diff options
| author | Holger Hans Peter Freyther <zecke@selfish.org> | 2012-12-26 10:17:42 +0100 |
|---|---|---|
| committer | Holger Hans Peter Freyther <zecke@selfish.org> | 2012-12-26 10:32:02 +0100 |
| commit | c121bb3188445dfc23a6daef3444031f447395bb (patch) | |
| tree | ffc105ad751a96d11816414bba8d0f60dea7acb8 /openbsc/src/osmo-bsc/osmo_bsc_audio.c | |
| parent | 006e3d87e019b202a38c5393ab8f5b6df763e664 (diff) | |
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): ...
Diffstat (limited to 'openbsc/src/osmo-bsc/osmo_bsc_audio.c')
| -rw-r--r-- | openbsc/src/osmo-bsc/osmo_bsc_audio.c | 4 |
1 files changed, 4 insertions, 0 deletions
diff --git a/openbsc/src/osmo-bsc/osmo_bsc_audio.c b/openbsc/src/osmo-bsc/osmo_bsc_audio.c index ed0ece761..660d88497 100644 --- a/openbsc/src/osmo-bsc/osmo_bsc_audio.c +++ b/openbsc/src/osmo-bsc/osmo_bsc_audio.c @@ -45,6 +45,10 @@ static int handle_abisip_signal(unsigned int subsys, unsigned int signal, 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); |