How to Improve VoIP Call Quality

Poor VoIP call quality — choppy audio, dropouts, “can you repeat that?” — is almost never about not having enough bandwidth. The real culprits are latency, jitter, and packet loss. This article explains what those mean, what targets your network needs to hit, and the specific changes (QoS, Ethernet, disabling SIP-ALG) that actually move the needle.

Recommended Network Requirements

Audio packets are small but extremely time-sensitive. The thresholds below describe the conditions a network needs to consistently maintain for good voice quality.

What Actually Helps

Use Ethernet, not Wi-Fi

Wi-Fi is convenient, but it is susceptible to interference and channel contention, which produce high latency, retransmissions, and packet loss — all enemies of clean audio. Wherever possible, run a Cat5e or better Ethernet cable to your phone or softphone host.

Disable SIP-ALG on your router

Many consumer-grade routers ship with a “SIP ALG” feature that tries to rewrite SIP traffic. In practice, it almost always corrupts SIP messages and creates one-way audio or registration drops. Disable it.

Consider TCP for the EMAK softphone

If you are deploying the EMAK softphone and your network has UDP issues, switching to TCP on a port other than 5060 sometimes helps. See the softphone setup guides for the exact setting.

Use a router with QoS to fight bufferbloat

The single biggest cause of jitter on a saturated home or office connection is bufferbloat — oversized buffers in the modem that queue traffic and inflate latency the moment the link gets busy (e.g., a backup or large download). A router with proper QoS keeps the upstream queue short and prioritizes voice packets, eliminating most of the jitter your ISP introduces.

Bufferbloat is not the only source of jitter, packet loss, and latency, but it is the most common — and the one you have direct control over.

Test Your Connection

Use the Waveform Bufferbloat & Internet Speed Test, which measures latency and jitter under load (which is when calls actually fail):

https://www.waveform.com/tools/bufferbloat

Sub-Par Connection (Bufferbloat D)

This connection is fast, but bufferbloat introduces large jitter under load — calls have audible cuts and pops.

Same Connection With QoS Enabled

Same line, same ISP — but routed through a router that enforces QoS. Bufferbloat is gone and call quality is excellent.

Slow but Clean Beats Fast but Bloated

An extreme example: a slower connection with low latency and Bufferbloat A produces dramatically better call quality than the high-bandwidth connection above with Bufferbloat D.

Still Bad — Even With Bufferbloat A or B?

If your test results look healthy but call quality is still poor, the problem may be elsewhere on your network or upstream. Send the following to [email protected] and we can dig in:

• The extension or phone number the call was placed from
• The extension or phone number that was dialed
• The time the call was placed
• The account used

Below is an example of a connection where jitter and packet loss are still problems despite reasonable speed:

Myth: More Bandwidth Means Better Calls

It does not — at least, not on any modern connection. A single voice call is roughly 64–128 Kbps; a 100 Mbps connection has more than enough headroom for hundreds of calls. Real-time audio depends on packets arriving quickly and consistently, not in bulk.

Latency vs. bandwidth: the carrier-pigeon analogy

Example 1. You send a carrier pigeon carrying a 1 TB USB drive to a friend’s house every hour. Your effective transfer rate is enormous — 2.2 Gbps — but your latency is one hour. You can move huge amounts of data, but you cannot have a conversation.

Example 2. You string a copper phone line between two houses. The line carries only 56 Kbps, yet your voice arrives near-instantly. A normal conversation works fine — sufficient (low) bandwidth with low, consistent latency.

Voice needs Example 2’s profile, not Example 1’s.

Jitter is worse than latency

Jitter is the variation in latency over time — packets arriving in inconsistent order or spacing. Consider this sentence with the word only in different positions:

• Only she told him that she loved him.
• She only told him that she loved him.
• She told only him that she loved him.
• She told him only that she loved him.
• She told him that only she loved him.
• She told him that she only loved him.
• She told him that she loved only him.
• She told him that she loved him only.

The meaning changes entirely depending on where only arrives. If the speaker said “She only told him that she loved him,” but the audio packets arrived out of order so the listener heard “She told him that she loved only him,” the result would be quite different.

To prevent this, audio buffers drop late-arriving packets so that what does play stays in order. That dropping is what produces most of the clicks, pops, and small audio gaps you hear on a jittery network.

VoIP Call Quality FAQ

How do I know if my router supports QoS?

Most stock ISP modems do not, or have very basic implementations. Routers running OpenWrt, pfSense/OPNsense, or modern prosumer routers (Ubiquiti, MikroTik, etc.) support proper FQ-CoDel or CAKE-based QoS. EMAK Telecom can recommend or supply a suitable model.

What is SIP-ALG and why is it bad?

SIP-ALG is a router feature that inspects and rewrites SIP packets passing through. In theory it helps with NAT; in practice it almost always mangles SIP messages, breaks registration, or causes one-way audio. Disable it on your router.

Should I open specific ports for VoIP?

You should not need to forward ports for outbound SIP/RTP. You do need to ensure your firewall is not blocking the EMAK Telecom IP ranges and ports — see EMAK firewall whitelist.

Wi-Fi works for everything else — why not phones?

VoIP exposes Wi-Fi’s weak points. A web page or video can buffer through a half-second of Wi-Fi interference; a live conversation cannot. Even on excellent Wi-Fi, Ethernet to your phone or softphone is the safer choice.

Does call quality depend on the other party’s network too?

Yes. Audio travels both directions, and jitter or loss on either side affects the call. If only certain remote parties sound bad, the issue may be on their network or with the route between you.