Network Audio Quality Display on 9600 Series IP Telephones
All 9600 Series IP Telephones give the user an opportunity to monitor network audio performance while on a call. For more information, see the telephone user guide.
While on a call, the telephones display network audio quality parameters in real-time, as shown in Table 4:
Table 4: Parameters in Real-Time
| Parameter |
Possible Values |
| Received Audio Coding |
G.711, G.722, G.726A, or G.729. |
| Packet Loss |
No data or a 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. |
| Packetization Delay |
No data or an integer number of milliseconds. The number reflects the amount of delay in received audio packets, and includes any potential delay associated with the codec. |
| One-way Network Delay |
No data or an integer number of milliseconds. The number is one-half the value RTCP computes for the round-trip delay. |
| Network Jitter Compensation Delay |
No data or an integer number of milliseconds reporting the average delay introduced by the jitter buffer of the telephone. |
The implication for LAN administration depends on the values the user reports and the specific nature of your LAN, like topology, loading, and QoS administration. This information gives the user an idea of how network conditions affect the audio quality of the current call. Avaya assumes you have more detailed tools available for LAN troubleshooting.
