User manual

4-21

Chapter 4

Analyze Network

VoIP

Select the VoIP button in the Bottom Area.

NOTE: Select another Test button visible on the screen or by scrolling to

the right or left with the white arrows or select the Exit button.

VoIP uses three different protocols. Detected phone lines could also be

video links or data links.

SIP is used for <Session Management> - the VoIP device/line

combination registers with the SIP server at intervals (usually

every few minutes). In this way, the location and status of the

device/line is maintained with the VoIP provider. This

authenticates the device/line also. Incoming and outgoing calls

are also managed through the SIP server.

RTP is used for the actual <streaming data> (i.e. audio in a

phone call) - when a call is made to/from a device/line then the

SIP server provides the device/line with the IPs of the RTP

server to use. The device/line then sends or receives a data

stream to/from the RTP server. Generally, this data is at a

consistent rate (usually 50fps for audio) and there are two data

streams (one for inbound data, the other for outbound).

RTCP is used to allow each end of an active RTP session to

communicate link quality information with each other. Not all RTP

devices or servers do this. If this occurs, the sender will

generally send one packet every few seconds. This allows the

VoIP provider to maintain knowledge of the performance of the

network.

There are separate entries for each line/device/server combination. The

example on the following page shows one device, one server and two

lines. The terminology used in this example is from the definition of SIP.