Network Audio Quality Display On 4600 Series Ip Telephones - Avaya 4600 Series Administrator's Manual

Server Administration
QoS
Network Audio Quality Display on 4600 Series
IP Telephones
As of Release 1.7 for the 4620 and 4630/4630SW telephones, and as of Release 1.8 for the
4602/4602SW/4610SW/4620SW IP Telephones, the telephone is by default administered to offer the end
user an opportunity to monitor network audio performance while on a call. The user guides for each
phone go into specific detail on how the user gets to the appropriate screen; this document tells you what
the end user can see, and what it means.
For 4610SW/4620/4620SW/4630/4630SW IP Telephones, the following parameters are displayed in
real-time to users on the appropriate screens, while on a call:
Table 5: Parameters in Real-Time
Parameter
Audio Connection
Present?
Received Audio
Coding
Silence
Suppression
Packet Loss
Packetization
Delay
One-way Network
Delay
Network Jitter
Compensation
Delay
For 4602/4602SW IP Telephones, the Network Audio Quality Screen presents the user with a qualitative
assessment of the overall audio quality currently being experienced. This assessment is based on separate
evaluations of the Packet Loss and the total Network Delay (the sum of Packetization Delay, One-way
Network Delay, and Network Jitter Compensation Delay), and consideration of the codec in use. You can
disable the display of the Network Audio Quality data and assessment for all sets by setting the system
value NTWKAUDIO to a value of "0" as explained in
Telephones on page 66.
The implication of this information for LAN administration depends, of course, on the values reported by
the user and the specific nature of your LAN (topology, loading, QoS administration, etc.). The major use
for this information is to give the user an idea of how network conditions are affecting the audio quality
of the current call. It is assumed you have more detailed tools available for troubleshooting the LAN.
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Possible Values
Yes (if a receive RTP stream has been established)
No (if a receive RTP stream has not been established)
G.711 or G.729
Yes (if the telephone knows the far-end has silence suppression Enabled)
No (if the telephone knows the far-end has silence suppression Disabled, or the
telephone does not know either way)
No data or a decimal percentage. Late and out-of-sequence packets are counted
as lost if they are discarded. Packets are not counted as lost until a subsequent
packet is received and the loss confirmed by the RTP sequence number.
No data or an integer number of milliseconds. The number reflects the amount
of delay in received audio packets, and includes any look-ahead delay
associated with the codec.
No data or an integer number of milliseconds. The number is one-half the value
RTCP computes for the round-trip delay.
No data or an integer number of milliseconds reporting the average delay
introduced by the telephone's jitter buffer.
4600 Series IP Telephone Release 2.1 LAN Administrator's Guide
Administering Options for the 4600 Series IP
July 2004