When a VoIP caller sounds underwater or muffled, the problem is almost always in the network. The call itself connects and stays active, but voice data packets arrive out of order, too late, or not at all, and the audio degrades in ways that are harder to diagnose than a simple dropped call.

The symptoms usually point to one of four causes: insufficient bandwidth, high latency, excessive jitter, or faulty hardware. If you're experiencing any of the following, this guide covers how to identify and fix the source:

• Callers that sound like they're in a tunnel
• Calls that are garbled or have an echo
• Inbound or outbound audio loss
• Audio delay or call latency

Unlike a dropped call, degraded audio is often attributed to the business rather than the technology. Clients who struggle to hear you lose confidence quickly, and in a business context, unlike a personal call, there's usually no second chance to make that impression. If your calls are choppy or distorted, the fix starts with understanding which layer of your network is failing.

What is VoIP muffled audio?

VoIP muffled audio, often described as callers sounding underwater or garbled, occurs when network issues disrupt the quality of voice data packets during transmission, causing degraded or unintelligible audio on either end of the call.

Common causes of VoIP muffled audio

VoIP muffled audio is almost always a network-layer problem. The four most consistent causes are listed below, and identifying which one applies is the first step toward a fix.

Insufficient bandwidth

VoIP calls require a minimum of 100 kbps upload and download per simultaneous call. When available bandwidth drops below that threshold, voice data packets are compressed, dropped, or delayed, resulting in the muffled or underwater sound quality that callers notice. Bandwidth problems tend to worsen during peak usage hours when other applications compete for the same connection.

High latency

Latency is the time it takes for a voice packet to travel from sender to receiver. VoIP calls require latency below 150ms for clear audio. Above that threshold, packets arrive too late to be reassembled in sequence, and the audio degrades into a muffled or robotic quality. High latency is commonly caused by network congestion, long routing paths, or an overloaded router.

Excessive jitter

Jitter is variation in packet delivery timing. Even when average latency is acceptable, inconsistent packet arrival causes the VoIP system to struggle reassembling audio in the correct order. Jitter above 30ms produces the fragmented, muffled, or underwater effect. A jitter buffer can compensate for minor variation but cannot correct severe or sustained jitter.

Faulty hardware or loose connections

A loose Ethernet cable, an aging modem, or an unnecessary extra router in the signal path can all introduce packet loss that degrades audio. VoIP networks perform best with direct, minimal-hop connections. Any unnecessary routing step adds latency and increases the chance of packet loss, which can cause muffled or distorted audio.

How to fix VoIP muffled audio

Once you have identified the likely cause, work through the five steps below in order. Start with the network test, then inspect hardware, and work through the remaining configuration checks until the problem clears.

Test the network with diagnostic tools

Start by running a speed test through your VoIP provider's built-in diagnostics or your ISP's tools. You need to confirm that your connection meets VoIP minimums: at least 100 kbps upload and download per simultaneous call, latency under 150ms, jitter under 30ms, and packet loss as close to zero as possible. There are also free online speed test tools if your provider doesn't offer them.

Check all four metrics, not just download speed. A fast connection with high jitter or elevated latency will still produce muffled audio. If any metric is out of range, that is your likely cause. Some of the most reliable free tools for testing VoIP-specific network quality include:

1. Speedtest by Ookla
2. SolarWinds tools
3. DNSstuff free tools
4. Ping-test tool

If the tests reveal insufficient bandwidth, contact your ISP about upgrading to a business-class internet plan. Consumer plans with variable performance are a common source of VoIP quality problems that don't consistently show up.

Double-check equipment and connections

The hardware connections are the next thing on your list. Even the slightest loose wire or an unnecessary extra router can cause a lot of quality issues with your calls. VoIP networks work best with fewer connections and the most direct routing possible. You'll want to check all of the routers, modems, Ethernet cables, and other connections on your network. This includes computers, as well as the connections going to them.

Reset your modem as the first line of defense against latency, echoes, and callers sounding underwater. Sometimes, a simple network reset can go a long way. While you're inspecting all of your equipment, look for places where you can shorten the journey or eliminate waypoints that could slow your network connection.

Finally, check the specs of your hardware. How old are your devices? Does your computer need to be upgraded to handle the VoIP bandwidth and speed requirements? Perhaps your modem or router is outdated and needs to be replaced. Although it's usually one of the last things you'll want to consider, it may just be time to get new networking hardware or upgrade your computers.

Enable Quality of Service on your router

Quality of Service (QoS) is a router setting that assigns priority to different types of network traffic. When VoIP packets compete with video streams, file downloads, or software updates for bandwidth, QoS ensures that voice traffic takes priority. Without it, a large file transfer can spike latency or jitter enough to produce the muffled, underwater effect mid-call.

To enable QoS, log into your router's admin panel and look for QoS or traffic prioritization settings. Set VoIP traffic to high priority using the UDP ports your provider specifies, typically port 5060 for SIP signaling and ports 10000 to 20000 for RTP audio. Most business-grade routers support this natively. If yours does not, it may be time to upgrade to one that does, since QoS is one of the most reliable fixes for sustained jitter and latency problems.

Switch to a wired connection

Wi-Fi introduces signal interference, channel congestion, and inconsistent packet timing, all of which degrade VoIP audio. A wireless connection that benchmarks well on a speed test can still produce jitter above 30ms during actual use, which is enough to cause muffled or garbled audio.

Switching to a wired Ethernet connection removes that variable entirely. If running a cable to a VoIP phone or computer is not practical, a powerline adapter can use your building's electrical wiring to carry a wired signal across rooms. The result is a more stable connection without the latency spikes that wireless introduces, particularly in environments with multiple devices on the same network.

Check your codec settings

A VoIP codec is the algorithm that compresses and decompresses voice data for transmission. When the codecs on both ends of a call do not match, the audio degrades to the muffled, robotic, or choppy quality callers notice. Codec mismatches are common after provider changes, software updates, or when calling across different VoIP systems.

Log into your VoIP phone or softphone settings and review the active codec. G.711 offers the best audio quality at around 87 kbps and is the right choice when bandwidth is not a constraint. 

G.729 compresses audio down to around 31 kbps, which is useful on limited connections but produces lower fidelity. If your provider supports G.722, it delivers wideband audio that sounds noticeably clearer than either of the others. 

Set both ends of the call to use the same codec, preferably G.711 unless bandwidth forces a compromise.

When outsourcing makes more sense than fixing

If the root cause is aging infrastructure or a network that can't reliably meet VoIP minimums, fixing the problem may cost more in time and hardware than it saves. In those cases, removing the VoIP dependency entirely is often the cleaner solution.

The AI Receptionist at Smith.ai answers calls 24/7 on its own infrastructure, so call quality is never your problem to manage. For businesses that want human judgment on every call, the Virtual Receptionist service puts trained professionals on the line who handle lead intake and appointment scheduling directly.

Book a free consultation to find the right coverage for your business.

Related reading:

• Choppy VoIP calls fix
• VoIP call latency
• Caller can't hear you
• Best VoIP phone systems
• What is VoIP?