/* GStreamer * Copyright (C) 2008 Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:rtsp-sdp * @short_description: Make SDP messages * @see_also: #GstRTSPMedia * * Last reviewed on 2013-07-11 (1.0.0) */ #include #include #include "rtsp-sdp.h" static gboolean get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data) { GstSDPMedia *media = (GstSDPMedia *) user_data; if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) { GstTagList *tags; guint bitrate = 0; gst_event_parse_tag (*event, &tags); if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM) return TRUE; if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE, &bitrate) || bitrate == 0) if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) || bitrate == 0) return TRUE; /* set bandwidth (kbits/s) */ gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000); return FALSE; } return TRUE; } static void update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media) { GstPad *src_pad; src_pad = gst_rtsp_stream_get_srcpad (stream); gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media); gst_object_unref (src_pad); } static void make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media, GstRTSPStream * stream, GstStructure * s, GstRTSPProfile profile) { GstSDPMedia *smedia; const gchar *caps_str, *caps_enc, *caps_params; gchar *tmp; gint caps_pt, caps_rate; guint n_fields, j; gboolean first; GString *fmtp; GstRTSPLowerTrans ltrans; GSocketFamily family; const gchar *addrtype, *proto; gchar *address; guint ttl; gst_sdp_media_new (&smedia); /* get media type and payload for the m= line */ caps_str = gst_structure_get_string (s, "media"); gst_sdp_media_set_media (smedia, caps_str); gst_structure_get_int (s, "payload", &caps_pt); tmp = g_strdup_printf ("%d", caps_pt); gst_sdp_media_add_format (smedia, tmp); g_free (tmp); gst_sdp_media_set_port_info (smedia, 0, 1); switch (profile) { case GST_RTSP_PROFILE_AVP: proto = "RTP/AVP"; break; case GST_RTSP_PROFILE_AVPF: proto = "RTP/AVPF"; break; case GST_RTSP_PROFILE_SAVP: proto = "RTP/SAVP"; break; case GST_RTSP_PROFILE_SAVPF: proto = "RTP/SAVPF"; break; default: proto = "udp"; break; } gst_sdp_media_set_proto (smedia, proto); if (info->is_ipv6) { addrtype = "IP6"; family = G_SOCKET_FAMILY_IPV6; } else { addrtype = "IP4"; family = G_SOCKET_FAMILY_IPV4; } ltrans = gst_rtsp_stream_get_protocols (stream); if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) { GstRTSPAddress *addr; addr = gst_rtsp_stream_get_multicast_address (stream, family); if (addr == NULL) goto no_multicast; address = g_strdup (addr->address); ttl = addr->ttl; gst_rtsp_address_free (addr); } else { ttl = 16; if (info->is_ipv6) address = g_strdup ("::"); else address = g_strdup ("0.0.0.0"); } /* for the c= line */ gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1); g_free (address); /* get clock-rate, media type and params for the rtpmap attribute */ gst_structure_get_int (s, "clock-rate", &caps_rate); caps_enc = gst_structure_get_string (s, "encoding-name"); caps_params = gst_structure_get_string (s, "encoding-params"); if (caps_enc) { if (caps_params) tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate, caps_params); else tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate); gst_sdp_media_add_attribute (smedia, "rtpmap", tmp); g_free (tmp); } /* the config uri */ tmp = gst_rtsp_stream_get_control (stream); gst_sdp_media_add_attribute (smedia, "control", tmp); g_free (tmp); /* check for srtp */ do { GstBuffer *srtpkey; const GValue *val; const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth; GstMIKEYMessage *msg; GstMIKEYPayload *payload, *pkd; GBytes *bytes; GstMapInfo info; const guint8 *data; gsize size; gchar *base64; guint8 byte; guint32 ssrc; val = gst_structure_get_value (s, "srtp-key"); if (val == NULL) break; srtpkey = gst_value_get_buffer (val); if (srtpkey == NULL) break; srtpcipher = gst_structure_get_string (s, "srtp-cipher"); srtpauth = gst_structure_get_string (s, "srtp-auth"); srtcpcipher = gst_structure_get_string (s, "srtcp-cipher"); srtcpauth = gst_structure_get_string (s, "srtcp-auth"); if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL || srtcpauth == NULL) break; msg = gst_mikey_message_new (); /* unencrypted MIKEY message, we send this over TLS so this is allowed */ gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT, FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP); /* add policy '0' for our SSRC */ gst_rtsp_stream_get_ssrc (stream, &ssrc); gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0); /* timestamp is now */ gst_mikey_message_add_t_now_ntp_utc (msg); /* add some random data */ gst_mikey_message_add_rand_len (msg, 16); /* the policy '0' is SRTP with the above discovered algorithms */ payload = gst_mikey_payload_new (GST_MIKEY_PT_SP); gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP); /* only AES-CM is supported */ byte = 1; gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte); /* only HMAC-SHA1 */ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1, &byte); /* we enable encryption on RTP and RTCP */ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1, &byte); gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1, &byte); /* we enable authentication on RTP and RTCP */ gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1, &byte); gst_mikey_message_add_payload (msg, payload); /* make unencrypted KEMAC */ payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC); gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL); /* add the key in key data */ pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA); gst_buffer_map (srtpkey, &info, GST_MAP_READ); gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size, info.data); gst_buffer_unmap (srtpkey, &info); /* add key data to KEMAC */ gst_mikey_payload_kemac_add_sub (payload, pkd); gst_mikey_message_add_payload (msg, payload); /* now serialize this to bytes */ bytes = gst_mikey_message_to_bytes (msg, NULL, NULL); gst_mikey_message_free (msg); /* and make it into base64 */ data = g_bytes_get_data (bytes, &size); base64 = g_base64_encode (data, size); g_bytes_unref (bytes); tmp = g_strdup_printf ("mikey %s", base64); g_free (base64); gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp); g_free (tmp); } while (FALSE); /* collect all other properties and add them to fmtp or attributes */ fmtp = g_string_new (""); g_string_append_printf (fmtp, "%d ", caps_pt); first = TRUE; n_fields = gst_structure_n_fields (s); for (j = 0; j < n_fields; j++) { const gchar *fname, *fval; fname = gst_structure_nth_field_name (s, j); /* filter out standard properties */ if (!strcmp (fname, "media")) continue; if (!strcmp (fname, "payload")) continue; if (!strcmp (fname, "clock-rate")) continue; if (!strcmp (fname, "encoding-name")) continue; if (!strcmp (fname, "encoding-params")) continue; if (!strcmp (fname, "ssrc")) continue; if (!strcmp (fname, "clock-base")) continue; if (!strcmp (fname, "seqnum-base")) continue; if (g_str_has_prefix (fname, "srtp-")) continue; if (g_str_has_prefix (fname, "srtcp-")) continue; if (g_str_has_prefix (fname, "a-")) { /* attribute */ if ((fval = gst_structure_get_string (s, fname))) gst_sdp_media_add_attribute (smedia, fname + 2, fval); continue; } if (g_str_has_prefix (fname, "x-")) { /* attribute */ if ((fval = gst_structure_get_string (s, fname))) gst_sdp_media_add_attribute (smedia, fname, fval); continue; } if ((fval = gst_structure_get_string (s, fname))) { g_string_append_printf (fmtp, "%s%s=%s", first ? "" : ";", fname, fval); first = FALSE; } } if (!first) { tmp = g_string_free (fmtp, FALSE); gst_sdp_media_add_attribute (smedia, "fmtp", tmp); g_free (tmp); } else { g_string_free (fmtp, TRUE); } update_sdp_from_tags (stream, smedia); gst_sdp_message_add_media (sdp, smedia); gst_sdp_media_free (smedia); return; /* ERRORS */ no_multicast: { gst_sdp_media_free (smedia); g_warning ("ignoring stream %d without multicast address", gst_rtsp_stream_get_index (stream)); return; } } /** * gst_rtsp_sdp_from_media: * @sdp: a #GstSDPMessage * @info: (transfer none): a #GstSDPInfo * @media: (transfer none): a #GstRTSPMedia * * Add @media specific info to @sdp. @info is used to configure the connection * information in the SDP. * * Returns: TRUE on success. */ gboolean gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media) { guint i, n_streams; gchar *rangestr; n_streams = gst_rtsp_media_n_streams (media); rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT); if (rangestr == NULL) goto not_prepared; gst_sdp_message_add_attribute (sdp, "range", rangestr); g_free (rangestr); for (i = 0; i < n_streams; i++) { GstRTSPStream *stream; GstCaps *caps; GstStructure *s; GstRTSPProfile profiles; guint mask; stream = gst_rtsp_media_get_stream (media, i); caps = gst_rtsp_stream_get_caps (stream); if (caps == NULL) { g_warning ("ignoring stream %d without media type", i); continue; } s = gst_caps_get_structure (caps, 0); if (s == NULL) { gst_caps_unref (caps); g_warning ("ignoring stream %d without media type", i); continue; } /* make a new media for each profile */ profiles = gst_rtsp_stream_get_profiles (stream); mask = 1; while (profiles >= mask) { GstRTSPProfile prof = profiles & mask; if (prof) make_media (sdp, info, media, stream, s, prof); mask <<= 1; } gst_caps_unref (caps); } { GstNetTimeProvider *provider; if ((provider = gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) { GstClock *clock; gchar *address, *str; gint port; g_object_get (provider, "clock", &clock, "address", &address, "port", &port, NULL); str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT, g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port, gst_clock_get_time (clock)); gst_sdp_message_add_attribute (sdp, "x-gst-clock", str); g_free (str); gst_object_unref (clock); g_free (address); gst_object_unref (provider); } } return TRUE; /* ERRORS */ not_prepared: { GST_ERROR ("media %p is not prepared", media); return FALSE; } }