aboutsummaryrefslogtreecommitdiffstats
path: root/src/osmo-bsc/osmo_bsc_audio.c
diff options
context:
space:
mode:
Diffstat (limited to 'src/osmo-bsc/osmo_bsc_audio.c')
-rw-r--r--src/osmo-bsc/osmo_bsc_audio.c70
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 11602090..b4ffa88f 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;