Voice testing overview, Supported protocols – Agilent Technologies N2620A User Manual
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N2620A User’s Guide
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Voice Testing with the FrameScope Pro
Voice Testing Overview
The VoIP Service Quality Testing feature of the FrameScope Pro
provides you analysis options for VoIP environments that
traditional tests do not. These options are listed as follows.
•
Accessing, managing, and running voice quality tests
end-to-end across widely deployed voice networks
•
Testing VoIP network components such as routers, gateways,
PBXs, and switches
•
Comparing VoIP quality directly with the existing toll quality
networks
•
Testing VoIP systems to gather end-to-end voice quality
information
•
Augmenting other traditional telephony test suites such as
the Transmission Impairment Measurement Set (TIMS)
•
Measuring fundamental voice quality metrics such as delay,
jitter, and Mean Opinion Score (MOS)
Supported Protocols
The VoIP test feature supports the following protocols.
•
Session Initiation Protocol (SIP), RFC 3261, which requires
license Option N2620A-030 or N2620A-03E
•
Peer-to-peer SIP, RFC 3261, which requires license Option
N2620A-030 or N2620A-03E
•
STUN, RFC 3489, which requires license Option N2620A-03E
•
Megaco/H.248, RFC 3525, which requires license Option
N2620A-032
•
VoIP traffic generation, which requires license Option
N2620A-03G
•
Background traffic generation, which requires license Option
N2620A-041
, “License Details” on page 163 for the
information on activating this feature.