diff --git a/openbsc/include/openbsc/mgcp_internal.h b/openbsc/include/openbsc/mgcp_internal.h
index 570d963..763d83b 100644
--- a/openbsc/include/openbsc/mgcp_internal.h
+++ b/openbsc/include/openbsc/mgcp_internal.h
@@ -22,6 +22,8 @@
 
 #pragma once
 
+#include <string.h>
+
 #include <osmocom/core/select.h>
 
 #define CI_UNUSED 0
@@ -203,8 +205,51 @@
 	} osmux;
 };
 
+#define for_each_line(line, save)			\
+	for (line = strline_r(NULL, &save); line;\
+	     line = strline_r(NULL, &save))
+
+static inline char *strline_r(char *str, char **saveptr)
+{
+	char *result;
+
+	if (str)
+		*saveptr = str;
+
+	result = *saveptr;
+
+	if (*saveptr != NULL) {
+		*saveptr = strpbrk(*saveptr, "\r\n");
+
+		if (*saveptr != NULL) {
+			char *eos = *saveptr;
+
+			if ((*saveptr)[0] == '\r' && (*saveptr)[1] == '\n')
+				(*saveptr)++;
+			(*saveptr)++;
+			if ((*saveptr)[0] == '\0')
+				*saveptr = NULL;
+
+			*eos = '\0';
+		}
+	}
+
+	return result;
+}
+
+
+
 #define ENDPOINT_NUMBER(endp) abs((int)(endp - endp->tcfg->endpoints))
 
+/**
+ * Internal structure while parsing a request
+ */
+struct mgcp_parse_data {
+	struct mgcp_config *cfg;
+	struct mgcp_endpoint *endp;
+	char *trans;
+	char *save;
+	int found;
 };
 
 int mgcp_send_dummy(struct mgcp_endpoint *endp);
@@ -257,5 +302,21 @@
 	MGCP_DEST_BTS,
 };
 
+
 #define MGCP_DUMMY_LOAD 0x23
 
+
+/**
+ * SDP related information
+ */
+/* Assume audio frame length of 20ms */
+#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
+#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
+#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
+#define DEFAULT_RTP_AUDIO_DEFAULT_RATE  8000
+#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1
+
+#define PTYPE_UNDEFINED (-1)
+int mgcp_parse_sdp_data(struct mgcp_rtp_end *rtp, struct mgcp_parse_data *p);
+int mgcp_set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
+			int payload_type, const char *audio_name);
diff --git a/openbsc/src/libmgcp/Makefile.am b/openbsc/src/libmgcp/Makefile.am
index d02b880..4403d60 100644
--- a/openbsc/src/libmgcp/Makefile.am
+++ b/openbsc/src/libmgcp/Makefile.am
@@ -8,7 +8,8 @@
 
 noinst_HEADERS = g711common.h
 
-libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c mgcp_osmux.c
+libmgcp_a_SOURCES = mgcp_protocol.c mgcp_network.c mgcp_vty.c mgcp_osmux.c \
+	mgcp_sdp.c
 
 if BUILD_MGCP_TRANSCODING
     libmgcp_a_SOURCES += mgcp_transcode.c
diff --git a/openbsc/src/libmgcp/mgcp_protocol.c b/openbsc/src/libmgcp/mgcp_protocol.c
index 62f6974..5f3eb3a 100644
--- a/openbsc/src/libmgcp/mgcp_protocol.c
+++ b/openbsc/src/libmgcp/mgcp_protocol.c
@@ -24,7 +24,6 @@
 #include <ctype.h>
 #include <stdio.h>
 #include <stdlib.h>
-#include <string.h>
 #include <time.h>
 #include <limits.h>
 #include <unistd.h>
@@ -41,57 +40,9 @@
 	for (line = strtok_r(NULL, "\r\n", &save); line;\
 	     line = strtok_r(NULL, "\r\n", &save))
 
-#define for_each_line(line, save)			\
-	for (line = strline_r(NULL, &save); line;\
-	     line = strline_r(NULL, &save))
-
-char *strline_r(char *str, char **saveptr)
-{
-	char *result;
-
-	if (str)
-		*saveptr = str;
-
-	result = *saveptr;
-
-	if (*saveptr != NULL) {
-		*saveptr = strpbrk(*saveptr, "\r\n");
-
-		if (*saveptr != NULL) {
-			char *eos = *saveptr;
-
-			if ((*saveptr)[0] == '\r' && (*saveptr)[1] == '\n')
-				(*saveptr)++;
-			(*saveptr)++;
-			if ((*saveptr)[0] == '\0')
-				*saveptr = NULL;
-
-			*eos = '\0';
-		}
-	}
-
-	return result;
-}
-
-/* Assume audio frame length of 20ms */
-#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
-#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
-#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
-#define DEFAULT_RTP_AUDIO_DEFAULT_RATE  8000
-#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1
-
-#define PTYPE_UNDEFINED (-1)
 
 static void mgcp_rtp_end_reset(struct mgcp_rtp_end *end);
 
-struct mgcp_parse_data {
-	struct mgcp_config *cfg;
-	struct mgcp_endpoint *endp;
-	char *trans;
-	char *save;
-	int found;
-};
-
 struct mgcp_request {
 	char *name;
 	struct msgb *(*handle_request) (struct mgcp_parse_data *data);
@@ -599,72 +550,6 @@
 	return ret;
 }
 
-static int set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
-			  int payload_type, const char *audio_name)
-{
-	int rate = codec->rate;
-	int channels = codec->channels;
-	char audio_codec[64];
-
-	talloc_free(codec->subtype_name);
-	codec->subtype_name = NULL;
-	talloc_free(codec->audio_name);
-	codec->audio_name = NULL;
-
-	if (payload_type != PTYPE_UNDEFINED)
-		codec->payload_type = payload_type;
-
-	if (!audio_name) {
-		switch (payload_type) {
-		case 3: audio_name = "GSM/8000/1"; break;
-		case 8: audio_name = "PCMA/8000/1"; break;
-		case 18: audio_name = "G729/8000/1"; break;
-		default:
-			 /* Payload type is unknown, don't change rate and
-			  * channels. */
-			 /* TODO: return value? */
-			 return 0;
-		}
-	}
-
-	if (sscanf(audio_name, "%63[^/]/%d/%d",
-		   audio_codec, &rate, &channels) < 1)
-		return -EINVAL;
-
-	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")) {
-		codec->frame_duration_num = 10;
-		codec->frame_duration_den = 1000;
-	} else {
-		codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
-		codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
-	}
-
-	if (payload_type < 0) {
-		payload_type = 96;
-		if (rate == 8000 && channels == 1) {
-			if (!strcmp(audio_codec, "GSM"))
-				payload_type = 3;
-			else if (!strcmp(audio_codec, "PCMA"))
-				payload_type = 8;
-			else if (!strcmp(audio_codec, "G729"))
-				payload_type = 18;
-		}
-
-		codec->payload_type = payload_type;
-	}
-
-	if (channels != 1)
-		LOGP(DMGCP, LOGL_NOTICE,
-		     "Channels != 1 in SDP: '%s'\n", audio_name);
-
-	return 0;
-}
-
 static int allocate_port(struct mgcp_endpoint *endp, struct mgcp_rtp_end *end,
 			 struct mgcp_port_range *range,
 			 int (*alloc)(struct mgcp_endpoint *endp, int port))
@@ -735,103 +620,6 @@
 	return 0;
 }
 
-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]) {
-		case 'o':
-		case 's':
-		case 't':
-		case 'v':
-			/* skip these SDP attributes */
-			break;
-		case 'a': {
-			int payload;
-			int ptime, ptime2 = 0;
-			char audio_name[64];
-
-			if (audio_payload == -1)
-				break;
-
-			if (sscanf(line, "a=rtpmap:%d %63s",
-				   &payload, audio_name) == 2) {
-				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)
-					rtp->packet_duration_ms = 0;
-				else
-					rtp->packet_duration_ms = ptime;
-			} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
-				/* 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, rc;
-			audio_payload = -1;
-			audio_payload_alt = -1;
-
-			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->codec, audio_payload, NULL);
-				if (rc == 3)
-					set_audio_info(p->cfg, &rtp->alt_codec,
-							audio_payload_alt, NULL);
-			}
-			break;
-		}
-		case 'c': {
-			char ipv4[16];
-
-			if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
-				inet_aton(ipv4, &rtp->addr);
-			}
-			break;
-		}
-		default:
-			if (p->endp)
-				LOGP(DMGCP, LOGL_NOTICE,
-				     "Unhandled SDP option: '%c'/%d on 0x%x\n",
-				     line[0], line[0], ENDPOINT_NUMBER(p->endp));
-			else
-				LOGP(DMGCP, LOGL_NOTICE,
-				     "Unhandled SDP option: '%c'/%d\n",
-				     line[0], line[0]);
-			break;
-		}
-	}
-
-	if (found_media)
-		LOGP(DMGCP, LOGL_NOTICE,
-		     "Got media info via SDP: port %d, payload %d (%s), "
-		     "duration %d, addr %s\n",
-		     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;
-}
-
 /* Set the LCO from a string (see RFC 3435).
  * The string is stored in the 'string' field. A NULL string is handled excatly
  * like an empty string, the 'string' field is never NULL after this function
@@ -1036,13 +824,13 @@
 	endp->allocated = 1;
 
 	/* set up RTP media parameters */
-	set_audio_info(p->cfg, &endp->bts_end.codec, tcfg->audio_payload, tcfg->audio_name);
+	mgcp_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);
+		mgcp_parse_sdp_data(&endp->net_end, p);
 	else if (endp->local_options.codec)
-		set_audio_info(p->cfg, &endp->net_end.codec,
+		mgcp_set_audio_info(p->cfg, &endp->net_end.codec,
 			       PTYPE_UNDEFINED, endp->local_options.codec);
 
 	if (p->cfg->bts_force_ptime) {
@@ -1143,7 +931,7 @@
 		case '\0':
 			/* SDP file begins */
 			have_sdp = 1;
-			parse_sdp_data(&endp->net_end, p);
+			mgcp_parse_sdp_data(&endp->net_end, p);
 			/* This will exhaust p->save, so the loop will
 			 * terminate next time.
 			 */
@@ -1159,7 +947,7 @@
 			     local_options);
 
 	if (!have_sdp && endp->local_options.codec)
-		set_audio_info(p->cfg, &endp->net_end.codec,
+		mgcp_set_audio_info(p->cfg, &endp->net_end.codec,
 			       PTYPE_UNDEFINED, endp->local_options.codec);
 
 	if (setup_rtp_processing(endp) != 0)
diff --git a/openbsc/src/libmgcp/mgcp_sdp.c b/openbsc/src/libmgcp/mgcp_sdp.c
new file mode 100644
index 0000000..a10b3ac
--- /dev/null
+++ b/openbsc/src/libmgcp/mgcp_sdp.c
@@ -0,0 +1,191 @@
+/*
+ * Some SDP file parsing...
+ *
+ * (C) 2009-2015 by Holger Hans Peter Freyther <zecke@selfish.org>
+ * (C) 2009-2014 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 <openbsc/mgcp.h>
+#include <openbsc/mgcp_internal.h>
+
+#include <errno.h>
+
+int mgcp_set_audio_info(void *ctx, struct mgcp_rtp_codec *codec,
+			int payload_type, const char *audio_name)
+{
+	int rate = codec->rate;
+	int channels = codec->channels;
+	char audio_codec[64];
+
+	talloc_free(codec->subtype_name);
+	codec->subtype_name = NULL;
+	talloc_free(codec->audio_name);
+	codec->audio_name = NULL;
+
+	if (payload_type != PTYPE_UNDEFINED)
+		codec->payload_type = payload_type;
+
+	if (!audio_name) {
+		switch (payload_type) {
+		case 3: audio_name = "GSM/8000/1"; break;
+		case 8: audio_name = "PCMA/8000/1"; break;
+		case 18: audio_name = "G729/8000/1"; break;
+		default:
+			 /* Payload type is unknown, don't change rate and
+			  * channels. */
+			 /* TODO: return value? */
+			 return 0;
+		}
+	}
+
+	if (sscanf(audio_name, "%63[^/]/%d/%d",
+		   audio_codec, &rate, &channels) < 1)
+		return -EINVAL;
+
+	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")) {
+		codec->frame_duration_num = 10;
+		codec->frame_duration_den = 1000;
+	} else {
+		codec->frame_duration_num = DEFAULT_RTP_AUDIO_FRAME_DUR_NUM;
+		codec->frame_duration_den = DEFAULT_RTP_AUDIO_FRAME_DUR_DEN;
+	}
+
+	if (payload_type < 0) {
+		payload_type = 96;
+		if (rate == 8000 && channels == 1) {
+			if (!strcmp(audio_codec, "GSM"))
+				payload_type = 3;
+			else if (!strcmp(audio_codec, "PCMA"))
+				payload_type = 8;
+			else if (!strcmp(audio_codec, "G729"))
+				payload_type = 18;
+		}
+
+		codec->payload_type = payload_type;
+	}
+
+	if (channels != 1)
+		LOGP(DMGCP, LOGL_NOTICE,
+		     "Channels != 1 in SDP: '%s'\n", audio_name);
+
+	return 0;
+}
+
+
+int mgcp_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]) {
+		case 'o':
+		case 's':
+		case 't':
+		case 'v':
+			/* skip these SDP attributes */
+			break;
+		case 'a': {
+			int payload;
+			int ptime, ptime2 = 0;
+			char audio_name[64];
+
+			if (audio_payload == -1)
+				break;
+
+			if (sscanf(line, "a=rtpmap:%d %63s",
+				   &payload, audio_name) == 2) {
+				if (payload == audio_payload)
+					mgcp_set_audio_info(p->cfg, &rtp->codec,
+							payload, audio_name);
+				else if (payload == audio_payload_alt)
+					mgcp_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)
+					rtp->packet_duration_ms = 0;
+				else
+					rtp->packet_duration_ms = ptime;
+			} else if (sscanf(line, "a=maxptime:%d", &ptime2) == 1) {
+				/* 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, rc;
+			audio_payload = -1;
+			audio_payload_alt = -1;
+
+			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;
+				mgcp_set_audio_info(p->cfg, &rtp->codec, audio_payload, NULL);
+				if (rc == 3)
+					mgcp_set_audio_info(p->cfg, &rtp->alt_codec,
+							audio_payload_alt, NULL);
+			}
+			break;
+		}
+		case 'c': {
+			char ipv4[16];
+
+			if (sscanf(line, "c=IN IP4 %15s", ipv4) == 1) {
+				inet_aton(ipv4, &rtp->addr);
+			}
+			break;
+		}
+		default:
+			if (p->endp)
+				LOGP(DMGCP, LOGL_NOTICE,
+				     "Unhandled SDP option: '%c'/%d on 0x%x\n",
+				     line[0], line[0], ENDPOINT_NUMBER(p->endp));
+			else
+				LOGP(DMGCP, LOGL_NOTICE,
+				     "Unhandled SDP option: '%c'/%d\n",
+				     line[0], line[0]);
+			break;
+		}
+	}
+
+	if (found_media)
+		LOGP(DMGCP, LOGL_NOTICE,
+		     "Got media info via SDP: port %d, payload %d (%s), "
+		     "duration %d, addr %s\n",
+		     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;
+}
+