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-rw-r--r--src/osmo-bsc/osmo_bsc_audio.c98
1 files changed, 0 insertions, 98 deletions
diff --git a/src/osmo-bsc/osmo_bsc_audio.c b/src/osmo-bsc/osmo_bsc_audio.c
deleted file mode 100644
index 8eef8d2..0000000
--- a/src/osmo-bsc/osmo_bsc_audio.c
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * 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 <osmocom/bsc/bsc_msc_data.h>
-#include <osmocom/bsc/osmo_bsc.h>
-#include <osmocom/bsc/abis_rsl.h>
-#include <osmocom/bsc/gsm_data.h>
-#include <osmocom/bsc/debug.h>
-#include <osmocom/bsc/signal.h>
-#include <osmocom/gsm/gsm0808.h>
-#include <osmocom/gsm/gsm0808_utils.h>
-#include <osmocom/bsc/osmo_bsc_sigtran.h>
-#include <osmocom/bsc/bsc_subscr_conn_fsm.h>
-#include <osmocom/bsc/bsc_subscriber.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;
- uint32_t rtp_ip;
-
- if (subsys != SS_ABISIP)
- return 0;
-
- con = lchan->conn;
- if (!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->user_plane.rtp_port, lchan->abis_ip.conn_id);
-
- /* If AoIP is in use, the rtp_ip, which has been communicated
- * via the A interface as connect_ip */
- if(con->user_plane.rtp_ip)
- rtp_ip = con->user_plane.rtp_ip;
- else
- rtp_ip = ntohl(INADDR_ANY);
-
- rc = rsl_ipacc_mdcx(lchan, rtp_ip,
- con->user_plane.rtp_port,
- lchan->abis_ip.rtp_payload2);
- if (rc < 0) {
- LOGP(DMSC, LOGL_ERROR, "Failed to send MDCX: %d\n", rc);
- return rc;
- }
- break;
-
- case S_ABISIP_MDCX_ACK:
- if (con->ho) {
- LOGPHO(con->ho, LOGL_DEBUG, "BTS sent MDCX ACK\n");
- /* No need to do anything for handover here. As soon as a HANDOVER DETECT
- * happens, handover_logic.c and bsc_subscr_conn_fsm.c will trigger the
- * MGCP MDCX towards MGW by receiving an S_LCHAN_HANDOVER_DETECT signal. */
- } else if (is_ipaccess_bts(conn_get_bts(con)) && con->user_plane.rtp_ip) {
- /* NOTE: This is only relevant on AoIP networks with
- * IPA based base stations. See also osmo_bsc_api.c,
- * function bsc_assign_compl() */
- LOGP(DMSC, LOGL_INFO, "Tx MSC ASSIGN COMPL (POSTPONED)\n");
- osmo_fsm_inst_dispatch(con->fi, GSCON_EV_RR_ASS_COMPL, NULL);
-
- }
- break;
- }
-
- return 0;
-}
-
-int osmo_bsc_audio_init(struct gsm_network *net)
-{
- osmo_signal_register_handler(SS_ABISIP, handle_abisip_signal, net);
- return 0;
-}