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Latency for VoIP


Voice over Internet Protocol (VoIP) leverages advanced digital compression enabling phone calls transmitted over the internet rather than traditional analog telephone lines. VoIP brings cost efficiencies and mobility advantages empowering modern telecommuting and globalized lifestyles. However, reliability remains contingent on consistent internet connectivity able to preserve real-time call quality sensitive to transmission lags known as latency.

This guide explores the concepts around VoIP network latency, quantifying its impact on user experience and outlining actionable tactics for benchmarking and reducing latency and jitter to uphold robust VoIP performance rivaling conventional phone infrastructure.

Understanding Network Latency

Defining Latency – In networking contexts, latency refers to delays between the initiation of data transfers and its eventual arrival at a destination measured in milliseconds (ms). Latency determines the responsiveness of network connections.

VoIP Latency Causes – Myriad technical factors introduce latency across internet pathways underlying VoIP call infrastructure:

  • Physical distance and speed of light/electricity lifespan needed to traverse cables and infrastructure induces propagation delay.
  • Network congestion where excessive competing traffic results in queuing backup adds latency.
  • Suboptimal routing decisions force indirect paths increasing transit times.
  • Inherent codec encoding and decoding processing time causes serialization delay along each network hop.
  • Faulty network gear or misconfigured firewalls blocking traffic and requiring repeat transmissions generates further lag at each waypoint.

Measuring Latency – Most latency benchmarks quantify one-way delay between client and server locations accounting for compounding lag. Common metrics include:

  • Round-trip time (RTT) averaging relays between endpoints.
  • Jitter or fluctuations indicating inconsistent latency.
  • Packet loss showing severe network deterioration losing call data bursts.

Managed network monitoring tools continuously test endpoint connections tracking performance trends managing acceptable 150 ms thresholds for VoIP reliability, where sub 75ms upholds HD call clarity.

Effects of Latency on VoIP

High VoIP latency degrades audio quality with distorted, robotic voice reproduction full of gaps or echoes making conversations unintelligible. Studies confirm customers perceive under 150ms latency as essentially real-time for natural call flows free of disruptions.

However, degraded quality from excessive latency quickly causes user frustration reflected in heightened churn rates and lost revenue if businesses cannot sustain call integrity. Each call drop averages $56 in lost productivity and IT troubleshooting burdens for enterprises.

Proactive monitoring ensures networks avoid breaching 150ms ceilings indicating infrastructure upgrades needed or misconfigurations causing unexpected bottlenecking evading standard redundancy. IT teams must quantify user latency exposure through active testing confirming where organizations fall on performance spectrums and how upgrades confer measurable gains.

Addressing VoIP Performance Issues

Myriad variables subtlety influence VoIP call excellence but broadly affect quality dimensions – delay, jitter, packet loss:

Delays manifest between initial signal transmission and eventual recipient processing from intermediary queuing lags. Solutions involve optimizing routing policies and upgrading overloaded links. Types include:

  • Serialization delay from voice encoding/decoding and packetization.
  • Network delays induced by congestion and suboptimal path directing.
  • Blocked traffic and retransmissions generating switch/router handling delays.

Jitter signifies uneven packet arrival spans disrupting real-time reconstruction with gaps or echoes. Improvements stem from Quality of Service (QoS) prioritization configurations favoring voice packets and added buffering smoothing playback.

Packet Loss completely omits frames of call data that then cannot contribute to conversational continuity. Fixes require boosting reliability and redundancy of delivery mechanisms to limit outage risks.

Supplementary performance issues like echo path modeling, PCM companding functions, smart endpoint resource adaptations through voice activity detection (VAD) all tuning VoIP network resiliency further demonstrate the intricate interdependent mechanisms underpinning call optimization efforts striving for indistinguishable comparisons to analog calls.

Causes of High VoIP Latency

Three predominant phenomenon cement VoIP latency emerging as a performance barrier:

  • Slow transmission or holding up of voice data packets moving through networks prevents timely delivery critical for real-time clarity. Complex encapsulation and encoding processes also add marginal delays.
  • Failed dispatched or out of sequence packets eventually expire if arrival exceeds session time-outs failing call reconstruction. Such impediments stem from network capacity constraints, intermittent ISP service degraded Quality of Service, faulty endpoint resources and poor Cabling.
  • Subpar internet connections lacking ample speed and stability for smooth multimedia transfers generates high latency and jitter destabilizing VoIP streams through issues like excessive hops andPoints of failure without redundancy.

Careful assessments pinpointing weak infrastructure links or software bottlenecks inform calculated upgrades strategically alleviating identified constraints impeding seamless VoIP connectivity

Reducing VoIP Latency

Robust Internet – Most latency solutions start with procuring enterprise-grade low-latency internet connectivity to access reliable backbone networks ensuring quality routing. Multi-carrier LAN/WAN links supply contingency coverage limiting downtime risks during an outage.

Traffic Prioritization – Next, enable QoS monitoring controls favoring voice and video packets through passive monitoring or DSCP tagging defining high-precedence treatment minimizing jitter and delays for latency-sensitive applications despite bandwidth competition. SD-WAN software simplifies configurations.

Proactive Testing – Continually quantify user latency exposure through active monitoring diagnosing network shortfalls before call quality suffers while validating infrastructure changes that enhance metrics. Routine speed tests checking major network segments distinguish localized as opposed to ISP-stemming lags.

Remedial Initiatives – Once diagnosing performancedrag culprits from quantifiable testing insights, undertake tactical infrastructure changes like added redundancy equipment, MPLS overlays protecting routing, SIP ALG firewall adjustments and VPN encryption securing transport flows for private preservation across public external networks.


While often overlooked, latent network delays profoundly shape VoIP call satisfaction equivalent to analog lines. As nanosecond-sensitive voice packets traverse convoluted internet pathways and intermediary hardware, managing pernicious lag time through research-driven understanding alongside purposeful infrastructure investments pays dividends ensuring enterprise VoIP solutions uphold reliability expectations alongside all the functionality and cost advantages over antiquated telephony.