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)
644 p += strspn(p, SPACE_CHARS);
645 v = strtol(p, (char **)&p, 10);
649 v = strtol(p, (char **)&p, 10);
658 /* XXX: only one transport specification is parsed */
659 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
661 char transport_protocol[16];
663 char lower_transport[16];
665 RTSPTransportField *th;
668 reply->nb_transports = 0;
671 p += strspn(p, SPACE_CHARS);
675 th = &reply->transports[reply->nb_transports];
677 get_word_sep(transport_protocol, sizeof(transport_protocol),
679 if (!av_strcasecmp (transport_protocol, "rtp")) {
680 get_word_sep(profile, sizeof(profile), "/;,", &p);
681 lower_transport[0] = '\0';
682 /* rtp/avp/<protocol> */
684 get_word_sep(lower_transport, sizeof(lower_transport),
687 th->transport = RTSP_TRANSPORT_RTP;
688 } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
689 !av_strcasecmp (transport_protocol, "x-real-rdt")) {
690 /* x-pn-tng/<protocol> */
691 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
693 th->transport = RTSP_TRANSPORT_RDT;
695 if (!av_strcasecmp(lower_transport, "TCP"))
696 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
698 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
702 /* get each parameter */
703 while (*p != '\0' && *p != ',') {
704 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
705 if (!strcmp(parameter, "port")) {
708 rtsp_parse_range(&th->port_min, &th->port_max, &p);
710 } else if (!strcmp(parameter, "client_port")) {
713 rtsp_parse_range(&th->client_port_min,
714 &th->client_port_max, &p);
716 } else if (!strcmp(parameter, "server_port")) {
719 rtsp_parse_range(&th->server_port_min,
720 &th->server_port_max, &p);
722 } else if (!strcmp(parameter, "interleaved")) {
725 rtsp_parse_range(&th->interleaved_min,
726 &th->interleaved_max, &p);
728 } else if (!strcmp(parameter, "multicast")) {
729 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
730 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
731 } else if (!strcmp(parameter, "ttl")) {
734 th->ttl = strtol(p, (char **)&p, 10);
736 } else if (!strcmp(parameter, "destination")) {
739 get_word_sep(buf, sizeof(buf), ";,", &p);
740 get_sockaddr(buf, &th->destination);
742 } else if (!strcmp(parameter, "source")) {
745 get_word_sep(buf, sizeof(buf), ";,", &p);
746 av_strlcpy(th->source, buf, sizeof(th->source));
750 while (*p != ';' && *p != '\0' && *p != ',')
758 reply->nb_transports++;
762 static void handle_rtp_info(RTSPState *rt, const char *url,
763 uint32_t seq, uint32_t rtptime)
766 if (!rtptime || !url[0])
768 if (rt->transport != RTSP_TRANSPORT_RTP)
770 for (i = 0; i < rt->nb_rtsp_streams; i++) {
771 RTSPStream *rtsp_st = rt->rtsp_streams[i];
772 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
775 if (!strcmp(rtsp_st->control_url, url)) {
776 rtpctx->base_timestamp = rtptime;
782 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
785 char key[20], value[1024], url[1024] = "";
786 uint32_t seq = 0, rtptime = 0;
789 p += strspn(p, SPACE_CHARS);
792 get_word_sep(key, sizeof(key), "=", &p);
796 get_word_sep(value, sizeof(value), ";, ", &p);
798 if (!strcmp(key, "url"))
799 av_strlcpy(url, value, sizeof(url));
800 else if (!strcmp(key, "seq"))
801 seq = strtoul(value, NULL, 10);
802 else if (!strcmp(key, "rtptime"))
803 rtptime = strtoul(value, NULL, 10);
805 handle_rtp_info(rt, url, seq, rtptime);
814 handle_rtp_info(rt, url, seq, rtptime);
817 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
818 RTSPState *rt, const char *method)
822 /* NOTE: we do case independent match for broken servers */
824 if (av_stristart(p, "Session:", &p)) {
826 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
827 if (av_stristart(p, ";timeout=", &p) &&
828 (t = strtol(p, NULL, 10)) > 0) {
831 } else if (av_stristart(p, "Content-Length:", &p)) {
832 reply->content_length = strtol(p, NULL, 10);
833 } else if (av_stristart(p, "Transport:", &p)) {
834 rtsp_parse_transport(reply, p);
835 } else if (av_stristart(p, "CSeq:", &p)) {
836 reply->seq = strtol(p, NULL, 10);
837 } else if (av_stristart(p, "Range:", &p)) {
838 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
839 } else if (av_stristart(p, "RealChallenge1:", &p)) {
840 p += strspn(p, SPACE_CHARS);
841 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
842 } else if (av_stristart(p, "Server:", &p)) {
843 p += strspn(p, SPACE_CHARS);
844 av_strlcpy(reply->server, p, sizeof(reply->server));
845 } else if (av_stristart(p, "Notice:", &p) ||
846 av_stristart(p, "X-Notice:", &p)) {
847 reply->notice = strtol(p, NULL, 10);
848 } else if (av_stristart(p, "Location:", &p)) {
849 p += strspn(p, SPACE_CHARS);
850 av_strlcpy(reply->location, p , sizeof(reply->location));
851 } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) {
852 p += strspn(p, SPACE_CHARS);
853 ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
854 } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {
855 p += strspn(p, SPACE_CHARS);
856 ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);
857 } else if (av_stristart(p, "Content-Base:", &p) && rt) {
858 p += strspn(p, SPACE_CHARS);
859 if (method && !strcmp(method, "DESCRIBE"))
860 av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));
861 } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
862 p += strspn(p, SPACE_CHARS);
863 if (method && !strcmp(method, "PLAY"))
864 rtsp_parse_rtp_info(rt, p);
865 } else if (av_stristart(p, "Public:", &p) && rt) {
866 if (strstr(p, "GET_PARAMETER") &&
867 method && !strcmp(method, "OPTIONS"))
868 rt->get_parameter_supported = 1;
869 } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
870 p += strspn(p, SPACE_CHARS);
871 rt->accept_dynamic_rate = atoi(p);
875 /* skip a RTP/TCP interleaved packet */
876 void ff_rtsp_skip_packet(AVFormatContext *s)
878 RTSPState *rt = s->priv_data;
882 ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
885 len = AV_RB16(buf + 1);
887 av_dlog(s, "skipping RTP packet len=%d\n", len);
892 if (len1 > sizeof(buf))
894 ret = ffurl_read_complete(rt->rtsp_hd, buf, len1);
901 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
902 unsigned char **content_ptr,
903 int return_on_interleaved_data, const char *method)
905 RTSPState *rt = s->priv_data;
906 char buf[4096], buf1[1024], *q;
909 int ret, content_length, line_count = 0, request = 0;
910 unsigned char *content = NULL;
916 memset(reply, 0, sizeof(*reply));
918 /* parse reply (XXX: use buffers) */
919 rt->last_reply[0] = '\0';
923 ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);
924 av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
930 /* XXX: only parse it if first char on line ? */
931 if (return_on_interleaved_data) {
934 ff_rtsp_skip_packet(s);
935 } else if (ch != '\r') {
936 if ((q - buf) < sizeof(buf) - 1)
942 av_dlog(s, "line='%s'\n", buf);
944 /* test if last line */
948 if (line_count == 0) {
950 get_word(buf1, sizeof(buf1), &p);
951 if (!strncmp(buf1, "RTSP/", 5)) {
952 get_word(buf1, sizeof(buf1), &p);
953 reply->status_code = atoi(buf1);
954 av_strlcpy(reply->reason, p, sizeof(reply->reason));
956 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
957 get_word(buf1, sizeof(buf1), &p); // object
961 ff_rtsp_parse_line(reply, p, rt, method);
962 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
963 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
968 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
969 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
971 content_length = reply->content_length;
972 if (content_length > 0) {
973 /* leave some room for a trailing '\0' (useful for simple parsing) */
974 content = av_malloc(content_length + 1);
975 ffurl_read_complete(rt->rtsp_hd, content, content_length);
976 content[content_length] = '\0';
979 *content_ptr = content;
985 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
986 const char* ptr = buf;
988 if (!strcmp(reply->reason, "OPTIONS")) {
989 snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
991 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
992 if (reply->session_id[0])
993 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
996 snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
998 av_strlcat(buf, "\r\n", sizeof(buf));
1000 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1001 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1004 ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
1006 rt->last_cmd_time = av_gettime();
1007 /* Even if the request from the server had data, it is not the data
1008 * that the caller wants or expects. The memory could also be leaked
1009 * if the actual following reply has content data. */
1011 av_freep(content_ptr);
1012 /* If method is set, this is called from ff_rtsp_send_cmd,
1013 * where a reply to exactly this request is awaited. For
1014 * callers from within packet receiving, we just want to
1015 * return to the caller and go back to receiving packets. */
1021 if (rt->seq != reply->seq) {
1022 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
1023 rt->seq, reply->seq);
1027 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
1028 reply->notice == 2104 /* Start-of-Stream Reached */ ||
1029 reply->notice == 2306 /* Continuous Feed Terminated */) {
1030 rt->state = RTSP_STATE_IDLE;
1031 } else if (reply->notice >= 4400 && reply->notice < 5500) {
1032 return AVERROR(EIO); /* data or server error */
1033 } else if (reply->notice == 2401 /* Ticket Expired */ ||
1034 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
1035 return AVERROR(EPERM);
1041 * Send a command to the RTSP server without waiting for the reply.
1043 * @param s RTSP (de)muxer context
1044 * @param method the method for the request
1045 * @param url the target url for the request
1046 * @param headers extra header lines to include in the request
1047 * @param send_content if non-null, the data to send as request body content
1048 * @param send_content_length the length of the send_content data, or 0 if
1049 * send_content is null
1051 * @return zero if success, nonzero otherwise
1053 static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1054 const char *method, const char *url,
1055 const char *headers,
1056 const unsigned char *send_content,
1057 int send_content_length)
1059 RTSPState *rt = s->priv_data;
1060 char buf[4096], *out_buf;
1061 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
1063 /* Add in RTSP headers */
1066 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
1068 av_strlcat(buf, headers, sizeof(buf));
1069 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1070 if (rt->session_id[0] != '\0' && (!headers ||
1071 !strstr(headers, "\nIf-Match:"))) {
1072 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1075 char *str = ff_http_auth_create_response(&rt->auth_state,
1076 rt->auth, url, method);
1078 av_strlcat(buf, str, sizeof(buf));
1081 if (send_content_length > 0 && send_content)
1082 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1083 av_strlcat(buf, "\r\n", sizeof(buf));
1085 /* base64 encode rtsp if tunneling */
1086 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1087 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
1088 out_buf = base64buf;
1091 av_dlog(s, "Sending:\n%s--\n", buf);
1093 ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
1094 if (send_content_length > 0 && send_content) {
1095 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1096 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
1097 "with content data not supported\n");
1098 return AVERROR_PATCHWELCOME;
1100 ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
1102 rt->last_cmd_time = av_gettime();
1107 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
1108 const char *url, const char *headers)
1110 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1113 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
1114 const char *headers, RTSPMessageHeader *reply,
1115 unsigned char **content_ptr)
1117 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
1118 content_ptr, NULL, 0);
1121 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1122 const char *method, const char *url,
1124 RTSPMessageHeader *reply,
1125 unsigned char **content_ptr,
1126 const unsigned char *send_content,
1127 int send_content_length)
1129 RTSPState *rt = s->priv_data;
1130 HTTPAuthType cur_auth_type;
1131 int ret, attempts = 0;
1134 cur_auth_type = rt->auth_state.auth_type;
1135 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
1137 send_content_length)))
1140 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
1144 if (reply->status_code == 401 &&
1145 (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
1146 rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
1149 if (reply->status_code > 400){
1150 av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n",
1154 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
1160 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
1161 int lower_transport, const char *real_challenge)
1163 RTSPState *rt = s->priv_data;
1164 int rtx = 0, j, i, err, interleave = 0, port_off;
1165 RTSPStream *rtsp_st;
1166 RTSPMessageHeader reply1, *reply = &reply1;
1168 const char *trans_pref;
1170 if (rt->transport == RTSP_TRANSPORT_RDT)
1171 trans_pref = "x-pn-tng";
1173 trans_pref = "RTP/AVP";
1175 /* default timeout: 1 minute */
1178 /* Choose a random starting offset within the first half of the
1179 * port range, to allow for a number of ports to try even if the offset
1180 * happens to be at the end of the random range. */
1181 port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);
1182 /* even random offset */
1183 port_off -= port_off & 0x01;
1185 for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) {
1186 char transport[2048];
1189 * WMS serves all UDP data over a single connection, the RTX, which
1190 * isn't necessarily the first in the SDP but has to be the first
1191 * to be set up, else the second/third SETUP will fail with a 461.
1193 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1194 rt->server_type == RTSP_SERVER_WMS) {
1197 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1198 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1200 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1204 if (rtx == rt->nb_rtsp_streams)
1205 return -1; /* no RTX found */
1206 rtsp_st = rt->rtsp_streams[rtx];
1208 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1210 rtsp_st = rt->rtsp_streams[i];
1213 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1216 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1217 port = reply->transports[0].client_port_min;
1221 /* first try in specified port range */
1222 while (j <= rt->rtp_port_max) {
1223 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1224 "?localport=%d", j);
1225 /* we will use two ports per rtp stream (rtp and rtcp) */
1227 if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
1228 &s->interrupt_callback, NULL))
1231 av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
1236 port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle);
1238 snprintf(transport, sizeof(transport) - 1,
1239 "%s/UDP;", trans_pref);
1240 if (rt->server_type != RTSP_SERVER_REAL)
1241 av_strlcat(transport, "unicast;", sizeof(transport));
1242 av_strlcatf(transport, sizeof(transport),
1243 "client_port=%d", port);
1244 if (rt->transport == RTSP_TRANSPORT_RTP &&
1245 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1246 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1250 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1251 /* For WMS streams, the application streams are only used for
1252 * UDP. When trying to set it up for TCP streams, the server
1253 * will return an error. Therefore, we skip those streams. */
1254 if (rt->server_type == RTSP_SERVER_WMS &&
1255 (rtsp_st->stream_index < 0 ||
1256 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1259 snprintf(transport, sizeof(transport) - 1,
1260 "%s/TCP;", trans_pref);
1261 if (rt->transport != RTSP_TRANSPORT_RDT)
1262 av_strlcat(transport, "unicast;", sizeof(transport));
1263 av_strlcatf(transport, sizeof(transport),
1264 "interleaved=%d-%d",
1265 interleave, interleave + 1);
1269 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1270 snprintf(transport, sizeof(transport) - 1,
1271 "%s/UDP;multicast", trans_pref);
1274 av_strlcat(transport, ";mode=receive", sizeof(transport));
1275 } else if (rt->server_type == RTSP_SERVER_REAL ||
1276 rt->server_type == RTSP_SERVER_WMS)
1277 av_strlcat(transport, ";mode=play", sizeof(transport));
1278 snprintf(cmd, sizeof(cmd),
1279 "Transport: %s\r\n",
1281 if (rt->accept_dynamic_rate)
1282 av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));
1283 if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {
1284 char real_res[41], real_csum[9];
1285 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1287 av_strlcatf(cmd, sizeof(cmd),
1289 "RealChallenge2: %s, sd=%s\r\n",
1290 rt->session_id, real_res, real_csum);
1292 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1293 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1296 } else if (reply->status_code != RTSP_STATUS_OK ||
1297 reply->nb_transports != 1) {
1298 err = AVERROR_INVALIDDATA;
1302 /* XXX: same protocol for all streams is required */
1304 if (reply->transports[0].lower_transport != rt->lower_transport ||
1305 reply->transports[0].transport != rt->transport) {
1306 err = AVERROR_INVALIDDATA;
1310 rt->lower_transport = reply->transports[0].lower_transport;
1311 rt->transport = reply->transports[0].transport;
1314 /* Fail if the server responded with another lower transport mode
1315 * than what we requested. */
1316 if (reply->transports[0].lower_transport != lower_transport) {
1317 av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
1318 err = AVERROR_INVALIDDATA;
1322 switch(reply->transports[0].lower_transport) {
1323 case RTSP_LOWER_TRANSPORT_TCP:
1324 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1325 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1328 case RTSP_LOWER_TRANSPORT_UDP: {
1329 char url[1024], options[30] = "";
1331 if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)
1332 av_strlcpy(options, "?connect=1", sizeof(options));
1333 /* Use source address if specified */
1334 if (reply->transports[0].source[0]) {
1335 ff_url_join(url, sizeof(url), "rtp", NULL,
1336 reply->transports[0].source,
1337 reply->transports[0].server_port_min, "%s", options);
1339 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1340 reply->transports[0].server_port_min, "%s", options);
1342 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1343 ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1344 err = AVERROR_INVALIDDATA;
1347 /* Try to initialize the connection state in a
1348 * potential NAT router by sending dummy packets.
1349 * RTP/RTCP dummy packets are used for RDT, too.
1351 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat &&
1353 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);
1356 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1357 char url[1024], namebuf[50], optbuf[20] = "";
1358 struct sockaddr_storage addr;
1361 if (reply->transports[0].destination.ss_family) {
1362 addr = reply->transports[0].destination;
1363 port = reply->transports[0].port_min;
1364 ttl = reply->transports[0].ttl;
1366 addr = rtsp_st->sdp_ip;
1367 port = rtsp_st->sdp_port;
1368 ttl = rtsp_st->sdp_ttl;
1371 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
1372 getnameinfo((struct sockaddr*) &addr, sizeof(addr),
1373 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1374 ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
1375 port, "%s", optbuf);
1376 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1377 &s->interrupt_callback, NULL) < 0) {
1378 err = AVERROR_INVALIDDATA;
1385 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1389 if (rt->nb_rtsp_streams && reply->timeout > 0)
1390 rt->timeout = reply->timeout;
1392 if (rt->server_type == RTSP_SERVER_REAL)
1393 rt->need_subscription = 1;
1398 ff_rtsp_undo_setup(s);
1402 void ff_rtsp_close_connections(AVFormatContext *s)
1404 RTSPState *rt = s->priv_data;
1405 if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
1406 ffurl_close(rt->rtsp_hd);
1407 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1410 int ff_rtsp_connect(AVFormatContext *s)
1412 RTSPState *rt = s->priv_data;
1413 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1414 int port, err, tcp_fd;
1415 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1416 int lower_transport_mask = 0;
1417 char real_challenge[64] = "";
1418 struct sockaddr_storage peer;
1419 socklen_t peer_len = sizeof(peer);
1421 if (rt->rtp_port_max < rt->rtp_port_min) {
1422 av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
1423 "than min port %d\n", rt->rtp_port_max,
1425 return AVERROR(EINVAL);
1428 if (!ff_network_init())
1429 return AVERROR(EIO);
1431 if (s->max_delay < 0) /* Not set by the caller */
1432 s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
1434 rt->control_transport = RTSP_MODE_PLAIN;
1435 if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
1436 rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
1437 rt->control_transport = RTSP_MODE_TUNNEL;
1439 /* Only pass through valid flags from here */
1440 rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1443 lower_transport_mask = rt->lower_transport_mask;
1444 /* extract hostname and port */
1445 av_url_split(NULL, 0, auth, sizeof(auth),
1446 host, sizeof(host), &port, path, sizeof(path), s->filename);
1448 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1451 port = RTSP_DEFAULT_PORT;
1453 if (!lower_transport_mask)
1454 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1457 /* Only UDP or TCP - UDP multicast isn't supported. */
1458 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1459 (1 << RTSP_LOWER_TRANSPORT_TCP);
1460 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1461 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1462 "only UDP and TCP are supported for output.\n");
1463 err = AVERROR(EINVAL);
1468 /* Construct the URI used in request; this is similar to s->filename,
1469 * but with authentication credentials removed and RTSP specific options
1471 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1472 host, port, "%s", path);
1474 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1475 /* set up initial handshake for tunneling */
1476 char httpname[1024];
1477 char sessioncookie[17];
1480 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1481 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1482 av_get_random_seed(), av_get_random_seed());
1485 if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
1486 &s->interrupt_callback) < 0) {
1491 /* generate GET headers */
1492 snprintf(headers, sizeof(headers),
1493 "x-sessioncookie: %s\r\n"
1494 "Accept: application/x-rtsp-tunnelled\r\n"
1495 "Pragma: no-cache\r\n"
1496 "Cache-Control: no-cache\r\n",
1498 av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);
1500 /* complete the connection */
1501 if (ffurl_connect(rt->rtsp_hd, NULL)) {
1507 if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
1508 &s->interrupt_callback) < 0 ) {
1513 /* generate POST headers */
1514 snprintf(headers, sizeof(headers),
1515 "x-sessioncookie: %s\r\n"
1516 "Content-Type: application/x-rtsp-tunnelled\r\n"
1517 "Pragma: no-cache\r\n"
1518 "Cache-Control: no-cache\r\n"
1519 "Content-Length: 32767\r\n"
1520 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1522 av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);
1523 av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0);
1525 /* Initialize the authentication state for the POST session. The HTTP
1526 * protocol implementation doesn't properly handle multi-pass
1527 * authentication for POST requests, since it would require one of
1529 * - implementing Expect: 100-continue, which many HTTP servers
1530 * don't support anyway, even less the RTSP servers that do HTTP
1532 * - sending the whole POST data until getting a 401 reply specifying
1533 * what authentication method to use, then resending all that data
1534 * - waiting for potential 401 replies directly after sending the
1535 * POST header (waiting for some unspecified time)
1536 * Therefore, we copy the full auth state, which works for both basic
1537 * and digest. (For digest, we would have to synchronize the nonce
1538 * count variable between the two sessions, if we'd do more requests
1539 * with the original session, though.)
1541 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1543 /* complete the connection */
1544 if (ffurl_connect(rt->rtsp_hd_out, NULL)) {
1549 /* open the tcp connection */
1550 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1551 if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
1552 &s->interrupt_callback, NULL) < 0) {
1556 rt->rtsp_hd_out = rt->rtsp_hd;
1560 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1561 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1562 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1563 NULL, 0, NI_NUMERICHOST);
1566 /* request options supported by the server; this also detects server
1568 for (rt->server_type = RTSP_SERVER_RTP;;) {
1570 if (rt->server_type == RTSP_SERVER_REAL)
1573 * The following entries are required for proper
1574 * streaming from a Realmedia server. They are
1575 * interdependent in some way although we currently
1576 * don't quite understand how. Values were copied
1577 * from mplayer SVN r23589.
1578 * ClientChallenge is a 16-byte ID in hex
1579 * CompanyID is a 16-byte ID in base64
1581 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1582 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1583 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1584 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1586 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1587 if (reply->status_code != RTSP_STATUS_OK) {
1588 err = AVERROR_INVALIDDATA;
1592 /* detect server type if not standard-compliant RTP */
1593 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1594 rt->server_type = RTSP_SERVER_REAL;
1596 } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) {
1597 rt->server_type = RTSP_SERVER_WMS;
1598 } else if (rt->server_type == RTSP_SERVER_REAL)
1599 strcpy(real_challenge, reply->real_challenge);
1603 if (s->iformat && CONFIG_RTSP_DEMUXER)
1604 err = ff_rtsp_setup_input_streams(s, reply);
1605 else if (CONFIG_RTSP_MUXER)
1606 err = ff_rtsp_setup_output_streams(s, host);
1611 int lower_transport = ff_log2_tab[lower_transport_mask &
1612 ~(lower_transport_mask - 1)];
1614 err = ff_rtsp_make_setup_request(s, host, port, lower_transport,
1615 rt->server_type == RTSP_SERVER_REAL ?
1616 real_challenge : NULL);
1619 lower_transport_mask &= ~(1 << lower_transport);
1620 if (lower_transport_mask == 0 && err == 1) {
1621 err = AVERROR(EPROTONOSUPPORT);
1626 rt->lower_transport_mask = lower_transport_mask;
1627 av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
1628 rt->state = RTSP_STATE_IDLE;
1629 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1632 ff_rtsp_close_streams(s);
1633 ff_rtsp_close_connections(s);
1634 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1635 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1636 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1644 #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */
1647 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1648 uint8_t *buf, int buf_size, int64_t wait_end)
1650 RTSPState *rt = s->priv_data;
1651 RTSPStream *rtsp_st;
1652 int n, i, ret, tcp_fd, timeout_cnt = 0;
1654 struct pollfd *p = rt->p;
1657 if (ff_check_interrupt(&s->interrupt_callback))
1658 return AVERROR_EXIT;
1659 if (wait_end && wait_end - av_gettime() < 0)
1660 return AVERROR(EAGAIN);
1663 tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);
1664 p[max_p].fd = tcp_fd;
1665 p[max_p++].events = POLLIN;
1669 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1670 rtsp_st = rt->rtsp_streams[i];
1671 if (rtsp_st->rtp_handle) {
1672 p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle);
1673 p[max_p++].events = POLLIN;
1674 p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle);
1675 p[max_p++].events = POLLIN;
1678 n = poll(p, max_p, POLL_TIMEOUT_MS);
1680 int j = 1 - (tcp_fd == -1);
1682 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1683 rtsp_st = rt->rtsp_streams[i];
1684 if (rtsp_st->rtp_handle) {
1685 if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {
1686 ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);
1688 *prtsp_st = rtsp_st;
1695 #if CONFIG_RTSP_DEMUXER
1696 if (tcp_fd != -1 && p[0].revents & POLLIN) {
1697 RTSPMessageHeader reply;
1699 ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
1702 /* XXX: parse message */
1703 if (rt->state != RTSP_STATE_STREAMING)
1707 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1708 return AVERROR(ETIMEDOUT);
1709 } else if (n < 0 && errno != EINTR)
1710 return AVERROR(errno);
1714 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1716 RTSPState *rt = s->priv_data;
1718 RTSPStream *rtsp_st, *first_queue_st = NULL;
1719 int64_t wait_end = 0;
1721 if (rt->nb_byes == rt->nb_rtsp_streams)
1724 /* get next frames from the same RTP packet */
1725 if (rt->cur_transport_priv) {
1726 if (rt->transport == RTSP_TRANSPORT_RDT) {
1727 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1729 ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1731 rt->cur_transport_priv = NULL;
1733 } else if (ret == 1) {
1736 rt->cur_transport_priv = NULL;
1739 if (rt->transport == RTSP_TRANSPORT_RTP) {
1741 int64_t first_queue_time = 0;
1742 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1743 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
1747 queue_time = ff_rtp_queued_packet_time(rtpctx);
1748 if (queue_time && (queue_time - first_queue_time < 0 ||
1749 !first_queue_time)) {
1750 first_queue_time = queue_time;
1751 first_queue_st = rt->rtsp_streams[i];
1754 if (first_queue_time)
1755 wait_end = first_queue_time + s->max_delay;
1758 /* read next RTP packet */
1761 rt->recvbuf = av_malloc(RECVBUF_SIZE);
1763 return AVERROR(ENOMEM);
1766 switch(rt->lower_transport) {
1768 #if CONFIG_RTSP_DEMUXER
1769 case RTSP_LOWER_TRANSPORT_TCP:
1770 len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);
1773 case RTSP_LOWER_TRANSPORT_UDP:
1774 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1775 len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);
1776 if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1777 ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1780 if (len == AVERROR(EAGAIN) && first_queue_st &&
1781 rt->transport == RTSP_TRANSPORT_RTP) {
1782 rtsp_st = first_queue_st;
1783 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);
1790 if (rt->transport == RTSP_TRANSPORT_RDT) {
1791 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1793 ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);
1795 /* Either bad packet, or a RTCP packet. Check if the
1796 * first_rtcp_ntp_time field was initialized. */
1797 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1798 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1799 /* first_rtcp_ntp_time has been initialized for this stream,
1800 * copy the same value to all other uninitialized streams,
1801 * in order to map their timestamp origin to the same ntp time
1804 AVStream *st = NULL;
1805 if (rtsp_st->stream_index >= 0)
1806 st = s->streams[rtsp_st->stream_index];
1807 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1808 RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;
1809 AVStream *st2 = NULL;
1810 if (rt->rtsp_streams[i]->stream_index >= 0)
1811 st2 = s->streams[rt->rtsp_streams[i]->stream_index];
1812 if (rtpctx2 && st && st2 &&
1813 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
1814 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1815 rtpctx2->rtcp_ts_offset = av_rescale_q(
1816 rtpctx->rtcp_ts_offset, st->time_base,
1821 if (ret == -RTCP_BYE) {
1824 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
1825 rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
1827 if (rt->nb_byes == rt->nb_rtsp_streams)
1836 /* more packets may follow, so we save the RTP context */
1837 rt->cur_transport_priv = rtsp_st->transport_priv;
1841 #endif /* CONFIG_RTPDEC */
1843 #if CONFIG_SDP_DEMUXER
1844 static int sdp_probe(AVProbeData *p1)
1846 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1848 /* we look for a line beginning "c=IN IP" */
1849 while (p < p_end && *p != '\0') {
1850 if (p + sizeof("c=IN IP") - 1 < p_end &&
1851 av_strstart(p, "c=IN IP", NULL))
1852 return AVPROBE_SCORE_MAX / 2;
1854 while (p < p_end - 1 && *p != '\n') p++;
1863 static int sdp_read_header(AVFormatContext *s)
1865 RTSPState *rt = s->priv_data;
1866 RTSPStream *rtsp_st;
1871 if (!ff_network_init())
1872 return AVERROR(EIO);
1874 if (s->max_delay < 0) /* Not set by the caller */
1875 s->max_delay = DEFAULT_REORDERING_DELAY;
1877 /* read the whole sdp file */
1878 /* XXX: better loading */
1879 content = av_malloc(SDP_MAX_SIZE);
1880 size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);
1883 return AVERROR_INVALIDDATA;
1885 content[size] ='\0';
1887 err = ff_sdp_parse(s, content);
1891 /* open each RTP stream */
1892 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1894 rtsp_st = rt->rtsp_streams[i];
1896 getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
1897 namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
1898 ff_url_join(url, sizeof(url), "rtp", NULL,
1899 namebuf, rtsp_st->sdp_port,
1900 "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port,
1902 rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0);
1903 if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
1904 &s->interrupt_callback, NULL) < 0) {
1905 err = AVERROR_INVALIDDATA;
1908 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1913 ff_rtsp_close_streams(s);
1918 static int sdp_read_close(AVFormatContext *s)
1920 ff_rtsp_close_streams(s);
1925 static const AVClass sdp_demuxer_class = {
1926 .class_name = "SDP demuxer",
1927 .item_name = av_default_item_name,
1928 .option = sdp_options,
1929 .version = LIBAVUTIL_VERSION_INT,
1932 AVInputFormat ff_sdp_demuxer = {
1934 .long_name = NULL_IF_CONFIG_SMALL("SDP"),
1935 .priv_data_size = sizeof(RTSPState),
1936 .read_probe = sdp_probe,
1937 .read_header = sdp_read_header,
1938 .read_packet = ff_rtsp_fetch_packet,
1939 .read_close = sdp_read_close,
1940 .priv_class = &sdp_demuxer_class,
1942 #endif /* CONFIG_SDP_DEMUXER */
1944 #if CONFIG_RTP_DEMUXER
1945 static int rtp_probe(AVProbeData *p)
1947 if (av_strstart(p->filename, "rtp:", NULL))
1948 return AVPROBE_SCORE_MAX;
1952 static int rtp_read_header(AVFormatContext *s)
1954 uint8_t recvbuf[1500];
1955 char host[500], sdp[500];
1957 URLContext* in = NULL;
1959 AVCodecContext codec = { 0 };
1960 struct sockaddr_storage addr;
1962 socklen_t addrlen = sizeof(addr);
1963 RTSPState *rt = s->priv_data;
1965 if (!ff_network_init())
1966 return AVERROR(EIO);
1968 ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,
1969 &s->interrupt_callback, NULL);
1974 ret = ffurl_read(in, recvbuf, sizeof(recvbuf));
1975 if (ret == AVERROR(EAGAIN))
1980 av_log(s, AV_LOG_WARNING, "Received too short packet\n");
1984 if ((recvbuf[0] & 0xc0) != 0x80) {
1985 av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
1990 if (RTP_PT_IS_RTCP(recvbuf[1]))
1993 payload_type = recvbuf[1] & 0x7f;
1996 getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);
2000 if (ff_rtp_get_codec_info(&codec, payload_type)) {
2001 av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
2002 "without an SDP file describing it\n",
2006 if (codec.codec_type != AVMEDIA_TYPE_DATA) {
2007 av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
2008 "properly you need an SDP file "
2012 av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
2013 NULL, 0, s->filename);
2015 snprintf(sdp, sizeof(sdp),
2016 "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
2017 addr.ss_family == AF_INET ? 4 : 6, host,
2018 codec.codec_type == AVMEDIA_TYPE_DATA ? "application" :
2019 codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
2020 port, payload_type);
2021 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
2023 ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);
2026 /* sdp_read_header initializes this again */
2029 rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
2031 ret = sdp_read_header(s);
2042 static const AVClass rtp_demuxer_class = {
2043 .class_name = "RTP demuxer",
2044 .item_name = av_default_item_name,
2045 .option = rtp_options,
2046 .version = LIBAVUTIL_VERSION_INT,
2049 AVInputFormat ff_rtp_demuxer = {
2051 .long_name = NULL_IF_CONFIG_SMALL("RTP input format"),
2052 .priv_data_size = sizeof(RTSPState),
2053 .read_probe = rtp_probe,
2054 .read_header = rtp_read_header,
2055 .read_packet = ff_rtsp_fetch_packet,
2056 .read_close = sdp_read_close,
2057 .flags = AVFMT_NOFILE,
2058 .priv_class = &rtp_demuxer_class,
2060 #endif /* CONFIG_RTP_DEMUXER */