diff --git a/openbsc/src/osmo-bsc_mgcp/Makefile.am b/openbsc/src/osmo-bsc_mgcp/Makefile.am
index be39977..fba76b4 100644
--- a/openbsc/src/osmo-bsc_mgcp/Makefile.am
+++ b/openbsc/src/osmo-bsc_mgcp/Makefile.am
@@ -1,17 +1,12 @@
 AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir)
 AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOGSM_CFLAGS) \
-	$(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS) \
-	$(LIBBCG729_CFLAGS)
+	$(LIBOSMOVTY_CFLAGS) $(LIBOSMOABIS_CFLAGS) $(COVERAGE_CFLAGS)
 
 bin_PROGRAMS = osmo-bsc_mgcp
 
 osmo_bsc_mgcp_SOURCES = mgcp_main.c
-if BUILD_MGCP_TRANSCODING
-    osmo_bsc_mgcp_SOURCES += mgcp_transcode.c
-endif
+
 osmo_bsc_mgcp_LDADD = $(top_builddir)/src/libcommon/libcommon.a \
 		 $(top_builddir)/src/libmgcp/libmgcp.a -lrt \
 		 $(LIBOSMOVTY_LIBS) $(LIBOSMOCORE_LIBS) \
 		 $(LIBOSMONETIF_LIBS) $(LIBBCG729_LIBS)
-
-noinst_HEADERS = g711common.h mgcp_transcode.h
diff --git a/openbsc/src/osmo-bsc_mgcp/g711common.h b/openbsc/src/osmo-bsc_mgcp/g711common.h
deleted file mode 100644
index cb35fc6..0000000
--- a/openbsc/src/osmo-bsc_mgcp/g711common.h
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  PCM - A-Law conversion
- *  Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
- *
- *  Wrapper for linphone Codec class by Simon Morlat <simon.morlat@linphone.org>
- *
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; either version 2 of the License, or
- *  (at your option) any later version.
- *
- *  This program is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *  GNU General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License
- *  along with this program; if not, write to the Free Software
- *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-static inline int val_seg(int val)
-{
-	int r = 0;
-	val >>= 7; /*7 = 4 + 3*/
-	if (val & 0xf0) {
-		val >>= 4;
-		r += 4;
-	}
-	if (val & 0x0c) {
-		val >>= 2;
-		r += 2;
-	}
-	if (val & 0x02)
-		r += 1;
-	return r;
-}
-
-/*
- * s16_to_alaw() - Convert a 16-bit linear PCM value to 8-bit A-law
- *
- * s16_to_alaw() accepts an 16-bit integer and encodes it as A-law data.
- *
- *		Linear Input Code	Compressed Code
- *	------------------------	---------------
- *	0000000wxyza			000wxyz
- *	0000001wxyza			001wxyz
- *	000001wxyzab			010wxyz
- *	00001wxyzabc			011wxyz
- *	0001wxyzabcd			100wxyz
- *	001wxyzabcde			101wxyz
- *	01wxyzabcdef			110wxyz
- *	1wxyzabcdefg			111wxyz
- *
- * For further information see John C. Bellamy's Digital Telephony, 1982,
- * John Wiley & Sons, pps 98-111 and 472-476.
- * G711 is designed for 13 bits input signal, this function add extra shifting to take this into account.
- */
-
-static inline unsigned char s16_to_alaw(int pcm_val)
-{
-	int		mask;
-	int		seg;
-	unsigned char	aval;
-
-	if (pcm_val >= 0) {
-		mask = 0xD5;
-	} else {
-		mask = 0x55;
-		pcm_val = -pcm_val;
-		if (pcm_val > 0x7fff)
-			pcm_val = 0x7fff;
-	}
-
-	if (pcm_val < 256) /*256 = 32 << 3*/
-		aval = pcm_val >> 4; /*4 = 1 + 3*/
-	else {
-		/* Convert the scaled magnitude to segment number. */
-		seg = val_seg(pcm_val);
-		aval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f);
-	}
-	return aval ^ mask;
-}
-
-/*
- * alaw_to_s16() - Convert an A-law value to 16-bit linear PCM
- *
- */
-static inline int alaw_to_s16(unsigned char a_val)
-{
-	int		t;
-	int		seg;
-
-	a_val ^= 0x55;
-	t = a_val & 0x7f;
-	if (t < 16)
-		t = (t << 4) + 8;
-	else {
-		seg = (t >> 4) & 0x07;
-		t = ((t & 0x0f) << 4) + 0x108;
-		t <<= seg -1;
-	}
-	return ((a_val & 0x80) ? t : -t);
-}
-/*
- * s16_to_ulaw() - Convert a linear PCM value to u-law
- *
- * In order to simplify the encoding process, the original linear magnitude
- * is biased by adding 33 which shifts the encoding range from (0 - 8158) to
- * (33 - 8191). The result can be seen in the following encoding table:
- *
- *	Biased Linear Input Code	Compressed Code
- *	------------------------	---------------
- *	00000001wxyza			000wxyz
- *	0000001wxyzab			001wxyz
- *	000001wxyzabc			010wxyz
- *	00001wxyzabcd			011wxyz
- *	0001wxyzabcde			100wxyz
- *	001wxyzabcdef			101wxyz
- *	01wxyzabcdefg			110wxyz
- *	1wxyzabcdefgh			111wxyz
- *
- * Each biased linear code has a leading 1 which identifies the segment
- * number. The value of the segment number is equal to 7 minus the number
- * of leading 0's. The quantization interval is directly available as the
- * four bits wxyz.  * The trailing bits (a - h) are ignored.
- *
- * Ordinarily the complement of the resulting code word is used for
- * transmission, and so the code word is complemented before it is returned.
- *
- * For further information see John C. Bellamy's Digital Telephony, 1982,
- * John Wiley & Sons, pps 98-111 and 472-476.
- */
-
-static inline unsigned char s16_to_ulaw(int pcm_val)	/* 2's complement (16-bit range) */
-{
-	int mask;
-	int seg;
-	unsigned char uval;
-
-	if (pcm_val < 0) {
-		pcm_val = 0x84 - pcm_val;
-		mask = 0x7f;
-	} else {
-		pcm_val += 0x84;
-		mask = 0xff;
-	}
-	if (pcm_val > 0x7fff)
-		pcm_val = 0x7fff;
-
-	/* Convert the scaled magnitude to segment number. */
-	seg = val_seg(pcm_val);
-
-	/*
-	 * Combine the sign, segment, quantization bits;
-	 * and complement the code word.
-	 */
-	uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0x0f);
-	return uval ^ mask;
-}
-
-/*
- * ulaw_to_s16() - Convert a u-law value to 16-bit linear PCM
- *
- * First, a biased linear code is derived from the code word. An unbiased
- * output can then be obtained by subtracting 33 from the biased code.
- *
- * Note that this function expects to be passed the complement of the
- * original code word. This is in keeping with ISDN conventions.
- */
-static inline int ulaw_to_s16(unsigned char u_val)
-{
-	int t;
-
-	/* Complement to obtain normal u-law value. */
-	u_val = ~u_val;
-
-	/*
-	 * Extract and bias the quantization bits. Then
-	 * shift up by the segment number and subtract out the bias.
-	 */
-	t = ((u_val & 0x0f) << 3) + 0x84;
-	t <<= (u_val & 0x70) >> 4;
-
-	return ((u_val & 0x80) ? (0x84 - t) : (t - 0x84));
-}
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c
index 6b72965..8c3808a 100644
--- a/openbsc/src/osmo-bsc_mgcp/mgcp_main.c
+++ b/openbsc/src/osmo-bsc_mgcp/mgcp_main.c
@@ -50,7 +50,7 @@
 #include "../../bscconfig.h"
 
 #ifdef BUILD_MGCP_TRANSCODING
-#include "mgcp_transcode.h"
+#include "openbsc/mgcp_transcode.h"
 #endif
 
 /* this is here for the vty... it will never be called */
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
deleted file mode 100644
index 581cd32..0000000
--- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.c
+++ /dev/null
@@ -1,550 +0,0 @@
-/*
- * (C) 2014 by On-Waves
- * All Rights Reserved
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU Affero General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Affero General Public License for more details.
- *
- * You should have received a copy of the GNU Affero General Public License
- * along with this program.  If not, see <http://www.gnu.org/licenses/>.
- *
- */
-
-#include <stdlib.h>
-#include <string.h>
-#include <errno.h>
-
-
-#include "../../bscconfig.h"
-
-#include "g711common.h"
-#include <gsm.h>
-#ifdef HAVE_BCG729
-#include <bcg729/decoder.h>
-#include <bcg729/encoder.h>
-#endif
-
-#include <openbsc/debug.h>
-#include <openbsc/mgcp.h>
-#include <openbsc/mgcp_internal.h>
-
-#include <osmocom/core/talloc.h>
-
-enum audio_format {
-	AF_INVALID,
-	AF_S16,
-	AF_L16,
-	AF_GSM,
-	AF_G729,
-	AF_PCMA
-};
-
-struct mgcp_process_rtp_state {
-	/* decoding */
-	enum audio_format src_fmt;
-	union {
-		gsm gsm_handle;
-#ifdef HAVE_BCG729
-		bcg729DecoderChannelContextStruct *g729_dec;
-#endif
-	} src;
-	size_t src_frame_size;
-	size_t src_samples_per_frame;
-
-	/* processing */
-
-	/* encoding */
-	enum audio_format dst_fmt;
-	union {
-		gsm gsm_handle;
-#ifdef HAVE_BCG729
-		bcg729EncoderChannelContextStruct *g729_enc;
-#endif
-	} dst;
-	size_t dst_frame_size;
-	size_t dst_samples_per_frame;
-	int dst_packet_duration;
-
-	int is_running;
-	uint16_t next_seq;
-	uint32_t next_time;
-	int16_t samples[10*160];
-	size_t sample_cnt;
-	size_t sample_offs;
-};
-
-int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
-{
-	struct mgcp_process_rtp_state *state = state_;
-	if (dst)
-		return (nsamples >= 0 ?
-			nsamples / state->dst_samples_per_frame :
-			1) * state->dst_frame_size;
-	else
-		return (nsamples >= 0 ?
-			nsamples / state->src_samples_per_frame :
-			1) * state->src_frame_size;
-}
-
-static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
-{
-	if (rtp_end->subtype_name) {
-		if (!strcmp("GSM", rtp_end->subtype_name))
-			return AF_GSM;
-		if (!strcmp("PCMA", rtp_end->subtype_name))
-			return AF_PCMA;
-#ifdef HAVE_BCG729
-		if (!strcmp("G729", rtp_end->subtype_name))
-			return AF_G729;
-#endif
-		if (!strcmp("L16", rtp_end->subtype_name))
-			return AF_L16;
-	}
-
-	switch (rtp_end->payload_type) {
-	case 3 /* GSM */:
-		return AF_GSM;
-	case 8 /* PCMA */:
-		return AF_PCMA;
-#ifdef HAVE_BCG729
-	case 18 /* G.729 */:
-		return AF_G729;
-#endif
-	case 11 /* L16 */:
-		return AF_L16;
-	default:
-		return AF_INVALID;
-	}
-}
-
-static void l16_encode(short *sample, unsigned char *buf, size_t n)
-{
-	for (; n > 0; --n, ++sample, buf += 2) {
-		buf[0] = sample[0] >> 8;
-		buf[1] = sample[0] & 0xff;
-	}
-}
-
-static void l16_decode(unsigned char *buf, short *sample, size_t n)
-{
-	for (; n > 0; --n, ++sample, buf += 2)
-		sample[0] = ((short)buf[0] << 8) | buf[1];
-}
-
-static void alaw_encode(short *sample, unsigned char *buf, size_t n)
-{
-	for (; n > 0; --n)
-		*(buf++) = s16_to_alaw(*(sample++));
-}
-
-static void alaw_decode(unsigned char *buf, short *sample, size_t n)
-{
-	for (; n > 0; --n)
-		*(sample++) = alaw_to_s16(*(buf++));
-}
-
-static int processing_state_destructor(struct mgcp_process_rtp_state *state)
-{
-	switch (state->src_fmt) {
-	case AF_GSM:
-		if (state->dst.gsm_handle)
-			gsm_destroy(state->src.gsm_handle);
-		break;
-#ifdef HAVE_BCG729
-	case AF_G729:
-		if (state->src.g729_dec)
-			closeBcg729DecoderChannel(state->src.g729_dec);
-		break;
-#endif
-	default:
-		break;
-	}
-	switch (state->dst_fmt) {
-	case AF_GSM:
-		if (state->dst.gsm_handle)
-			gsm_destroy(state->dst.gsm_handle);
-		break;
-#ifdef HAVE_BCG729
-	case AF_G729:
-		if (state->dst.g729_enc)
-			closeBcg729EncoderChannel(state->dst.g729_enc);
-		break;
-#endif
-	default:
-		break;
-	}
-	return 0;
-}
-
-int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
-			   struct mgcp_rtp_end *dst_end,
-			   struct mgcp_rtp_end *src_end)
-{
-	struct mgcp_process_rtp_state *state;
-	enum audio_format src_fmt, dst_fmt;
-
-	/* cleanup first */
-	if (dst_end->rtp_process_data) {
-		talloc_free(dst_end->rtp_process_data);
-		dst_end->rtp_process_data = NULL;
-	}
-
-	if (!src_end)
-		return 0;
-
-	src_fmt = get_audio_format(src_end);
-	dst_fmt = get_audio_format(dst_end);
-
-	LOGP(DMGCP, LOGL_ERROR,
-	     "Checking transcoding: %s (%d) -> %s (%d)\n",
-	     src_end->subtype_name, src_end->payload_type,
-	     dst_end->subtype_name, dst_end->payload_type);
-
-	if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
-		if (!src_end->subtype_name || !dst_end->subtype_name)
-			/* Not enough info, do nothing */
-			return 0;
-
-		if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
-			/* Nothing to do */
-			return 0;
-
-		LOGP(DMGCP, LOGL_ERROR,
-		     "Cannot transcode: %s codec not supported (%s -> %s).\n",
-		     src_fmt != AF_INVALID ? "destination" : "source",
-		     src_end->audio_name, dst_end->audio_name);
-		return -EINVAL;
-	}
-
-	if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
-		LOGP(DMGCP, LOGL_ERROR,
-		     "Cannot transcode: rate conversion (%d -> %d) not supported.\n",
-		     src_end->rate, dst_end->rate);
-		return -EINVAL;
-	}
-
-	state = talloc_zero(endp->tcfg->cfg, struct mgcp_process_rtp_state);
-	talloc_set_destructor(state, processing_state_destructor);
-	dst_end->rtp_process_data = state;
-
-	state->src_fmt = src_fmt;
-
-	switch (state->src_fmt) {
-	case AF_L16:
-	case AF_S16:
-		state->src_frame_size = 80 * sizeof(short);
-		state->src_samples_per_frame = 80;
-		break;
-	case AF_GSM:
-		state->src_frame_size = sizeof(gsm_frame);
-		state->src_samples_per_frame = 160;
-		state->src.gsm_handle = gsm_create();
-		if (!state->src.gsm_handle) {
-			LOGP(DMGCP, LOGL_ERROR,
-			     "Failed to initialize GSM decoder.\n");
-			return -EINVAL;
-		}
-		break;
-#ifdef HAVE_BCG729
-	case AF_G729:
-		state->src_frame_size = 10;
-		state->src_samples_per_frame = 80;
-		state->src.g729_dec = initBcg729DecoderChannel();
-		if (!state->src.g729_dec) {
-			LOGP(DMGCP, LOGL_ERROR,
-			     "Failed to initialize G.729 decoder.\n");
-			return -EINVAL;
-		}
-		break;
-#endif
-	case AF_PCMA:
-		state->src_frame_size = 80;
-		state->src_samples_per_frame = 80;
-		break;
-	default:
-		break;
-	}
-
-	state->dst_fmt = dst_fmt;
-
-	switch (state->dst_fmt) {
-	case AF_L16:
-	case AF_S16:
-		state->dst_frame_size = 80*sizeof(short);
-		state->dst_samples_per_frame = 80;
-		break;
-	case AF_GSM:
-		state->dst_frame_size = sizeof(gsm_frame);
-		state->dst_samples_per_frame = 160;
-		state->dst.gsm_handle = gsm_create();
-		if (!state->dst.gsm_handle) {
-			LOGP(DMGCP, LOGL_ERROR,
-			     "Failed to initialize GSM encoder.\n");
-			return -EINVAL;
-		}
-		break;
-#ifdef HAVE_BCG729
-	case AF_G729:
-		state->dst_frame_size = 10;
-		state->dst_samples_per_frame = 80;
-		state->dst.g729_enc = initBcg729EncoderChannel();
-		if (!state->dst.g729_enc) {
-			LOGP(DMGCP, LOGL_ERROR,
-			     "Failed to initialize G.729 decoder.\n");
-			return -EINVAL;
-		}
-		break;
-#endif
-	case AF_PCMA:
-		state->dst_frame_size = 80;
-		state->dst_samples_per_frame = 80;
-		break;
-	default:
-		break;
-	}
-
-	if (dst_end->force_output_ptime)
-		state->dst_packet_duration = mgcp_rtp_packet_duration(endp, dst_end);
-
-	LOGP(DMGCP, LOGL_INFO,
-	     "Initialized RTP processing on: 0x%x "
-	     "conv: %d (%d, %d, %s) -> %d (%d, %d, %s)\n",
-	     ENDPOINT_NUMBER(endp),
-	     src_fmt, src_end->payload_type, src_end->rate, src_end->fmtp_extra,
-	     dst_fmt, dst_end->payload_type, dst_end->rate, dst_end->fmtp_extra);
-
-	return 0;
-}
-
-void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
-					  int *payload_type,
-					  const char**audio_name,
-					  const char**fmtp_extra)
-{
-	struct mgcp_process_rtp_state *state = endp->net_end.rtp_process_data;
-	if (!state || endp->net_end.payload_type < 0) {
-		*payload_type = endp->bts_end.payload_type;
-		*audio_name = endp->bts_end.audio_name;
-		*fmtp_extra = endp->bts_end.fmtp_extra;
-		return;
-	}
-
-	*payload_type = endp->net_end.payload_type;
-	*fmtp_extra = endp->net_end.fmtp_extra;
-	*audio_name = endp->net_end.audio_name;
-}
-
-static int decode_audio(struct mgcp_process_rtp_state *state,
-			uint8_t **src, size_t *nbytes)
-{
-	while (*nbytes >= state->src_frame_size) {
-		if (state->sample_cnt + state->src_samples_per_frame > ARRAY_SIZE(state->samples)) {
-			LOGP(DMGCP, LOGL_ERROR,
-			     "Sample buffer too small: %d > %d.\n",
-			     state->sample_cnt + state->src_samples_per_frame,
-			     ARRAY_SIZE(state->samples));
-			return -ENOSPC;
-		}
-		switch (state->src_fmt) {
-		case AF_GSM:
-			if (gsm_decode(state->src.gsm_handle,
-				       (gsm_byte *)*src, state->samples + state->sample_cnt) < 0) {
-				LOGP(DMGCP, LOGL_ERROR,
-				     "Failed to decode GSM.\n");
-				return -EINVAL;
-			}
-			break;
-#ifdef HAVE_BCG729
-		case AF_G729:
-			bcg729Decoder(state->src.g729_dec, *src, 0, state->samples + state->sample_cnt);
-			break;
-#endif
-		case AF_PCMA:
-			alaw_decode(*src, state->samples + state->sample_cnt,
-				    state->src_samples_per_frame);
-			break;
-		case AF_S16:
-			memmove(state->samples + state->sample_cnt, *src,
-				state->src_frame_size);
-			break;
-		case AF_L16:
-			l16_decode(*src, state->samples + state->sample_cnt,
-				   state->src_samples_per_frame);
-			break;
-		default:
-			break;
-		}
-		*src        += state->src_frame_size;
-		*nbytes     -= state->src_frame_size;
-		state->sample_cnt += state->src_samples_per_frame;
-	}
-	return 0;
-}
-
-static int encode_audio(struct mgcp_process_rtp_state *state,
-			uint8_t *dst, size_t buf_size, size_t max_samples)
-{
-	int nbytes = 0;
-	size_t nsamples = 0;
-	/* Encode samples into dst */
-	while (nsamples + state->dst_samples_per_frame <= max_samples) {
-		if (nbytes + state->dst_frame_size > buf_size) {
-			if (nbytes > 0)
-				break;
-
-			/* Not even one frame fits into the buffer */
-			LOGP(DMGCP, LOGL_INFO,
-			     "Encoding (RTP) buffer too small: %d > %d.\n",
-			     nbytes + state->dst_frame_size, buf_size);
-			return -ENOSPC;
-		}
-		switch (state->dst_fmt) {
-		case AF_GSM:
-			gsm_encode(state->dst.gsm_handle,
-				   state->samples + state->sample_offs, dst);
-			break;
-#ifdef HAVE_BCG729
-		case AF_G729:
-			bcg729Encoder(state->dst.g729_enc,
-				      state->samples + state->sample_offs, dst);
-			break;
-#endif
-		case AF_PCMA:
-			alaw_encode(state->samples + state->sample_offs, dst,
-				    state->src_samples_per_frame);
-			break;
-		case AF_S16:
-			memmove(dst, state->samples + state->sample_offs,
-				state->dst_frame_size);
-			break;
-		case AF_L16:
-			l16_encode(state->samples + state->sample_offs, dst,
-				   state->src_samples_per_frame);
-			break;
-		default:
-			break;
-		}
-		dst        += state->dst_frame_size;
-		nbytes     += state->dst_frame_size;
-		state->sample_offs += state->dst_samples_per_frame;
-		nsamples   += state->dst_samples_per_frame;
-	}
-	state->sample_cnt -= nsamples;
-	return nbytes;
-}
-
-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;
-	size_t rtp_hdr_size = 12;
-	char *payload_data = data + rtp_hdr_size;
-	int payload_len = *len - rtp_hdr_size;
-	uint8_t *src = (uint8_t *)payload_data;
-	uint8_t *dst = (uint8_t *)payload_data;
-	size_t nbytes = payload_len;
-	size_t nsamples;
-	size_t max_samples;
-	uint32_t ts_no;
-	int rc;
-
-	if (!state)
-		return 0;
-
-	if (state->src_fmt == state->dst_fmt) {
-		if (!state->dst_packet_duration)
-			return 0;
-
-		/* TODO: repackage without transcoding */
-	}
-
-	/* If the remaining samples do not fit into a fixed ptime,
-	 * a) discard them, if the next packet is much later
-	 * b) add silence and * send it, if the current packet is not
-	 *    yet too late
-	 * c) append the sample data, if the timestamp matches exactly
-	 */
-
-	/* TODO: check payload type (-> G.711 comfort noise) */
-
-	if (payload_len > 0) {
-		ts_no = ntohl(*(uint32_t*)(data+4));
-		if (!state->is_running)
-			state->next_seq = ntohs(*(uint32_t*)(data+4));
-
-		state->is_running = 1;
-
-		if (state->sample_cnt > 0) {
-			int32_t delta = ts_no - state->next_time;
-			/* TODO: check sequence? reordering? packet loss? */
-
-			if (delta > state->sample_cnt)
-				/* There is a time gap between the last packet
-				 * and the current one. Just discard the
-				 * partial data that is left in the buffer.
-				 * TODO: This can be improved by adding silence
-				 * instead if the delta is small enough.
-				 */
-				state->sample_cnt = 0;
-			else if (delta < 0) {
-				LOGP(DMGCP, LOGL_NOTICE,
-				     "RTP time jumps backwards, delta = %d, "
-				     "discarding buffered samples\n",
-				     delta);
-				state->sample_cnt = 0;
-				state->sample_offs = 0;
-				return -EAGAIN;
-			}
-
-			/* Make sure the samples start without offset */
-			if (state->sample_offs && state->sample_cnt)
-				memmove(&state->samples[0],
-					&state->samples[state->sample_offs],
-					state->sample_cnt *
-					sizeof(state->samples[0]));
-		}
-
-		state->sample_offs = 0;
-
-		/* Append decoded audio to samples */
-		decode_audio(state, &src, &nbytes);
-
-		if (nbytes > 0)
-			LOGP(DMGCP, LOGL_NOTICE,
-			     "Skipped audio frame in RTP packet: %d octets\n",
-			     nbytes);
-	} else
-		ts_no = state->next_time;
-
-	if (state->sample_cnt < state->dst_packet_duration)
-		return -EAGAIN;
-
-	max_samples =
-		state->dst_packet_duration ?
-		state->dst_packet_duration : state->sample_cnt;
-
-	nsamples = state->sample_cnt;
-
-	rc = encode_audio(state, dst, buf_size, max_samples);
-	if (rc <= 0)
-		return rc;
-
-	nsamples -= state->sample_cnt;
-
-	*len = rtp_hdr_size + rc;
-	*(uint16_t*)(data+2) = htonl(state->next_seq);
-	*(uint32_t*)(data+4) = htonl(ts_no);
-
-	state->next_seq += 1;
-	state->next_time = ts_no + nsamples;
-
-	return nsamples ? rtp_hdr_size : 0;
-}
diff --git a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h b/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
deleted file mode 100644
index 0961634..0000000
--- a/openbsc/src/osmo-bsc_mgcp/mgcp_transcode.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * (C) 2014 by On-Waves
- * All Rights Reserved
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU Affero General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU Affero General Public License for more details.
- *
- * You should have received a copy of the GNU Affero General Public License
- * along with this program.  If not, see <http://www.gnu.org/licenses/>.
- *
- */
-#ifndef OPENBSC_MGCP_TRANSCODE_H
-#define OPENBSC_MGCP_TRANSCODE_H
-
-int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
-			   struct mgcp_rtp_end *dst_end,
-			   struct mgcp_rtp_end *src_end);
-
-void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
-					  int *payload_type,
-					  const char**audio_name,
-					  const char**fmtp_extra);
-
-int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
-				 struct mgcp_rtp_end *dst_end,
-				 char *data, int *len, int buf_size);
-
-int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst);
-#endif /* OPENBSC_MGCP_TRANSCODE_H */