/* * GStreamer - GStreamer SRTP encoder and decoder * * Copyright 2009-2013 Collabora Ltd. * @author: Gabriel Millaire * @author: Olivier Crete * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #include "gstsrtp.h" #include #include "gstsrtpenc.h" #include "gstsrtpdec.h" #ifndef HAVE_SRTP2 srtp_err_status_t srtp_set_stream_roc (srtp_t session, guint32 ssrc, guint32 roc) { srtp_stream_t stream; stream = srtp_get_stream (session, htonl (ssrc)); if (stream == NULL) { return srtp_err_status_bad_param; } rdbx_set_roc (&stream->rtp_rdbx, roc); return srtp_err_status_ok; } srtp_err_status_t srtp_get_stream_roc (srtp_t session, guint32 ssrc, guint32 * roc) { srtp_stream_t stream; stream = srtp_get_stream (session, htonl (ssrc)); if (stream == NULL) { return srtp_err_status_bad_param; } *roc = stream->rtp_rdbx.index >> 16; return srtp_err_status_ok; } #endif static void free_reporter_data (gpointer data); GPrivate current_callback = G_PRIVATE_INIT (free_reporter_data); struct GstSrtpEventReporterData { gboolean soft_limit_reached; }; static void free_reporter_data (gpointer data) { g_slice_free (struct GstSrtpEventReporterData, data); } static void srtp_event_reporter (srtp_event_data_t * data) { struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback); if (!dat) return; switch (data->event) { case event_key_soft_limit: dat->soft_limit_reached = TRUE; break; default: break; } } void gst_srtp_init_event_reporter (void) { struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback); if (!dat) { dat = g_slice_new (struct GstSrtpEventReporterData); g_private_set (¤t_callback, dat); } dat->soft_limit_reached = FALSE; srtp_install_event_handler (srtp_event_reporter); } const gchar * enum_nick_from_value (GType enum_gtype, gint value) { GEnumClass *enum_class = g_type_class_ref (enum_gtype); GEnumValue *enum_value; const gchar *nick; if (!enum_gtype) return NULL; enum_value = g_enum_get_value (enum_class, value); if (!enum_value) return NULL; nick = enum_value->value_nick; g_type_class_unref (enum_class); return nick; } gint enum_value_from_nick (GType enum_gtype, const gchar * nick) { GEnumClass *enum_class = g_type_class_ref (enum_gtype); GEnumValue *enum_value; gint value; if (!enum_gtype) return -1; enum_value = g_enum_get_value_by_nick (enum_class, nick); if (!enum_value) return -1; value = enum_value->value; g_type_class_unref (enum_class); return value; } gboolean gst_srtp_get_soft_limit_reached (void) { struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback); if (dat) return dat->soft_limit_reached; return FALSE; } /* Get SSRC from RTCP buffer */ gboolean rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc) { gboolean ret = FALSE; GstRTCPBuffer rtcpbuf = GST_RTCP_BUFFER_INIT; GstRTCPPacket packet; /* Get SSRC from RR or SR packet (RTCP) */ if (!gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcpbuf)) return FALSE; if (gst_rtcp_buffer_get_first_packet (&rtcpbuf, &packet)) { GstRTCPType type; do { type = gst_rtcp_packet_get_type (&packet); switch (type) { case GST_RTCP_TYPE_RR: *ssrc = gst_rtcp_packet_rr_get_ssrc (&packet); ret = TRUE; break; case GST_RTCP_TYPE_SR: gst_rtcp_packet_sr_get_sender_info (&packet, ssrc, NULL, NULL, NULL, NULL); ret = TRUE; break; case GST_RTCP_TYPE_RTPFB: case GST_RTCP_TYPE_PSFB: *ssrc = gst_rtcp_packet_fb_get_sender_ssrc (&packet); ret = TRUE; break; case GST_RTCP_TYPE_APP: *ssrc = gst_rtcp_packet_app_get_ssrc (&packet); ret = TRUE; break; case GST_RTCP_TYPE_BYE: *ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, 0); ret = TRUE; break; default: break; } } while ((ret == FALSE) && (type != GST_RTCP_TYPE_INVALID) && gst_rtcp_packet_move_to_next (&packet)); } gst_rtcp_buffer_unmap (&rtcpbuf); return ret; } void set_crypto_policy_cipher_auth (GstSrtpCipherType cipher, GstSrtpAuthType auth, srtp_crypto_policy_t * policy) { switch (cipher) { case GST_SRTP_CIPHER_AES_128_ICM: policy->cipher_type = SRTP_AES_ICM_128; break; case GST_SRTP_CIPHER_AES_256_ICM: policy->cipher_type = SRTP_AES_ICM_256; break; case GST_SRTP_CIPHER_AES_128_GCM: policy->cipher_type = SRTP_AES_GCM_128; break; case GST_SRTP_CIPHER_AES_256_GCM: policy->cipher_type = SRTP_AES_GCM_256; break; case GST_SRTP_CIPHER_NULL: policy->cipher_type = SRTP_NULL_CIPHER; break; default: g_assert_not_reached (); } policy->cipher_key_len = cipher_key_size (cipher); switch (auth) { case GST_SRTP_AUTH_HMAC_SHA1_80: policy->auth_type = SRTP_HMAC_SHA1; policy->auth_key_len = 20; policy->auth_tag_len = 10; break; case GST_SRTP_AUTH_HMAC_SHA1_32: policy->auth_type = SRTP_HMAC_SHA1; policy->auth_key_len = 20; policy->auth_tag_len = 4; break; case GST_SRTP_AUTH_NULL: policy->auth_type = SRTP_NULL_AUTH; policy->auth_key_len = 0; if (cipher == GST_SRTP_CIPHER_AES_128_GCM || cipher == GST_SRTP_CIPHER_AES_256_GCM) { policy->auth_tag_len = 16; } else { policy->auth_tag_len = 0; } break; } if (cipher == GST_SRTP_CIPHER_NULL && auth == GST_SRTP_AUTH_NULL) policy->sec_serv = sec_serv_none; else if (cipher == GST_SRTP_CIPHER_NULL) policy->sec_serv = sec_serv_auth; else if (auth == GST_SRTP_AUTH_NULL) policy->sec_serv = sec_serv_conf; else policy->sec_serv = sec_serv_conf_and_auth; } guint cipher_key_size (GstSrtpCipherType cipher) { guint size = 0; switch (cipher) { case GST_SRTP_CIPHER_AES_128_ICM: size = SRTP_AES_ICM_128_KEY_LEN_WSALT; break; case GST_SRTP_CIPHER_AES_256_ICM: size = SRTP_AES_ICM_256_KEY_LEN_WSALT; break; case GST_SRTP_CIPHER_AES_128_GCM: size = SRTP_AES_GCM_128_KEY_LEN_WSALT; break; case GST_SRTP_CIPHER_AES_256_GCM: size = SRTP_AES_GCM_256_KEY_LEN_WSALT; break; case GST_SRTP_CIPHER_NULL: break; default: g_assert_not_reached (); } return size; } static gboolean plugin_init (GstPlugin * plugin) { srtp_init (); if (!gst_srtp_enc_plugin_init (plugin)) return FALSE; if (!gst_srtp_dec_plugin_init (plugin)) return FALSE; return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, srtp, "GStreamer SRTP", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)