To exit the Network Statistics screen, press Back.
Network Statistics Fields
The following table lists the Network Statistics Message information.
Table 2 Network Statistics Message Information for the Cisco Unified SIP Phone 3905
Number of packets received by the phone
Number of packets sent by the phone
Number of broadcast packets received by the phone
Cause of the last reset of the phone - One of these values:
Hardware Reset (Power-on reset)
Software Reset (memory controller also reset)
Software Reset (memory controller not reset)
Link state and connection of the PC port (for example, Auto 100 Mb Full-Duplex means that the PC port is in a link-up state and has auto-negotiated a full-duplex, 100-Mbps connection)
Link state and connection of the Network port
Information on the DHCP status. This includes the following states:
DHCP WAITING COLDBOOT TIMEOUT
SET DHCP COLDBOOT
SET DHCP DISABLED
DISABLED DUPLICATE IP
SET DHCP FAST
Access Call Statistics Screen
You can access the Call Statistics screen on the phone to display counters, statistics, and voice-quality metrics of the most recent call.
You can also remotely view the call statistics information by using a web browser to access the Streaming Statistics web page. This web page contains additional RTCP statistics not available on the phone. For more information about remote monitoring, see Remote Monitoring
A single call can have multiple voice streams, but data is captured for only the last voice stream. A voice stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops even though the call is still connected. When the call resumes, a new voice packet stream begins, and the new call data overwrites the former call data.
To display the Call Statistics screen for information about the latest voice stream, perform these steps:
The following table contains the fields in the Call Statistics screen.
Table 3 Call Statistics Items for the Cisco Unified SIP Phone 3905
Type of voice stream received (RTP streaming audio from codec): G.729, G.711 u-law, G.711 A-law.
Type of voice stream transmitted (RTP streaming audio from codec): G.729, G.711 u-law, G.711 A-law.
Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network) observed since the receiving voice stream was opened.
Maximum jitter observed since the receiving voice stream was opened.
Voice Quality Metrics
Objective estimate of the Mean Opinion Score (MOS) for Listening Quality (LQK) that ranks audio quality from 5 (excellent) to 1 (bad). This score is based on audible-concealment events due to a frame loss in the preceding 8 seconds of the voice stream.
The MOS LQK score can vary based on the type of codec that the Cisco Unified IP Phone uses.
Avg MOS LQK
Average MOS LQK score for the entire voice stream.
Min MOS LQK
Lowest MOS LQK score from the start of the voice stream.
Max MOS LQK
Baseline or highest MOS LQK score from the start of the voice stream.
The following codecs provide the corresponding maximum MOS LQK scores under normal conditions with no frame loss:
MOS LQK Version
Version of the Cisco-proprietary algorithm used to calculate the MOS LQK scores.
Estimate of the network latency, expressed in milliseconds. Represents a running average of the round-trip delay, measured when RTCP receiver report blocks are received.