diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
new file mode 100644
index 0000000..e5da138
--- /dev/null
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c
@@ -0,0 +1,377 @@
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <string.h>
+#include <err.h>
+
+#include <osmocom/core/talloc.h>
+#include <osmocom/core/application.h>
+
+#include <openbsc/debug.h>
+#include <openbsc/gsm_data.h>
+#include <openbsc/mgcp.h>
+#include <openbsc/mgcp_internal.h>
+
+#include "bscconfig.h"
+#ifndef BUILD_MGCP_TRANSCODING
+#error "Requires MGCP transcoding enabled (see --enable-mgcp-transcoding)"
+#endif
+
+#include "src/osmo-bsc_mgcp/mgcp_transcode.h"
+
+uint8_t *audio_frame_l16[] = {
+};
+
+struct rtp_packets {
+	float t;
+	int len;
+	char *data;
+};
+
+struct rtp_packets audio_packets_l16[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 332,
+		"\x80\x0B\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+		"\x00\x00\x40\x13\x5A\x9E\x40\x13\x00\x00\xBF\xED\xA5\x62\xBF\xED"
+	},
+};
+
+struct rtp_packets audio_packets_gsm[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 45,
+		"\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+		"\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+		"\xDE"
+	},
+};
+
+struct rtp_packets audio_packets_gsm_invalid_size[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 41,
+		"\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+		"\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+		"\xDE"
+	},
+};
+
+struct rtp_packets audio_packets_gsm_invalid_data[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 45,
+		"\x80\x03\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE"
+		"\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE\xEE"
+		"\xEE"
+	},
+};
+
+struct rtp_packets audio_packets_gsm_invalid_ptype[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 45,
+		"\x80\x08\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xD4\x7C\xE3\xE9\x62\x50\x39\xF0\xF8\xB4\x68\xEA\x6C\x0E\x81\x1B"
+		"\x56\x2A\xD5\xBC\x69\x9C\xD1\xF0\x66\x7A\xEC\x49\x7A\x33\x3D\x0A"
+		"\xDE"
+	},
+};
+
+struct rtp_packets audio_packets_g729[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 32,
+		"\x80\x12\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xAF\xC2\x81\x40\x00\xFA\xCE\xA4\x21\x7C\xC5\xC3\x4F\xA5\x98\xF5"
+		"\xB2\x95\xC4\xAD"
+	},
+};
+
+struct rtp_packets audio_packets_pcma[] = {
+	/* RTP: SeqNo=1, TS=160 */
+	{0.020000, 172,
+		"\x80\x08\x00\x01\x00\x00\x00\xA0\x11\x22\x33\x44"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+		"\xD5\xA5\xA3\xA5\xD5\x25\x23\x25\xD5\xA5\xA3\xA5\xD5\x25\x23\x25"
+	},
+};
+
+
+
+static int audio_name_to_type(const char *name)
+{
+	if (!strcasecmp(name, "gsm"))
+		return 3;
+#ifdef HAVE_BCG729
+	else if (!strcasecmp(name, "g729"))
+		return 18;
+#endif
+	else if (!strcasecmp(name, "pcma"))
+		return 8;
+	else if (!strcasecmp(name, "l16"))
+		return 11;
+	return -1;
+}
+
+int mgcp_get_trans_frame_size(void *state_, int nsamples, int dst);
+
+static int transcode_test(const char *srcfmt, const char *dstfmt,
+			  uint8_t *src_pkts, size_t src_pkt_size)
+{
+	char buf[4096] = {0x80, 0};
+	int rc;
+	struct mgcp_rtp_end *dst_end;
+	struct mgcp_rtp_end *src_end;
+	struct mgcp_trunk_config tcfg = {{0}};
+	struct mgcp_endpoint endp = {0};
+	struct mgcp_process_rtp_state *state;
+	int in_size;
+	int in_samples = 160;
+	int len, cont;
+
+	printf("== Transcoding test ==\n");
+	printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+	tcfg.endpoints = &endp;
+	tcfg.number_endpoints = 1;
+	endp.tcfg = &tcfg;
+	mgcp_free_endp(&endp);
+
+	dst_end = &endp.bts_end;
+	src_end = &endp.net_end;
+
+	src_end->payload_type = audio_name_to_type(srcfmt);
+	dst_end->payload_type = audio_name_to_type(dstfmt);
+
+	rc = mgcp_transcoding_setup(&endp, dst_end, src_end);
+	if (rc < 0)
+		errx(1, "setup failed: %s", strerror(-rc));
+
+	state = dst_end->rtp_process_data;
+	OSMO_ASSERT(state != NULL);
+
+	in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
+	OSMO_ASSERT(sizeof(buf) >= in_size + 12);
+
+	memcpy(buf, src_pkts, src_pkt_size);
+
+	len = src_pkt_size;
+
+	cont = mgcp_transcoding_process_rtp(&endp, dst_end,
+					    buf, &len, sizeof(buf));
+	if (cont < 0)
+		errx(1, "processing failed: %s", strerror(-cont));
+
+	if (len < 24) {
+		printf("encoded: %s\n", osmo_hexdump((unsigned char *)buf, len));
+	} else {
+		const char *str = osmo_hexdump((unsigned char *)buf, len);
+		int i = 0;
+		const int prefix = 4;
+		const int cutlen = 48;
+		int nchars = 0;
+
+		printf("encoded:\n");
+		do {
+			nchars = printf("%*s%-.*s", prefix, "", cutlen, str + i);
+			i += nchars - prefix;
+			printf("\n");
+		} while (nchars - prefix >= cutlen);
+	}
+	return 0;
+}
+
+static int test_repacking(int in_samples, int out_samples, int no_transcode)
+{
+	char buf[4096] = {0x80, 0};
+	int cc, rc;
+	struct mgcp_rtp_end *dst_end;
+	struct mgcp_rtp_end *src_end;
+	struct mgcp_config *cfg;
+	struct mgcp_trunk_config tcfg = {{0}};
+	struct mgcp_endpoint endp = {0};
+	struct mgcp_process_rtp_state *state;
+	int in_cnt;
+	int out_size;
+	int in_size;
+	uint32_t ts = 0;
+	uint16_t seq = 0;
+	const char *srcfmt = "pcma";
+	const char *dstfmt = no_transcode ? "pcma" : "l16";
+
+	cfg = mgcp_config_alloc();
+
+	tcfg.endpoints = &endp;
+	tcfg.number_endpoints = 1;
+	tcfg.cfg = cfg;
+	endp.tcfg = &tcfg;
+	endp.cfg = cfg;
+	mgcp_free_endp(&endp);
+
+	dst_end = &endp.bts_end;
+	src_end = &endp.net_end;
+
+	printf("== Transcoding test ==\n");
+	printf("converting %s -> %s\n", srcfmt, dstfmt);
+
+	src_end->payload_type = audio_name_to_type(srcfmt);
+	dst_end->payload_type = audio_name_to_type(dstfmt);
+
+	if (out_samples) {
+		dst_end->frame_duration_den = dst_end->rate;
+		dst_end->frame_duration_num = out_samples;
+		dst_end->frames_per_packet = 1;
+		dst_end->force_output_ptime = 1;
+	}
+
+	rc = mgcp_transcoding_setup(&endp, dst_end, src_end);
+	if (rc < 0)
+		errx(1, "setup failed: %s", strerror(-rc));
+
+	state = dst_end->rtp_process_data;
+	OSMO_ASSERT(state != NULL);
+
+	in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
+	OSMO_ASSERT(sizeof(buf) >= in_size + 12);
+
+	out_size = mgcp_transcoding_get_frame_size(state, -1, 1);
+	OSMO_ASSERT(sizeof(buf) >= out_size + 12);
+
+	buf[1] = src_end->payload_type;
+	*(uint16_t*)(buf+2) = htons(1);
+	*(uint32_t*)(buf+4) = htonl(0);
+	*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
+
+	for (in_cnt = 0; in_cnt < 16; in_cnt++) {
+		int cont;
+		int len;
+
+		/* fake PCMA data */
+		printf("generating %d %s input samples\n", in_samples, srcfmt);
+		for (cc = 0; cc < in_samples; cc++)
+			buf[12+cc] = cc;
+
+		*(uint16_t*)(buf+2) = htonl(seq);
+		*(uint32_t*)(buf+4) = htonl(ts);
+
+		seq += 1;
+		ts += in_samples;
+
+		cc += 12; /* include RTP header */
+
+		len = cc;
+
+		do {
+			cont = mgcp_transcoding_process_rtp(&endp, dst_end,
+							    buf, &len, sizeof(buf));
+			if (cont == -EAGAIN) {
+				fprintf(stderr, "Got EAGAIN\n");
+				break;
+			}
+
+			if (cont < 0)
+				errx(1, "processing failed: %s", strerror(-cont));
+
+			len -= 12; /* ignore RTP header */
+
+			printf("got %d %s output frames (%d octets)\n",
+			       len / out_size, dstfmt, len);
+
+			len = cont;
+		} while (len > 0);
+	}
+	return 0;
+}
+
+int main(int argc, char **argv)
+{
+	osmo_init_logging(&log_info);
+
+	printf("=== Transcoding Good Cases ===\n");
+
+	transcode_test("l16", "l16",
+		       (uint8_t *)audio_packets_l16[0].data,
+		       audio_packets_l16[0].len);
+	transcode_test("l16", "gsm",
+		       (uint8_t *)audio_packets_l16[0].data,
+		       audio_packets_l16[0].len);
+	transcode_test("l16", "pcma",
+		       (uint8_t *)audio_packets_l16[0].data,
+		       audio_packets_l16[0].len);
+	transcode_test("gsm", "l16",
+		       (uint8_t *)audio_packets_gsm[0].data,
+		       audio_packets_gsm[0].len);
+	transcode_test("gsm", "gsm",
+		       (uint8_t *)audio_packets_gsm[0].data,
+		       audio_packets_gsm[0].len);
+	transcode_test("gsm", "pcma",
+		       (uint8_t *)audio_packets_gsm[0].data,
+		       audio_packets_gsm[0].len);
+	transcode_test("pcma", "l16",
+		       (uint8_t *)audio_packets_pcma[0].data,
+		       audio_packets_pcma[0].len);
+	transcode_test("pcma", "gsm",
+		       (uint8_t *)audio_packets_pcma[0].data,
+		       audio_packets_pcma[0].len);
+	transcode_test("pcma", "pcma",
+		       (uint8_t *)audio_packets_pcma[0].data,
+		       audio_packets_pcma[0].len);
+
+	printf("=== Transcoding Bad Cases ===\n");
+
+	printf("Invalid size:\n");
+	transcode_test("gsm", "pcma",
+		       (uint8_t *)audio_packets_gsm_invalid_size[0].data,
+		       audio_packets_gsm_invalid_size[0].len);
+
+	printf("Invalid data:\n");
+	transcode_test("gsm", "pcma",
+		       (uint8_t *)audio_packets_gsm_invalid_data[0].data,
+		       audio_packets_gsm_invalid_data[0].len);
+
+	printf("Invalid payload type:\n");
+	transcode_test("gsm", "pcma",
+		       (uint8_t *)audio_packets_gsm_invalid_ptype[0].data,
+		       audio_packets_gsm_invalid_ptype[0].len);
+
+	printf("=== Repacking ===\n");
+
+	test_repacking(160, 160, 0);
+	test_repacking(160, 160, 1);
+	test_repacking(160, 80, 0);
+	test_repacking(160, 80, 1);
+	test_repacking(160, 320, 0);
+	test_repacking(160, 320, 1);
+	test_repacking(160, 240, 0);
+	test_repacking(160, 240, 1);
+	test_repacking(160, 100, 0);
+	test_repacking(160, 100, 1);
+
+	return 0;
+}
+