Ensuring a quality VoIP service

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By Rob Lith and David Meintjes dave_rob

This article examines the quality problems that are typical on voice over IP (VoIP) calls, the methods of detecting and countering them, and a solution to resolve VoIP quality issues.

This article examines the quality problems that are typical on voice over IP (VoIP) calls, the methods of detecting and countering them, and a solution to resolve VoIP quality issues.

Quality is a fundamental preoccupation of VoIP architects. The primary challenge is ensuring voice traffic is not delayed or otherwise compromised due to interference from non-voice traffic on the network. The concern is valid since internet protocol (IP) allows the transmission of multiple types of data, often simultaneously, on a best-effort (non-dedicated) network.

Bandwidth constraints

The internet connection always tops the list of factors affecting voice quality in VoIP conversations, since bandwidth is the key for voice quality on VoIP. A broadband connection will work well as long as it is not "bursty", doesn’t suffer high latency and isn’t shared with too many other communication applications.

It’s all about order

Following the convention of all data transmissions, voice traffic on an IP network is broken down into data packets that are again reassembled at the destination. But while the order in which packets arrive is immaterial in data applications like email or web downloads, voice coherency and quality depends very much on their timely arrival. When packets arrive out of order or late, quality issues occur. Typically, these and other packet problems manifest in the following quality issues:

  • Dropped packets: If the receiving router doesn’t have enough buffer space to store all the packets when they arrive, they may be dropped. Even if a request is issued for the missing packets to be re-sent, this could delay transmission.
  • Delay: Packets may be held up in server queues or sent via an alternative route and hence take longer to reach its destination – a tactic paradoxically undertaken to avoid congestion in the first place. A delay of more than 300 ms or more will affect the flow of conversation, while anything longer could render the service unusable.
  • Jitter: Because the fact and length of delay is unpredictable, variable delay known as jitter can seriously affect the quality of streaming audio and/or video.
  • Out-of-order delivery: When routing a collection of related packets through the Internet, their routes may, resulting in packets sometimes arriving out of order.
  • Error: Sometimes packets are misdirected, erroneously combined or corrupted en route. The receiver has to detect this and – just as if the packet was dropped – ask the sender to re-send.


The VoIP hardware equipment used can greatly affect your voice quality. The cheapest solutions are normally the poorest quality (but not always). It is always advisable to have as much information as possible on an analogue telephone adapter, router or IP phone before investing in it and starting to use it.

For these and other reasons, quality of VoIP can only be assured if the issue is tackled on a number of fronts, such as network choice, design, and software applications aimed at overcoming quality problems.

Measures to ensure VoIP quality

Experienced VoIP providers can employ a range of tactics to offer quality assurance, from infrastructural interventions including local area network (LAN) design and analysis, to managed interventions including monitoring of the solution, to a range of network applications to prevent and remedy quality issues. Of these, infrastructural preparation is the first and most fundamental.

Network infrastructure – LAN

The best possible VoIP quality is achieved over a network that is engineered to provide the necessary capacity and capability to bear the expected voice traffic of the company. LAN design capability considerations include the following:

  • Employ full duplex Ethernet connections (enabling simultaneous two-way transmission)
  • Enable auto-negotiation, to establish a match in transmission speed, duplex behaviour and required special protocols
  • Use Layer 2 or 3 switches
  • Put the correct transmission protocols in place (such as SMTP, an e-mail protocol), and provide enough bandwidth for applications using those protocols
  • Ensure power supply is reliable.

In addition, the network must be of high enough capacity (standard QoS techniques are not very effective for voice carried over links with throughput of less than 1000 Kbps), offer redundancy and adhere to other design principles to handle the customer’s traffic throughput requirements and ensure problems can be dealt with easily.

To achieve this, the provider should draw up a detailed network diagram, including the LAN and WAN (wide area network) – to understand traffic flows; networking devices like routers and switches – for monitoring and management; and the lengths of cables and circuits – to ensure strong signal propagation. A prolonged "soak test" will reveal the success or otherwise of an installation.


Network infrastructure – access

The IP-based access network of the enterprise must be adequately provisioned to handle voice, and also dedicated to voice traffic. Quite often, the access portion of the network will be a copper-based ADSL line.

Unlike dedicated leased lines (Diginet circuits in South Africa), DSL offers only a "best-effor" service. In such cases, the ADSL network over which the VoIP service runs must feature "un-contended" (dedicated) capacity, provided in the form of "IP connect" capacity – a service from Telkom, the South African telco operator – to VoIP providers.

But even in cases where IP connect is in place, local ADSL exchanges may be over-subscribed. In such cases, no matter the quality assurance that has been implemented on the client’s side, VoIP traffic will suffer unless alternatives are sought to ADSL, or additional measures are taken to protect the voice traffic on it.

The alternative to ADSL in such a case is a leased line, which significantly increases cost for the sake of guaranteed quality. This will be offset by the benefits of VoIP, which include call cost savings, infrastructure rationalisation and VoIP’s feature benefits of enabling collaboration, unified communications and business process integration. If this alternative is out of financial range for the customer, QoS can be pursued on ADSL in other ways.

Applications and tools for enabling quality of service

A wide array of mechanisms is available to test for, enable and remediate VoIP quality. Depending on the provider’s own technology supply portfolio, these may include:

  • Network and software features such as jitter buffering.
  • End-to-end QoS, by ensuring that traffic is treated consistently across the network. There are various methods for achieving QoS, including bandwidth reservation, traffic shaping, various forms of queueing (such as first-in, first-out or priority queueing), preferential forwarding, IP fragmentation, policy management and various forms of prioritisation (such as virtual LANs, IP address prioritisation, port-based prioritisation, application and network service classes and priority bit sets).
  • Special protocols to rearrange packets into the correct order once they reach their destination
  • QoS applications providing fully-redundant VoIP channels.

Managed service

In addition, a managed service, while not strictly speaking a quality-enhancing feature, can be effective in countering quality problems. VoIP providers should manage the entire service – from customer premises equipment through to the hosted PBX solution, if any.

VoIP quality solution: ViBE by VoIPex

VoIP call quality degrades quickly over low-bandwidth links, and lines can easily become over-subscribed. In countries such as South Africa, where bandwidth is at a premium, unchecked acquisition of more capacity will nullify the biggest benefit of VoIP – cost savings. Solutions that streamline traffic rather than drive up the cost of bandwidth are therefore in high demand. While compression can be one component to an overall solution, significant compression can also reduce call quality.

ViBE from VoIPex is gaining widespread recognition in South Africa and abroad. Its ability to strip away unnecessary data from voice packets has manifold benefits – both as a QoS and redundancy application.

How ViBE works

ViBE prioritises voice calls by stripping away unnecessary data overhead in packets (as much as 80% of a packet). In this way the technology allows sending an unprecedented number of VoIP calls over a data network (ADSL, 3G, wireless, ISDN or satellite). This happens within a dedicated voice connection over a data link, meaning VoIP calls are "insulated" from interference of non-voice data, avoiding typical problems such as jitter and latency. At the same time, non-voice data outside this dedicated connection is allowed to flow freely. ViBE calls are therefore always of business class quality without compromising other data activities.

Without ViBE, three to four voice calls can be carried over an ADSL link with 256 K uplink speed (with voice being full duplex the maximum speed available for voice is the uplink speed). With ViBE on a standard ADSL line with 256 Kbps upload speed, 28 coexistent calls along with other data – such as email and web browsing – are possible. With multiple IP link bonding, this can be doubled.

IP-based voice communication for business use has made aggressive strides forward while overcoming some of its past barriers and misconceptions. Today, VoIP technology coupled with a quality-of-service enabled network can provide businesses with enterprise-grade reliability and high-definition voice quality at a lower cost.


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