3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/mathematics.h"
26 #include "libavutil/parseutils.h"
27 #include "libavutil/random_seed.h"
28 #include "libavutil/dict.h"
29 #include "libavutil/opt.h"
31 #include "avio_internal.h"
39 #include "os_support.h"
45 #include "rtpdec_formats.h"
46 #include "rtpenc_chain.h"
52 /* Timeout values for socket poll, in ms,
53 * and read_packet(), in seconds */
54 #define POLL_TIMEOUT_MS 100
55 #define READ_PACKET_TIMEOUT_S 10
56 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
57 #define SDP_MAX_SIZE 16384
58 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
59 #define DEFAULT_REORDERING_DELAY 100000
61 #define OFFSET(x) offsetof(RTSPState, x)
62 #define DEC AV_OPT_FLAG_DECODING_PARAM
63 #define ENC AV_OPT_FLAG_ENCODING_PARAM
65 #define RTSP_FLAG_OPTS(name, longname) \
66 { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
67 { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
69 #define RTSP_MEDIATYPE_OPTS(name, longname) \
70 { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
71 { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
72 { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
73 { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
75 const AVOption ff_rtsp_options[] = {
76 { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
77 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags)
78 { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \
79 { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
80 { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
81 { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
82 { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
83 RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
84 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
85 { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
86 { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
90 static const AVOption sdp_options[] = {
91 RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
92 RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
96 static const AVOption rtp_options[] = {
97 RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
101 static void get_word_until_chars(char *buf, int buf_size,
102 const char *sep, const char **pp)
108 p += strspn(p, SPACE_CHARS);
110 while (!strchr(sep, *p) && *p != '\0') {
111 if ((q - buf) < buf_size - 1)
120 static void get_word_sep(char *buf, int buf_size, const char *sep,
123 if (**pp == '/') (*pp)++;
124 get_word_until_chars(buf, buf_size, sep, pp);
127 static void get_word(char *buf, int buf_size, const char **pp)
129 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
132 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
134 * Used for seeking in the rtp stream.
136 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
140 p += strspn(p, SPACE_CHARS);
141 if (!av_stristart(p, "npt=", &p))
144 *start = AV_NOPTS_VALUE;
145 *end = AV_NOPTS_VALUE;
147 get_word_sep(buf, sizeof(buf), "-", &p);
148 av_parse_time(start, buf, 1);
151 get_word_sep(buf, sizeof(buf), "-", &p);
152 av_parse_time(end, buf, 1);
154 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
155 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
158 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
160 struct addrinfo hints = { 0 }, *ai = NULL;
161 hints.ai_flags = AI_NUMERICHOST;
162 if (getaddrinfo(buf, NULL, &hints, &ai))
164 memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
170 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
171 RTSPStream *rtsp_st, AVCodecContext *codec)
175 codec->codec_id = handler->codec_id;
176 rtsp_st->dynamic_handler = handler;
177 if (handler->alloc) {
178 rtsp_st->dynamic_protocol_context = handler->alloc();
179 if (!rtsp_st->dynamic_protocol_context)
180 rtsp_st->dynamic_handler = NULL;
184 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
185 static int sdp_parse_rtpmap(AVFormatContext *s,
186 AVStream *st, RTSPStream *rtsp_st,
187 int payload_type, const char *p)
189 AVCodecContext *codec = st->codec;
195 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
196 * see if we can handle this kind of payload.
197 * The space should normally not be there but some Real streams or
198 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
199 * have a trailing space. */
200 get_word_sep(buf, sizeof(buf), "/ ", &p);
201 if (payload_type >= RTP_PT_PRIVATE) {
202 RTPDynamicProtocolHandler *handler =
203 ff_rtp_handler_find_by_name(buf, codec->codec_type);
204 init_rtp_handler(handler, rtsp_st, codec);
205 /* If no dynamic handler was found, check with the list of standard
206 * allocated types, if such a stream for some reason happens to
207 * use a private payload type. This isn't handled in rtpdec.c, since
208 * the format name from the rtpmap line never is passed into rtpdec. */
209 if (!rtsp_st->dynamic_handler)
210 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
212 /* We are in a standard case
213 * (from http://www.iana.org/assignments/rtp-parameters). */
214 /* search into AVRtpPayloadTypes[] */
215 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
218 c = avcodec_find_decoder(codec->codec_id);
224 get_word_sep(buf, sizeof(buf), "/", &p);
226 switch (codec->codec_type) {
227 case AVMEDIA_TYPE_AUDIO:
228 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
229 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
230 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
232 codec->sample_rate = i;
233 avpriv_set_pts_info(st, 32, 1, codec->sample_rate);
234 get_word_sep(buf, sizeof(buf), "/", &p);
238 // TODO: there is a bug here; if it is a mono stream, and
239 // less than 22000Hz, faad upconverts to stereo and twice
240 // the frequency. No problem, but the sample rate is being
241 // set here by the sdp line. Patch on its way. (rdm)
243 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
245 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
248 case AVMEDIA_TYPE_VIDEO:
249 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
251 avpriv_set_pts_info(st, 32, 1, i);
256 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
257 rtsp_st->dynamic_handler->init(s, st->index,
258 rtsp_st->dynamic_protocol_context);
262 /* parse the attribute line from the fmtp a line of an sdp response. This
263 * is broken out as a function because it is used in rtp_h264.c, which is
265 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
266 char *value, int value_size)
268 *p += strspn(*p, SPACE_CHARS);
270 get_word_sep(attr, attr_size, "=", p);
273 get_word_sep(value, value_size, ";", p);
281 typedef struct SDPParseState {
283 struct sockaddr_storage default_ip;
285 int skip_media; ///< set if an unknown m= line occurs
288 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
289 int letter, const char *buf)
291 RTSPState *rt = s->priv_data;
292 char buf1[64], st_type[64];
294 enum AVMediaType codec_type;
298 struct sockaddr_storage sdp_ip;
301 av_dlog(s, "sdp: %c='%s'\n", letter, buf);
304 if (s1->skip_media && letter != 'm')
308 get_word(buf1, sizeof(buf1), &p);
309 if (strcmp(buf1, "IN") != 0)
311 get_word(buf1, sizeof(buf1), &p);
312 if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6"))
314 get_word_sep(buf1, sizeof(buf1), "/", &p);
315 if (get_sockaddr(buf1, &sdp_ip))
320 get_word_sep(buf1, sizeof(buf1), "/", &p);
323 if (s->nb_streams == 0) {
324 s1->default_ip = sdp_ip;
325 s1->default_ttl = ttl;
327 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
328 rtsp_st->sdp_ip = sdp_ip;
329 rtsp_st->sdp_ttl = ttl;
333 av_dict_set(&s->metadata, "title", p, 0);
336 if (s->nb_streams == 0) {
337 av_dict_set(&s->metadata, "comment", p, 0);
344 codec_type = AVMEDIA_TYPE_UNKNOWN;
345 get_word(st_type, sizeof(st_type), &p);
346 if (!strcmp(st_type, "audio")) {
347 codec_type = AVMEDIA_TYPE_AUDIO;
348 } else if (!strcmp(st_type, "video")) {
349 codec_type = AVMEDIA_TYPE_VIDEO;
350 } else if (!strcmp(st_type, "application")) {
351 codec_type = AVMEDIA_TYPE_DATA;
353 if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
357 rtsp_st = av_mallocz(sizeof(RTSPStream));
360 rtsp_st->stream_index = -1;
361 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
363 rtsp_st->sdp_ip = s1->default_ip;
364 rtsp_st->sdp_ttl = s1->default_ttl;
366 get_word(buf1, sizeof(buf1), &p); /* port */
367 rtsp_st->sdp_port = atoi(buf1);
369 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
371 /* XXX: handle list of formats */
372 get_word(buf1, sizeof(buf1), &p); /* format list */
373 rtsp_st->sdp_payload_type = atoi(buf1);
375 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
376 /* no corresponding stream */
377 } else if (rt->server_type == RTSP_SERVER_WMS &&
378 codec_type == AVMEDIA_TYPE_DATA) {
379 /* RTX stream, a stream that carries all the other actual
380 * audio/video streams. Don't expose this to the callers. */
382 st = avformat_new_stream(s, NULL);
385 st->id = rt->nb_rtsp_streams - 1;
386 rtsp_st->stream_index = st->index;
387 st->codec->codec_type = codec_type;
388 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
389 RTPDynamicProtocolHandler *handler;
390 /* if standard payload type, we can find the codec right now */
391 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
392 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
393 st->codec->sample_rate > 0)
394 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
395 /* Even static payload types may need a custom depacketizer */
396 handler = ff_rtp_handler_find_by_id(
397 rtsp_st->sdp_payload_type, st->codec->codec_type);
398 init_rtp_handler(handler, rtsp_st, st->codec);
399 if (handler && handler->init)
400 handler->init(s, st->index,
401 rtsp_st->dynamic_protocol_context);
404 /* put a default control url */
405 av_strlcpy(rtsp_st->control_url, rt->control_uri,
406 sizeof(rtsp_st->control_url));
409 if (av_strstart(p, "control:", &p)) {
410 if (s->nb_streams == 0) {
411 if (!strncmp(p, "rtsp://", 7))
412 av_strlcpy(rt->control_uri, p,
413 sizeof(rt->control_uri));
416 /* get the control url */
417 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
419 /* XXX: may need to add full url resolution */
420 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
422 if (proto[0] == '\0') {
423 /* relative control URL */
424 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
425 av_strlcat(rtsp_st->control_url, "/",
426 sizeof(rtsp_st->control_url));
427 av_strlcat(rtsp_st->control_url, p,
428 sizeof(rtsp_st->control_url));
430 av_strlcpy(rtsp_st->control_url, p,
431 sizeof(rtsp_st->control_url));
433 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
434 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
435 get_word(buf1, sizeof(buf1), &p);
436 payload_type = atoi(buf1);
437 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
438 if (rtsp_st->stream_index >= 0) {
439 st = s->streams[rtsp_st->stream_index];
440 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
442 } else if (av_strstart(p, "fmtp:", &p) ||
443 av_strstart(p, "framesize:", &p)) {
444 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
445 // let dynamic protocol handlers have a stab at the line.
446 get_word(buf1, sizeof(buf1), &p);
447 payload_type = atoi(buf1);
448 for (i = 0; i < rt->nb_rtsp_streams; i++) {
449 rtsp_st = rt->rtsp_streams[i];
450 if (rtsp_st->sdp_payload_type == payload_type &&
451 rtsp_st->dynamic_handler &&
452 rtsp_st->dynamic_handler->parse_sdp_a_line)
453 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
454 rtsp_st->dynamic_protocol_context, buf);
456 } else if (av_strstart(p, "range:", &p)) {
459 // this is so that seeking on a streamed file can work.
460 rtsp_parse_range_npt(p, &start, &end);
461 s->start_time = start;
462 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
463 s->duration = (end == AV_NOPTS_VALUE) ?
464 AV_NOPTS_VALUE : end - start;
465 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
467 rt->transport = RTSP_TRANSPORT_RDT;
468 } else if (av_strstart(p, "SampleRate:integer;", &p) &&
470 st = s->streams[s->nb_streams - 1];
471 st->codec->sample_rate = atoi(p);
473 if (rt->server_type == RTSP_SERVER_WMS)
474 ff_wms_parse_sdp_a_line(s, p);
475 if (s->nb_streams > 0) {
476 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
478 if (rt->server_type == RTSP_SERVER_REAL)
479 ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
481 if (rtsp_st->dynamic_handler &&
482 rtsp_st->dynamic_handler->parse_sdp_a_line)
483 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
484 rtsp_st->stream_index,
485 rtsp_st->dynamic_protocol_context, buf);
492 int ff_sdp_parse(AVFormatContext *s, const char *content)
494 RTSPState *rt = s->priv_data;
497 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
498 * contain long SDP lines containing complete ASF Headers (several
499 * kB) or arrays of MDPR (RM stream descriptor) headers plus
500 * "rulebooks" describing their properties. Therefore, the SDP line
503 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
504 * in rtpdec_xiph.c. */
506 SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state;
510 p += strspn(p, SPACE_CHARS);
518 /* get the content */
520 while (*p != '\n' && *p != '\r' && *p != '\0') {
521 if ((q - buf) < sizeof(buf) - 1)
526 sdp_parse_line(s, s1, letter, buf);
528 while (*p != '\n' && *p != '\0')
533 rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1));
534 if (!rt->p) return AVERROR(ENOMEM);
537 #endif /* CONFIG_RTPDEC */
539 void ff_rtsp_undo_setup(AVFormatContext *s)
541 RTSPState *rt = s->priv_data;
544 for (i = 0; i < rt->nb_rtsp_streams; i++) {
545 RTSPStream *rtsp_st = rt->rtsp_streams[i];
548 if (rtsp_st->transport_priv) {
550 AVFormatContext *rtpctx = rtsp_st->transport_priv;
551 av_write_trailer(rtpctx);
552 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
554 avio_close_dyn_buf(rtpctx->pb, &ptr);
557 avio_close(rtpctx->pb);
559 avformat_free_context(rtpctx);
560 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
561 ff_rdt_parse_close(rtsp_st->transport_priv);
562 else if (CONFIG_RTPDEC)
563 ff_rtp_parse_close(rtsp_st->transport_priv);
565 rtsp_st->transport_priv = NULL;
566 if (rtsp_st->rtp_handle)
567 ffurl_close(rtsp_st->rtp_handle);
568 rtsp_st->rtp_handle = NULL;
572 /* close and free RTSP streams */
573 void ff_rtsp_close_streams(AVFormatContext *s)
575 RTSPState *rt = s->priv_data;
579 ff_rtsp_undo_setup(s);
580 for (i = 0; i < rt->nb_rtsp_streams; i++) {
581 rtsp_st = rt->rtsp_streams[i];
583 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
584 rtsp_st->dynamic_handler->free(
585 rtsp_st->dynamic_protocol_context);
589 av_free(rt->rtsp_streams);
591 avformat_close_input(&rt->asf_ctx);
594 av_free(rt->recvbuf);
597 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
599 RTSPState *rt = s->priv_data;
602 /* open the RTP context */
603 if (rtsp_st->stream_index >= 0)
604 st = s->streams[rtsp_st->stream_index];
606 s->ctx_flags |= AVFMTCTX_NOHEADER;
608 if (s->oformat && CONFIG_RTSP_MUXER) {
609 rtsp_st->transport_priv = ff_rtp_chain_mux_open(s, st,
611 RTSP_TCP_MAX_PACKET_SIZE);
612 /* Ownership of rtp_handle is passed to the rtp mux context */
613 rtsp_st->rtp_handle = NULL;
614 } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
615 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
616 rtsp_st->dynamic_protocol_context,
617 rtsp_st->dynamic_handler);
618 else if (CONFIG_RTPDEC)
619 rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
620 rtsp_st->sdp_payload_type,
621 (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
622 ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
624 if (!rtsp_st->transport_priv) {
625 return AVERROR(ENOMEM);
626 } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) {
627 if (rtsp_st->dynamic_handler) {
628 ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
629 rtsp_st->dynamic_protocol_context,
630 rtsp_st->dynamic_handler);
637 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
638 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
645 q += strspn(q, SPACE_CHARS);
646 v = strtol(q, &p, 10);
650 v = strtol(p, &p, 10);
659 /* XXX: only one transport specification is parsed */
660 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
662 char transport_protocol[16];
664 char lower_transport[16];
666 RTSPTransportField *th;
669 reply->nb_transports = 0;
672 p += strspn(p, SPACE_CHARS);
676 th = &reply->transports[reply->nb_transports];
678 get_word_sep(transport_protocol, sizeof(transport_protocol),
680 if (!av_strcasecmp (transport_protocol, "rtp")) {
681 get_word_sep(profile, sizeof(profile), "/;,", &p);
682 lower_transport[0] = '\0';
683 /* rtp/avp/<protocol> */
685 get_word_sep(lower_transport, sizeof(lower_transport),
688 th->transport = RTSP_TRANSPORT_RTP;
689 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
690 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
691 /* x-pn-tng/<protocol> */
692 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
694 th->transport = RTSP_TRANSPORT_RDT;
696 if (!av_strcasecmp(lower_transport, "TCP"))
697 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
699 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
703 /* get each parameter */
704 while (*p != '\0' && *p != ',') {
705 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
706 if (!strcmp(parameter, "port")) {
709 rtsp_parse_range(&th->port_min, &th->port_max, &p);
711 } else if (!strcmp(parameter, "client_port")) {
714 rtsp_parse_range(&th->client_port_min,
715 &th->client_port_max, &p);
717 } else if (!strcmp(parameter, "server_port")) {
720 rtsp_parse_range(&th->server_port_min,
721 &th->server_port_max, &p);
723 } else if (!strcmp(parameter, "interleaved")) {
726 rtsp_parse_range(&th->interleaved_min,
727 &th->interleaved_max, &p);
729 } else if (!strcmp(parameter, "multicast")) {
730 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
731 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
732 } else if (!strcmp(parameter, "ttl")) {
735 th->ttl = strtol(p, (char **)&p, 10);
737 } else if (!strcmp(parameter, "destination")) {
740 get_word_sep(buf, sizeof(buf), ";,", &p);
741 get_sockaddr(buf, &th->destination);
743 } else if (!strcmp(parameter, "source")) {
746 get_word_sep(buf, sizeof(buf), ";,", &p);
747 av_strlcpy(th->source, buf, sizeof(th->source));
751 while (*p != ';' && *p != '\0' && *p != ',')
759 reply->nb_transports++;
763 static void handle_rtp_info(RTSPState *rt, const char *url,
764 uint32_t seq, uint32_t rtptime)
767 if (!rtptime || !url[0])
769 if (rt->transport != RTSP_TRANSPORT_RTP)
771 for (i = 0; i < rt->nb_rtsp_streams; i++) {
772 RTSPStream *rtsp_st = rt->rtsp_streams[i];
773 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
776 if (!strcmp(rtsp_st->control_url, url)) {
777 rtpctx->base_timestamp = rtptime;
783 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
786 char key[20], value[1024], url[1024] = "";
787 uint32_t seq = 0, rtptime = 0;
790 p += strspn(p, SPACE_CHARS);
793 get_word_sep(key, sizeof(key), "=", &p);
797 get_word_sep(value, sizeof(value), ";, ", &p);
799 if (!strcmp(key, "url"))
800 av_strlcpy(url, value, sizeof(url));
801 else if (!strcmp(key, "seq"))
802 seq = strtoul(value, NULL, 10);
803 else if (!strcmp(key, "rtptime"))
804 rtptime = strtoul(value, NULL, 10);
806 handle_rtp_info(rt, url, seq, rtptime);
815 handle_rtp_info(rt, url, seq, rtptime);
818 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
819 RTSPState *rt, const char *method)
823 /* NOTE: we do case independent match for broken servers */
825 if (av_stristart(p, "Session:", &p)) {
827 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
828 if (av_stristart(p, ";timeout=", &p) &&
829 (t = strtol(p, NULL, 10)) > 0) {
832 } else if (av_stristart(p, "Content-Length:", &p)) {
833 reply->content_length = strtol(p, NULL, 10);
834 } else if (av_stristart(p, "Transport:", &p)) {
835 rtsp_parse_transport(reply, p);
836 } else if (av_stristart(p, "CSeq:", &p)) {
837 reply->seq = strtol(p, NULL, 10);
838 } else if (av_stristart(p, "Range:", &p)) {
839 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
840 } else if (av_stristart(p, "RealChallenge1:", &p)) {
841 p += strspn(p, SPACE_CHARS);
842 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
843 } else if (av_stristart(p, "Server:", &p)) {
844 p += strspn(p, SPACE_CHARS);
845 av_strlcpy(reply->server, p, sizeof(reply->server));
846 } else if (av_stristart(p, "Notice:", &p) ||
847 av_stristart(p, "X-Notice:", &p)) {
848 reply->notice = strtol(p, NULL, 10);
849 } else if (av_stristart(p, "Location:", &p)) {
850 p += strspn(p, SPACE_CHARS);
851 av_strlcpy(reply->location, p , sizeof(reply->location));
852 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
853 p += strspn(p, SPACE_CHARS);
854 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
855 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
856 p += strspn(p, SPACE_CHARS);
857 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
858 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
859 p += strspn(p, SPACE_CHARS);
860 if (method && !strcmp(method, "DESCRIBE"))
861 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
862 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
863 p += strspn(p, SPACE_CHARS);
864 if (method && !strcmp(method, "PLAY"))
865 rtsp_parse_rtp_info(rt, p);
866 } else if (av_stristart(p, "Public:", &p) && rt) {
867 if (strstr(p, "GET_PARAMETER") &&
868 method && !strcmp(method, "OPTIONS"))
869 rt->get_parameter_supported = 1;
870 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
871 p += strspn(p, SPACE_CHARS);
872 rt->accept_dynamic_rate = atoi(p);
873 } else if (av_stristart(p, "Content-Type:", &p)) {
874 p += strspn(p, SPACE_CHARS);
875 av_strlcpy(reply->content_type, p, sizeof(reply->content_type));
879 /* skip a RTP/TCP interleaved packet */
880 void ff_rtsp_skip_packet(AVFormatContext *s)
882 RTSPState *rt = s->priv_data;
886 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
889 len = AV_RB16(buf + 1);
891 av_dlog(s, "skipping RTP packet len=%d\n", len);
896 if (len1 > sizeof(buf))
898 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
905 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
906 unsigned char **content_ptr,
907 int return_on_interleaved_data, const char *method)
909 RTSPState *rt = s->priv_data;
910 char buf[4096], buf1[1024], *q;
913 int ret, content_length, line_count = 0, request = 0;
914 unsigned char *content = NULL;
920 memset(reply, 0, sizeof(*reply));
922 /* parse reply (XXX: use buffers) */
923 rt->last_reply[0] = '\0';
927 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
928 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
934 /* XXX: only parse it if first char on line ? */
935 if (return_on_interleaved_data) {
938 ff_rtsp_skip_packet(s);
939 } else if (ch != '\r') {
940 if ((q - buf) < sizeof(buf) - 1)
946 av_dlog(s, "line='%s'\n", buf);
948 /* test if last line */
952 if (line_count == 0) {
954 get_word(buf1, sizeof(buf1), &p);
955 if (!strncmp(buf1, "RTSP/", 5)) {
956 get_word(buf1, sizeof(buf1), &p);
957 reply->status_code = atoi(buf1);
958 av_strlcpy(reply->reason, p, sizeof(reply->reason));
960 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
961 get_word(buf1, sizeof(buf1), &p); // object
965 ff_rtsp_parse_line(reply, p, rt, method);
966 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
967 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
972 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
973 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
975 content_length = reply->content_length;
976 if (content_length > 0) {
977 /* leave some room for a trailing '\0' (useful for simple parsing) */
978 content = av_malloc(content_length + 1);
979 ffurl_read_complete(rt->rtsp_hd, content, content_length);
980 content[content_length] = '\0';
983 *content_ptr = content;
989 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
990 const char* ptr = buf;
992 if (!strcmp(reply->reason, "OPTIONS")) {
993 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
995 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
996 if (reply->session_id[0])
997 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
1000 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
1002 av_strlcat(buf, "\r\n", sizeof(buf));
1004 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1005 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1008 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1010 rt->last_cmd_time = av_gettime();
1011 /* Even if the request from the server had data, it is not the data
1012 * that the caller wants or expects. The memory could also be leaked
1013 * if the actual following reply has content data. */
1015 av_freep(content_ptr);
1016 /* If method is set, this is called from ff_rtsp_send_cmd,
1017 * where a reply to exactly this request is awaited. For
1018 * callers from within packet receiving, we just want to
1019 * return to the caller and go back to receiving packets. */
1025 if (rt->seq != reply->seq) {
1026 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1027 rt->seq, reply->seq);
1031 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1032 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1033 reply->notice == 2306 /* Continuous Feed Terminated */) {
1034 rt->state = RTSP_STATE_IDLE;
1035 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1036 return AVERROR(EIO); /* data or server error */
1037 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1038 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1039 return AVERROR(EPERM);
1045 * Send a command to the RTSP server without waiting for the reply.
1047 * @param s RTSP (de)muxer context
1048 * @param method the method for the request
1049 * @param url the target url for the request
1050 * @param headers extra header lines to include in the request
1051 * @param send_content if non-null, the data to send as request body content
1052 * @param send_content_length the length of the send_content data, or 0 if
1053 * send_content is null
1055 * @return zero if success, nonzero otherwise
1057 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1058 const char *method, const char *url,
1059 const char *headers,
1060 const unsigned char *send_content,
1061 int send_content_length)
1063 RTSPState *rt = s->priv_data;
1064 char buf[4096], *out_buf;
1065 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1067 /* Add in RTSP headers */
1070 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1072 av_strlcat(buf, headers, sizeof(buf));
1073 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1074 if (rt->session_id[0] != '\0' && (!headers ||
1075 !strstr(headers, "\nIf-Match:"))) {
1076 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1079 char *str = ff_http_auth_create_response(&rt->auth_state,
1080 rt->auth, url, method);
1082 av_strlcat(buf, str, sizeof(buf));
1085 if (send_content_length > 0 && send_content)
1086 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1087 av_strlcat(buf, "\r\n", sizeof(buf));
1089 /* base64 encode rtsp if tunneling */
1090 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1091 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1092 out_buf = base64buf;
1095 av_dlog(s, "Sending:\n%s--\n", buf);
1097 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1098 if (send_content_length > 0 && send_content) {
1099 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1100 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1101 "with content data not supported\n");
1102 return AVERROR_PATCHWELCOME;
1104 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1106 rt->last_cmd_time = av_gettime();
1111 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1112 const char *url, const char *headers)
1114 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1117 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1118 const char *headers, RTSPMessageHeader *reply,
1119 unsigned char **content_ptr)
1121 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1122 content_ptr, NULL, 0);
1125 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1126 const char *method, const char *url,
1128 RTSPMessageHeader *reply,
1129 unsigned char **content_ptr,
1130 const unsigned char *send_content,
1131 int send_content_length)
1133 RTSPState *rt = s->priv_data;
1134 HTTPAuthType cur_auth_type;
1135 int ret, attempts = 0;
1138 cur_auth_type = rt->auth_state.auth_type;
1139 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1141 send_content_length)))
1144 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1148 if (reply->status_code == 401 &&
1149 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1150 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1153 if (reply->status_code > 400){
1154 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1158 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1164 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1165 int lower_transport, const char *real_challenge)
1167 RTSPState *rt = s->priv_data;
1168 int rtx = 0, j, i, err, interleave = 0, port_off;
1169 RTSPStream *rtsp_st;
1170 RTSPMessageHeader reply1, *reply = &reply1;
1172 const char *trans_pref;
1174 if (rt->transport == RTSP_TRANSPORT_RDT)
1175 trans_pref = "x-pn-tng";
1177 trans_pref = "RTP/AVP";
1179 /* default timeout: 1 minute */
1182 /* Choose a random starting offset within the first half of the
1183 * port range, to allow for a number of ports to try even if the offset
1184 * happens to be at the end of the random range. */
1185 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1186 /* even random offset */
1187 port_off -= port_off & 0x01;
1189 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1190 char transport[2048];
1193 * WMS serves all UDP data over a single connection, the RTX, which
1194 * isn't necessarily the first in the SDP but has to be the first
1195 * to be set up, else the second/third SETUP will fail with a 461.
1197 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1198 rt->server_type == RTSP_SERVER_WMS) {
1201 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1202 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1204 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1208 if (rtx == rt->nb_rtsp_streams)
1209 return -1; /* no RTX found */
1210 rtsp_st = rt->rtsp_streams[rtx];
1212 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1214 rtsp_st = rt->rtsp_streams[i];
1217 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1220 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1221 port = reply->transports[0].client_port_min;
1225 /* first try in specified port range */
1226 while (j <= rt->rtp_port_max) {
1227 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1228 "?localport=%d", j);
1229 /* we will use two ports per rtp stream (rtp and rtcp) */
1231 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1232 &s->interrupt_callback, NULL))
1235 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1240 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1242 snprintf(transport, sizeof(transport) - 1,
1243 "%s/UDP;", trans_pref);
1244 if (rt->server_type != RTSP_SERVER_REAL)
1245 av_strlcat(transport, "unicast;", sizeof(transport));
1246 av_strlcatf(transport, sizeof(transport),
1247 "client_port=%d", port);
1248 if (rt->transport == RTSP_TRANSPORT_RTP &&
1249 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1250 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1254 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1255 /* For WMS streams, the application streams are only used for
1256 * UDP. When trying to set it up for TCP streams, the server
1257 * will return an error. Therefore, we skip those streams. */
1258 if (rt->server_type == RTSP_SERVER_WMS &&
1259 (rtsp_st->stream_index < 0 ||
1260 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1263 snprintf(transport, sizeof(transport) - 1,
1264 "%s/TCP;", trans_pref);
1265 if (rt->transport != RTSP_TRANSPORT_RDT)
1266 av_strlcat(transport, "unicast;", sizeof(transport));
1267 av_strlcatf(transport, sizeof(transport),
1268 "interleaved=%d-%d",
1269 interleave, interleave + 1);
1273 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1274 snprintf(transport, sizeof(transport) - 1,
1275 "%s/UDP;multicast", trans_pref);
1278 av_strlcat(transport, ";mode=receive", sizeof(transport));
1279 } else if (rt->server_type == RTSP_SERVER_REAL ||
1280 rt->server_type == RTSP_SERVER_WMS)
1281 av_strlcat(transport, ";mode=play", sizeof(transport));
1282 snprintf(cmd, sizeof(cmd),
1283 "Transport: %s\r\n",
1285 if (rt->accept_dynamic_rate)
1286 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1287 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1288 char real_res[41], real_csum[9];
1289 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1291 av_strlcatf(cmd, sizeof(cmd),
1293 "RealChallenge2: %s, sd=%s\r\n",
1294 rt->session_id, real_res, real_csum);
1296 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1297 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1300 } else if (reply->status_code != RTSP_STATUS_OK ||
1301 reply->nb_transports != 1) {
1302 err = AVERROR_INVALIDDATA;
1306 /* XXX: same protocol for all streams is required */
1308 if (reply->transports[0].lower_transport != rt->lower_transport ||
1309 reply->transports[0].transport != rt->transport) {
1310 err = AVERROR_INVALIDDATA;
1314 rt->lower_transport = reply->transports[0].lower_transport;
1315 rt->transport = reply->transports[0].transport;
1318 /* Fail if the server responded with another lower transport mode
1319 * than what we requested. */
1320 if (reply->transports[0].lower_transport != lower_transport) {
1321 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1322 err = AVERROR_INVALIDDATA;
1326 switch(reply->transports[0].lower_transport) {
1327 case RTSP_LOWER_TRANSPORT_TCP:
1328 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1329 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1332 case RTSP_LOWER_TRANSPORT_UDP: {
1333 char url[1024], options[30] = "";
1335 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1336 av_strlcpy(options, "?connect=1", sizeof(options));
1337 /* Use source address if specified */
1338 if (reply->transports[0].source[0]) {
1339 ff_url_join(url, sizeof(url), "rtp", NULL,
1340 reply->transports[0].source,
1341 reply->transports[0].server_port_min, "%s", options);
1343 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1344 reply->transports[0].server_port_min, "%s", options);
1346 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1347 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1348 err = AVERROR_INVALIDDATA;
1351 /* Try to initialize the connection state in a
1352 * potential NAT router by sending dummy packets.
1353 * RTP/RTCP dummy packets are used for RDT, too.
1355 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1357 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1360 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1361 char url[1024], namebuf[50], optbuf[20] = "";
1362 struct sockaddr_storage addr;
1365 if (reply->transports[0].destination.ss_family) {
1366 addr = reply->transports[0].destination;
1367 port = reply->transports[0].port_min;
1368 ttl = reply->transports[0].ttl;
1370 addr = rtsp_st->sdp_ip;
1371 port = rtsp_st->sdp_port;
1372 ttl = rtsp_st->sdp_ttl;
1375 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1376 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1377 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1378 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1379 port, "%s", optbuf);
1380 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1381 &s->interrupt_callback, NULL) < 0) {
1382 err = AVERROR_INVALIDDATA;
1389 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1393 if (rt->nb_rtsp_streams && reply->timeout > 0)
1394 rt->timeout = reply->timeout;
1396 if (rt->server_type == RTSP_SERVER_REAL)
1397 rt->need_subscription = 1;
1402 ff_rtsp_undo_setup(s);
1406 void ff_rtsp_close_connections(AVFormatContext *s)
1408 RTSPState *rt = s->priv_data;
1409 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1410 ffurl_close(rt->rtsp_hd);
1411 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1414 int ff_rtsp_connect(AVFormatContext *s)
1416 RTSPState *rt = s->priv_data;
1417 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1418 int port, err, tcp_fd;
1419 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1420 int lower_transport_mask = 0;
1421 char real_challenge[64] = "";
1422 struct sockaddr_storage peer;
1423 socklen_t peer_len = sizeof(peer);
1425 if (rt->rtp_port_max < rt->rtp_port_min) {
1426 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1427 "than min port %d\n", rt->rtp_port_max,
1429 return AVERROR(EINVAL);
1432 if (!ff_network_init())
1433 return AVERROR(EIO);
1435 if (s->max_delay < 0) /* Not set by the caller */
1436 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1438 rt->control_transport = RTSP_MODE_PLAIN;
1439 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1440 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1441 rt->control_transport = RTSP_MODE_TUNNEL;
1443 /* Only pass through valid flags from here */
1444 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1447 lower_transport_mask = rt->lower_transport_mask;
1448 /* extract hostname and port */
1449 av_url_split(NULL, 0, auth, sizeof(auth),
1450 host, sizeof(host), &port, path, sizeof(path), s->filename);
1452 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1455 port = RTSP_DEFAULT_PORT;
1457 if (!lower_transport_mask)
1458 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1461 /* Only UDP or TCP - UDP multicast isn't supported. */
1462 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1463 (1 << RTSP_LOWER_TRANSPORT_TCP);
1464 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1465 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1466 "only UDP and TCP are supported for output.\n");
1467 err = AVERROR(EINVAL);
1472 /* Construct the URI used in request; this is similar to s->filename,
1473 * but with authentication credentials removed and RTSP specific options
1475 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1476 host, port, "%s", path);
1478 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1479 /* set up initial handshake for tunneling */
1480 char httpname[1024];
1481 char sessioncookie[17];
1484 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1485 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1486 av_get_random_seed(), av_get_random_seed());
1489 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1490 &s->interrupt_callback) < 0) {
1495 /* generate GET headers */
1496 snprintf(headers, sizeof(headers),
1497 "x-sessioncookie: %s\r\n"
1498 "Accept: application/x-rtsp-tunnelled\r\n"
1499 "Pragma: no-cache\r\n"
1500 "Cache-Control: no-cache\r\n",
1502 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1504 /* complete the connection */
1505 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1511 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1512 &s->interrupt_callback) < 0 ) {
1517 /* generate POST headers */
1518 snprintf(headers, sizeof(headers),
1519 "x-sessioncookie: %s\r\n"
1520 "Content-Type: application/x-rtsp-tunnelled\r\n"
1521 "Pragma: no-cache\r\n"
1522 "Cache-Control: no-cache\r\n"
1523 "Content-Length: 32767\r\n"
1524 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1526 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1527 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1529 /* Initialize the authentication state for the POST session. The HTTP
1530 * protocol implementation doesn't properly handle multi-pass
1531 * authentication for POST requests, since it would require one of
1533 * - implementing Expect: 100-continue, which many HTTP servers
1534 * don't support anyway, even less the RTSP servers that do HTTP
1536 * - sending the whole POST data until getting a 401 reply specifying
1537 * what authentication method to use, then resending all that data
1538 * - waiting for potential 401 replies directly after sending the
1539 * POST header (waiting for some unspecified time)
1540 * Therefore, we copy the full auth state, which works for both basic
1541 * and digest. (For digest, we would have to synchronize the nonce
1542 * count variable between the two sessions, if we'd do more requests
1543 * with the original session, though.)
1545 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1547 /* complete the connection */
1548 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1553 /* open the tcp connection */
1554 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1555 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1556 &s->interrupt_callback, NULL) < 0) {
1560 rt->rtsp_hd_out = rt->rtsp_hd;
1564 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1565 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1566 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1567 NULL, 0, NI_NUMERICHOST);
1570 /* request options supported by the server; this also detects server
1572 for (rt->server_type = RTSP_SERVER_RTP;;) {
1574 if (rt->server_type == RTSP_SERVER_REAL)
1577 * The following entries are required for proper
1578 * streaming from a Realmedia server. They are
1579 * interdependent in some way although we currently
1580 * don't quite understand how. Values were copied
1581 * from mplayer SVN r23589.
1582 * ClientChallenge is a 16-byte ID in hex
1583 * CompanyID is a 16-byte ID in base64
1585 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1586 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1587 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1588 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1590 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1591 if (reply->status_code != RTSP_STATUS_OK) {
1592 err = AVERROR_INVALIDDATA;
1596 /* detect server type if not standard-compliant RTP */
1597 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1598 rt->server_type = RTSP_SERVER_REAL;
1600 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1601 rt->server_type = RTSP_SERVER_WMS;
1602 } else if (rt->server_type == RTSP_SERVER_REAL)
1603 strcpy(real_challenge, reply->real_challenge);
1607 if (s->iformat && CONFIG_RTSP_DEMUXER)
1608 err = ff_rtsp_setup_input_streams(s, reply);
1609 else if (CONFIG_RTSP_MUXER)
1610 err = ff_rtsp_setup_output_streams(s, host);
1615 int lower_transport = ff_log2_tab[lower_transport_mask &
1616 ~(lower_transport_mask - 1)];
1618 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1619 rt->server_type == RTSP_SERVER_REAL ?
1620 real_challenge : NULL);
1623 lower_transport_mask &= ~(1 << lower_transport);
1624 if (lower_transport_mask == 0 && err == 1) {
1625 err = AVERROR(EPROTONOSUPPORT);
1630 rt->lower_transport_mask = lower_transport_mask;
1631 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1632 rt->state = RTSP_STATE_IDLE;
1633 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1636 ff_rtsp_close_streams(s);
1637 ff_rtsp_close_connections(s);
1638 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1639 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1640 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1648 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1651 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1652 uint8_t *buf, int buf_size, int64_t wait_end)
1654 RTSPState *rt = s->priv_data;
1655 RTSPStream *rtsp_st;
1656 int n, i, ret, tcp_fd, timeout_cnt = 0;
1658 struct pollfd *p = rt->p;
1661 if (ff_check_interrupt(&s->interrupt_callback))
1662 return AVERROR_EXIT;
1663 if (wait_end && wait_end - av_gettime() < 0)
1664 return AVERROR(EAGAIN);
1667 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1668 p[max_p].fd = tcp_fd;
1669 p[max_p++].events = POLLIN;
1673 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1674 rtsp_st = rt->rtsp_streams[i];
1675 if (rtsp_st->rtp_handle) {
1676 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1677 p[max_p++].events = POLLIN;
1678 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1679 p[max_p++].events = POLLIN;
1682 n = poll(p, max_p, POLL_TIMEOUT_MS);
1684 int j = 1 - (tcp_fd == -1);
1686 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1687 rtsp_st = rt->rtsp_streams[i];
1688 if (rtsp_st->rtp_handle) {
1689 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1690 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1692 *prtsp_st = rtsp_st;
1699 #if CONFIG_RTSP_DEMUXER
1700 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1701 RTSPMessageHeader reply;
1703 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1706 /* XXX: parse message */
1707 if (rt->state != RTSP_STATE_STREAMING)
1711 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1712 return AVERROR(ETIMEDOUT);
1713 } else if (n < 0 && errno != EINTR)
1714 return AVERROR(errno);
1718 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1720 RTSPState *rt = s->priv_data;
1722 RTSPStream *rtsp_st, *first_queue_st = NULL;
1723 int64_t wait_end = 0;
1725 if (rt->nb_byes == rt->nb_rtsp_streams)
1728 /* get next frames from the same RTP packet */
1729 if (rt->cur_transport_priv) {
1730 if (rt->transport == RTSP_TRANSPORT_RDT) {
1731 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1733 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1735 rt->cur_transport_priv = NULL;
1737 } else if (ret == 1) {
1740 rt->cur_transport_priv = NULL;
1743 if (rt->transport == RTSP_TRANSPORT_RTP) {
1745 int64_t first_queue_time = 0;
1746 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1747 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1751 queue_time = ff_rtp_queued_packet_time(rtpctx);
1752 if (queue_time && (queue_time - first_queue_time < 0 ||
1753 !first_queue_time)) {
1754 first_queue_time = queue_time;
1755 first_queue_st = rt->rtsp_streams[i];
1758 if (first_queue_time)
1759 wait_end = first_queue_time + s->max_delay;
1762 /* read next RTP packet */
1765 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1767 return AVERROR(ENOMEM);
1770 switch(rt->lower_transport) {
1772 #if CONFIG_RTSP_DEMUXER
1773 case RTSP_LOWER_TRANSPORT_TCP:
1774 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1777 case RTSP_LOWER_TRANSPORT_UDP:
1778 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1779 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1780 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1781 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1784 if (len == AVERROR(EAGAIN) && first_queue_st &&
1785 rt->transport == RTSP_TRANSPORT_RTP) {
1786 rtsp_st = first_queue_st;
1787 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1794 if (rt->transport == RTSP_TRANSPORT_RDT) {
1795 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1797 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1799 /* Either bad packet, or a RTCP packet. Check if the
1800 * first_rtcp_ntp_time field was initialized. */
1801 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1802 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1803 /* first_rtcp_ntp_time has been initialized for this stream,
1804 * copy the same value to all other uninitialized streams,
1805 * in order to map their timestamp origin to the same ntp time
1808 AVStream *st = NULL;
1809 if (rtsp_st->stream_index >= 0)
1810 st = s->streams[rtsp_st->stream_index];
1811 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1812 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1813 AVStream *st2 = NULL;
1814 if (rt->rtsp_streams[i]->stream_index >= 0)
1815 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1816 if (rtpctx2 && st && st2 &&
1817 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1818 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1819 rtpctx2->rtcp_ts_offset = av_rescale_q(
1820 rtpctx->rtcp_ts_offset, st->time_base,
1825 if (ret == -RTCP_BYE) {
1828 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1829 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1831 if (rt->nb_byes == rt->nb_rtsp_streams)
1840 /* more packets may follow, so we save the RTP context */
1841 rt->cur_transport_priv = rtsp_st->transport_priv;
1845 #endif /* CONFIG_RTPDEC */
1847 #if CONFIG_SDP_DEMUXER
1848 static int sdp_probe(AVProbeData *p1)
1850 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1852 /* we look for a line beginning "c=IN IP" */
1853 while (p < p_end && *p != '\0') {
1854 if (p + sizeof("c=IN IP") - 1 < p_end &&
1855 av_strstart(p, "c=IN IP", NULL))
1856 return AVPROBE_SCORE_MAX / 2;
1858 while (p < p_end - 1 && *p != '\n') p++;
1867 static int sdp_read_header(AVFormatContext *s)
1869 RTSPState *rt = s->priv_data;
1870 RTSPStream *rtsp_st;
1875 if (!ff_network_init())
1876 return AVERROR(EIO);
1878 if (s->max_delay < 0) /* Not set by the caller */
1879 s->max_delay = DEFAULT_REORDERING_DELAY;
1881 /* read the whole sdp file */
1882 /* XXX: better loading */
1883 content = av_malloc(SDP_MAX_SIZE);
1884 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1887 return AVERROR_INVALIDDATA;
1889 content[size] ='\0';
1891 err = ff_sdp_parse(s, content);
1895 /* open each RTP stream */
1896 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1898 rtsp_st = rt->rtsp_streams[i];
1900 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1901 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1902 ff_url_join(url, sizeof(url), "rtp", NULL,
1903 namebuf, rtsp_st->sdp_port,
1904 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1906 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1907 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1908 &s->interrupt_callback, NULL) < 0) {
1909 err = AVERROR_INVALIDDATA;
1912 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1917 ff_rtsp_close_streams(s);
1922 static int sdp_read_close(AVFormatContext *s)
1924 ff_rtsp_close_streams(s);
1929 static const AVClass sdp_demuxer_class = {
1930 .class_name = "SDP demuxer",
1931 .item_name = av_default_item_name,
1932 .option = sdp_options,
1933 .version = LIBAVUTIL_VERSION_INT,
1936 AVInputFormat ff_sdp_demuxer = {
1938 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1939 .priv_data_size = sizeof(RTSPState),
1940 .read_probe = sdp_probe,
1941 .read_header = sdp_read_header,
1942 .read_packet = ff_rtsp_fetch_packet,
1943 .read_close = sdp_read_close,
1944 .priv_class = &sdp_demuxer_class,
1946 #endif /* CONFIG_SDP_DEMUXER */
1948 #if CONFIG_RTP_DEMUXER
1949 static int rtp_probe(AVProbeData *p)
1951 if (av_strstart(p->filename, "rtp:", NULL))
1952 return AVPROBE_SCORE_MAX;
1956 static int rtp_read_header(AVFormatContext *s)
1958 uint8_t recvbuf[1500];
1959 char host[500], sdp[500];
1961 URLContext* in = NULL;
1963 AVCodecContext codec = { 0 };
1964 struct sockaddr_storage addr;
1966 socklen_t addrlen = sizeof(addr);
1967 RTSPState *rt = s->priv_data;
1969 if (!ff_network_init())
1970 return AVERROR(EIO);
1972 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1973 &s->interrupt_callback, NULL);
1978 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1979 if (ret == AVERROR(EAGAIN))
1984 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1988 if ((recvbuf[0] & 0xc0) != 0x80) {
1989 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1994 if (RTP_PT_IS_RTCP(recvbuf[1]))
1997 payload_type = recvbuf[1] & 0x7f;
2000 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2004 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2005 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2006 "without an SDP file describing it\n",
2010 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2011 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2012 "properly you need an SDP file "
2016 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2017 NULL, 0, s->filename);
2019 snprintf(sdp, sizeof(sdp),
2020 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2021 addr.ss_family == AF_INET ? 4 : 6, host,
2022 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2023 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2024 port, payload_type);
2025 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2027 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2030 /* sdp_read_header initializes this again */
2033 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2035 ret = sdp_read_header(s);
2046 static const AVClass rtp_demuxer_class = {
2047 .class_name = "RTP demuxer",
2048 .item_name = av_default_item_name,
2049 .option = rtp_options,
2050 .version = LIBAVUTIL_VERSION_INT,
2053 AVInputFormat ff_rtp_demuxer = {
2055 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2056 .priv_data_size = sizeof(RTSPState),
2057 .read_probe = rtp_probe,
2058 .read_header = rtp_read_header,
2059 .read_packet = ff_rtsp_fetch_packet,
2060 .read_close = sdp_read_close,
2061 .flags = AVFMT_NOFILE,
2062 .priv_class = &rtp_demuxer_class,
2064 #endif /* CONFIG_RTP_DEMUXER */