diff --git a/src/libmsc/rtp_stream.c b/src/libmsc/rtp_stream.c
new file mode 100644
index 0000000..6163a98
--- /dev/null
+++ b/src/libmsc/rtp_stream.c
@@ -0,0 +1,389 @@
+/*
+ * (C) 2019 by sysmocom - s.m.f.c. GmbH <info@sysmocom.de>
+ * All Rights Reserved
+ *
+ * SPDX-License-Identifier: AGPL-3.0+
+ *
+ * Author: Neels Hofmeyr
+ *
+ * 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 <osmocom/core/fsm.h>
+
+#include <osmocom/mgcp_client/mgcp_client_endpoint_fsm.h>
+
+#include <osmocom/msc/debug.h>
+#include <osmocom/msc/transaction.h>
+#include <osmocom/msc/call_leg.h>
+#include <osmocom/msc/rtp_stream.h>
+
+#define LOG_RTPS(rtps, level, fmt, args...) \
+	LOGPFSML(rtps->fi, level, fmt, ##args)
+
+enum rtp_stream_event {
+	RTP_STREAM_EV_CRCX_OK,
+	RTP_STREAM_EV_CRCX_FAIL,
+	RTP_STREAM_EV_MDCX_OK,
+	RTP_STREAM_EV_MDCX_FAIL,
+};
+
+enum rtp_stream_state {
+	RTP_STREAM_ST_UNINITIALIZED,
+	RTP_STREAM_ST_ESTABLISHING,
+	RTP_STREAM_ST_ESTABLISHED,
+	RTP_STREAM_ST_DISCARDING,
+};
+
+static struct osmo_fsm rtp_stream_fsm;
+
+static struct osmo_tdef_state_timeout rtp_stream_fsm_timeouts[32] = {
+	[RTP_STREAM_ST_ESTABLISHING] = { .T = -2 },
+};
+
+#define rtp_stream_state_chg(rtps, state) \
+	osmo_tdef_fsm_inst_state_chg((rtps)->fi, state, rtp_stream_fsm_timeouts, g_mgw_tdefs, 5)
+
+static __attribute__((constructor)) void rtp_stream_init()
+{
+	OSMO_ASSERT(osmo_fsm_register(&rtp_stream_fsm) == 0);
+}
+
+void rtp_stream_update_id(struct rtp_stream *rtps)
+{
+	char buf[256];
+	char *p;
+	struct osmo_strbuf sb = { .buf = buf, .len = sizeof(buf) };
+	OSMO_STRBUF_PRINTF(sb, "%s", rtps->fi->proc.parent->id);
+	if (rtps->for_trans)
+		OSMO_STRBUF_PRINTF(sb, ":trans-%u", rtps->for_trans->transaction_id);
+	OSMO_STRBUF_PRINTF(sb, ":call-%u", rtps->call_id);
+	OSMO_STRBUF_PRINTF(sb, ":%s", rtp_direction_name(rtps->dir));
+	if (!osmo_mgcpc_ep_ci_id(rtps->ci)) {
+		OSMO_STRBUF_PRINTF(sb, ":no-CI");
+	} else {
+		OSMO_STRBUF_PRINTF(sb, ":CI-%s", osmo_mgcpc_ep_ci_id(rtps->ci));
+		if (!osmo_sockaddr_str_is_set(&rtps->remote))
+			OSMO_STRBUF_PRINTF(sb, ":no-remote-port");
+		else if (!rtps->remote_sent_to_mgw)
+			OSMO_STRBUF_PRINTF(sb, ":remote-port-not-sent");
+		if (!rtps->codec_known)
+			OSMO_STRBUF_PRINTF(sb, ":no-codec");
+		else if (!rtps->codec_sent_to_mgw)
+			OSMO_STRBUF_PRINTF(sb, ":codec-not-sent");
+	}
+	if (osmo_sockaddr_str_is_set(&rtps->local))
+		OSMO_STRBUF_PRINTF(sb, ":local-%s-%u", rtps->local.ip, rtps->local.port);
+	if (osmo_sockaddr_str_is_set(&rtps->remote))
+		OSMO_STRBUF_PRINTF(sb, ":remote-%s-%u", rtps->remote.ip, rtps->remote.port);
+
+	/* Replace any dots in the IP address, dots not allowed as FSM instance name */
+	for (p = buf; *p; p++)
+		if (*p == '.')
+			*p = '-';
+
+	osmo_fsm_inst_update_id_f(rtps->fi, "%s", buf);
+}
+
+/* Allocate RTP stream under a call leg. This is one RTP connection from some remote entity with address and port to a
+ * local RTP address and port. call_id is stored for sending in MGCP transactions and as logging context. for_trans is
+ * optional, merely stored for reference by callers, and appears as log context if not NULL. */
+struct rtp_stream *rtp_stream_alloc(struct call_leg *parent_call_leg, enum rtp_direction dir,
+				    uint32_t call_id, struct gsm_trans *for_trans)
+{
+	struct osmo_fsm_inst *fi;
+	struct rtp_stream *rtps;
+
+	fi = osmo_fsm_inst_alloc_child(&rtp_stream_fsm, parent_call_leg->fi, CALL_LEG_EV_RTP_STREAM_GONE);
+	OSMO_ASSERT(fi);
+
+	rtps = talloc(fi, struct rtp_stream);
+	OSMO_ASSERT(rtps);
+	fi->priv = rtps;
+	*rtps = (struct rtp_stream){
+		.fi = fi,
+		.parent_call_leg = parent_call_leg,
+		.call_id = call_id,
+		.for_trans = for_trans,
+		.dir = dir,
+	};
+
+	rtp_stream_update_id(rtps);
+
+	return rtps;
+}
+
+static void check_established(struct rtp_stream *rtps)
+{
+	if (rtps->fi->state != RTP_STREAM_ST_ESTABLISHED
+	    && osmo_sockaddr_str_is_set(&rtps->local)
+	    && osmo_sockaddr_str_is_set(&rtps->remote)
+	    && rtps->remote_sent_to_mgw
+	    && rtps->codec_known)
+		rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHED);
+}
+
+static void rtp_stream_fsm_establishing_established(struct osmo_fsm_inst *fi, uint32_t event, void *data)
+{
+	struct rtp_stream *rtps = fi->priv;
+	const struct mgcp_conn_peer *crcx_info;
+	switch (event) {
+	case RTP_STREAM_EV_CRCX_OK:
+		crcx_info = osmo_mgcpc_ep_ci_get_rtp_info(rtps->ci);
+		osmo_sockaddr_str_from_str(&rtps->local, crcx_info->addr, crcx_info->port);
+		rtp_stream_update_id(rtps);
+		osmo_fsm_inst_dispatch(fi->proc.parent, CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE, rtps);
+		check_established(rtps);
+
+		if ((!rtps->remote_sent_to_mgw || !rtps->codec_sent_to_mgw)
+		    && osmo_sockaddr_str_is_set(&rtps->remote)
+		    && rtps->codec_known) {
+			LOG_RTPS(rtps, LOGL_DEBUG,
+				 "local ip:port set;%s%s triggering MDCX to send the new settings\n",
+				 (!rtps->remote_sent_to_mgw)? " remote ip:port not yet sent," : "",
+				 (!rtps->codec_sent_to_mgw)? " codec not yet sent," : "");
+			rtp_stream_do_mdcx(rtps);
+		}
+		return;
+
+	case RTP_STREAM_EV_MDCX_OK:
+		rtp_stream_update_id(rtps);
+		check_established(rtps);
+		return;
+
+	case RTP_STREAM_EV_CRCX_FAIL:
+	case RTP_STREAM_EV_MDCX_FAIL:
+		rtps->remote_sent_to_mgw = false;
+		rtps->codec_sent_to_mgw = false;
+		rtp_stream_update_id(rtps);
+		rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING);
+		return;
+
+	default:
+		OSMO_ASSERT(false);
+	};
+}
+
+void rtp_stream_fsm_established_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state)
+{
+	struct rtp_stream *rtps = fi->priv;
+	osmo_fsm_inst_dispatch(fi->proc.parent, CALL_LEG_EV_RTP_STREAM_ESTABLISHED, rtps);
+}
+
+static int rtp_stream_fsm_timer_cb(struct osmo_fsm_inst *fi)
+{
+	struct rtp_stream *rtps = fi->priv;
+	rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING);
+	return 0;
+}
+
+static void rtp_stream_fsm_cleanup(struct osmo_fsm_inst *fi, enum osmo_fsm_term_cause cause)
+{
+	struct rtp_stream *rtps = fi->priv;
+	if (rtps->ci) {
+		osmo_mgcpc_ep_ci_dlcx(rtps->ci);
+		rtps->ci = NULL;
+	}
+}
+
+void rtp_stream_fsm_discarding_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state)
+{
+	osmo_fsm_inst_term(fi, OSMO_FSM_TERM_REGULAR, NULL);
+}
+
+static const struct value_string rtp_stream_fsm_event_names[] = {
+	OSMO_VALUE_STRING(RTP_STREAM_EV_CRCX_OK),
+	OSMO_VALUE_STRING(RTP_STREAM_EV_CRCX_FAIL),
+	OSMO_VALUE_STRING(RTP_STREAM_EV_MDCX_OK),
+	OSMO_VALUE_STRING(RTP_STREAM_EV_MDCX_FAIL),
+	{}
+};
+
+#define S(x)	(1 << (x))
+
+static const struct osmo_fsm_state rtp_stream_fsm_states[] = {
+	[RTP_STREAM_ST_UNINITIALIZED] = {
+		.name = "UNINITIALIZED",
+		.out_state_mask = 0
+			| S(RTP_STREAM_ST_ESTABLISHING)
+			| S(RTP_STREAM_ST_DISCARDING)
+			,
+	},
+	[RTP_STREAM_ST_ESTABLISHING] = {
+		.name = "ESTABLISHING",
+		.in_event_mask = 0
+			| S(RTP_STREAM_EV_CRCX_OK)
+			| S(RTP_STREAM_EV_CRCX_FAIL)
+			| S(RTP_STREAM_EV_MDCX_OK)
+			| S(RTP_STREAM_EV_MDCX_FAIL)
+			,
+		.out_state_mask = 0
+			| S(RTP_STREAM_ST_ESTABLISHED)
+			| S(RTP_STREAM_ST_DISCARDING)
+			,
+		.action = rtp_stream_fsm_establishing_established,
+	},
+	[RTP_STREAM_ST_ESTABLISHED] = {
+		.name = "ESTABLISHED",
+		.out_state_mask = 0
+			| S(RTP_STREAM_ST_ESTABLISHING)
+			| S(RTP_STREAM_ST_DISCARDING)
+			,
+		.onenter = rtp_stream_fsm_established_onenter,
+		.action = rtp_stream_fsm_establishing_established,
+	},
+	[RTP_STREAM_ST_DISCARDING] = {
+		.name = "DISCARDING",
+		.onenter = rtp_stream_fsm_discarding_onenter,
+		.out_state_mask = 0
+			| S(RTP_STREAM_ST_DISCARDING)
+			,
+	},
+};
+
+static struct osmo_fsm rtp_stream_fsm = {
+	.name = "rtp_stream",
+	.states = rtp_stream_fsm_states,
+	.num_states = ARRAY_SIZE(rtp_stream_fsm_states),
+	.log_subsys = DCC,
+	.event_names = rtp_stream_fsm_event_names,
+	.timer_cb = rtp_stream_fsm_timer_cb,
+	.cleanup = rtp_stream_fsm_cleanup,
+};
+
+static int rtp_stream_do_mgcp_verb(struct rtp_stream *rtps, enum mgcp_verb verb, uint32_t ok_event, uint32_t fail_event)
+{
+	struct mgcp_conn_peer verb_info;
+
+	if (!rtps->ci) {
+		LOG_RTPS(rtps, LOGL_ERROR, "Cannot send %s, no endpoint CI allocated\n", osmo_mgcp_verb_name(verb));
+		return -EINVAL;
+	}
+
+	verb_info = (struct mgcp_conn_peer){
+		.call_id = rtps->call_id,
+		.ptime = 20,
+	};
+
+	if (verb == MGCP_VERB_CRCX)
+		verb_info.conn_mode = rtps->crcx_conn_mode;
+
+	if (rtps->codec_known) {
+		verb_info.codecs[0] = rtps->codec;
+		verb_info.codecs_len = 1;
+		rtps->codec_sent_to_mgw = true;
+	}
+	if (osmo_sockaddr_str_is_set(&rtps->remote)) {
+		int rc = osmo_strlcpy(verb_info.addr, rtps->remote.ip, sizeof(verb_info.addr));
+		if (rc <= 0 || rc >= sizeof(verb_info.addr)) {
+			LOG_RTPS(rtps, LOGL_ERROR, "Failure to write IP address to MGCP message (rc=%d)\n", rc);
+			return -ENOSPC;
+		}
+		verb_info.port = rtps->remote.port;
+		rtps->remote_sent_to_mgw = true;
+	}
+
+	osmo_mgcpc_ep_ci_request(rtps->ci, verb, &verb_info, rtps->fi, ok_event, fail_event, NULL);
+	return 0;
+}
+
+int rtp_stream_ensure_ci(struct rtp_stream *rtps, struct osmo_mgcpc_ep *at_endpoint)
+{
+	if (rtps->ci)
+		return rtp_stream_commit(rtps);
+
+	rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING);
+
+	rtps->ci = osmo_mgcpc_ep_ci_add(at_endpoint, "%s", rtp_direction_name(rtps->dir));
+	if (!rtps->ci)
+		return -ENODEV;
+
+	return rtp_stream_do_mgcp_verb(rtps, MGCP_VERB_CRCX, RTP_STREAM_EV_CRCX_OK, RTP_STREAM_EV_CRCX_FAIL);
+}
+
+int rtp_stream_do_mdcx(struct rtp_stream *rtps)
+{
+	return rtp_stream_do_mgcp_verb(rtps, MGCP_VERB_MDCX, RTP_STREAM_EV_MDCX_OK, RTP_STREAM_EV_MDCX_FAIL);
+}
+
+void rtp_stream_release(struct rtp_stream *rtps)
+{
+	if (!rtps)
+		return;
+
+	rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING);
+}
+
+/* After setting up a remote RTP address or a new codec, call this to trigger an MDCX.
+ * The MDCX will only trigger if all data needed by an endpoint is available (both RTP address and codec) and if at
+ * least one of them has not yet been sent to the MGW in a previous CRCX or MDCX. */
+int rtp_stream_commit(struct rtp_stream *rtps)
+{
+	if (!rtps->ci) {
+		LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no MGW endpoint CI set up\n");
+		return -1;
+	}
+	if (!osmo_sockaddr_str_is_set(&rtps->remote)) {
+		LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no remote RTP address known\n");
+		return -1;
+	}
+	if (!rtps->codec_known) {
+		LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no codec known\n");
+		return -1;
+	}
+	if (rtps->remote_sent_to_mgw && rtps->codec_sent_to_mgw) {
+		LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: both remote RTP address and codec already set up at MGW\n");
+		return 0;
+	}
+
+	LOG_RTPS(rtps, LOGL_DEBUG, "Committing: Tx MDCX to update the MGW: updating%s%s\n",
+		 rtps->remote_sent_to_mgw ? "" : " remote-RTP-IP-port",
+		 rtps->codec_sent_to_mgw ? "" : " codec");
+	return rtp_stream_do_mdcx(rtps);
+}
+
+void rtp_stream_set_codec(struct rtp_stream *rtps, enum mgcp_codecs codec)
+{
+	if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED)
+		rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING);
+	LOG_RTPS(rtps, LOGL_DEBUG, "setting codec to %s\n", osmo_mgcpc_codec_name(codec));
+	rtps->codec = codec;
+	rtps->codec_known = true;
+	rtps->codec_sent_to_mgw = false;
+	rtp_stream_update_id(rtps);
+}
+
+void rtp_stream_set_remote_addr(struct rtp_stream *rtps, const struct osmo_sockaddr_str *r)
+{
+	if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED)
+		rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING);
+	LOG_RTPS(rtps, LOGL_DEBUG, "setting remote addr to " OSMO_SOCKADDR_STR_FMT "\n", OSMO_SOCKADDR_STR_FMT_ARGS(r));
+	rtps->remote = *r;
+	rtps->remote_sent_to_mgw = false;
+	rtp_stream_update_id(rtps);
+}
+
+bool rtp_stream_is_established(struct rtp_stream *rtps)
+{
+	if (!rtps)
+		return false;
+	if (!rtps->fi)
+		return false;
+	if (rtps->fi->state != RTP_STREAM_ST_ESTABLISHED)
+		return false;
+	if (!rtps->remote_sent_to_mgw
+	    || !rtps->codec_sent_to_mgw)
+		return false;
+	return true;
+}