3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/time.h"
32 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
53 /* Timeout values for socket poll, in ms,
54 * and read_packet(), in seconds */
55 #define POLL_TIMEOUT_MS 100
56 #define READ_PACKET_TIMEOUT_S 10
57 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
58 #define SDP_MAX_SIZE 16384
59 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
60 #define DEFAULT_REORDERING_DELAY 100000
62 #define OFFSET(x) offsetof(RTSPState, x)
63 #define DEC AV_OPT_FLAG_DECODING_PARAM
64 #define ENC AV_OPT_FLAG_ENCODING_PARAM
66 #define RTSP_FLAG_OPTS(name, longname) \
67 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
68 { "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" }, \
69 { "listen", "Wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" }
71 #define RTSP_MEDIATYPE_OPTS(name, longname) \
72 { 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" }, \
73 { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
74 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
75 { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
77 const AVOption ff_rtsp_options[] = {
78 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
79 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
80 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
81 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
82 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
83 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
84 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
85 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
86 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
87 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
88 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
89 { "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 },
93 static const AVOption sdp_options[] = {
94 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
95 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
99 static const AVOption rtp_options[] = {
100 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
104 static void get_word_until_chars(char *buf, int buf_size,
105 const char *sep, const char **pp)
111 p += strspn(p, SPACE_CHARS);
113 while (!strchr(sep, *p) && *p != '\0') {
114 if ((q - buf) < buf_size - 1)
123 static void get_word_sep(char *buf, int buf_size, const char *sep,
126 if (**pp == '/') (*pp)++;
127 get_word_until_chars(buf, buf_size, sep, pp);
130 static void get_word(char *buf, int buf_size, const char **pp)
132 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
135 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
137 * Used for seeking in the rtp stream.
139 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
143 p += strspn(p, SPACE_CHARS);
144 if (!av_stristart(p, "npt=", &p))
147 *start = AV_NOPTS_VALUE;
148 *end = AV_NOPTS_VALUE;
150 get_word_sep(buf, sizeof(buf), "-", &p);
151 av_parse_time(start, buf, 1);
154 get_word_sep(buf, sizeof(buf), "-", &p);
155 av_parse_time(end, buf, 1);
159 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
161 struct addrinfo hints = { 0 }, *ai = NULL;
162 hints.ai_flags = AI_NUMERICHOST;
163 if (getaddrinfo(buf, NULL, &hints, &ai))
165 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
171 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
172 RTSPStream *rtsp_st, AVCodecContext *codec)
176 codec->codec_id = handler->codec_id;
177 rtsp_st->dynamic_handler = handler;
178 if (handler->alloc) {
179 rtsp_st->dynamic_protocol_context = handler->alloc();
180 if (!rtsp_st->dynamic_protocol_context)
181 rtsp_st->dynamic_handler = NULL;
185 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
186 static int sdp_parse_rtpmap(AVFormatContext *s,
187 AVStream *st, RTSPStream *rtsp_st,
188 int payload_type, const char *p)
190 AVCodecContext *codec = st->codec;
196 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
197 * see if we can handle this kind of payload.
198 * The space should normally not be there but some Real streams or
199 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
200 * have a trailing space. */
201 get_word_sep(buf, sizeof(buf), "/ ", &p);
202 if (payload_type < RTP_PT_PRIVATE) {
203 /* We are in a standard case
204 * (from http://www.iana.org/assignments/rtp-parameters). */
205 /* search into AVRtpPayloadTypes[] */
206 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
209 if (codec->codec_id == AV_CODEC_ID_NONE) {
210 RTPDynamicProtocolHandler *handler =
211 ff_rtp_handler_find_by_name(buf, codec->codec_type);
212 init_rtp_handler(handler, rtsp_st, codec);
213 /* If no dynamic handler was found, check with the list of standard
214 * allocated types, if such a stream for some reason happens to
215 * use a private payload type. This isn't handled in rtpdec.c, since
216 * the format name from the rtpmap line never is passed into rtpdec. */
217 if (!rtsp_st->dynamic_handler)
218 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
221 c = avcodec_find_decoder(codec->codec_id);
227 get_word_sep(buf, sizeof(buf), "/", &p);
229 switch (codec->codec_type) {
230 case AVMEDIA_TYPE_AUDIO:
231 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
232 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
233 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
235 codec->sample_rate = i;
236 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
237 get_word_sep(buf, sizeof(buf), "/", &p);
241 // TODO: there is a bug here; if it is a mono stream, and
242 // less than 22000Hz, faad upconverts to stereo and twice
243 // the frequency. No problem, but the sample rate is being
244 // set here by the sdp line. Patch on its way. (rdm)
246 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
248 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
251 case AVMEDIA_TYPE_VIDEO:
252 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
254 avpriv_set_pts_info(st, 32, 1, i);
259 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
260 rtsp_st->dynamic_handler->init(s, st->index,
261 rtsp_st->dynamic_protocol_context);
265 /* parse the attribute line from the fmtp a line of an sdp response. This
266 * is broken out as a function because it is used in rtp_h264.c, which is
268 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
269 char *value, int value_size)
271 *p += strspn(*p, SPACE_CHARS);
273 get_word_sep(attr, attr_size, "=", p);
276 get_word_sep(value, value_size, ";", p);
284 typedef struct SDPParseState {
286 struct sockaddr_storage default_ip;
288 int skip_media; ///< set if an unknown m= line occurs
291 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
292 int letter, const char *buf)
294 RTSPState *rt = s->priv_data;
295 char buf1[64], st_type[64];
297 enum AVMediaType codec_type;
301 struct sockaddr_storage sdp_ip;
304 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
307 if (s1->skip_media && letter != 'm')
311 get_word(buf1, sizeof(buf1), &p);
312 if (strcmp(buf1, "IN") != 0)
314 get_word(buf1, sizeof(buf1), &p);
315 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
317 get_word_sep(buf1, sizeof(buf1), "/", &p);
318 if (get_sockaddr(buf1, &sdp_ip))
323 get_word_sep(buf1, sizeof(buf1), "/", &p);
326 if (s->nb_streams == 0) {
327 s1->default_ip = sdp_ip;
328 s1->default_ttl = ttl;
330 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
331 rtsp_st->sdp_ip = sdp_ip;
332 rtsp_st->sdp_ttl = ttl;
336 av_dict_set(&s->metadata, "title", p, 0);
339 if (s->nb_streams == 0) {
340 av_dict_set(&s->metadata, "comment", p, 0);
347 codec_type = AVMEDIA_TYPE_UNKNOWN;
348 get_word(st_type, sizeof(st_type), &p);
349 if (!strcmp(st_type, "audio")) {
350 codec_type = AVMEDIA_TYPE_AUDIO;
351 } else if (!strcmp(st_type, "video")) {
352 codec_type = AVMEDIA_TYPE_VIDEO;
353 } else if (!strcmp(st_type, "application")) {
354 codec_type = AVMEDIA_TYPE_DATA;
356 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
360 rtsp_st = av_mallocz(sizeof(RTSPStream));
363 rtsp_st->stream_index = -1;
364 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
366 rtsp_st->sdp_ip = s1->default_ip;
367 rtsp_st->sdp_ttl = s1->default_ttl;
369 get_word(buf1, sizeof(buf1), &p); /* port */
370 rtsp_st->sdp_port = atoi(buf1);
372 get_word(buf1, sizeof(buf1), &p); /* protocol */
373 if (!strcmp(buf1, "udp"))
374 rt->transport = RTSP_TRANSPORT_RAW;
376 /* XXX: handle list of formats */
377 get_word(buf1, sizeof(buf1), &p); /* format list */
378 rtsp_st->sdp_payload_type = atoi(buf1);
380 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
381 /* no corresponding stream */
382 if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
383 rt->ts = ff_mpegts_parse_open(s);
384 } else if (rt->server_type == RTSP_SERVER_WMS &&
385 codec_type == AVMEDIA_TYPE_DATA) {
386 /* RTX stream, a stream that carries all the other actual
387 * audio/video streams. Don't expose this to the callers. */
389 st = avformat_new_stream(s, NULL);
392 st->id = rt->nb_rtsp_streams - 1;
393 rtsp_st->stream_index = st->index;
394 st->codec->codec_type = codec_type;
395 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
396 RTPDynamicProtocolHandler *handler;
397 /* if standard payload type, we can find the codec right now */
398 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
399 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
400 st->codec->sample_rate > 0)
401 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
402 /* Even static payload types may need a custom depacketizer */
403 handler = ff_rtp_handler_find_by_id(
404 rtsp_st->sdp_payload_type, st->codec->codec_type);
405 init_rtp_handler(handler, rtsp_st, st->codec);
406 if (handler && handler->init)
407 handler->init(s, st->index,
408 rtsp_st->dynamic_protocol_context);
411 /* put a default control url */
412 av_strlcpy(rtsp_st->control_url, rt->control_uri,
413 sizeof(rtsp_st->control_url));
416 if (av_strstart(p, "control:", &p)) {
417 if (s->nb_streams == 0) {
418 if (!strncmp(p, "rtsp://", 7))
419 av_strlcpy(rt->control_uri, p,
420 sizeof(rt->control_uri));
423 /* get the control url */
424 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
426 /* XXX: may need to add full url resolution */
427 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
429 if (proto[0] == '\0') {
430 /* relative control URL */
431 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
432 av_strlcat(rtsp_st->control_url, "/",
433 sizeof(rtsp_st->control_url));
434 av_strlcat(rtsp_st->control_url, p,
435 sizeof(rtsp_st->control_url));
437 av_strlcpy(rtsp_st->control_url, p,
438 sizeof(rtsp_st->control_url));
440 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
441 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
442 get_word(buf1, sizeof(buf1), &p);
443 payload_type = atoi(buf1);
444 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
445 if (rtsp_st->stream_index >= 0) {
446 st = s->streams[rtsp_st->stream_index];
447 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
449 } else if (av_strstart(p, "fmtp:", &p) ||
450 av_strstart(p, "framesize:", &p)) {
451 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
452 // let dynamic protocol handlers have a stab at the line.
453 get_word(buf1, sizeof(buf1), &p);
454 payload_type = atoi(buf1);
455 for (i = 0; i < rt->nb_rtsp_streams; i++) {
456 rtsp_st = rt->rtsp_streams[i];
457 if (rtsp_st->sdp_payload_type == payload_type &&
458 rtsp_st->dynamic_handler &&
459 rtsp_st->dynamic_handler->parse_sdp_a_line)
460 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
461 rtsp_st->dynamic_protocol_context, buf);
463 } else if (av_strstart(p, "range:", &p)) {
466 // this is so that seeking on a streamed file can work.
467 rtsp_parse_range_npt(p, &start, &end);
468 s->start_time = start;
469 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
470 s->duration = (end == AV_NOPTS_VALUE) ?
471 AV_NOPTS_VALUE : end - start;
472 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
474 rt->transport = RTSP_TRANSPORT_RDT;
475 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
477 st = s->streams[s->nb_streams - 1];
478 st->codec->sample_rate = atoi(p);
480 if (rt->server_type == RTSP_SERVER_WMS)
481 ff_wms_parse_sdp_a_line(s, p);
482 if (s->nb_streams > 0) {
483 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
485 if (rt->server_type == RTSP_SERVER_REAL)
486 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
488 if (rtsp_st->dynamic_handler &&
489 rtsp_st->dynamic_handler->parse_sdp_a_line)
490 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
491 rtsp_st->stream_index,
492 rtsp_st->dynamic_protocol_context, buf);
499 int ff_sdp_parse(AVFormatContext *s, const char *content)
501 RTSPState *rt = s->priv_data;
504 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
505 * contain long SDP lines containing complete ASF Headers (several
506 * kB) or arrays of MDPR (RM stream descriptor) headers plus
507 * "rulebooks" describing their properties. Therefore, the SDP line
510 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
511 * in rtpdec_xiph.c. */
513 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
517 p += strspn(p, SPACE_CHARS);
525 /* get the content */
527 while (*p != '\n' && *p != '\r' && *p != '\0') {
528 if ((q - buf) < sizeof(buf) - 1)
533 sdp_parse_line(s, s1, letter, buf);
535 while (*p != '\n' && *p != '\0')
540 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
541 if (!rt->p) return AVERROR(ENOMEM);
544 #endif /* CONFIG_RTPDEC */
546 void ff_rtsp_undo_setup(AVFormatContext *s)
548 RTSPState *rt = s->priv_data;
551 for (i = 0; i < rt->nb_rtsp_streams; i++) {
552 RTSPStream *rtsp_st = rt->rtsp_streams[i];
555 if (rtsp_st->transport_priv) {
557 AVFormatContext *rtpctx = rtsp_st->transport_priv;
558 av_write_trailer(rtpctx);
559 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
561 avio_close_dyn_buf(rtpctx->pb, &ptr);
564 avio_close(rtpctx->pb);
566 avformat_free_context(rtpctx);
567 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
568 ff_rdt_parse_close(rtsp_st->transport_priv);
569 else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)
570 ff_rtp_parse_close(rtsp_st->transport_priv);
572 rtsp_st->transport_priv = NULL;
573 if (rtsp_st->rtp_handle)
574 ffurl_close(rtsp_st->rtp_handle);
575 rtsp_st->rtp_handle = NULL;
579 /* close and free RTSP streams */
580 void ff_rtsp_close_streams(AVFormatContext *s)
582 RTSPState *rt = s->priv_data;
586 ff_rtsp_undo_setup(s);
587 for (i = 0; i < rt->nb_rtsp_streams; i++) {
588 rtsp_st = rt->rtsp_streams[i];
590 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
591 rtsp_st->dynamic_handler->free(
592 rtsp_st->dynamic_protocol_context);
596 av_free(rt->rtsp_streams);
598 avformat_close_input(&rt->asf_ctx);
600 if (rt->ts && CONFIG_RTPDEC)
601 ff_mpegts_parse_close(rt->ts);
603 av_free(rt->recvbuf);
606 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
608 RTSPState *rt = s->priv_data;
611 /* open the RTP context */
612 if (rtsp_st->stream_index >= 0)
613 st = s->streams[rtsp_st->stream_index];
615 s->ctx_flags |= AVFMTCTX_NOHEADER;
617 if (s->oformat && CONFIG_RTSP_MUXER) {
618 int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
620 RTSP_TCP_MAX_PACKET_SIZE);
621 /* Ownership of rtp_handle is passed to the rtp mux context */
622 rtsp_st->rtp_handle = NULL;
625 } else if (rt->transport == RTSP_TRANSPORT_RAW) {
626 return 0; // Don't need to open any parser here
627 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
628 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
629 rtsp_st->dynamic_protocol_context,
630 rtsp_st->dynamic_handler);
631 else if (CONFIG_RTPDEC)
632 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
633 rtsp_st->sdp_payload_type,
634 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
635 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
637 if (!rtsp_st->transport_priv) {
638 return AVERROR(ENOMEM);
639 } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {
640 if (rtsp_st->dynamic_handler) {
641 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
642 rtsp_st->dynamic_protocol_context,
643 rtsp_st->dynamic_handler);
650 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
651 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
658 q += strspn(q, SPACE_CHARS);
659 v = strtol(q, &p, 10);
663 v = strtol(p, &p, 10);
672 /* XXX: only one transport specification is parsed */
673 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
675 char transport_protocol[16];
677 char lower_transport[16];
679 RTSPTransportField *th;
682 reply->nb_transports = 0;
685 p += strspn(p, SPACE_CHARS);
689 th = &reply->transports[reply->nb_transports];
691 get_word_sep(transport_protocol, sizeof(transport_protocol),
693 if (!av_strcasecmp (transport_protocol, "rtp")) {
694 get_word_sep(profile, sizeof(profile), "/;,", &p);
695 lower_transport[0] = '\0';
696 /* rtp/avp/<protocol> */
698 get_word_sep(lower_transport, sizeof(lower_transport),
701 th->transport = RTSP_TRANSPORT_RTP;
702 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
703 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
704 /* x-pn-tng/<protocol> */
705 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
707 th->transport = RTSP_TRANSPORT_RDT;
708 } else if (!av_strcasecmp(transport_protocol, "raw")) {
709 get_word_sep(profile, sizeof(profile), "/;,", &p);
710 lower_transport[0] = '\0';
711 /* raw/raw/<protocol> */
713 get_word_sep(lower_transport, sizeof(lower_transport),
716 th->transport = RTSP_TRANSPORT_RAW;
718 if (!av_strcasecmp(lower_transport, "TCP"))
719 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
721 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
725 /* get each parameter */
726 while (*p != '\0' && *p != ',') {
727 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
728 if (!strcmp(parameter, "port")) {
731 rtsp_parse_range(&th->port_min, &th->port_max, &p);
733 } else if (!strcmp(parameter, "client_port")) {
736 rtsp_parse_range(&th->client_port_min,
737 &th->client_port_max, &p);
739 } else if (!strcmp(parameter, "server_port")) {
742 rtsp_parse_range(&th->server_port_min,
743 &th->server_port_max, &p);
745 } else if (!strcmp(parameter, "interleaved")) {
748 rtsp_parse_range(&th->interleaved_min,
749 &th->interleaved_max, &p);
751 } else if (!strcmp(parameter, "multicast")) {
752 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
753 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
754 } else if (!strcmp(parameter, "ttl")) {
757 th->ttl = strtol(p, (char **)&p, 10);
759 } else if (!strcmp(parameter, "destination")) {
762 get_word_sep(buf, sizeof(buf), ";,", &p);
763 get_sockaddr(buf, &th->destination);
765 } else if (!strcmp(parameter, "source")) {
768 get_word_sep(buf, sizeof(buf), ";,", &p);
769 av_strlcpy(th->source, buf, sizeof(th->source));
771 } else if (!strcmp(parameter, "mode")) {
774 get_word_sep(buf, sizeof(buf), ";, ", &p);
775 if (!strcmp(buf, "record") ||
776 !strcmp(buf, "receive"))
781 while (*p != ';' && *p != '\0' && *p != ',')
789 reply->nb_transports++;
793 static void handle_rtp_info(RTSPState *rt, const char *url,
794 uint32_t seq, uint32_t rtptime)
797 if (!rtptime || !url[0])
799 if (rt->transport != RTSP_TRANSPORT_RTP)
801 for (i = 0; i < rt->nb_rtsp_streams; i++) {
802 RTSPStream *rtsp_st = rt->rtsp_streams[i];
803 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
806 if (!strcmp(rtsp_st->control_url, url)) {
807 rtpctx->base_timestamp = rtptime;
813 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
816 char key[20], value[1024], url[1024] = "";
817 uint32_t seq = 0, rtptime = 0;
820 p += strspn(p, SPACE_CHARS);
823 get_word_sep(key, sizeof(key), "=", &p);
827 get_word_sep(value, sizeof(value), ";, ", &p);
829 if (!strcmp(key, "url"))
830 av_strlcpy(url, value, sizeof(url));
831 else if (!strcmp(key, "seq"))
832 seq = strtoul(value, NULL, 10);
833 else if (!strcmp(key, "rtptime"))
834 rtptime = strtoul(value, NULL, 10);
836 handle_rtp_info(rt, url, seq, rtptime);
845 handle_rtp_info(rt, url, seq, rtptime);
848 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
849 RTSPState *rt, const char *method)
853 /* NOTE: we do case independent match for broken servers */
855 if (av_stristart(p, "Session:", &p)) {
857 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
858 if (av_stristart(p, ";timeout=", &p) &&
859 (t = strtol(p, NULL, 10)) > 0) {
862 } else if (av_stristart(p, "Content-Length:", &p)) {
863 reply->content_length = strtol(p, NULL, 10);
864 } else if (av_stristart(p, "Transport:", &p)) {
865 rtsp_parse_transport(reply, p);
866 } else if (av_stristart(p, "CSeq:", &p)) {
867 reply->seq = strtol(p, NULL, 10);
868 } else if (av_stristart(p, "Range:", &p)) {
869 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
870 } else if (av_stristart(p, "RealChallenge1:", &p)) {
871 p += strspn(p, SPACE_CHARS);
872 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
873 } else if (av_stristart(p, "Server:", &p)) {
874 p += strspn(p, SPACE_CHARS);
875 av_strlcpy(reply->server, p, sizeof(reply->server));
876 } else if (av_stristart(p, "Notice:", &p) ||
877 av_stristart(p, "X-Notice:", &p)) {
878 reply->notice = strtol(p, NULL, 10);
879 } else if (av_stristart(p, "Location:", &p)) {
880 p += strspn(p, SPACE_CHARS);
881 av_strlcpy(reply->location, p , sizeof(reply->location));
882 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
883 p += strspn(p, SPACE_CHARS);
884 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
885 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
886 p += strspn(p, SPACE_CHARS);
887 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
888 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
889 p += strspn(p, SPACE_CHARS);
890 if (method && !strcmp(method, "DESCRIBE"))
891 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
892 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
893 p += strspn(p, SPACE_CHARS);
894 if (method && !strcmp(method, "PLAY"))
895 rtsp_parse_rtp_info(rt, p);
896 } else if (av_stristart(p, "Public:", &p) && rt) {
897 if (strstr(p, "GET_PARAMETER") &&
898 method && !strcmp(method, "OPTIONS"))
899 rt->get_parameter_supported = 1;
900 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
901 p += strspn(p, SPACE_CHARS);
902 rt->accept_dynamic_rate = atoi(p);
903 } else if (av_stristart(p, "Content-Type:", &p)) {
904 p += strspn(p, SPACE_CHARS);
905 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
909 /* skip a RTP/TCP interleaved packet */
910 void ff_rtsp_skip_packet(AVFormatContext *s)
912 RTSPState *rt = s->priv_data;
916 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
919 len = AV_RB16(buf + 1);
921 av_dlog(s, "skipping RTP packet len=%d\n", len);
926 if (len1 > sizeof(buf))
928 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
935 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
936 unsigned char **content_ptr,
937 int return_on_interleaved_data, const char *method)
939 RTSPState *rt = s->priv_data;
940 char buf[4096], buf1[1024], *q;
943 int ret, content_length, line_count = 0, request = 0;
944 unsigned char *content = NULL;
950 memset(reply, 0, sizeof(*reply));
952 /* parse reply (XXX: use buffers) */
953 rt->last_reply[0] = '\0';
957 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
958 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
964 /* XXX: only parse it if first char on line ? */
965 if (return_on_interleaved_data) {
968 ff_rtsp_skip_packet(s);
969 } else if (ch != '\r') {
970 if ((q - buf) < sizeof(buf) - 1)
976 av_dlog(s, "line='%s'\n", buf);
978 /* test if last line */
982 if (line_count == 0) {
984 get_word(buf1, sizeof(buf1), &p);
985 if (!strncmp(buf1, "RTSP/", 5)) {
986 get_word(buf1, sizeof(buf1), &p);
987 reply->status_code = atoi(buf1);
988 av_strlcpy(reply->reason, p, sizeof(reply->reason));
990 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
991 get_word(buf1, sizeof(buf1), &p); // object
995 ff_rtsp_parse_line(reply, p, rt, method);
996 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
997 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
1002 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
1003 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
1005 content_length = reply->content_length;
1006 if (content_length > 0) {
1007 /* leave some room for a trailing '\0' (useful for simple parsing) */
1008 content = av_malloc(content_length + 1);
1009 ffurl_read_complete(rt->rtsp_hd, content, content_length);
1010 content[content_length] = '\0';
1013 *content_ptr = content;
1019 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1020 const char* ptr = buf;
1022 if (!strcmp(reply->reason, "OPTIONS")) {
1023 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
1025 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
1026 if (reply->session_id[0])
1027 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1030 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1032 av_strlcat(buf, "\r\n", sizeof(buf));
1034 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1035 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1038 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1040 rt->last_cmd_time = av_gettime();
1041 /* Even if the request from the server had data, it is not the data
1042 * that the caller wants or expects. The memory could also be leaked
1043 * if the actual following reply has content data. */
1045 av_freep(content_ptr);
1046 /* If method is set, this is called from ff_rtsp_send_cmd,
1047 * where a reply to exactly this request is awaited. For
1048 * callers from within packet receiving, we just want to
1049 * return to the caller and go back to receiving packets. */
1055 if (rt->seq != reply->seq) {
1056 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1057 rt->seq, reply->seq);
1061 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1062 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1063 reply->notice == 2306 /* Continuous Feed Terminated */) {
1064 rt->state = RTSP_STATE_IDLE;
1065 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1066 return AVERROR(EIO); /* data or server error */
1067 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1068 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1069 return AVERROR(EPERM);
1075 * Send a command to the RTSP server without waiting for the reply.
1077 * @param s RTSP (de)muxer context
1078 * @param method the method for the request
1079 * @param url the target url for the request
1080 * @param headers extra header lines to include in the request
1081 * @param send_content if non-null, the data to send as request body content
1082 * @param send_content_length the length of the send_content data, or 0 if
1083 * send_content is null
1085 * @return zero if success, nonzero otherwise
1087 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1088 const char *method, const char *url,
1089 const char *headers,
1090 const unsigned char *send_content,
1091 int send_content_length)
1093 RTSPState *rt = s->priv_data;
1094 char buf[4096], *out_buf;
1095 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1097 /* Add in RTSP headers */
1100 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1102 av_strlcat(buf, headers, sizeof(buf));
1103 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1104 if (rt->session_id[0] != '\0' && (!headers ||
1105 !strstr(headers, "\nIf-Match:"))) {
1106 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1109 char *str = ff_http_auth_create_response(&rt->auth_state,
1110 rt->auth, url, method);
1112 av_strlcat(buf, str, sizeof(buf));
1115 if (send_content_length > 0 && send_content)
1116 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1117 av_strlcat(buf, "\r\n", sizeof(buf));
1119 /* base64 encode rtsp if tunneling */
1120 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1121 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1122 out_buf = base64buf;
1125 av_dlog(s, "Sending:\n%s--\n", buf);
1127 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1128 if (send_content_length > 0 && send_content) {
1129 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1130 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1131 "with content data not supported\n");
1132 return AVERROR_PATCHWELCOME;
1134 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1136 rt->last_cmd_time = av_gettime();
1141 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1142 const char *url, const char *headers)
1144 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1147 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1148 const char *headers, RTSPMessageHeader *reply,
1149 unsigned char **content_ptr)
1151 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1152 content_ptr, NULL, 0);
1155 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1156 const char *method, const char *url,
1158 RTSPMessageHeader *reply,
1159 unsigned char **content_ptr,
1160 const unsigned char *send_content,
1161 int send_content_length)
1163 RTSPState *rt = s->priv_data;
1164 HTTPAuthType cur_auth_type;
1165 int ret, attempts = 0;
1168 cur_auth_type = rt->auth_state.auth_type;
1169 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1171 send_content_length)))
1174 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1178 if (reply->status_code == 401 &&
1179 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1180 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1183 if (reply->status_code > 400){
1184 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1188 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1194 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1195 int lower_transport, const char *real_challenge)
1197 RTSPState *rt = s->priv_data;
1198 int rtx = 0, j, i, err, interleave = 0, port_off;
1199 RTSPStream *rtsp_st;
1200 RTSPMessageHeader reply1, *reply = &reply1;
1202 const char *trans_pref;
1204 if (rt->transport == RTSP_TRANSPORT_RDT)
1205 trans_pref = "x-pn-tng";
1206 else if (rt->transport == RTSP_TRANSPORT_RAW)
1207 trans_pref = "RAW/RAW";
1209 trans_pref = "RTP/AVP";
1211 /* default timeout: 1 minute */
1214 /* for each stream, make the setup request */
1215 /* XXX: we assume the same server is used for the control of each
1218 /* Choose a random starting offset within the first half of the
1219 * port range, to allow for a number of ports to try even if the offset
1220 * happens to be at the end of the random range. */
1221 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1222 /* even random offset */
1223 port_off -= port_off & 0x01;
1225 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1226 char transport[2048];
1229 * WMS serves all UDP data over a single connection, the RTX, which
1230 * isn't necessarily the first in the SDP but has to be the first
1231 * to be set up, else the second/third SETUP will fail with a 461.
1233 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1234 rt->server_type == RTSP_SERVER_WMS) {
1237 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1238 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1240 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1244 if (rtx == rt->nb_rtsp_streams)
1245 return -1; /* no RTX found */
1246 rtsp_st = rt->rtsp_streams[rtx];
1248 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1250 rtsp_st = rt->rtsp_streams[i];
1253 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1256 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1257 port = reply->transports[0].client_port_min;
1261 /* first try in specified port range */
1262 while (j <= rt->rtp_port_max) {
1263 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1264 "?localport=%d", j);
1265 /* we will use two ports per rtp stream (rtp and rtcp) */
1267 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1268 &s->interrupt_callback, NULL))
1272 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1277 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1279 snprintf(transport, sizeof(transport) - 1,
1280 "%s/UDP;", trans_pref);
1281 if (rt->server_type != RTSP_SERVER_REAL)
1282 av_strlcat(transport, "unicast;", sizeof(transport));
1283 av_strlcatf(transport, sizeof(transport),
1284 "client_port=%d", port);
1285 if (rt->transport == RTSP_TRANSPORT_RTP &&
1286 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1287 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1291 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1292 /* For WMS streams, the application streams are only used for
1293 * UDP. When trying to set it up for TCP streams, the server
1294 * will return an error. Therefore, we skip those streams. */
1295 if (rt->server_type == RTSP_SERVER_WMS &&
1296 (rtsp_st->stream_index < 0 ||
1297 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1300 snprintf(transport, sizeof(transport) - 1,
1301 "%s/TCP;", trans_pref);
1302 if (rt->transport != RTSP_TRANSPORT_RDT)
1303 av_strlcat(transport, "unicast;", sizeof(transport));
1304 av_strlcatf(transport, sizeof(transport),
1305 "interleaved=%d-%d",
1306 interleave, interleave + 1);
1310 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1311 snprintf(transport, sizeof(transport) - 1,
1312 "%s/UDP;multicast", trans_pref);
1315 av_strlcat(transport, ";mode=record", sizeof(transport));
1316 } else if (rt->server_type == RTSP_SERVER_REAL ||
1317 rt->server_type == RTSP_SERVER_WMS)
1318 av_strlcat(transport, ";mode=play", sizeof(transport));
1319 snprintf(cmd, sizeof(cmd),
1320 "Transport: %s\r\n",
1322 if (rt->accept_dynamic_rate)
1323 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1324 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1325 char real_res[41], real_csum[9];
1326 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1328 av_strlcatf(cmd, sizeof(cmd),
1330 "RealChallenge2: %s, sd=%s\r\n",
1331 rt->session_id, real_res, real_csum);
1333 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1334 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1337 } else if (reply->status_code != RTSP_STATUS_OK ||
1338 reply->nb_transports != 1) {
1339 err = AVERROR_INVALIDDATA;
1343 /* XXX: same protocol for all streams is required */
1345 if (reply->transports[0].lower_transport != rt->lower_transport ||
1346 reply->transports[0].transport != rt->transport) {
1347 err = AVERROR_INVALIDDATA;
1351 rt->lower_transport = reply->transports[0].lower_transport;
1352 rt->transport = reply->transports[0].transport;
1355 /* Fail if the server responded with another lower transport mode
1356 * than what we requested. */
1357 if (reply->transports[0].lower_transport != lower_transport) {
1358 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1359 err = AVERROR_INVALIDDATA;
1363 switch(reply->transports[0].lower_transport) {
1364 case RTSP_LOWER_TRANSPORT_TCP:
1365 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1366 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1369 case RTSP_LOWER_TRANSPORT_UDP: {
1370 char url[1024], options[30] = "";
1372 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1373 av_strlcpy(options, "?connect=1", sizeof(options));
1374 /* Use source address if specified */
1375 if (reply->transports[0].source[0]) {
1376 ff_url_join(url, sizeof(url), "rtp", NULL,
1377 reply->transports[0].source,
1378 reply->transports[0].server_port_min, "%s", options);
1380 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1381 reply->transports[0].server_port_min, "%s", options);
1383 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1384 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1385 err = AVERROR_INVALIDDATA;
1388 /* Try to initialize the connection state in a
1389 * potential NAT router by sending dummy packets.
1390 * RTP/RTCP dummy packets are used for RDT, too.
1392 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1394 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1397 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1398 char url[1024], namebuf[50], optbuf[20] = "";
1399 struct sockaddr_storage addr;
1402 if (reply->transports[0].destination.ss_family) {
1403 addr = reply->transports[0].destination;
1404 port = reply->transports[0].port_min;
1405 ttl = reply->transports[0].ttl;
1407 addr = rtsp_st->sdp_ip;
1408 port = rtsp_st->sdp_port;
1409 ttl = rtsp_st->sdp_ttl;
1412 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1413 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1414 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1415 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1416 port, "%s", optbuf);
1417 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1418 &s->interrupt_callback, NULL) < 0) {
1419 err = AVERROR_INVALIDDATA;
1426 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1430 if (rt->nb_rtsp_streams && reply->timeout > 0)
1431 rt->timeout = reply->timeout;
1433 if (rt->server_type == RTSP_SERVER_REAL)
1434 rt->need_subscription = 1;
1439 ff_rtsp_undo_setup(s);
1443 void ff_rtsp_close_connections(AVFormatContext *s)
1445 RTSPState *rt = s->priv_data;
1446 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1447 ffurl_close(rt->rtsp_hd);
1448 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1451 int ff_rtsp_connect(AVFormatContext *s)
1453 RTSPState *rt = s->priv_data;
1454 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1455 int port, err, tcp_fd;
1456 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1457 int lower_transport_mask = 0;
1458 char real_challenge[64] = "";
1459 struct sockaddr_storage peer;
1460 socklen_t peer_len = sizeof(peer);
1462 if (rt->rtp_port_max < rt->rtp_port_min) {
1463 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1464 "than min port %d\n", rt->rtp_port_max,
1466 return AVERROR(EINVAL);
1469 if (!ff_network_init())
1470 return AVERROR(EIO);
1472 if (s->max_delay < 0) /* Not set by the caller */
1473 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1475 rt->control_transport = RTSP_MODE_PLAIN;
1476 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1477 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1478 rt->control_transport = RTSP_MODE_TUNNEL;
1480 /* Only pass through valid flags from here */
1481 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1484 lower_transport_mask = rt->lower_transport_mask;
1485 /* extract hostname and port */
1486 av_url_split(NULL, 0, auth, sizeof(auth),
1487 host, sizeof(host), &port, path, sizeof(path), s->filename);
1489 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1492 port = RTSP_DEFAULT_PORT;
1494 if (!lower_transport_mask)
1495 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1498 /* Only UDP or TCP - UDP multicast isn't supported. */
1499 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1500 (1 << RTSP_LOWER_TRANSPORT_TCP);
1501 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1502 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1503 "only UDP and TCP are supported for output.\n");
1504 err = AVERROR(EINVAL);
1509 /* Construct the URI used in request; this is similar to s->filename,
1510 * but with authentication credentials removed and RTSP specific options
1512 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1513 host, port, "%s", path);
1515 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1516 /* set up initial handshake for tunneling */
1517 char httpname[1024];
1518 char sessioncookie[17];
1521 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1522 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1523 av_get_random_seed(), av_get_random_seed());
1526 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1527 &s->interrupt_callback) < 0) {
1532 /* generate GET headers */
1533 snprintf(headers, sizeof(headers),
1534 "x-sessioncookie: %s\r\n"
1535 "Accept: application/x-rtsp-tunnelled\r\n"
1536 "Pragma: no-cache\r\n"
1537 "Cache-Control: no-cache\r\n",
1539 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1541 /* complete the connection */
1542 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1548 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1549 &s->interrupt_callback) < 0 ) {
1554 /* generate POST headers */
1555 snprintf(headers, sizeof(headers),
1556 "x-sessioncookie: %s\r\n"
1557 "Content-Type: application/x-rtsp-tunnelled\r\n"
1558 "Pragma: no-cache\r\n"
1559 "Cache-Control: no-cache\r\n"
1560 "Content-Length: 32767\r\n"
1561 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1563 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1564 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1566 /* Initialize the authentication state for the POST session. The HTTP
1567 * protocol implementation doesn't properly handle multi-pass
1568 * authentication for POST requests, since it would require one of
1570 * - implementing Expect: 100-continue, which many HTTP servers
1571 * don't support anyway, even less the RTSP servers that do HTTP
1573 * - sending the whole POST data until getting a 401 reply specifying
1574 * what authentication method to use, then resending all that data
1575 * - waiting for potential 401 replies directly after sending the
1576 * POST header (waiting for some unspecified time)
1577 * Therefore, we copy the full auth state, which works for both basic
1578 * and digest. (For digest, we would have to synchronize the nonce
1579 * count variable between the two sessions, if we'd do more requests
1580 * with the original session, though.)
1582 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1584 /* complete the connection */
1585 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1590 /* open the tcp connection */
1591 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1592 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1593 &s->interrupt_callback, NULL) < 0) {
1597 rt->rtsp_hd_out = rt->rtsp_hd;
1601 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1602 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1603 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1604 NULL, 0, NI_NUMERICHOST);
1607 /* request options supported by the server; this also detects server
1609 for (rt->server_type = RTSP_SERVER_RTP;;) {
1611 if (rt->server_type == RTSP_SERVER_REAL)
1614 * The following entries are required for proper
1615 * streaming from a Realmedia server. They are
1616 * interdependent in some way although we currently
1617 * don't quite understand how. Values were copied
1618 * from mplayer SVN r23589.
1619 * ClientChallenge is a 16-byte ID in hex
1620 * CompanyID is a 16-byte ID in base64
1622 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1623 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1624 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1625 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1627 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1628 if (reply->status_code != RTSP_STATUS_OK) {
1629 err = AVERROR_INVALIDDATA;
1633 /* detect server type if not standard-compliant RTP */
1634 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1635 rt->server_type = RTSP_SERVER_REAL;
1637 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1638 rt->server_type = RTSP_SERVER_WMS;
1639 } else if (rt->server_type == RTSP_SERVER_REAL)
1640 strcpy(real_challenge, reply->real_challenge);
1644 if (s->iformat && CONFIG_RTSP_DEMUXER)
1645 err = ff_rtsp_setup_input_streams(s, reply);
1646 else if (CONFIG_RTSP_MUXER)
1647 err = ff_rtsp_setup_output_streams(s, host);
1652 int lower_transport = ff_log2_tab[lower_transport_mask &
1653 ~(lower_transport_mask - 1)];
1655 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1656 rt->server_type == RTSP_SERVER_REAL ?
1657 real_challenge : NULL);
1660 lower_transport_mask &= ~(1 << lower_transport);
1661 if (lower_transport_mask == 0 && err == 1) {
1662 err = AVERROR(EPROTONOSUPPORT);
1667 rt->lower_transport_mask = lower_transport_mask;
1668 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1669 rt->state = RTSP_STATE_IDLE;
1670 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1673 ff_rtsp_close_streams(s);
1674 ff_rtsp_close_connections(s);
1675 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1676 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1677 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1685 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1688 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1689 uint8_t *buf, int buf_size, int64_t wait_end)
1691 RTSPState *rt = s->priv_data;
1692 RTSPStream *rtsp_st;
1693 int n, i, ret, tcp_fd, timeout_cnt = 0;
1695 struct pollfd *p = rt->p;
1696 int *fds = NULL, fdsnum, fdsidx;
1699 if (ff_check_interrupt(&s->interrupt_callback))
1700 return AVERROR_EXIT;
1701 if (wait_end && wait_end - av_gettime() < 0)
1702 return AVERROR(EAGAIN);
1705 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1706 p[max_p].fd = tcp_fd;
1707 p[max_p++].events = POLLIN;
1711 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1712 rtsp_st = rt->rtsp_streams[i];
1713 if (rtsp_st->rtp_handle) {
1714 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
1716 av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
1720 av_log(s, AV_LOG_ERROR,
1721 "Number of fds %d not supported\n", fdsnum);
1722 return AVERROR_INVALIDDATA;
1724 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
1725 p[max_p].fd = fds[fdsidx];
1726 p[max_p++].events = POLLIN;
1731 n = poll(p, max_p, POLL_TIMEOUT_MS);
1733 int j = 1 - (tcp_fd == -1);
1735 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1736 rtsp_st = rt->rtsp_streams[i];
1737 if (rtsp_st->rtp_handle) {
1738 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1739 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1741 *prtsp_st = rtsp_st;
1748 #if CONFIG_RTSP_DEMUXER
1749 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1750 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
1751 if (rt->state == RTSP_STATE_STREAMING) {
1752 if (!ff_rtsp_parse_streaming_commands(s))
1755 av_log(s, AV_LOG_WARNING,
1756 "Unable to answer to TEARDOWN\n");
1760 RTSPMessageHeader reply;
1761 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1764 /* XXX: parse message */
1765 if (rt->state != RTSP_STATE_STREAMING)
1770 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1771 return AVERROR(ETIMEDOUT);
1772 } else if (n < 0 && errno != EINTR)
1773 return AVERROR(errno);
1777 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1779 RTSPState *rt = s->priv_data;
1781 RTSPStream *rtsp_st, *first_queue_st = NULL;
1782 int64_t wait_end = 0;
1784 if (rt->nb_byes == rt->nb_rtsp_streams)
1787 /* get next frames from the same RTP packet */
1788 if (rt->cur_transport_priv) {
1789 if (rt->transport == RTSP_TRANSPORT_RDT) {
1790 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1791 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1792 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1793 } else if (rt->ts && CONFIG_RTPDEC) {
1794 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);
1796 rt->recvbuf_pos += ret;
1797 ret = rt->recvbuf_pos < rt->recvbuf_len;
1801 rt->cur_transport_priv = NULL;
1803 } else if (ret == 1) {
1806 rt->cur_transport_priv = NULL;
1809 if (rt->transport == RTSP_TRANSPORT_RTP) {
1811 int64_t first_queue_time = 0;
1812 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1813 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1817 queue_time = ff_rtp_queued_packet_time(rtpctx);
1818 if (queue_time && (queue_time - first_queue_time < 0 ||
1819 !first_queue_time)) {
1820 first_queue_time = queue_time;
1821 first_queue_st = rt->rtsp_streams[i];
1824 if (first_queue_time)
1825 wait_end = first_queue_time + s->max_delay;
1828 /* read next RTP packet */
1831 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1833 return AVERROR(ENOMEM);
1836 switch(rt->lower_transport) {
1838 #if CONFIG_RTSP_DEMUXER
1839 case RTSP_LOWER_TRANSPORT_TCP:
1840 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1843 case RTSP_LOWER_TRANSPORT_UDP:
1844 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1845 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1846 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1847 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1850 if (len == AVERROR(EAGAIN) && first_queue_st &&
1851 rt->transport == RTSP_TRANSPORT_RTP) {
1852 rtsp_st = first_queue_st;
1853 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1860 if (rt->transport == RTSP_TRANSPORT_RDT) {
1861 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1862 } else if (rt->transport == RTSP_TRANSPORT_RTP) {
1863 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1865 /* Either bad packet, or a RTCP packet. Check if the
1866 * first_rtcp_ntp_time field was initialized. */
1867 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1868 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1869 /* first_rtcp_ntp_time has been initialized for this stream,
1870 * copy the same value to all other uninitialized streams,
1871 * in order to map their timestamp origin to the same ntp time
1874 AVStream *st = NULL;
1875 if (rtsp_st->stream_index >= 0)
1876 st = s->streams[rtsp_st->stream_index];
1877 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1878 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1879 AVStream *st2 = NULL;
1880 if (rt->rtsp_streams[i]->stream_index >= 0)
1881 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1882 if (rtpctx2 && st && st2 &&
1883 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1884 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1885 rtpctx2->rtcp_ts_offset = av_rescale_q(
1886 rtpctx->rtcp_ts_offset, st->time_base,
1891 if (ret == -RTCP_BYE) {
1894 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1895 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1897 if (rt->nb_byes == rt->nb_rtsp_streams)
1901 } else if (rt->ts && CONFIG_RTPDEC) {
1902 ret = ff_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);
1905 rt->recvbuf_len = len;
1906 rt->recvbuf_pos = ret;
1907 rt->cur_transport_priv = rt->ts;
1914 return AVERROR_INVALIDDATA;
1920 /* more packets may follow, so we save the RTP context */
1921 rt->cur_transport_priv = rtsp_st->transport_priv;
1925 #endif /* CONFIG_RTPDEC */
1927 #if CONFIG_SDP_DEMUXER
1928 static int sdp_probe(AVProbeData *p1)
1930 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1932 /* we look for a line beginning "c=IN IP" */
1933 while (p < p_end && *p != '\0') {
1934 if (p + sizeof("c=IN IP") - 1 < p_end &&
1935 av_strstart(p, "c=IN IP", NULL))
1936 return AVPROBE_SCORE_MAX / 2;
1938 while (p < p_end - 1 && *p != '\n') p++;
1947 static int sdp_read_header(AVFormatContext *s)
1949 RTSPState *rt = s->priv_data;
1950 RTSPStream *rtsp_st;
1955 if (!ff_network_init())
1956 return AVERROR(EIO);
1958 if (s->max_delay < 0) /* Not set by the caller */
1959 s->max_delay = DEFAULT_REORDERING_DELAY;
1961 /* read the whole sdp file */
1962 /* XXX: better loading */
1963 content = av_malloc(SDP_MAX_SIZE);
1964 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1967 return AVERROR_INVALIDDATA;
1969 content[size] ='\0';
1971 err = ff_sdp_parse(s, content);
1975 /* open each RTP stream */
1976 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1978 rtsp_st = rt->rtsp_streams[i];
1980 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1981 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1982 ff_url_join(url, sizeof(url), "rtp", NULL,
1983 namebuf, rtsp_st->sdp_port,
1984 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1986 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1987 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1988 &s->interrupt_callback, NULL) < 0) {
1989 err = AVERROR_INVALIDDATA;
1992 if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))
1997 ff_rtsp_close_streams(s);
2002 static int sdp_read_close(AVFormatContext *s)
2004 ff_rtsp_close_streams(s);
2009 static const AVClass sdp_demuxer_class = {
2010 .class_name = "SDP demuxer",
2011 .item_name = av_default_item_name,
2012 .option = sdp_options,
2013 .version = LIBAVUTIL_VERSION_INT,
2016 AVInputFormat ff_sdp_demuxer = {
2018 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
2019 .priv_data_size = sizeof(RTSPState),
2020 .read_probe = sdp_probe,
2021 .read_header = sdp_read_header,
2022 .read_packet = ff_rtsp_fetch_packet,
2023 .read_close = sdp_read_close,
2024 .priv_class = &sdp_demuxer_class,
2026 #endif /* CONFIG_SDP_DEMUXER */
2028 #if CONFIG_RTP_DEMUXER
2029 static int rtp_probe(AVProbeData *p)
2031 if (av_strstart(p->filename, "rtp:", NULL))
2032 return AVPROBE_SCORE_MAX;
2036 static int rtp_read_header(AVFormatContext *s)
2038 uint8_t recvbuf[1500];
2039 char host[500], sdp[500];
2041 URLContext* in = NULL;
2043 AVCodecContext codec = { 0 };
2044 struct sockaddr_storage addr;
2046 socklen_t addrlen = sizeof(addr);
2047 RTSPState *rt = s->priv_data;
2049 if (!ff_network_init())
2050 return AVERROR(EIO);
2052 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
2053 &s->interrupt_callback, NULL);
2058 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
2059 if (ret == AVERROR(EAGAIN))
2064 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
2068 if ((recvbuf[0] & 0xc0) != 0x80) {
2069 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
2074 if (RTP_PT_IS_RTCP(recvbuf[1]))
2077 payload_type = recvbuf[1] & 0x7f;
2080 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2084 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2085 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2086 "without an SDP file describing it\n",
2090 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2091 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2092 "properly you need an SDP file "
2096 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2097 NULL, 0, s->filename);
2099 snprintf(sdp, sizeof(sdp),
2100 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2101 addr.ss_family == AF_INET ? 4 : 6, host,
2102 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2103 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2104 port, payload_type);
2105 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2107 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2110 /* sdp_read_header initializes this again */
2113 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2115 ret = sdp_read_header(s);
2126 static const AVClass rtp_demuxer_class = {
2127 .class_name = "RTP demuxer",
2128 .item_name = av_default_item_name,
2129 .option = rtp_options,
2130 .version = LIBAVUTIL_VERSION_INT,
2133 AVInputFormat ff_rtp_demuxer = {
2135 .long_name = NULL_IF_CONFIG_SMALL("RTP input"),
2136 .priv_data_size = sizeof(RTSPState),
2137 .read_probe = rtp_probe,
2138 .read_header = rtp_read_header,
2139 .read_packet = ff_rtsp_fetch_packet,
2140 .read_close = sdp_read_close,
2141 .flags = AVFMT_NOFILE,
2142 .priv_class = &rtp_demuxer_class,
2144 #endif /* CONFIG_RTP_DEMUXER */