add full SDP codec information to the MNCC socket

This way osmo-msc can benefit from the complete codec information received via
SIP, which was so far terminated at osmo-sip-connector. osmo-sip-connector
could/should have translated the received SDP to MNCC bearer_cap, but this was
never implemented properly. Since osmo-msc already handles SDP towards the MGW,
it makes most sense to pass SDP to osmo-msc transparently.

To be able to send a valid RTP IP:port in the SDP upon the first MNCC_SETUP_IND
going out, move the CN side CRCX to the very start of establishing a voice
call. As a result, first create MGW conns for both RAN and CN before starting.

The voice_call_full.msc chart shows the change in message sequence for MO and
MT voice calls.

Implement cc_sdp.c, which accumulates codec information from various sources
(MS, BSS, Assignment, remote call leg) and provides filtering to get the
available set of codecs at any point in time.

Implement codec_sdp_cc_t9n.c, to translate between SDP and the various
libosmo-mgcp-client, CC and BSSMAP representations of codecs:
- Speech Version,
- Permitted Speech,
- Speech Codec Type,
- default Payload Type numbers,
- enum mgcp_codecs,
- FR/HR compatibility
- SDP audio codec names,
- various AMR configurations.
A codec_map lists these relations in one large data record.
Various functions provide conversions by traversing this map.

Add trans->cc.mnccc_release_sent: so far, avoiding to send an MNCC release
during trans_free() was done by setting the callref = 0. But that also skips CC
Release. On codec mismatch, we send a specific MNCC error code but still want a
normal CC Release: hence send the MNCC message, set mnccc_release_sent = true
and do normal CC Release in trans_free().
(A better way to do this would be to adopt the mncc_call FSM from inter-MSC
handover also for local voice calls, but that is out of scope for now. I want
to try that soon, as time permits.)

Change-Id: I8c3b2de53ffae4ec3a66b9dabf308c290a2c999f

1 files changed, 12 insertions, 12 deletions

diff --git a/tests/sdp_msg/sdp_msg_test.ok b/tests/sdp_msg/sdp_msg_test.ok
index a7a3d5156..6a21ce7d2 100644
--- a/tests/sdp_msg/sdp_msg_test.ok
+++ b/tests/sdp_msg/sdp_msg_test.ok
@@ -566,25 +566,25 @@ sdp_msg_intersection(a,b): a=ptime:20\r\n
 test_select
 
 [0]
-SDP: AMR:octet-align=1,GSM,GSM-HR-08,GSM-EFR
-Select: AMR:octet-align=1
-SDP: AMR:octet-align=1,GSM,GSM-HR-08,GSM-EFR
+SDP: AMR:octet-align=1#112,GSM#3,GSM-HR-08#111,GSM-EFR#110
+Select: AMR:octet-align=1#112
+SDP: AMR:octet-align=1#112,GSM#3,GSM-HR-08#111,GSM-EFR#110
 [0] ok
 
 [1]
-SDP: AMR:octet-align=1,GSM,GSM-HR-08,GSM-EFR
-Select: GSM
-SDP: GSM,AMR:octet-align=1,GSM-HR-08,GSM-EFR
+SDP: AMR:octet-align=1#112,GSM#3,GSM-HR-08#111,GSM-EFR#110
+Select: GSM#3
+SDP: GSM#3,AMR:octet-align=1#112,GSM-HR-08#111,GSM-EFR#110
 [1] ok
 
 [2]
-SDP: AMR:octet-align=1,GSM,GSM-HR-08,GSM-EFR
-Select: GSM-HR-08
-SDP: GSM-HR-08,AMR:octet-align=1,GSM,GSM-EFR
+SDP: AMR:octet-align=1#112,GSM#3,GSM-HR-08#111,GSM-EFR#110
+Select: GSM-HR-08#111
+SDP: GSM-HR-08#111,AMR:octet-align=1#112,GSM#3,GSM-EFR#110
 [2] ok
 
 [3]
-SDP: AMR:octet-align=1,GSM,GSM-HR-08,GSM-EFR
-Select: GSM-EFR
-SDP: GSM-EFR,AMR:octet-align=1,GSM,GSM-HR-08
+SDP: AMR:octet-align=1#112,GSM#3,GSM-HR-08#111,GSM-EFR#110
+Select: GSM-EFR#110
+SDP: GSM-EFR#110,AMR:octet-align=1#112,GSM#3,GSM-HR-08#111
 [3] ok