diff --git a/daemon/.ycm_extra_conf.py b/daemon/.ycm_extra_conf.py
index cf64ff7da..aefe9b1ba 100644
--- a/daemon/.ycm_extra_conf.py
+++ b/daemon/.ycm_extra_conf.py
@@ -20,6 +20,7 @@ flags = [
     '-I/usr/lib/x86_64-linux-gnu/glib-2.0/include',
     '-pthread',
     '-I../kernel-module/',
+    '-I../lib/',
     '-D_GNU_SOURCE',
     '-D__DEBUG=1',
     '-D__YCM=1',
diff --git a/daemon/Makefile b/daemon/Makefile
index 21c5870c0..f81d1a2a0 100644
--- a/daemon/Makefile
+++ b/daemon/Makefile
@@ -55,7 +55,8 @@ include ../lib/lib.Makefile
 SRCS=		main.c kernel.c poller.c aux.c control_tcp.c streambuf.c call.c control_udp.c redis.c \
 		bencode.c cookie_cache.c udp_listener.c control_ng.c sdp.c str.c stun.c rtcp.c \
 		crypto.c rtp.c call_interfaces.c dtls.c log.c cli.c graphite.c ice.c socket.c \
-		media_socket.c homer.c recording.c statistics.c cdr.c ssrc.c iptables.c tcp_listener.c
+		media_socket.c homer.c recording.c statistics.c cdr.c ssrc.c iptables.c tcp_listener.c \
+		codec.c
 LIBSRCS=	loglib.c auxlib.c rtplib.c
 OBJS=		$(SRCS:.c=.o) $(LIBSRCS:.c=.o)
 
diff --git a/daemon/aux.h b/daemon/aux.h
index 4feb1250b..ea82a4b9b 100644
--- a/daemon/aux.h
+++ b/daemon/aux.h
@@ -158,6 +158,19 @@ INLINE void g_tree_add_all(GTree *t, GQueue *q) {
 
 
 
+/* GHASHTABLE */
+
+INLINE GQueue *g_hash_table_lookup_queue_new(GHashTable *ht, void *key) {
+	GQueue *ret = g_hash_table_lookup(ht, key);
+	if (ret)
+		return ret;
+	ret = g_queue_new();
+	g_hash_table_insert(ht, key, ret);
+	return ret;
+}
+
+
+
 /*** STRING HELPERS ***/
 
 INLINE void strmove(char **d, char **s) {
diff --git a/daemon/call.c b/daemon/call.c
index 553970cf7..486b16606 100644
--- a/daemon/call.c
+++ b/daemon/call.c
@@ -43,6 +43,7 @@
 #include "ssrc.h"
 #include "main.h"
 #include "graphite.h"
+#include "codec.h"
 
 
 /* also serves as array index for callstream->peers[] */
@@ -683,6 +684,7 @@ static struct call_media *__get_media(struct call_monologue *ml, GList **it, con
 	med->index = sp->index;
 	call_str_cpy(ml->call, &med->type, &sp->type);
 	med->codecs = g_hash_table_new_full(g_int_hash, g_int_equal, NULL, __payload_type_free);
+	med->codec_names = g_hash_table_new_full(str_hash, str_equal, NULL, (void (*)(void*)) g_queue_free);
 
 	g_queue_push_tail(&ml->medias, med);
 
@@ -1448,27 +1450,57 @@ static void __dtls_logic(const struct sdp_ng_flags *flags,
 		MEDIA_SET(other_media, DTLS);
 }
 
-static void __rtp_payload_type_add(struct call_media *media, struct rtp_payload_type *pt) {
+static void __rtp_payload_type_add(struct call_media *media, struct call_media *other_media,
+		struct rtp_payload_type *pt)
+{
 	struct call *call = media->call;
+
+	if (g_hash_table_lookup(media->codecs, &pt->payload_type)) {
+		// collision/duplicate - ignore
+		__payload_type_free(pt);
+		return;
+	}
 	/* we must duplicate the contents */
 	call_str_cpy(call, &pt->encoding_with_params, &pt->encoding_with_params);
 	call_str_cpy(call, &pt->encoding, &pt->encoding);
 	call_str_cpy(call, &pt->encoding_parameters, &pt->encoding_parameters);
 	call_str_cpy(call, &pt->format_parameters, &pt->format_parameters);
 	g_hash_table_replace(media->codecs, &pt->payload_type, pt);
-	g_queue_push_tail(&media->codecs_prefs, pt);
+
+	GQueue *q = g_hash_table_lookup_queue_new(media->codec_names, &pt->encoding);
+	g_queue_push_tail(q, GUINT_TO_POINTER(pt->payload_type));
+
+	g_queue_push_tail(&media->codecs_prefs_recv, pt);
+
+	// for the other side, we need a new 'pt' struct
+	struct rtp_payload_type *pt_copy = g_slice_alloc(sizeof(*pt));
+	*pt_copy = *pt; // contents are allocated from the 'call'
+	g_queue_push_tail(&other_media->codecs_prefs_send, pt_copy);
+
+	// make sure we have at least an empty queue here to indicate support for this code.
+	// don't add anything to the queue as we don't know the reverse RTP payload type.
+	g_hash_table_lookup_queue_new(other_media->codec_names, &pt->encoding);
 }
 
-static void __rtp_payload_types(struct call_media *media, GQueue *types, GHashTable *strip,
+static void __payload_queue_free(void *qq) {
+	GQueue *q = qq;
+	g_queue_free_full(q, __payload_type_free);
+}
+static void __rtp_payload_types(struct call_media *media, struct call_media *other_media,
+		GQueue *types, GHashTable *strip,
 		const GQueue *offer)
 {
+	// 'media' = receiver of this offer/answer; 'other_media' = sender of this offer/answer
 	struct rtp_payload_type *pt;
 	static const str str_all = STR_CONST_INIT("all");
-	GHashTable *removed = g_hash_table_new_full(str_hash, str_equal, NULL, __payload_type_free);
+	GHashTable *removed = g_hash_table_new_full(str_hash, str_equal, NULL, __payload_queue_free);
 	int remove_all = 0;
 
 	// start fresh
-	g_queue_clear(&media->codecs_prefs);
+	g_queue_clear(&media->codecs_prefs_recv);
+	g_queue_clear_full(&other_media->codecs_prefs_send, __payload_type_free);
+	g_hash_table_remove_all(media->codecs);
+	g_hash_table_remove_all(media->codec_names);
 
 	if (strip && g_hash_table_lookup(strip, &str_all))
 		remove_all = 1;
@@ -1478,22 +1510,27 @@ static void __rtp_payload_types(struct call_media *media, GQueue *types, GHashTa
 		// codec stripping
 		if (strip) {
 			if (remove_all || g_hash_table_lookup(strip, &pt->encoding)) {
-				g_hash_table_replace(removed, &pt->encoding, pt);
+				GQueue *q = g_hash_table_lookup_queue_new(removed, &pt->encoding);
+				g_queue_push_tail(q, pt);
 				continue;
 			}
 		}
-		__rtp_payload_type_add(media, pt);
+		__rtp_payload_type_add(media, other_media, pt);
 	}
 
 	if (offer) {
 		// now restore codecs that have been removed, but should be offered
 		for (GList *l = offer->head; l; l = l->next) {
 			str *codec = l->data;
-			pt = g_hash_table_lookup(removed, codec);
-			if (!pt)
+			GQueue *q = g_hash_table_lookup(removed, codec);
+			if (!q)
 				continue;
 			g_hash_table_steal(removed, codec);
-			__rtp_payload_type_add(media, pt);
+			for (GList *l = q->head; l; l = l->next) {
+				pt = l->data;
+				__rtp_payload_type_add(media, other_media, pt);
+			}
+			g_queue_free(q);
 		}
 	}
 
@@ -1620,7 +1657,10 @@ int monologue_offer_answer(struct call_monologue *other_ml, GQueue *streams,
 				MEDIA_SET(other_media, SDES);
 		}
 
-		__rtp_payload_types(media, &sp->rtp_payload_types, flags->codec_strip, &flags->codec_offer);
+		// codec and RTP payload types handling
+		__rtp_payload_types(media, other_media, &sp->rtp_payload_types,
+				flags->codec_strip, &flags->codec_offer);
+		codec_handlers_update(media, other_media);
 
 		/* send and recv are from our POV */
 		bf_copy_same(&media->media_flags, &sp->sp_flags,
@@ -2030,7 +2070,10 @@ static void __call_free(void *p) {
 		g_queue_clear(&md->streams);
 		g_queue_clear(&md->endpoint_maps);
 		g_hash_table_destroy(md->codecs);
-		g_queue_clear(&md->codecs_prefs);
+		g_hash_table_destroy(md->codec_names);
+		g_queue_clear(&md->codecs_prefs_recv);
+		g_queue_clear_full(&md->codecs_prefs_send, __payload_type_free);
+		codec_handlers_free(md);
 		g_slice_free1(sizeof(*md), md);
 	}
 
diff --git a/daemon/call.h b/daemon/call.h
index f6d88fd86..4bddf6c2c 100644
--- a/daemon/call.h
+++ b/daemon/call.h
@@ -199,6 +199,7 @@ struct local_interface;
 struct call_monologue;
 struct ice_agent;
 struct ssrc_hash;
+struct codec_handler;
 
 
 typedef bencode_buffer_t call_buffer_t;
@@ -323,8 +324,13 @@ struct call_media {
 	GQueue			streams; /* normally RTP + RTCP */
 	GQueue			endpoint_maps;
 
-	GHashTable		*codecs;
-	GQueue			codecs_prefs;
+	GHashTable		*codecs; // int payload type -> struct rtp_payload_type; storage container
+	GHashTable		*codec_names; // codec name -> GQueue of int payload types; storage container
+	GQueue			codecs_prefs_recv, // preference by order in SDP; values shared with 'codecs'
+				codecs_prefs_send; // ditto for outgoing media; storage container
+	GHashTable		*codec_handlers; // int payload type -> struct codec_handler
+						// XXX combine this with 'codecs' hash table?
+	volatile struct codec_handler *codec_handler_cache;
 
 	volatile unsigned int	media_flags;
 };
diff --git a/daemon/codec.c b/daemon/codec.c
new file mode 100644
index 000000000..9c2ebc7f1
--- /dev/null
+++ b/daemon/codec.c
@@ -0,0 +1,130 @@
+#include "codec.h"
+#include <glib.h>
+#include "call.h"
+#include "log.h"
+#include "rtplib.h"
+
+
+
+
+static codec_handler_func handler_func_stub;
+
+
+static struct codec_handler codec_handler_stub = {
+	.rtp_payload_type = -1,
+	.func = handler_func_stub,
+};
+
+
+
+static void __make_stub(struct codec_handler *handler) {
+	handler->func = handler_func_stub;
+}
+
+static void __codec_handler_free(void *pp) {
+	struct codec_handler *h = pp;
+	g_slice_free1(sizeof(*h), h);
+}
+
+// call must be locked in W
+void codec_handlers_update(struct call_media *receiver, struct call_media *sink) {
+	if (!receiver->codec_handlers)
+		receiver->codec_handlers = g_hash_table_new_full(g_int_hash, g_int_equal,
+				NULL, __codec_handler_free);
+
+	// we go through the list of codecs that the receiver supports and compare it
+	// with the list of codecs supported by the sink. if the receiver supports
+	// a codec that the sink doesn't support, we must transcode.
+	//
+	// if we transcode, we transcode to the highest-preference supported codec
+	// that the sink specified. determine this first.
+	struct rtp_payload_type *pref_dest_codec = NULL;
+	for (GList *l = sink->codecs_prefs_send.head; l; l = l->next) {
+		struct rtp_payload_type *pt = l->data;
+		// XXX if supported ...
+		ilog(LOG_DEBUG, "Default sink codec is " STR_FORMAT, STR_FMT(&pt->encoding));
+		pref_dest_codec = pt;
+		break;
+	}
+
+	for (GList *l = receiver->codecs_prefs_recv.head; l; l = l->next) {
+		struct rtp_payload_type *pt = l->data;
+
+		// first, make sure we have a codec_handler struct for this
+		struct codec_handler *handler;
+		handler = g_hash_table_lookup(receiver->codec_handlers, &pt->payload_type);
+		if (!handler) {
+			ilog(LOG_DEBUG, "Creating codec handler for " STR_FORMAT, STR_FMT(&pt->encoding));
+			handler = g_slice_alloc0(sizeof(*handler));
+			handler->rtp_payload_type = pt->payload_type;
+			g_hash_table_insert(receiver->codec_handlers, &handler->rtp_payload_type,
+					handler);
+		}
+
+		// if the sink's codec preferences are unknown (empty), or there are
+		// no supported codecs to transcode to, then we have nothing
+		// to do. most likely this is an initial offer without a received answer.
+		// we default to forwarding without transcoding.
+		if (!pref_dest_codec) {
+			ilog(LOG_DEBUG, "No known/supported sink codec for " STR_FORMAT, STR_FMT(&pt->encoding));
+			__make_stub(handler);
+			continue;
+		}
+
+		if (g_hash_table_lookup(sink->codec_names, &pt->encoding)) {
+			// the sink supports this codec. forward without transcoding.
+			ilog(LOG_DEBUG, "Sink supports codec " STR_FORMAT, STR_FMT(&pt->encoding));
+			__make_stub(handler);
+			continue;
+		}
+
+		// the sink does not support this codec XXX do something
+		ilog(LOG_DEBUG, "Sink does not support codec " STR_FORMAT, STR_FMT(&pt->encoding));
+		__make_stub(handler);
+	}
+}
+
+// call must be locked in R
+struct codec_handler *codec_handler_get(struct call_media *m, int payload_type) {
+	struct codec_handler *h;
+
+	if (payload_type < 0)
+		goto out;
+
+	h = g_atomic_pointer_get(&m->codec_handler_cache);
+	if (G_LIKELY(G_LIKELY(h) && G_LIKELY(h->rtp_payload_type == payload_type)))
+		return h;
+
+	h = g_hash_table_lookup(m->codec_handlers, &payload_type);
+	if (!h)
+		goto out;
+
+	g_atomic_pointer_set(&m->codec_handler_cache, h);
+
+	return h;
+
+out:
+	return &codec_handler_stub;
+}
+
+void codec_handlers_free(struct call_media *m) {
+	g_hash_table_destroy(m->codec_handlers);
+	m->codec_handlers = NULL;
+	m->codec_handler_cache = NULL;
+}
+
+
+static int handler_func_stub(struct codec_handler *h, struct call_media *media, const str *s, GQueue *out) {
+	struct codec_packet *p = g_slice_alloc(sizeof(*p));
+	p->s = *s;
+	p->free_func = NULL;
+	g_queue_push_tail(out, p);
+	return 0;
+}
+
+void codec_packet_free(void *pp) {
+	struct codec_packet *p = pp;
+	if (p->free_func)
+		p->free_func(p->s.s);
+	g_slice_free1(sizeof(*p), p);
+}
diff --git a/daemon/codec.h b/daemon/codec.h
new file mode 100644
index 000000000..83beeb66c
--- /dev/null
+++ b/daemon/codec.h
@@ -0,0 +1,38 @@
+#ifndef __CODEC_H__
+#define __CODEC_H__
+
+
+#include <glib.h>
+#include "str.h"
+
+
+struct call_media;
+struct codec_handler;
+
+
+typedef int codec_handler_func(struct codec_handler *, struct call_media *, const str *, GQueue *);
+
+
+struct codec_handler {
+	int rtp_payload_type;
+	codec_handler_func *func;
+};
+
+struct codec_packet {
+	str s;
+	void (*free_func)(void *);
+};
+
+
+void codec_handlers_update(struct call_media *receiver, struct call_media *sink);
+
+struct codec_handler *codec_handler_get(struct call_media *, int payload_type);
+
+void codec_handlers_free(struct call_media *);
+
+void codec_packet_free(void *);
+
+
+
+
+#endif
diff --git a/daemon/media_socket.c b/daemon/media_socket.c
index ef57a2820..02b7580dd 100644
--- a/daemon/media_socket.c
+++ b/daemon/media_socket.c
@@ -24,6 +24,7 @@
 #include "ssrc.h"
 #include "iptables.h"
 #include "main.h"
+#include "codec.h"
 
 
 #ifndef PORT_RANDOM_MIN
@@ -68,6 +69,7 @@ struct packet_handler_ctx {
 	struct packet_stream *sink; // where to send output packets to (forward destination)
 	rewrite_func decrypt_func, encrypt_func; // handlers for decrypt/encrypt
 	struct packet_stream *in_srtp, *out_srtp; // SRTP contexts for decrypt/encrypt (relevant for muxed RTCP)
+	int payload_type; // -1 if unknown or not RTP
 	struct ssrc_ctx *ssrc_in, *ssrc_out; // SSRC contexts from in_srtp and out_srtp
 	int rtcp; // true if this is an RTCP packet
 
@@ -75,6 +77,9 @@ struct packet_handler_ctx {
 	int update; // true if Redis info needs to be updated
 	int unkernelize; // true if stream ought to be removed from kernel
 	int kernelize; // true if stream can be kernelized
+
+	// output:
+	GQueue packets_out;
 };
 
 
@@ -1281,6 +1286,8 @@ static void media_packet_rtp(struct packet_handler_ctx *phc)
 	struct rtp_header *rtp_h;
 	struct rtcp_packet *rtcp_h;
 
+	phc->payload_type = -1;
+
 	if (G_UNLIKELY(!phc->media->protocol))
 		return;
 	if (G_UNLIKELY(!phc->media->protocol->rtp))
@@ -1292,15 +1299,16 @@ static void media_packet_rtp(struct packet_handler_ctx *phc)
 					&phc->ssrc_out, phc->call->ssrc_hash);
 
 		// check the payload type
-		int i = (rtp_h->m_pt & 0x7f);
+		// XXX redundant between SSRC handling and codec_handler stuff -> combine
+		phc->payload_type = (rtp_h->m_pt & 0x7f);
 		if (G_LIKELY(phc->ssrc_in))
-			phc->ssrc_in->parent->payload_type = i;
+			phc->ssrc_in->parent->payload_type = phc->payload_type;
 
 		// XXX convert to array? or keep last pointer?
-		struct rtp_stats *rtp_s = g_hash_table_lookup(phc->stream->rtp_stats, &i);
+		struct rtp_stats *rtp_s = g_hash_table_lookup(phc->stream->rtp_stats, &phc->payload_type);
 		if (!rtp_s) {
 			ilog(LOG_WARNING | LOG_FLAG_LIMIT,
-					"RTP packet with unknown payload type %u received", i);
+					"RTP packet with unknown payload type %u received", phc->payload_type);
 			atomic64_inc(&phc->stream->stats.errors);
 			atomic64_inc(&rtpe_statsps.errors);
 		}
@@ -1348,21 +1356,25 @@ static int media_packet_decrypt(struct packet_handler_ctx *phc)
 	return ret;
 }
 
-static int media_packet_encrypt(struct packet_handler_ctx *phc)
-{
+static int media_packet_encrypt(struct packet_handler_ctx *phc) {
+	int ret = 0;
+
 	if (!phc->encrypt_func)
 		return 0;
 
 	mutex_lock(&phc->out_srtp->out_lock);
 
-	int ret = phc->encrypt_func(&phc->s, phc->out_srtp, NULL, NULL, NULL, phc->ssrc_out);
+	for (GList *l = phc->packets_out.head; l; l = l->next) {
+		struct codec_packet *p = l->data;
+		int encret = phc->encrypt_func(&p->s, phc->out_srtp, NULL, NULL, NULL, phc->ssrc_out);
+		if (encret == 1)
+			phc->update = 1;
+		else if (encret != 0)
+			ret = -1;
+	}
 
 	mutex_unlock(&phc->out_srtp->out_lock);
 
-	if (ret == 1) {
-		phc->update = 1;
-		ret = 0;
-	}
 	return ret;
 }
 
@@ -1558,7 +1570,7 @@ static int stream_packet(struct packet_handler_ctx *phc) {
 	// this sets rtcp, in_srtp, out_srtp, and sink
 	media_packet_rtcp_demux(phc);
 
-	// this set ssrc_in and ssrc_out
+	// this set payload_type, ssrc_in and ssrc_out
 	media_packet_rtp(phc);
 
 
@@ -1580,6 +1592,11 @@ static int stream_packet(struct packet_handler_ctx *phc) {
 	if (phc->call->recording)
 		dump_packet(phc->call->recording, phc->stream, &phc->s);
 
+	// XXX use a handler for RTCP
+	struct codec_handler *transcoder = codec_handler_get(phc->media, phc->payload_type);
+	// this transfers the packet from 's' to 'packets_out'
+	transcoder->func(transcoder, phc->media, &phc->s, &phc->packets_out);
+
 	if (G_LIKELY(handler_ret >= 0))
 		handler_ret = media_packet_encrypt(phc);
 
@@ -1606,9 +1623,19 @@ static int stream_packet(struct packet_handler_ctx *phc) {
 		goto drop;
 	}
 
-	ret = socket_sendto(&phc->sink->selected_sfd->socket, phc->s.s, phc->s.len, &phc->sink->endpoint);
-	__C_DBG("Forward to sink endpoint: %s:%d", sockaddr_print_buf(&phc->sink->endpoint.address),
-			phc->sink->endpoint.port);
+	struct codec_packet *p;
+	ret = 0;
+	while ((p = g_queue_pop_head(&phc->packets_out))) {
+		__C_DBG("Forward to sink endpoint: %s:%d", sockaddr_print_buf(&phc->sink->endpoint.address),
+				phc->sink->endpoint.port);
+
+		ret = socket_sendto(&phc->sink->selected_sfd->socket, p->s.s, p->s.len, &phc->sink->endpoint);
+
+		codec_packet_free(p);
+
+		if (ret == -1)
+			break;
+	}
 
 	mutex_unlock(&phc->sink->out_lock);
 
@@ -1622,6 +1649,7 @@ static int stream_packet(struct packet_handler_ctx *phc) {
 
 drop:
 	ret = 0;
+	// XXX separate stats for received/sent
 	atomic64_inc(&phc->stream->stats.packets);
 	atomic64_add(&phc->stream->stats.bytes, phc->s.len);
 	atomic64_set(&phc->stream->last_packet, rtpe_now.tv_sec);
@@ -1637,6 +1665,8 @@ out:
 
 	rwlock_unlock_r(&phc->call->master_lock);
 
+	g_queue_clear_full(&phc->packets_out, codec_packet_free);
+
 	return ret;
 }
 
@@ -1662,7 +1692,8 @@ static void stream_fd_readable(int fd, void *p, uintptr_t u) {
 		}
 #endif
 
-		struct packet_handler_ctx phc = { 0, };
+		struct packet_handler_ctx phc;
+		ZERO(phc);
 		phc.sfd = sfd;
 
 		ret = socket_recvfrom_ts(&sfd->socket, buf + RTP_BUFFER_HEAD_ROOM, MAX_RTP_PACKET_SIZE,
diff --git a/daemon/rtp.c b/daemon/rtp.c
index 4ae11db16..4d90547df 100644
--- a/daemon/rtp.c
+++ b/daemon/rtp.c
@@ -91,7 +91,6 @@ int rtp_avp2savp(str *s, struct crypto_context *c, struct ssrc_ctx *ssrc_ctx) {
 	struct rtp_header *rtp;
 	str payload, to_auth;
 	u_int64_t index;
-	int ret = 0;
 
 	if (G_UNLIKELY(!ssrc_ctx))
 		return -1;
@@ -115,7 +114,7 @@ int rtp_avp2savp(str *s, struct crypto_context *c, struct ssrc_ctx *ssrc_ctx) {
 		s->len += c->params.crypto_suite->srtp_auth_tag;
 	}
 
-	return ret;
+	return 0;
 }
 
 /* rfc 3711, section 3.3 */
@@ -124,7 +123,6 @@ int rtp_savp2avp(str *s, struct crypto_context *c, struct ssrc_ctx *ssrc_ctx) {
 	u_int64_t index;
 	str payload, to_auth, to_decrypt, auth_tag;
 	char hmac[20];
-	int ret = 0;
 
 	if (G_UNLIKELY(!ssrc_ctx))
 		return -1;
@@ -176,7 +174,7 @@ decrypt:
 
 	*s = to_auth;
 
-	return ret;
+	return 0;
 
 error:
 	ilog(LOG_WARNING | LOG_FLAG_LIMIT, "Discarded invalid SRTP packet: authentication failed");
diff --git a/daemon/sdp.c b/daemon/sdp.c
index 540e2ae46..9cde6bccb 100644
--- a/daemon/sdp.c
+++ b/daemon/sdp.c
@@ -1003,15 +1003,9 @@ int sdp_parse(str *body, GQueue *sessions) {
 				/* if (attr->key.s)
 					g_hash_table_insert(attrs->name_hash, &attr->key, attr); */
 
-				/* attr_queue = g_hash_table_lookup(attrs->name_lists_hash, &attr->name);
-				if (!attr_queue)
-					g_hash_table_insert(attrs->name_lists_hash, &attr->name,
-							(attr_queue = g_queue_new()));
+				/* attr_queue = g_hash_table_lookup_queue_new(attrs->name_lists_hash, &attr->name);
 				g_queue_push_tail(attr_queue, attr); */
-				attr_queue = g_hash_table_lookup(attrs->id_lists_hash, &attr->attr);
-				if (!attr_queue)
-					g_hash_table_insert(attrs->id_lists_hash, &attr->attr,
-							(attr_queue = g_queue_new()));
+				attr_queue = g_hash_table_lookup_queue_new(attrs->id_lists_hash, &attr->attr);
 				g_queue_push_tail(attr_queue, attr);
 
 				break;
@@ -1470,10 +1464,10 @@ static int replace_transport_protocol(struct sdp_chopper *chop,
 static int replace_codec_list(struct sdp_chopper *chop,
 		struct sdp_media *media, struct call_media *cm)
 {
-	if (cm->codecs_prefs.length == 0)
+	if (cm->codecs_prefs_recv.length == 0)
 		return 0; // legacy protocol or usage error
 
-	for (GList *l = cm->codecs_prefs.head; l; l = l->next) {
+	for (GList *l = cm->codecs_prefs_recv.head; l; l = l->next) {
 		struct rtp_payload_type *pt = l->data;
 		chopper_append_printf(chop, " %u", pt->payload_type);
 	}
@@ -1718,7 +1712,7 @@ static int process_media_attributes(struct sdp_chopper *chop, struct sdp_media *
 				break;
 
 			case ATTR_RTPMAP:
-				if (media->codecs_prefs.length == 0)
+				if (media->codecs_prefs_recv.length == 0)
 					break; // legacy protocol or usage error
 				if (!g_hash_table_lookup(media->codecs,
 							&attr->u.rtpmap.rtp_pt.payload_type))
@@ -1726,7 +1720,7 @@ static int process_media_attributes(struct sdp_chopper *chop, struct sdp_media *
 				break;
 
 			case ATTR_FMTP:
-				if (media->codecs_prefs.length == 0)
+				if (media->codecs_prefs_recv.length == 0)
 					break; // legacy protocol or usage error
 				if (!g_hash_table_lookup(media->codecs,
 							&attr->u.fmtp.payload_type))