Network capacity can be expressed in terms of the amount of traffic that it can manage. With VoIP, network capacity is measured with respect to the number of simultaneous calls that it can process.
VoIP capacity planning for a network should be based on the peak load that the network will be called upon to handle. Factors such as LAN/WAN design, existing data traffic load, type of voice codecs, hardware capabilities, redundancy in the network, etc need to be considered. The first step in implementing VoIP is to have knowledge of the bandwidth capacity of each link in the existing network as this helps in identifying the potential bottlenecks. VoIP deployment on a single-site network that has a high-speed infrastructure is not likely to be encumbered by network capacity; instead it is the network layout that may result in bandwidth-related issues.
VoIP communications on WANs can suffer as bandwidth bottlenecks are created on the serial-based connections that work using the T1 or fractional T1 lines. VoIP QoS guidelines state that voice traffic be prioritized but if this is followed, then data traffic will slow down during peak traffic hours. If standard PRI or voice T1 lines are used, it may not be possible to place additional VoIP calls once the available channels are used. If the number of VoIP calls is more than what the network can handle, users will face operator error or fast-busy messages. The voice quality will drop even if data is compressed and QoS standards are in place.
The mix of PSTN connectivity and additional bandwidth can be arrived at by evaluating the average and peak usage for each connection. The decision to either increase the speed of existing circuits or add more lines is influenced by the number of users in each location and peak usage. The increased use of services such as Metropolitan-area networks (MAN) will make the addition of higher bandwidth lines a cheaper alternative.
Bandwidth monitoring techniques should be employed to study the effect of VoIP traffic on bandwidth utilization. VoIP-specific tools are useful for recreating voice traffic scenarios, checking for errors, and monitoring for problems like jitter and delay. Call loads vary with the sampling rates provided by different codecs such as G.711 and G.729. Compression and call loads should be tested in real-time. Remote locations can either have PSTN connectivity or they can be managed from a central location by using a WAN.
Centralized PSTN connectivity helps to reduce deployment costs and increase network redundancy. The hardware should be able to cope with the increased overall traffic; the distribution-level points should not turn into bottlenecks. Almost all current Ethernet hardware supports a 100 Mbps connection to the phone. Modular network hardware allows users to increase the port density while using the existing equipment.
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