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, 9 insertions, 8 deletions

diff --git a/src/libmsc/call_leg.c b/src/libmsc/call_leg.c
index b1d0b1e48..725b8bbdb 100644
--- a/src/libmsc/call_leg.c
+++ b/src/libmsc/call_leg.c
@@ -316,7 +316,8 @@ struct osmo_sockaddr_str *call_leg_local_ip(struct call_leg *cl, enum rtp_direct
  * MDCX.
  */
 int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t call_id, struct gsm_trans *for_trans,
-		       const enum mgcp_codecs *codec_if_known, const struct osmo_sockaddr_str *remote_addr_if_known)
+		       const struct sdp_audio_codecs *codecs_if_known,
+		       const struct osmo_sockaddr_str *remote_addr_if_known)
 {
 	if (call_leg_ensure_rtp_alloc(cl, dir, call_id, for_trans))
 		return -EIO;
@@ -325,8 +326,8 @@ int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t cal
 		cl->rtp[dir]->use_osmux = true;
 		cl->rtp[dir]->remote_osmux_cid = -1; /* wildcard */
 	}
-	if (codec_if_known)
-		rtp_stream_set_codec(cl->rtp[dir], *codec_if_known);
+	if (codecs_if_known)
+		rtp_stream_set_codecs(cl->rtp[dir], codecs_if_known);
 	if (remote_addr_if_known && osmo_sockaddr_str_is_nonzero(remote_addr_if_known))
 		rtp_stream_set_remote_addr(cl->rtp[dir], remote_addr_if_known);
 	return rtp_stream_ensure_ci(cl->rtp[dir], cl->mgw_endpoint);
@@ -335,22 +336,22 @@ int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t cal
 int call_leg_local_bridge(struct call_leg *cl1, uint32_t call_id1, struct gsm_trans *trans1,
 			  struct call_leg *cl2, uint32_t call_id2, struct gsm_trans *trans2)
 {
-	enum mgcp_codecs codec;
+	struct sdp_audio_codecs *codecs;
 
 	cl1->local_bridge = cl2;
 	cl2->local_bridge = cl1;
 
 	/* We may just copy the codec info we have for the RAN side of the first leg to the CN side of both legs. This
 	 * also means that if both legs use different codecs the MGW must perform transcoding on the second leg. */
-	if (!cl1->rtp[RTP_TO_RAN] || !cl1->rtp[RTP_TO_RAN]->codec_known) {
+	if (!cl1->rtp[RTP_TO_RAN] || !cl1->rtp[RTP_TO_RAN]->codecs_known) {
 		LOG_CALL_LEG(cl1, LOGL_ERROR, "RAN-side RTP stream codec is not known, not ready for bridging\n");
 		return -EINVAL;
 	}
-	codec = cl1->rtp[RTP_TO_RAN]->codec;
+	codecs = &cl1->rtp[RTP_TO_RAN]->codecs;
 
 	call_leg_ensure_ci(cl1, RTP_TO_CN, call_id1, trans1,
-			   &codec, &cl2->rtp[RTP_TO_CN]->local);
+			   codecs, &cl2->rtp[RTP_TO_CN]->local);
 	call_leg_ensure_ci(cl2, RTP_TO_CN, call_id2, trans2,
-			   &codec, &cl1->rtp[RTP_TO_CN]->local);
+			   codecs, &cl1->rtp[RTP_TO_CN]->local);
 	return 0;
 }