338 * of strcpy since the strings may overlap. */
1339 int len
= strlen(option
);
1340 memmove(++filename
, option
, len
);
1342 if (option_list
) *filename
= '&';
1348 if (!lower_transport_mask
)
1349 lower_transport_mask
= (1 << RTSP_LOWER_TRANSPORT_NB
) - 1;
1352 /* Only UDP or TCP - UDP multicast isn't supported. */
1353 lower_transport_mask
&= (1 << RTSP_LOWER_TRANSPORT_UDP
) |
1354 (1 << RTSP_LOWER_TRANSPORT_TCP
);
1355 if (!lower_transport_mask
|| rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1356 av_log(s
, AV_LOG_ERROR
, "Unsupported lower transport method, "
1357 "only UDP and TCP are supported for output.\n");
1358 err
= AVERROR(EINVAL
);
1363 /* Construct the URI used in request; this is similar to s->filename,
1364 * but with authentication credentials removed and RTSP specific options
1366 ff_url_join(rt
->control_uri
, sizeof(rt
->control_uri
), "rtsp", NULL
,
1367 host
, port
, "%s", path
);
1369 if (rt
->control_transport
== RTSP_MODE_TUNNEL
) {
1370 /* set up initial handshake for tunneling */
1371 char httpname
[1024];
1372 char sessioncookie
[17];
1375 ff_url_join(httpname
, sizeof(httpname
), "http", auth
, host
, port
, "%s", path
);
1376 snprintf(sessioncookie
, sizeof(sessioncookie
), "%08x%08x",
1377 av_get_random_seed(), av_get_random_seed());
1380 if (url_alloc(&rt
->rtsp_hd
, httpname
, URL_RDONLY
) < 0) {
1385 /* generate GET headers */
1386 snprintf(headers
, sizeof(headers
),
1387 "x-sessioncookie: %s\r\n"
1388 "Accept: application/x-rtsp-tunnelled\r\n"
1389 "Pragma: no-cache\r\n"
1390 "Cache-Control: no-cache\r\n",
1392 ff_http_set_headers(rt
->rtsp_hd
, headers
);
1394 /* complete the connection */
1395 if (url_connect(rt
->rtsp_hd
)) {
1401 if (url_alloc(&rt
->rtsp_hd_out
, httpname
, URL_WRONLY
) < 0 ) {
1406 /* generate POST headers */
1407 snprintf(headers
, sizeof(headers
),
1408 "x-sessioncookie: %s\r\n"
1409 "Content-Type: application/x-rtsp-tunnelled\r\n"
1410 "Pragma: no-cache\r\n"
1411 "Cache-Control: no-cache\r\n"
1412 "Content-Length: 32767\r\n"
1413 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1415 ff_http_set_headers(rt
->rtsp_hd_out
, headers
);
1416 ff_http_set_chunked_transfer_encoding(rt
->rtsp_hd_out
, 0);
1418 /* Initialize the authentication state for the POST session. The HTTP
1419 * protocol implementation doesn't properly handle multi-pass
1420 * authentication for POST requests, since it would require one of
1422 * - implementing Expect: 100-continue, which many HTTP servers
1423 * don't support anyway, even less the RTSP servers that do HTTP
1425 * - sending the whole POST data until getting a 401 reply specifying
1426 * what authentication method to use, then resending all that data
1427 * - waiting for potential 401 replies directly after sending the
1428 * POST header (waiting for some unspecified time)
1429 * Therefore, we copy the full auth state, which works for both basic
1430 * and digest. (For digest, we would have to synchronize the nonce
1431 * count variable between the two sessions, if we'd do more requests
1432 * with the original session, though.)
1434 ff_http_init_auth_state(rt
->rtsp_hd_out
, rt
->rtsp_hd
);
1436 /* complete the connection */
1437 if (url_connect(rt
->rtsp_hd_out
)) {
1442 /* open the tcp connection */
1443 ff_url_join(tcpname
, sizeof(tcpname
), "tcp", NULL
, host
, port
, NULL
);
1444 if (url_open(&rt
->rtsp_hd
, tcpname
, URL_RDWR
) < 0) {
1448 rt
->rtsp_hd_out
= rt
->rtsp_hd
;
1452 tcp_fd
= url_get_file_handle(rt
->rtsp_hd
);
1453 if (!getpeername(tcp_fd
, (struct sockaddr
*) &peer
, &peer_len
)) {
1454 getnameinfo((struct sockaddr
*) &peer
, peer_len
, host
, sizeof(host
),
1455 NULL
, 0, NI_NUMERICHOST
);
1458 /* request options supported by the server; this also detects server
1460 for (rt
->server_type
= RTSP_SERVER_RTP
;;) {
1462 if (rt
->server_type
== RTSP_SERVER_REAL
)
1465 * The following entries are required for proper
1466 * streaming from a Realmedia server. They are
1467 * interdependent in some way although we currently
1468 * don't quite understand how. Values were copied
1469 * from mplayer SVN r23589.
1470 * @param CompanyID is a 16-byte ID in base64
1471 * @param ClientChallenge is a 16-byte ID in hex
1473 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1474 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1475 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1476 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1478 ff_rtsp_send_cmd(s
, "OPTIONS", rt
->control_uri
, cmd
, reply
, NULL
);
1479 if (reply
->status_code
!= RTSP_STATUS_OK
) {
1480 err
= AVERROR_INVALIDDATA
;
1484 /* detect server type if not standard-compliant RTP */
1485 if (rt
->server_type
!= RTSP_SERVER_REAL
&& reply
->real_challenge
[0]) {
1486 rt
->server_type
= RTSP_SERVER_REAL
;
1488 } else if (!strncasecmp(reply
->server
, "WMServer/", 9)) {
1489 rt
->server_type
= RTSP_SERVER_WMS
;
1490 } else if (rt
->server_type
== RTSP_SERVER_REAL
)
1491 strcpy(real_challenge
, reply
->real_challenge
);
1495 if (s
->iformat
&& CONFIG_RTSP_DEMUXER
)
1496 err
= ff_rtsp_setup_input_streams(s
, reply
);
1497 else if (CONFIG_RTSP_MUXER
)
1498 err
= ff_rtsp_setup_output_streams(s
, host
);
1503 int lower_transport
= ff_log2_tab
[lower_transport_mask
&
1504 ~(lower_transport_mask
- 1)];
1506 err
= ff_rtsp_make_setup_request(s
, host
, port
, lower_transport
,
1507 rt
->server_type
== RTSP_SERVER_REAL ?
1508 real_challenge
: NULL
);
1511 lower_transport_mask
&= ~(1 << lower_transport
);
1512 if (lower_transport_mask
== 0 && err
== 1) {
1513 err
= FF_NETERROR(EPROTONOSUPPORT
);
1518 rt
->lower_transport_mask
= lower_transport_mask
;
1519 av_strlcpy(rt
->real_challenge
, real_challenge
, sizeof(rt
->real_challenge
));
1520 rt
->state
= RTSP_STATE_IDLE
;
1521 rt
->seek_timestamp
= 0; /* default is to start stream at position zero */
1524 ff_rtsp_close_streams(s
);
1525 ff_rtsp_close_connections(s
);
1526 if (reply
->status_code
>=300 && reply
->status_code
< 400 && s
->iformat
) {
1527 av_strlcpy(s
->filename
, reply
->location
, sizeof(s
->filename
));
1528 av_log(s
, AV_LOG_INFO
, "Status %d: Redirecting to %s\n",
1536 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1539 static int udp_read_packet(AVFormatContext
*s
, RTSPStream
**prtsp_st
,
1540 uint8_t *buf
, int buf_size
, int64_t wait_end
)
1542 RTSPState
*rt
= s
->priv_data
;
1543 RTSPStream
*rtsp_st
;
1545 int fd
, fd_rtcp
, fd_max
, n
, i
, ret
, tcp_fd
, timeout_cnt
= 0;
1549 if (url_interrupt_cb())
1550 return AVERROR(EINTR
);
1551 if (wait_end
&& wait_end
- av_gettime() < 0)
1552 return AVERROR(EAGAIN
);
1555 tcp_fd
= fd_max
= url_get_file_handle(rt
->rtsp_hd
);
1556 FD_SET(tcp_fd
, &rfds
);
1561 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1562 rtsp_st
= rt
->rtsp_streams
[i
];
1563 if (rtsp_st
->rtp_handle
) {
1564 fd
= url_get_file_handle(rtsp_st
->rtp_handle
);
1565 fd_rtcp
= rtp_get_rtcp_file_handle(rtsp_st
->rtp_handle
);
1566 if (FFMAX(fd
, fd_rtcp
) > fd_max
)
1567 fd_max
= FFMAX(fd
, fd_rtcp
);
1569 FD_SET(fd_rtcp
, &rfds
);
1573 tv
.tv_usec
= SELECT_TIMEOUT_MS
* 1000;
1574 n
= select(fd_max
+ 1, &rfds
, NULL
, NULL
, &tv
);
1577 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1578 rtsp_st
= rt
->rtsp_streams
[i
];
1579 if (rtsp_st
->rtp_handle
) {
1580 fd
= url_get_file_handle(rtsp_st
->rtp_handle
);
1581 fd_rtcp
= rtp_get_rtcp_file_handle(rtsp_st
->rtp_handle
);
1582 if (FD_ISSET(fd_rtcp
, &rfds
) || FD_ISSET(fd
, &rfds
)) {
1583 ret
= url_read(rtsp_st
->rtp_handle
, buf
, buf_size
);
1585 *prtsp_st
= rtsp_st
;
1591 #if CONFIG_RTSP_DEMUXER
1592 if (tcp_fd
!= -1 && FD_ISSET(tcp_fd
, &rfds
)) {
1593 RTSPMessageHeader reply
;
1595 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 0, NULL
);
1598 /* XXX: parse message */
1599 if (rt
->state
!= RTSP_STATE_STREAMING
)
1603 } else if (n
== 0 && ++timeout_cnt
>= MAX_TIMEOUTS
) {
1604 return FF_NETERROR(ETIMEDOUT
);
1605 } else if (n
< 0 && errno
!= EINTR
)
1606 return AVERROR(errno
);
1610 int ff_rtsp_fetch_packet(AVFormatContext
*s
, AVPacket
*pkt
)
1612 RTSPState
*rt
= s
->priv_data
;
1614 RTSPStream
*rtsp_st
, *first_queue_st
= NULL
;
1615 int64_t wait_end
= 0;
1617 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1620 /* get next frames from the same RTP packet */
1621 if (rt
->cur_transport_priv
) {
1622 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1623 ret
= ff_rdt_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1625 ret
= rtp_parse_packet(rt
->cur_transport_priv
, pkt
, NULL
, 0);
1627 rt
->cur_transport_priv
= NULL
;
1629 } else if (ret
== 1) {
1632 rt
->cur_transport_priv
= NULL
;
1635 if (rt
->transport
== RTSP_TRANSPORT_RTP
) {
1637 int64_t first_queue_time
= 0;
1638 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1639 RTPDemuxContext
*rtpctx
= rt
->rtsp_streams
[i
]->transport_priv
;
1643 queue_time
= ff_rtp_queued_packet_time(rtpctx
);
1644 if (queue_time
&& (queue_time
- first_queue_time
< 0 ||
1645 !first_queue_time
)) {
1646 first_queue_time
= queue_time
;
1647 first_queue_st
= rt
->rtsp_streams
[i
];
1650 if (first_queue_time
)
1651 wait_end
= first_queue_time
+ s
->max_delay
;
1654 /* read next RTP packet */
1657 rt
->recvbuf
= av_malloc(RECVBUF_SIZE
);
1659 return AVERROR(ENOMEM
);
1662 switch(rt
->lower_transport
) {
1664 #if CONFIG_RTSP_DEMUXER
1665 case RTSP_LOWER_TRANSPORT_TCP
:
1666 len
= ff_rtsp_tcp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
);
1669 case RTSP_LOWER_TRANSPORT_UDP
:
1670 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST
:
1671 len
= udp_read_packet(s
, &rtsp_st
, rt
->recvbuf
, RECVBUF_SIZE
, wait_end
);
1672 if (len
>=0 && rtsp_st
->transport_priv
&& rt
->transport
== RTSP_TRANSPORT_RTP
)
1673 rtp_check_and_send_back_rr(rtsp_st
->transport_priv
, len
);
1676 if (len
== AVERROR(EAGAIN
) && first_queue_st
&&
1677 rt
->transport
== RTSP_TRANSPORT_RTP
) {
1678 rtsp_st
= first_queue_st
;
1679 ret
= rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, NULL
, 0);
1686 if (rt
->transport
== RTSP_TRANSPORT_RDT
) {
1687 ret
= ff_rdt_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1689 ret
= rtp_parse_packet(rtsp_st
->transport_priv
, pkt
, &rt
->recvbuf
, len
);
1691 /* Either bad packet, or a RTCP packet. Check if the
1692 * first_rtcp_ntp_time field was initialized. */
1693 RTPDemuxContext
*rtpctx
= rtsp_st
->transport_priv
;
1694 if (rtpctx
->first_rtcp_ntp_time
!= AV_NOPTS_VALUE
) {
1695 /* first_rtcp_ntp_time has been initialized for this stream,
1696 * copy the same value to all other uninitialized streams,
1697 * in order to map their timestamp origin to the same ntp time
1700 AVStream
*st
= NULL
;
1701 if (rtsp_st
->stream_index
>= 0)
1702 st
= s
->streams
[rtsp_st
->stream_index
];
1703 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1704 RTPDemuxContext
*rtpctx2
= rt
->rtsp_streams
[i
]->transport_priv
;
1705 AVStream
*st2
= NULL
;
1706 if (rt
->rtsp_streams
[i
]->stream_index
>= 0)
1707 st2
= s
->streams
[rt
->rtsp_streams
[i
]->stream_index
];
1708 if (rtpctx2
&& st
&& st2
&&
1709 rtpctx2
->first_rtcp_ntp_time
== AV_NOPTS_VALUE
) {
1710 rtpctx2
->first_rtcp_ntp_time
= rtpctx
->first_rtcp_ntp_time
;
1711 rtpctx2
->rtcp_ts_offset
= av_rescale_q(
1712 rtpctx
->rtcp_ts_offset
, st
->time_base
,
1717 if (ret
== -RTCP_BYE
) {
1720 av_log(s
, AV_LOG_DEBUG
, "Received BYE for stream %d (%d/%d)\n",
1721 rtsp_st
->stream_index
, rt
->nb_byes
, rt
->nb_rtsp_streams
);
1723 if (rt
->nb_byes
== rt
->nb_rtsp_streams
)
1732 /* more packets may follow, so we save the RTP context */
1733 rt
->cur_transport_priv
= rtsp_st
->transport_priv
;
1737 #endif /* CONFIG_RTPDEC */
1739 #if CONFIG_SDP_DEMUXER
1740 static int sdp_probe(AVProbeData
*p1
)
1742 const char *p
= p1
->buf
, *p_end
= p1
->buf
+ p1
->buf_size
;
1744 /* we look for a line beginning "c=IN IP" */
1745 while (p
< p_end
&& *p
!= '\0') {
1746 if (p
+ sizeof("c=IN IP") - 1 < p_end
&&
1747 av_strstart(p
, "c=IN IP", NULL
))
1748 return AVPROBE_SCORE_MAX
/ 2;
1750 while (p
< p_end
- 1 && *p
!= '\n') p
++;
1759 static int sdp_read_header(AVFormatContext
*s
, AVFormatParameters
*ap
)
1761 RTSPState
*rt
= s
->priv_data
;
1762 RTSPStream
*rtsp_st
;
1767 if (!ff_network_init())
1768 return AVERROR(EIO
);
1770 /* read the whole sdp file */
1771 /* XXX: better loading */
1772 content
= av_malloc(SDP_MAX_SIZE
);
1773 size
= get_buffer(s
->pb
, content
, SDP_MAX_SIZE
- 1);
1776 return AVERROR_INVALIDDATA
;
1778 content
[size
] ='\0';
1780 ff_sdp_parse(s
, content
);
1783 /* open each RTP stream */
1784 for (i
= 0; i
< rt
->nb_rtsp_streams
; i
++) {
1786 rtsp_st
= rt
->rtsp_streams
[i
];
1788 getnameinfo((struct sockaddr
*) &rtsp_st
->sdp_ip
, sizeof(rtsp_st
->sdp_ip
),
1789 namebuf
, sizeof(namebuf
), NULL
, 0, NI_NUMERICHOST
);
1790 ff_url_join(url
, sizeof(url
), "rtp", NULL
,
1791 namebuf
, rtsp_st
->sdp_port
,
1792 "?localport=%d&ttl=%d", rtsp_st
->sdp_port
,
1794 if (url_open(&rtsp_st
->rtp_handle
, url
, URL_RDWR
) < 0) {
1795 err
= AVERROR_INVALIDDATA
;
1798 if ((err
= rtsp_open_transport_ctx(s
, rtsp_st
)))
1803 ff_rtsp_close_streams(s
);
1808 static int sdp_read_close(AVFormatContext
*s
)
1810 ff_rtsp_close_streams(s
);
1815 AVInputFormat sdp_demuxer
= {
1817 NULL_IF_CONFIG_SMALL("SDP"),
1821 ff_rtsp_fetch_packet
,
1824 #endif /* CONFIG_SDP_DEMUXER */
1826 #if CONFIG_RTP_DEMUXER
1827 static int rtp_probe(AVProbeData
*p
)
1829 if (av_strstart(p
->filename
, "rtp:", NULL
))
1830 return AVPROBE_SCORE_MAX
;
1834 static int rtp_read_header(AVFormatContext
*s
,
1835 AVFormatParameters
*ap
)
1837 uint8_t recvbuf
[1500];
1838 char host
[500], sdp
[500];
1840 URLContext
* in
= NULL
;
1842 AVCodecContext codec
;
1843 struct sockaddr_storage addr
;
1845 socklen_t addrlen
= sizeof(addr
);
1847 if (!ff_network_init())
1848 return AVERROR(EIO
);
1850 ret
= url_open(&in
, s
->filename
, URL_RDONLY
);
1855 ret
= url_read(in
, recvbuf
, sizeof(recvbuf
));
1856 if (ret
== AVERROR(EAGAIN
))
1861 av_log(s
, AV_LOG_WARNING
, "Received too short packet\n");
1865 if ((recvbuf
[0] & 0xc0) != 0x80) {
1866 av_log(s
, AV_LOG_WARNING
, "Unsupported RTP version packet "
1871 payload_type
= recvbuf
[1] & 0x7f;
1874 getsockname(url_get_file_handle(in
), (struct sockaddr
*) &addr
, &addrlen
);
1878 memset(&codec
, 0, sizeof(codec
));
1879 if (ff_rtp_get_codec_info(&codec
, payload_type
)) {
1880 av_log(s
, AV_LOG_ERROR
, "Unable to receive RTP payload type %d "
1881 "without an SDP file describing it\n",
1885 if (codec
.codec_type
!= AVMEDIA_TYPE_DATA
) {
1886 av_log(s
, AV_LOG_WARNING
, "Guessing on RTP content - if not received "
1887 "properly you need an SDP file "
1891 av_url_split(NULL
, 0, NULL
, 0, host
, sizeof(host
), &port
,
1892 NULL
, 0, s
->filename
);
1894 snprintf(sdp
, sizeof(sdp
),
1895 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
1896 addr
.ss_family
== AF_INET ?
4 : 6, host
,
1897 codec
.codec_type
== AVMEDIA_TYPE_DATA ?
"application" :
1898 codec
.codec_type
== AVMEDIA_TYPE_VIDEO ?
"video" : "audio",
1899 port
, payload_type
);
1900 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
);
1902 init_put_byte(&pb
, sdp
, strlen(sdp
), 0, NULL
, NULL
, NULL
, NULL
);
1905 /* sdp_read_header initializes this again */
1908 ret
= sdp_read_header(s
, ap
);
1919 AVInputFormat rtp_demuxer
= {
1921 NULL_IF_CONFIG_SMALL("RTP input format"),
1925 ff_rtsp_fetch_packet
,
1927 .flags
= AVFMT_NOFILE
,
1929 #endif /* CONFIG_RTP_DEMUXER */