VoIP (Voice over Internet Protocol) has come a long way since Skype brought it to the mainstream consumer market in 2003. Once a cool way to make free internet calls, VoIP has evolved into a communication powerhouse, reshaping how businesses connect and collaborate.
Companies of all sizes are jumping on the VoIP bandwagon, ditching traditional phone systems for this more flexible, cost-effective alternative. It’s not just about saving money though; VoIP plays well with other digital tools, making it a key layer in the modern office tech stack. The cloud has also given VoIP a major boost, with cloud-based VoIP solutions offering businesses unparalleled flexibility to scale up quickly or adapt to changing needs without a tech headache. Perhaps where VoIP has shone most brightly in recent years is the work-from-home revolution, when keeping teams connected across living rooms, home offices, and kitchen tables became crucial almost overnight.
And while businesses are loving VoIP, everyday users are changing the game too. Messaging apps like WhatsApp, Facebook Messenger and FaceTime are all sneaking VoIP into our daily lives, often without us even realizing it. Voice calls, video chats, funny cat videos—it’s all riding on VoIP technology.
For all its advantages, VoiP has one caveat. Because it depends on real-time, two-way communication, VoiP is sensitive to network issues and requires consistent, low latency and low jitter to maintain call quality. In this increasingly competitive market, telecoms can’t afford to be the ones dropping calls or sounding like they’re transmitting from the bottom of a fish tank. Customers expect crystal-clear conversations, and one too many “Can you hear me now?” moments can send customers running to the competition. To step up their game, telecoms must embrace best practices that include optimizing networks, implementing fancy codecs to make voices sound good, and prioritizing voice traffic.
Smart telecoms are also playing the long game—educating customers on the best ways to set up and use their VoIP systems. In the end, great call quality isn’t just about being heard—it’s about being the one that everyone wants to listen to.
For all its advantages, VoiP has one caveat. Because it depends on real-time, two-way communication, VoiP is sensitive to network issues and requires consistent, low latency and low jitter to maintain call quality.
Common VoIP Call Quality Issues and How Telecoms Can Help
One of the leading causes of poor VoIP call quality is network bottlenecks, where insufficient bandwidth or congested routers delay packet transmission. These bottlenecks can manifest as choppy audio, where parts of the conversation are cut out, echo/feedback, where users hear their own voices looping back with a delay, and dropped calls, which are particularly disruptive in business settings, affecting productivity and damaging client relationships.
Unlike streaming, which can pre-load content into a buffer and allows for smoother playback even with some network fluctuations, VoIP requires immediate transmission and reception of voice data packets to maintain natural conversation flow.
- High latency (>150ms) can cause noticeable delays in conversation, making communication difficult due to “talk over”.
- Jitter can cause choppy or distorted audio, severely impacting call quality.
- Even small packet loss (1-2%) can result in noticeable audio quality degradation.
- In terms of bandwidth, VoIP requires consistent, but relatively low bandwidth (64-100 kbps per call). For comparison, streaming often requires higher overall bandwidth but can adapt to available network conditions.
The most common way for telecoms to measure overall voice quality is by determining MOS Score (Mean Opinion Score). MOS is a subjective metric that uses a group of people or an Objective Measurement Method (R-factor) that approximates human ranking to rate the perceived quality of audio samples on a scale from 1 (bad) to 5 (excellent). Standardized by the International Telecommunications Union, MOS has been used for decades as the leading metric to measure overall voice quality. For VoIP the MOS score can be calculated from the transmission line latency, jitter, and packet loss based on ITU-T G.107 standard.
Telecoms play a critical role in supporting VoIP reliability by providing the infrastructure and services needed for smooth call quality. They ensure that network paths are optimized for voice traffic, employ Quality of Service (QoS) mechanisms, and manage bandwidth allocations to prioritize VoIP packets over less critical data streams. Telecom providers also offer dedicated SIP trunks or private links to businesses, helping reduce latency and packet loss by avoiding congested public networks. Their ability to manage interconnections and provide redundancy significantly enhances VoIP call reliability, reducing the likelihood of disruptions that stem from external network issues.
How to Improve VoIP Call Quality: Best Practices
To ensure high-quality VoIP calls, it’s critical to address these underlying causes through various best practices.
Optimize Network Performance
The foundation of reliable VoIP call quality is a well-optimized network. Even though the bandwidth requirement per call is low, ensuring sufficient bandwidth is key. Network monitoring tools should be used to detect potential bottlenecks and ensure traffic is flowing smoothly. Without proper bandwidth allocation and continuous oversight, businesses may experience voice degradation that can disrupt communications.
Use Quality of Service (QoS) Policies
QoS policies are vital for prioritizing voice traffic, which is sensitive to latency, jitter, and packet loss. By configuring routers and switches to prioritize VoIP data packets, QoS minimizes delays and ensures that voice traffic is given precedence over less time-sensitive data. This results in clearer, uninterrupted calls and improved overall call performance.
Upgrade Network Equipment
Outdated routers, switches, and VoIP devices can introduce inefficiencies that impact call quality. Telecoms and businesses should regularly upgrade their network equipment to support higher speeds and more advanced protocols. Modern hardware can handle increased data loads more effectively, which helps reduce issues like jitter and packet loss.
Leverage SD-WAN Technology
SD-WAN (Software-Defined Wide Area Network) technology enhances VoIP performance by dynamically selecting the best available network paths for voice traffic. This technology not only improves call stability by rerouting traffic away from congested networks but also reduces packet loss and latency SD-WAN also allows for the efficient use of multiple network links, combining MPLS, broadband, and 4G/5G connections, to ensure consistent call quality across different locations, including remote offices.
A note on SD-WAN limitations
While SD-WAN can significantly improve performance in the first-mile (client-side) and last-mile (end-user-side) by optimizing how traffic enters and exits the network, it does have limitations when it comes to managing the mid-mile—the segment of the network controlled by third-party carriers. Once traffic leaves the enterprise or service provider’s direct control, it is subject to the performance of intermediate networks, which SD-WAN cannot directly influence. Issues such as carrier network congestion, route inefficiencies, or unexpected latency can arise in the mid-mile, potentially degrading VoIP performance despite SD-WAN’s optimizations on the edges of the network.
To mitigate these limitations, telecom providers must ensure that the mid-mile traffic they handle is optimized for real-time communication.
Collaborate with Clients to Implement VoIP Best Practices
Effective collaboration between service providers and clients ensures that VoIP systems are optimized for performance and reliability. Providers should offer tailored recommendations for network upgrades, bandwidth adjustments, and security configurations that align with the client’s specific communication needs. Ongoing education and support are also critical to ensuring clients are aware of the best practices for VoIP maintenance. Additionally, businesses may benefit from routine health checks of their VoIP infrastructure, where providers evaluate call quality metrics and make necessary adjustments. A proactive, collaborative approach helps clients maintain the highest level of performance and reduce disruptions in communication.
Regularly Updating VoIP Software and Hardware
Keeping VoIP software and hardware up to date is essential for ensuring compatibility with new features, security patches, and network technologies. Outdated systems can experience compatibility issues, performance lags, or vulnerabilities that compromise call quality and security. Regular updates ensure that VoIP systems operate with the latest protocols and enhancements, providing smoother calls and better overall performance. Telecoms and IT teams should set up automated updates or a routine schedule for checking and installing the latest versions of both hardware firmware and VoIP applications.
Fine-Tuning Router and Firewall Settings for VoIP
Properly configuring routers and firewalls for VoIP traffic is crucial to prevent common issues such as dropped calls or blocked connections. VoIP traffic often needs specific Quality of Service (QoS) settings to prioritize voice packets over other types of network traffic. Without proper prioritization, voice data can be delayed or lost, resulting in poor call quality. Additionally, firewall settings need to be adjusted to allow VoIP protocols (such as SIP and RTP) to pass through without being blocked. Telecoms and IT professionals should work closely with clients to fine-tune these settings to balance security with performance.
Using Ethernet Instead of Wi-Fi for Stable Connections
One of the simplest yet most effective ways to improve VoIP call quality is to use wired Ethernet connections instead of relying on Wi-Fi. While Wi-Fi is convenient, it is prone to interference, signal drops, and inconsistent bandwidth, all of which can severely impact VoIP performance. Ethernet connections offer more stable, consistent data transmission, reducing latency and packet loss that commonly affect call quality over wireless networks. Telecoms should encourage clients to use wired connections for VoIP whenever possible, especially in environments where stable communication is critical, such as call centers or customer service departments.
Enhancing Codec Selection for Voice Clarity
The choice of codec—the algorithm used to compress and decompress voice data—has a significant impact on VoIP call quality. Different codecs vary in terms of compression levels, bandwidth usage, and audio quality. For instance, codecs like G.711 offer high voice clarity but require more bandwidth, while codecs like G.729 use less bandwidth but may compress audio to the point of degrading quality. Telecoms should work with clients to determine the most suitable codec based on their network capacity and desired audio quality. By selecting the right codec, clients can achieve clearer, more natural-sounding voice communications without overwhelming their network resources.
How Teridion Helps Telecoms Enhance VoIP Call Quality
Teridion’s private global backbone network uses AI-powered route detection to dynamically find the fastest and most stable routes while in transit to ensure fast, reliable performance in a cloud-first world. As an overlay network, Teridion gives telecoms the ability to utilize peering agreements when the internet performs optimally, but once it detects that performance has dropped below the SLA threshold, telecoms can apply real-time traffic steering across middle and last mile for full network control.
However, not always the lower latency path is better, and sometimes the best path has higher latency but far better jitter and packet loss, resulting in an overall better MOS score despite the higher latency. The Teridion dynamic path selection at the middle-mile plus the performance based selection at the last mile provide targeted optimization to any location and with any edge device with the right combination of latency, jitter, and packet loss to ensure consistently high MOS scores.
Using Teridion’s global reach and intelligent routing, telecoms can promise and deliver a high-quality VoIP experience with a network that adapts to the speed of business.
