Move RTP socket handling out of signal handlers into abis_rsl
This is not really nice, but we will soon have multiple users of
the CRCX / MDCX / DLCX signals, and we cannot guarantee the ordering
of them. So as a workaround, we move the RTP socket creation and
deletion into the core abis_rsl codebase.
diff --git a/openbsc/src/abis_rsl.c b/openbsc/src/abis_rsl.c
index 2d55806..05735ee 100644
--- a/openbsc/src/abis_rsl.c
+++ b/openbsc/src/abis_rsl.c
@@ -39,6 +39,7 @@
#include <openbsc/paging.h>
#include <openbsc/signal.h>
#include <openbsc/meas_rep.h>
+#include <openbsc/rtp_proxy.h>
#define RSL_ALLOC_SIZE 1024
#define RSL_ALLOC_HEADROOM 128
@@ -1649,10 +1650,35 @@
LOGP(DRSL, LOGL_NOTICE, "mandatory IE missing");
return -EINVAL;
}
+
ipac_parse_rtp(lchan, &tv);
+
+ /* in case we don't use direct BTS-to-BTS RTP */
+ if (!ipacc_rtp_direct) {
+ int rc;
+ /* the BTS has successfully bound a TCH to a local ip/port,
+ * which means we can connect our UDP socket to it */
+ if (lchan->abis_ip.rtp_socket) {
+ rtp_socket_free(lchan->abis_ip.rtp_socket);
+ lchan->abis_ip.rtp_socket = NULL;
+ }
+
+ lchan->abis_ip.rtp_socket = rtp_socket_create();
+ if (!lchan->abis_ip.rtp_socket)
+ goto out_err;
+
+ rc = rtp_socket_connect(lchan->abis_ip.rtp_socket,
+ lchan->abis_ip.bound_ip,
+ lchan->abis_ip.bound_port);
+ if (rc < 0)
+ goto out_err;
+ }
+
dispatch_signal(SS_ABISIP, S_ABISIP_CRCX_ACK, msg->lchan);
return 0;
+out_err:
+ return -EIO;
}
static int abis_rsl_rx_ipacc_mdcx_ack(struct msgb *msg)
@@ -1676,6 +1702,7 @@
{
struct abis_rsl_dchan_hdr *dh = msgb_l2(msg);
struct tlv_parsed tv;
+ struct gsm_lchan *lchan = msg->lchan;
rsl_tlv_parse(&tv, dh->data, msgb_l2len(msg)-sizeof(*dh));
@@ -1683,6 +1710,12 @@
print_rsl_cause(TLVP_VAL(&tv, RSL_IE_CAUSE),
TLVP_LEN(&tv, RSL_IE_CAUSE));
+ /* the BTS tells us a RTP stream has been disconnected */
+ if (lchan->abis_ip.rtp_socket) {
+ rtp_socket_free(lchan->abis_ip.rtp_socket);
+ lchan->abis_ip.rtp_socket = NULL;
+ }
+
dispatch_signal(SS_ABISIP, S_ABISIP_DLCX_IND, msg->lchan);
return 0;