1 files changed, 69 insertions, 1 deletions

diff --git a/src/osmo-bsc/osmo_bsc_audio.c b/src/osmo-bsc/osmo_bsc_audio.c
index 116020900..b4ffa88f1 100644
--- a/src/osmo-bsc/osmo_bsc_audio.c
+++ b/src/osmo-bsc/osmo_bsc_audio.c
@@ -26,15 +26,57 @@
 #include <openbsc/gsm_data.h>
 #include <openbsc/debug.h>
 #include <openbsc/signal.h>
+#include <osmocom/gsm/gsm0808.h>
+#include <osmocom/gsm/gsm0808_utils.h>
+#include <openbsc/osmo_bsc_sigtran.h>
 
 #include <arpa/inet.h>
 
+/* Generate and send assignment complete message */
+static int send_aoip_ass_compl(struct gsm_subscriber_connection *conn, struct gsm_lchan *lchan)
+{
+	struct msgb *resp;
+	struct sockaddr_storage rtp_addr;
+	struct sockaddr_in rtp_addr_in;
+	struct gsm0808_speech_codec sc;
+
+	OSMO_ASSERT(lchan->abis_ip.ass_compl.valid == true);
+
+	/* Package RTP-Address data */
+	memset(&rtp_addr_in, 0, sizeof(rtp_addr_in));
+	rtp_addr_in.sin_family = AF_INET;
+	rtp_addr_in.sin_port = htons(lchan->abis_ip.bound_port);
+	rtp_addr_in.sin_addr.s_addr = htonl(lchan->abis_ip.bound_ip);
+	memset(&rtp_addr, 0, sizeof(rtp_addr));
+	memcpy(&rtp_addr, &rtp_addr_in, sizeof(rtp_addr_in));
+
+	/* Extrapolate speech codec from speech mode */
+	gsm0808_speech_codec_from_chan_type(&sc, lchan->abis_ip.ass_compl.speech_mode);
+
+	/* Generate message */
+	resp = gsm0808_create_ass_compl(lchan->abis_ip.ass_compl.rr_cause,
+					lchan->abis_ip.ass_compl.chosen_channel,
+					lchan->abis_ip.ass_compl.encr_alg_id,
+					lchan->abis_ip.ass_compl.speech_mode,
+					&rtp_addr,
+					&sc,
+					NULL);
+
+	if (!resp) {
+		LOGP(DMSC, LOGL_ERROR, "Failed to generate assignment completed message!\n"); \
+		return -EINVAL;
+	}
+
+	return osmo_bsc_sigtran_send(conn->sccp_con, resp);
+}
+
 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;
@@ -49,11 +91,19 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
 		 * TODO: handle handover here... then the audio should go to
 		 * the old mgcp port..
 		 */
+
 		/* 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),
+		/* If AoIP is in use, the rtp_ip, which has been communicated
+		 * via the A interface as connect_ip */
+		if(con->sccp_con->rtp_ip)
+			rtp_ip = con->sccp_con->rtp_ip;
+		else
+			rtp_ip = ntohl(INADDR_ANY);
+
+		rc = rsl_ipacc_mdcx(lchan, rtp_ip,
 				    con->sccp_con->rtp_port,
 				    lchan->abis_ip.rtp_payload2);
 		if (rc < 0) {
@@ -61,6 +111,24 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
 			return rc;
 		}
 		break;
+
+	case S_ABISIP_MDCX_ACK:
+		if (con->ho_lchan) {
+			/* NOTE: When an ho_lchan exists, the MDCX is part of an
+			 * handover operation (intra-bsc). This means we will not
+			 * inform the MSC about the event, which means that no
+			 * assignment complete message is transmitted */
+			LOGP(DMSC, LOGL_INFO," RTP connection handover complete\n");
+		} else if (is_ipaccess_bts(con->bts) && con->sccp_con->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");
+			if (send_aoip_ass_compl(con, lchan) != 0)
+				return -EINVAL;
+		}
+		break;
+	break;
 	}
 
 	return 0;