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2.1.2
Local Network Connectivity
The Local Area Network (LAN) is the communication platform used by the OM20/50 IP Telephony System
for allowing communications among the telephone users and between the telephone users and the external
VoIP and PSTN services. This LAN is composed of the data wiring (UTP cabling), networking equipment
(switches and routers/access device) and the telecommunication (PSTN) lines.
The Local Area Network (LAN) may be already installed or it can be installed and configured at the time of
installing the OM20/50 IP Telephony System Below are the general recommendations to ensure proper
operation of the OM20/50 IP Telephony System.
Ethernet cabling: Ensure there is an Ethernet cabling system and that there is an outlet for each NewRock
VoIP device. It is recommended that Ethernet cables are UTP CAT 5e or better.
2.1.3
Customer Phone Requirements
The IP phones require access to power outlets or can receive power from a Power over Ethernet (PoE)
switch and are not supplied with power supplies. If you are not using the recommended PoE switch, you
need to purchase a suitable power supply.
2.2 Deployment Considerations
When installing and configuring the OM20/50 IP Telephony System, it is necessary to analyze and meet
some design considerations to ensure the best quality and user experience. The design considerations cover
available bandwidth and quality of service.
2.2.1
Codec and Network bandwidth
The available connection bandwidth determines the maximum number of simultaneous calls that the system
can support with the appropriate audio quality. Before installing and configuring the OM 20/50 IP Telephony
System, use this information to determine the maximum number of simultaneous VoIP connections that the
system can support. In general it is a best practice to use no more than 75% of the total available bandwidth
for calls. This provides space for data traffic and helps ensure good voice quality.
NOTE: Some ITSP SIP trunk services limit the maximum number of simultaneous calls. Please check with
your Service Provider to understand the maximum number of simultaneous calls each SIP trunk supports.
The following table provides the approximate bandwidth allocation for different codecs.
Codec
Approximate bandwidth allocation for
each side of conversation (kbit/s)
2 calls
(kbit/s)
10 calls
(kbit/s)
G.729A
55
110
550
G.711U/G.711A
110
220
1100
