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)
184 codec->codec_id = handler->codec_id;
185 rtsp_st->dynamic_handler = handler;
186 if (handler->alloc) {
187 rtsp_st->dynamic_protocol_context = handler->alloc();
188 if (!rtsp_st->dynamic_protocol_context)
189 rtsp_st->dynamic_handler = NULL;
193 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
194 static int sdp_parse_rtpmap(AVFormatContext *s,
195 AVStream *st, RTSPStream *rtsp_st,
196 int payload_type, const char *p)
198 AVCodecContext *codec = st->codec;
204 /* See if we can handle this kind of payload.
205 * The space should normally not be there but some Real streams or
206 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
207 * have a trailing space. */
208 get_word_sep(buf, sizeof(buf), "/ ", &p);
209 if (payload_type < RTP_PT_PRIVATE) {
210 /* We are in a standard case
211 * (from http://www.iana.org/assignments/rtp-parameters). */
212 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
215 if (codec->codec_id == AV_CODEC_ID_NONE) {
216 RTPDynamicProtocolHandler *handler =
217 ff_rtp_handler_find_by_name(buf, codec->codec_type);
218 init_rtp_handler(handler, rtsp_st, codec);
219 /* If no dynamic handler was found, check with the list of standard
220 * allocated types, if such a stream for some reason happens to
221 * use a private payload type. This isn't handled in rtpdec.c, since
222 * the format name from the rtpmap line never is passed into rtpdec. */
223 if (!rtsp_st->dynamic_handler)
224 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
227 c = avcodec_find_decoder(codec->codec_id);
233 get_word_sep(buf, sizeof(buf), "/", &p);
235 switch (codec->codec_type) {
236 case AVMEDIA_TYPE_AUDIO:
237 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
238 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
239 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
241 codec->sample_rate = i;
242 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
243 get_word_sep(buf, sizeof(buf), "/", &p);
248 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
250 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
253 case AVMEDIA_TYPE_VIDEO:
254 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
256 avpriv_set_pts_info(st, 32, 1, i);
261 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
262 rtsp_st->dynamic_handler->init(s, st->index,
263 rtsp_st->dynamic_protocol_context);
267 /* parse the attribute line from the fmtp a line of an sdp response. This
268 * is broken out as a function because it is used in rtp_h264.c, which is
270 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
271 char *value, int value_size)
273 *p += strspn(*p, SPACE_CHARS);
275 get_word_sep(attr, attr_size, "=", p);
278 get_word_sep(value, value_size, ";", p);
286 typedef struct SDPParseState {
288 struct sockaddr_storage default_ip;
290 int skip_media; ///< set if an unknown m= line occurs
293 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
294 int letter, const char *buf)
296 RTSPState *rt = s->priv_data;
297 char buf1[64], st_type[64];
299 enum AVMediaType codec_type;
303 struct sockaddr_storage sdp_ip;
306 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
309 if (s1->skip_media && letter != 'm')
313 get_word(buf1, sizeof(buf1), &p);
314 if (strcmp(buf1, "IN") != 0)
316 get_word(buf1, sizeof(buf1), &p);
317 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
319 get_word_sep(buf1, sizeof(buf1), "/", &p);
320 if (get_sockaddr(buf1, &sdp_ip))
325 get_word_sep(buf1, sizeof(buf1), "/", &p);
328 if (s->nb_streams == 0) {
329 s1->default_ip = sdp_ip;
330 s1->default_ttl = ttl;
332 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
333 rtsp_st->sdp_ip = sdp_ip;
334 rtsp_st->sdp_ttl = ttl;
338 av_dict_set(&s->metadata, "title", p, 0);
341 if (s->nb_streams == 0) {
342 av_dict_set(&s->metadata, "comment", p, 0);
349 codec_type = AVMEDIA_TYPE_UNKNOWN;
350 get_word(st_type, sizeof(st_type), &p);
351 if (!strcmp(st_type, "audio")) {
352 codec_type = AVMEDIA_TYPE_AUDIO;
353 } else if (!strcmp(st_type, "video")) {
354 codec_type = AVMEDIA_TYPE_VIDEO;
355 } else if (!strcmp(st_type, "application")) {
356 codec_type = AVMEDIA_TYPE_DATA;
358 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
362 rtsp_st = av_mallocz(sizeof(RTSPStream));
365 rtsp_st->stream_index = -1;
366 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
368 rtsp_st->sdp_ip = s1->default_ip;
369 rtsp_st->sdp_ttl = s1->default_ttl;
371 get_word(buf1, sizeof(buf1), &p); /* port */
372 rtsp_st->sdp_port = atoi(buf1);
374 get_word(buf1, sizeof(buf1), &p); /* protocol */
375 if (!strcmp(buf1, "udp"))
376 rt->transport = RTSP_TRANSPORT_RAW;
377 else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
378 rtsp_st->feedback = 1;
380 /* XXX: handle list of formats */
381 get_word(buf1, sizeof(buf1), &p); /* format list */
382 rtsp_st->sdp_payload_type = atoi(buf1);
384 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
385 /* no corresponding stream */
386 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
387 rt->ts = ff_mpegts_parse_open(s);
388 } else if (rt->server_type == RTSP_SERVER_WMS &&
389 codec_type == AVMEDIA_TYPE_DATA) {
390 /* RTX stream, a stream that carries all the other actual
391 * audio/video streams. Don't expose this to the callers. */
393 st = avformat_new_stream(s, NULL);
396 st->id = rt->nb_rtsp_streams - 1;
397 rtsp_st->stream_index = st->index;
398 st->codec->codec_type = codec_type;
399 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
400 RTPDynamicProtocolHandler *handler;
401 /* if standard payload type, we can find the codec right now */
402 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
403 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
404 st->codec->sample_rate > 0)
405 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
406 /* Even static payload types may need a custom depacketizer */
407 handler = ff_rtp_handler_find_by_id(
408 rtsp_st->sdp_payload_type, st->codec->codec_type);
409 init_rtp_handler(handler, rtsp_st, st->codec);
410 if (handler && handler->init)
411 handler->init(s, st->index,
412 rtsp_st->dynamic_protocol_context);
415 /* put a default control url */
416 av_strlcpy(rtsp_st->control_url, rt->control_uri,
417 sizeof(rtsp_st->control_url));
420 if (av_strstart(p, "control:", &p)) {
421 if (s->nb_streams == 0) {
422 if (!strncmp(p, "rtsp://", 7))
423 av_strlcpy(rt->control_uri, p,
424 sizeof(rt->control_uri));
427 /* get the control url */
428 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
430 /* XXX: may need to add full url resolution */
431 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
433 if (proto[0] == '\0') {
434 /* relative control URL */
435 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
436 av_strlcat(rtsp_st->control_url, "/",
437 sizeof(rtsp_st->control_url));
438 av_strlcat(rtsp_st->control_url, p,
439 sizeof(rtsp_st->control_url));
441 av_strlcpy(rtsp_st->control_url, p,
442 sizeof(rtsp_st->control_url));
444 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
445 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
446 get_word(buf1, sizeof(buf1), &p);
447 payload_type = atoi(buf1);
448 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
449 if (rtsp_st->stream_index >= 0) {
450 st = s->streams[rtsp_st->stream_index];
451 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
453 } else if (av_strstart(p, "fmtp:", &p) ||
454 av_strstart(p, "framesize:", &p)) {
455 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
456 // let dynamic protocol handlers have a stab at the line.
457 get_word(buf1, sizeof(buf1), &p);
458 payload_type = atoi(buf1);
459 for (i = 0; i < rt->nb_rtsp_streams; i++) {
460 rtsp_st = rt->rtsp_streams[i];
461 if (rtsp_st->sdp_payload_type == payload_type &&
462 rtsp_st->dynamic_handler &&
463 rtsp_st->dynamic_handler->parse_sdp_a_line)
464 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
465 rtsp_st->dynamic_protocol_context, buf);
467 } else if (av_strstart(p, "range:", &p)) {
470 // this is so that seeking on a streamed file can work.
471 rtsp_parse_range_npt(p, &start, &end);
472 s->start_time = start;
473 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
474 s->duration = (end == AV_NOPTS_VALUE) ?
475 AV_NOPTS_VALUE : end - start;
476 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
478 rt->transport = RTSP_TRANSPORT_RDT;
479 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
481 st = s->streams[s->nb_streams - 1];
482 st->codec->sample_rate = atoi(p);
483 } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
485 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
486 get_word(buf1, sizeof(buf1), &p); // ignore tag
487 get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
488 p += strspn(p, SPACE_CHARS);
489 if (av_strstart(p, "inline:", &p))
490 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);
492 if (rt->server_type == RTSP_SERVER_WMS)
493 ff_wms_parse_sdp_a_line(s, p);
494 if (s->nb_streams > 0) {
495 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
497 if (rt->server_type == RTSP_SERVER_REAL)
498 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
500 if (rtsp_st->dynamic_handler &&
501 rtsp_st->dynamic_handler->parse_sdp_a_line)
502 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
503 rtsp_st->stream_index,
504 rtsp_st->dynamic_protocol_context, buf);
511 int ff_sdp_parse(AVFormatContext *s, const char *content)
513 RTSPState *rt = s->priv_data;
516 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
517 * contain long SDP lines containing complete ASF Headers (several
518 * kB) or arrays of MDPR (RM stream descriptor) headers plus
519 * "rulebooks" describing their properties. Therefore, the SDP line
522 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
523 * in rtpdec_xiph.c. */
525 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
529 p += strspn(p, SPACE_CHARS);
537 /* get the content */
539 while (*p != '\n' && *p != '\r' && *p != '\0') {
540 if ((q - buf) < sizeof(buf) - 1)
545 sdp_parse_line(s, s1, letter, buf);
547 while (*p != '\n' && *p != '\0')
552 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
553 if (!rt->p) return AVERROR(ENOMEM);
556 #endif /* CONFIG_RTPDEC */
558 void ff_rtsp_undo_setup(AVFormatContext *s)
560 RTSPState *rt = s->priv_data;
563 for (i = 0; i < rt->nb_rtsp_streams; i++) {
564 RTSPStream *rtsp_st = rt->rtsp_streams[i];
567 if (rtsp_st->transport_priv) {
569 AVFormatContext *rtpctx = rtsp_st->transport_priv;
570 av_write_trailer(rtpctx);
571 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
573 avio_close_dyn_buf(rtpctx->pb, &ptr);
576 avio_close(rtpctx->pb);
578 avformat_free_context(rtpctx);
579 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
580 ff_rdt_parse_close(rtsp_st->transport_priv);
581 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
582 ff_rtp_parse_close(rtsp_st->transport_priv);
584 rtsp_st->transport_priv = NULL;
585 if (rtsp_st->rtp_handle)
586 ffurl_close(rtsp_st->rtp_handle);
587 rtsp_st->rtp_handle = NULL;
591 /* close and free RTSP streams */
592 void ff_rtsp_close_streams(AVFormatContext *s)
594 RTSPState *rt = s->priv_data;
598 ff_rtsp_undo_setup(s);
599 for (i = 0; i < rt->nb_rtsp_streams; i++) {
600 rtsp_st = rt->rtsp_streams[i];
602 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
603 rtsp_st->dynamic_handler->free(
604 rtsp_st->dynamic_protocol_context);
608 av_free(rt->rtsp_streams);
610 avformat_close_input(&rt->asf_ctx);
612 if (rt->ts && CONFIG_RTPDEC)
613 ff_mpegts_parse_close(rt->ts);
615 av_free(rt->recvbuf);
618 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
620 RTSPState *rt = s->priv_data;
622 int reordering_queue_size = rt->reordering_queue_size;
623 if (reordering_queue_size < 0) {
624 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
625 reordering_queue_size = 0;
627 reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
630 /* open the RTP context */
631 if (rtsp_st->stream_index >= 0)
632 st = s->streams[rtsp_st->stream_index];
634 s->ctx_flags |= AVFMTCTX_NOHEADER;
636 if (s->oformat && CONFIG_RTSP_MUXER) {
637 int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
639 RTSP_TCP_MAX_PACKET_SIZE,
640 rtsp_st->stream_index);
641 /* Ownership of rtp_handle is passed to the rtp mux context */
642 rtsp_st->rtp_handle = NULL;
645 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
646 return 0; // Don't need to open any parser here
647 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
648 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
649 rtsp_st->dynamic_protocol_context,
650 rtsp_st->dynamic_handler);
651 else if (CONFIG_RTPDEC)
652 rtsp_st->transport_priv = ff_rtp_parse_open(s, st,
653 rtsp_st->sdp_payload_type,
654 reordering_queue_size);
656 if (!rtsp_st->transport_priv) {
657 return AVERROR(ENOMEM);
658 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
659 if (rtsp_st->dynamic_handler) {
660 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
661 rtsp_st->dynamic_protocol_context,
662 rtsp_st->dynamic_handler);
664 if (rtsp_st->crypto_suite[0])
665 ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
666 rtsp_st->crypto_suite,
667 rtsp_st->crypto_params);
673 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
674 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
681 q += strspn(q, SPACE_CHARS);
682 v = strtol(q, &p, 10);
686 v = strtol(p, &p, 10);
695 /* XXX: only one transport specification is parsed */
696 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
698 char transport_protocol[16];
700 char lower_transport[16];
702 RTSPTransportField *th;
705 reply->nb_transports = 0;
708 p += strspn(p, SPACE_CHARS);
712 th = &reply->transports[reply->nb_transports];
714 get_word_sep(transport_protocol, sizeof(transport_protocol),
716 if (!av_strcasecmp (transport_protocol, "rtp")) {
717 get_word_sep(profile, sizeof(profile), "/;,", &p);
718 lower_transport[0] = '\0';
719 /* rtp/avp/<protocol> */
721 get_word_sep(lower_transport, sizeof(lower_transport),
724 th->transport = RTSP_TRANSPORT_RTP;
725 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
726 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
727 /* x-pn-tng/<protocol> */
728 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
730 th->transport = RTSP_TRANSPORT_RDT;
731 } else if (!av_strcasecmp(transport_protocol, "raw")) {
732 get_word_sep(profile, sizeof(profile), "/;,", &p);
733 lower_transport[0] = '\0';
734 /* raw/raw/<protocol> */
736 get_word_sep(lower_transport, sizeof(lower_transport),
739 th->transport = RTSP_TRANSPORT_RAW;
741 if (!av_strcasecmp(lower_transport, "TCP"))
742 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
744 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
748 /* get each parameter */
749 while (*p != '\0' && *p != ',') {
750 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
751 if (!strcmp(parameter, "port")) {
754 rtsp_parse_range(&th->port_min, &th->port_max, &p);
756 } else if (!strcmp(parameter, "client_port")) {
759 rtsp_parse_range(&th->client_port_min,
760 &th->client_port_max, &p);
762 } else if (!strcmp(parameter, "server_port")) {
765 rtsp_parse_range(&th->server_port_min,
766 &th->server_port_max, &p);
768 } else if (!strcmp(parameter, "interleaved")) {
771 rtsp_parse_range(&th->interleaved_min,
772 &th->interleaved_max, &p);
774 } else if (!strcmp(parameter, "multicast")) {
775 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
776 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
777 } else if (!strcmp(parameter, "ttl")) {
781 th->ttl = strtol(p, &end, 10);
784 } else if (!strcmp(parameter, "destination")) {
787 get_word_sep(buf, sizeof(buf), ";,", &p);
788 get_sockaddr(buf, &th->destination);
790 } else if (!strcmp(parameter, "source")) {
793 get_word_sep(buf, sizeof(buf), ";,", &p);
794 av_strlcpy(th->source, buf, sizeof(th->source));
796 } else if (!strcmp(parameter, "mode")) {
799 get_word_sep(buf, sizeof(buf), ";, ", &p);
800 if (!strcmp(buf, "record") ||
801 !strcmp(buf, "receive"))
806 while (*p != ';' && *p != '\0' && *p != ',')
814 reply->nb_transports++;
818 static void handle_rtp_info(RTSPState *rt, const char *url,
819 uint32_t seq, uint32_t rtptime)
822 if (!rtptime || !url[0])
824 if (rt->transport != RTSP_TRANSPORT_RTP)
826 for (i = 0; i < rt->nb_rtsp_streams; i++) {
827 RTSPStream *rtsp_st = rt->rtsp_streams[i];
828 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
831 if (!strcmp(rtsp_st->control_url, url)) {
832 rtpctx->base_timestamp = rtptime;
838 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
841 char key[20], value[1024], url[1024] = "";
842 uint32_t seq = 0, rtptime = 0;
845 p += strspn(p, SPACE_CHARS);
848 get_word_sep(key, sizeof(key), "=", &p);
852 get_word_sep(value, sizeof(value), ";, ", &p);
854 if (!strcmp(key, "url"))
855 av_strlcpy(url, value, sizeof(url));
856 else if (!strcmp(key, "seq"))
857 seq = strtoul(value, NULL, 10);
858 else if (!strcmp(key, "rtptime"))
859 rtptime = strtoul(value, NULL, 10);
861 handle_rtp_info(rt, url, seq, rtptime);
870 handle_rtp_info(rt, url, seq, rtptime);
873 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
874 RTSPState *rt, const char *method)
878 /* NOTE: we do case independent match for broken servers */
880 if (av_stristart(p, "Session:", &p)) {
882 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
883 if (av_stristart(p, ";timeout=", &p) &&
884 (t = strtol(p, NULL, 10)) > 0) {
887 } else if (av_stristart(p, "Content-Length:", &p)) {
888 reply->content_length = strtol(p, NULL, 10);
889 } else if (av_stristart(p, "Transport:", &p)) {
890 rtsp_parse_transport(reply, p);
891 } else if (av_stristart(p, "CSeq:", &p)) {
892 reply->seq = strtol(p, NULL, 10);
893 } else if (av_stristart(p, "Range:", &p)) {
894 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
895 } else if (av_stristart(p, "RealChallenge1:", &p)) {
896 p += strspn(p, SPACE_CHARS);
897 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
898 } else if (av_stristart(p, "Server:", &p)) {
899 p += strspn(p, SPACE_CHARS);
900 av_strlcpy(reply->server, p, sizeof(reply->server));
901 } else if (av_stristart(p, "Notice:", &p) ||
902 av_stristart(p, "X-Notice:", &p)) {
903 reply->notice = strtol(p, NULL, 10);
904 } else if (av_stristart(p, "Location:", &p)) {
905 p += strspn(p, SPACE_CHARS);
906 av_strlcpy(reply->location, p , sizeof(reply->location));
907 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
908 p += strspn(p, SPACE_CHARS);
909 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
910 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
911 p += strspn(p, SPACE_CHARS);
912 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
913 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
914 p += strspn(p, SPACE_CHARS);
915 if (method && !strcmp(method, "DESCRIBE"))
916 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
917 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
918 p += strspn(p, SPACE_CHARS);
919 if (method && !strcmp(method, "PLAY"))
920 rtsp_parse_rtp_info(rt, p);
921 } else if (av_stristart(p, "Public:", &p) && rt) {
922 if (strstr(p, "GET_PARAMETER") &&
923 method && !strcmp(method, "OPTIONS"))
924 rt->get_parameter_supported = 1;
925 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
926 p += strspn(p, SPACE_CHARS);
927 rt->accept_dynamic_rate = atoi(p);
928 } else if (av_stristart(p, "Content-Type:", &p)) {
929 p += strspn(p, SPACE_CHARS);
930 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
934 /* skip a RTP/TCP interleaved packet */
935 void ff_rtsp_skip_packet(AVFormatContext *s)
937 RTSPState *rt = s->priv_data;
941 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
944 len = AV_RB16(buf + 1);
946 av_dlog(s, "skipping RTP packet len=%d\n", len);
951 if (len1 > sizeof(buf))
953 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
960 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
961 unsigned char **content_ptr,
962 int return_on_interleaved_data, const char *method)
964 RTSPState *rt = s->priv_data;
965 char buf[4096], buf1[1024], *q;
968 int ret, content_length, line_count = 0, request = 0;
969 unsigned char *content = NULL;
975 memset(reply, 0, sizeof(*reply));
977 /* parse reply (XXX: use buffers) */
978 rt->last_reply[0] = '\0';
982 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
983 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
989 /* XXX: only parse it if first char on line ? */
990 if (return_on_interleaved_data) {
993 ff_rtsp_skip_packet(s);
994 } else if (ch != '\r') {
995 if ((q - buf) < sizeof(buf) - 1)
1001 av_dlog(s, "line='%s'\n", buf);
1003 /* test if last line */
1007 if (line_count == 0) {
1008 /* get reply code */
1009 get_word(buf1, sizeof(buf1), &p);
1010 if (!strncmp(buf1, "RTSP/", 5)) {
1011 get_word(buf1, sizeof(buf1), &p);
1012 reply->status_code = atoi(buf1);
1013 av_strlcpy(reply->reason, p, sizeof(reply->reason));
1015 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
1016 get_word(buf1, sizeof(buf1), &p); // object
1020 ff_rtsp_parse_line(reply, p, rt, method);
1021 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
1022 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1027 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1028 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1030 content_length = reply->content_length;
1031 if (content_length > 0) {
1032 /* leave some room for a trailing '\0' (useful for simple parsing) */
1033 content = av_malloc(content_length + 1);
1034 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1035 content[content_length] = '\0';
1038 *content_ptr = content;
1044 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1045 const char* ptr = buf;
1047 if (!strcmp(reply->reason, "OPTIONS")) {
1048 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1050 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1051 if (reply->session_id[0])
1052 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1055 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1057 av_strlcat(buf, "\r\n", sizeof(buf));
1059 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1060 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1063 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1065 rt->last_cmd_time = av_gettime();
1066 /* Even if the request from the server had data, it is not the data
1067 * that the caller wants or expects. The memory could also be leaked
1068 * if the actual following reply has content data. */
1070 av_freep(content_ptr);
1071 /* If method is set, this is called from ff_rtsp_send_cmd,
1072 * where a reply to exactly this request is awaited. For
1073 * callers from within packet receiving, we just want to
1074 * return to the caller and go back to receiving packets. */
1080 if (rt->seq != reply->seq) {
1081 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1082 rt->seq, reply->seq);
1086 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1087 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1088 reply->notice == 2306 /* Continuous Feed Terminated */) {
1089 rt->state = RTSP_STATE_IDLE;
1090 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1091 return AVERROR(EIO); /* data or server error */
1092 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1093 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1094 return AVERROR(EPERM);
1100 * Send a command to the RTSP server without waiting for the reply.
1102 * @param s RTSP (de)muxer context
1103 * @param method the method for the request
1104 * @param url the target url for the request
1105 * @param headers extra header lines to include in the request
1106 * @param send_content if non-null, the data to send as request body content
1107 * @param send_content_length the length of the send_content data, or 0 if
1108 * send_content is null
1110 * @return zero if success, nonzero otherwise
1112 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1113 const char *method, const char *url,
1114 const char *headers,
1115 const unsigned char *send_content,
1116 int send_content_length)
1118 RTSPState *rt = s->priv_data;
1119 char buf[4096], *out_buf;
1120 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1122 /* Add in RTSP headers */
1125 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1127 av_strlcat(buf, headers, sizeof(buf));
1128 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1129 if (rt->session_id[0] != '\0' && (!headers ||
1130 !strstr(headers, "\nIf-Match:"))) {
1131 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1134 char *str = ff_http_auth_create_response(&rt->auth_state,
1135 rt->auth, url, method);
1137 av_strlcat(buf, str, sizeof(buf));
1140 if (send_content_length > 0 && send_content)
1141 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1142 av_strlcat(buf, "\r\n", sizeof(buf));
1144 /* base64 encode rtsp if tunneling */
1145 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1146 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1147 out_buf = base64buf;
1150 av_dlog(s, "Sending:\n%s--\n", buf);
1152 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1153 if (send_content_length > 0 && send_content) {
1154 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1155 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1156 "with content data not supported\n");
1157 return AVERROR_PATCHWELCOME;
1159 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1161 rt->last_cmd_time = av_gettime();
1166 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1167 const char *url, const char *headers)
1169 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1172 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1173 const char *headers, RTSPMessageHeader *reply,
1174 unsigned char **content_ptr)
1176 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1177 content_ptr, NULL, 0);
1180 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1181 const char *method, const char *url,
1183 RTSPMessageHeader *reply,
1184 unsigned char **content_ptr,
1185 const unsigned char *send_content,
1186 int send_content_length)
1188 RTSPState *rt = s->priv_data;
1189 HTTPAuthType cur_auth_type;
1190 int ret, attempts = 0;
1193 cur_auth_type = rt->auth_state.auth_type;
1194 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1196 send_content_length)))
1199 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1203 if (reply->status_code == 401 &&
1204 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1205 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1208 if (reply->status_code > 400){
1209 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1213 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1219 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1220 int lower_transport, const char *real_challenge)
1222 RTSPState *rt = s->priv_data;
1223 int rtx = 0, j, i, err, interleave = 0, port_off;
1224 RTSPStream *rtsp_st;
1225 RTSPMessageHeader reply1, *reply = &reply1;
1227 const char *trans_pref;
1229 if (rt->transport == RTSP_TRANSPORT_RDT)
1230 trans_pref = "x-pn-tng";
1231 else if (rt->transport == RTSP_TRANSPORT_RAW)
1232 trans_pref = "RAW/RAW";
1234 trans_pref = "RTP/AVP";
1236 /* default timeout: 1 minute */
1239 /* Choose a random starting offset within the first half of the
1240 * port range, to allow for a number of ports to try even if the offset
1241 * happens to be at the end of the random range. */
1242 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1243 /* even random offset */
1244 port_off -= port_off & 0x01;
1246 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1247 char transport[2048];
1250 * WMS serves all UDP data over a single connection, the RTX, which
1251 * isn't necessarily the first in the SDP but has to be the first
1252 * to be set up, else the second/third SETUP will fail with a 461.
1254 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1255 rt->server_type == RTSP_SERVER_WMS) {
1258 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1259 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1261 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1265 if (rtx == rt->nb_rtsp_streams)
1266 return -1; /* no RTX found */
1267 rtsp_st = rt->rtsp_streams[rtx];
1269 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1271 rtsp_st = rt->rtsp_streams[i];
1274 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1277 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1278 port = reply->transports[0].client_port_min;
1282 /* first try in specified port range */
1283 while (j <= rt->rtp_port_max) {
1284 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1285 "?localport=%d", j);
1286 /* we will use two ports per rtp stream (rtp and rtcp) */
1288 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1289 &s->interrupt_callback, NULL))
1292 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1297 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1299 snprintf(transport, sizeof(transport) - 1,
1300 "%s/UDP;", trans_pref);
1301 if (rt->server_type != RTSP_SERVER_REAL)
1302 av_strlcat(transport, "unicast;", sizeof(transport));
1303 av_strlcatf(transport, sizeof(transport),
1304 "client_port=%d", port);
1305 if (rt->transport == RTSP_TRANSPORT_RTP &&
1306 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1307 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1311 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1312 /* For WMS streams, the application streams are only used for
1313 * UDP. When trying to set it up for TCP streams, the server
1314 * will return an error. Therefore, we skip those streams. */
1315 if (rt->server_type == RTSP_SERVER_WMS &&
1316 (rtsp_st->stream_index < 0 ||
1317 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1320 snprintf(transport, sizeof(transport) - 1,
1321 "%s/TCP;", trans_pref);
1322 if (rt->transport != RTSP_TRANSPORT_RDT)
1323 av_strlcat(transport, "unicast;", sizeof(transport));
1324 av_strlcatf(transport, sizeof(transport),
1325 "interleaved=%d-%d",
1326 interleave, interleave + 1);
1330 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1331 snprintf(transport, sizeof(transport) - 1,
1332 "%s/UDP;multicast", trans_pref);
1335 av_strlcat(transport, ";mode=record", sizeof(transport));
1336 } else if (rt->server_type == RTSP_SERVER_REAL ||
1337 rt->server_type == RTSP_SERVER_WMS)
1338 av_strlcat(transport, ";mode=play", sizeof(transport));
1339 snprintf(cmd, sizeof(cmd),
1340 "Transport: %s\r\n",
1342 if (rt->accept_dynamic_rate)
1343 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1344 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1345 char real_res[41], real_csum[9];
1346 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1348 av_strlcatf(cmd, sizeof(cmd),
1350 "RealChallenge2: %s, sd=%s\r\n",
1351 rt->session_id, real_res, real_csum);
1353 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1354 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1357 } else if (reply->status_code != RTSP_STATUS_OK ||
1358 reply->nb_transports != 1) {
1359 err = AVERROR_INVALIDDATA;
1363 /* XXX: same protocol for all streams is required */
1365 if (reply->transports[0].lower_transport != rt->lower_transport ||
1366 reply->transports[0].transport != rt->transport) {
1367 err = AVERROR_INVALIDDATA;
1371 rt->lower_transport = reply->transports[0].lower_transport;
1372 rt->transport = reply->transports[0].transport;
1375 /* Fail if the server responded with another lower transport mode
1376 * than what we requested. */
1377 if (reply->transports[0].lower_transport != lower_transport) {
1378 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1379 err = AVERROR_INVALIDDATA;
1383 switch(reply->transports[0].lower_transport) {
1384 case RTSP_LOWER_TRANSPORT_TCP:
1385 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1386 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1389 case RTSP_LOWER_TRANSPORT_UDP: {
1390 char url[1024], options[30] = "";
1392 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1393 av_strlcpy(options, "?connect=1", sizeof(options));
1394 /* Use source address if specified */
1395 if (reply->transports[0].source[0]) {
1396 ff_url_join(url, sizeof(url), "rtp", NULL,
1397 reply->transports[0].source,
1398 reply->transports[0].server_port_min, "%s", options);
1400 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1401 reply->transports[0].server_port_min, "%s", options);
1403 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1404 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1405 err = AVERROR_INVALIDDATA;
1408 /* Try to initialize the connection state in a
1409 * potential NAT router by sending dummy packets.
1410 * RTP/RTCP dummy packets are used for RDT, too.
1412 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1414 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1417 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1418 char url[1024], namebuf[50], optbuf[20] = "";
1419 struct sockaddr_storage addr;
1422 if (reply->transports[0].destination.ss_family) {
1423 addr = reply->transports[0].destination;
1424 port = reply->transports[0].port_min;
1425 ttl = reply->transports[0].ttl;
1427 addr = rtsp_st->sdp_ip;
1428 port = rtsp_st->sdp_port;
1429 ttl = rtsp_st->sdp_ttl;
1432 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1433 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1434 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1435 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1436 port, "%s", optbuf);
1437 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1438 &s->interrupt_callback, NULL) < 0) {
1439 err = AVERROR_INVALIDDATA;
1446 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1450 if (rt->nb_rtsp_streams && reply->timeout > 0)
1451 rt->timeout = reply->timeout;
1453 if (rt->server_type == RTSP_SERVER_REAL)
1454 rt->need_subscription = 1;
1459 ff_rtsp_undo_setup(s);
1463 void ff_rtsp_close_connections(AVFormatContext *s)
1465 RTSPState *rt = s->priv_data;
1466 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1467 ffurl_close(rt->rtsp_hd);
1468 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1471 int ff_rtsp_connect(AVFormatContext *s)
1473 RTSPState *rt = s->priv_data;
1474 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1475 int port, err, tcp_fd;
1476 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1477 int lower_transport_mask = 0;
1478 char real_challenge[64] = "";
1479 struct sockaddr_storage peer;
1480 socklen_t peer_len = sizeof(peer);
1482 if (rt->rtp_port_max < rt->rtp_port_min) {
1483 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1484 "than min port %d\n", rt->rtp_port_max,
1486 return AVERROR(EINVAL);
1489 if (!ff_network_init())
1490 return AVERROR(EIO);
1492 if (s->max_delay < 0) /* Not set by the caller */
1493 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1495 rt->control_transport = RTSP_MODE_PLAIN;
1496 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1497 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1498 rt->control_transport = RTSP_MODE_TUNNEL;
1500 /* Only pass through valid flags from here */
1501 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1504 lower_transport_mask = rt->lower_transport_mask;
1505 /* extract hostname and port */
1506 av_url_split(NULL, 0, auth, sizeof(auth),
1507 host, sizeof(host), &port, path, sizeof(path), s->filename);
1509 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1512 port = RTSP_DEFAULT_PORT;
1514 if (!lower_transport_mask)
1515 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1518 /* Only UDP or TCP - UDP multicast isn't supported. */
1519 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1520 (1 << RTSP_LOWER_TRANSPORT_TCP);
1521 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1522 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1523 "only UDP and TCP are supported for output.\n");
1524 err = AVERROR(EINVAL);
1529 /* Construct the URI used in request; this is similar to s->filename,
1530 * but with authentication credentials removed and RTSP specific options
1532 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1533 host, port, "%s", path);
1535 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1536 /* set up initial handshake for tunneling */
1537 char httpname[1024];
1538 char sessioncookie[17];
1541 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1542 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1543 av_get_random_seed(), av_get_random_seed());
1546 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1547 &s->interrupt_callback) < 0) {
1552 /* generate GET headers */
1553 snprintf(headers, sizeof(headers),
1554 "x-sessioncookie: %s\r\n"
1555 "Accept: application/x-rtsp-tunnelled\r\n"
1556 "Pragma: no-cache\r\n"
1557 "Cache-Control: no-cache\r\n",
1559 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1561 /* complete the connection */
1562 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1568 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1569 &s->interrupt_callback) < 0 ) {
1574 /* generate POST headers */
1575 snprintf(headers, sizeof(headers),
1576 "x-sessioncookie: %s\r\n"
1577 "Content-Type: application/x-rtsp-tunnelled\r\n"
1578 "Pragma: no-cache\r\n"
1579 "Cache-Control: no-cache\r\n"
1580 "Content-Length: 32767\r\n"
1581 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1583 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1584 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1586 /* Initialize the authentication state for the POST session. The HTTP
1587 * protocol implementation doesn't properly handle multi-pass
1588 * authentication for POST requests, since it would require one of
1590 * - implementing Expect: 100-continue, which many HTTP servers
1591 * don't support anyway, even less the RTSP servers that do HTTP
1593 * - sending the whole POST data until getting a 401 reply specifying
1594 * what authentication method to use, then resending all that data
1595 * - waiting for potential 401 replies directly after sending the
1596 * POST header (waiting for some unspecified time)
1597 * Therefore, we copy the full auth state, which works for both basic
1598 * and digest. (For digest, we would have to synchronize the nonce
1599 * count variable between the two sessions, if we'd do more requests
1600 * with the original session, though.)
1602 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1604 /* complete the connection */
1605 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1610 /* open the tcp connection */
1611 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1612 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1613 &s->interrupt_callback, NULL) < 0) {
1617 rt->rtsp_hd_out = rt->rtsp_hd;
1621 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1622 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1623 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1624 NULL, 0, NI_NUMERICHOST);
1627 /* request options supported by the server; this also detects server
1629 for (rt->server_type = RTSP_SERVER_RTP;;) {
1631 if (rt->server_type == RTSP_SERVER_REAL)
1634 * The following entries are required for proper
1635 * streaming from a Realmedia server. They are
1636 * interdependent in some way although we currently
1637 * don't quite understand how. Values were copied
1638 * from mplayer SVN r23589.
1639 * ClientChallenge is a 16-byte ID in hex
1640 * CompanyID is a 16-byte ID in base64
1642 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1643 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1644 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1645 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1647 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1648 if (reply->status_code != RTSP_STATUS_OK) {
1649 err = AVERROR_INVALIDDATA;
1653 /* detect server type if not standard-compliant RTP */
1654 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1655 rt->server_type = RTSP_SERVER_REAL;
1657 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1658 rt->server_type = RTSP_SERVER_WMS;
1659 } else if (rt->server_type == RTSP_SERVER_REAL)
1660 strcpy(real_challenge, reply->real_challenge);
1664 if (s->iformat && CONFIG_RTSP_DEMUXER)
1665 err = ff_rtsp_setup_input_streams(s, reply);
1666 else if (CONFIG_RTSP_MUXER)
1667 err = ff_rtsp_setup_output_streams(s, host);
1672 int lower_transport = ff_log2_tab[lower_transport_mask &
1673 ~(lower_transport_mask - 1)];
1675 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1676 rt->server_type == RTSP_SERVER_REAL ?
1677 real_challenge : NULL);
1680 lower_transport_mask &= ~(1 << lower_transport);
1681 if (lower_transport_mask == 0 && err == 1) {
1682 err = AVERROR(EPROTONOSUPPORT);
1687 rt->lower_transport_mask = lower_transport_mask;
1688 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1689 rt->state = RTSP_STATE_IDLE;
1690 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1693 ff_rtsp_close_streams(s);
1694 ff_rtsp_close_connections(s);
1695 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1696 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1697 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1705 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1708 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1709 uint8_t *buf, int buf_size, int64_t wait_end)
1711 RTSPState *rt = s->priv_data;
1712 RTSPStream *rtsp_st;
1713 int n, i, ret, tcp_fd, timeout_cnt = 0;
1715 struct pollfd *p = rt->p;
1716 int *fds = NULL, fdsnum, fdsidx;
1719 if (ff_check_interrupt(&s->interrupt_callback))
1720 return AVERROR_EXIT;
1721 if (wait_end && wait_end - av_gettime() < 0)
1722 return AVERROR(EAGAIN);
1725 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1726 p[max_p].fd = tcp_fd;
1727 p[max_p++].events = POLLIN;
1731 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1732 rtsp_st = rt->rtsp_streams[i];
1733 if (rtsp_st->rtp_handle) {
1734 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1736 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1740 av_log(s, AV_LOG_ERROR,
1741 "Number of fds %d not supported\n", fdsnum);
1742 return AVERROR_INVALIDDATA;
1744 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1745 p[max_p].fd = fds[fdsidx];
1746 p[max_p++].events = POLLIN;
1751 n = poll(p, max_p, POLL_TIMEOUT_MS);
1753 int j = 1 - (tcp_fd == -1);
1755 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1756 rtsp_st = rt->rtsp_streams[i];
1757 if (rtsp_st->rtp_handle) {
1758 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1759 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1761 *prtsp_st = rtsp_st;
1768 #if CONFIG_RTSP_DEMUXER
1769 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1770 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1771 if (rt->state == RTSP_STATE_STREAMING) {
1772 if (!ff_rtsp_parse_streaming_commands(s))
1775 av_log(s, AV_LOG_WARNING,
1776 "Unable to answer to TEARDOWN\n");
1780 RTSPMessageHeader reply;
1781 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1784 /* XXX: parse message */
1785 if (rt->state != RTSP_STATE_STREAMING)
1790 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1791 return AVERROR(ETIMEDOUT);
1792 } else if (n < 0 && errno != EINTR)
1793 return AVERROR(errno);
1797 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
1798 const uint8_t *buf, int len)
1800 RTSPState *rt = s->priv_data;
1804 if (rt->nb_rtsp_streams == 1) {
1805 *rtsp_st = rt->rtsp_streams[0];
1808 if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
1809 if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
1811 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1812 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1815 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
1816 *rtsp_st = rt->rtsp_streams[i];
1823 av_log(s, AV_LOG_WARNING,
1824 "Unable to pick stream for packet - SSRC not known for "
1826 return AVERROR(EAGAIN);
1829 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1830 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
1831 *rtsp_st = rt->rtsp_streams[i];
1837 av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
1838 return AVERROR(EAGAIN);
1841 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1843 RTSPState *rt = s->priv_data;
1845 RTSPStream *rtsp_st, *first_queue_st = NULL;
1846 int64_t wait_end = 0;
1848 if (rt->nb_byes == rt->nb_rtsp_streams)
1851 /* get next frames from the same RTP packet */
1852 if (rt->cur_transport_priv) {
1853 if (rt->transport == RTSP_TRANSPORT_RDT) {
1854 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1855 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1856 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1857 } else if (rt->ts && CONFIG_RTPDEC) {
1858 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1860 rt->recvbuf_pos += ret;
1861 ret = rt->recvbuf_pos < rt->recvbuf_len;
1866 rt->cur_transport_priv = NULL;
1868 } else if (ret == 1) {
1871 rt->cur_transport_priv = NULL;
1875 if (rt->transport == RTSP_TRANSPORT_RTP) {
1877 int64_t first_queue_time = 0;
1878 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1879 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1883 queue_time = ff_rtp_queued_packet_time(rtpctx);
1884 if (queue_time && (queue_time - first_queue_time < 0 ||
1885 !first_queue_time)) {
1886 first_queue_time = queue_time;
1887 first_queue_st = rt->rtsp_streams[i];
1890 if (first_queue_time) {
1891 wait_end = first_queue_time + s->max_delay;
1894 first_queue_st = NULL;
1898 /* read next RTP packet */
1900 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1902 return AVERROR(ENOMEM);
1905 switch(rt->lower_transport) {
1907 #if CONFIG_RTSP_DEMUXER
1908 case RTSP_LOWER_TRANSPORT_TCP:
1909 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1912 case RTSP_LOWER_TRANSPORT_UDP:
1913 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1914 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1915 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1916 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
1918 case RTSP_LOWER_TRANSPORT_CUSTOM:
1919 if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
1920 wait_end && wait_end < av_gettime())
1921 len = AVERROR(EAGAIN);
1923 len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
1924 len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
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, NULL, s->pb, len);
1929 if (len == AVERROR(EAGAIN) && first_queue_st &&
1930 rt->transport == RTSP_TRANSPORT_RTP) {
1931 rtsp_st = first_queue_st;
1932 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1939 if (rt->transport == RTSP_TRANSPORT_RDT) {
1940 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1941 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1942 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1943 if (rtsp_st->feedback) {
1944 AVIOContext *pb = NULL;
1945 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
1947 ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
1950 /* Either bad packet, or a RTCP packet. Check if the
1951 * first_rtcp_ntp_time field was initialized. */
1952 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1953 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1954 /* first_rtcp_ntp_time has been initialized for this stream,
1955 * copy the same value to all other uninitialized streams,
1956 * in order to map their timestamp origin to the same ntp time
1959 AVStream *st = NULL;
1960 if (rtsp_st->stream_index >= 0)
1961 st = s->streams[rtsp_st->stream_index];
1962 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1963 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1964 AVStream *st2 = NULL;
1965 if (rt->rtsp_streams[i]->stream_index >= 0)
1966 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1967 if (rtpctx2 && st && st2 &&
1968 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1969 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1970 rtpctx2->rtcp_ts_offset = av_rescale_q(
1971 rtpctx->rtcp_ts_offset, st->time_base,
1976 if (ret == -RTCP_BYE) {
1979 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1980 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1982 if (rt->nb_byes == rt->nb_rtsp_streams)
1986 } else if (rt->ts && CONFIG_RTPDEC) {
1987 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1990 rt->recvbuf_len = len;
1991 rt->recvbuf_pos = ret;
1992 rt->cur_transport_priv = rt->ts;
1999 return AVERROR_INVALIDDATA;
2005 /* more packets may follow, so we save the RTP context */
2006 rt->cur_transport_priv = rtsp_st->transport_priv;
2010 #endif /* CONFIG_RTPDEC */
2012 #if CONFIG_SDP_DEMUXER
2013 static int sdp_probe(AVProbeData *p1)
2015 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
2017 /* we look for a line beginning "c=IN IP" */
2018 while (p < p_end && *p != '\0') {
2019 if (p + sizeof("c=IN IP") - 1 < p_end &&
2020 av_strstart(p, "c=IN IP", NULL))
2021 return AVPROBE_SCORE_MAX / 2;
2023 while (p < p_end - 1 && *p != '\n') p++;
2032 static int sdp_read_header(AVFormatContext *s)
2034 RTSPState *rt = s->priv_data;
2035 RTSPStream *rtsp_st;
2040 if (!ff_network_init())
2041 return AVERROR(EIO);
2043 if (s->max_delay < 0) /* Not set by the caller */
2044 s->max_delay = DEFAULT_REORDERING_DELAY;
2045 if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
2046 rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;
2048 /* read the whole sdp file */
2049 /* XXX: better loading */
2050 content = av_malloc(SDP_MAX_SIZE);
2051 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
2054 return AVERROR_INVALIDDATA;
2056 content[size] ='\0';
2058 err = ff_sdp_parse(s, content);
2062 /* open each RTP stream */
2063 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2065 rtsp_st = rt->rtsp_streams[i];
2067 if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
2068 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
2069 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
2070 ff_url_join(url, sizeof(url), "rtp", NULL,
2071 namebuf, rtsp_st->sdp_port,
2072 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
2074 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
2075 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
2076 &s->interrupt_callback, NULL) < 0) {
2077 err = AVERROR_INVALIDDATA;
2081 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
2086 ff_rtsp_close_streams(s);
2091 static int sdp_read_close(AVFormatContext *s)
2093 ff_rtsp_close_streams(s);
2098 static const AVClass sdp_demuxer_class = {
2099 .class_name = "SDP demuxer",
2100 .item_name = av_default_item_name,
2101 .option = sdp_options,
2102 .version = LIBAVUTIL_VERSION_INT,
2105 AVInputFormat ff_sdp_demuxer = {
2107 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2108 .priv_data_size = sizeof(RTSPState),
2109 .read_probe = sdp_probe,
2110 .read_header = sdp_read_header,
2111 .read_packet = ff_rtsp_fetch_packet,
2112 .read_close = sdp_read_close,
2113 .priv_class = &sdp_demuxer_class,
2115 #endif /* CONFIG_SDP_DEMUXER */
2117 #if CONFIG_RTP_DEMUXER
2118 static int rtp_probe(AVProbeData *p)
2120 if (av_strstart(p->filename, "rtp:", NULL))
2121 return AVPROBE_SCORE_MAX;
2125 static int rtp_read_header(AVFormatContext *s)
2127 uint8_t recvbuf[1500];
2128 char host[500], sdp[500];
2130 URLContext* in = NULL;
2132 AVCodecContext codec = { 0 };
2133 struct sockaddr_storage addr;
2135 socklen_t addrlen = sizeof(addr);
2136 RTSPState *rt = s->priv_data;
2138 if (!ff_network_init())
2139 return AVERROR(EIO);
2141 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2142 &s->interrupt_callback, NULL);
2147 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2148 if (ret == AVERROR(EAGAIN))
2153 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2157 if ((recvbuf[0] & 0xc0) != 0x80) {
2158 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2163 if (RTP_PT_IS_RTCP(recvbuf[1]))
2166 payload_type = recvbuf[1] & 0x7f;
2169 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2173 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2174 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2175 "without an SDP file describing it\n",
2179 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2180 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2181 "properly you need an SDP file "
2185 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2186 NULL, 0, s->filename);
2188 snprintf(sdp, sizeof(sdp),
2189 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2190 addr.ss_family == AF_INET ? 4 : 6, host,
2191 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2192 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2193 port, payload_type);
2194 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2196 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2199 /* sdp_read_header initializes this again */
2202 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2204 ret = sdp_read_header(s);
2215 static const AVClass rtp_demuxer_class = {
2216 .class_name = "RTP demuxer",
2217 .item_name = av_default_item_name,
2218 .option = rtp_options,
2219 .version = LIBAVUTIL_VERSION_INT,
2222 AVInputFormat ff_rtp_demuxer = {
2224 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2225 .priv_data_size = sizeof(RTSPState),
2226 .read_probe = rtp_probe,
2227 .read_header = rtp_read_header,
2228 .read_packet = ff_rtsp_fetch_packet,
2229 .read_close = sdp_read_close,
2230 .flags = AVFMT_NOFILE,
2231 .priv_class = &rtp_demuxer_class,
2233 #endif /* CONFIG_RTP_DEMUXER */