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Ivan Kluchnikov gerrit-no-reply at lists.osmocom.orgReview at https://gerrit.osmocom.org/1784 rsl: Output RTP stats before closing the socket. It's useful to know RTP statistics (number of packets lost, jitter, etc) when looking at voice call quality issues. Right now this information is not avialable anywhere and this looks like the best place to start. Change-Id: Ife9f27c43157b4a1bf71aba41cd7b0f5f41ac99f --- M src/common/rsl.c 1 file changed, 4 insertions(+), 0 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmo-bts refs/changes/84/1784/1 diff --git a/src/common/rsl.c b/src/common/rsl.c index 5c4e4ff..deacdc3 100644 --- a/src/common/rsl.c +++ b/src/common/rsl.c @@ -1023,6 +1023,8 @@ if (lchan->abis_ip.rtp_socket) { rsl_tx_ipac_dlcx_ind(lchan, RSL_ERR_NORMAL_UNSPEC); + osmo_rtp_socket_log_stats(lchan->abis_ip.rtp_socket, DRSL, LOGL_INFO, + "Closing RTP socket on Channel Release "); osmo_rtp_socket_free(lchan->abis_ip.rtp_socket); lchan->abis_ip.rtp_socket = NULL; msgb_queue_flush(&lchan->dl_tch_queue); @@ -1786,6 +1788,8 @@ inc_conn_id = 1; rc = rsl_tx_ipac_dlcx_ack(lchan, inc_conn_id); + osmo_rtp_socket_log_stats(lchan->abis_ip.rtp_socket, DRSL, LOGL_INFO, + "Closing RTP socket on DLCX "); osmo_rtp_socket_free(lchan->abis_ip.rtp_socket); lchan->abis_ip.rtp_socket = NULL; msgb_queue_flush(&lchan->dl_tch_queue); -- To view, visit https://gerrit.osmocom.org/1784 To unsubscribe, visit https://gerrit.osmocom.org/settings Gerrit-MessageType: newchange Gerrit-Change-Id: Ife9f27c43157b4a1bf71aba41cd7b0f5f41ac99f Gerrit-PatchSet: 1 Gerrit-Project: osmo-bts Gerrit-Branch: master Gerrit-Owner: Ivan Kluchnikov <kluchnikovi at gmail.com> Gerrit-Reviewer: Alexander Chemeris <Alexander.Chemeris at gmail.com>