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gstreamer/ext/srtp/gstsrtpenc.c
Aleix Conchillo Flaqué da30669589 srtp: add support for rollover counters and replay protection window size 
We add a new signal, get-rollover-counter, to the SRTP encoder. Given a
ssrc the signal will return the currently internal SRTP rollover counter
for the given stream.

For the SRTP decoder we have a new SRTP caps parameter "roc" that needs
to be set when a new SRTP stream is created for a given SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=726861
2014-06-03 16:18:25 -04:00

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/*
* GStreamer - GStreamer SRTP encoder
*
* Copyright 2009-2011 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.com>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* 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:gst-plugin-bad-plugins-srtpenc
* @see_also: srtpdec
*
* gstrtpenc acts as an encoder that adds security to RTP and RTCP
* packets in the form of encryption and authentication. It outs SRTP
* and SRTCP.
*
* An application can request multiple RTP and RTCP pads to protect,
* but every sink pad requested must receive packets from the same
* source (identical SSRC). If a packet received contains a different
* SSRC, a warning is emited and the valid SSRC is forced on the packet.
*
* This element uses libsrtp library. When receiving the first packet,
* the library is initialized with a new stream (based on the SSRC). It
* uses the default RTP and RTCP encryption and authentication mechanisms,
* unless the user has set the relevant properties first. It also uses
* a master key that MUST be set by property (key) at the beginning. The
* master key must be of a maximum length of 46 characters (14 characters
* for the salt plus the key). The encryption and authentication mecanisms
* available are :
*
* Encryption (properties rtp-cipher and rtcp-cipher)
* - AES_ICM 256 bits (maximum security)
* - AES_ICM 128 bits (default)
* - NULL
*
* Authentication (properties rtp-auth and rtcp-auth)
* - 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).
*
* When requested to create a sink pad, a linked source pad is created.
* Each packet received is first analysed (checked for valid SSRC) then
* its buffer is protected 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. The packets pushed on the source
* pad are of type 'application/x-srtp' or 'application/x-srtcp'.
*
* When the maximum usage of the master key is reached, a soft-limit
* signal is sent to the user. The user must then set a new master key
* by property. 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 it is also
* possible to request the internal SRTP rollover counter for a given
* SSRC. The rollover counter should be then transmitted and used by the
* clients to authenticate and decrypt the packets. Failing to do that
* the clients will start with a rollover counter of 0 which will
* probably be incorrect if the stream has been transmitted for a
* while to other clients.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <string.h>
#include "gstsrtpenc.h"
#include "gstsrtp.h"
#include "gstsrtp-enumtypes.h"
#include <srtp/srtp_priv.h>
GST_DEBUG_CATEGORY_STATIC (gst_srtp_enc_debug);
#define GST_CAT_DEFAULT gst_srtp_enc_debug
/* 128 bit key size: 14 (salt) + 16 */
#define MASTER_128_KEY_SIZE 30
/* 256 bit key size: 14 (salt) + 16 + 16 */
#define MASTER_256_KEY_SIZE 46
/* Properties default values */
#define DEFAULT_MASTER_KEY NULL
#define DEFAULT_RTP_CIPHER GST_SRTP_CIPHER_AES_128_ICM
#define DEFAULT_RTP_AUTH GST_SRTP_AUTH_HMAC_SHA1_80
#define DEFAULT_RTCP_CIPHER DEFAULT_RTP_CIPHER
#define DEFAULT_RTCP_AUTH DEFAULT_RTP_AUTH
#define DEFAULT_RANDOM_KEY FALSE
#define DEFAULT_REPLAY_WINDOW_SIZE 128
#define HAS_CRYPTO(filter) (filter->rtp_cipher != GST_SRTP_CIPHER_NULL || \
filter->rtcp_cipher != GST_SRTP_CIPHER_NULL || \
filter->rtp_auth != GST_SRTP_AUTH_NULL || \
filter->rtcp_auth != GST_SRTP_AUTH_NULL)
/* Filter signals and args */
enum
{
SIGNAL_SOFT_LIMIT,
SIGNAL_GET_ROLLOVER_COUNTER,
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MKEY,
PROP_RTP_CIPHER,
PROP_RTP_AUTH,
PROP_RTCP_CIPHER,
PROP_RTCP_AUTH,
PROP_RANDOM_KEY,
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_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-srtp")
);
static GstStaticPadTemplate rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-srtcp")
);
G_DEFINE_TYPE (GstSrtpEnc, gst_srtp_enc, GST_TYPE_ELEMENT);
static guint gst_srtp_enc_signals[LAST_SIGNAL] = { 0 };
static void gst_srtp_enc_dispose (GObject * object);
static void gst_srtp_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_srtp_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_srtp_enc_sink_query_rtp (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_srtp_enc_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstIterator *gst_srtp_enc_iterate_internal_links_rtp (GstPad * pad,
GstObject * parent);
static GstIterator *gst_srtp_enc_iterate_internal_links_rtcp (GstPad * pad,
GstObject * parent);
static GstFlowReturn gst_srtp_enc_chain_rtp (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static GstFlowReturn gst_srtp_enc_chain_rtcp (GstPad * pad, GstObject * parent,
GstBuffer * buf);
static gboolean gst_srtp_enc_sink_event_rtp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_enc_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstStateChangeReturn gst_srtp_enc_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_srtp_enc_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_srtp_enc_release_pad (GstElement * element, GstPad * pad);
static guint32
gst_srtp_enc_get_rollover_counter (GstSrtpEnc * filter, guint32 ssrc)
{
guint32 roc = 0;
srtp_stream_t stream;
GST_OBJECT_LOCK (filter);
GST_DEBUG_OBJECT (filter, "retrieving SRTP Rollover Counter, ssrc: %u", ssrc);
if (filter->session) {
stream = srtp_get_stream (filter->session, htonl (ssrc));
if (stream)
roc = stream->rtp_rdbx.index >> 16;
}
GST_OBJECT_UNLOCK (filter);
return roc;
}
/* initialize the srtpenc's class
*/
static void
gst_srtp_enc_class_init (GstSrtpEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_sink_template));
gst_element_class_set_static_metadata (gstelement_class, "SRTP encoder",
"Filter/Network/SRTP",
"A SRTP and SRTCP encoder",
"Gabriel Millaire <millaire.gabriel@collabora.com>");
/* Install callbacks */
gobject_class->set_property = gst_srtp_enc_set_property;
gobject_class->get_property = gst_srtp_enc_get_property;
gobject_class->dispose = gst_srtp_enc_dispose;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_srtp_enc_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_srtp_enc_release_pad);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_srtp_enc_change_state);
/* Install properties */
g_object_class_install_property (gobject_class, PROP_MKEY,
g_param_spec_boxed ("key", "Key", "Master key (minimum of "
G_STRINGIFY (MASTER_128_KEY_SIZE) " and maximum of "
G_STRINGIFY (MASTER_256_KEY_SIZE) " bytes)",
GST_TYPE_BUFFER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
g_object_class_install_property (gobject_class, PROP_RTP_CIPHER,
g_param_spec_enum ("rtp-cipher", "RTP Cipher", "RTP Cipher",
GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTP_CIPHER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTP_AUTH,
g_param_spec_enum ("rtp-auth", "RTP Authentication",
"RTP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTP_AUTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_CIPHER,
g_param_spec_enum ("rtcp-cipher", "RTCP Cipher",
"RTCP Cipher", GST_TYPE_SRTP_CIPHER_TYPE, DEFAULT_RTCP_CIPHER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_AUTH,
g_param_spec_enum ("rtcp-auth", "RTCP Authentication",
"RTCP Authentication", GST_TYPE_SRTP_AUTH_TYPE, DEFAULT_RTCP_AUTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RANDOM_KEY,
g_param_spec_boolean ("random-key", "Generate random key",
"Generate a random key if TRUE",
DEFAULT_RANDOM_KEY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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));
/**
* GstSrtpEnc::soft-limit:
* @gstsrtpenc: the element on which the signal is emitted
*
* 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 by setting the #GstSrtpEnc:key property.
*/
gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT] =
g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
/**
* GstSrtpEnc::get-rollover-counter:
* @gstsrtpenc: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Request the SRTP rollover counter for the stream with @ssrc.
*/
gst_srtp_enc_signals[SIGNAL_GET_ROLLOVER_COUNTER] =
g_signal_new ("get-rollover-counter", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstSrtpEncClass,
get_rollover_counter), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_UINT, 1, G_TYPE_UINT);
klass->get_rollover_counter =
GST_DEBUG_FUNCPTR (gst_srtp_enc_get_rollover_counter);
}
/* initialize the new element
*/
static void
gst_srtp_enc_init (GstSrtpEnc * filter)
{
filter->key_changed = TRUE;
filter->first_session = TRUE;
filter->key = DEFAULT_MASTER_KEY;
filter->rtp_cipher = DEFAULT_RTP_CIPHER;
filter->rtp_auth = DEFAULT_RTP_AUTH;
filter->rtcp_cipher = DEFAULT_RTCP_CIPHER;
filter->rtcp_auth = DEFAULT_RTCP_AUTH;
filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE;
}
static guint
max_cipher_key_size (GstSrtpEnc * filter)
{
guint rtp_size, rtcp_size;
rtp_size = cipher_key_size (filter->rtp_cipher);
rtcp_size = cipher_key_size (filter->rtcp_cipher);
return (rtp_size > rtcp_size) ? rtp_size : rtcp_size;
}
/* Create stream
*/
static err_status_t
gst_srtp_enc_create_session (GstSrtpEnc * filter)
{
err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
guchar tmp[1];
memset (&policy, 0, sizeof (srtp_policy_t));
GST_OBJECT_LOCK (filter);
if (HAS_CRYPTO (filter)) {
guint expected;
gsize keysize;
if (filter->key == NULL) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("Cipher is not NULL, key must be set"),
("Cipher is not NULL, key must be set"));
return FALSE;
}
expected = max_cipher_key_size (filter);
keysize = gst_buffer_get_size (filter->key);
if (expected != keysize) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("Master key size is wrong"),
("Expected master key of %d bytes, but received %" G_GSIZE_FORMAT
" bytes", expected, keysize));
return FALSE;
}
}
GST_DEBUG_OBJECT (filter, "Setting RTP/RTCP policy to %d / %d",
filter->rtp_cipher, filter->rtcp_cipher);
set_crypto_policy_cipher_auth (filter->rtp_cipher, filter->rtp_auth,
&policy.rtp);
set_crypto_policy_cipher_auth (filter->rtcp_cipher, filter->rtcp_auth,
&policy.rtcp);
if (HAS_CRYPTO (filter)) {
gst_buffer_map (filter->key, &map, GST_MAP_READ);
policy.key = (guchar *) map.data;
} else {
policy.key = tmp;
}
policy.ssrc.value = 0;
policy.ssrc.type = ssrc_any_outbound;
policy.next = NULL;
policy.window_size = filter->replay_window_size;
/* If it is the first stream, create the session
* If not, add the stream to the session
*/
ret = srtp_create (&filter->session, &policy);
filter->first_session = FALSE;
if (HAS_CRYPTO (filter))
gst_buffer_unmap (filter->key, &map);
GST_OBJECT_UNLOCK (filter);
return ret;
}
/* Release ressources and set default values
*/
static void
gst_srtp_enc_reset (GstSrtpEnc * filter)
{
GST_OBJECT_LOCK (filter);
if (!filter->first_session)
srtp_dealloc (filter->session);
filter->first_session = TRUE;
filter->key_changed = FALSE;
GST_OBJECT_UNLOCK (filter);
}
/* Create sinkpad to receive RTP packets from encers
* and a srcpad for the RTP packets
*/
static GstPad *
create_rtp_sink (GstSrtpEnc * filter, const gchar * name)
{
GstPad *sinkpad, *srcpad;
gchar *sinkpadname, *srcpadname;
guint nb = 0;
GST_DEBUG_OBJECT (filter, "creating RTP sink pad");
sinkpad = gst_pad_new_from_static_template (&rtp_sink_template, name);
sinkpadname = gst_pad_get_name (sinkpad);
sscanf (sinkpadname, "rtp_sink_%u", &nb);
srcpadname = g_strdup_printf ("rtp_src_%u", nb);
GST_DEBUG_OBJECT (filter, "creating RTP source pad");
srcpad = gst_pad_new_from_static_template (&rtp_src_template, srcpadname);
g_free (srcpadname);
g_free (sinkpadname);
gst_pad_set_element_private (sinkpad, srcpad);
gst_pad_set_element_private (srcpad, sinkpad);
gst_pad_set_query_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_query_rtp));
gst_pad_set_iterate_internal_links_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtp));
gst_pad_set_chain_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_rtp));
gst_pad_set_event_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_event_rtp));
gst_pad_set_active (sinkpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), sinkpad);
gst_pad_set_iterate_internal_links_function (srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtp));
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), srcpad);
return sinkpad;
}
/* Create sinkpad to receive RTCP packets from encers
* and a srcpad for the RTCP packets
*/
static GstPad *
create_rtcp_sink (GstSrtpEnc * filter, const gchar * name)
{
GstPad *srcpad, *sinkpad;
gchar *sinkpadname, *srcpadname;
guint nb = 0;
GST_DEBUG_OBJECT (filter, "creating RTCP sink pad");
sinkpad = gst_pad_new_from_static_template (&rtcp_sink_template, name);
sinkpadname = gst_pad_get_name (sinkpad);
sscanf (sinkpadname, "rtcp_sink_%u", &nb);
srcpadname = g_strdup_printf ("rtcp_src_%u", nb);
GST_DEBUG_OBJECT (filter, "creating RTCP source pad");
srcpad = gst_pad_new_from_static_template (&rtcp_src_template, srcpadname);
g_free (srcpadname);
g_free (sinkpadname);
gst_pad_set_element_private (sinkpad, srcpad);
gst_pad_set_element_private (srcpad, sinkpad);
gst_pad_set_query_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_query_rtcp));
gst_pad_set_iterate_internal_links_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtcp));
gst_pad_set_chain_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_chain_rtcp));
gst_pad_set_event_function (sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_sink_event_rtcp));
gst_pad_set_active (sinkpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), sinkpad);
gst_pad_set_iterate_internal_links_function (srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_enc_iterate_internal_links_rtcp));
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (filter), srcpad);
return sinkpad;
}
/* Handling new pad request
*/
static GstPad *
gst_srtp_enc_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstElementClass *klass;
GstSrtpEnc *filter;
filter = GST_SRTP_ENC (element);
klass = GST_ELEMENT_GET_CLASS (element);
GST_INFO_OBJECT (element, "New pad requested");
if (templ == gst_element_class_get_pad_template (klass, "rtp_sink_%u"))
return create_rtp_sink (filter, name);
if (templ == gst_element_class_get_pad_template (klass, "rtcp_sink_%u"))
return create_rtcp_sink (filter, name);
GST_ERROR_OBJECT (element, "Could not find specified template");
return NULL;
}
/* Dispose
*/
static void
gst_srtp_enc_dispose (GObject * object)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GstIterator *it;
GValue val = { 0 };
GST_DEBUG_OBJECT (object, "Dispose...");
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (object));
while (gst_iterator_next (it, &val) == GST_ITERATOR_OK) {
gst_srtp_enc_release_pad (GST_ELEMENT_CAST (object),
g_value_get_object (&val));
g_value_unset (&val);
gst_iterator_resync (it);
}
gst_iterator_free (it);
if (filter->key)
gst_buffer_unref (filter->key);
filter->key = NULL;
G_OBJECT_CLASS (gst_srtp_enc_parent_class)->dispose (object);
}
static void
gst_srtp_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_MKEY:
if (filter->key)
gst_buffer_unref (filter->key);
filter->key = g_value_dup_boxed (value);
filter->key_changed = TRUE;
GST_INFO_OBJECT (object, "Set property: key=[%p]", filter->key);
break;
case PROP_RTP_CIPHER:
filter->rtp_cipher = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtp cipher=%d",
filter->rtp_cipher);
break;
case PROP_RTP_AUTH:
filter->rtp_auth = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtp auth=%d", filter->rtp_auth);
break;
case PROP_RTCP_CIPHER:
filter->rtcp_cipher = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtcp cipher=%d",
filter->rtcp_cipher);
break;
case PROP_RTCP_AUTH:
filter->rtcp_auth = g_value_get_enum (value);
GST_INFO_OBJECT (object, "Set property: rtcp auth=%d", filter->rtcp_auth);
break;
case PROP_RANDOM_KEY:
filter->random_key = g_value_get_boolean (value);
break;
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_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstSrtpEnc *filter = GST_SRTP_ENC (object);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
case PROP_MKEY:
if (filter->key)
g_value_set_boxed (value, filter->key);
break;
case PROP_RTP_CIPHER:
g_value_set_enum (value, filter->rtp_cipher);
break;
case PROP_RTCP_CIPHER:
g_value_set_enum (value, filter->rtcp_cipher);
break;
case PROP_RTP_AUTH:
g_value_set_enum (value, filter->rtp_auth);
break;
case PROP_RTCP_AUTH:
g_value_set_enum (value, filter->rtcp_auth);
break;
case PROP_RANDOM_KEY:
g_value_set_boolean (value, filter->random_key);
break;
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);
}
/* Returns the source pad linked with the sink pad
*/
static GstPad *
get_rtp_other_pad (GstPad * pad)
{
return GST_PAD (gst_pad_get_element_private (pad));
}
/* Release a sink pad and it's linked source pad
*/
static void
gst_srtp_enc_release_pad (GstElement * element, GstPad * sinkpad)
{
GstPad *srcpad;
GST_INFO_OBJECT (element, "Releasing pad %s:%s",
GST_DEBUG_PAD_NAME (sinkpad));
srcpad = GST_PAD (gst_pad_get_element_private (sinkpad));
gst_pad_set_element_private (sinkpad, NULL);
gst_pad_set_element_private (srcpad, NULL);
/* deactivate from source to sink */
gst_pad_set_active (srcpad, FALSE);
gst_pad_set_active (sinkpad, FALSE);
/* remove pads */
gst_element_remove_pad (element, srcpad);
gst_element_remove_pad (element, sinkpad);
}
/* Common setcaps function
* Handles the link with other elements
*/
static gboolean
gst_srtp_enc_sink_setcaps (GstPad * pad, GstSrtpEnc * filter,
GstCaps * caps, gboolean is_rtcp)
{
GstPad *otherpad = NULL;
GstStructure *ps = NULL;
gboolean ret = FALSE;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
caps = gst_caps_copy (caps);
ps = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (pad, "Sink caps: %" GST_PTR_FORMAT, caps);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
GST_OBJECT_LOCK (filter);
if (HAS_CRYPTO (filter))
gst_structure_set (ps, "srtp-key", GST_TYPE_BUFFER, filter->key, NULL);
/* Add srtp-specific params to source caps */
gst_structure_set (ps,
"srtp-cipher", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_CIPHER_TYPE, filter->rtp_cipher),
"srtp-auth", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_AUTH_TYPE, filter->rtp_auth),
"srtcp-cipher", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_CIPHER_TYPE, filter->rtcp_cipher),
"srtcp-auth", G_TYPE_STRING,
enum_nick_from_value (GST_TYPE_SRTP_AUTH_TYPE, filter->rtcp_auth), NULL);
GST_OBJECT_UNLOCK (filter);
GST_DEBUG_OBJECT (pad, "Source caps: %" GST_PTR_FORMAT, caps);
/* Set caps on source pad */
otherpad = get_rtp_other_pad (pad);
ret = gst_pad_set_caps (otherpad, caps);
gst_caps_unref (caps);
return ret;
}
static gboolean
gst_srtp_enc_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;
GstPad *otherpad;
GstCaps *other_caps;
GstCaps *ret;
GstCaps *template_caps;
int i;
otherpad = get_rtp_other_pad (pad);
gst_query_parse_caps (query, &filter);
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-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
}
}
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-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);
}
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_enc_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query)
{
return gst_srtp_enc_sink_query (pad, parent, query, FALSE);
}
static gboolean
gst_srtp_enc_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query)
{
return gst_srtp_enc_sink_query (pad, parent, query, TRUE);
}
static GstIterator *
gst_srtp_enc_iterate_internal_links (GstPad * pad, GstObject * parent,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
GstPad *otherpad = NULL;
GstIterator *it = NULL;
otherpad = get_rtp_other_pad (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_enc_iterate_internal_links_rtp (GstPad * pad, GstObject * parent)
{
return gst_srtp_enc_iterate_internal_links (pad, parent, FALSE);
}
static GstIterator *
gst_srtp_enc_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent)
{
return gst_srtp_enc_iterate_internal_links (pad, parent, TRUE);
}
static void
gst_srtp_enc_replace_random_key (GstSrtpEnc * filter)
{
guint i;
guint key_size;
GstMapInfo map;
GST_DEBUG_OBJECT (filter, "Generating random key");
if (filter->key)
gst_buffer_unref (filter->key);
key_size = max_cipher_key_size (filter);
filter->key = gst_buffer_new_allocate (NULL, key_size, NULL);
gst_buffer_map (filter->key, &map, GST_MAP_WRITE);
for (i = 0; i < map.size; i += 4)
GST_WRITE_UINT32_BE (map.data + i, g_random_int ());
gst_buffer_unmap (filter->key, &map);
filter->key_changed = TRUE;
}
static GstFlowReturn
gst_srtp_enc_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
GstFlowReturn ret = GST_FLOW_OK;
GstPad *otherpad = NULL;
err_status_t err = err_status_ok;
gint size_max, size;
GstBuffer *bufout = NULL;
gboolean do_setcaps = FALSE;
GstMapInfo mapin, mapout;
if (!is_rtcp) {
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
if (!gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf)) {
GST_ELEMENT_ERROR (filter, STREAM, WRONG_TYPE, (NULL),
("Could not map RTP buffer"));
ret = GST_FLOW_ERROR;
goto out;
}
gst_rtp_buffer_unmap (&rtpbuf);
} else {
GstRTCPBuffer rtcpbuf = GST_RTCP_BUFFER_INIT;
if (!gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcpbuf)) {
GST_ELEMENT_ERROR (filter, STREAM, WRONG_TYPE, (NULL),
("Could not map RTCP buffer"));
ret = GST_FLOW_ERROR;
goto out;
}
gst_rtcp_buffer_unmap (&rtcpbuf);
}
do_setcaps = filter->key_changed;
if (filter->key_changed)
gst_srtp_enc_reset (filter);
if (filter->first_session) {
err_status_t status = gst_srtp_enc_create_session (filter);
if (status != err_status_ok) {
GST_ELEMENT_ERROR (filter, LIBRARY, INIT,
("Could not initialize SRTP encoder"),
("Failed to add stream to SRTP encoder (err: %d)", status));
ret = GST_FLOW_ERROR;
goto out;
}
}
GST_OBJECT_LOCK (filter);
/* Update source caps if asked */
if (do_setcaps) {
GstCaps *caps;
GST_OBJECT_UNLOCK (filter);
caps = gst_pad_get_current_caps (pad);
if (!gst_srtp_enc_sink_setcaps (pad, filter, caps, is_rtcp)) {
gst_caps_unref (caps);
ret = GST_FLOW_NOT_NEGOTIATED;
goto out;
}
gst_caps_unref (caps);
GST_OBJECT_LOCK (filter);
}
if (!HAS_CRYPTO (filter)) {
GST_OBJECT_UNLOCK (filter);
otherpad = get_rtp_other_pad (pad);
return gst_pad_push (otherpad, buf);
}
/* Create a bigger buffer to add protection */
size_max = gst_buffer_get_size (buf) + SRTP_MAX_TRAILER_LEN + 10;
bufout = gst_buffer_new_allocate (NULL, size_max, NULL);
gst_buffer_map (buf, &mapin, GST_MAP_READ);
gst_buffer_map (bufout, &mapout, GST_MAP_READWRITE);
size = mapin.size;
memcpy (mapout.data, mapin.data, mapin.size);
gst_buffer_unmap (buf, &mapin);
gst_srtp_init_event_reporter ();
if (is_rtcp)
err = srtp_protect_rtcp (filter->session, mapout.data, &size);
else
err = srtp_protect (filter->session, mapout.data, &size);
gst_buffer_unmap (bufout, &mapout);
GST_OBJECT_UNLOCK (filter);
if (err == err_status_ok) {
/* Buffer protected */
gst_buffer_set_size (bufout, size);
gst_buffer_copy_into (bufout, buf, GST_BUFFER_COPY_METADATA, 0, -1);
GST_LOG_OBJECT (pad, "Encing %s buffer of size %d",
is_rtcp ? "RTCP" : "RTP", size);
/* Push buffer to source pad */
otherpad = get_rtp_other_pad (pad);
ret = gst_pad_push (otherpad, bufout);
bufout = NULL;
if (ret != GST_FLOW_OK)
goto out;
} else if (err == err_status_key_expired) {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), STREAM, ENCODE,
("Key usage limit has been reached"),
("Unable to protect buffer (hard key usage limit reached)"));
gst_buffer_unref (bufout);
goto fail;
} else {
/* srtp_protect failed */
GST_ELEMENT_ERROR (filter, LIBRARY, FAILED, (NULL),
("Unable to protect buffer (protect failed) code %d", err));
gst_buffer_unref (bufout);
goto fail;
}
if (gst_srtp_get_soft_limit_reached ()) {
g_signal_emit (filter, gst_srtp_enc_signals[SIGNAL_SOFT_LIMIT], 0);
if (filter->random_key && !filter->key_changed)
gst_srtp_enc_replace_random_key (filter);
}
out:
gst_buffer_unref (buf);
return ret;
fail:
ret = GST_FLOW_ERROR;
goto out;
}
static GstFlowReturn
gst_srtp_enc_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_enc_chain (pad, parent, buf, FALSE);
}
static GstFlowReturn
gst_srtp_enc_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_enc_chain (pad, parent, buf, TRUE);
}
/* Change state
*/
static GstStateChangeReturn
gst_srtp_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstSrtpEnc *filter;
filter = GST_SRTP_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (filter->rtp_cipher != GST_SRTP_CIPHER_NULL ||
filter->rtcp_cipher != GST_SRTP_CIPHER_NULL ||
filter->rtp_auth != GST_SRTP_AUTH_NULL ||
filter->rtcp_auth != GST_SRTP_AUTH_NULL) {
if (!filter->key) {
if (filter->random_key) {
gst_srtp_enc_replace_random_key (filter);
} else {
GST_ERROR_OBJECT (element, "Need a key to get to READY");
return GST_STATE_CHANGE_FAILURE;
}
}
}
if ((filter->rtcp_cipher != NULL_CIPHER)
&& (filter->rtcp_auth == NULL_AUTH)) {
GST_ERROR_OBJECT (filter,
"RTCP authentication can't be NULL if encryption is not NULL.");
return GST_STATE_CHANGE_FAILURE;
}
if (!filter->first_session)
gst_srtp_enc_reset (filter);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
res = GST_ELEMENT_CLASS (gst_srtp_enc_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_enc_reset (filter);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
static gboolean
gst_srtp_enc_sink_event (GstPad * pad, GstObject * parent, GstEvent * event,
gboolean is_rtcp)
{
GstSrtpEnc *filter = GST_SRTP_ENC (parent);
gboolean ret;
GstPad *otherpad;
otherpad = get_rtp_other_pad (pad);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_DEBUG_OBJECT (pad, "Encing event Flush stop (%d)",
GST_EVENT_TYPE (event));
gst_srtp_enc_reset (filter);
ret = gst_pad_push_event (otherpad, event);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_srtp_enc_sink_setcaps (pad, filter, caps, is_rtcp);
break;
}
default:
GST_DEBUG_OBJECT (pad, "Encing event default (%d)",
GST_EVENT_TYPE (event));
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static gboolean
gst_srtp_enc_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
return gst_srtp_enc_sink_event (pad, parent, event, FALSE);
}
static gboolean
gst_srtp_enc_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
return gst_srtp_enc_sink_event (pad, parent, event, TRUE);
}
/* entry point to initialize the plug-in
* initialize the plug-in itself
* register the element factories and other features
*/
gboolean
gst_srtp_enc_plugin_init (GstPlugin * srtpenc)
{
GST_DEBUG_CATEGORY_INIT (gst_srtp_enc_debug, "srtpenc", 0, "SRTP Enc");
return gst_element_register (srtpenc, "srtpenc", GST_RANK_NONE,
GST_TYPE_SRTP_ENC);
}
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