The Essential Guide to VoIP QoS

Updated: August 20, 2012

Despite widespread perceptions to the contrary, VoIP telephony can actually supply service levels matching and even exceeding those of traditional phone service. How well (or poorly) VoIP performs is simply a QoS (quality-of-service) matter.

Improving and managing VoIP QoS is an issue that's central to nearly all IP telephony users. Most VoIP adopters soon come to realize that QoS is an ongoing challenge that requires a solid understanding of several technical and business topics, as well as a commitment to address service quality on multiple levels.

QoS Challenges

VoIP QoS is a general term that encompasses several different technical areas, including:

Latency: Latency is the time between the point that a voice packet is transmitted and the point that it reaches its destination. In a VoIP system, long latency periods can create echoes and overlapping voices, disrupting communications. Latency begins to become noticeable at about 250 milliseconds. The ITU ( International Telecommunication Union) suggests that latency should be kept to less than 150 milliseconds from speaker to listener.

Jitter: When some of the transmitted data packets are delayed, usually due to congested or slow network links, jitter can result. Jitter affects QoS by creating audio slowdowns that can give a VoIP call small gaps or stutters. Since jitter is by nature a variable type of latency, the problem can be difficult to quantify.

Packet Loss: Packet loss can happen for many reasons, such as when network congestion causes a router buffer overflow, when a network link fails and when packets are misrouted. Mild to moderate cases of packet loss tend to generate audio with a metallic sound. Packet loss can also lead to occasional blips in the audio steam, lengthy audio-stream gaps, and even total audio failure and call collapse. In any event, packet loss should never be greater than 1 percent of the audio steam; 0.5 percent or less is considered by many industry experts to be a benchmark acceptable level. A 1 percent packet loss equals one audio skip every 3 minutes; a 0.25 percent packet loss means that there is one error every 53 minutes.

Addressing QoS

Latency, jitter and packet-loss challenges hinge on several factors, including service-provider performance, the condition of the customer's internal network, and the condition of the connection between the business and the service provider.

Businesses can take several steps to help ensure that on-site network deficiencies don't degrade VoIP performance. Network administrators can reduce latency and jitter at the endpoint by optimizing buffer and packet-size settings. Using the G.711 codec to establish a uniform packet size, as well as steering clear of asynchronous transcoding, can also alleviate latency and jitter symptoms.

Traditional data networks tend to be optimized for data traffic that is not time-based, but retuning for VoIP packets doesn't necessarily mean that data traffic suffers — it is more a matter of setting the correct priorities. Network administrators can also enhance QoS at the network endpoint by prioritizing VoIP network traffic at layer 2 and layer 3. Existing data networks should also be carefully analyzed for their ability to absorb extra traffic before a VoIP infrastructure is dropped on to it. It is also important to examine the network's physical condition to make sure that worn or outdated cabling isn't contributing to QoS issues.

Service-Provider Issues

VoIP service-provider QoS varies widely, with plan pricing often dictating service levels. Many VoIP service providers offer their customers an SLA (service-level agreement) that guarantees basic performance, reliability and survivability standards, usually in exchange for a long-term service commitment. SLAs are often calculated on an MOS (Mean Opinion Score) that's based on call-completion rates, as well as the time required for a user to hear a dial tone or to connect to a destination.

ISPs are at the root of some QoS problems. Recently, several ISPs have been accused of intentionally degrading — or even blocking — VoIP traffic in order to limit network traffic and/or give priority treatment to favored IP telephony services. This issue, which sits directly at the center of today's network-neutrality debate, threatens to hurt IP telephony competition — and customer freedom of choice — by degrading the QoS of non-ISP-preferred VoIP services. In contrast, rather than relying on a third-party network, some business VoIP networks use private IP network backbones in order to guarantee service levels.