4600 Series Ip Telephones - Avaya 4600 Series Administrator's Manual

A network assessment should include:
A network audit to review existing equipment and evaluate its capabilities, including its
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ability to meet planned voice and data needs.
A determination of network objectives, including the dominant traffic type, selection of
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technologies, and setting voice quality objectives.
The assessment should leave you confident that the implemented network will have the
capacity for the foreseen data and voice traffic, and can support H.323, SIP, DHCP, TFTP,
HTTP, and jitter buffers in all applications.
It is important to distinguish between compliance with the minimal VoIP standards and QoS
support, the latter being a requirement to run VoIP on your configuration.

4600 Series IP Telephones

The 4600 Series IP Telephones support either of two signaling protocol families - H.323 and
Session Initiation Protocol (SIP).
The H.323 standard, developed by ITU-T, provides for real time audio, video, and data
communications transmission over a packet network. An H.323 telephone protocol stack
comprises several protocols:
H.225 for registration, admission, status (RAS), and call signaling,
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H.245 for control signaling,
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Real Time Transfer Protocol (RTP), and
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Real Time Control Protocol (RTCP)
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SIP was developed by the IETF. Like H.323, SIP also provides for real time audio, video, and
data communications transmission over a packet network. SIP uses various messages, or
methods, to provide:
Registration (REGISTER),
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Call signaling (INVITE, BYE)
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Control signaling (SUBSCRIBE, NOTIFY)
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SIP also supports RTP and RCTP using the Session Description Protocol.
A telephone is loaded with either H.323 or SIP software as part of its initial script file
administration and initialization.
4600 Series IP Telephones
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