Q: What is Digital Voice Quality of Service (QoS)?

A: Quality of Service or QoS is the quality of a call over a network. It also refers to the ability to prioritize certain types of traffic on an IP network. In the case of Digital Voice, this typically means prioritizing voice traffic at a higher level than other forms of traffic such as data so that voice traffic will not be delayed or dropped. Most QoS solutions focus on either resource reservation or resource prioritization.

Q: What is Digital Voice latency?

A: Digital Voice latency causes delays in packet delivery. Physical distance, the number of router hops, encryption, and voice/data conversion all impact latency. Users begin noticing latency as a service level issue when roundtrip latency is greater than 250 milliseconds (ms). The International Telecommunications Union recommends that latency never exceed 150 ms one way (from speaker to listener).

Q: What is Digital Voice jitter?

A: Digital Voice jitter occurs when voice packets are sent and received with timing variations. Jitter is effectively a variation of packet delay where delays actually impact the quality of the conversation. Think of jitter as variable delays in packet delivery. Participants will notice delays in the conversations impacted by jitter. As a result, many service providers now account for maximum jitter levels.

Q: What is Digital Voice packet loss?

A: Digital Voice packet loss takes place when a large amount of traffic hits the network and causes it to drop packets. It usually manifest itself as dropped conversations or “tinny” sounds. Packet loss should never exceed 1% and most service providers guarantee service levels with .5% or less packet loss. Packet loss of 1% translates into one voice clip or skip every three minutes, while packet loss of .25% translates into one error every 53 minutes.

Q: What steps should be taken to improve Digital Voice service levels?

A: Latency and jitter really go hand in hand in most Digital Voice deployments. To effectively manage both, administrators should focus on reducing delay at the network endpoint and prioritizing traffic over the network. Optimization of jitter buffering and packet size are good first steps to improving service quality at the endpoint. Endpoint delays can also be reduced by adhering to a standard packet size, using the G.711 codec, and avoiding asynchronous transcoding.

Prioritizing Digital Voice traffic over the network at Layers 2 and 3 also yields latency and jitter improvements. Policy based network management, bandwidth reservation, Type of Service, Class of Service, and Multi-Protocol Label Switching (MPLS) are all widely used techniques for prioritizing Digital Voice traffic at Layers 2 and 3.

Various solutions are available for Digital Voice packet loss. These include: packet loss concealment which hides the audible effects of lost packets; root cause analysis of individual switches and routers along the network path that may be the sole cause of loss; and, when packet loss manifests itself as jitter, monitoring for congestion.

E-mail: Digital Voice Information