Merge branch 'zecke/features/dynamic-codec-switch'

12 files changed, 384 insertions, 112 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..34c3d973a 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 +88,14 @@ struct mgcp_rtp_end {
 	/* in network byte order */
 	int rtp_port, rtcp_port;
 
+	/* audio codec information */
+	struct mgcp_rtp_codec codec;
+	struct mgcp_rtp_codec alt_codec; /* TODO/XXX: make it generic */
+
 	/* 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/include/openbsc/rtp.h b/openbsc/include/openbsc/rtp.h
index 451d0defa..718fa84e6 100644
--- a/openbsc/include/openbsc/rtp.h
+++ b/openbsc/include/openbsc/rtp.h
@@ -35,4 +35,5 @@ struct rtp_hdr {
 	uint16_t sequence;
 	uint32_t timestamp;
 	uint32_t ssrc;
+	uint8_t data[0];
 } __attribute__((packed));
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..a728b67c4 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)
@@ -738,7 +738,9 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 {
 	char *line;
 	int found_media = 0;
+	/* TODO/XXX make it more generic */
 	int audio_payload = -1;
+	int audio_payload_alt = -1;
 
 	for_each_line(line, p->save) {
 		switch (line[0]) {
@@ -758,10 +760,12 @@ static int parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p)
 
 			if (sscanf(line, "a=rtpmap:%d %63s",
 				   &payload, audio_name) == 2) {
-				if (payload != audio_payload)
-					break;
-
-				set_audio_info(p->cfg, rtp, payload, audio_name);
+				if (payload == audio_payload)
+					set_audio_info(p->cfg, &rtp->codec,
+							payload, audio_name);
+				else if (payload == audio_payload_alt)
+					set_audio_info(p->cfg, &rtp->alt_codec,
+							payload, audio_name);
 			} else if (sscanf(line, "a=ptime:%d-%d",
 					  &ptime, &ptime2) >= 1) {
 				if (ptime2 > 0 && ptime2 != ptime)
@@ -769,23 +773,29 @@ 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)
+				/* TODO/XXX: Store this per codec and derive it on use */
+				if (ptime2 * rtp->codec.frame_duration_den >
+				    rtp->codec.frame_duration_num * 1500)
 					/* more than 1 frame */
 					rtp->packet_duration_ms = 0;
 			}
 			break;
 		}
 		case 'm': {
-			int port;
+			int port, rc;
 			audio_payload = -1;
+			audio_payload_alt = -1;
 
-			if (sscanf(line, "m=audio %d RTP/AVP %d",
-				   &port, &audio_payload) == 2) {
+			rc = sscanf(line, "m=audio %d RTP/AVP %d %d",
+				   &port, &audio_payload, &audio_payload_alt);
+			if (rc >= 2) {
 				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);
+				if (rc == 3)
+					set_audio_info(p->cfg, &rtp->alt_codec,
+							audio_payload_alt, NULL);
 			}
 			break;
 		}
@@ -814,8 +824,8 @@ 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 +882,13 @@ 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 +1035,13 @@ 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 +1157,7 @@ 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);
@@ -1449,6 +1459,19 @@ struct mgcp_trunk_config *mgcp_trunk_num(struct mgcp_config *cfg, int index)
 	return NULL;
 }
 
+static void mgcp_rtp_codec_reset(struct mgcp_rtp_codec *codec)
+{
+	codec->payload_type = -1;
+	talloc_free(codec->subtype_name);
+	codec->subtype_name = NULL;
+	talloc_free(codec->audio_name);
+	codec->audio_name = NULL;
+	codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+	codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+	codec->rate               = DEFAULT_RTP_AUDIO_DEFAULT_RATE;
+	codec->channels           = DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS;
+}
+
 static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end)
 {
 	if (end->local_alloc == PORT_ALLOC_DYNAMIC) {
@@ -1461,25 +1484,19 @@ 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->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->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->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->output_enabled	= 0;
+
+	mgcp_rtp_codec_reset(&end->codec);
+	mgcp_rtp_codec_reset(&end->alt_codec);
 }
 
 static void mgcp_rtp_end_init(struct mgcp_rtp_end *end)
diff --git a/openbsc/src/libmgcp/mgcp_transcode.c b/openbsc/src/libmgcp/mgcp_transcode.c
index 4d4cec8a2..38daeb8ae 100644
--- a/openbsc/src/libmgcp/mgcp_transcode.c
+++ b/openbsc/src/libmgcp/mgcp_transcode.c
@@ -28,6 +28,7 @@
 #include <openbsc/mgcp.h>
 #include <openbsc/mgcp_internal.h>
 #include <openbsc/mgcp_transcode.h>
+#include <openbsc/rtp.h>
 
 #include <osmocom/core/talloc.h>
 
@@ -44,22 +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 */:
@@ -140,6 +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) {
@@ -150,34 +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;
 	}
 
@@ -268,8 +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;
 }
@@ -280,16 +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,
@@ -391,13 +397,65 @@ 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;
-	size_t rtp_hdr_size = 12;
-	char *payload_data = data + rtp_hdr_size;
+	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];
 	int payload_len = *len - rtp_hdr_size;
 	uint8_t *src = (uint8_t *)payload_data;
 	uint8_t *dst = (uint8_t *)payload_data;
@@ -407,6 +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;
 
@@ -427,9 +486,9 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
 	/* TODO: check payload type (-> G.711 comfort noise) */
 
 	if (payload_len > 0) {
-		ts_no = ntohl(*(uint32_t*)(data+4));
+		ts_no = ntohl(rtp_hdr->timestamp);
 		if (!state->is_running) {
-			state->next_seq = ntohs(*(uint16_t*)(data+2));
+			state->next_seq = ntohs(rtp_hdr->sequence);
 			state->next_time = ts_no;
 			state->is_running = 1;
 		}
@@ -504,8 +563,8 @@ int mgcp_transcoding_process_rtp(struct mgcp_endpoint *endp,
 	nsamples -= state->sample_cnt;
 
 	*len = rtp_hdr_size + rc;
-	*(uint16_t*)(data+2) = htons(state->next_seq);
-	*(uint32_t*)(data+4) = htonl(ts_no);
+	rtp_hdr->sequence = htons(state->next_seq);
+	rtp_hdr->timestamp = htonl(ts_no);
 
 	state->next_seq += 1;
 	state->next_time = ts_no + nsamples;
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..a0575032c 100644
--- a/openbsc/tests/mgcp/mgcp_test.c
+++ b/openbsc/tests/mgcp/mgcp_test.c
@@ -1,6 +1,6 @@
 /*
- * (C) 2011-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
- * (C) 2011-2012 by On-Waves
+ * (C) 2011-2012,2014 by Holger Hans Peter Freyther <zecke@selfish.org>
+ * (C) 2011-2012,2014 by On-Waves
  * All Rights Reserved
  *
  * This program is free software; you can redistribute it and/or modify
@@ -268,6 +268,61 @@ static void test_strline(void)
 #define PTYPE_NONE 128
 #define PTYPE_NYI  PTYPE_NONE
 
+#define CRCX_MULT_1 "CRCX 2 1@mgw MGCP 1.0\r\n"	\
+		 "M: recvonly\r\n"		\
+		 "C: 2\r\n"			\
+		 "X\r\n"			\
+		 "L: p:20\r\n"		\
+		 "\r\n"				\
+		 "v=0\r\n"			\
+		 "c=IN IP4 123.12.12.123\r\n"	\
+		 "m=audio 5904 RTP/AVP 18 97\r\n"\
+		 "a=rtpmap:18 G729/8000\r\n"	\
+		 "a=rtpmap:97 GSM-EFR/8000\r\n"	\
+		 "a=ptime:40\r\n"
+
+#define CRCX_MULT_2 "CRCX 2 2@mgw MGCP 1.0\r\n"	\
+		 "M: recvonly\r\n"		\
+		 "C: 2\r\n"			\
+		 "X\r\n"			\
+		 "L: p:20\r\n"		\
+		 "\r\n"				\
+		 "v=0\r\n"			\
+		 "c=IN IP4 123.12.12.123\r\n"	\
+		 "m=audio 5904 RTP/AVP 18 97 101\r\n"\
+		 "a=rtpmap:18 G729/8000\r\n"	\
+		 "a=rtpmap:97 GSM-EFR/8000\r\n"	\
+		 "a=rtpmap:101 FOO/8000\r\n"	\
+		 "a=ptime:40\r\n"
+
+#define CRCX_MULT_3 "CRCX 2 3@mgw MGCP 1.0\r\n"	\
+		 "M: recvonly\r\n"		\
+		 "C: 2\r\n"			\
+		 "X\r\n"			\
+		 "L: p:20\r\n"		\
+		 "\r\n"				\
+		 "v=0\r\n"			\
+		 "c=IN IP4 123.12.12.123\r\n"	\
+		 "m=audio 5904 RTP/AVP\r\n"	\
+		 "a=rtpmap:18 G729/8000\r\n"	\
+		 "a=rtpmap:97 GSM-EFR/8000\r\n"	\
+		 "a=rtpmap:101 FOO/8000\r\n"	\
+		 "a=ptime:40\r\n"
+
+#define CRCX_MULT_4 "CRCX 2 4@mgw MGCP 1.0\r\n"	\
+		 "M: recvonly\r\n"		\
+		 "C: 2\r\n"			\
+		 "X\r\n"			\
+		 "L: p:20\r\n"		\
+		 "\r\n"				\
+		 "v=0\r\n"			\
+		 "c=IN IP4 123.12.12.123\r\n"	\
+		 "m=audio 5904 RTP/AVP 18\r\n"	\
+		 "a=rtpmap:18 G729/8000\r\n"	\
+		 "a=rtpmap:97 GSM-EFR/8000\r\n"	\
+		 "a=rtpmap:101 FOO/8000\r\n"	\
+		 "a=ptime:40\r\n"
+
 struct mgcp_test {
 	const char *name;
 	const char *req;
@@ -410,7 +465,7 @@ 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 +553,18 @@ 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 +885,7 @@ 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;
@@ -877,6 +932,72 @@ static void test_packet_error_detection(int patch_ssrc, int patch_ts)
 	force_monotonic_time_us = -1;
 }
 
+static void test_multilple_codec(void)
+{
+	struct mgcp_config *cfg;
+	struct mgcp_endpoint *endp;
+	struct msgb *inp, *resp;
+
+	printf("Testing multiple payload types\n");
+
+	cfg = mgcp_config_alloc();
+	cfg->trunk.number_endpoints = 64;
+	mgcp_endpoints_allocate(&cfg->trunk);
+	cfg->policy_cb = mgcp_test_policy_cb;
+	mgcp_endpoints_allocate(mgcp_trunk_alloc(cfg, 1));
+
+	/* Allocate endpoint 1@mgw with two codecs */
+	last_endpoint = -1;
+	inp = create_msg(CRCX_MULT_1);
+	resp = mgcp_handle_message(cfg, inp);
+	msgb_free(inp);
+	msgb_free(resp);
+
+	OSMO_ASSERT(last_endpoint == 1);
+	endp = &cfg->trunk.endpoints[last_endpoint];
+	OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
+	OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 97);
+
+	/* Allocate 2@mgw with three codecs, last one ignored */
+	last_endpoint = -1;
+	inp = create_msg(CRCX_MULT_2);
+	resp = mgcp_handle_message(cfg, inp);
+	msgb_free(inp);
+	msgb_free(resp);
+
+	OSMO_ASSERT(last_endpoint == 2);
+	endp = &cfg->trunk.endpoints[last_endpoint];
+	OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
+	OSMO_ASSERT(endp->net_end.alt_codec.payload_type == 97);
+
+	/* Allocate 3@mgw with no codecs, check for PT == -1 */
+	last_endpoint = -1;
+	inp = create_msg(CRCX_MULT_3);
+	resp = mgcp_handle_message(cfg, inp);
+	msgb_free(inp);
+	msgb_free(resp);
+
+	OSMO_ASSERT(last_endpoint == 3);
+	endp = &cfg->trunk.endpoints[last_endpoint];
+	OSMO_ASSERT(endp->net_end.codec.payload_type == -1);
+	OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
+
+	/* Allocate 4@mgw with a single codec */
+	last_endpoint = -1;
+	inp = create_msg(CRCX_MULT_4);
+	resp = mgcp_handle_message(cfg, inp);
+	msgb_free(inp);
+	msgb_free(resp);
+
+	OSMO_ASSERT(last_endpoint == 4);
+	endp = &cfg->trunk.endpoints[last_endpoint];
+	OSMO_ASSERT(endp->net_end.codec.payload_type == 18);
+	OSMO_ASSERT(endp->net_end.alt_codec.payload_type == -1);
+
+
+	talloc_free(cfg);
+}
+
 int main(int argc, char **argv)
 {
 	osmo_init_logging(&log_info);
@@ -892,6 +1013,7 @@ int main(int argc, char **argv)
 	test_packet_error_detection(0, 0);
 	test_packet_error_detection(0, 1);
 	test_packet_error_detection(1, 1);
+	test_multilple_codec();
 
 	printf("Done\n");
 	return EXIT_SUCCESS;
diff --git a/openbsc/tests/mgcp/mgcp_test.ok b/openbsc/tests/mgcp/mgcp_test.ok
index 033f783f3..a56a3fd1b 100644
--- a/openbsc/tests/mgcp/mgcp_test.ok
+++ b/openbsc/tests/mgcp/mgcp_test.ok
@@ -474,4 +474,5 @@ Stats: Jitter = 0, Transit = -144000
 In TS: 160320, dTS: 160, Seq: 1002
 Out TS change: 160, dTS: 160, Seq change: 1, TS Err change: in +0, out +0
 Stats: Jitter = 0, Transit = -144000
+Testing multiple payload types
 Done
diff --git a/openbsc/tests/mgcp/mgcp_transcoding_test.c b/openbsc/tests/mgcp/mgcp_transcoding_test.c
index 44f307251..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;
@@ -183,14 +186,14 @@ 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;
 	}
@@ -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};
@@ -463,7 +523,7 @@ 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);
@@ -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