/* * RTSP/SDP client * Copyright (c) 2002 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/base64.h" #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" #include "libavutil/parseutils.h" #include "libavutil/random_seed.h" #include "libavutil/dict.h" #include "libavutil/opt.h" #include "avformat.h" #include "avio_internal.h" #if HAVE_POLL_H #include #endif #include "internal.h" #include "network.h" #include "os_support.h" #include "http.h" #include "rtsp.h" #include "rtpdec.h" #include "rdt.h" #include "rtpdec_formats.h" #include "rtpenc_chain.h" #include "url.h" #include "rtpenc.h" //#define DEBUG /* Timeout values for socket poll, in ms, * and read_packet(), in seconds */ #define POLL_TIMEOUT_MS 100 #define READ_PACKET_TIMEOUT_S 10 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS #define SDP_MAX_SIZE 16384 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH #define DEFAULT_REORDERING_DELAY 100000 #define OFFSET(x) offsetof(RTSPState, x) #define DEC AV_OPT_FLAG_DECODING_PARAM #define ENC AV_OPT_FLAG_ENCODING_PARAM #define RTSP_FLAG_OPTS(name, longname) \ { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \ { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" } #define RTSP_MEDIATYPE_OPTS(name, longname) \ { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \ { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \ { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \ { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" } const AVOption ff_rtsp_options[] = { { "initial_pause", "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC }, FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags) { "rtsp_transport", "RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \ { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \ { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" }, { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" }, RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"), RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC }, { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC }, { NULL }, }; static const AVOption sdp_options[] = { RTSP_FLAG_OPTS("sdp_flags", "SDP flags"), RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"), { NULL }, }; static const AVOption rtp_options[] = { RTSP_FLAG_OPTS("rtp_flags", "RTP flags"), { NULL }, }; static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp) { const char *p; char *q; p = *pp; p += strspn(p, SPACE_CHARS); q = buf; while (!strchr(sep, *p) && *p != '\0') { if ((q - buf) < buf_size - 1) *q++ = *p; p++; } if (buf_size > 0) *q = '\0'; *pp = p; } static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp) { if (**pp == '/') (*pp)++; get_word_until_chars(buf, buf_size, sep, pp); } static void get_word(char *buf, int buf_size, const char **pp) { get_word_until_chars(buf, buf_size, SPACE_CHARS, pp); } /** Parse a string p in the form of Range:npt=xx-xx, and determine the start * and end time. * Used for seeking in the rtp stream. */ static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end) { char buf[256]; p += strspn(p, SPACE_CHARS); if (!av_stristart(p, "npt=", &p)) return; *start = AV_NOPTS_VALUE; *end = AV_NOPTS_VALUE; get_word_sep(buf, sizeof(buf), "-", &p); av_parse_time(start, buf, 1); if (*p == '-') { p++; get_word_sep(buf, sizeof(buf), "-", &p); av_parse_time(end, buf, 1); } // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start); // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end); } static int get_sockaddr(const char *buf, struct sockaddr_storage *sock) { struct addrinfo hints = { 0 }, *ai = NULL; hints.ai_flags = AI_NUMERICHOST; if (getaddrinfo(buf, NULL, &hints, &ai)) return -1; memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen)); freeaddrinfo(ai); return 0; } #if CONFIG_RTPDEC static void init_rtp_handler(RTPDynamicProtocolHandler *handler, RTSPStream *rtsp_st, AVCodecContext *codec) { if (!handler) return; codec->codec_id = handler->codec_id; rtsp_st->dynamic_handler = handler; if (handler->alloc) { rtsp_st->dynamic_protocol_context = handler->alloc(); if (!rtsp_st->dynamic_protocol_context) rtsp_st->dynamic_handler = NULL; } } /* parse the rtpmap description: /[/] */ static int sdp_parse_rtpmap(AVFormatContext *s, AVStream *st, RTSPStream *rtsp_st, int payload_type, const char *p) { AVCodecContext *codec = st->codec; char buf[256]; int i; AVCodec *c; const char *c_name; /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and * see if we can handle this kind of payload. * The space should normally not be there but some Real streams or * particular servers ("RealServer Version 6.1.3.970", see issue 1658) * have a trailing space. */ get_word_sep(buf, sizeof(buf), "/ ", &p); if (payload_type < RTP_PT_PRIVATE) { /* We are in a standard case * (from http://www.iana.org/assignments/rtp-parameters). */ /* search into AVRtpPayloadTypes[] */ codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); } if (codec->codec_id == CODEC_ID_NONE) { RTPDynamicProtocolHandler *handler = ff_rtp_handler_find_by_name(buf, codec->codec_type); init_rtp_handler(handler, rtsp_st, codec); /* If no dynamic handler was found, check with the list of standard * allocated types, if such a stream for some reason happens to * use a private payload type. This isn't handled in rtpdec.c, since * the format name from the rtpmap line never is passed into rtpdec. */ if (!rtsp_st->dynamic_handler) codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); } c = avcodec_find_decoder(codec->codec_id); if (c && c->name) c_name = c->name; else c_name = "(null)"; get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); switch (codec->codec_type) { case AVMEDIA_TYPE_AUDIO: av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name); codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE; codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; if (i > 0) { codec->sample_rate = i; avpriv_set_pts_info(st, 32, 1, codec->sample_rate); get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); if (i > 0) codec->channels = i; // TODO: there is a bug here; if it is a mono stream, and // less than 22000Hz, faad upconverts to stereo and twice // the frequency. No problem, but the sample rate is being // set here by the sdp line. Patch on its way. (rdm) } av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n", codec->sample_rate); av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n", codec->channels); break; case AVMEDIA_TYPE_VIDEO: av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name); if (i > 0) avpriv_set_pts_info(st, 32, 1, i); break; default: break; } if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) rtsp_st->dynamic_handler->init(s, st->index, rtsp_st->dynamic_protocol_context); return 0; } /* parse the attribute line from the fmtp a line of an sdp response. This * is broken out as a function because it is used in rtp_h264.c, which is * forthcoming. */ int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size) { *p += strspn(*p, SPACE_CHARS); if (**p) { get_word_sep(attr, attr_size, "=", p); if (**p == '=') (*p)++; get_word_sep(value, value_size, ";", p); if (**p == ';') (*p)++; return 1; } return 0; } typedef struct SDPParseState { /* SDP only */ struct sockaddr_storage default_ip; int default_ttl; int skip_media; ///< set if an unknown m= line occurs } SDPParseState; static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, int letter, const char *buf) { RTSPState *rt = s->priv_data; char buf1[64], st_type[64]; const char *p; enum AVMediaType codec_type; int payload_type, i; AVStream *st; RTSPStream *rtsp_st; struct sockaddr_storage sdp_ip; int ttl; av_dlog(s, "sdp: %c='%s'\n", letter, buf); p = buf; if (s1->skip_media && letter != 'm') return; switch (letter) { case 'c': get_word(buf1, sizeof(buf1), &p); if (strcmp(buf1, "IN") != 0) return; get_word(buf1, sizeof(buf1), &p); if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6")) return; get_word_sep(buf1, sizeof(buf1), "/", &p); if (get_sockaddr(buf1, &sdp_ip)) return; ttl = 16; if (*p == '/') { p++; get_word_sep(buf1, sizeof(buf1), "/", &p); ttl = atoi(buf1); } if (s->nb_streams == 0) { s1->default_ip = sdp_ip; s1->default_ttl = ttl; } else { rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; rtsp_st->sdp_ip = sdp_ip; rtsp_st->sdp_ttl = ttl; } break; case 's': av_dict_set(&s->metadata, "title", p, 0); break; case 'i': if (s->nb_streams == 0) { av_dict_set(&s->metadata, "comment", p, 0); break; } break; case 'm': /* new stream */ s1->skip_media = 0; codec_type = AVMEDIA_TYPE_UNKNOWN; get_word(st_type, sizeof(st_type), &p); if (!strcmp(st_type, "audio")) { codec_type = AVMEDIA_TYPE_AUDIO; } else if (!strcmp(st_type, "video")) { codec_type = AVMEDIA_TYPE_VIDEO; } else if (!strcmp(st_type, "application")) { codec_type = AVMEDIA_TYPE_DATA; } if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) { s1->skip_media = 1; return; } rtsp_st = av_mallocz(sizeof(RTSPStream)); if (!rtsp_st) return; rtsp_st->stream_index = -1; dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); rtsp_st->sdp_ip = s1->default_ip; rtsp_st->sdp_ttl = s1->default_ttl; get_word(buf1, sizeof(buf1), &p); /* port */ rtsp_st->sdp_port = atoi(buf1); get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */ /* XXX: handle list of formats */ get_word(buf1, sizeof(buf1), &p); /* format list */ rtsp_st->sdp_payload_type = atoi(buf1); if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) { /* no corresponding stream */ } else if (rt->server_type == RTSP_SERVER_WMS && codec_type == AVMEDIA_TYPE_DATA) { /* RTX stream, a stream that carries all the other actual * audio/video streams. Don't expose this to the callers. */ } else { st = avformat_new_stream(s, NULL); if (!st) return; st->id = rt->nb_rtsp_streams - 1; rtsp_st->stream_index = st->index; st->codec->codec_type = codec_type; if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { RTPDynamicProtocolHandler *handler; /* if standard payload type, we can find the codec right now */ ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && st->codec->sample_rate > 0) avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); /* Even static payload types may need a custom depacketizer */ handler = ff_rtp_handler_find_by_id( rtsp_st->sdp_payload_type, st->codec->codec_type); init_rtp_handler(handler, rtsp_st, st->codec); if (handler && handler->init) handler->init(s, st->index, rtsp_st->dynamic_protocol_context); } } /* put a default control url */ av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); break; case 'a': if (av_strstart(p, "control:", &p)) { if (s->nb_streams == 0) { if (!strncmp(p, "rtsp://", 7)) av_strlcpy(rt->control_uri, p, sizeof(rt->control_uri)); } else { char proto[32]; /* get the control url */ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; /* XXX: may need to add full url resolution */ av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p); if (proto[0] == '\0') { /* relative control URL */ if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/') av_strlcat(rtsp_st->control_url, "/", sizeof(rtsp_st->control_url)); av_strlcat(rtsp_st->control_url, p, sizeof(rtsp_st->control_url)); } else av_strlcpy(rtsp_st->control_url, p, sizeof(rtsp_st->control_url)); } } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) { /* NOTE: rtpmap is only supported AFTER the 'm=' tag */ get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; if (rtsp_st->stream_index >= 0) { st = s->streams[rtsp_st->stream_index]; sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p); } } else if (av_strstart(p, "fmtp:", &p) || av_strstart(p, "framesize:", &p)) { /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ // let dynamic protocol handlers have a stab at the line. get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->sdp_payload_type == payload_type && rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, rtsp_st->dynamic_protocol_context, buf); } } else if (av_strstart(p, "range:", &p)) { int64_t start, end; // this is so that seeking on a streamed file can work. rtsp_parse_range_npt(p, &start, &end); s->start_time = start; /* AV_NOPTS_VALUE means live broadcast (and can't seek) */ s->duration = (end == AV_NOPTS_VALUE) ? AV_NOPTS_VALUE : end - start; } else if (av_strstart(p, "IsRealDataType:integer;",&p)) { if (atoi(p) == 1) rt->transport = RTSP_TRANSPORT_RDT; } else if (av_strstart(p, "SampleRate:integer;", &p) && s->nb_streams > 0) { st = s->streams[s->nb_streams - 1]; st->codec->sample_rate = atoi(p); } else { if (rt->server_type == RTSP_SERVER_WMS) ff_wms_parse_sdp_a_line(s, p); if (s->nb_streams > 0) { rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1]; if (rt->server_type == RTSP_SERVER_REAL) ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p); if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) rtsp_st->dynamic_handler->parse_sdp_a_line(s, rtsp_st->stream_index, rtsp_st->dynamic_protocol_context, buf); } } break; } } int ff_sdp_parse(AVFormatContext *s, const char *content) { RTSPState *rt = s->priv_data; const char *p; int letter; /* Some SDP lines, particularly for Realmedia or ASF RTSP streams, * contain long SDP lines containing complete ASF Headers (several * kB) or arrays of MDPR (RM stream descriptor) headers plus * "rulebooks" describing their properties. Therefore, the SDP line * buffer is large. * * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line * in rtpdec_xiph.c. */ char buf[16384], *q; SDPParseState sdp_parse_state = { { 0 } }, *s1 = &sdp_parse_state; p = content; for (;;) { p += strspn(p, SPACE_CHARS); letter = *p; if (letter == '\0') break; p++; if (*p != '=') goto next_line; p++; /* get the content */ q = buf; while (*p != '\n' && *p != '\r' && *p != '\0') { if ((q - buf) < sizeof(buf) - 1) *q++ = *p; p++; } *q = '\0'; sdp_parse_line(s, s1, letter, buf); next_line: while (*p != '\n' && *p != '\0') p++; if (*p == '\n') p++; } rt->p = av_malloc(sizeof(struct pollfd)*2*(rt->nb_rtsp_streams+1)); if (!rt->p) return AVERROR(ENOMEM); return 0; } #endif /* CONFIG_RTPDEC */ void ff_rtsp_undo_setup(AVFormatContext *s) { RTSPState *rt = s->priv_data; int i; for (i = 0; i < rt->nb_rtsp_streams; i++) { RTSPStream *rtsp_st = rt->rtsp_streams[i]; if (!rtsp_st) continue; if (rtsp_st->transport_priv) { if (s->oformat) { AVFormatContext *rtpctx = rtsp_st->transport_priv; av_write_trailer(rtpctx); if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { uint8_t *ptr; avio_close_dyn_buf(rtpctx->pb, &ptr); av_free(ptr); } else { avio_close(rtpctx->pb); } avformat_free_context(rtpctx); } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) ff_rdt_parse_close(rtsp_st->transport_priv); else if (CONFIG_RTPDEC) ff_rtp_parse_close(rtsp_st->transport_priv); } rtsp_st->transport_priv = NULL; if (rtsp_st->rtp_handle) ffurl_close(rtsp_st->rtp_handle); rtsp_st->rtp_handle = NULL; } } /* close and free RTSP streams */ void ff_rtsp_close_streams(AVFormatContext *s) { RTSPState *rt = s->priv_data; int i; RTSPStream *rtsp_st; ff_rtsp_undo_setup(s); for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st) { if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) rtsp_st->dynamic_handler->free( rtsp_st->dynamic_protocol_context); av_free(rtsp_st); } } av_free(rt->rtsp_streams); if (rt->asf_ctx) { avformat_close_input(&rt->asf_ctx); } av_free(rt->p); av_free(rt->recvbuf); } static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) { RTSPState *rt = s->priv_data; AVStream *st = NULL; /* open the RTP context */ if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; if (!st) s->ctx_flags |= AVFMTCTX_NOHEADER; if (s->oformat && CONFIG_RTSP_MUXER) { int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st, rtsp_st->rtp_handle, RTSP_TCP_MAX_PACKET_SIZE); /* Ownership of rtp_handle is passed to the rtp mux context */ rtsp_st->rtp_handle = NULL; if (ret < 0) return ret; } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index, rtsp_st->dynamic_protocol_context, rtsp_st->dynamic_handler); else if (CONFIG_RTPDEC) rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay) ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE); if (!rtsp_st->transport_priv) { return AVERROR(ENOMEM); } else if (rt->transport != RTSP_TRANSPORT_RDT && CONFIG_RTPDEC) { if (rtsp_st->dynamic_handler) { ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, rtsp_st->dynamic_protocol_context, rtsp_st->dynamic_handler); } } return 0; } #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) { const char *q; char *p; int v; q = *pp; q += strspn(q, SPACE_CHARS); v = strtol(q, &p, 10); if (*p == '-') { p++; *min_ptr = v; v = strtol(p, &p, 10); *max_ptr = v; } else { *min_ptr = v; *max_ptr = v; } *pp = p; } /* XXX: only one transport specification is parsed */ static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p) { char transport_protocol[16]; char profile[16]; char lower_transport[16]; char parameter[16]; RTSPTransportField *th; char buf[256]; reply->nb_transports = 0; for (;;) { p += strspn(p, SPACE_CHARS); if (*p == '\0') break; th = &reply->transports[reply->nb_transports]; get_word_sep(transport_protocol, sizeof(transport_protocol), "/", &p); if (!av_strcasecmp (transport_protocol, "rtp")) { get_word_sep(profile, sizeof(profile), "/;,", &p); lower_transport[0] = '\0'; /* rtp/avp/ */ if (*p == '/') { get_word_sep(lower_transport, sizeof(lower_transport), ";,", &p); } th->transport = RTSP_TRANSPORT_RTP; } else if (!av_strcasecmp (transport_protocol, "x-pn-tng") || !av_strcasecmp (transport_protocol, "x-real-rdt")) { /* x-pn-tng/ */ get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p); profile[0] = '\0'; th->transport = RTSP_TRANSPORT_RDT; } if (!av_strcasecmp(lower_transport, "TCP")) th->lower_transport = RTSP_LOWER_TRANSPORT_TCP; else th->lower_transport = RTSP_LOWER_TRANSPORT_UDP; if (*p == ';') p++; /* get each parameter */ while (*p != '\0' && *p != ',') { get_word_sep(parameter, sizeof(parameter), "=;,", &p); if (!strcmp(parameter, "port")) { if (*p == '=') { p++; rtsp_parse_range(&th->port_min, &th->port_max, &p); } } else if (!strcmp(parameter, "client_port")) { if (*p == '=') { p++; rtsp_parse_range(&th->client_port_min, &th->client_port_max, &p); } } else if (!strcmp(parameter, "server_port")) { if (*p == '=') { p++; rtsp_parse_range(&th->server_port_min, &th->server_port_max, &p); } } else if (!strcmp(parameter, "interleaved")) { if (*p == '=') { p++; rtsp_parse_range(&th->interleaved_min, &th->interleaved_max, &p); } } else if (!strcmp(parameter, "multicast")) { if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP) th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST; } else if (!strcmp(parameter, "ttl")) { if (*p == '=') { p++; th->ttl = strtol(p, (char **)&p, 10); } } else if (!strcmp(parameter, "destination")) { if (*p == '=') { p++; get_word_sep(buf, sizeof(buf), ";,", &p); get_sockaddr(buf, &th->destination); } } else if (!strcmp(parameter, "source")) { if (*p == '=') { p++; get_word_sep(buf, sizeof(buf), ";,", &p); av_strlcpy(th->source, buf, sizeof(th->source)); } } while (*p != ';' && *p != '\0' && *p != ',') p++; if (*p == ';') p++; } if (*p == ',') p++; reply->nb_transports++; } } static void handle_rtp_info(RTSPState *rt, const char *url, uint32_t seq, uint32_t rtptime) { int i; if (!rtptime || !url[0]) return; if (rt->transport != RTSP_TRANSPORT_RTP) return; for (i = 0; i < rt->nb_rtsp_streams; i++) { RTSPStream *rtsp_st = rt->rtsp_streams[i]; RTPDemuxContext *rtpctx = rtsp_st->transport_priv; if (!rtpctx) continue; if (!strcmp(rtsp_st->control_url, url)) { rtpctx->base_timestamp = rtptime; break; } } } static void rtsp_parse_rtp_info(RTSPState *rt, const char *p) { int read = 0; char key[20], value[1024], url[1024] = ""; uint32_t seq = 0, rtptime = 0; for (;;) { p += strspn(p, SPACE_CHARS); if (!*p) break; get_word_sep(key, sizeof(key), "=", &p); if (*p != '=') break; p++; get_word_sep(value, sizeof(value), ";, ", &p); read++; if (!strcmp(key, "url")) av_strlcpy(url, value, sizeof(url)); else if (!strcmp(key, "seq")) seq = strtoul(value, NULL, 10); else if (!strcmp(key, "rtptime")) rtptime = strtoul(value, NULL, 10); if (*p == ',') { handle_rtp_info(rt, url, seq, rtptime); url[0] = '\0'; seq = rtptime = 0; read = 0; } if (*p) p++; } if (read > 0) handle_rtp_info(rt, url, seq, rtptime); } void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, RTSPState *rt, const char *method) { const char *p; /* NOTE: we do case independent match for broken servers */ p = buf; if (av_stristart(p, "Session:", &p)) { int t; get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); if (av_stristart(p, ";timeout=", &p) && (t = strtol(p, NULL, 10)) > 0) { reply->timeout = t; } } else if (av_stristart(p, "Content-Length:", &p)) { reply->content_length = strtol(p, NULL, 10); } else if (av_stristart(p, "Transport:", &p)) { rtsp_parse_transport(reply, p); } else if (av_stristart(p, "CSeq:", &p)) { reply->seq = strtol(p, NULL, 10); } else if (av_stristart(p, "Range:", &p)) { rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); } else if (av_stristart(p, "RealChallenge1:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge)); } else if (av_stristart(p, "Server:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->server, p, sizeof(reply->server)); } else if (av_stristart(p, "Notice:", &p) || av_stristart(p, "X-Notice:", &p)) { reply->notice = strtol(p, NULL, 10); } else if (av_stristart(p, "Location:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->location, p , sizeof(reply->location)); } else if (av_stristart(p, "WWW-Authenticate:", &p) && rt) { p += strspn(p, SPACE_CHARS); ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p); } else if (av_stristart(p, "Authentication-Info:", &p) && rt) { p += strspn(p, SPACE_CHARS); ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p); } else if (av_stristart(p, "Content-Base:", &p) && rt) { p += strspn(p, SPACE_CHARS); if (method && !strcmp(method, "DESCRIBE")) av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri)); } else if (av_stristart(p, "RTP-Info:", &p) && rt) { p += strspn(p, SPACE_CHARS); if (method && !strcmp(method, "PLAY")) rtsp_parse_rtp_info(rt, p); } else if (av_stristart(p, "Public:", &p) && rt) { if (strstr(p, "GET_PARAMETER") && method && !strcmp(method, "OPTIONS")) rt->get_parameter_supported = 1; } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) { p += strspn(p, SPACE_CHARS); rt->accept_dynamic_rate = atoi(p); } else if (av_stristart(p, "Content-Type:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->content_type, p, sizeof(reply->content_type)); } } /* skip a RTP/TCP interleaved packet */ void ff_rtsp_skip_packet(AVFormatContext *s) { RTSPState *rt = s->priv_data; int ret, len, len1; uint8_t buf[1024]; ret = ffurl_read_complete(rt->rtsp_hd, buf, 3); if (ret != 3) return; len = AV_RB16(buf + 1); av_dlog(s, "skipping RTP packet len=%d\n", len); /* skip payload */ while (len > 0) { len1 = len; if (len1 > sizeof(buf)) len1 = sizeof(buf); ret = ffurl_read_complete(rt->rtsp_hd, buf, len1); if (ret != len1) return; len -= len1; } } int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method) { RTSPState *rt = s->priv_data; char buf[4096], buf1[1024], *q; unsigned char ch; const char *p; int ret, content_length, line_count = 0, request = 0; unsigned char *content = NULL; start: line_count = 0; request = 0; content = NULL; memset(reply, 0, sizeof(*reply)); /* parse reply (XXX: use buffers) */ rt->last_reply[0] = '\0'; for (;;) { q = buf; for (;;) { ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1); av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch); if (ret != 1) return AVERROR_EOF; if (ch == '\n') break; if (ch == '$') { /* XXX: only parse it if first char on line ? */ if (return_on_interleaved_data) { return 1; } else ff_rtsp_skip_packet(s); } else if (ch != '\r') { if ((q - buf) < sizeof(buf) - 1) *q++ = ch; } } *q = '\0'; av_dlog(s, "line='%s'\n", buf); /* test if last line */ if (buf[0] == '\0') break; p = buf; if (line_count == 0) { /* get reply code */ get_word(buf1, sizeof(buf1), &p); if (!strncmp(buf1, "RTSP/", 5)) { get_word(buf1, sizeof(buf1), &p); reply->status_code = atoi(buf1); av_strlcpy(reply->reason, p, sizeof(reply->reason)); } else { av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method get_word(buf1, sizeof(buf1), &p); // object request = 1; } } else { ff_rtsp_parse_line(reply, p, rt, method); av_strlcat(rt->last_reply, p, sizeof(rt->last_reply)); av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply)); } line_count++; } if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request) av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id)); content_length = reply->content_length; if (content_length > 0) { /* leave some room for a trailing '\0' (useful for simple parsing) */ content = av_malloc(content_length + 1); ffurl_read_complete(rt->rtsp_hd, content, content_length); content[content_length] = '\0'; } if (content_ptr) *content_ptr = content; else av_free(content); if (request) { char buf[1024]; char base64buf[AV_BASE64_SIZE(sizeof(buf))]; const char* ptr = buf; if (!strcmp(reply->reason, "OPTIONS")) { snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n"); if (reply->seq) av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq); if (reply->session_id[0]) av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", reply->session_id); } else { snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n"); } av_strlcat(buf, "\r\n", sizeof(buf)); if (rt->control_transport == RTSP_MODE_TUNNEL) { av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf)); ptr = base64buf; } ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr)); rt->last_cmd_time = av_gettime(); /* Even if the request from the server had data, it is not the data * that the caller wants or expects. The memory could also be leaked * if the actual following reply has content data. */ if (content_ptr) av_freep(content_ptr); /* If method is set, this is called from ff_rtsp_send_cmd, * where a reply to exactly this request is awaited. For * callers from within packet receiving, we just want to * return to the caller and go back to receiving packets. */ if (method) goto start; return 0; } if (rt->seq != reply->seq) { av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n", rt->seq, reply->seq); } /* EOS */ if (reply->notice == 2101 /* End-of-Stream Reached */ || reply->notice == 2104 /* Start-of-Stream Reached */ || reply->notice == 2306 /* Continuous Feed Terminated */) { rt->state = RTSP_STATE_IDLE; } else if (reply->notice >= 4400 && reply->notice < 5500) { return AVERROR(EIO); /* data or server error */ } else if (reply->notice == 2401 /* Ticket Expired */ || (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ ) return AVERROR(EPERM); return 0; } /** * Send a command to the RTSP server without waiting for the reply. * * @param s RTSP (de)muxer context * @param method the method for the request * @param url the target url for the request * @param headers extra header lines to include in the request * @param send_content if non-null, the data to send as request body content * @param send_content_length the length of the send_content data, or 0 if * send_content is null * * @return zero if success, nonzero otherwise */ static int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s, const char *method, const char *url, const char *headers, const unsigned char *send_content, int send_content_length) { RTSPState *rt = s->priv_data; char buf[4096], *out_buf; char base64buf[AV_BASE64_SIZE(sizeof(buf))]; /* Add in RTSP headers */ out_buf = buf; rt->seq++; snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url); if (headers) av_strlcat(buf, headers, sizeof(buf)); av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq); if (rt->session_id[0] != '\0' && (!headers || !strstr(headers, "\nIf-Match:"))) { av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id); } if (rt->auth[0]) { char *str = ff_http_auth_create_response(&rt->auth_state, rt->auth, url, method); if (str) av_strlcat(buf, str, sizeof(buf)); av_free(str); } if (send_content_length > 0 && send_content) av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length); av_strlcat(buf, "\r\n", sizeof(buf)); /* base64 encode rtsp if tunneling */ if (rt->control_transport == RTSP_MODE_TUNNEL) { av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf)); out_buf = base64buf; } av_dlog(s, "Sending:\n%s--\n", buf); ffurl_write(rt->rtsp_hd_out, out_buf, strlen(out_buf)); if (send_content_length > 0 && send_content) { if (rt->control_transport == RTSP_MODE_TUNNEL) { av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests " "with content data not supported\n"); return AVERROR_PATCHWELCOME; } ffurl_write(rt->rtsp_hd_out, send_content, send_content_length); } rt->last_cmd_time = av_gettime(); return 0; } int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers) { return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0); } int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr) { return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply, content_ptr, NULL, 0); } int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *header, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length) { RTSPState *rt = s->priv_data; HTTPAuthType cur_auth_type; int ret, attempts = 0; retry: cur_auth_type = rt->auth_state.auth_type; if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header, send_content, send_content_length))) return ret; if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0) return ret; attempts++; if (reply->status_code == 401 && (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) && rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2) goto retry; if (reply->status_code > 400){ av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n", method, reply->status_code, reply->reason); av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply); } return 0; } int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge) { RTSPState *rt = s->priv_data; int rtx = 0, j, i, err, interleave = 0, port_off; RTSPStream *rtsp_st; RTSPMessageHeader reply1, *reply = &reply1; char cmd[2048]; const char *trans_pref; if (rt->transport == RTSP_TRANSPORT_RDT) trans_pref = "x-pn-tng"; else trans_pref = "RTP/AVP"; /* default timeout: 1 minute */ rt->timeout = 60; /* Choose a random starting offset within the first half of the * port range, to allow for a number of ports to try even if the offset * happens to be at the end of the random range. */ port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2); /* even random offset */ port_off -= port_off & 0x01; for (j = rt->rtp_port_min + port_off, i = 0; i < rt->nb_rtsp_streams; ++i) { char transport[2048]; /* * WMS serves all UDP data over a single connection, the RTX, which * isn't necessarily the first in the SDP but has to be the first * to be set up, else the second/third SETUP will fail with a 461. */ if (lower_transport == RTSP_LOWER_TRANSPORT_UDP && rt->server_type == RTSP_SERVER_WMS) { if (i == 0) { /* rtx first */ for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) { int len = strlen(rt->rtsp_streams[rtx]->control_url); if (len >= 4 && !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4, "/rtx")) break; } if (rtx == rt->nb_rtsp_streams) return -1; /* no RTX found */ rtsp_st = rt->rtsp_streams[rtx]; } else rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1]; } else rtsp_st = rt->rtsp_streams[i]; /* RTP/UDP */ if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) { char buf[256]; if (rt->server_type == RTSP_SERVER_WMS && i > 1) { port = reply->transports[0].client_port_min; goto have_port; } /* first try in specified port range */ while (j <= rt->rtp_port_max) { ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1, "?localport=%d", j); /* we will use two ports per rtp stream (rtp and rtcp) */ j += 2; if (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE, &s->interrupt_callback, NULL)) goto rtp_opened; } av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n"); err = AVERROR(EIO); goto fail; rtp_opened: port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle); have_port: snprintf(transport, sizeof(transport) - 1, "%s/UDP;", trans_pref); if (rt->server_type != RTSP_SERVER_REAL) av_strlcat(transport, "unicast;", sizeof(transport)); av_strlcatf(transport, sizeof(transport), "client_port=%d", port); if (rt->transport == RTSP_TRANSPORT_RTP && !(rt->server_type == RTSP_SERVER_WMS && i > 0)) av_strlcatf(transport, sizeof(transport), "-%d", port + 1); } /* RTP/TCP */ else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) { /* For WMS streams, the application streams are only used for * UDP. When trying to set it up for TCP streams, the server * will return an error. Therefore, we skip those streams. */ if (rt->server_type == RTSP_SERVER_WMS && (rtsp_st->stream_index < 0 || s->streams[rtsp_st->stream_index]->codec->codec_type == AVMEDIA_TYPE_DATA)) continue; snprintf(transport, sizeof(transport) - 1, "%s/TCP;", trans_pref); if (rt->transport != RTSP_TRANSPORT_RDT) av_strlcat(transport, "unicast;", sizeof(transport)); av_strlcatf(transport, sizeof(transport), "interleaved=%d-%d", interleave, interleave + 1); interleave += 2; } else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) { snprintf(transport, sizeof(transport) - 1, "%s/UDP;multicast", trans_pref); } if (s->oformat) { av_strlcat(transport, ";mode=record", sizeof(transport)); } else if (rt->server_type == RTSP_SERVER_REAL || rt->server_type == RTSP_SERVER_WMS) av_strlcat(transport, ";mode=play", sizeof(transport)); snprintf(cmd, sizeof(cmd), "Transport: %s\r\n", transport); if (rt->accept_dynamic_rate) av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd)); if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) { char real_res[41], real_csum[9]; ff_rdt_calc_response_and_checksum(real_res, real_csum, real_challenge); av_strlcatf(cmd, sizeof(cmd), "If-Match: %s\r\n" "RealChallenge2: %s, sd=%s\r\n", rt->session_id, real_res, real_csum); } ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL); if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) { err = 1; goto fail; } else if (reply->status_code != RTSP_STATUS_OK || reply->nb_transports != 1) { err = AVERROR_INVALIDDATA; goto fail; } /* XXX: same protocol for all streams is required */ if (i > 0) { if (reply->transports[0].lower_transport != rt->lower_transport || reply->transports[0].transport != rt->transport) { err = AVERROR_INVALIDDATA; goto fail; } } else { rt->lower_transport = reply->transports[0].lower_transport; rt->transport = reply->transports[0].transport; } /* Fail if the server responded with another lower transport mode * than what we requested. */ if (reply->transports[0].lower_transport != lower_transport) { av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n"); err = AVERROR_INVALIDDATA; goto fail; } switch(reply->transports[0].lower_transport) { case RTSP_LOWER_TRANSPORT_TCP: rtsp_st->interleaved_min = reply->transports[0].interleaved_min; rtsp_st->interleaved_max = reply->transports[0].interleaved_max; break; case RTSP_LOWER_TRANSPORT_UDP: { char url[1024], options[30] = ""; if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC) av_strlcpy(options, "?connect=1", sizeof(options)); /* Use source address if specified */ if (reply->transports[0].source[0]) { ff_url_join(url, sizeof(url), "rtp", NULL, reply->transports[0].source, reply->transports[0].server_port_min, "%s", options); } else { ff_url_join(url, sizeof(url), "rtp", NULL, host, reply->transports[0].server_port_min, "%s", options); } if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && ff_rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { err = AVERROR_INVALIDDATA; goto fail; } /* Try to initialize the connection state in a * potential NAT router by sending dummy packets. * RTP/RTCP dummy packets are used for RDT, too. */ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat && CONFIG_RTPDEC) ff_rtp_send_punch_packets(rtsp_st->rtp_handle); break; } case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: { char url[1024], namebuf[50], optbuf[20] = ""; struct sockaddr_storage addr; int port, ttl; if (reply->transports[0].destination.ss_family) { addr = reply->transports[0].destination; port = reply->transports[0].port_min; ttl = reply->transports[0].ttl; } else { addr = rtsp_st->sdp_ip; port = rtsp_st->sdp_port; ttl = rtsp_st->sdp_ttl; } if (ttl > 0) snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl); getnameinfo((struct sockaddr*) &addr, sizeof(addr), namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); ff_url_join(url, sizeof(url), "rtp", NULL, namebuf, port, "%s", optbuf); if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, &s->interrupt_callback, NULL) < 0) { err = AVERROR_INVALIDDATA; goto fail; } break; } } if ((err = rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } if (rt->nb_rtsp_streams && reply->timeout > 0) rt->timeout = reply->timeout; if (rt->server_type == RTSP_SERVER_REAL) rt->need_subscription = 1; return 0; fail: ff_rtsp_undo_setup(s); return err; } void ff_rtsp_close_connections(AVFormatContext *s) { RTSPState *rt = s->priv_data; if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out); ffurl_close(rt->rtsp_hd); rt->rtsp_hd = rt->rtsp_hd_out = NULL; } int ff_rtsp_connect(AVFormatContext *s) { RTSPState *rt = s->priv_data; char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128]; int port, err, tcp_fd; RTSPMessageHeader reply1 = {0}, *reply = &reply1; int lower_transport_mask = 0; char real_challenge[64] = ""; struct sockaddr_storage peer; socklen_t peer_len = sizeof(peer); if (rt->rtp_port_max < rt->rtp_port_min) { av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less " "than min port %d\n", rt->rtp_port_max, rt->rtp_port_min); return AVERROR(EINVAL); } if (!ff_network_init()) return AVERROR(EIO); if (s->max_delay < 0) /* Not set by the caller */ s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0; rt->control_transport = RTSP_MODE_PLAIN; if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) { rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP; rt->control_transport = RTSP_MODE_TUNNEL; } /* Only pass through valid flags from here */ rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1; redirect: lower_transport_mask = rt->lower_transport_mask; /* extract hostname and port */ av_url_split(NULL, 0, auth, sizeof(auth), host, sizeof(host), &port, path, sizeof(path), s->filename); if (*auth) { av_strlcpy(rt->auth, auth, sizeof(rt->auth)); } if (port < 0) port = RTSP_DEFAULT_PORT; if (!lower_transport_mask) lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1; if (s->oformat) { /* Only UDP or TCP - UDP multicast isn't supported. */ lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) | (1 << RTSP_LOWER_TRANSPORT_TCP); if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) { av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, " "only UDP and TCP are supported for output.\n"); err = AVERROR(EINVAL); goto fail; } } /* Construct the URI used in request; this is similar to s->filename, * but with authentication credentials removed and RTSP specific options * stripped out. */ ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL, host, port, "%s", path); if (rt->control_transport == RTSP_MODE_TUNNEL) { /* set up initial handshake for tunneling */ char httpname[1024]; char sessioncookie[17]; char headers[1024]; ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path); snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x", av_get_random_seed(), av_get_random_seed()); /* GET requests */ if (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ, &s->interrupt_callback) < 0) { err = AVERROR(EIO); goto fail; } /* generate GET headers */ snprintf(headers, sizeof(headers), "x-sessioncookie: %s\r\n" "Accept: application/x-rtsp-tunnelled\r\n" "Pragma: no-cache\r\n" "Cache-Control: no-cache\r\n", sessioncookie); av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0); /* complete the connection */ if (ffurl_connect(rt->rtsp_hd, NULL)) { err = AVERROR(EIO); goto fail; } /* POST requests */ if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE, &s->interrupt_callback) < 0 ) { err = AVERROR(EIO); goto fail; } /* generate POST headers */ snprintf(headers, sizeof(headers), "x-sessioncookie: %s\r\n" "Content-Type: application/x-rtsp-tunnelled\r\n" "Pragma: no-cache\r\n" "Cache-Control: no-cache\r\n" "Content-Length: 32767\r\n" "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n", sessioncookie); av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0); av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 0); /* Initialize the authentication state for the POST session. The HTTP * protocol implementation doesn't properly handle multi-pass * authentication for POST requests, since it would require one of * the following: * - implementing Expect: 100-continue, which many HTTP servers * don't support anyway, even less the RTSP servers that do HTTP * tunneling * - sending the whole POST data until getting a 401 reply specifying * what authentication method to use, then resending all that data * - waiting for potential 401 replies directly after sending the * POST header (waiting for some unspecified time) * Therefore, we copy the full auth state, which works for both basic * and digest. (For digest, we would have to synchronize the nonce * count variable between the two sessions, if we'd do more requests * with the original session, though.) */ ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd); /* complete the connection */ if (ffurl_connect(rt->rtsp_hd_out, NULL)) { err = AVERROR(EIO); goto fail; } } else { /* open the tcp connection */ ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL); if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE, &s->interrupt_callback, NULL) < 0) { err = AVERROR(EIO); goto fail; } rt->rtsp_hd_out = rt->rtsp_hd; } rt->seq = 0; tcp_fd = ffurl_get_file_handle(rt->rtsp_hd); if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) { getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host), NULL, 0, NI_NUMERICHOST); } /* request options supported by the server; this also detects server * type */ for (rt->server_type = RTSP_SERVER_RTP;;) { cmd[0] = 0; if (rt->server_type == RTSP_SERVER_REAL) av_strlcat(cmd, /* * The following entries are required for proper * streaming from a Realmedia server. They are * interdependent in some way although we currently * don't quite understand how. Values were copied * from mplayer SVN r23589. * ClientChallenge is a 16-byte ID in hex * CompanyID is a 16-byte ID in base64 */ "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n" "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n" "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n" "GUID: 00000000-0000-0000-0000-000000000000\r\n", sizeof(cmd)); ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { err = AVERROR_INVALIDDATA; goto fail; } /* detect server type if not standard-compliant RTP */ if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) { rt->server_type = RTSP_SERVER_REAL; continue; } else if (!av_strncasecmp(reply->server, "WMServer/", 9)) { rt->server_type = RTSP_SERVER_WMS; } else if (rt->server_type == RTSP_SERVER_REAL) strcpy(real_challenge, reply->real_challenge); break; } if (s->iformat && CONFIG_RTSP_DEMUXER) err = ff_rtsp_setup_input_streams(s, reply); else if (CONFIG_RTSP_MUXER) err = ff_rtsp_setup_output_streams(s, host); if (err) goto fail; do { int lower_transport = ff_log2_tab[lower_transport_mask & ~(lower_transport_mask - 1)]; err = ff_rtsp_make_setup_request(s, host, port, lower_transport, rt->server_type == RTSP_SERVER_REAL ? real_challenge : NULL); if (err < 0) goto fail; lower_transport_mask &= ~(1 << lower_transport); if (lower_transport_mask == 0 && err == 1) { err = AVERROR(EPROTONOSUPPORT); goto fail; } } while (err); rt->lower_transport_mask = lower_transport_mask; av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge)); rt->state = RTSP_STATE_IDLE; rt->seek_timestamp = 0; /* default is to start stream at position zero */ return 0; fail: ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) { av_strlcpy(s->filename, reply->location, sizeof(s->filename)); av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n", reply->status_code, s->filename); goto redirect; } ff_network_close(); return err; } #endif /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */ #if CONFIG_RTPDEC static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size, int64_t wait_end) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int n, i, ret, tcp_fd, timeout_cnt = 0; int max_p = 0; struct pollfd *p = rt->p; for (;;) { if (ff_check_interrupt(&s->interrupt_callback)) return AVERROR_EXIT; if (wait_end && wait_end - av_gettime() < 0) return AVERROR(EAGAIN); max_p = 0; if (rt->rtsp_hd) { tcp_fd = ffurl_get_file_handle(rt->rtsp_hd); p[max_p].fd = tcp_fd; p[max_p++].events = POLLIN; } else { tcp_fd = -1; } for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->rtp_handle) { p[max_p].fd = ffurl_get_file_handle(rtsp_st->rtp_handle); p[max_p++].events = POLLIN; p[max_p].fd = ff_rtp_get_rtcp_file_handle(rtsp_st->rtp_handle); p[max_p++].events = POLLIN; } } n = poll(p, max_p, POLL_TIMEOUT_MS); if (n > 0) { int j = 1 - (tcp_fd == -1); timeout_cnt = 0; for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->rtp_handle) { if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) { ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size); if (ret > 0) { *prtsp_st = rtsp_st; return ret; } } j+=2; } } #if CONFIG_RTSP_DEMUXER if (tcp_fd != -1 && p[0].revents & POLLIN) { RTSPMessageHeader reply; ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL); if (ret < 0) return ret; /* XXX: parse message */ if (rt->state != RTSP_STATE_STREAMING) return 0; } #endif } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { return AVERROR(ETIMEDOUT); } else if (n < 0 && errno != EINTR) return AVERROR(errno); } } int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) { RTSPState *rt = s->priv_data; int ret, len; RTSPStream *rtsp_st, *first_queue_st = NULL; int64_t wait_end = 0; if (rt->nb_byes == rt->nb_rtsp_streams) return AVERROR_EOF; /* get next frames from the same RTP packet */ if (rt->cur_transport_priv) { if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); } else ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); if (ret == 0) { rt->cur_transport_priv = NULL; return 0; } else if (ret == 1) { return 0; } else rt->cur_transport_priv = NULL; } if (rt->transport == RTSP_TRANSPORT_RTP) { int i; int64_t first_queue_time = 0; for (i = 0; i < rt->nb_rtsp_streams; i++) { RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv; int64_t queue_time; if (!rtpctx) continue; queue_time = ff_rtp_queued_packet_time(rtpctx); if (queue_time && (queue_time - first_queue_time < 0 || !first_queue_time)) { first_queue_time = queue_time; first_queue_st = rt->rtsp_streams[i]; } } if (first_queue_time) wait_end = first_queue_time + s->max_delay; } /* read next RTP packet */ redo: if (!rt->recvbuf) { rt->recvbuf = av_malloc(RECVBUF_SIZE); if (!rt->recvbuf) return AVERROR(ENOMEM); } switch(rt->lower_transport) { default: #if CONFIG_RTSP_DEMUXER case RTSP_LOWER_TRANSPORT_TCP: len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE); break; #endif case RTSP_LOWER_TRANSPORT_UDP: case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end); if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, len); break; } if (len == AVERROR(EAGAIN) && first_queue_st && rt->transport == RTSP_TRANSPORT_RTP) { rtsp_st = first_queue_st; ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0); goto end; } if (len < 0) return len; if (len == 0) return AVERROR_EOF; if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); } else { ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len); if (ret < 0) { /* Either bad packet, or a RTCP packet. Check if the * first_rtcp_ntp_time field was initialized. */ RTPDemuxContext *rtpctx = rtsp_st->transport_priv; if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) { /* first_rtcp_ntp_time has been initialized for this stream, * copy the same value to all other uninitialized streams, * in order to map their timestamp origin to the same ntp time * as this one. */ int i; AVStream *st = NULL; if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; for (i = 0; i < rt->nb_rtsp_streams; i++) { RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv; AVStream *st2 = NULL; if (rt->rtsp_streams[i]->stream_index >= 0) st2 = s->streams[rt->rtsp_streams[i]->stream_index]; if (rtpctx2 && st && st2 && rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) { rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time; rtpctx2->rtcp_ts_offset = av_rescale_q( rtpctx->rtcp_ts_offset, st->time_base, st2->time_base); } } } if (ret == -RTCP_BYE) { rt->nb_byes++; av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n", rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams); if (rt->nb_byes == rt->nb_rtsp_streams) return AVERROR_EOF; } } } end: if (ret < 0) goto redo; if (ret == 1) /* more packets may follow, so we save the RTP context */ rt->cur_transport_priv = rtsp_st->transport_priv; return ret; } #endif /* CONFIG_RTPDEC */ #if CONFIG_SDP_DEMUXER static int sdp_probe(AVProbeData *p1) { const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; /* we look for a line beginning "c=IN IP" */ while (p < p_end && *p != '\0') { if (p + sizeof("c=IN IP") - 1 < p_end && av_strstart(p, "c=IN IP", NULL)) return AVPROBE_SCORE_MAX / 2; while (p < p_end - 1 && *p != '\n') p++; if (++p >= p_end) break; if (*p == '\r') p++; } return 0; } static int sdp_read_header(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int size, i, err; char *content; char url[1024]; if (!ff_network_init()) return AVERROR(EIO); if (s->max_delay < 0) /* Not set by the caller */ s->max_delay = DEFAULT_REORDERING_DELAY; /* read the whole sdp file */ /* XXX: better loading */ content = av_malloc(SDP_MAX_SIZE); size = avio_read(s->pb, content, SDP_MAX_SIZE - 1); if (size <= 0) { av_free(content); return AVERROR_INVALIDDATA; } content[size] ='\0'; err = ff_sdp_parse(s, content); av_free(content); if (err) goto fail; /* open each RTP stream */ for (i = 0; i < rt->nb_rtsp_streams; i++) { char namebuf[50]; rtsp_st = rt->rtsp_streams[i]; getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip), namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST); ff_url_join(url, sizeof(url), "rtp", NULL, namebuf, rtsp_st->sdp_port, "?localport=%d&ttl=%d&connect=%d", rtsp_st->sdp_port, rtsp_st->sdp_ttl, rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0); if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE, &s->interrupt_callback, NULL) < 0) { err = AVERROR_INVALIDDATA; goto fail; } if ((err = rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } return 0; fail: ff_rtsp_close_streams(s); ff_network_close(); return err; } static int sdp_read_close(AVFormatContext *s) { ff_rtsp_close_streams(s); ff_network_close(); return 0; } static const AVClass sdp_demuxer_class = { .class_name = "SDP demuxer", .item_name = av_default_item_name, .option = sdp_options, .version = LIBAVUTIL_VERSION_INT, }; AVInputFormat ff_sdp_demuxer = { .name = "sdp", .long_name = NULL_IF_CONFIG_SMALL("SDP"), .priv_data_size = sizeof(RTSPState), .read_probe = sdp_probe, .read_header = sdp_read_header, .read_packet = ff_rtsp_fetch_packet, .read_close = sdp_read_close, .priv_class = &sdp_demuxer_class, }; #endif /* CONFIG_SDP_DEMUXER */ #if CONFIG_RTP_DEMUXER static int rtp_probe(AVProbeData *p) { if (av_strstart(p->filename, "rtp:", NULL)) return AVPROBE_SCORE_MAX; return 0; } static int rtp_read_header(AVFormatContext *s) { uint8_t recvbuf[1500]; char host[500], sdp[500]; int ret, port; URLContext* in = NULL; int payload_type; AVCodecContext codec = { 0 }; struct sockaddr_storage addr; AVIOContext pb; socklen_t addrlen = sizeof(addr); RTSPState *rt = s->priv_data; if (!ff_network_init()) return AVERROR(EIO); ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ, &s->interrupt_callback, NULL); if (ret) goto fail; while (1) { ret = ffurl_read(in, recvbuf, sizeof(recvbuf)); if (ret == AVERROR(EAGAIN)) continue; if (ret < 0) goto fail; if (ret < 12) { av_log(s, AV_LOG_WARNING, "Received too short packet\n"); continue; } if ((recvbuf[0] & 0xc0) != 0x80) { av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet " "received\n"); continue; } if (RTP_PT_IS_RTCP(recvbuf[1])) continue; payload_type = recvbuf[1] & 0x7f; break; } getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen); ffurl_close(in); in = NULL; if (ff_rtp_get_codec_info(&codec, payload_type)) { av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d " "without an SDP file describing it\n", payload_type); goto fail; } if (codec.codec_type != AVMEDIA_TYPE_DATA) { av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received " "properly you need an SDP file " "describing it\n"); } av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, NULL, 0, s->filename); snprintf(sdp, sizeof(sdp), "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n", addr.ss_family == AF_INET ? 4 : 6, host, codec.codec_type == AVMEDIA_TYPE_DATA ? "application" : codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio", port, payload_type); av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp); ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL); s->pb = &pb; /* sdp_read_header initializes this again */ ff_network_close(); rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1; ret = sdp_read_header(s); s->pb = NULL; return ret; fail: if (in) ffurl_close(in); ff_network_close(); return ret; } static const AVClass rtp_demuxer_class = { .class_name = "RTP demuxer", .item_name = av_default_item_name, .option = rtp_options, .version = LIBAVUTIL_VERSION_INT, }; AVInputFormat ff_rtp_demuxer = { .name = "rtp", .long_name = NULL_IF_CONFIG_SMALL("RTP input format"), .priv_data_size = sizeof(RTSPState), .read_probe = rtp_probe, .read_header = rtp_read_header, .read_packet = ff_rtsp_fetch_packet, .read_close = sdp_read_close, .flags = AVFMT_NOFILE, .priv_class = &rtp_demuxer_class, }; #endif /* CONFIG_RTP_DEMUXER */