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/avassert.h"
23 #include "libavutil/base64.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/mathematics.h"
27 #include "libavutil/parseutils.h"
28 #include "libavutil/random_seed.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/time.h"
33 #include "avio_internal.h"
40 #include "os_support.h"
46 #include "rtpdec_formats.h"
47 #include "rtpenc_chain.h"
54 /* Timeout values for socket poll, in ms,
55 * and read_packet(), in seconds */
56 #define POLL_TIMEOUT_MS 100
57 #define READ_PACKET_TIMEOUT_S 10
58 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
59 #define SDP_MAX_SIZE 16384
60 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
61 #define DEFAULT_REORDERING_DELAY 100000
63 #define OFFSET(x) offsetof(RTSPState, x)
64 #define DEC AV_OPT_FLAG_DECODING_PARAM
65 #define ENC AV_OPT_FLAG_ENCODING_PARAM
67 #define RTSP_FLAG_OPTS(name, longname) \
68 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
69 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }, \
70 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
72 #define RTSP_MEDIATYPE_OPTS(name, longname) \
73 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
74 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
76 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
78 #define RTSP_REORDERING_OPTS() \
79 { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
81 const AVOption ff_rtsp_options[] = {
82 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
83 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
84 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
85 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
86 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
87 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
88 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
89 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
90 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
91 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
92 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
93 { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
94 RTSP_REORDERING_OPTS(),
98 static const AVOption sdp_options[] = {
99 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
100 { "custom_io", "Use custom IO", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
101 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
102 RTSP_REORDERING_OPTS(),
106 static const AVOption rtp_options[] = {
107 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
108 RTSP_REORDERING_OPTS(),
112 static void get_word_until_chars(char *buf, int buf_size,
113 const char *sep, const char **pp)
119 p += strspn(p, SPACE_CHARS);
121 while (!strchr(sep, *p) && *p != '\0') {
122 if ((q - buf) < buf_size - 1)
131 static void get_word_sep(char *buf, int buf_size, const char *sep,
134 if (**pp == '/') (*pp)++;
135 get_word_until_chars(buf, buf_size, sep, pp);
138 static void get_word(char *buf, int buf_size, const char **pp)
140 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
143 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
145 * Used for seeking in the rtp stream.
147 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
151 p += strspn(p, SPACE_CHARS);
152 if (!av_stristart(p, "npt=", &p))
155 *start = AV_NOPTS_VALUE;
156 *end = AV_NOPTS_VALUE;
158 get_word_sep(buf, sizeof(buf), "-", &p);
159 av_parse_time(start, buf, 1);
162 get_word_sep(buf, sizeof(buf), "-", &p);
163 av_parse_time(end, buf, 1);
167 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
169 struct addrinfo hints = { 0 }, *ai = NULL;
170 hints.ai_flags = AI_NUMERICHOST;
171 if (getaddrinfo(buf, NULL, &hints, &ai))
173 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
179 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
180 RTSPStream *rtsp_st, AVCodecContext *codec)
185 codec->codec_id = handler->codec_id;
186 rtsp_st->dynamic_handler = handler;
187 if (handler->alloc) {
188 rtsp_st->dynamic_protocol_context = handler->alloc();
189 if (!rtsp_st->dynamic_protocol_context)
190 rtsp_st->dynamic_handler = NULL;
194 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
195 static int sdp_parse_rtpmap(AVFormatContext *s,
196 AVStream *st, RTSPStream *rtsp_st,
197 int payload_type, const char *p)
199 AVCodecContext *codec = st->codec;
205 /* See if we can handle this kind of payload.
206 * The space should normally not be there but some Real streams or
207 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
208 * have a trailing space. */
209 get_word_sep(buf, sizeof(buf), "/ ", &p);
210 if (payload_type < RTP_PT_PRIVATE) {
211 /* We are in a standard case
212 * (from http://www.iana.org/assignments/rtp-parameters). */
213 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
216 if (codec->codec_id == AV_CODEC_ID_NONE) {
217 RTPDynamicProtocolHandler *handler =
218 ff_rtp_handler_find_by_name(buf, codec->codec_type);
219 init_rtp_handler(handler, rtsp_st, codec);
220 /* If no dynamic handler was found, check with the list of standard
221 * allocated types, if such a stream for some reason happens to
222 * use a private payload type. This isn't handled in rtpdec.c, since
223 * the format name from the rtpmap line never is passed into rtpdec. */
224 if (!rtsp_st->dynamic_handler)
225 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
228 c = avcodec_find_decoder(codec->codec_id);
234 get_word_sep(buf, sizeof(buf), "/", &p);
236 switch (codec->codec_type) {
237 case AVMEDIA_TYPE_AUDIO:
238 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
239 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
240 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
242 codec->sample_rate = i;
243 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
244 get_word_sep(buf, sizeof(buf), "/", &p);
249 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
251 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
254 case AVMEDIA_TYPE_VIDEO:
255 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
257 avpriv_set_pts_info(st, 32, 1, i);
262 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
263 rtsp_st->dynamic_handler->init(s, st->index,
264 rtsp_st->dynamic_protocol_context);
268 /* parse the attribute line from the fmtp a line of an sdp response. This
269 * is broken out as a function because it is used in rtp_h264.c, which is
271 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
272 char *value, int value_size)
274 *p += strspn(*p, SPACE_CHARS);
276 get_word_sep(attr, attr_size, "=", p);
279 get_word_sep(value, value_size, ";", p);
287 typedef struct SDPParseState {
289 struct sockaddr_storage default_ip;
291 int skip_media; ///< set if an unknown m= line occurs
294 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
295 int letter, const char *buf)
297 RTSPState *rt = s->priv_data;
298 char buf1[64], st_type[64];
300 enum AVMediaType codec_type;
304 struct sockaddr_storage sdp_ip;
307 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
310 if (s1->skip_media && letter != 'm')
314 get_word(buf1, sizeof(buf1), &p);
315 if (strcmp(buf1, "IN") != 0)
317 get_word(buf1, sizeof(buf1), &p);
318 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
320 get_word_sep(buf1, sizeof(buf1), "/", &p);
321 if (get_sockaddr(buf1, &sdp_ip))
326 get_word_sep(buf1, sizeof(buf1), "/", &p);
329 if (s->nb_streams == 0) {
330 s1->default_ip = sdp_ip;
331 s1->default_ttl = ttl;
333 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
334 rtsp_st->sdp_ip = sdp_ip;
335 rtsp_st->sdp_ttl = ttl;
339 av_dict_set(&s->metadata, "title", p, 0);
342 if (s->nb_streams == 0) {
343 av_dict_set(&s->metadata, "comment", p, 0);
350 codec_type = AVMEDIA_TYPE_UNKNOWN;
351 get_word(st_type, sizeof(st_type), &p);
352 if (!strcmp(st_type, "audio")) {
353 codec_type = AVMEDIA_TYPE_AUDIO;
354 } else if (!strcmp(st_type, "video")) {
355 codec_type = AVMEDIA_TYPE_VIDEO;
356 } else if (!strcmp(st_type, "application")) {
357 codec_type = AVMEDIA_TYPE_DATA;
359 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
363 rtsp_st = av_mallocz(sizeof(RTSPStream));
366 rtsp_st->stream_index = -1;
367 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
369 rtsp_st->sdp_ip = s1->default_ip;
370 rtsp_st->sdp_ttl = s1->default_ttl;
372 get_word(buf1, sizeof(buf1), &p); /* port */
373 rtsp_st->sdp_port = atoi(buf1);
375 get_word(buf1, sizeof(buf1), &p); /* protocol */
376 if (!strcmp(buf1, "udp"))
377 rt->transport = RTSP_TRANSPORT_RAW;
378 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
379 rtsp_st->feedback = 1;
381 /* XXX: handle list of formats */
382 get_word(buf1, sizeof(buf1), &p); /* format list */
383 rtsp_st->sdp_payload_type = atoi(buf1);
385 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
386 /* no corresponding stream */
387 if (rt->transport == RTSP_TRANSPORT_RAW) {
388 if (!rt->ts && CONFIG_RTPDEC)
389 rt->ts = ff_mpegts_parse_open(s);
391 RTPDynamicProtocolHandler *handler;
392 handler = ff_rtp_handler_find_by_id(
393 rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
394 init_rtp_handler(handler, rtsp_st, NULL);
395 if (handler && handler->init)
396 handler->init(s, -1, rtsp_st->dynamic_protocol_context);
398 } else if (rt->server_type == RTSP_SERVER_WMS &&
399 codec_type == AVMEDIA_TYPE_DATA) {
400 /* RTX stream, a stream that carries all the other actual
401 * audio/video streams. Don't expose this to the callers. */
403 st = avformat_new_stream(s, NULL);
406 st->id = rt->nb_rtsp_streams - 1;
407 rtsp_st->stream_index = st->index;
408 st->codec->codec_type = codec_type;
409 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
410 RTPDynamicProtocolHandler *handler;
411 /* if standard payload type, we can find the codec right now */
412 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
413 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
414 st->codec->sample_rate > 0)
415 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
416 /* Even static payload types may need a custom depacketizer */
417 handler = ff_rtp_handler_find_by_id(
418 rtsp_st->sdp_payload_type, st->codec->codec_type);
419 init_rtp_handler(handler, rtsp_st, st->codec);
420 if (handler && handler->init)
421 handler->init(s, st->index,
422 rtsp_st->dynamic_protocol_context);
425 /* put a default control url */
426 av_strlcpy(rtsp_st->control_url, rt->control_uri,
427 sizeof(rtsp_st->control_url));
430 if (av_strstart(p, "control:", &p)) {
431 if (s->nb_streams == 0) {
432 if (!strncmp(p, "rtsp://", 7))
433 av_strlcpy(rt->control_uri, p,
434 sizeof(rt->control_uri));
437 /* get the control url */
438 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
440 /* XXX: may need to add full url resolution */
441 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
443 if (proto[0] == '\0') {
444 /* relative control URL */
445 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
446 av_strlcat(rtsp_st->control_url, "/",
447 sizeof(rtsp_st->control_url));
448 av_strlcat(rtsp_st->control_url, p,
449 sizeof(rtsp_st->control_url));
451 av_strlcpy(rtsp_st->control_url, p,
452 sizeof(rtsp_st->control_url));
454 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
455 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
456 get_word(buf1, sizeof(buf1), &p);
457 payload_type = atoi(buf1);
458 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
459 if (rtsp_st->stream_index >= 0) {
460 st = s->streams[rtsp_st->stream_index];
461 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
463 } else if (av_strstart(p, "fmtp:", &p) ||
464 av_strstart(p, "framesize:", &p)) {
465 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
466 // let dynamic protocol handlers have a stab at the line.
467 get_word(buf1, sizeof(buf1), &p);
468 payload_type = atoi(buf1);
469 for (i = 0; i < rt->nb_rtsp_streams; i++) {
470 rtsp_st = rt->rtsp_streams[i];
471 if (rtsp_st->sdp_payload_type == payload_type &&
472 rtsp_st->dynamic_handler &&
473 rtsp_st->dynamic_handler->parse_sdp_a_line)
474 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
475 rtsp_st->dynamic_protocol_context, buf);
477 } else if (av_strstart(p, "range:", &p)) {
480 // this is so that seeking on a streamed file can work.
481 rtsp_parse_range_npt(p, &start, &end);
482 s->start_time = start;
483 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
484 s->duration = (end == AV_NOPTS_VALUE) ?
485 AV_NOPTS_VALUE : end - start;
486 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
488 rt->transport = RTSP_TRANSPORT_RDT;
489 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
491 st = s->streams[s->nb_streams - 1];
492 st->codec->sample_rate = atoi(p);
493 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
495 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
496 get_word(buf1, sizeof(buf1), &p); // ignore tag
497 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
498 p += strspn(p, SPACE_CHARS);
499 if (av_strstart(p, "inline:", &p))
500 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
502 if (rt->server_type == RTSP_SERVER_WMS)
503 ff_wms_parse_sdp_a_line(s, p);
504 if (s->nb_streams > 0) {
505 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
507 if (rt->server_type == RTSP_SERVER_REAL)
508 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
510 if (rtsp_st->dynamic_handler &&
511 rtsp_st->dynamic_handler->parse_sdp_a_line)
512 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
513 rtsp_st->stream_index,
514 rtsp_st->dynamic_protocol_context, buf);
521 int ff_sdp_parse(AVFormatContext *s, const char *content)
523 RTSPState *rt = s->priv_data;
526 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
527 * contain long SDP lines containing complete ASF Headers (several
528 * kB) or arrays of MDPR (RM stream descriptor) headers plus
529 * "rulebooks" describing their properties. Therefore, the SDP line
532 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
533 * in rtpdec_xiph.c. */
535 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
539 p += strspn(p, SPACE_CHARS);
547 /* get the content */
549 while (*p != '\n' && *p != '\r' && *p != '\0') {
550 if ((q - buf) < sizeof(buf) - 1)
555 sdp_parse_line(s, s1, letter, buf);
557 while (*p != '\n' && *p != '\0')
562 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
563 if (!rt->p) return AVERROR(ENOMEM);
566 #endif /* CONFIG_RTPDEC */
568 void ff_rtsp_undo_setup(AVFormatContext *s)
570 RTSPState *rt = s->priv_data;
573 for (i = 0; i < rt->nb_rtsp_streams; i++) {
574 RTSPStream *rtsp_st = rt->rtsp_streams[i];
577 if (rtsp_st->transport_priv) {
579 AVFormatContext *rtpctx = rtsp_st->transport_priv;
580 av_write_trailer(rtpctx);
581 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
583 avio_close_dyn_buf(rtpctx->pb, &ptr);
586 avio_close(rtpctx->pb);
588 avformat_free_context(rtpctx);
589 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
590 ff_rdt_parse_close(rtsp_st->transport_priv);
591 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
592 ff_rtp_parse_close(rtsp_st->transport_priv);
594 rtsp_st->transport_priv = NULL;
595 if (rtsp_st->rtp_handle)
596 ffurl_close(rtsp_st->rtp_handle);
597 rtsp_st->rtp_handle = NULL;
601 /* close and free RTSP streams */
602 void ff_rtsp_close_streams(AVFormatContext *s)
604 RTSPState *rt = s->priv_data;
608 ff_rtsp_undo_setup(s);
609 for (i = 0; i < rt->nb_rtsp_streams; i++) {
610 rtsp_st = rt->rtsp_streams[i];
612 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
613 rtsp_st->dynamic_handler->free(
614 rtsp_st->dynamic_protocol_context);
618 av_free(rt->rtsp_streams);
620 avformat_close_input(&rt->asf_ctx);
622 if (rt->ts && CONFIG_RTPDEC)
623 ff_mpegts_parse_close(rt->ts);
625 av_free(rt->recvbuf);
628 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
630 RTSPState *rt = s->priv_data;
632 int reordering_queue_size = rt->reordering_queue_size;
633 if (reordering_queue_size < 0) {
634 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
635 reordering_queue_size = 0;
637 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
640 /* open the RTP context */
641 if (rtsp_st->stream_index >= 0)
642 st = s->streams[rtsp_st->stream_index];
644 s->ctx_flags |= AVFMTCTX_NOHEADER;
646 if (s->oformat && CONFIG_RTSP_MUXER) {
647 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
649 RTSP_TCP_MAX_PACKET_SIZE,
650 rtsp_st->stream_index);
651 /* Ownership of rtp_handle is passed to the rtp mux context */
652 rtsp_st->rtp_handle = NULL;
655 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
656 return 0; // Don't need to open any parser here
657 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
658 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
659 rtsp_st->dynamic_protocol_context,
660 rtsp_st->dynamic_handler);
661 else if (CONFIG_RTPDEC)
662 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
663 rtsp_st->sdp_payload_type,
664 reordering_queue_size);
666 if (!rtsp_st->transport_priv) {
667 return AVERROR(ENOMEM);
668 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
669 if (rtsp_st->dynamic_handler) {
670 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
671 rtsp_st->dynamic_protocol_context,
672 rtsp_st->dynamic_handler);
674 if (rtsp_st->crypto_suite[0])
675 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
676 rtsp_st->crypto_suite,
677 rtsp_st->crypto_params);
683 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
684 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
691 q += strspn(q, SPACE_CHARS);
692 v = strtol(q, &p, 10);
696 v = strtol(p, &p, 10);
705 /* XXX: only one transport specification is parsed */
706 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
708 char transport_protocol[16];
710 char lower_transport[16];
712 RTSPTransportField *th;
715 reply->nb_transports = 0;
718 p += strspn(p, SPACE_CHARS);
722 th = &reply->transports[reply->nb_transports];
724 get_word_sep(transport_protocol, sizeof(transport_protocol),
726 if (!av_strcasecmp (transport_protocol, "rtp")) {
727 get_word_sep(profile, sizeof(profile), "/;,", &p);
728 lower_transport[0] = '\0';
729 /* rtp/avp/<protocol> */
731 get_word_sep(lower_transport, sizeof(lower_transport),
734 th->transport = RTSP_TRANSPORT_RTP;
735 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
736 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
737 /* x-pn-tng/<protocol> */
738 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
740 th->transport = RTSP_TRANSPORT_RDT;
741 } else if (!av_strcasecmp(transport_protocol, "raw")) {
742 get_word_sep(profile, sizeof(profile), "/;,", &p);
743 lower_transport[0] = '\0';
744 /* raw/raw/<protocol> */
746 get_word_sep(lower_transport, sizeof(lower_transport),
749 th->transport = RTSP_TRANSPORT_RAW;
751 if (!av_strcasecmp(lower_transport, "TCP"))
752 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
754 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
758 /* get each parameter */
759 while (*p != '\0' && *p != ',') {
760 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
761 if (!strcmp(parameter, "port")) {
764 rtsp_parse_range(&th->port_min, &th->port_max, &p);
766 } else if (!strcmp(parameter, "client_port")) {
769 rtsp_parse_range(&th->client_port_min,
770 &th->client_port_max, &p);
772 } else if (!strcmp(parameter, "server_port")) {
775 rtsp_parse_range(&th->server_port_min,
776 &th->server_port_max, &p);
778 } else if (!strcmp(parameter, "interleaved")) {
781 rtsp_parse_range(&th->interleaved_min,
782 &th->interleaved_max, &p);
784 } else if (!strcmp(parameter, "multicast")) {
785 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
786 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
787 } else if (!strcmp(parameter, "ttl")) {
791 th->ttl = strtol(p, &end, 10);
794 } else if (!strcmp(parameter, "destination")) {
797 get_word_sep(buf, sizeof(buf), ";,", &p);
798 get_sockaddr(buf, &th->destination);
800 } else if (!strcmp(parameter, "source")) {
803 get_word_sep(buf, sizeof(buf), ";,", &p);
804 av_strlcpy(th->source, buf, sizeof(th->source));
806 } else if (!strcmp(parameter, "mode")) {
809 get_word_sep(buf, sizeof(buf), ";, ", &p);
810 if (!strcmp(buf, "record") ||
811 !strcmp(buf, "receive"))
816 while (*p != ';' && *p != '\0' && *p != ',')
824 reply->nb_transports++;
828 static void handle_rtp_info(RTSPState *rt, const char *url,
829 uint32_t seq, uint32_t rtptime)
832 if (!rtptime || !url[0])
834 if (rt->transport != RTSP_TRANSPORT_RTP)
836 for (i = 0; i < rt->nb_rtsp_streams; i++) {
837 RTSPStream *rtsp_st = rt->rtsp_streams[i];
838 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
841 if (!strcmp(rtsp_st->control_url, url)) {
842 rtpctx->base_timestamp = rtptime;
848 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
851 char key[20], value[1024], url[1024] = "";
852 uint32_t seq = 0, rtptime = 0;
855 p += strspn(p, SPACE_CHARS);
858 get_word_sep(key, sizeof(key), "=", &p);
862 get_word_sep(value, sizeof(value), ";, ", &p);
864 if (!strcmp(key, "url"))
865 av_strlcpy(url, value, sizeof(url));
866 else if (!strcmp(key, "seq"))
867 seq = strtoul(value, NULL, 10);
868 else if (!strcmp(key, "rtptime"))
869 rtptime = strtoul(value, NULL, 10);
871 handle_rtp_info(rt, url, seq, rtptime);
880 handle_rtp_info(rt, url, seq, rtptime);
883 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
884 RTSPState *rt, const char *method)
888 /* NOTE: we do case independent match for broken servers */
890 if (av_stristart(p, "Session:", &p)) {
892 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
893 if (av_stristart(p, ";timeout=", &p) &&
894 (t = strtol(p, NULL, 10)) > 0) {
897 } else if (av_stristart(p, "Content-Length:", &p)) {
898 reply->content_length = strtol(p, NULL, 10);
899 } else if (av_stristart(p, "Transport:", &p)) {
900 rtsp_parse_transport(reply, p);
901 } else if (av_stristart(p, "CSeq:", &p)) {
902 reply->seq = strtol(p, NULL, 10);
903 } else if (av_stristart(p, "Range:", &p)) {
904 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
905 } else if (av_stristart(p, "RealChallenge1:", &p)) {
906 p += strspn(p, SPACE_CHARS);
907 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
908 } else if (av_stristart(p, "Server:", &p)) {
909 p += strspn(p, SPACE_CHARS);
910 av_strlcpy(reply->server, p, sizeof(reply->server));
911 } else if (av_stristart(p, "Notice:", &p) ||
912 av_stristart(p, "X-Notice:", &p)) {
913 reply->notice = strtol(p, NULL, 10);
914 } else if (av_stristart(p, "Location:", &p)) {
915 p += strspn(p, SPACE_CHARS);
916 av_strlcpy(reply->location, p , sizeof(reply->location));
917 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
918 p += strspn(p, SPACE_CHARS);
919 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
920 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
921 p += strspn(p, SPACE_CHARS);
922 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
923 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
924 p += strspn(p, SPACE_CHARS);
925 if (method && !strcmp(method, "DESCRIBE"))
926 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
927 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
928 p += strspn(p, SPACE_CHARS);
929 if (method && !strcmp(method, "PLAY"))
930 rtsp_parse_rtp_info(rt, p);
931 } else if (av_stristart(p, "Public:", &p) && rt) {
932 if (strstr(p, "GET_PARAMETER") &&
933 method && !strcmp(method, "OPTIONS"))
934 rt->get_parameter_supported = 1;
935 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
936 p += strspn(p, SPACE_CHARS);
937 rt->accept_dynamic_rate = atoi(p);
938 } else if (av_stristart(p, "Content-Type:", &p)) {
939 p += strspn(p, SPACE_CHARS);
940 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
944 /* skip a RTP/TCP interleaved packet */
945 void ff_rtsp_skip_packet(AVFormatContext *s)
947 RTSPState *rt = s->priv_data;
951 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
954 len = AV_RB16(buf + 1);
956 av_dlog(s, "skipping RTP packet len=%d\n", len);
961 if (len1 > sizeof(buf))
963 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
970 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
971 unsigned char **content_ptr,
972 int return_on_interleaved_data, const char *method)
974 RTSPState *rt = s->priv_data;
975 char buf[4096], buf1[1024], *q;
978 int ret, content_length, line_count = 0, request = 0;
979 unsigned char *content = NULL;
985 memset(reply, 0, sizeof(*reply));
987 /* parse reply (XXX: use buffers) */
988 rt->last_reply[0] = '\0';
992 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
993 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
999 /* XXX: only parse it if first char on line ? */
1000 if (return_on_interleaved_data) {
1003 ff_rtsp_skip_packet(s);
1004 } else if (ch != '\r') {
1005 if ((q - buf) < sizeof(buf) - 1)
1011 av_dlog(s, "line='%s'\n", buf);
1013 /* test if last line */
1017 if (line_count == 0) {
1018 /* get reply code */
1019 get_word(buf1, sizeof(buf1), &p);
1020 if (!strncmp(buf1, "RTSP/", 5)) {
1021 get_word(buf1, sizeof(buf1), &p);
1022 reply->status_code = atoi(buf1);
1023 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1025 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1026 get_word(buf1, sizeof(buf1), &p); // object
1030 ff_rtsp_parse_line(reply, p, rt, method);
1031 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1032 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1037 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1038 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1040 content_length = reply->content_length;
1041 if (content_length > 0) {
1042 /* leave some room for a trailing '\0' (useful for simple parsing) */
1043 content = av_malloc(content_length + 1);
1044 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1045 content[content_length] = '\0';
1048 *content_ptr = content;
1054 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1055 const char* ptr = buf;
1057 if (!strcmp(reply->reason, "OPTIONS")) {
1058 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1060 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1061 if (reply->session_id[0])
1062 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1065 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1067 av_strlcat(buf, "\r\n", sizeof(buf));
1069 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1070 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1073 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1075 rt->last_cmd_time = av_gettime();
1076 /* Even if the request from the server had data, it is not the data
1077 * that the caller wants or expects. The memory could also be leaked
1078 * if the actual following reply has content data. */
1080 av_freep(content_ptr);
1081 /* If method is set, this is called from ff_rtsp_send_cmd,
1082 * where a reply to exactly this request is awaited. For
1083 * callers from within packet receiving, we just want to
1084 * return to the caller and go back to receiving packets. */
1090 if (rt->seq != reply->seq) {
1091 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1092 rt->seq, reply->seq);
1096 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1097 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1098 reply->notice == 2306 /* Continuous Feed Terminated */) {
1099 rt->state = RTSP_STATE_IDLE;
1100 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1101 return AVERROR(EIO); /* data or server error */
1102 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1103 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1104 return AVERROR(EPERM);
1110 * Send a command to the RTSP server without waiting for the reply.
1112 * @param s RTSP (de)muxer context
1113 * @param method the method for the request
1114 * @param url the target url for the request
1115 * @param headers extra header lines to include in the request
1116 * @param send_content if non-null, the data to send as request body content
1117 * @param send_content_length the length of the send_content data, or 0 if
1118 * send_content is null
1120 * @return zero if success, nonzero otherwise
1122 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1123 const char *method, const char *url,
1124 const char *headers,
1125 const unsigned char *send_content,
1126 int send_content_length)
1128 RTSPState *rt = s->priv_data;
1129 char buf[4096], *out_buf;
1130 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1132 /* Add in RTSP headers */
1135 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1137 av_strlcat(buf, headers, sizeof(buf));
1138 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1139 if (rt->session_id[0] != '\0' && (!headers ||
1140 !strstr(headers, "\nIf-Match:"))) {
1141 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1144 char *str = ff_http_auth_create_response(&rt->auth_state,
1145 rt->auth, url, method);
1147 av_strlcat(buf, str, sizeof(buf));
1150 if (send_content_length > 0 && send_content)
1151 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1152 av_strlcat(buf, "\r\n", sizeof(buf));
1154 /* base64 encode rtsp if tunneling */
1155 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1156 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1157 out_buf = base64buf;
1160 av_dlog(s, "Sending:\n%s--\n", buf);
1162 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1163 if (send_content_length > 0 && send_content) {
1164 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1165 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1166 "with content data not supported\n");
1167 return AVERROR_PATCHWELCOME;
1169 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1171 rt->last_cmd_time = av_gettime();
1176 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1177 const char *url, const char *headers)
1179 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1182 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1183 const char *headers, RTSPMessageHeader *reply,
1184 unsigned char **content_ptr)
1186 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1187 content_ptr, NULL, 0);
1190 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1191 const char *method, const char *url,
1193 RTSPMessageHeader *reply,
1194 unsigned char **content_ptr,
1195 const unsigned char *send_content,
1196 int send_content_length)
1198 RTSPState *rt = s->priv_data;
1199 HTTPAuthType cur_auth_type;
1200 int ret, attempts = 0;
1203 cur_auth_type = rt->auth_state.auth_type;
1204 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1206 send_content_length)))
1209 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1213 if (reply->status_code == 401 &&
1214 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1215 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1218 if (reply->status_code > 400){
1219 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1223 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1229 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1230 int lower_transport, const char *real_challenge)
1232 RTSPState *rt = s->priv_data;
1233 int rtx = 0, j, i, err, interleave = 0, port_off;
1234 RTSPStream *rtsp_st;
1235 RTSPMessageHeader reply1, *reply = &reply1;
1237 const char *trans_pref;
1239 if (rt->transport == RTSP_TRANSPORT_RDT)
1240 trans_pref = "x-pn-tng";
1241 else if (rt->transport == RTSP_TRANSPORT_RAW)
1242 trans_pref = "RAW/RAW";
1244 trans_pref = "RTP/AVP";
1246 /* default timeout: 1 minute */
1249 /* Choose a random starting offset within the first half of the
1250 * port range, to allow for a number of ports to try even if the offset
1251 * happens to be at the end of the random range. */
1252 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1253 /* even random offset */
1254 port_off -= port_off & 0x01;
1256 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1257 char transport[2048];
1260 * WMS serves all UDP data over a single connection, the RTX, which
1261 * isn't necessarily the first in the SDP but has to be the first
1262 * to be set up, else the second/third SETUP will fail with a 461.
1264 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1265 rt->server_type == RTSP_SERVER_WMS) {
1268 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1269 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1271 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1275 if (rtx == rt->nb_rtsp_streams)
1276 return -1; /* no RTX found */
1277 rtsp_st = rt->rtsp_streams[rtx];
1279 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1281 rtsp_st = rt->rtsp_streams[i];
1284 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1287 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1288 port = reply->transports[0].client_port_min;
1292 /* first try in specified port range */
1293 while (j <= rt->rtp_port_max) {
1294 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1295 "?localport=%d", j);
1296 /* we will use two ports per rtp stream (rtp and rtcp) */
1298 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1299 &s->interrupt_callback, NULL))
1302 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1307 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1309 snprintf(transport, sizeof(transport) - 1,
1310 "%s/UDP;", trans_pref);
1311 if (rt->server_type != RTSP_SERVER_REAL)
1312 av_strlcat(transport, "unicast;", sizeof(transport));
1313 av_strlcatf(transport, sizeof(transport),
1314 "client_port=%d", port);
1315 if (rt->transport == RTSP_TRANSPORT_RTP &&
1316 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1317 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1321 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1322 /* For WMS streams, the application streams are only used for
1323 * UDP. When trying to set it up for TCP streams, the server
1324 * will return an error. Therefore, we skip those streams. */
1325 if (rt->server_type == RTSP_SERVER_WMS &&
1326 (rtsp_st->stream_index < 0 ||
1327 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1330 snprintf(transport, sizeof(transport) - 1,
1331 "%s/TCP;", trans_pref);
1332 if (rt->transport != RTSP_TRANSPORT_RDT)
1333 av_strlcat(transport, "unicast;", sizeof(transport));
1334 av_strlcatf(transport, sizeof(transport),
1335 "interleaved=%d-%d",
1336 interleave, interleave + 1);
1340 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1341 snprintf(transport, sizeof(transport) - 1,
1342 "%s/UDP;multicast", trans_pref);
1345 av_strlcat(transport, ";mode=record", sizeof(transport));
1346 } else if (rt->server_type == RTSP_SERVER_REAL ||
1347 rt->server_type == RTSP_SERVER_WMS)
1348 av_strlcat(transport, ";mode=play", sizeof(transport));
1349 snprintf(cmd, sizeof(cmd),
1350 "Transport: %s\r\n",
1352 if (rt->accept_dynamic_rate)
1353 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1354 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1355 char real_res[41], real_csum[9];
1356 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1358 av_strlcatf(cmd, sizeof(cmd),
1360 "RealChallenge2: %s, sd=%s\r\n",
1361 rt->session_id, real_res, real_csum);
1363 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1364 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1367 } else if (reply->status_code != RTSP_STATUS_OK ||
1368 reply->nb_transports != 1) {
1369 err = AVERROR_INVALIDDATA;
1373 /* XXX: same protocol for all streams is required */
1375 if (reply->transports[0].lower_transport != rt->lower_transport ||
1376 reply->transports[0].transport != rt->transport) {
1377 err = AVERROR_INVALIDDATA;
1381 rt->lower_transport = reply->transports[0].lower_transport;
1382 rt->transport = reply->transports[0].transport;
1385 /* Fail if the server responded with another lower transport mode
1386 * than what we requested. */
1387 if (reply->transports[0].lower_transport != lower_transport) {
1388 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1389 err = AVERROR_INVALIDDATA;
1393 switch(reply->transports[0].lower_transport) {
1394 case RTSP_LOWER_TRANSPORT_TCP:
1395 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1396 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1399 case RTSP_LOWER_TRANSPORT_UDP: {
1400 char url[1024], options[30] = "";
1402 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1403 av_strlcpy(options, "?connect=1", sizeof(options));
1404 /* Use source address if specified */
1405 if (reply->transports[0].source[0]) {
1406 ff_url_join(url, sizeof(url), "rtp", NULL,
1407 reply->transports[0].source,
1408 reply->transports[0].server_port_min, "%s", options);
1410 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1411 reply->transports[0].server_port_min, "%s", options);
1413 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1414 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1415 err = AVERROR_INVALIDDATA;
1418 /* Try to initialize the connection state in a
1419 * potential NAT router by sending dummy packets.
1420 * RTP/RTCP dummy packets are used for RDT, too.
1422 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1424 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1427 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1428 char url[1024], namebuf[50], optbuf[20] = "";
1429 struct sockaddr_storage addr;
1432 if (reply->transports[0].destination.ss_family) {
1433 addr = reply->transports[0].destination;
1434 port = reply->transports[0].port_min;
1435 ttl = reply->transports[0].ttl;
1437 addr = rtsp_st->sdp_ip;
1438 port = rtsp_st->sdp_port;
1439 ttl = rtsp_st->sdp_ttl;
1442 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1443 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1444 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1445 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1446 port, "%s", optbuf);
1447 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1448 &s->interrupt_callback, NULL) < 0) {
1449 err = AVERROR_INVALIDDATA;
1456 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1460 if (rt->nb_rtsp_streams && reply->timeout > 0)
1461 rt->timeout = reply->timeout;
1463 if (rt->server_type == RTSP_SERVER_REAL)
1464 rt->need_subscription = 1;
1469 ff_rtsp_undo_setup(s);
1473 void ff_rtsp_close_connections(AVFormatContext *s)
1475 RTSPState *rt = s->priv_data;
1476 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1477 ffurl_close(rt->rtsp_hd);
1478 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1481 int ff_rtsp_connect(AVFormatContext *s)
1483 RTSPState *rt = s->priv_data;
1484 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1485 int port, err, tcp_fd;
1486 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1487 int lower_transport_mask = 0;
1488 char real_challenge[64] = "";
1489 struct sockaddr_storage peer;
1490 socklen_t peer_len = sizeof(peer);
1492 if (rt->rtp_port_max < rt->rtp_port_min) {
1493 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1494 "than min port %d\n", rt->rtp_port_max,
1496 return AVERROR(EINVAL);
1499 if (!ff_network_init())
1500 return AVERROR(EIO);
1502 if (s->max_delay < 0) /* Not set by the caller */
1503 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1505 rt->control_transport = RTSP_MODE_PLAIN;
1506 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1507 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1508 rt->control_transport = RTSP_MODE_TUNNEL;
1510 /* Only pass through valid flags from here */
1511 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1514 lower_transport_mask = rt->lower_transport_mask;
1515 /* extract hostname and port */
1516 av_url_split(NULL, 0, auth, sizeof(auth),
1517 host, sizeof(host), &port, path, sizeof(path), s->filename);
1519 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1522 port = RTSP_DEFAULT_PORT;
1524 if (!lower_transport_mask)
1525 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1528 /* Only UDP or TCP - UDP multicast isn't supported. */
1529 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1530 (1 << RTSP_LOWER_TRANSPORT_TCP);
1531 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1532 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1533 "only UDP and TCP are supported for output.\n");
1534 err = AVERROR(EINVAL);
1539 /* Construct the URI used in request; this is similar to s->filename,
1540 * but with authentication credentials removed and RTSP specific options
1542 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1543 host, port, "%s", path);
1545 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1546 /* set up initial handshake for tunneling */
1547 char httpname[1024];
1548 char sessioncookie[17];
1551 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1552 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1553 av_get_random_seed(), av_get_random_seed());
1556 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1557 &s->interrupt_callback) < 0) {
1562 /* generate GET headers */
1563 snprintf(headers, sizeof(headers),
1564 "x-sessioncookie: %s\r\n"
1565 "Accept: application/x-rtsp-tunnelled\r\n"
1566 "Pragma: no-cache\r\n"
1567 "Cache-Control: no-cache\r\n",
1569 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1571 /* complete the connection */
1572 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1578 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1579 &s->interrupt_callback) < 0 ) {
1584 /* generate POST headers */
1585 snprintf(headers, sizeof(headers),
1586 "x-sessioncookie: %s\r\n"
1587 "Content-Type: application/x-rtsp-tunnelled\r\n"
1588 "Pragma: no-cache\r\n"
1589 "Cache-Control: no-cache\r\n"
1590 "Content-Length: 32767\r\n"
1591 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1593 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1594 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1596 /* Initialize the authentication state for the POST session. The HTTP
1597 * protocol implementation doesn't properly handle multi-pass
1598 * authentication for POST requests, since it would require one of
1600 * - implementing Expect: 100-continue, which many HTTP servers
1601 * don't support anyway, even less the RTSP servers that do HTTP
1603 * - sending the whole POST data until getting a 401 reply specifying
1604 * what authentication method to use, then resending all that data
1605 * - waiting for potential 401 replies directly after sending the
1606 * POST header (waiting for some unspecified time)
1607 * Therefore, we copy the full auth state, which works for both basic
1608 * and digest. (For digest, we would have to synchronize the nonce
1609 * count variable between the two sessions, if we'd do more requests
1610 * with the original session, though.)
1612 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1614 /* complete the connection */
1615 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1620 /* open the tcp connection */
1621 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1622 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1623 &s->interrupt_callback, NULL) < 0) {
1627 rt->rtsp_hd_out = rt->rtsp_hd;
1631 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1632 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1633 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1634 NULL, 0, NI_NUMERICHOST);
1637 /* request options supported by the server; this also detects server
1639 for (rt->server_type = RTSP_SERVER_RTP;;) {
1641 if (rt->server_type == RTSP_SERVER_REAL)
1644 * The following entries are required for proper
1645 * streaming from a Realmedia server. They are
1646 * interdependent in some way although we currently
1647 * don't quite understand how. Values were copied
1648 * from mplayer SVN r23589.
1649 * ClientChallenge is a 16-byte ID in hex
1650 * CompanyID is a 16-byte ID in base64
1652 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1653 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1654 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1655 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1657 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1658 if (reply->status_code != RTSP_STATUS_OK) {
1659 err = AVERROR_INVALIDDATA;
1663 /* detect server type if not standard-compliant RTP */
1664 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1665 rt->server_type = RTSP_SERVER_REAL;
1667 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1668 rt->server_type = RTSP_SERVER_WMS;
1669 } else if (rt->server_type == RTSP_SERVER_REAL)
1670 strcpy(real_challenge, reply->real_challenge);
1674 if (s->iformat && CONFIG_RTSP_DEMUXER)
1675 err = ff_rtsp_setup_input_streams(s, reply);
1676 else if (CONFIG_RTSP_MUXER)
1677 err = ff_rtsp_setup_output_streams(s, host);
1682 int lower_transport = ff_log2_tab[lower_transport_mask &
1683 ~(lower_transport_mask - 1)];
1685 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1686 rt->server_type == RTSP_SERVER_REAL ?
1687 real_challenge : NULL);
1690 lower_transport_mask &= ~(1 << lower_transport);
1691 if (lower_transport_mask == 0 && err == 1) {
1692 err = AVERROR(EPROTONOSUPPORT);
1697 rt->lower_transport_mask = lower_transport_mask;
1698 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1699 rt->state = RTSP_STATE_IDLE;
1700 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1703 ff_rtsp_close_streams(s);
1704 ff_rtsp_close_connections(s);
1705 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1706 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1707 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1715 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1718 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1719 uint8_t *buf, int buf_size, int64_t wait_end)
1721 RTSPState *rt = s->priv_data;
1722 RTSPStream *rtsp_st;
1723 int n, i, ret, tcp_fd, timeout_cnt = 0;
1725 struct pollfd *p = rt->p;
1726 int *fds = NULL, fdsnum, fdsidx;
1729 if (ff_check_interrupt(&s->interrupt_callback))
1730 return AVERROR_EXIT;
1731 if (wait_end && wait_end - av_gettime() < 0)
1732 return AVERROR(EAGAIN);
1735 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1736 p[max_p].fd = tcp_fd;
1737 p[max_p++].events = POLLIN;
1741 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1742 rtsp_st = rt->rtsp_streams[i];
1743 if (rtsp_st->rtp_handle) {
1744 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1746 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1750 av_log(s, AV_LOG_ERROR,
1751 "Number of fds %d not supported\n", fdsnum);
1752 return AVERROR_INVALIDDATA;
1754 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1755 p[max_p].fd = fds[fdsidx];
1756 p[max_p++].events = POLLIN;
1761 n = poll(p, max_p, POLL_TIMEOUT_MS);
1763 int j = 1 - (tcp_fd == -1);
1765 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1766 rtsp_st = rt->rtsp_streams[i];
1767 if (rtsp_st->rtp_handle) {
1768 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1769 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1771 *prtsp_st = rtsp_st;
1778 #if CONFIG_RTSP_DEMUXER
1779 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1780 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1781 if (rt->state == RTSP_STATE_STREAMING) {
1782 if (!ff_rtsp_parse_streaming_commands(s))
1785 av_log(s, AV_LOG_WARNING,
1786 "Unable to answer to TEARDOWN\n");
1790 RTSPMessageHeader reply;
1791 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1794 /* XXX: parse message */
1795 if (rt->state != RTSP_STATE_STREAMING)
1800 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1801 return AVERROR(ETIMEDOUT);
1802 } else if (n < 0 && errno != EINTR)
1803 return AVERROR(errno);
1807 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1808 const uint8_t *buf, int len)
1810 RTSPState *rt = s->priv_data;
1814 if (rt->nb_rtsp_streams == 1) {
1815 *rtsp_st = rt->rtsp_streams[0];
1818 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1819 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1821 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1822 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1825 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1826 *rtsp_st = rt->rtsp_streams[i];
1833 av_log(s, AV_LOG_WARNING,
1834 "Unable to pick stream for packet - SSRC not known for "
1836 return AVERROR(EAGAIN);
1839 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1840 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1841 *rtsp_st = rt->rtsp_streams[i];
1847 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1848 return AVERROR(EAGAIN);
1851 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1853 RTSPState *rt = s->priv_data;
1855 RTSPStream *rtsp_st, *first_queue_st = NULL;
1856 int64_t wait_end = 0;
1858 if (rt->nb_byes == rt->nb_rtsp_streams)
1861 /* get next frames from the same RTP packet */
1862 if (rt->cur_transport_priv) {
1863 if (rt->transport == RTSP_TRANSPORT_RDT) {
1864 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1865 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1866 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1867 } else if (rt->ts && CONFIG_RTPDEC) {
1868 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1870 rt->recvbuf_pos += ret;
1871 ret = rt->recvbuf_pos < rt->recvbuf_len;
1876 rt->cur_transport_priv = NULL;
1878 } else if (ret == 1) {
1881 rt->cur_transport_priv = NULL;
1885 if (rt->transport == RTSP_TRANSPORT_RTP) {
1887 int64_t first_queue_time = 0;
1888 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1889 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1893 queue_time = ff_rtp_queued_packet_time(rtpctx);
1894 if (queue_time && (queue_time - first_queue_time < 0 ||
1895 !first_queue_time)) {
1896 first_queue_time = queue_time;
1897 first_queue_st = rt->rtsp_streams[i];
1900 if (first_queue_time) {
1901 wait_end = first_queue_time + s->max_delay;
1904 first_queue_st = NULL;
1908 /* read next RTP packet */
1910 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1912 return AVERROR(ENOMEM);
1915 switch(rt->lower_transport) {
1917 #if CONFIG_RTSP_DEMUXER
1918 case RTSP_LOWER_TRANSPORT_TCP:
1919 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1922 case RTSP_LOWER_TRANSPORT_UDP:
1923 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1924 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1925 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1926 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1928 case RTSP_LOWER_TRANSPORT_CUSTOM:
1929 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1930 wait_end && wait_end < av_gettime())
1931 len = AVERROR(EAGAIN);
1933 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1934 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
1935 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1936 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);
1939 if (len == AVERROR(EAGAIN) && first_queue_st &&
1940 rt->transport == RTSP_TRANSPORT_RTP) {
1941 rtsp_st = first_queue_st;
1942 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1949 if (rt->transport == RTSP_TRANSPORT_RDT) {
1950 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1951 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1952 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1953 if (rtsp_st->feedback) {
1954 AVIOContext *pb = NULL;
1955 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1957 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1960 /* Either bad packet, or a RTCP packet. Check if the
1961 * first_rtcp_ntp_time field was initialized. */
1962 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1963 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1964 /* first_rtcp_ntp_time has been initialized for this stream,
1965 * copy the same value to all other uninitialized streams,
1966 * in order to map their timestamp origin to the same ntp time
1969 AVStream *st = NULL;
1970 if (rtsp_st->stream_index >= 0)
1971 st = s->streams[rtsp_st->stream_index];
1972 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1973 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1974 AVStream *st2 = NULL;
1975 if (rt->rtsp_streams[i]->stream_index >= 0)
1976 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1977 if (rtpctx2 && st && st2 &&
1978 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1979 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1980 rtpctx2->rtcp_ts_offset = av_rescale_q(
1981 rtpctx->rtcp_ts_offset, st->time_base,
1986 if (ret == -RTCP_BYE) {
1989 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1990 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1992 if (rt->nb_byes == rt->nb_rtsp_streams)
1996 } else if (rt->ts && CONFIG_RTPDEC) {
1997 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
2000 rt->recvbuf_len = len;
2001 rt->recvbuf_pos = ret;
2002 rt->cur_transport_priv = rt->ts;
2009 return AVERROR_INVALIDDATA;
2015 /* more packets may follow, so we save the RTP context */
2016 rt->cur_transport_priv = rtsp_st->transport_priv;
2020 #endif /* CONFIG_RTPDEC */
2022 #if CONFIG_SDP_DEMUXER
2023 static int sdp_probe(AVProbeData *p1)
2025 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2027 /* we look for a line beginning "c=IN IP" */
2028 while (p < p_end && *p != '\0') {
2029 if (p + sizeof("c=IN IP") - 1 < p_end &&
2030 av_strstart(p, "c=IN IP", NULL))
2031 return AVPROBE_SCORE_MAX / 2;
2033 while (p < p_end - 1 && *p != '\n') p++;
2042 static int sdp_read_header(AVFormatContext *s)
2044 RTSPState *rt = s->priv_data;
2045 RTSPStream *rtsp_st;
2050 if (!ff_network_init())
2051 return AVERROR(EIO);
2053 if (s->max_delay < 0) /* Not set by the caller */
2054 s->max_delay = DEFAULT_REORDERING_DELAY;
2055 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2056 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2058 /* read the whole sdp file */
2059 /* XXX: better loading */
2060 content = av_malloc(SDP_MAX_SIZE);
2061 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2064 return AVERROR_INVALIDDATA;
2066 content[size] ='\0';
2068 err = ff_sdp_parse(s, content);
2072 /* open each RTP stream */
2073 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2075 rtsp_st = rt->rtsp_streams[i];
2077 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2078 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2079 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2080 ff_url_join(url, sizeof(url), "rtp", NULL,
2081 namebuf, rtsp_st->sdp_port,
2082 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2084 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2085 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2086 &s->interrupt_callback, NULL) < 0) {
2087 err = AVERROR_INVALIDDATA;
2091 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2096 ff_rtsp_close_streams(s);
2101 static int sdp_read_close(AVFormatContext *s)
2103 ff_rtsp_close_streams(s);
2108 static const AVClass sdp_demuxer_class = {
2109 .class_name = "SDP demuxer",
2110 .item_name = av_default_item_name,
2111 .option = sdp_options,
2112 .version = LIBAVUTIL_VERSION_INT,
2115 AVInputFormat ff_sdp_demuxer = {
2117 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2118 .priv_data_size = sizeof(RTSPState),
2119 .read_probe = sdp_probe,
2120 .read_header = sdp_read_header,
2121 .read_packet = ff_rtsp_fetch_packet,
2122 .read_close = sdp_read_close,
2123 .priv_class = &sdp_demuxer_class,
2125 #endif /* CONFIG_SDP_DEMUXER */
2127 #if CONFIG_RTP_DEMUXER
2128 static int rtp_probe(AVProbeData *p)
2130 if (av_strstart(p->filename, "rtp:", NULL))
2131 return AVPROBE_SCORE_MAX;
2135 static int rtp_read_header(AVFormatContext *s)
2137 uint8_t recvbuf[1500];
2138 char host[500], sdp[500];
2140 URLContext* in = NULL;
2142 AVCodecContext codec = { 0 };
2143 struct sockaddr_storage addr;
2145 socklen_t addrlen = sizeof(addr);
2146 RTSPState *rt = s->priv_data;
2148 if (!ff_network_init())
2149 return AVERROR(EIO);
2151 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2152 &s->interrupt_callback, NULL);
2157 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2158 if (ret == AVERROR(EAGAIN))
2163 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2167 if ((recvbuf[0] & 0xc0) != 0x80) {
2168 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2173 if (RTP_PT_IS_RTCP(recvbuf[1]))
2176 payload_type = recvbuf[1] & 0x7f;
2179 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2183 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2184 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2185 "without an SDP file describing it\n",
2189 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2190 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2191 "properly you need an SDP file "
2195 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2196 NULL, 0, s->filename);
2198 snprintf(sdp, sizeof(sdp),
2199 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2200 addr.ss_family == AF_INET ? 4 : 6, host,
2201 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2202 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2203 port, payload_type);
2204 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2206 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2209 /* sdp_read_header initializes this again */
2212 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2214 ret = sdp_read_header(s);
2225 static const AVClass rtp_demuxer_class = {
2226 .class_name = "RTP demuxer",
2227 .item_name = av_default_item_name,
2228 .option = rtp_options,
2229 .version = LIBAVUTIL_VERSION_INT,
2232 AVInputFormat ff_rtp_demuxer = {
2234 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2235 .priv_data_size = sizeof(RTSPState),
2236 .read_probe = rtp_probe,
2237 .read_header = rtp_read_header,
2238 .read_packet = ff_rtsp_fetch_packet,
2239 .read_close = sdp_read_close,
2240 .flags = AVFMT_NOFILE,
2241 .priv_class = &rtp_demuxer_class,
2243 #endif /* CONFIG_RTP_DEMUXER */