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/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);