pespin has submitted this change. ( https://gerrit.osmocom.org/c/osmo-mgw/+/39214?usp=email )
Change subject: mgw: Introduce mgcp_rtp_end_init() ......................................................................
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(-)
Approvals: laforge: Looks good to me, but someone else must approve Jenkins Builder: Verified daniel: Looks good to me, approved
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) &&
-
pespin