/* * GStreamer - GStreamer SRTP decoder * * Copyright 2009-2011 Collabora Ltd. * @author: Gabriel Millaire * @author: Olivier Crete * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. * * Alternatively, the contents of this file may be used under the * GNU Lesser General Public License Version 2.1 (the "LGPL"), in * which case the following provisions apply instead of the ones * mentioned above: * * 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. */ /** * SECTION:element-srtpdec * @see_also: srtpenc * * gstrtpdec acts as a decoder that removes security from SRTP and SRTCP * packets (encryption and authentication) and out RTP and RTCP. It * receives packet of type 'application/x-srtp' or 'application/x-srtcp' * on its sink pad, and outs packets of type 'application/x-rtp' or * 'application/x-rtcp' on its source pad. * * For each packet received, it checks if the internal SSRC is in the list * of streams already in use. If this is not the case, it sends a signal to * the user to get the needed parameters to create a new stream : master * key, encryption and authentication mecanisms for both RTP and RTCP. If * the user can't provide those parameters, the buffer is dropped and a * warning is emitted. * * This element uses libsrtp library. The encryption and authentication * mecanisms available are : * * Encryption * - AES_ICM 256 bits (maximum security) * - AES_ICM 128 bits (default) * - NULL * * Authentication * - HMAC_SHA1 80 bits (default, maximum protection) * - HMAC_SHA1 32 bits * - NULL * * Note that for SRTP protection, authentication is mandatory (non-null) * if encryption is used (non-null). * * Each packet received is first analysed (checked for valid SSRC) then * its buffer is unprotected with libsrtp, then pushed on the source pad. * If protection failed or the stream could not be created, the buffer * is dropped and a warning is emitted. * * When the maximum usage of the master key is reached, a soft-limit * signal is sent to the user, and new parameters (master key) are needed * in return. If the hard limit is reached, a flag is set and every * subsequent packet is dropped, until a new key is set and the stream * has been updated. * * If a stream is to be shared between multiple clients the SRTP * rollover counter for a given SSRC must be set in the caps "roc" field * when the request-key signal is emitted by the decoder. The rollover * counters should have been transmitted by a signaling protocol by some * other means. If no rollover counter is provided by the user, 0 is * used by default. * * * Example pipelines * |[ * gst-launch-1.0 udpsrc port=5004 caps='application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80' ! srtpdec ! rtppcmadepay ! alawdec ! pulsesink * ]| Receive PCMA SRTP packets through UDP using caps to specify * master key and protection. * |[ * gst-launch-1.0 audiotestsrc ! alawenc ! rtppcmapay ! 'application/x-rtp, payload=(int)8, ssrc=(uint)1356955624' ! srtpenc key="012345678901234567890123456789012345678901234567890123456789" ! udpsink port=5004 * ]| Send PCMA SRTP packets through UDP, nothing how the SSRC is forced so * that the receiver will recognize it. * */ #ifdef HAVE_CONFIG_H #include #endif #include #include #include #include "gstsrtp.h" #include "gstsrtp-enumtypes.h" #include "gstsrtpdec.h" #include GST_DEBUG_CATEGORY_STATIC (gst_srtp_dec_debug); #define GST_CAT_DEFAULT gst_srtp_dec_debug #define DEFAULT_REPLAY_WINDOW_SIZE 128 /* Filter signals and args */ enum { SIGNAL_REQUEST_KEY = 1, SIGNAL_CLEAR_KEYS, SIGNAL_SOFT_LIMIT, SIGNAL_HARD_LIMIT, SIGNAL_REMOVE_KEY, LAST_SIGNAL }; enum { PROP_0, PROP_REPLAY_WINDOW_SIZE }; /* the capabilities of the inputs and outputs. * * describe the real formats here. */ static GstStaticPadTemplate rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("rtp_sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-srtp") ); static GstStaticPadTemplate rtp_src_template = GST_STATIC_PAD_TEMPLATE ("rtp_src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate rtcp_sink_template = GST_STATIC_PAD_TEMPLATE ("rtcp_sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-srtcp") ); static GstStaticPadTemplate rtcp_src_template = GST_STATIC_PAD_TEMPLATE ("rtcp_src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtcp") ); static guint gst_srtp_dec_signals[LAST_SIGNAL] = { 0 }; G_DEFINE_TYPE (GstSrtpDec, gst_srtp_dec, GST_TYPE_ELEMENT); static void gst_srtp_dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_srtp_dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_srtp_dec_clear_streams (GstSrtpDec * filter); static void gst_srtp_dec_remove_stream (GstSrtpDec * filter, guint ssrc); static gboolean gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_srtp_dec_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_srtp_dec_sink_query_rtcp (GstPad * pad, GstObject * parent, GstQuery * query); static GstIterator *gst_srtp_dec_iterate_internal_links_rtp (GstPad * pad, GstObject * parent); static GstIterator *gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent); static GstFlowReturn gst_srtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf); static GstFlowReturn gst_srtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf); static GstStateChangeReturn gst_srtp_dec_change_state (GstElement * element, GstStateChange transition); static GstSrtpDecSsrcStream *request_key_with_signal (GstSrtpDec * filter, guint32 ssrc, gint signal); struct _GstSrtpDecSsrcStream { guint32 ssrc; guint32 roc; GstBuffer *key; GstSrtpCipherType rtp_cipher; GstSrtpAuthType rtp_auth; GstSrtpCipherType rtcp_cipher; GstSrtpAuthType rtcp_auth; }; #define STREAM_HAS_CRYPTO(stream) \ (stream->rtp_cipher != GST_SRTP_CIPHER_NULL || \ stream->rtcp_cipher != GST_SRTP_CIPHER_NULL || \ stream->rtp_auth != GST_SRTP_AUTH_NULL || \ stream->rtcp_auth != GST_SRTP_AUTH_NULL) /* initialize the srtpdec's class */ static void gst_srtp_dec_class_init (GstSrtpDecClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_srtp_dec_set_property; gobject_class->get_property = gst_srtp_dec_get_property; gst_element_class_add_static_pad_template (gstelement_class, &rtp_src_template); gst_element_class_add_static_pad_template (gstelement_class, &rtp_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &rtcp_src_template); gst_element_class_add_static_pad_template (gstelement_class, &rtcp_sink_template); gst_element_class_set_static_metadata (gstelement_class, "SRTP decoder", "Filter/Network/SRTP", "A SRTP and SRTCP decoder", "Gabriel Millaire "); /* Install callbacks */ gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_srtp_dec_change_state); klass->clear_streams = GST_DEBUG_FUNCPTR (gst_srtp_dec_clear_streams); klass->remove_stream = GST_DEBUG_FUNCPTR (gst_srtp_dec_remove_stream); /* Install properties */ g_object_class_install_property (gobject_class, PROP_REPLAY_WINDOW_SIZE, g_param_spec_uint ("replay-window-size", "Replay window size", "Size of the replay protection window", 64, 0x8000, DEFAULT_REPLAY_WINDOW_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /* Install signals */ /** * GstSrtpDec::request-key: * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * * Signal emited to get the parameters relevant to stream * with @ssrc. User should provide the key and the RTP and * RTCP encryption ciphers and authentication, and return * them wrapped in a GstCaps. */ gst_srtp_dec_signals[SIGNAL_REQUEST_KEY] = g_signal_new ("request-key", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstSrtpDec::clear-keys: * @gstsrtpdec: the element on which the signal is emitted * * Clear the internal list of streams */ gst_srtp_dec_signals[SIGNAL_CLEAR_KEYS] = g_signal_new ("clear-keys", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstSrtpDecClass, clear_streams), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstSrtpDec::soft-limit: * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * * Signal emited when the stream with @ssrc has reached the * soft limit of utilisation of it's master encryption key. * User should provide a new key and new RTP and RTCP encryption * ciphers and authentication, and return them wrapped in a * GstCaps. */ gst_srtp_dec_signals[SIGNAL_SOFT_LIMIT] = g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstSrtpDec::hard-limit: * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * * Signal emited when the stream with @ssrc has reached the * hard limit of utilisation of it's master encryption key. * User should provide a new key and new RTP and RTCP encryption * ciphers and authentication, and return them wrapped in a * GstCaps. If user could not provide those parameters or signal * is not answered, the buffers of this stream will be dropped. */ gst_srtp_dec_signals[SIGNAL_HARD_LIMIT] = g_signal_new ("hard-limit", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstSrtpDec::remove-key: * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The SSRC for which to remove the key. * * Removes keys for a specific SSRC */ gst_srtp_dec_signals[SIGNAL_REMOVE_KEY] = g_signal_new ("remove-key", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstSrtpDecClass, remove_stream), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); } /* initialize the new element * instantiate pads and add them to element * set pad calback functions * initialize instance structure */ static void gst_srtp_dec_init (GstSrtpDec * filter) { filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE; filter->rtp_sinkpad = gst_pad_new_from_static_template (&rtp_sink_template, "rtp_sink"); gst_pad_set_event_function (filter->rtp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtp)); gst_pad_set_query_function (filter->rtp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtp)); gst_pad_set_iterate_internal_links_function (filter->rtp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp)); gst_pad_set_chain_function (filter->rtp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtp)); filter->rtp_srcpad = gst_pad_new_from_static_template (&rtp_src_template, "rtp_src"); gst_pad_set_iterate_internal_links_function (filter->rtp_srcpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp)); gst_pad_set_element_private (filter->rtp_sinkpad, filter->rtp_srcpad); gst_pad_set_element_private (filter->rtp_srcpad, filter->rtp_sinkpad); gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_sinkpad); gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_srcpad); filter->rtcp_sinkpad = gst_pad_new_from_static_template (&rtcp_sink_template, "rtcp_sink"); gst_pad_set_event_function (filter->rtcp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtcp)); gst_pad_set_query_function (filter->rtcp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtcp)); gst_pad_set_iterate_internal_links_function (filter->rtcp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp)); gst_pad_set_chain_function (filter->rtcp_sinkpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtcp)); filter->rtcp_srcpad = gst_pad_new_from_static_template (&rtcp_src_template, "rtcp_src"); gst_pad_set_iterate_internal_links_function (filter->rtcp_srcpad, GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp)); gst_pad_set_element_private (filter->rtcp_sinkpad, filter->rtcp_srcpad); gst_pad_set_element_private (filter->rtcp_srcpad, filter->rtcp_sinkpad); gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_sinkpad); gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_srcpad); filter->first_session = TRUE; filter->roc_changed = FALSE; } static void gst_srtp_dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstSrtpDec *filter = GST_SRTP_DEC (object); GST_OBJECT_LOCK (filter); switch (prop_id) { case PROP_REPLAY_WINDOW_SIZE: filter->replay_window_size = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (filter); } static void gst_srtp_dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstSrtpDec *filter = GST_SRTP_DEC (object); GST_OBJECT_LOCK (filter); switch (prop_id) { case PROP_REPLAY_WINDOW_SIZE: g_value_set_uint (value, filter->replay_window_size); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } GST_OBJECT_UNLOCK (filter); } static void gst_srtp_dec_remove_stream (GstSrtpDec * filter, guint ssrc) { GstSrtpDecSsrcStream *stream = NULL; if (filter->streams == NULL) return; stream = g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc)); if (stream) { srtp_remove_stream (filter->session, ssrc); g_hash_table_remove (filter->streams, GUINT_TO_POINTER (ssrc)); } } static GstSrtpDecSsrcStream * find_stream_by_ssrc (GstSrtpDec * filter, guint32 ssrc) { return g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc)); } /* get info from buffer caps */ static GstSrtpDecSsrcStream * get_stream_from_caps (GstSrtpDec * filter, GstCaps * caps, guint32 ssrc) { GstSrtpDecSsrcStream *stream; GstStructure *s; GstBuffer *buf; const gchar *rtp_cipher, *rtp_auth, *rtcp_cipher, *rtcp_auth; /* Create new stream structure and set default values */ stream = g_slice_new0 (GstSrtpDecSsrcStream); stream->ssrc = ssrc; stream->key = NULL; /* Get info from caps */ s = gst_caps_get_structure (caps, 0); if (!s) goto error; rtp_cipher = gst_structure_get_string (s, "srtp-cipher"); rtp_auth = gst_structure_get_string (s, "srtp-auth"); rtcp_cipher = gst_structure_get_string (s, "srtcp-cipher"); rtcp_auth = gst_structure_get_string (s, "srtcp-auth"); if (!rtp_cipher || !rtp_auth || !rtcp_cipher || !rtcp_auth) goto error; gst_structure_get_uint (s, "roc", &stream->roc); stream->rtp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE, rtp_cipher); stream->rtp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtp_auth); stream->rtcp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE, rtcp_cipher); stream->rtcp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtcp_auth); if ((gint) stream->rtp_cipher == -1 || (gint) stream->rtp_auth == -1 || (gint) stream->rtcp_cipher == -1 || (gint) stream->rtcp_auth == -1) { GST_WARNING_OBJECT (filter, "Invalid caps for stream," " unknown cipher or auth type"); goto error; } if (stream->rtcp_cipher != NULL_CIPHER && stream->rtcp_auth == NULL_AUTH) { GST_WARNING_OBJECT (filter, "Cannot have SRTP NULL authentication with a not-NULL encryption" " cipher."); goto error; } if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) { GST_DEBUG_OBJECT (filter, "Got key [%p] for SSRC %u", buf, ssrc); stream->key = buf; } else if (STREAM_HAS_CRYPTO (stream)) { goto error; } return stream; error: g_slice_free (GstSrtpDecSsrcStream, stream); return NULL; } /* Get SRTP params by signal */ static GstCaps * signal_get_srtp_params (GstSrtpDec * filter, guint32 ssrc, gint signal) { GstCaps *caps = NULL; g_signal_emit (filter, gst_srtp_dec_signals[signal], 0, ssrc, &caps); if (caps != NULL) GST_DEBUG_OBJECT (filter, "Caps received"); return caps; } /* Create a stream in the session */ static err_status_t init_session_stream (GstSrtpDec * filter, guint32 ssrc, GstSrtpDecSsrcStream * stream) { err_status_t ret; srtp_policy_t policy; GstMapInfo map; guchar tmp[1]; memset (&policy, 0, sizeof (srtp_policy_t)); if (!stream) return err_status_bad_param; GST_INFO_OBJECT (filter, "Setting RTP policy..."); set_crypto_policy_cipher_auth (stream->rtp_cipher, stream->rtp_auth, &policy.rtp); GST_INFO_OBJECT (filter, "Setting RTCP policy..."); set_crypto_policy_cipher_auth (stream->rtcp_cipher, stream->rtcp_auth, &policy.rtcp); if (stream->key) { gst_buffer_map (stream->key, &map, GST_MAP_READ); policy.key = (guchar *) map.data; } else { policy.key = tmp; } policy.ssrc.value = ssrc; policy.ssrc.type = ssrc_specific; policy.window_size = filter->replay_window_size; policy.next = NULL; /* If it is the first stream, create the session * If not, add the stream policy to the session */ if (filter->first_session) ret = srtp_create (&filter->session, &policy); else ret = srtp_add_stream (filter->session, &policy); if (stream->key) gst_buffer_unmap (stream->key, &map); if (ret == err_status_ok) { srtp_stream_t srtp_stream; srtp_stream = srtp_get_stream (filter->session, htonl (ssrc)); if (srtp_stream) { /* Here, we just set the ROC, but we also need to set the initial * RTP sequence number later, otherwise libsrtp will not be able * to get the right packet index. */ rdbx_set_roc (&srtp_stream->rtp_rdbx, stream->roc); filter->roc_changed = TRUE; } filter->first_session = FALSE; g_hash_table_insert (filter->streams, GUINT_TO_POINTER (stream->ssrc), stream); } return ret; } /* Return a stream structure for a given buffer */ static GstSrtpDecSsrcStream * validate_buffer (GstSrtpDec * filter, GstBuffer * buf, guint32 * ssrc, gboolean * is_rtcp) { GstSrtpDecSsrcStream *stream = NULL; GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT; if (gst_rtp_buffer_map (buf, GST_MAP_READ | GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING, &rtpbuf)) { if (gst_rtp_buffer_get_payload_type (&rtpbuf) < 64 || gst_rtp_buffer_get_payload_type (&rtpbuf) > 80) { *ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf); gst_rtp_buffer_unmap (&rtpbuf); *is_rtcp = FALSE; goto have_ssrc; } gst_rtp_buffer_unmap (&rtpbuf); } if (rtcp_buffer_get_ssrc (buf, ssrc)) { *is_rtcp = TRUE; } else { GST_WARNING_OBJECT (filter, "No SSRC found in buffer"); return NULL; } have_ssrc: stream = find_stream_by_ssrc (filter, *ssrc); if (stream) return stream; return request_key_with_signal (filter, *ssrc, SIGNAL_REQUEST_KEY); } static void free_stream (GstSrtpDecSsrcStream * stream) { if (stream->key) gst_buffer_unref (stream->key); g_slice_free (GstSrtpDecSsrcStream, stream); } /* Create new stream from params in caps */ static GstSrtpDecSsrcStream * update_session_stream_from_caps (GstSrtpDec * filter, guint32 ssrc, GstCaps * caps) { GstSrtpDecSsrcStream *stream = NULL; GstSrtpDecSsrcStream *old_stream = NULL; err_status_t err; g_return_val_if_fail (GST_IS_SRTP_DEC (filter), NULL); g_return_val_if_fail (GST_IS_CAPS (caps), NULL); stream = get_stream_from_caps (filter, caps, ssrc); old_stream = find_stream_by_ssrc (filter, ssrc); if (stream && old_stream && stream->rtp_cipher == old_stream->rtp_cipher && stream->rtcp_cipher == old_stream->rtcp_cipher && stream->rtp_auth == old_stream->rtp_auth && stream->rtcp_auth == old_stream->rtcp_auth && stream->key && old_stream->key && gst_buffer_get_size (stream->key) == gst_buffer_get_size (old_stream->key)) { GstMapInfo info; if (gst_buffer_map (old_stream->key, &info, GST_MAP_READ)) { gboolean equal; equal = (gst_buffer_memcmp (stream->key, 0, info.data, info.size) == 0); gst_buffer_unmap (old_stream->key, &info); if (equal) { free_stream (stream); return old_stream; } } } /* Remove existing stream, if any */ gst_srtp_dec_remove_stream (filter, ssrc); if (stream) { /* Create new session stream */ err = init_session_stream (filter, ssrc, stream); if (err != err_status_ok) { if (stream->key) gst_buffer_unref (stream->key); g_slice_free (GstSrtpDecSsrcStream, stream); stream = NULL; } } return stream; } static gboolean remove_yes (gpointer key, gpointer value, gpointer user_data) { return TRUE; } /* Clear the policy list */ static void gst_srtp_dec_clear_streams (GstSrtpDec * filter) { guint nb = 0; GST_OBJECT_LOCK (filter); if (!filter->first_session) srtp_dealloc (filter->session); if (filter->streams) nb = g_hash_table_foreach_remove (filter->streams, remove_yes, NULL); filter->first_session = TRUE; GST_OBJECT_UNLOCK (filter); GST_DEBUG_OBJECT (filter, "Cleared %d streams", nb); } /* Send a signal */ static GstSrtpDecSsrcStream * request_key_with_signal (GstSrtpDec * filter, guint32 ssrc, gint signal) { GstCaps *caps; GstSrtpDecSsrcStream *stream = NULL; caps = signal_get_srtp_params (filter, ssrc, signal); if (caps) { stream = update_session_stream_from_caps (filter, ssrc, caps); if (stream) GST_DEBUG_OBJECT (filter, "New stream set with SSRC %u", ssrc); else GST_WARNING_OBJECT (filter, "Could not set stream with SSRC %u", ssrc); gst_caps_unref (caps); } else { GST_WARNING_OBJECT (filter, "Could not get caps for stream with SSRC %u", ssrc); } return stream; } static gboolean gst_srtp_dec_sink_setcaps (GstPad * pad, GstObject * parent, GstCaps * caps, gboolean is_rtcp) { GstSrtpDec *filter = GST_SRTP_DEC (parent); GstPad *otherpad; GstStructure *ps; gboolean ret = FALSE; g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE); ps = gst_caps_get_structure (caps, 0); if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT) && gst_structure_has_field_typed (ps, "srtp-cipher", G_TYPE_STRING) && gst_structure_has_field_typed (ps, "srtp-auth", G_TYPE_STRING) && gst_structure_has_field_typed (ps, "srtcp-cipher", G_TYPE_STRING) && gst_structure_has_field_typed (ps, "srtcp-auth", G_TYPE_STRING)) { guint ssrc; gst_structure_get_uint (ps, "ssrc", &ssrc); if (!update_session_stream_from_caps (filter, ssrc, caps)) { GST_WARNING_OBJECT (pad, "Could not create session from pad caps: %" GST_PTR_FORMAT, caps); return FALSE; } } caps = gst_caps_copy (caps); ps = gst_caps_get_structure (caps, 0); gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth", "srtcp-cipher", "srtcp-auth", NULL); if (is_rtcp) gst_structure_set_name (ps, "application/x-rtcp"); else gst_structure_set_name (ps, "application/x-rtp"); otherpad = gst_pad_get_element_private (pad); ret = gst_pad_set_caps (otherpad, caps); gst_caps_unref (caps); return ret; } static gboolean gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean ret; GstCaps *caps; GstSrtpDec *filter = GST_SRTP_DEC (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: gst_event_parse_caps (event, &caps); ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, FALSE); gst_event_unref (event); return ret; case GST_EVENT_SEGMENT: /* Make sure to send a caps event downstream before the segment event, * even if upstream didn't */ if (!gst_pad_has_current_caps (filter->rtp_srcpad)) { GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtp"); gst_pad_set_caps (filter->rtp_srcpad, caps); gst_caps_unref (caps); } filter->rtp_has_segment = TRUE; break; case GST_EVENT_FLUSH_STOP: filter->rtp_has_segment = FALSE; break; default: break; } return gst_pad_event_default (pad, parent, event); } static gboolean gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean ret; GstCaps *caps; GstSrtpDec *filter = GST_SRTP_DEC (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: gst_event_parse_caps (event, &caps); ret = gst_srtp_dec_sink_setcaps (pad, parent, caps, TRUE); gst_event_unref (event); return ret; case GST_EVENT_SEGMENT: /* Make sure to send a caps event downstream before the segment event, * even if upstream didn't */ if (!gst_pad_has_current_caps (filter->rtcp_srcpad)) { GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp"); gst_pad_set_caps (filter->rtcp_srcpad, caps); gst_caps_unref (caps); } filter->rtcp_has_segment = TRUE; break; case GST_EVENT_FLUSH_STOP: filter->rtcp_has_segment = FALSE; break; default: break; } return gst_pad_event_default (pad, parent, event); } static gboolean gst_srtp_dec_sink_query (GstPad * pad, GstObject * parent, GstQuery * query, gboolean is_rtcp) { switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter = NULL; GstCaps *other_filter = NULL; GstCaps *template_caps; GstPad *otherpad; GstCaps *other_caps; GstCaps *ret; int i; gst_query_parse_caps (query, &filter); otherpad = (GstPad *) gst_pad_get_element_private (pad); if (filter) { other_filter = gst_caps_copy (filter); for (i = 0; i < gst_caps_get_size (other_filter); i++) { GstStructure *ps = gst_caps_get_structure (other_filter, i); if (is_rtcp) gst_structure_set_name (ps, "application/x-rtcp"); else gst_structure_set_name (ps, "application/x-rtp"); gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth", "srtcp-cipher", "srtcp-auth", NULL); } } other_caps = gst_pad_peer_query_caps (otherpad, other_filter); if (other_filter) gst_caps_unref (other_filter); if (!other_caps) { goto return_template; } template_caps = gst_pad_get_pad_template_caps (otherpad); ret = gst_caps_intersect_full (other_caps, template_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (other_caps); gst_caps_unref (template_caps); ret = gst_caps_make_writable (ret); for (i = 0; i < gst_caps_get_size (ret); i++) { GstStructure *ps = gst_caps_get_structure (ret, i); if (is_rtcp) gst_structure_set_name (ps, "application/x-srtcp"); else gst_structure_set_name (ps, "application/x-srtp"); } if (filter) { GstCaps *tmp; tmp = gst_caps_intersect (ret, filter); gst_caps_unref (ret); ret = tmp; } gst_query_set_caps_result (query, ret); gst_caps_unref (ret); return TRUE; return_template: ret = gst_pad_get_pad_template_caps (pad); gst_query_set_caps_result (query, ret); gst_caps_unref (ret); return TRUE; } default: return gst_pad_query_default (pad, parent, query); } } static gboolean gst_srtp_dec_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query) { return gst_srtp_dec_sink_query (pad, parent, query, FALSE); } static gboolean gst_srtp_dec_sink_query_rtcp (GstPad * pad, GstObject * parent, GstQuery * query) { return gst_srtp_dec_sink_query (pad, parent, query, TRUE); } static GstIterator * gst_srtp_dec_iterate_internal_links (GstPad * pad, GstObject * parent, gboolean is_rtcp) { GstSrtpDec *filter = GST_SRTP_DEC (parent); GstPad *otherpad = NULL; GstIterator *it = NULL; otherpad = (GstPad *) gst_pad_get_element_private (pad); if (otherpad) { GValue val = { 0 }; g_value_init (&val, GST_TYPE_PAD); g_value_set_object (&val, otherpad); it = gst_iterator_new_single (GST_TYPE_PAD, &val); g_value_unset (&val); } else { GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), CORE, PAD, (NULL), ("Unable to get linked pad")); } return it; } static GstIterator * gst_srtp_dec_iterate_internal_links_rtp (GstPad * pad, GstObject * parent) { return gst_srtp_dec_iterate_internal_links (pad, parent, FALSE); } static GstIterator * gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent) { return gst_srtp_dec_iterate_internal_links (pad, parent, TRUE); } static void gst_srtp_dec_push_early_events (GstSrtpDec * filter, GstPad * pad, GstPad * otherpad, gboolean is_rtcp) { GstEvent *otherev, *ev; ev = gst_pad_get_sticky_event (pad, GST_EVENT_STREAM_START, 0); if (ev) { gst_event_unref (ev); } else { gchar *new_stream_id; otherev = gst_pad_get_sticky_event (otherpad, GST_EVENT_STREAM_START, 0); if (otherev) { const gchar *other_stream_id; gst_event_parse_stream_start (otherev, &other_stream_id); new_stream_id = g_strdup_printf ("%s/%s", other_stream_id, is_rtcp ? "rtcp" : "rtp"); gst_event_unref (otherev); } else { new_stream_id = gst_pad_create_stream_id (pad, GST_ELEMENT (filter), is_rtcp ? "rtcp" : "rtp"); } ev = gst_event_new_stream_start (new_stream_id); g_free (new_stream_id); gst_pad_push_event (pad, ev); } ev = gst_pad_get_sticky_event (pad, GST_EVENT_CAPS, 0); if (ev) { gst_event_unref (ev); } else { GstCaps *caps; if (is_rtcp) caps = gst_caps_new_empty_simple ("application/x-rtcp"); else caps = gst_caps_new_empty_simple ("application/x-rtp"); gst_pad_set_caps (pad, caps); gst_caps_unref (caps); } ev = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0); if (ev) { gst_event_unref (ev); } else { ev = gst_pad_get_sticky_event (otherpad, GST_EVENT_SEGMENT, 0); if (ev) gst_pad_push_event (pad, ev); } if (is_rtcp) filter->rtcp_has_segment = TRUE; else filter->rtp_has_segment = TRUE; } /* * This function should be called while holding the filter lock */ static gboolean gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf, gboolean is_rtcp, guint32 ssrc) { GstMapInfo map; err_status_t err; gint size; GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT " with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf), ssrc); /* Change buffer to remove protection */ buf = gst_buffer_make_writable (buf); gst_buffer_map (buf, &map, GST_MAP_READWRITE); size = map.size; unprotect: gst_srtp_init_event_reporter (); if (is_rtcp) err = srtp_unprotect_rtcp (filter->session, map.data, &size); else { /* If ROC has changed, we know we need to set the initial RTP * sequence number too. */ if (filter->roc_changed) { srtp_stream_t stream; stream = srtp_get_stream (filter->session, htonl (ssrc)); if (stream) { guint16 seqnum = 0; GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT; gst_rtp_buffer_map (buf, GST_MAP_READ | GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING, &rtpbuf); seqnum = gst_rtp_buffer_get_seq (&rtpbuf); gst_rtp_buffer_unmap (&rtpbuf); /* We finally add the RTP sequence number to the current * rollover counter. */ stream->rtp_rdbx.index &= ~0xFFFF; stream->rtp_rdbx.index |= seqnum; } filter->roc_changed = FALSE; } err = srtp_unprotect (filter->session, map.data, &size); } GST_OBJECT_UNLOCK (filter); if (err != err_status_ok) { GST_WARNING_OBJECT (pad, "Unable to unprotect buffer (unprotect failed code %d)", err); /* Signal user depending on type of error */ switch (err) { case err_status_key_expired: GST_OBJECT_LOCK (filter); /* Update stream */ if (find_stream_by_ssrc (filter, ssrc)) { GST_OBJECT_UNLOCK (filter); if (request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT)) { GST_OBJECT_LOCK (filter); goto unprotect; } else { GST_WARNING_OBJECT (filter, "Hard limit reached, no new key, " "dropping"); } } else { GST_WARNING_OBJECT (filter, "Could not find matching stream, " "dropping"); } break; case err_status_auth_fail: GST_WARNING_OBJECT (filter, "Error authentication packet, dropping"); break; case err_status_cipher_fail: GST_WARNING_OBJECT (filter, "Error while decrypting packet, dropping"); break; default: GST_WARNING_OBJECT (filter, "Other error, dropping"); break; } gst_buffer_unmap (buf, &map); GST_OBJECT_LOCK (filter); return FALSE; } gst_buffer_unmap (buf, &map); gst_buffer_set_size (buf, size); GST_OBJECT_LOCK (filter); return TRUE; } static GstFlowReturn gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf, gboolean is_rtcp) { GstSrtpDec *filter = GST_SRTP_DEC (parent); GstPad *otherpad; GstSrtpDecSsrcStream *stream = NULL; GstFlowReturn ret = GST_FLOW_OK; guint32 ssrc = 0; GST_OBJECT_LOCK (filter); /* Check if this stream exists, if not create a new stream */ if (!(stream = validate_buffer (filter, buf, &ssrc, &is_rtcp))) { GST_OBJECT_UNLOCK (filter); GST_WARNING_OBJECT (filter, "Invalid buffer, dropping"); goto drop_buffer; } if (!STREAM_HAS_CRYPTO (stream)) { GST_OBJECT_UNLOCK (filter); goto push_out; } if (!gst_srtp_dec_decode_buffer (filter, pad, buf, is_rtcp, ssrc)) { GST_OBJECT_UNLOCK (filter); goto drop_buffer; } GST_OBJECT_UNLOCK (filter); /* If all is well, we may have reached soft limit */ if (gst_srtp_get_soft_limit_reached ()) request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT); push_out: /* Push buffer to source pad */ if (is_rtcp) { otherpad = filter->rtcp_srcpad; if (!filter->rtcp_has_segment) gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad, filter->rtp_srcpad, TRUE); } else { otherpad = filter->rtp_srcpad; if (!filter->rtp_has_segment) gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad, filter->rtcp_srcpad, FALSE); } ret = gst_pad_push (otherpad, buf); return ret; drop_buffer: /* Drop buffer, except if gst_pad_push returned OK or an error */ gst_buffer_unref (buf); return ret; } static GstFlowReturn gst_srtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf) { return gst_srtp_dec_chain (pad, parent, buf, FALSE); } static GstFlowReturn gst_srtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf) { return gst_srtp_dec_chain (pad, parent, buf, TRUE); } static GstStateChangeReturn gst_srtp_dec_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn res; GstSrtpDec *filter; filter = GST_SRTP_DEC (element); GST_OBJECT_LOCK (filter); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: filter->streams = g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL, (GDestroyNotify) free_stream); filter->rtp_has_segment = FALSE; filter->rtcp_has_segment = FALSE; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; default: break; } GST_OBJECT_UNLOCK (filter); res = GST_ELEMENT_CLASS (gst_srtp_dec_parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_srtp_dec_clear_streams (filter); g_hash_table_unref (filter->streams); filter->streams = NULL; break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; } /* entry point to initialize the plug-in * initialize the plug-in itself * register the element factories and other features */ gboolean gst_srtp_dec_plugin_init (GstPlugin * srtpdec) { GST_DEBUG_CATEGORY_INIT (gst_srtp_dec_debug, "srtpdec", 0, "SRTP dec"); return gst_element_register (srtpdec, "srtpdec", GST_RANK_NONE, GST_TYPE_SRTP_DEC); }