mgcp: Move the "codec" params to a struct

We might be offered multiple codecs by the remote and need to
switch between them once we receive data. Do this by moving it
to a struct so we can separate between proposed and current
codec. In SDP we can have multiple codecs but a global ptime.
The current code doesn't separate that clearly instead we write
it to the main codec.

9 files changed, 111 insertions, 97 deletions

diff --git a/openbsc/contrib/testconv/testconv_main.c b/openbsc/contrib/testconv/testconv_main.c
index 773be26a3..6c95c5542 100644
--- a/openbsc/contrib/testconv/testconv_main.c
+++ b/openbsc/contrib/testconv/testconv_main.c
@@ -64,16 +64,16 @@ int main(int argc, char **argv)
 	if (argc <= 2)
 		errx(1, "Usage: {gsm|g729|pcma|l16} {gsm|g729|pcma|l16} [SPP]");
 
-	if ((src_end->payload_type = audio_name_to_type(argv[1])) == -1)
+	if ((src_end->codec.payload_type = audio_name_to_type(argv[1])) == -1)
 		errx(1, "invalid input format '%s'", argv[1]);
-	if ((dst_end->payload_type = audio_name_to_type(argv[2])) == -1)
+	if ((dst_end->codec.payload_type = audio_name_to_type(argv[2])) == -1)
 		errx(1, "invalid output format '%s'", argv[2]);
 	if (argc > 3)
 		out_samples = atoi(argv[3]);
 
 	if (out_samples) {
-		dst_end->frame_duration_den = dst_end->rate;
-		dst_end->frame_duration_num = out_samples;
+		dst_end->codec.frame_duration_den = dst_end->codec.rate;
+		dst_end->codec.frame_duration_num = out_samples;
 		dst_end->frames_per_packet = 1;
 	}
 
@@ -87,7 +87,7 @@ int main(int argc, char **argv)
 	in_size = mgcp_transcoding_get_frame_size(state, in_samples, 0);
 	OSMO_ASSERT(sizeof(buf) >= in_size + 12);
 
-	buf[1] = src_end->payload_type;
+	buf[1] = src_end->codec.payload_type;
 	*(uint16_t*)(buf+2) = htons(1);
 	*(uint32_t*)(buf+4) = htonl(0);
 	*(uint32_t*)(buf+8) = htonl(0xaabbccdd);
diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h
index 3d308835e..245890994 100644
--- a/openbsc/include/openbsc/mgcp_internal.h
+++ b/openbsc/include/openbsc/mgcp_internal.h
@@ -67,6 +67,17 @@ struct mgcp_rtp_state {
 	struct mgcp_rtp_stream_state out_stream;
 };
 
+struct mgcp_rtp_codec {
+	uint32_t rate;
+	int channels;
+	uint32_t frame_duration_num;
+	uint32_t frame_duration_den;
+
+	int payload_type;
+	char *audio_name;
+	char *subtype_name;
+};
+
 struct mgcp_rtp_end {
 	/* statistics */
 	unsigned int packets;
@@ -77,17 +88,13 @@ struct mgcp_rtp_end {
 	/* in network byte order */
 	int rtp_port, rtcp_port;
 
+	/* audio codec information */
+	struct mgcp_rtp_codec codec;
+
 	/* per endpoint data */
-	int payload_type;
-	uint32_t rate;
-	int channels;
-	uint32_t frame_duration_num;
-	uint32_t frame_duration_den;
 	int  frames_per_packet;
 	uint32_t packet_duration_ms;
 	char *fmtp_extra;
-	char *audio_name;
-	char *subtype_name;
 	int output_enabled;
 	int force_output_ptime;
 
diff --git a/openbsc/src/libmgcp/mgcp_network.c b/openbsc/src/libmgcp/mgcp_network.c
index 7dcf3f3ae..587d4e88c 100644
--- a/openbsc/src/libmgcp/mgcp_network.c
+++ b/openbsc/src/libmgcp/mgcp_network.c
@@ -239,7 +239,7 @@ static int adjust_rtp_timestamp_offset(struct mgcp_endpoint *endp,
 			     inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
 			     endp->conn_mode);
 		} else {
-			tsdelta = rtp_end->rate * 20 / 1000;
+			tsdelta = rtp_end->codec.rate * 20 / 1000;
 			LOGP(DMGCP, LOGL_NOTICE,
 			     "Fixed packet duration and last timestamp delta "
 			     "are not available on 0x%x, "
@@ -326,8 +326,8 @@ void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
 	/* Use the BTS side parameters when passing the SDP data (for
 	 * downlink) to the net peer.
 	 */
-	*payload_type = endp->bts_end.payload_type;
-	*audio_name = endp->bts_end.audio_name;
+	*payload_type = endp->bts_end.codec.payload_type;
+	*audio_name = endp->bts_end.codec.audio_name;
 	*fmtp_extra = endp->bts_end.fmtp_extra;
 }
 
@@ -348,7 +348,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
 	uint16_t seq, udelta;
 	uint32_t timestamp, ssrc;
 	struct rtp_hdr *rtp_hdr;
-	int payload = rtp_end->payload_type;
+	int payload = rtp_end->codec.payload_type;
 
 	if (len < sizeof(*rtp_hdr))
 		return;
@@ -356,7 +356,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
 	rtp_hdr = (struct rtp_hdr *) data;
 	seq = ntohs(rtp_hdr->sequence);
 	timestamp = ntohl(rtp_hdr->timestamp);
-	arrival_time = get_current_ts(rtp_end->rate);
+	arrival_time = get_current_ts(rtp_end->codec.rate);
 	ssrc = ntohl(rtp_hdr->ssrc);
 	transit = arrival_time - timestamp;
 
@@ -380,7 +380,7 @@ void mgcp_patch_and_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *sta
 			inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
 			endp->conn_mode);
 		if (state->packet_duration == 0) {
-			state->packet_duration = rtp_end->rate * 20 / 1000;
+			state->packet_duration = rtp_end->codec.rate * 20 / 1000;
 			LOGP(DMGCP, LOGL_NOTICE,
 			     "Fixed packet duration is not available on 0x%x, "
 			     "using fixed 20ms instead: %d from %s:%d in %d\n",
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index ae275a890..d2a9f093e 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -598,20 +598,20 @@ static int parse_conn_mode(const char *msg, struct mgcp_endpoint *endp)
 	return ret;
 }
 
-static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
+static int set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
 			  int payload_type, const char *audio_name)
 {
-	int rate = rtp->rate;
-	int channels = rtp->channels;
+	int rate = codec->rate;
+	int channels = codec->channels;
 	char audio_codec[64];
 
-	talloc_free(rtp->subtype_name);
-	rtp->subtype_name = NULL;
-	talloc_free(rtp->audio_name);
-	rtp->audio_name = NULL;
+	talloc_free(codec->subtype_name);
+	codec->subtype_name = NULL;
+	talloc_free(codec->audio_name);
+	codec->audio_name = NULL;
 
 	if (payload_type != PTYPE_UNDEFINED)
-		rtp->payload_type = payload_type;
+		codec->payload_type = payload_type;
 
 	if (!audio_name) {
 		switch (payload_type) {
@@ -630,17 +630,17 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
 		   audio_codec, &rate, &channels) < 1)
 		return -EINVAL;
 
-	rtp->rate = rate;
-	rtp->channels = channels;
-	rtp->subtype_name = talloc_strdup(ctx, audio_codec);
-	rtp->audio_name = talloc_strdup(ctx, audio_name);
+	codec->rate = rate;
+	codec->channels = channels;
+	codec->subtype_name = talloc_strdup(ctx, audio_codec);
+	codec->audio_name = talloc_strdup(ctx, audio_name);
 
 	if (!strcmp(audio_codec, "G729")) {
-		rtp->frame_duration_num = 10;
-		rtp->frame_duration_den = 1000;
+		codec->frame_duration_num = 10;
+		codec->frame_duration_den = 1000;
 	} else {
-		rtp->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
-		rtp->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+		codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+		codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
 	}
 
 	if (payload_type < 0) {
@@ -654,7 +654,7 @@ static int set_audio_info(void *ctx, struct mgcp_rtp_end *rtp,
 				payload_type = 18;
 		}
 
-		rtp->payload_type = payload_type;
+		codec->payload_type = payload_type;
 	}
 
 	if (channels != 1)
@@ -761,7 +761,7 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 				if (payload != audio_payload)
 					break;
 
-				set_audio_info(p->cfg, rtp, payload, audio_name);
+				set_audio_info(p->cfg, &rtp->codec, payload, audio_name);
 			} else if (sscanf(line, "a=ptime:%d-%d",
 					  &ptime, &ptime2) >= 1) {
 				if (ptime2 > 0 && ptime2 != ptime)
@@ -769,8 +769,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 				else
 					rtp->packet_duration_ms = ptime;
 			} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
-				if (ptime2 * rtp->frame_duration_den >
-				    rtp->frame_duration_num * 1500)
+				if (ptime2 * rtp->codec.frame_duration_den >
+				    rtp->codec.frame_duration_num * 1500)
 					/* more than 1 frame */
 					rtp->packet_duration_ms = 0;
 			}
@@ -785,7 +785,7 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 				rtp->rtp_port = htons(port);
 				rtp->rtcp_port = htons(port + 1);
 				found_media = 1;
-				set_audio_info(p->cfg, rtp, audio_payload, NULL);
+				set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
 			}
 			break;
 		}
@@ -814,8 +814,8 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 		LOGP(DMGCP, LOGL_NOTICE,
 		     "Got media info via SDP: port %d, payload %d (%s), "
 		     "duration %d, addr %s\n",
-		     ntohs(rtp->rtp_port), rtp->payload_type,
-		     rtp->subtype_name ? rtp->subtype_name : "unknown",
+		     ntohs(rtp->rtp_port), rtp->codec.payload_type,
+		     rtp->codec.subtype_name ? rtp->codec.subtype_name : "unknown",
 		     rtp->packet_duration_ms, inet_ntoa(rtp->addr));
 
 	return found_media;
@@ -872,13 +872,13 @@ uint32_t mgcp_rtp_packet_duration(struct mgcp_endpoint *endp,
 	/* Get the number of frames per channel and packet */
 	if (rtp->frames_per_packet)
 		f = rtp->frames_per_packet;
-	else if (rtp->packet_duration_ms && rtp->frame_duration_num) {
-		int den = 1000 * rtp->frame_duration_num;
-		f = (rtp->packet_duration_ms * rtp->frame_duration_den + den/2)
+	else if (rtp->packet_duration_ms && rtp->codec.frame_duration_num) {
+		int den = 1000 * rtp->codec.frame_duration_num;
+		f = (rtp->packet_duration_ms * rtp->codec.frame_duration_den + den/2)
 			/ den;
 	}
 
-	return rtp->rate * f * rtp->frame_duration_num / rtp->frame_duration_den;
+	return rtp->codec.rate * f * rtp->codec.frame_duration_num / rtp->codec.frame_duration_den;
 }
 
 static int mgcp_parse_osmux_cid(const char *line)
@@ -1025,13 +1025,13 @@ mgcp_header_done:
 	endp->allocated = 1;
 
 	/* set up RTP media parameters */
-	set_audio_info(p->cfg, &endp->bts_end, tcfg->audio_payload, tcfg->audio_name);
+	set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
 	endp->bts_end.fmtp_extra = talloc_strdup(tcfg->endpoints,
 						tcfg->audio_fmtp_extra);
 	if (have_sdp)
 		parse_sdp_data(&endp->net_end, p);
 	else if (endp->local_options.codec)
-		set_audio_info(p->cfg, &endp->net_end,
+		set_audio_info(p->cfg, &endp->net_end.codec,
 			       PTYPE_UNDEFINED, endp->local_options.codec);
 
 	if (p->cfg->bts_force_ptime) {
@@ -1147,7 +1147,7 @@ static struct msgb *handle_modify_con(struct mgcp_parse_data *p)
 			     local_options);
 
 	if (!have_sdp && endp->local_options.codec)
-		set_audio_info(p->cfg, &endp->net_end,
+		set_audio_info(p->cfg, &endp->net_end.codec,
 			       PTYPE_UNDEFINED, endp->local_options.codec);
 
 	setup_rtp_processing(endp);
@@ -1461,24 +1461,24 @@ static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end)
 	end->dropped_packets = 0;
 	memset(&end->addr, 0, sizeof(end->addr));
 	end->rtp_port = end->rtcp_port = 0;
-	end->payload_type = -1;
+	end->codec.payload_type = -1;
 	end->local_alloc = -1;
 	talloc_free(end->fmtp_extra);
 	end->fmtp_extra = NULL;
-	talloc_free(end->subtype_name);
-	end->subtype_name = NULL;
-	talloc_free(end->audio_name);
-	end->audio_name = NULL;
+	talloc_free(end->codec.subtype_name);
+	end->codec.subtype_name = NULL;
+	talloc_free(end->codec.audio_name);
+	end->codec.audio_name = NULL;
 	talloc_free(end->rtp_process_data);
 	end->rtp_process_data = NULL;
 
 	/* Set default values */
-	end->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
-	end->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+	end->codec.frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+	end->codec.frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
 	end->frames_per_packet  = 0; /* unknown */
 	end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
-	end->rate               = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
-	end->channels           = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
+	end->codec.rate               = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
+	end->codec.channels           = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
 	end->output_enabled	= 0;
 }
 
diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
index 88de7d2b2..e961ba6cc 100644
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -45,22 +45,22 @@ int mgcp_transcoding_get_frame_size(void *state_, int nsamples, int dst)
 			1) * state->src_frame_size;
 }
 
-static enum audio_format get_audio_format(const struct mgcp_rtp_end *rtp_end)
+static enum audio_format get_audio_format(const struct mgcp_rtp_codec *codec)
 {
-	if (rtp_end->subtype_name) {
-		if (!strcmp("GSM", rtp_end->subtype_name))
+	if (codec->subtype_name) {
+		if (!strcmp("GSM", codec->subtype_name))
 			return AF_GSM;
-		if (!strcmp("PCMA", rtp_end->subtype_name))
+		if (!strcmp("PCMA", codec->subtype_name))
 			return AF_PCMA;
 #ifdef HAVE_BCG729
-		if (!strcmp("G729", rtp_end->subtype_name))
+		if (!strcmp("G729", codec->subtype_name))
 			return AF_G729;
 #endif
-		if (!strcmp("L16", rtp_end->subtype_name))
+		if (!strcmp("L16", codec->subtype_name))
 			return AF_L16;
 	}
 
-	switch (rtp_end->payload_type) {
+	switch (codec->payload_type) {
 	case 3 /* GSM */:
 		return AF_GSM;
 	case 8 /* PCMA */:
@@ -141,6 +141,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
 {
 	struct mgcp_process_rtp_state *state;
 	enum audio_format src_fmt, dst_fmt;
+	const struct mgcp_rtp_codec *src_codec = &src_end->codec;
+	const struct mgcp_rtp_codec *dst_codec = &dst_end->codec;
 
 	/* cleanup first */
 	if (dst_end->rtp_process_data) {
@@ -151,34 +153,34 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
 	if (!src_end)
 		return 0;
 
-	src_fmt = get_audio_format(src_end);
-	dst_fmt = get_audio_format(dst_end);
+	src_fmt = get_audio_format(src_codec);
+	dst_fmt = get_audio_format(dst_codec);
 
 	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);
+	     src_codec->subtype_name, src_codec->payload_type,
+	     dst_codec->subtype_name, dst_codec->payload_type);
 
 	if (src_fmt == AF_INVALID || dst_fmt == AF_INVALID) {
-		if (!src_end->subtype_name || !dst_end->subtype_name)
+		if (!src_codec->subtype_name || !dst_codec->subtype_name)
 			/* Not enough info, do nothing */
 			return 0;
 
-		if (strcmp(src_end->subtype_name, dst_end->subtype_name) == 0)
+		if (strcmp(src_codec->subtype_name, dst_codec->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);
+		     src_codec->audio_name, dst_codec->audio_name);
 		return -EINVAL;
 	}
 
-	if (src_end->rate && dst_end->rate && src_end->rate != dst_end->rate) {
+	if (src_codec->rate && dst_codec->rate && src_codec->rate != dst_codec->rate) {
 		LOGP(DMGCP, LOGL_ERROR,
 		     "Cannot transcode: rate conversion (%d -> %d) not supported.\n",
-		     src_end->rate, dst_end->rate);
+		     src_codec->rate, dst_codec->rate);
 		return -EINVAL;
 	}
 
@@ -269,8 +271,8 @@ int mgcp_transcoding_setup(struct mgcp_endpoint *endp,
 	     "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);
+	     src_fmt, src_codec->payload_type, src_codec->rate, src_end->fmtp_extra,
+	     dst_fmt, dst_codec->payload_type, dst_codec->rate, dst_end->fmtp_extra);
 
 	return 0;
 }
@@ -281,16 +283,19 @@ void mgcp_transcoding_net_downlink_format(struct mgcp_endpoint *endp,
 					  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;
+	struct mgcp_rtp_codec *net_codec = &endp->net_end.codec;
+	struct mgcp_rtp_codec *bts_codec = &endp->bts_end.codec;
+
+	if (!state || net_codec->payload_type < 0) {
+		*payload_type = bts_codec->payload_type;
+		*audio_name = bts_codec->audio_name;
 		*fmtp_extra = endp->bts_end.fmtp_extra;
 		return;
 	}
 
-	*payload_type = endp->net_end.payload_type;
+	*payload_type = net_codec->payload_type;
+	*audio_name = net_codec->audio_name;
 	*fmtp_extra = endp->net_end.fmtp_extra;
-	*audio_name = endp->net_end.audio_name;
 }
 
 static int decode_audio(struct mgcp_process_rtp_state *state,
diff --git a/openbsc/src/libmgcp/mgcp_vty.c b/openbsc/src/libmgcp/mgcp_vty.c
index 9ae0bc9fa..b29eb6b4d 100644
--- a/openbsc/src/libmgcp/mgcp_vty.c
+++ b/openbsc/src/libmgcp/mgcp_vty.c
@@ -147,6 +147,8 @@ static int config_write_mgcp(struct vty *vty)
 static void dump_rtp_end(const char *end_name, struct vty *vty,
 			struct mgcp_rtp_state *state, struct mgcp_rtp_end *end)
 {
+	struct mgcp_rtp_codec *codec = &end->codec;
+
 	vty_out(vty,
 		"  %s%s"
 		"   Timestamp Errs: %d->%d%s"
@@ -160,10 +162,10 @@ static void dump_rtp_end(const char *end_name, struct vty *vty,
 		state->in_stream.err_ts_counter,
 		state->out_stream.err_ts_counter, VTY_NEWLINE,
 		end->dropped_packets, VTY_NEWLINE,
-		end->payload_type, end->rate, end->channels, VTY_NEWLINE,
-		end->frame_duration_num, end->frame_duration_den, VTY_NEWLINE,
+		codec->payload_type, codec->rate, codec->channels, VTY_NEWLINE,
+		codec->frame_duration_num, codec->frame_duration_den, VTY_NEWLINE,
 		end->frames_per_packet, end->packet_duration_ms, VTY_NEWLINE,
-		end->fmtp_extra, end->audio_name, end->subtype_name, VTY_NEWLINE,
+		end->fmtp_extra, codec->audio_name, codec->subtype_name, VTY_NEWLINE,
 		end->output_enabled, end->force_output_ptime, VTY_NEWLINE);
 }
 
diff --git a/openbsc/src/osmo-bsc_nat/bsc_mgcp_utils.c b/openbsc/src/osmo-bsc_nat/bsc_mgcp_utils.c
index d58039788..97593031c 100644
--- a/openbsc/src/osmo-bsc_nat/bsc_mgcp_utils.c
+++ b/openbsc/src/osmo-bsc_nat/bsc_mgcp_utils.c
@@ -552,7 +552,7 @@ static int bsc_mgcp_policy_cb(struct mgcp_trunk_config *tcfg, int endpoint, int
 	bsc_msg = bsc_mgcp_rewrite((char *) nat->mgcp_msg, nat->mgcp_length,
 				   sccp->bsc_endp, nat->mgcp_cfg->source_addr,
 				   mgcp_endp->bts_end.local_port, osmux_cid,
-				   &mgcp_endp->net_end.payload_type);
+				   &mgcp_endp->net_end.codec.payload_type);
 	if (!bsc_msg) {
 		LOGP(DMGCP, LOGL_ERROR, "Failed to patch the msg.\n");
 		return MGCP_POLICY_CONT;
@@ -746,7 +746,7 @@ void bsc_mgcp_forward(struct bsc_connection *bsc, struct msgb *msg)
 	output = bsc_mgcp_rewrite((char * ) msg->l2h, msgb_l2len(msg), -1,
 				  bsc->nat->mgcp_cfg->source_addr,
 				  endp->net_end.local_port, -1,
-				  &endp->bts_end.payload_type);
+				  &endp->bts_end.codec.payload_type);
 	if (!output) {
 		LOGP(DMGCP, LOGL_ERROR, "Failed to rewrite MGCP msg.\n");
 		return;
diff --git a/openbsc/tests/mgcp/mgcp_test.c b/openbsc/tests/mgcp/mgcp_test.c
index 50c8b2d0a..06981dd44 100644
--- a/openbsc/tests/mgcp/mgcp_test.c
+++ b/openbsc/tests/mgcp/mgcp_test.c
@@ -410,7 +410,7 @@ static void test_messages(void)
 	/* reset endpoints */
 	for (i = 0; i < cfg->trunk.number_endpoints; i++) {
 		endp = &cfg->trunk.endpoints[i];
-		endp->net_end.payload_type = PTYPE_NONE;
+		endp->net_end.codec.payload_type = PTYPE_NONE;
 		endp->net_end.packet_duration_ms = -1;
 
 		OSMO_ASSERT(endp->conn_mode == MGCP_CONN_NONE);
@@ -498,18 +498,18 @@ static void test_messages(void)
 			fprintf(stderr, "endpoint %d: "
 				"payload type BTS %d (exp %d), NET %d (exp %d)\n",
 				last_endpoint,
-				endp->bts_end.payload_type, t->exp_bts_ptype,
-				endp->net_end.payload_type, t->exp_net_ptype);
+				endp->bts_end.codec.payload_type, t->exp_bts_ptype,
+				endp->net_end.codec.payload_type, t->exp_net_ptype);
 
 			if (t->exp_bts_ptype != PTYPE_IGNORE)
-				OSMO_ASSERT(endp->bts_end.payload_type ==
+				OSMO_ASSERT(endp->bts_end.codec.payload_type ==
 					    t->exp_bts_ptype);
 			if (t->exp_net_ptype != PTYPE_IGNORE)
-				OSMO_ASSERT(endp->net_end.payload_type ==
+				OSMO_ASSERT(endp->net_end.codec.payload_type ==
 					    t->exp_net_ptype);
 
 			/* Reset them again for next test */
-			endp->net_end.payload_type = PTYPE_NONE;
+			endp->net_end.codec.payload_type = PTYPE_NONE;
 		}
 	}
 
@@ -830,7 +830,7 @@ static void test_packet_error_detection(int patch_ssrc, int patch_ts)
 
 	mgcp_initialize_endp(&endp);
 
-	rtp->payload_type = 98;
+	rtp->codec.payload_type = 98;
 
 	for (i = 0; i < ARRAY_SIZE(test_rtp_packets1); ++i) {
 		struct rtp_packet_info *info = test_rtp_packets1 + i;
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
index 44f307251..079f62eba 100644
--- a/openbsc/tests/mgcp/mgcp_transcoding_test.c
+++ b/openbsc/tests/mgcp/mgcp_transcoding_test.c
@@ -183,14 +183,14 @@ static int given_configured_endpoint(int in_samples, int out_samples,
 	mgcp_initialize_endp(endp);
 
 	dst_end = &endp->bts_end;
-	dst_end->payload_type = audio_name_to_type(dstfmt);
+	dst_end->codec.payload_type = audio_name_to_type(dstfmt);
 
 	src_end = &endp->net_end;
-	src_end->payload_type = audio_name_to_type(srcfmt);
+	src_end->codec.payload_type = audio_name_to_type(srcfmt);
 
 	if (out_samples) {
-		dst_end->frame_duration_den = dst_end->rate;
-		dst_end->frame_duration_num = out_samples;
+		dst_end->codec.frame_duration_den = dst_end->codec.rate;
+		dst_end->codec.frame_duration_num = out_samples;
 		dst_end->frames_per_packet = 1;
 		dst_end->force_output_ptime = 1;
 	}
@@ -463,7 +463,7 @@ static int test_repacking(int in_samples, int out_samples, int no_transcode)
 	out_size = mgcp_transcoding_get_frame_size(state, -1, 1);
 	OSMO_ASSERT(sizeof(buf) >= out_size + 12);
 
-	buf[1] = endp->net_end.payload_type;
+	buf[1] = endp->net_end.codec.payload_type;
 	*(uint16_t*)(buf+2) = htons(1);
 	*(uint32_t*)(buf+4) = htonl(0);
 	*(uint32_t*)(buf+8) = htonl(0xaabbccdd);