The primary challenge in maintaining a unified communications system is minimizing network delay. Delay is the latency from infrastructure components such as:
Properly configured QoS policies ensure that other data applications do not contend with voice-allocated bandwidth, but they can add delay. Encryption technologies introduce more latency by ignoring priority flags (ToS) and adding additional header information, such as IPsec headers.
Non-uniform packet delays, or jitter, are often more detrimental to VoIP and video call quality than latency. When it is not affecting every voice packet in a stream, delay creates jitter. Jitter affects call quality as the signal starts to sound or look garbled. Jitter can also cause packets to arrive out of order, which introduces additional latency at the application layer when reassembly of the signal occurs.
VoIP-endpoint buffers, network devices that support QoS, and RTP header compression can minimize jitter. But there is always a tradeoff. RTP header compression, for example, adds latency due to the extra processing that is required on the routers.
Packet loss can also result from excess latency or jitter. VoIP and video are also more sensitive to packet loss than other network applications. Loss rates greater than three percent are considered intolerable when compared to plain old telephone service (POTS) calls.
Finally, specialized VoIP equipment presents its own monitoring and maintenance challenge. Test and monitor your call servers to ensure they can handle voice traffic and can route it properly.
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