pespin has uploaded this change for review.

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mgw: Introduce mgcp_rtp_end_init()

Change-Id: I29f57dfde60e3134f1db422dc63e7c58d3db7d54
---
M include/osmocom/mgcp/mgcp_rtp_end.h
M src/libosmo-mgcp/mgcp_conn.c
M src/libosmo-mgcp/mgcp_rtp_end.c
3 files changed, 18 insertions(+), 12 deletions(-)

git pull ssh://gerrit.osmocom.org:29418/osmo-mgw refs/changes/14/39214/1

diff --git a/include/osmocom/mgcp/mgcp_rtp_end.h b/include/osmocom/mgcp/mgcp_rtp_end.h
index 9c3bb4d..2ab300a 100644
--- a/include/osmocom/mgcp/mgcp_rtp_end.h
+++ b/include/osmocom/mgcp/mgcp_rtp_end.h
@@ -52,5 +52,6 @@
 	char local_addr[INET6_ADDRSTRLEN];
 };
 
+void mgcp_rtp_end_init(struct mgcp_rtp_end *end);
 bool mgcp_rtp_end_remote_addr_available(const struct mgcp_rtp_end *rtp_end);
 void mgcp_rtp_end_free_port(struct mgcp_rtp_end *end);
diff --git a/src/libosmo-mgcp/mgcp_conn.c b/src/libosmo-mgcp/mgcp_conn.c
index e7ff30b..8682e4b 100644
--- a/src/libosmo-mgcp/mgcp_conn.c
+++ b/src/libosmo-mgcp/mgcp_conn.c
@@ -88,7 +88,6 @@
 /* Initialize rtp connection struct with default values */
 static int mgcp_rtp_conn_init(struct mgcp_conn_rtp *conn_rtp, struct mgcp_conn *conn)
 {
-	struct mgcp_rtp_end *end = &conn_rtp->end;
 	/* FIXME: Each new rate counter group requires an unique index. At the
 	 * moment we generate this index using this counter, but perhaps there
 	 * is a more concious way to assign the indexes. */
@@ -106,17 +105,6 @@
 	/* backpointer to the generic part of the connection */
 	conn->u.rtp.conn = conn;
 
-	end->rtp = NULL;
-	end->rtcp = NULL;
-	memset(&end->addr, 0, sizeof(end->addr));
-	end->rtcp_port = 0;
-
-	/* Set default values */
-	end->frames_per_packet = 0;	/* unknown */
-	end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
-	end->output_enabled = false;
-	end->maximum_packet_time = -1;
-
 	conn_rtp->ctrg = rate_ctr_group_alloc(conn, &rate_ctr_group_desc, rate_ctr_index++);
 	if (!conn_rtp->ctrg)
 		return -1;
@@ -124,6 +112,8 @@
 	conn_rtp->state.in_stream.err_ts_ctr = rate_ctr_group_get_ctr(conn_rtp->ctrg, IN_STREAM_ERR_TSTMP_CTR);
 	conn_rtp->state.out_stream.err_ts_ctr = rate_ctr_group_get_ctr(conn_rtp->ctrg, OUT_STREAM_ERR_TSTMP_CTR);
 
+	mgcp_rtp_end_init(&conn_rtp->end);
+
 	/* Make sure codec table is reset */
 	mgcp_codec_reset_all(conn_rtp);
 
diff --git a/src/libosmo-mgcp/mgcp_rtp_end.c b/src/libosmo-mgcp/mgcp_rtp_end.c
index ddc1533..422c199 100644
--- a/src/libosmo-mgcp/mgcp_rtp_end.c
+++ b/src/libosmo-mgcp/mgcp_rtp_end.c
@@ -29,11 +29,26 @@
 #include <osmocom/core/socket.h>
 #include <osmocom/core/osmo_io.h>
 #include <osmocom/mgcp/mgcp_rtp_end.h>
+#include <osmocom/mgcp/mgcp_codec.h>
 
 /***********************
  * mgcp_rtp_end
  **********************/
 
+void mgcp_rtp_end_init(struct mgcp_rtp_end *end)
+{
+	end->rtp = NULL;
+	end->rtcp = NULL;
+	memset(&end->addr, 0, sizeof(end->addr));
+	end->rtcp_port = 0;
+
+	/* Set default values */
+	end->frames_per_packet = 0;	/* unknown */
+	end->packet_duration_ms = DEFAULT_RTP_AUDIO_PACKET_DURATION_MS;
+	end->output_enabled = false;
+	end->maximum_packet_time = -1;
+}
+
 bool mgcp_rtp_end_remote_addr_available(const struct mgcp_rtp_end *rtp_end)
 {
 	return (osmo_sockaddr_port(&rtp_end->addr.u.sa) != 0) &&

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