implement re-assignment to match codecsrhizomatica/testing

This is the last missing piece that allows osmo-msc to make good TFO
codecs choices.

Since the codec_filter, osmo-msc properly gathers codec options and
limitations. But the MO call leg still assigns a voice channel before
getting a response from the MT call leg, and is then stuck with that.

Add the capability to adjust the MO call leg's codec in case the MT side
needs a different codec for TFO.

This is only relevant for 2G; on 3G we always have AMR/IuUP.

For inter-MSC handover, keep the behavior unchanged: offer only the
currently assigned codec to the remote side. Codec-changing HO should be
equally trivial to implement, but that is for another day.

msc_vlr_test_call's codec tests are adjusted to test the new feature in
Ib933554f826c1b4347dfa3f6c4f6fe086be8b133. For now, avoid change in
these tests by validating the first codec in SDP lists only.

Related: OS#6258
Related: osmo-ttcn3-hacks I402ed0523a2a87b83f29c5577b2c828102005d53
Change-Id: I8760feaa8598047369ef8c3ab2673013bac8ac8a

7 files changed, 82 insertions, 61 deletions

diff --git a/include/osmocom/msc/msc_a.h b/include/osmocom/msc/msc_a.h
index 0276d62e2..5e1672aca 100644
--- a/include/osmocom/msc/msc_a.h
+++ b/include/osmocom/msc/msc_a.h
@@ -216,6 +216,7 @@ bool msc_a_is_establishing_auth_ciph(const struct msc_a *msc_a);
 
 int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans);
 int msc_a_try_call_assignment(struct gsm_trans *cc_trans);
+void msc_a_tx_assignment(struct msc_a *msc_a);
 
 const char *msc_a_cm_service_type_to_use(struct msc_a *msc_a, enum osmo_cm_service_type cm_service_type);
 
diff --git a/src/libmsc/codec_filter.c b/src/libmsc/codec_filter.c
index a9d93a70d..7511f9026 100644
--- a/src/libmsc/codec_filter.c
+++ b/src/libmsc/codec_filter.c
@@ -98,46 +98,16 @@ int codec_filter_run(struct codec_filter *codec_filter, struct sdp_msg *result,
 	if (remote->audio_codecs.count)
 		sdp_audio_codecs_intersection(r, &remote->audio_codecs, true);
 
-#if 0
-	/* Future: If osmo-msc were able to trigger a re-assignment after the remote side has picked a codec mismatching
-	 * the initial Assignment, then this code here would make sense: keep the other codecs as available to choose
-	 * from, but put the currently assigned codec in the first position. So far we only offer the single assigned
-	 * codec, because we have no way to deal with the remote side picking a different codec.
-	 * Another approach would be to postpone assignment until we know the codecs from the remote side. */
 	if (sdp_audio_codec_is_set(a)) {
 		/* Assignment has completed, the chosen codec should be the first of the resulting SDP.
-		 * Make sure this is actually listed in the result SDP and move to first place. */
+		 * If present, make sure this is listed in first place.
+		 * If 'select' is NULL, the assigned codec is not present in the intersection of possible choices for
+		 * TFO. Just omit the assigned codec from the filter result, and it is the CC code's responsibility to
+		 * detect this and assign a working codec instead. */
 		struct sdp_audio_codec *select = sdp_audio_codecs_by_descr(r, a);
-
-		if (!select) {
-			/* Not present. Add. */
-			if (sdp_audio_codec_by_payload_type(r, a->payload_type, false)) {
-				/* Oh crunch, that payload type number is already in use.
-				 * Find an unused one. */
-				for (a->payload_type = 96; a->payload_type <= 127; a->payload_type++) {
-					if (!sdp_audio_codec_by_payload_type(r, a->payload_type, false))
-						break;
-				}
-
-				if (a->payload_type > 127)
-					return -ENOSPC;
-			}
-			select = sdp_audio_codecs_add_copy(r, a);
-		}
-
-		sdp_audio_codecs_select(r, select);
-	}
-#else
-	/* Currently, osmo-msc does not trigger re-assignment if the remote side has picked a codec that is different
-	 * from the already assigned codec.
-	 * So, if locally, Assignment has already chosen a codec, this is the single definitive result to be used
-	 * towards the CN. */
-	if (sdp_audio_codec_is_set(a)) {
-		/* Assignment has completed, the chosen codec should be the the only possible one. */
-		*r = (struct sdp_audio_codecs){};
-		sdp_audio_codecs_add_copy(r, a);
+		if (select)
+			sdp_audio_codecs_select(r, select);
 	}
-#endif
 	return 0;
 }
 
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index a4c2ba258..7bab2554a 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -283,7 +283,16 @@ static void _log_mncc_rx_tx(const char *file, int line,
 		break;
 	}
 
-	if (sdp && sdp[0] && (sdp_msg_from_sdp_str(&sdp_msg, sdp) == 0)) {
+	if (sdp && sdp[0]) {
+		int rc = sdp_msg_from_sdp_str(&sdp_msg, sdp);
+		if (rc != 0) {
+			LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_ERROR, file, line, "%s %s: invalid SDP message (trying anyway)\n",
+					  rx_tx,
+					  get_mncc_name(mncc->msg_type));
+			LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "erratic SDP: %s\n",
+					  osmo_quote_cstr_c(OTC_SELECT, sdp, -1));
+			return;
+		}
 		LOG_TRANS_CAT_SRC(trans, DMNCC, LOGL_DEBUG, file, line, "%s %s (RTP=%s)\n",
 				  rx_tx,
 				  get_mncc_name(mncc->msg_type),
@@ -1130,6 +1139,7 @@ static int gsm48_cc_tx_alerting(struct gsm_trans *trans, void *arg)
 	struct gsm_mncc *alerting = arg;
 	struct msgb *msg = gsm48_msgb_alloc_name("GSM 04.08 CC ALERT");
 	struct gsm48_hdr *gh = (struct gsm48_hdr *) msgb_put(msg, sizeof(*gh));
+	struct codec_filter *codecs = &trans->cc.codecs;
 	int rc;
 
 	gh->msg_type = GSM48_MT_CC_ALERTING;
@@ -1162,7 +1172,23 @@ static int gsm48_cc_tx_alerting(struct gsm_trans *trans, void *arg)
 		}
 	}
 
-	return trans_tx_gsm48(trans, msg);
+	/* First handle the MNCC event */
+	rc = trans_tx_gsm48(trans, msg);
+
+	/* Now see if the codecs are fine for TFO:
+	 * This is the first time we are told the MT call leg's codec capabilities, via the MNCC_ALERT_REQ from MT to
+	 * MO. Here, at MO, we have already assigned a specific codec. If the MT call leg does not support this codec,
+	 * but the MO does support one of MT's codecs, we need to re-assign our assigned codec to match MT. */
+	if (sdp_audio_codec_is_set(&codecs->assignment)
+	    && trans->cc.remote.audio_codecs.count
+	    && !sdp_audio_codecs_by_descr(&trans->cc.remote.audio_codecs, &codecs->assignment)) {
+		LOG_TRANS(trans, LOGL_INFO, "Remote call leg mismatches assigned codec: %s\n",
+			  codec_filter_to_str(&trans->cc.codecs, &trans->cc.local, &trans->cc.remote));
+
+		msc_a_tx_assignment(trans->msc_a);
+	}
+
+	return rc;
 }
 
 static int gsm48_cc_tx_progress(struct gsm_trans *trans, void *arg)
@@ -2055,17 +2081,23 @@ int cc_on_assignment_done(struct gsm_trans *trans)
 	switch (trans->cc.state) {
 	case GSM_CSTATE_INITIATED:
 	case GSM_CSTATE_MO_CALL_PROC:
-		/* MO call */
+		/* MO call, send ACK in form of an MNCC_RTP_CREATE (below) */
 		break;
 
 	case GSM_CSTATE_CALL_RECEIVED:
 	case GSM_CSTATE_MO_TERM_CALL_CONF:
-		/* MT call */
+		/* MT call, send ACK in form of an MNCC_RTP_CREATE (below) */
 		break;
 
 	case GSM_CSTATE_ACTIVE:
-		/* already active. MNCC finished before Abis completed the Assignment. */
-		break;
+		/* already active. We decided to re-assign later on during the call - at time of writing this never
+		 * happens. */
+	case GSM_CSTATE_CALL_DELIVERED:
+	case GSM_CSTATE_CONNECT_IND:
+		/* MNCC has progressed past the initial assignment. Usually it means that this happened: after
+		 * MNCC_ALERT_REQ, MO has triggered a re-assignment, to adjust MO's codec to MT's codec. */
+		LOG_TRANS(trans, LOGL_DEBUG, "Re-Assignment complete\n");
+		return 0;
 
 	default:
 		LOG_TRANS(trans, LOGL_ERROR, "Assignment done in unexpected CC state: %d\n", trans->cc.state);
diff --git a/src/libmsc/msc_a.c b/src/libmsc/msc_a.c
index e64b54d8f..c10afb8e3 100644
--- a/src/libmsc/msc_a.c
+++ b/src/libmsc/msc_a.c
@@ -636,7 +636,7 @@ int msc_a_ensure_cn_local_rtp(struct msc_a *msc_a, struct gsm_trans *cc_trans)
 }
 
 /* The MGW has given us a local IP address for the RAN side. Ready to start the Assignment of a voice channel. */
-static void msc_a_call_leg_ran_local_addr_available(struct msc_a *msc_a)
+void msc_a_tx_assignment(struct msc_a *msc_a)
 {
 	struct ran_msg msg;
 	struct gsm_trans *cc_trans = msc_a->cc.active_trans;
@@ -804,7 +804,7 @@ static void msc_a_fsm_communicating(struct osmo_fsm_inst *fi, uint32_t event, vo
 			  rtps->use_osmux ? "yes" : "no", rtps->local_osmux_cid);
 		switch (rtps->dir) {
 		case RTP_TO_RAN:
-			msc_a_call_leg_ran_local_addr_available(msc_a);
+			msc_a_tx_assignment(msc_a);
 			return;
 		case RTP_TO_CN:
 			cc_on_cn_local_rtp_port_known(rtps->for_trans);
diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c
index f82697503..47f000b2f 100644
--- a/src/libmsc/msc_ho.c
+++ b/src/libmsc/msc_ho.c
@@ -380,7 +380,7 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a)
 	struct vlr_subscr *vsub = msc_a_vsub(msc_a);
 	struct gsm_network *net = msc_a_net(msc_a);
 	struct gsm0808_channel_type channel_type;
-	struct gsm0808_speech_codec_list scl;
+	struct gsm0808_speech_codec_list scl = {};
 	struct gsm_trans *cc_trans = msc_a->cc.active_trans;
 	struct ran_msg ran_enc_msg = {
 		.msg_type = RAN_MSG_HANDOVER_REQUEST,
@@ -442,7 +442,13 @@ static void msc_ho_send_handover_request(struct msc_a *msc_a)
 		ran_enc_msg.handover_request.call_id_present = true;
 		ran_enc_msg.handover_request.call_id = cc_trans->call_id;
 
-		sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
+		/* Call assignment is now capable of re-assigning to overcome a codec mismatch with the remote call leg.
+		 * But for inter-MSC handover, that is not supported yet. So keep here the old limitation of only
+		 * offering the assigned codec. */
+		if (sdp_audio_codec_is_set(&cc_trans->cc.codecs.assignment))
+			sdp_audio_codec_to_speech_codec_list(&scl, &cc_trans->cc.codecs.assignment);
+		else
+			sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs);
 		if (!scl.len) {
 			msc_ho_failed(msc_a, GSM0808_CAUSE_EQUIPMENT_FAILURE, "Failed to compose"
 				      " Codec List (MSC Preferred) for Handover Request message\n");
diff --git a/tests/msc_vlr/msc_vlr_test_call.c b/tests/msc_vlr/msc_vlr_test_call.c
index cb3c77bb1..3b91524b1 100644
--- a/tests/msc_vlr/msc_vlr_test_call.c
+++ b/tests/msc_vlr/msc_vlr_test_call.c
@@ -1083,6 +1083,9 @@ static bool validate_sdp(const char *func, const char *desc,
 			return false;
 		}
 		expect_pos++;
+
+		/* only match first codec */
+		return true;
 	}
 	if (*expect_pos) {
 		BTW("%s: %s: ERROR: mismatch: expected %s to be listed, but not found", func, desc, *expect_pos);
diff --git a/tests/msc_vlr/msc_vlr_test_call.err b/tests/msc_vlr/msc_vlr_test_call.err
index a1da0f778..18bc2c18a 100644
--- a/tests/msc_vlr/msc_vlr_test_call.err
+++ b/tests/msc_vlr/msc_vlr_test_call.err
@@ -2634,19 +2634,22 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-6:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000004 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
   MSC --> MNCC: callref 0x80000004: MNCC_RTP_CREATE
 v=0

 o=OsmoMSC 0 0 IN IP4 10.23.23.1

 s=GSM Call

 c=IN IP4 10.23.23.1

 t=0 0

-m=audio 23 RTP/AVP 112

+m=audio 23 RTP/AVP 112 110 3 111

 a=rtpmap:112 AMR/8000

 a=fmtp:112 octet-align=1

+a=rtpmap:110 GSM-EFR/8000

+a=rtpmap:3 GSM/8000

+a=rtpmap:111 GSM-HR-08/8000

 a=ptime:20

 
 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR
@@ -4458,19 +4461,22 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-12:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000007 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
   MSC --> MNCC: callref 0x80000007: MNCC_RTP_CREATE
 v=0

 o=OsmoMSC 0 0 IN IP4 10.23.23.1

 s=GSM Call

 c=IN IP4 10.23.23.1

 t=0 0

-m=audio 23 RTP/AVP 112

+m=audio 23 RTP/AVP 112 110 3 111

 a=rtpmap:112 AMR/8000

 a=fmtp:112 octet-align=1

+a=rtpmap:110 GSM-EFR/8000

+a=rtpmap:3 GSM/8000

+a=rtpmap:111 GSM-HR-08/8000

 a=ptime:20

 
 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR
@@ -4859,19 +4865,22 @@ DBSSAP msc_a(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ){MSC_A_ST_
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI){UNINITIALIZED}: setting codecs to AMR:octet-align=1#112
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI:local-10-23-23-1-23){UNINITIALIZED}: setting remote addr to 1.2.3.4:1234
 DCC rtp_stream(IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ:trans-8:call-14:RTP_TO_RAN:no-CI:local-10-23-23-1-23:remote-1-2-3-4-1234){UNINITIALIZED}: Not committing: no MGW endpoint CI set up
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112
-DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
-DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) Assignment Complete: RAN: AMR:octet-align=1#112, CN: AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111
+DCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) codecs: 10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} (from: assigned=AMR:octet-align=1#112 MS={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111} bss={GSM#3,GSM-EFR#110,AMR:octet-align=1#112,GSM-HR-08#111} RAN={AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
+DMNCC trans(CC:INITIATED IMSI-901700000010650:MSISDN-46071:GERAN-A:CM_SERVICE_REQ callref-0x80000008 tid-8) tx MNCC_RTP_CREATE (RTP=10.23.23.1:23{AMR:octet-align=1#112,GSM-EFR#110,GSM#3,GSM-HR-08#111})
   MSC --> MNCC: callref 0x80000008: MNCC_RTP_CREATE
 v=0

 o=OsmoMSC 0 0 IN IP4 10.23.23.1

 s=GSM Call

 c=IN IP4 10.23.23.1

 t=0 0

-m=audio 23 RTP/AVP 112

+m=audio 23 RTP/AVP 112 110 3 111

 a=rtpmap:112 AMR/8000

 a=fmtp:112 octet-align=1

+a=rtpmap:110 GSM-EFR/8000

+a=rtpmap:3 GSM/8000

+a=rtpmap:111 GSM-HR-08/8000

 a=ptime:20

 
 - VALIDATE_SDP OK: cc_to_mncc_tx_last_sdp == t->mo_tx_sdp_mncc_rtp_create == AMR