diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
index e961ba6cc..38daeb8ae 100644
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -397,11 +397,62 @@ static int encode_audio(struct mgcp_process_rtp_state *state,
 	return nbytes;
 }
 
+static struct mgcp_rtp_end *source_for_dest(struct mgcp_endpoint *endp,
+					struct mgcp_rtp_end *dst_end)
+{
+	if (&endp->bts_end == dst_end)
+		return &endp->net_end;
+	else if (&endp->net_end == dst_end)
+		return &endp->bts_end;
+	OSMO_ASSERT(0);
+}
+
+/*
+ * With some modems we get offered multiple codecs
+ * and we have selected one of them. It might not
+ * be the right one and we need to detect this with
+ * the first audio packets. One difficulty is that
+ * we patch the rtp payload type in place, so we
+ * need to discuss this.
+ */
+struct mgcp_process_rtp_state *check_transcode_state(
+				struct mgcp_endpoint *endp,
+				struct mgcp_rtp_end *dst_end,
+				struct rtp_hdr *rtp_hdr)
+{
+	struct mgcp_rtp_end *src_end;
+
+	/* Only deal with messages from net to bts */
+	if (&endp->bts_end != dst_end)
+		goto done;
+
+	src_end = source_for_dest(endp, dst_end);
+
+	/* Already patched */
+	if (rtp_hdr->payload_type == dst_end->codec.payload_type)
+		goto done;
+	/* The payload we expect */
+	if (rtp_hdr->payload_type == src_end->codec.payload_type)
+		goto done;
+	/* The matching alternate payload type? Then switch */
+	if (rtp_hdr->payload_type == src_end->alt_codec.payload_type) {
+		struct mgcp_config *cfg = endp->cfg;
+		struct mgcp_rtp_codec tmp_codec = src_end->alt_codec;
+		src_end->alt_codec = src_end->codec;
+		src_end->codec = tmp_codec;
+		cfg->setup_rtp_processing_cb(endp, &endp->net_end, &endp->bts_end);
+		cfg->setup_rtp_processing_cb(endp, &endp->bts_end, &endp->net_end);
+	}
+
+done:
+	return dst_end->rtp_process_data;
+}
+
 int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
 				struct mgcp_rtp_end *dst_end,
 			     char *data, int *len, int buf_size)
 {
-	struct mgcp_process_rtp_state *state = dst_end->rtp_process_data;
+	struct mgcp_process_rtp_state *state;
 	const size_t rtp_hdr_size = sizeof(struct rtp_hdr);
 	struct rtp_hdr *rtp_hdr = (struct rtp_hdr *) data;
 	char *payload_data = (char *) &rtp_hdr->data[0];
@@ -414,6 +465,7 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
 	uint32_t ts_no;
 	int rc;
 
+	state = check_transcode_state(endp, dst_end, rtp_hdr);
 	if (!state)
 		return 0;
 
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
index 079f62eba..cf679b356 100644
--- a/openbsc/tests/mgcp/mgcp_transcoding_test.c
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c
@@ -174,6 +174,9 @@ static int given_configured_endpoint(int in_samples, int out_samples,
 	tcfg = talloc_zero(cfg, struct mgcp_trunk_config);
 	endp = talloc_zero(tcfg, struct mgcp_endpoint);
 
+	cfg->setup_rtp_processing_cb = mgcp_transcoding_setup;
+	cfg->rtp_processing_cb = mgcp_transcoding_process_rtp;
+	cfg->get_net_downlink_format_cb = mgcp_transcoding_net_downlink_format;
 
 	tcfg->endpoints = endp;
 	tcfg->number_endpoints = 1;
@@ -433,6 +436,63 @@ static void test_transcode_result(void)
 	}
 }
 
+static void test_transcode_change(void)
+{
+	char buf[4096] = {0x80, 0};
+	void *ctx;
+
+	struct mgcp_endpoint *endp;
+	struct mgcp_process_rtp_state *state;
+	struct rtp_hdr *hdr;
+
+	int len, res;
+
+	{
+		/* from GSM to PCMA and same ptime */
+		printf("Testing Initial G729->GSM, PCMA->GSM\n");
+		given_configured_endpoint(160, 0, "g729", "gsm", &ctx, &endp);
+		endp->net_end.alt_codec = endp->net_end.codec;
+		endp->net_end.alt_codec.payload_type = audio_name_to_type("pcma");
+		state = endp->bts_end.rtp_process_data;
+
+		/* initial transcoding work */
+		OSMO_ASSERT(state->src_fmt == AF_G729);
+		OSMO_ASSERT(state->dst_fmt == AF_GSM);
+		OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 8);
+		OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
+
+		/* result */
+		len = audio_packets_pcma[0].len;
+		memcpy(buf, audio_packets_pcma[0].data, len);
+		res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+		OSMO_ASSERT(res == sizeof(struct rtp_hdr));
+		OSMO_ASSERT(state->sample_cnt == 0);
+		OSMO_ASSERT(state->src_fmt == AF_PCMA);
+		OSMO_ASSERT(state->dst_fmt == AF_GSM);
+		OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 18);
+		OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
+
+		len = res;
+		res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+		OSMO_ASSERT(res == -ENOMSG);
+
+
+		/* now check that comfort noise doesn't change anything */
+		len = audio_packets_pcma[1].len;
+		memcpy(buf, audio_packets_pcma[1].data, len);
+		hdr = (struct rtp_hdr *) buf;
+		hdr->payload_type = 11;
+		res = mgcp_transcoding_process_rtp(endp, &endp->bts_end, buf, &len, ARRAY_SIZE(buf));
+		OSMO_ASSERT(state->sample_cnt == 80);
+		OSMO_ASSERT(state->src_fmt == AF_PCMA);
+		OSMO_ASSERT(state->dst_fmt == AF_GSM);
+		OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 18);
+		OSMO_ASSERT(endp->net_end.codec.payload_type == 8);
+
+		talloc_free(ctx);
+	}
+}
+
 static int test_repacking(int in_samples, int out_samples, int no_transcode)
 {
 	char buf[4096] = {0x80, 0};
@@ -582,6 +642,7 @@ int main(int argc, char **argv)
 	test_repacking(160, 100, 1);
 	test_rtp_seq_state();
 	test_transcode_result();
+	test_transcode_change();
 
 	return 0;
 }
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.ok b/openbsc/tests/mgcp/mgcp_transcoding_test.ok
index 7c1c8cebd..5cfc2897e 100644
--- a/openbsc/tests/mgcp/mgcp_transcoding_test.ok
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.ok
@@ -536,3 +536,4 @@ got 1 pcma output frames (80 octets) count=12
 generating 160 pcma input samples
 got 1 pcma output frames (80 octets) count=12
 got 1 pcma output frames (80 octets) count=12
+Testing Initial G729->GSM, PCMA->GSM