VoIP Call Quality: Why Some Calls Sound Great and Others Sound Terrible

“VoIP sounds terrible” is a sentence I still hear from people whose last experience with internet calling was Skype in 2012 on hotel WiFi. And yeah — that did sound terrible. But modern VoIP on a decent internet connection sounds better than a traditional phone line. Noticeably better.

The gap between “crystal clear” and “sounds like you’re calling from a submarine” usually comes down to a handful of factors that are mostly fixable. You don’t need a networking degree. You just need to understand what’s actually happening when voice travels over the internet.

How VoIP Audio Works (Quick Version)

When you talk on a VoIP call, here’s what happens in the roughly 20 milliseconds between you saying a word and the other person hearing it:

1. Your microphone captures the sound — analog sound waves become a digital signal
2. The codec compresses it — the audio gets compressed to reduce the amount of data that needs to travel over the network
3. It gets chopped into packets — the compressed audio is split into tiny chunks (packets), each representing about 20ms of audio
4. Packets travel across the internet — through your router, your ISP, various internet backbone networks, the recipient’s ISP, and their router
5. The other end reassembles them — packets get reordered (they don’t always arrive in sequence), decompressed, and played through the speaker

When everything works, this process is seamless. When it doesn’t, you get choppy audio, robotic voices, echoes, delays, or one-way audio.

The Three Enemies of Call Quality

1. Jitter

Jitter is the variation in packet arrival times. Ideally, audio packets arrive at perfectly regular intervals — every 20ms, like clockwork. In reality, internet traffic doesn’t work that way. One packet takes 15ms, the next takes 45ms, the one after that takes 22ms.

Your VoIP app has a “jitter buffer” that absorbs this variation by holding packets briefly and playing them back at a steady rate. Small amounts of jitter (under 30ms) are handled transparently. Once jitter exceeds 50ms, the buffer can’t compensate anymore and you start hearing artifacts — choppy audio, brief dropouts, or words that sound like they’re being chewed up.

Acceptable jitter: Under 30ms Noticeable quality degradation: 30-50ms Calls become frustrating: Above 50ms

2. Packet loss

Sometimes packets don’t arrive at all. They get dropped by a congested router, lost in transit, or arrive so late they’re useless. VoIP can tolerate a small amount of packet loss — the codec can interpolate and fill in brief gaps. But once you’re losing more than 1-2% of packets, the audio quality falls off fast.

At 1% packet loss, you might notice the occasional “glitch” — a word slightly garbled. At 3%, conversations become difficult. At 5%+, the call is essentially unusable.

Acceptable: Under 1% Noticeable: 1-3% Unusable: Above 5%

3. Latency

Latency is the total time it takes for audio to travel from your mouth to the other person’s ear. Some latency is unavoidable — the speed of light through fiber optic cables, plus processing time at each hop. Domestic calls typically have 50-100ms of latency, which is fine. International calls might hit 150-250ms, which creates the slight awkwardness where both people start talking at the same time.

Imperceptible: Under 100ms Slightly awkward: 100-200ms Conversation becomes difficult: Above 300ms

Codecs: The Audio Quality Gatekeepers

A codec (coder-decoder) determines how your voice is compressed for transmission. Different codecs make different tradeoffs between audio quality and bandwidth usage.

Opus is the standout. It adapts its bitrate in real time based on available bandwidth. On a strong connection, it delivers full HD audio that sounds better than a landline. When the network gets congested, it gracefully reduces quality to maintain the call rather than dropping it. It also handles packet loss better than older codecs by building in redundancy.

VestaCall uses Opus as the default codec with automatic fallback to G.722 and G.711 when connecting to systems that don’t support Opus. This means you get the best possible quality your connection can handle, without any manual configuration.

The practical difference between G.711 (standard) and Opus (HD) is significant — it’s like the difference between AM radio and FM radio. Voices sound fuller, clearer, and more natural. Background noise is better filtered. You can actually hear subtle vocal cues that get lost in narrowband audio.

Why Your Calls Might Sound Bad (And How to Fix It)

Let me walk through the most common problems and their fixes, from easiest to hardest:

Problem: Calls sound fine in the morning but terrible in the afternoon

Cause: Network congestion. Other people and devices on your network are using more bandwidth in the afternoon — video calls, streaming, file transfers.

Fix: Enable Quality of Service (QoS) on your router. QoS prioritizes voice traffic over other types of data. Even when your teenager is streaming in 4K, your VoIP calls get first dibs on bandwidth. Most modern routers have QoS settings in their admin panel — look for it under “Traffic Management” or “Bandwidth Prioritization.”

Problem: Calls are choppy on WiFi but fine when wired

Cause: WiFi introduces jitter and packet loss that wired connections don’t. Wireless signals compete with other devices on the same frequency, get disrupted by physical obstacles (walls, floors, appliances), and degrade with distance from the router.

Fix: Use a wired Ethernet connection. Seriously — this single change fixes about 60% of all VoIP quality complaints. If wired isn’t possible, use 5GHz WiFi instead of 2.4GHz (less interference, faster), sit within 15 feet of the router, and make sure you’re not competing with 15 other devices on the same access point.

Problem: Echo on calls

Cause: Usually a hardware issue, not a network issue. The other person’s speaker output is being picked up by their microphone and sent back to you.

Fix: Use a headset. This physically separates the speaker from the microphone and eliminates echo. If using a speakerphone, make sure acoustic echo cancellation (AEC) is enabled in the VoIP app settings.

Problem: One-way audio (you can hear them but they can’t hear you, or vice versa)

Cause: Firewall or NAT traversal issues. Your network’s firewall is blocking the media stream in one direction.

Fix: Make sure your firewall allows UDP traffic on the ports your VoIP provider uses (VestaCall uses standard SIP/RTP ports). Alternatively, a VoIP provider that uses WebRTC-based calling often avoids firewall issues entirely because the traffic looks like regular web traffic to your firewall.

Problem: Calls are consistently poor quality regardless of fixes

Cause: Your ISP’s connection has fundamental quality issues — high baseline jitter, frequent packet loss, or insufficient upload speed.

Fix: Run a VoIP-specific speed test (not just a regular speed test — you need one that measures jitter and packet loss, not just throughput). If the numbers consistently show jitter above 30ms or packet loss above 1%, contact your ISP. If that doesn’t help, consider a dedicated internet connection or an SD-WAN solution for your office.

VestaCall’s Approach to Call Quality

We take quality pretty seriously because honestly, if the calls don’t sound good, nothing else matters. Here’s what we do on our end:

• Opus codec by default with adaptive bitrate — best quality your connection can support
• Global network with data centers in 15 regions, so calls take the shortest path
• Jitter buffer optimization — automatically tuned per call based on observed network conditions
• Redundant routing — if one network path degrades, calls are rerouted in real time
• 99.999% uptime SLA — backed by our infrastructure, not just a marketing number

The result: HD voice quality on virtually any modern internet connection. Most of our customers tell us calls sound noticeably better than their previous phone system. Some don’t notice a difference — which, for VoIP, is actually a compliment. It means it just works.

Try it yourself — 14-day free trial, no credit card required. If the call quality doesn’t meet your standards, nothing else about the platform matters anyway.

Sarah Chen

Head of Product, VestaCall