SBC : Jitter, No Voice, Connection Drop on SIP calls

Doc ID		SOLN216068
Version:		3.0
Status:		Published
Published date:		28 Jul 2021
Created Date:		09 Dec 2012

Author:

kharev

Details

Employees still complain about voice interruptions and connection drops in about 10-15% of all SIP calls. In general in wireshark traces, the statistics of the RTP streams from the far end look much better than the statistics of the all streams that we send to the far end. Operating system: Linux 2.6.11-AV18h i686 i686
Built: Aug 7 17:01 2008

Contains: 01.5.642.3
Reports as: R013x.01.5.642.3
Release String: S8700-013-01.5.642.3
UPDATES:
01.5.642.3-16210 unpacked cold patch 16210 for 01.5.642.3
01.5.642.3-17196 unpacked cold patch 17196 for 01.5.642.3
01.5.642.3-17944 activated cold patch 17944 for 01.5.642.3

Problem Clarification

Currently we are using only G.711 codec to simplify the investigation. Although in some RTP streams (in both directions) we can see lost packets. RTP from FE show around 1% packet loss, however RTP from PBX shows 80% Packet loss.

On enabling G.729 the loss reduced to 5-10%.

Cause

A big jitter can cause voice quality degradation in case of G.711 easily.

In RTP from far end there were mostly less than 1% of lost packets. Normally we expect RTP packets every 20 ms. But we can see 5ms and even 36ms in Wireshark traces.

Solution

Check with Service Provider for such big Jitters. No Issues at Avaya's end.
Use ifconfig command to check the drop packet value

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