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-rw-r--r--src/osmo-bsc/osmo_bsc_audio.c70
1 files changed, 70 insertions, 0 deletions
diff --git a/src/osmo-bsc/osmo_bsc_audio.c b/src/osmo-bsc/osmo_bsc_audio.c
new file mode 100644
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--- /dev/null
+++ b/src/osmo-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 Affero General Public License as published by
+ * the Free Software Foundation; either version 3 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 Affero General Public License for more details.
+ *
+ * You should have received a copy of the GNU Affero General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ *
+ */
+
+#include <openbsc/osmo_msc_data.h>
+#include <openbsc/osmo_bsc.h>
+#include <openbsc/abis_rsl.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;
+}