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Effects of Network Conditions on Voice Quality


The Internet provides "best-effort" data delivery. Best-effort IP allows the complexity to stay in the end-hosts, so the network can remain relatively simple. As more devices are connected, network service demands eventually exceed capacity, but service is not denied. Instead, it degrades gracefully. Although the resulting variability in delivery delays (jitter) and packet loss do not adversely affect typical Internet applications--email, file transfer, and Web applications-other applications cannot adapt to inconsistent service levels. Delivery delays cause problems for applications with real-time requirements, such as those that deliver multimedia, the most demanding of which are two-way applications like VoIP.

Increasing bandwidth is a necessary first step for accommodating these real-time applications, but it is still not enough to avoid jitter during traffic bursts. Even on a relatively unloaded network, delivery delays can vary enough to continue to adversely affect real-time applications. To provide adequate service -- some level of quantitative or qualitative determinism -- IP services must be supplemented. This requires adding some "smarts" to the network to distinguish traffic with strict timing requirements from those that can tolerate delay, jitter, and loss. That is what Quality of Service (QoS) protocols are designed to do. QoS does not create bandwidth but manages it so it is used more effectively to meet the wide range of application requirements. The goal of QoS is to provide some level of predictability and control beyond the current "best-effort" service.

A number of QoS protocols have been created to support a variety of application needs. The challenge of QoS technologies is to provide differentiated delivery services for individual flows or aggregates without breaking the Internet in the process. Adding "smarts" to the Internet and improving on "best-effort" service represents a fundamental change to the design of the Internet itself. This would also require a "standardization" of all QoS implementations. 

Quality of Service (QoS)

Quality of Service is defined as a given data stream priority over a network (WAN, LAN etc.). There is more than one way to characterize Quality of Service (QoS). Generally speaking, QoS is the ability of a network device to provide some level of assurance for consistent network data delivery. Some applications are more stringent about their QoS requirements than others, and for this reason, we are going to focus on Prioritization (differentiated services): network traffic is classified and apportioned network resources according to bandwidth management policy criteria. To enable QoS, network devices give preferential treatment to classifications identified as having more demanding requirements. QoS is only implemented (automatically by network devices) when the network is at or near 100% utilization.

Note-Icon.png Note: QoS does NOT apply to the "open internet."
eye.png Tip:
For more information on 8x8's recommendations for implementing QoS, and how 8x8 software and devices implement QoS, see our X Series Technical Requirements.

Things to Consider

  • Latency: Delay for packet delivery
  • Jitter: Variations in delay of packet delivery
  • Packet loss: Too much traffic in the network causes the network to drop packets
  • Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts

Packet Loss

VoIP is not tolerant of packet loss. Even 3% packet loss can "significantly degrade" a VoIP call using the default codec of G.711 (or G.722).  G.729 codec requires packet loss of far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP. 


This is the time it takes to move voice packets from the phone to 8x8 (and back). In general, this should not exceed 150ms in one direction to prevent deterioration of voice quality. 


Jitter is a variation in packet transit delay caused by queuing, contention, and serialization effects on the path through the network. In general, higher levels of jitter are more likely to occur on either slow or heavily congested links. This is the variation in packet delay. As far as the source endpoint is concerned, the packets would be sent in a continuous stream. But since each packet may take a different route to its destination, network congestion or improper configuration can result in variations in packet delay. If the packets are not received in the same order or were dropped entirely on the way, this results in Jitter. Jitter that exceeds 20ms will cause severe voice quality issues. High levels of jitter are usually a consequence of slow speeds or congested networks.

8x8 Recommended Maximum Acceptable Limits

  • Packet Loss Less than 0.3
  • Jitter Rate Less than 20
  • Latency Less than 120 ms

Differentiated Services (DiffServ)

Differentiated Services [DiffServ] provides a simple and coarse method of classifying services of various applications. 

  • Expedited Forwarding (EF): Has a single codepoint (DiffServ value). EF minimizes delay and jitter and provides the highest level of aggregate quality of service. Any traffic that exceeds the traffic profile (which is defined by local policy) is discarded [DiffServ EF].
  • Assured Forwarding (AF): Has four classes and three drop precedences within each class (so a total of twelve codepoints). Excess AF traffic is not delivered with as high a probability as the traffic "within the profile," which means it may be demoted but not necessarily dropped [DiffServ AF]

What is the result?


Consistently high latency can slow down conversations and also lead to the dreaded 'talk over' effect where one speaker interrupts the other unknowingly. Latency can also cause echoes making it hard for the listener to distinguish between spoken words. High latency can cause severe problems in normal one-to-one calls but these issues can be exacerbated during multiparty video or conference calls. It can cause the audio to be out of sync with the video which can quickly derail project meetings.

Packet Loss

Packet loss results in "lost" parts of the conversation so words or parts of words are lost/trimmed. Packet loss in VoIP is typically a slowly degrading impact on Voice Quality. The human ear is very good at handling short gaps that are typical of packet loss. So it may take a significant amount of packet loss for the user community to be annoyed enough to report it. Speech accents are typically are less forgiving of packet loss. 


Under optimal conditions, voice packets are sent in a steady stream at a constant rate, VoIP jitter occurs when the even spacing of packets is disrupted, causing: Delays, and distortion, or even "packet loss" if the delay is too long.

Additional Information

To test for and identify issues related to Packet Loss, Jitter and Latency 8x8 provides the Network Utility Test. For details on using the test and reading the results, see:

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