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

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.

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.

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

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