diff --git a/include/osmocom/msc/msc_a.h b/include/osmocom/msc/msc_a.h
index 0276d62..4099d4c 100644
--- a/include/osmocom/msc/msc_a.h
+++ b/include/osmocom/msc/msc_a.h
@@ -216,6 +216,7 @@
 
 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_cmd(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 a9d93a7..7511f90 100644
--- a/src/libmsc/codec_filter.c
+++ b/src/libmsc/codec_filter.c
@@ -98,46 +98,16 @@
 	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);
+		if (select)
+			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);
-	}
-#endif
 	return 0;
 }
 
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index fe34127..f6ec81b 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -270,7 +270,16 @@
 		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),
@@ -748,6 +757,7 @@
 static void rx_mncc_sdp(struct gsm_trans *trans, uint32_t mncc_msg_type, const char *sdp,
 			const struct gsm_mncc_bearer_cap *bcap)
 {
+	struct codec_filter *codecs = &trans->cc.codecs;
 	struct call_leg *cl = trans->msc_a ? trans->msc_a->cc.call_leg : NULL;
 	struct rtp_stream *rtp_cn = cl ? cl->rtp[RTP_TO_CN] : NULL;
 
@@ -775,6 +785,30 @@
 		rtp_stream_set_remote_addr_and_codecs(rtp_cn, &trans->cc.remote);
 		rtp_stream_commit(rtp_cn);
 	}
+
+	/* See if we need to switch codecs to maintain TFO: has the remote side changed the codecs information? If we
+	 * have already assigned a specific codec here, but the remote call leg has now chosen a different codec, we
+	 * need to re-assign this call leg to match the remote leg. */
+	if (!sdp_audio_codec_is_set(&codecs->assignment)) {
+		/* Voice channel assignment has not completed. Do not interfere. */
+		return;
+	}
+	if (!trans->cc.remote.audio_codecs.count) {
+		/* Don't know remote codecs, nothing to do. */
+		return;
+	}
+	if (sdp_audio_codecs_by_descr(&trans->cc.remote.audio_codecs, &codecs->assignment)) {
+		/* The assigned codec is part of the remote codec set. All is well. */
+		/* TODO: maybe this should require exactly the *first* remote codec to match, because we cannot flexibly
+		 * transcode, and assume the actual payload we will receive is listed in the first place? */
+		return;
+	}
+
+	/* We've already completed Assignment of a voice channel (some time ago), and now the remote side has changed
+	 * to a mismatching codec (list). Try to re-assign this side to a matching codec. */
+	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_cmd(trans->msc_a);
 }
 
 static int gsm48_cc_tx_setup(struct gsm_trans *trans, void *arg)
@@ -2049,17 +2083,23 @@
 	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 e64b54d..a933bd2 100644
--- a/src/libmsc/msc_a.c
+++ b/src/libmsc/msc_a.c
@@ -636,7 +636,7 @@
 }
 
 /* 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_cmd(struct msc_a *msc_a)
 {
 	struct ran_msg msg;
 	struct gsm_trans *cc_trans = msc_a->cc.active_trans;
@@ -804,7 +804,7 @@
 			  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_cmd(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 f826975..47f000b 100644
--- a/src/libmsc/msc_ho.c
+++ b/src/libmsc/msc_ho.c
@@ -380,7 +380,7 @@
 	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 @@
 		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 cb3c77b..3b91524 100644
--- a/tests/msc_vlr/msc_vlr_test_call.c
+++ b/tests/msc_vlr/msc_vlr_test_call.c
@@ -1083,6 +1083,9 @@
 			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 4af1bce..5758175 100644
--- a/tests/msc_vlr/msc_vlr_test_call.err
+++ b/tests/msc_vlr/msc_vlr_test_call.err
@@ -2636,19 +2636,22 @@
 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
@@ -4457,19 +4460,22 @@
 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 @@
 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