diff --git a/src/libmsc/a_iface_bssap.c b/src/libmsc/a_iface_bssap.c
index 1ace43d..11d3673 100644
--- a/src/libmsc/a_iface_bssap.c
+++ b/src/libmsc/a_iface_bssap.c
@@ -502,11 +502,50 @@
 	return 0;
 }
 
+/* Use the speech codec info we go with the assignment complete to dtermine
+ * which codec we will signal to the MGW */
+static enum mgcp_codecs mgcp_codec_from_sc(struct gsm0808_speech_codec *sc)
+{
+	switch (sc->type) {
+	case GSM0808_SCT_FR1:
+		return CODEC_GSM_8000_1;
+		break;
+	case GSM0808_SCT_FR2:
+		return CODEC_GSMEFR_8000_1;
+		break;
+	case GSM0808_SCT_FR3:
+		return CODEC_AMR_8000_1;
+		break;
+	case GSM0808_SCT_FR4:
+		return CODEC_AMRWB_16000_1;
+		break;
+	case GSM0808_SCT_FR5:
+		return CODEC_AMRWB_16000_1;
+		break;
+	case GSM0808_SCT_HR1:
+		return CODEC_GSMHR_8000_1;
+		break;
+	case GSM0808_SCT_HR3:
+		return CODEC_AMR_8000_1;
+		break;
+	case GSM0808_SCT_HR4:
+		return CODEC_AMRWB_16000_1;
+		break;
+	case GSM0808_SCT_HR6:
+		return CODEC_AMRWB_16000_1;
+		break;
+	default:
+		return CODEC_PCMU_8000_1;
+		break;
+	}
+}
+
 /* Endpoint to handle assignment complete */
 static int bssmap_rx_ass_compl(struct gsm_subscriber_connection *conn, struct msgb *msg,
 			       struct tlv_parsed *tp)
 {
 	struct sockaddr_storage rtp_addr;
+	struct gsm0808_speech_codec sc;
 	struct sockaddr_in *rtp_addr_in;
 	int rc;
 
@@ -525,6 +564,15 @@
 		return -EINVAL;
 	}
 
+	/* Decode speech codec (choosen) element */
+	rc = gsm0808_dec_speech_codec(&sc, TLVP_VAL(tp, GSM0808_IE_SPEECH_CODEC),
+					 TLVP_LEN(tp, GSM0808_IE_SPEECH_CODEC));
+	if (rc < 0) {
+		LOGPCONN(conn, LOGL_ERROR, "Unable to decode speech codec (choosen).\n");
+		return -EINVAL;
+	}
+	conn->rtp.codec_ran = mgcp_codec_from_sc(&sc);
+
 	/* use address / port supplied with the AoIP
 	 * transport address element */
 	if (rtp_addr.ss_family == AF_INET) {
diff --git a/src/libmsc/gsm_04_08_cc.c b/src/libmsc/gsm_04_08_cc.c
index 8becd05..2c17e22 100644
--- a/src/libmsc/gsm_04_08_cc.c
+++ b/src/libmsc/gsm_04_08_cc.c
@@ -319,6 +319,15 @@
 	/* Which subscriber do we want to track trans1 or trans2? */
 	log_set_context(LOG_CTX_VLR_SUBSCR, trans1->vsub);
 
+	/* This call briding mechanism is only used with the internal MNCC.
+	 * functionality (with external MNCC briding would be done by the PBX)
+	 * This means 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. */
+	trans1->conn->rtp.codec_cn = trans1->conn->rtp.codec_ran;
+	trans2->conn->rtp.codec_cn = trans1->conn->rtp.codec_ran;
+
 	/* Bridge RTP streams */
 	rc = msc_mgcp_call_complete(trans1, trans2->conn->rtp.local_port_cn,
 				    trans2->conn->rtp.local_addr_cn);
@@ -1716,6 +1725,16 @@
 	struct gsm_mncc_rtp *rtp = arg;
 	struct in_addr addr;
 
+	/* FIXME: in *rtp we should get the codec information of the remote
+	 * leg. We will have to populate trans->conn->rtp.codec_cn with a
+	 * meaningful value based on this information but unfortunately we
+	 * can't do that yet because the mncc API can not signal dynamic
+	 * payload types yet. This must be fixed first. Also there may be
+	 * additional members necessary in trans->conn->rtp because we
+	 * somehow need to deal with dynamic payload types that do not
+	 * comply to 3gpp's assumptions of payload type numbers on the A
+	 * interface. See also related tickets: OS#3399 and OS1683 */
+
 	/* Find callref */
 	trans = trans_find_by_callref(net, rtp->callref);
 	if (!trans) {
diff --git a/src/libmsc/msc_mgcp.c b/src/libmsc/msc_mgcp.c
index f5bdeb7..e58b249 100644
--- a/src/libmsc/msc_mgcp.c
+++ b/src/libmsc/msc_mgcp.c
@@ -277,22 +277,16 @@
 	struct mgcp_msg mgcp_msg;
 	struct msgb *msg;
 	int rc;
-
-#ifdef BUILD_IU
 	struct gsm_trans *trans;
 	struct gsm_subscriber_connection *conn;
-#endif
 
 	OSMO_ASSERT(mgcp_ctx);
 	mgcp = mgcp_ctx->mgcp;
 	OSMO_ASSERT(mgcp);
-
-#ifdef BUILD_IU
 	trans = mgcp_ctx->trans;
 	OSMO_ASSERT(trans);
 	conn = trans->conn;
 	OSMO_ASSERT(conn);
-#endif
 
 	/* NOTE: In case of error, we will not be able to perform any DLCX
 	 * operation because until this point we do not have requested any
@@ -396,22 +390,16 @@
 	struct mgcp_msg mgcp_msg;
 	struct msgb *msg;
 	int rc;
-
-#ifdef BUILD_IU
 	struct gsm_trans *trans;
 	struct gsm_subscriber_connection *conn;
-#endif
 
 	OSMO_ASSERT(mgcp_ctx);
 	mgcp = mgcp_ctx->mgcp;
 	OSMO_ASSERT(mgcp);
-
-#ifdef BUILD_IU
 	trans = mgcp_ctx->trans;
 	OSMO_ASSERT(trans);
 	conn = trans->conn;
 	OSMO_ASSERT(conn);
-#endif
 
 	switch (event) {
 	case EV_CRCX_RAN_RESP:
@@ -593,7 +581,9 @@
 		.conn_id = mgcp_ctx->conn_id_cn,
 		.conn_mode = MGCP_CONN_RECV_SEND,
 		.audio_ip = conn->rtp.remote_addr_cn,
-		.audio_port = conn->rtp.remote_port_cn
+		.audio_port = conn->rtp.remote_port_cn,
+		.codecs[0] = conn->rtp.codec_cn,
+		.codecs_len = 1
 	};
 	if (osmo_strlcpy(mgcp_msg.endpoint, mgcp_ctx->rtp_endpoint, sizeof(mgcp_msg.endpoint)) >=
 	    MGCP_ENDPOINT_MAXLEN) {
@@ -710,7 +700,9 @@
 		.conn_id = mgcp_ctx->conn_id_ran,
 		.conn_mode = MGCP_CONN_RECV_SEND,
 		.audio_ip = conn->rtp.remote_addr_ran,
-		.audio_port = conn->rtp.remote_port_ran
+		.audio_port = conn->rtp.remote_port_ran,
+		.codecs[0] = conn->rtp.codec_ran,
+		.codecs_len = 1
 	};
 	if (osmo_strlcpy(mgcp_msg.endpoint, mgcp_ctx->rtp_endpoint, sizeof(mgcp_msg.endpoint)) >=
 	    MGCP_ENDPOINT_MAXLEN) {