VoIP Service Considerations
In traditional telephony networks, dedicated bandwidth for each voice stream provides
voice with a guaranteed delay across the network. Because bandwidth is guaranteed in a
TDM environment, no variable delay exists (that is, jitter). Configuring voice in a data
network requires network services with low delay, minimal jitter, and minimal packet loss.
Bandwidth requirements must be properly calculated based on the codec used and the
number of concurrent connections. QoS must be configured to minimize jitter and loss
of voice packets. The PSTN provides 99.999 percent availability (that is, the five nines of
availability). To match the availability of the PSTN, an IP network must be designed
with redundancy and failover mechanisms. Security policies must be established to
address both network stability and voice-stream security.
Table 1-1 lists issues associated with implementing VoIP in a converged network and solutions
that address these issues.
Table 1-1 Issues and Solutions for VoIP in a Converged Network
Issue Solutions
Latency Increase bandwidth.
Choose a different codec type.
Fragment data packets.
Prioritize voice packets.
Jitter Use dejitter buffers.
Prioritize voice packets.
Bandwidth Calculate bandwidth requirements, including voice payload, overhead, and data.
Packet loss Design the network to minimize congestion.
Prioritize voice packets.
Use codecs to minimize small amounts of packet loss.
Reliability Provide redundancy for hardware, links, and power (uninterruptible power
supply [UPS]).
Perform proactive network management.
Security Secure the following components:
■ Network infrastructure
■ Call-processing systems
■ Endpoints
■ Applications