Avaya 1000 Manual page 92

Note 2: Insecure shells, including FTP, Telnet, and rlogin access, may be enabled or disabled and the status of insecure shells may be displayed,
using either the command line interface (CLI) commands, or the Element Manager. See reference 6 (Communication Server 1000 Security
Management Fundamentals) for details.
Note 3: Campus Redundancy increases the redundancy of a single CS 1000E system through the physical separation of the CS 1000E High
Availability (HA) Core Call Servers. To separate the redundant Call Servers, the ELAN subnet and the subnet of the High Speed Pipe (HSP)
may be extended between the two processors using networking equipment that provides layer 2 end to end connectivity. See reference 5
(Communication Server 1000 System Redundancy Fundamentals) for details.
Note 4: Packet loss on the Enterprise IP network must be less than 0.5% (0% packet loss is recommended), and round-trip delay between Call Server
and MG 1000E must be less than 80 ms. The specified network requirements for ELAN (80 ms round trip delay and 0.5% packet loss) apply to
all traffic. Note that because the ELAN network is a Layer 2 Switched LAN, the packet loss must be zero. If packet loss is experienced, its
source must be investigated and eliminated. For reliable signaling communication on the ELAN network interface, the packet loss must be <
1%. See reference 4 (Communication Server 1000 Converging the Data Network with VoIP Fundamentals) for details.
Note 5: All critical signaling flows between the MGC and CSs must be tagged at the same DSCP value.
Note 6: It is strongly recommends that suitable Quality of Service (QoS) mechanisms be implemented on any IP network that carries VoIP
Apply QoS mechanisms to the following VoIP media and signaling paths:

TLAN connections

VoIP traffic between IP Phones

VoIP traffic between IP Phones and Voice Gateway Media Cards on the TLAN subnet
It is recommended that an end-to-end delay of <= 50 ms be used on the IP network to ensure good voice quality. The 50 ms does not include
the built-in delay of the Voice Gateway Media Card and IP Phone.
For high-quality voice transmission, the long-term average packet loss between the IP Phones and the Voice Gateway Media Card TLAN
network interface must be < 1%, and the short-term packet loss must not exceed 5% in any 10-second interval.
It is strongly recommends that the G.711 codec be used with the following configuration:

end-to end delay less than 150 ms one way (network delay packetization delay + jitter buffer delay < 150)

packet loss less than 0.5% (approaching 0%)

maximum jitter buffer setting for IP Phone as low as possible (maximum 100 ms)
The default jitter buffer delay for voice on the receiving IP Trunk 3.0 (or later) node is 60 ms.
The jitter buffer setting is configured on the voice gateway channels of the Voice Gateway Media Card, and are sent out to IP Phones. The
jitter buffer size is set when you configure the DSP Profiles:
Avaya Communication Server 1000 Port Utilization – Issue 4.04
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Use pursuant to the terms of your signed agreement or Avaya policy.
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