diff --git a/openbsc/src/libmgcp/Makefile.am b/openbsc/src/libmgcp/Makefile.am
index 262ad34..e5dab1a 100644
--- a/openbsc/src/libmgcp/Makefile.am
+++ b/openbsc/src/libmgcp/Makefile.am
@@ -1,9 +1,15 @@
 AM_CPPFLAGS = $(all_includes) -I$(top_srcdir)/include -I$(top_builddir)
-AM_CFLAGS=-Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) \
-	   $(LIBOSMONETIF_CFLAGS) $(COVERAGE_CFLAGS)
+AM_CFLAGS = -Wall $(LIBOSMOCORE_CFLAGS) $(LIBOSMOVTY_CFLAGS) \
+	    $(LIBOSMONETIF_CFLAGS) $(COVERAGE_CFLAGS) $(LIBBCG729_CFLAGS)
 AM_LDFLAGS = $(LIBOSMOCORE_LIBS) $(LIBOSMOGSM_LIBS) \
-	     $(LIBOSMONETIF_LIBS) $(COVERAGE_LDFLAGS)
+	     $(LIBOSMONETIF_LIBS) $(COVERAGE_LDFLAGS) $(LIBBCG729_LIBS)
 
 noinst_LIBRARIES = libmgcp.a
 
+noinst_HEADERS = g711common.h
+
 libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c osmux.c
+
+if BUILD_MGCP_TRANSCODING
+    libmgcp_a_SOURCES += mgcp_transcode.c
+endif
diff --git a/openbsc/src/libmgcp/g711common.h b/openbsc/src/libmgcp/g711common.h
new file mode 100644
index 0000000..cb35fc6
--- /dev/null
+++ b/openbsc/src/libmgcp/g711common.h
@@ -0,0 +1,187 @@
+/*
+ *  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/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
new file mode 100644
index 0000000..581cd32
--- /dev/null
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -0,0 +1,550 @@
+/*
+ * (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;
+}