handover: Fix the handover signalling for IP based BTSes
This was reported by Kevin when he was testing handover. The problem
is the order of the signal handlers for S_ABISIP_CRCX_ACK. Right now
the handover signal handler is called before the one inside the libmsc
gsm_04_08.c. This means S_HANDOVER_ACK is signalled _before_ there is a
rtp socket created for the channel. The result is that the MDCX will
never be sent and the called will not be properly switched _after_ the
handover detection.
I do not want to play with the order of signal handlers, remove the
CRCX ack handling from the handover_logic.c and force the NITB (and
later the BSC) to check if the lchan is involved with a handover and
do the switching in there. This means right now we do what two signal
handlers did in one.
Reproduced and tested with the FakeBTS Handover test.
Log message:
<0004> abis_rsl.c:1954 (bts=1,trx=0,ts=3,ss=0) IPAC_CRCX_ACK ...
<000c> gsm_04_08.c:1400 no RTP socket for new_lchan
<001a> rtp_proxy.c:533 rtp_socket_create(): success
<001a> rtp_proxy.c:615 rtp_socket_bind(rs=0x48703c8, IP=0.0.0.0): ...
diff --git a/openbsc/include/openbsc/handover.h b/openbsc/include/openbsc/handover.h
index 9d9a90b..bd0d8ad 100644
--- a/openbsc/include/openbsc/handover.h
+++ b/openbsc/include/openbsc/handover.h
@@ -11,4 +11,7 @@
/* clear any operation for this connection */
void bsc_clear_handover(struct gsm_subscriber_connection *conn, int free_lchan);
+/* Return the old lchan or NULL. This is meant for audio handling */
+struct gsm_lchan *bsc_handover_pending(struct gsm_lchan *new_lchan);
+
#endif /* _HANDOVER_H */
diff --git a/openbsc/include/openbsc/signal.h b/openbsc/include/openbsc/signal.h
index 598d470..fd13ac7 100644
--- a/openbsc/include/openbsc/signal.h
+++ b/openbsc/include/openbsc/signal.h
@@ -226,16 +226,6 @@
struct osmo_msc_data *data;
};
-/* handover */
-enum signal_ho {
- S_HANDOVER_ACK,
-};
-
-struct ho_signal_data {
- struct gsm_lchan *old_lchan;
- struct gsm_lchan *new_lchan;
-};
-
/* SS_CCCH signals */
enum signal_ccch {
S_CCCH_PAGING_LOAD,
diff --git a/openbsc/src/libbsc/handover_logic.c b/openbsc/src/libbsc/handover_logic.c
index d2cd5ef..9cf26af 100644
--- a/openbsc/src/libbsc/handover_logic.c
+++ b/openbsc/src/libbsc/handover_logic.c
@@ -329,23 +329,6 @@
return 0;
}
-static int ho_ipac_crcx_ack(struct gsm_lchan *new_lchan)
-{
- struct bsc_handover *ho;
- struct ho_signal_data sig_ho;
-
- ho = bsc_ho_by_new_lchan(new_lchan);
- if (!ho) {
- /* it is perfectly normal, we have CRCX even in non-HO cases */
- return 0;
- }
-
- sig_ho.old_lchan = ho->old_lchan;
- sig_ho.new_lchan = new_lchan;
- osmo_signal_dispatch(SS_HO, S_HANDOVER_ACK, &sig_ho);
- return 0;
-}
-
static int ho_logic_sig_cb(unsigned int subsys, unsigned int signal,
void *handler_data, void *signal_data)
{
@@ -369,14 +352,6 @@
return ho_gsm48_ho_fail(lchan);
}
break;
- case SS_ABISIP:
- lchan = signal_data;
- switch (signal) {
- case S_ABISIP_CRCX_ACK:
- return ho_ipac_crcx_ack(lchan);
- break;
- }
- break;
default:
break;
}
@@ -384,8 +359,16 @@
return 0;
}
+struct gsm_lchan *bsc_handover_pending(struct gsm_lchan *new_lchan)
+{
+ struct bsc_handover *ho;
+ ho = bsc_ho_by_new_lchan(new_lchan);
+ if (!ho)
+ return NULL;
+ return ho->old_lchan;
+}
+
static __attribute__((constructor)) void on_dso_load_ho_logic(void)
{
osmo_signal_register_handler(SS_LCHAN, ho_logic_sig_cb, NULL);
- osmo_signal_register_handler(SS_ABISIP, ho_logic_sig_cb, NULL);
}
diff --git a/openbsc/src/libmsc/gsm_04_08.c b/openbsc/src/libmsc/gsm_04_08.c
index d1dfa4a..65e6fcc 100644
--- a/openbsc/src/libmsc/gsm_04_08.c
+++ b/openbsc/src/libmsc/gsm_04_08.c
@@ -2,7 +2,7 @@
* 3GPP TS 04.08 version 7.21.0 Release 1998 / ETSI TS 100 940 V7.21.0 */
/* (C) 2008-2009 by Harald Welte <laforge@gnumonks.org>
- * (C) 2008-2010 by Holger Hans Peter Freyther <zecke@selfish.org>
+ * (C) 2008-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
*
* All Rights Reserved
*
@@ -51,6 +51,7 @@
#include <openbsc/silent_call.h>
#include <openbsc/bsc_api.h>
#include <openbsc/osmo_msc.h>
+#include <openbsc/handover.h>
#include <osmocom/abis/e1_input.h>
#include <osmocom/core/bitvec.h>
@@ -1378,14 +1379,10 @@
static int tch_recv_mncc(struct gsm_network *net, uint32_t callref, int enable);
/* handle audio path for handover */
-static int handle_ho_signal(unsigned int subsys, unsigned int signal,
- void *handler_data, void *signal_data)
+static int switch_for_handover(struct gsm_lchan *old_lchan,
+ struct gsm_lchan *new_lchan)
{
struct rtp_socket *old_rs, *new_rs, *other_rs;
- struct ho_signal_data *sig = signal_data;
-
- if (subsys != SS_HO || signal != S_HANDOVER_ACK)
- return 0;
if (ipacc_rtp_direct) {
LOGP(DHO, LOGL_ERROR, "unable to handover in direct RTP mode\n");
@@ -1393,15 +1390,15 @@
}
/* RTP Proxy mode */
- new_rs = sig->new_lchan->abis_ip.rtp_socket;
- old_rs = sig->old_lchan->abis_ip.rtp_socket;
+ new_rs = new_lchan->abis_ip.rtp_socket;
+ old_rs = old_lchan->abis_ip.rtp_socket;
if (!new_rs) {
LOGP(DHO, LOGL_ERROR, "no RTP socket for new_lchan\n");
return -EIO;
}
- rsl_ipacc_mdcx_to_rtpsock(sig->new_lchan);
+ rsl_ipacc_mdcx_to_rtpsock(new_lchan);
if (!old_rs) {
LOGP(DHO, LOGL_ERROR, "no RTP socket for old_lchan\n");
@@ -1413,7 +1410,7 @@
new_rs->tx_action = old_rs->tx_action;
new_rs->transmit = old_rs->transmit;
- switch (sig->old_lchan->abis_ip.rtp_socket->rx_action) {
+ switch (old_lchan->abis_ip.rtp_socket->rx_action) {
case RTP_PROXY:
other_rs = old_rs->proxy.other_sock;
rtp_socket_proxy(new_rs, other_rs);
@@ -1435,7 +1432,7 @@
static int handle_abisip_signal(unsigned int subsys, unsigned int signal,
void *handler_data, void *signal_data)
{
- struct gsm_lchan *lchan = signal_data;
+ struct gsm_lchan *lchan = signal_data, *old_lchan;
int rc;
struct gsm_network *net;
struct gsm_trans *trans;
@@ -1476,6 +1473,17 @@
tch_recv_mncc(net, trans->callref, 1);
}
}
+
+ /*
+ * TODO: this appears to be too early? Why not until after
+ * the handover detect or the handover complete?
+ *
+ * Do we have a handover pending for this new lchan? In that
+ * case re-route the audio from the old channel to the new one.
+ */
+ old_lchan = bsc_handover_pending(lchan);
+ if (old_lchan)
+ switch_for_handover(old_lchan, lchan);
break;
case S_ABISIP_DLCX_IND:
/* the BTS tells us a RTP stream has been disconnected */
@@ -3253,6 +3261,5 @@
*/
static __attribute__((constructor)) void on_dso_load_0408(void)
{
- osmo_signal_register_handler(SS_HO, handle_ho_signal, NULL);
osmo_signal_register_handler(SS_ABISIP, handle_abisip_signal, NULL);
}
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_audio.c b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
index ed0ece7..660d884 100644
--- a/openbsc/src/osmo-bsc/osmo_bsc_audio.c
+++ b/openbsc/src/osmo-bsc/osmo_bsc_audio.c
@@ -45,6 +45,10 @@
switch (signal) {
case S_ABISIP_CRCX_ACK:
+ /*
+ * 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);