diff --git a/openbsc/src/bsc/osmo_bsc_audio.c b/openbsc/src/bsc/osmo_bsc_audio.c
new file mode 100644
index 0000000..1349263
--- /dev/null
+++ b/openbsc/src/bsc/osmo_bsc_audio.c
@@ -0,0 +1,70 @@
+/*
+ * ipaccess audio handling
+ *
+ * (C) 2009-2010 by Holger Hans Peter Freyther <zecke@selfish.org>
+ * (C) 2009-2010 by On-Waves
+ * All Rights Reserved
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ *
+ */
+
+#include <openbsc/osmo_msc_data.h>
+#include <openbsc/osmo_bsc.h>
+#include <openbsc/gsm_data.h>
+#include <openbsc/debug.h>
+#include <openbsc/signal.h>
+
+#include <arpa/inet.h>
+
+static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
+				 void *handler_data, void *signal_data)
+{
+	struct gsm_subscriber_connection *con;
+	struct gsm_lchan *lchan = signal_data;
+	int rc;
+
+	if (subsys != SS_ABISIP)
+		return 0;
+
+	con = lchan->conn;
+	if (!con || !con->sccp_con)
+		return 0;
+
+	switch (signal) {
+	case S_ABISIP_CRCX_ACK:
+		/* we can ask it to connect now */
+		LOGP(DMSC, LOGL_DEBUG, "Connecting BTS to port: %d conn: %d\n",
+		     con->sccp_con->rtp_port, lchan->abis_ip.conn_id);
+
+		rc = rsl_ipacc_mdcx(lchan, ntohl(INADDR_ANY),
+				    con->sccp_con->rtp_port,
+				    lchan->abis_ip.rtp_payload2);
+		if (rc < 0) {
+			LOGP(DMSC, LOGL_ERROR, "Failed to send MDCX: %d\n", rc);
+			return rc;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+int osmo_bsc_audio_init(struct gsm_network *net)
+{
+	net->hardcoded_rtp_payload = 98;
+	register_signal_handler(SS_ABISIP, handle_abisip_signal, net);
+	return 0;
+}