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Administrator’s Handbook
56
Voice
If you click the
Voice
ink, the
Voice
page appears.
Voice-over-IP (VoIP) refers to the ability to make voice telephone calls over the Internet. This differs from tradi-
tional phone calls that use the Public Switched Telephone Network (PSTN). VoIP calls use an Internet protocol,
Session Initiation Protocol (SIP), to transmit sound over a network or the Internet in the form of data packets.
The Voice page displays information about your VoIP phone lines, if configured. Your Gateway supports two
phones,
Line 1
and
Line 2
.
If either one or both are registered with a SIP server by your service provider or not registered, the Voice page
will display their
Registration Details
.
The links at the top of the Voice page access a series of pages to allow you to configure and monitor
features of your device. The following sections give brief descriptions of these pages.
Line Details
” on page
57
Call Statistics
” on page
58
Page 57 / 216
57
Link: Line Details
When you click the
Line Details
link, the
Line Details
page appears.
If your service provider has enabled your VoIP phone lines, you can register them by clicking the
Register
Line 1
or
Register Line 2
button(s).
To test if the lines are enabled, click the
Ring Line 1
or
Ring Line 2
button(s). If enabled and registered, the
respective phone will ring until you click the
Stop Ring Line 1
or
Stop Ring Line 2
buttons.
To clear the current state of each phone line, click the
Reset Line 1
or
Reset Line 2
button(s). This will dis-
connect any calls currently in progress as well.
To update the display, click the
Refresh
button.
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Administrator’s Handbook
58
Link: Call Statistics
When you click
Call Statistics
, the
Call Statistics
page appears.
For
Line 1
and
Line 2
:, the two available phone lines, the Call Statistics page displays the following information:
Call Statistics - Line 1 and Line 2
Last Call/Cumulative – Incoming/Outgoing
RTP Packet Loss
Real-time Transport Protocol packets dropped
RTP Packet Loss percent-
age
Percent of Real-time Transport Protocol packets dropped
Total RTCP Packets
Total Real-time Transport Control Protocol packets
Average Inter Arrival Jitter
This is calculated continuously in milliseconds as each data packet is received
and averaged.
Max Inter Arrival Jitter
This is the maximum value in milliseconds recorded as each data packet is
received.
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59
Sum of Inter Arrival Jitter
This is calculated continuously in milliseconds as each data packet is received
and totalled.
Sum of Inter Arrival Jitter
Squared
This is calculated continuously in milliseconds as each data packet is received
and the total is squared.
Sum of Franc Loss
Fraction Lost: The fraction of RTP data packets lost since the previous SR or
RR packet was sent. This fraction is defined to be the number of packets lost
divided by the number of packets expected. This will be calculated on every
RTCP SR packet. Sum of the fraction lost is calculated with all the RTCP pack-
ets.
Sum of Franc Loss
Squared
Fraction lost is squared with every RTCP SR or RR packet. Sum of all this will
give the Sum of Franc Loss Squared.
Max One Way Delay
One Way Delay will be calculated in milliseconds on every RTCP SR or RR
packet. This value is (systime - lsr - dslr) / 2
lsr means last SR timestamp
dslr means delay since last SR.
Sum of One Way Delay
The sum of all the one way delays calculated in milliseconds on every RTCP
packet is displayed as Sum of One Way Delay.
Sum of One Way Delay
Squared
One Way Delay is squared with every RTCP SR or RR packet. Sum of all this
will give the Sum of One Way Delay Squared.
Avg Round Trip Time
Average time in milliseconds from this local source to destination address and
back again for all logged calls
Max Round Trip Time
Maximum amount of time in milliseconds from this local source to destination
address and back again for all logged calls
Sum of Round Trip Time
Sum of time in milliseconds from this local source to destination address and
back again for all logged calls
Sum of Round Trip Time
Squared
Sum squared of time from this local source to destination address and back
again for all logged calls
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Administrator’s Handbook
60
For
Line 1
and
Line 2
:, the two available phone lines, the Call Summary section displays the following informa-
tion:
Call Summary - Line 1 and Line 2
Current Call/Last Completed Call
Call Timestamp
Date and Time of the current call
Type
May be Incoming or Outgoing
Duration
Length of time in seconds of call connection
Codec in Use
Audio codec used for decoding the call packet traffic.
Far-End Host Information
SIP server IP information: IP address and port number
Far-End Caller Information
Caller ID information, if available
Cumulative Since Last Reset
Last Reset Timestamp
Date and Time of the last call
Number of Calls
Total number of calls for each VoIP line
Duration
Time in seconds since the last call
Number of Incoming Calls Failed
Number of Incoming calls that fail to connect
Number of Outgoing Calls Failed
Number of Outgoing calls that fail to connect

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