Network Audio Quality Display on 4600 Series IP Telephones
With the exceptions of the 4601, 4601+, 4606, 4612, 4624, and 4690 IP Telephones, all Series 4600 IP Telephones are by default administered to allow the end user an opportunity to monitor network audio performance while on a call. The user guides for each telephone provide specific detail on getting to the appropriate screen, what the end user sees, and what the information means.
For 4610SW/4620/4620SW/4621SW/4622SW/4625SW/4630/4630SW IP Telephones, these parameters display in real-time to users on the appropriate screens, while on a call:
Table 5: Parameters in Real-Time
| Parameter |
Possible Values |
| Audio Connection Present? |
Yes if a receive RTP stream was established. No if a receive RTP stream was not established. |
| Received Audio Coding |
G.711, G.726A, or G.729. |
| Silence Suppression |
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. |
| Packet Loss |
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. |
| Packetization Delay |
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. |
| 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 telephone’s jitter buffer. |
For 4602/4602SW/4602SW+ IP Telephones, the Network Audio Quality Screen gives the user a qualitative assessment of the current overall audio quality. This assessment is based on separate evaluations of:
- the Packet Loss, and
- the total Network Delay, which is the sum of Packetization Delay, One-way Network Delay, and Network Jitter Compensation Delay, and
- consideration of the codec in use.
This information’s implication for LAN administration depends, of course, on the values the user reports and the specific nature of your LAN, like topology, loading, QoS administration, etc. This information’s major use is to give the user an idea of how network conditions affect the current call’s audio quality. It is assumed you have more detailed tools available for troubleshooting the LAN.
