mncc: Implement the direct RTP mode for ip based systems

For the LCR rtp-bridge audio should directly flow to the
remote system. In contrast to the original patch audio
will now flow directly from the BTS to the remote system.
This assumes that BTS and the remote system are in the
same network segment and can directly communicate.

There are various limitations in the first iteration of
the implementation:

We could (and in the future) should delay the assignment
but currently we are forced to pick the channel and move
it to the audio state. In case we are located on a SDCCH
we always need to change but if we are on a TCH we could
send the ipa.CRCX and change the audio state a lot later.
The net effect is that the audio codec selection needs to
be done in the NITB code and not in the system connected
to it.

This only works with ip based systems. For E1 systems one
could still use the RTP socket or even try to move this
out of the process.

There is no code for handover handling and it relies on
the remote system dealing with the SSRC change of the
system.

1 files changed, 4 insertions, 0 deletions

diff --git a/openbsc/include/openbsc/gsm_data.h b/openbsc/include/openbsc/gsm_data.h
index 90f3c800d..bed04e380 100644
--- a/openbsc/include/openbsc/gsm_data.h
+++ b/openbsc/include/openbsc/gsm_data.h
@@ -120,6 +120,10 @@ struct gsm_subscriber_connection {
 	/* Are we part of a special "silent" call */
 	int silent_call;
 
+	/* MNCC rtp bridge markers */
+	int mncc_rtp_bridge;
+	int mncc_rtp_create_pending;
+
 	/* bsc structures */
 	struct osmo_bsc_sccp_con *sccp_con;