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rtsp-client: Send KeyMgmt header in ANNOUNCE response

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When sending back an encrypted RTCP back channel, it is useful
for the client to know the encryption key.

https://bugzilla.gnome.org/show_bug.cgi?id=794813
This commit is contained in:
Mathieu Duponchelle 2018-03-29 16:12:26 +02:00
parent a093f4442b
commit ae0e08dac2
1 changed files with 84 additions and 3 deletions
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87
gst/rtsp-server/rtsp-client.c
View file

@ -2062,6 +2062,73 @@ rtsp_ctrl_timeout_remove (GstRTSPClientPrivate * priv)
g_mutex_unlock (&priv->lock);
}
static gchar *
stream_make_keymgmt (GstRTSPClient * client, const gchar *location, GstRTSPStream * stream)
{
gchar *base64, *result = NULL;
GstMIKEYMessage *mikey_msg;
GstCaps *srtcpparams;
GstElement *rtcp_encoder;
gint srtcp_cipher, srtp_cipher;
gint srtcp_auth, srtp_auth;
GstBuffer *key;
GType ciphertype, authtype;
GEnumClass *cipher_enum, *auth_enum;
GEnumValue *srtcp_cipher_value, *srtp_cipher_value, *srtcp_auth_value, *srtp_auth_value;
rtcp_encoder = gst_rtsp_stream_get_srtp_encoder (stream);
if (!rtcp_encoder)
goto done;
ciphertype = g_type_from_name ("GstSrtpCipherType");
authtype = g_type_from_name ("GstSrtpAuthType");
cipher_enum = g_type_class_ref (ciphertype);
auth_enum = g_type_class_ref (authtype);
/* We need to bring the encoder to READY so that it generates its key */
gst_element_set_state (rtcp_encoder, GST_STATE_READY);
g_object_get (rtcp_encoder, "rtcp-cipher", &srtcp_cipher, "rtcp-auth", &srtcp_auth,
"rtp-cipher", &srtp_cipher, "rtp-auth", &srtp_auth, "key", &key, NULL);
g_object_unref (rtcp_encoder);
srtcp_cipher_value = g_enum_get_value (cipher_enum, srtcp_cipher);
srtp_cipher_value = g_enum_get_value (cipher_enum, srtp_cipher);
srtcp_auth_value = g_enum_get_value (auth_enum, srtcp_auth);
srtp_auth_value = g_enum_get_value (auth_enum, srtp_auth);
g_type_class_unref (cipher_enum);
g_type_class_unref (auth_enum);
srtcpparams = gst_caps_new_simple ("application/x-srtcp",
"srtcp-cipher", G_TYPE_STRING, srtcp_cipher_value->value_nick,
"srtcp-auth", G_TYPE_STRING, srtcp_auth_value->value_nick,
"srtp-cipher", G_TYPE_STRING, srtp_cipher_value->value_nick,
"srtp-auth", G_TYPE_STRING, srtp_auth_value->value_nick,
"srtp-key", GST_TYPE_BUFFER, key, NULL);
mikey_msg = gst_mikey_message_new_from_caps (srtcpparams);
if (mikey_msg) {
guint send_ssrc;
gst_rtsp_stream_get_ssrc (stream, &send_ssrc);
gst_mikey_message_add_cs_srtp (mikey_msg, 0, send_ssrc, 0);
base64 = gst_mikey_message_base64_encode (mikey_msg);
gst_mikey_message_unref (mikey_msg);
if (base64) {
result = gst_sdp_make_keymgmt (location, base64);
g_free (base64);
}
}
done:
return result;
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
@ -2718,6 +2785,7 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
guint8 *data;
guint size;
GstRTSPStatusCode sig_result;
guint i, n_streams;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
@ -2771,13 +2839,26 @@ handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
if (!klass->handle_sdp (client, ctx, media, sdp))
goto unhandled_sdp;
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
gchar *location = g_strdup_printf ("rtsp://%s%s:8554/stream=%d", priv->server_ip, path, i);
gchar *keymgmt = stream_make_keymgmt (client, location, stream);
if (keymgmt)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_KEYMGMT, keymgmt);
g_free (location);
g_free (keymgmt);
}
/* we suspend after the announce */
gst_rtsp_media_suspend (media);
g_object_unref (media);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],