Mitel MiVOICE BUSINESS, Инструкция по инженерии

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Engineering Guidelines
210
NETWORK MEASUREMENT CRITERIA
Assuming that jitter and packet loss are not an issue, the one parameter left that affects the
voice and conversation quality is end-to-end delay. From ITU-T recommendations (and practical
experience), the end-to-end delay for a voice call should not exceed 150 ms. The characteristics
of the end devices such as the gateway (Ethernet and TDM bridge in the 3300 ICP) and the
IP phones are known.
In assessing a network, consider the network limits shown in the following table.
“Ping” delay is the value obtained using a PC ping utility. The ping utility sends a message from
one PC to a second PC. When the second PC receives the message, it sends a message back
to the first PC. The first PC determines the propagation delay encountered on the network
between the two PCs. Typically the send and receive paths have equal delays. Estimate jitter
by using ping over a short and longer-term period. Estimate packet loss by using ping over a
longer period (24 hours or more). Networks that are used for both voice and data can have
variations in the amount of network delay. For instance, if computer backup utilities run on a
regularly scheduled basis, network delay can increase. Perform longer-period delay
measurements over a time period that represents the customer's core operational hours.
Other tools, such as network analyzers, can also be used to determine packet loss. Many
analyzers look for VoIP and RTP packets, and can identify when a packet is missing as well
as average jitter.
Although ping can be used as a quick check or as a backup method, it is recommended that
networks be fully evaluated before installation. Mitel Consultants and Integrators, can provide
Professional Services to perform a full VoIP network pre-installation evaluation.
BANDWIDTH REQUIREMENTS
Consider the following when calculating bandwidth requirements:
• Level of call traffic (more phone calls means more bandwidth)
• Bandwidth required for speech connections (that is, codec to be used)
• Bandwidth required for signalling.
In general, the level of call traffic defines the number of Erlangs (busy channels) and hence,
the number of “channels.” As a simple rule of thumb, add 10% to the voice bandwidth to ensure
adequate signalling bandwidth. In practice, the signalling is needed only to set up a call and
clear it down. The signalling messages are also sent via TCP and acknowledged. Some delay
is tolerated in this case, unlike the voice case.
Table 62: Network limits
PACKET LOSS JITTER
END-TO-END
DELAY
PING DELAY
Go! <0.5% <20 ms <50 ms <100 ms
Caution <2% <60 ms <80 ms <160 ms
Stop! >2% >60 ms >80 ms >160 ms