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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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stream: add SRTP support
Install srtp encoder and decoder elements in rtpbin Add MIKEY in SDP
This commit is contained in:
Wim Taymans
parent
bfbf393925
commit
3d6175c745
5 changed files with 205 additions and 2 deletions
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@ -59,6 +59,7 @@ dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.3.0.1
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GSTPB_REQ=1.3.0.1
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GSTPG_REQ=1.3.0.1
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GSTPD_REQ=1.3.0.1
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dnl *** autotools stuff ****
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@ -176,6 +177,11 @@ GSTPG_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-good-$GST_API_VERSION --variabl
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AC_SUBST(GSTPG_PLUGINS_DIR)
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AC_MSG_NOTICE(Using GStreamer Good Plugins in $GSTPG_PLUGINS_DIR)
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AG_GST_CHECK_GST_PLUGINS_BAD($GST_API_VERSION, [$GSTPD_REQ], [yes])
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GSTPD_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-bad-$GST_API_VERSION --variable pluginsdir`
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AC_SUBST(GSTPD_PLUGINS_DIR)
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AC_MSG_NOTICE(Using GStreamer Bad Plugins in $GSTPD_PLUGINS_DIR)
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AG_GST_CHECK_GST_CHECK($GST_API_VERSION, [$GST_REQ], no)
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AM_CONDITIONAL(HAVE_CHECK, test "x$HAVE_GST_CHECK" = "xyes")
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@ -136,6 +136,7 @@ main (int argc, char *argv[])
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gst_rtsp_media_factory_set_permissions (factory, permissions);
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gst_rtsp_permissions_unref (permissions);
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#endif
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gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_SAVP);
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/* attach the test factory to the /test url */
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gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
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@ -26,6 +26,8 @@
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#include <string.h>
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#include <gst/sdp/gstmikey.h>
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#include "rtsp-sdp.h"
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static gboolean
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@ -171,6 +173,93 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
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gst_sdp_media_add_attribute (smedia, "control", tmp);
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g_free (tmp);
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/* check for srtp */
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do {
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GstBuffer *srtpkey;
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const GValue *val;
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const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth;
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GstMIKEYMessage *msg;
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GstMIKEYPayload *payload;
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GBytes *bytes;
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GstMapInfo info;
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const guint8 *data;
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gsize size;
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gchar *base64;
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guint8 byte;
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guint32 ssrc;
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val = gst_structure_get_value (s, "srtp-key");
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if (val == NULL)
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break;
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srtpkey = gst_value_get_buffer (val);
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if (srtpkey == NULL)
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break;
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srtpcipher = gst_structure_get_string (s, "srtp-cipher");
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srtpauth = gst_structure_get_string (s, "srtp-auth");
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srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
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srtcpauth = gst_structure_get_string (s, "srtcp-auth");
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if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL ||
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srtcpauth == NULL)
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break;
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msg = gst_mikey_message_new ();
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/* unencrypted MIKEY message, we send this over TLS so this is allowed */
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gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
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FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP);
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/* add policy '0' for our SSRC */
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gst_rtsp_stream_get_ssrc (stream, &ssrc);
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gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0);
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/* timestamp is now */
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gst_mikey_message_add_t_now_ntp_utc (msg);
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/* add some random data */
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gst_mikey_message_add_rand_len (msg, 16);
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/* the policy '0' is SRTP with the above discovered algorithms */
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payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
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gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
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/* only AES-CM is supported */
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byte = 1;
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1,
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&byte);
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/* only HMAC-SHA1 */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
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&byte);
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/* we enable encryption on RTP and RTCP */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
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&byte);
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
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&byte);
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/* we enable authentication on RTP and RTCP */
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gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
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&byte);
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gst_mikey_message_add_payload (msg, payload);
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/* add the key in KEMAC */
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gst_buffer_map (srtpkey, &info, GST_MAP_READ);
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gst_mikey_message_add_kemac (msg, GST_MIKEY_ENC_NULL, info.size, info.data,
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GST_MIKEY_MAC_NULL, NULL);
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gst_buffer_unmap (srtpkey, &info);
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/* now serialize this to bytes */
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bytes = gst_mikey_message_to_bytes (msg);
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gst_mikey_message_free (msg);
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/* and make it into base64 */
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data = g_bytes_get_data (bytes, &size);
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base64 = g_base64_encode (data, size);
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g_bytes_unref (bytes);
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tmp = g_strdup_printf ("mikey %s", base64);
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g_free (base64);
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gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
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g_free (tmp);
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} while (FALSE);
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/* collect all other properties and add them to fmtp or attributes */
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fmtp = g_string_new ("");
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g_string_append_printf (fmtp, "%d ", caps_pt);
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@ -198,6 +287,10 @@ make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
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continue;
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if (!strcmp (fname, "seqnum-base"))
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continue;
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if (g_str_has_prefix (fname, "srtp-"))
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continue;
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if (g_str_has_prefix (fname, "srtcp-"))
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continue;
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if (g_str_has_prefix (fname, "a-")) {
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/* attribute */
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@ -79,6 +79,10 @@ struct _GstRTSPStreamPrivate
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/* the RTPSession object */
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GObject *session;
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/* SRTP encoder/decoder */
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GstElement *srtpenc;
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GstElement *srtpdec;
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/* sinks used for sending and receiving RTP and RTCP over ipv4, they share
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* sockets */
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GstElement *udpsrc_v4[2];
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@ -1503,6 +1507,87 @@ static GstAppSinkCallbacks sink_cb = {
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handle_new_sample,
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};
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static GstElement *
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get_rtp_encoder (GstRTSPStream * stream, guint session)
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{
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GstRTSPStreamPrivate *priv = stream->priv;
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if (priv->srtpenc == NULL) {
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gchar *name;
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name = g_strdup_printf ("srtpenc_%u", session);
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priv->srtpenc = gst_element_factory_make ("srtpenc", name);
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g_free (name);
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g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
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}
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return gst_object_ref (priv->srtpenc);
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}
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static GstElement *
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request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
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{
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GstRTSPStreamPrivate *priv = stream->priv;
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GstElement *enc;
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GstPad *pad;
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gchar *name;
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if (priv->idx != session)
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return NULL;
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GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
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enc = get_rtp_encoder (stream, session);
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name = g_strdup_printf ("rtp_sink_%d", session);
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pad = gst_element_get_request_pad (enc, name);
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g_free (name);
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gst_object_unref (pad);
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return enc;
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}
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static GstElement *
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request_rtcp_encoder (GstElement * rtpbin, guint session,
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GstRTSPStream * stream)
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{
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GstRTSPStreamPrivate *priv = stream->priv;
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GstElement *enc;
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GstPad *pad;
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gchar *name;
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if (priv->idx != session)
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return NULL;
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GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
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enc = get_rtp_encoder (stream, session);
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name = g_strdup_printf ("rtcp_sink_%d", session);
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pad = gst_element_get_request_pad (enc, name);
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g_free (name);
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gst_object_unref (pad);
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return enc;
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}
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static GstElement *
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request_rtcp_decoder (GstElement * rtpbin, guint session,
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GstRTSPStream * stream)
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{
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GstRTSPStreamPrivate *priv = stream->priv;
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if (priv->idx != session)
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return NULL;
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if (priv->srtpdec == NULL) {
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gchar *name;
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name = g_strdup_printf ("srtpdec_%u", session);
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priv->srtpdec = gst_element_factory_make ("srtpdec", name);
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g_free (name);
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}
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return gst_object_ref (priv->srtpdec);
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}
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/**
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* gst_rtsp_stream_join_bin:
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* @stream: a #GstRTSPStream
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@ -1549,6 +1634,21 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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/* update the dscp qos field in the sinks */
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update_dscp_qos (stream);
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if (priv->profiles & GST_RTSP_PROFILE_SAVP
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|| priv->profiles & GST_RTSP_PROFILE_SAVPF) {
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/* For SRTP */
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g_signal_connect (rtpbin, "request-rtp-encoder",
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(GCallback) request_rtp_encoder, stream);
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g_signal_connect (rtpbin, "request-rtcp-encoder",
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(GCallback) request_rtcp_encoder, stream);
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#if 0
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g_signal_connect (rtpbin, "request-rtp-decoder",
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(GCallback) request_rtp_decoder, stream);
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#endif
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g_signal_connect (rtpbin, "request-rtcp-decoder",
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(GCallback) request_rtcp_decoder, stream);
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}
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/* get a pad for sending RTP */
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name = g_strdup_printf ("send_rtp_sink_%u", idx);
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priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
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@ -1873,6 +1973,9 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
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gst_caps_unref (priv->caps);
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priv->caps = NULL;
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if (priv->srtpenc)
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gst_object_unref (priv->srtpenc);
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priv->is_joined = FALSE;
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g_mutex_unlock (&priv->lock);
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@ -11,8 +11,8 @@ TESTS_ENVIRONMENT = \
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GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)" \
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$(REGISTRY_ENVIRONMENT) \
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GST_PLUGIN_SYSTEM_PATH_1_0= \
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GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR) \
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GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good"
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GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR) \
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GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
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# ths core dumps of some machines have PIDs appended
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