Why Your VoIP Calls Sound Bad (And Whose Fault It Is)

Nearly every case of VoIP call quality issues traces to one of three causes: jitter, packet loss, or a router rewriting SIP traffic. Robotic or warbling audio is jitter. Clipped, dropped-syllable audio is packet loss. Audio in one direction only is almost always SIP ALG or NAT. Two of the three live on your network.

That last point is not a deflection. It is why a provider's status page can be green while your calls sound terrible.

What do the symptoms actually mean?

Voice travels as a stream of small packets — with the G.711 codec most business VoIP uses, one every 20 milliseconds, so 50 packets per second each direction. Every quality complaint is a story about what happened to those packets.

Notice how few rows say "provider." That reflects where these problems actually originate.

Jitter: why uneven arrival sounds robotic

Jitter is variation in the arrival time of packets sent at a constant rate. The stream leaves every 20 ms; if it arrives at 18 ms, then 41 ms, then 12 ms, the receiving phone must reassemble even audio out of an uneven feed.

Phones absorb this with a jitter buffer, typically 30 to 50 ms of audio. That works until variation exceeds the buffer. Then the phone has nothing to play at the moment it must play something, and you hear the artifact people describe as robotic or underwater.

Three causes, in the order they appear in small offices. Congestion on your own upload — a backup, a file sync, or a video upload on the same link; voice needs almost no bandwidth but it needs it on a strict schedule. Wi-Fi, which introduces variable delay even when it is good. And consumer routers with deep buffers, which queue rather than drop and turn congestion into latency spikes — bufferbloat, and the reason "my speed test is fine" and "my calls sound awful" coexist so often.

Packet loss: how much is too much?

Under 1% is generally imperceptible. Between 1% and 3% is audible as occasional clipped words. Above 3%, calls become difficult and people give up on them.

Voice runs over UDP with no retransmission, which is correct — a packet re-sent 200 ms late is useless in a conversation. Lost packets are gone, and the codec conceals what it can.

Loss and jitter usually share a root cause, which is why they arrive together. A saturated link both delays and drops; so does a failing patch cable. Ask where the loss occurs: between the phone and your router it is a LAN problem, and between your router and the internet it is a circuit problem worth taking to your ISP with data attached.

SIP ALG: the router feature that breaks the thing it claims to fix

SIP ALG (Application Layer Gateway) is a feature in most consumer and small-business routers that inspects SIP signaling and rewrites the addresses inside it. It was meant to help SIP traverse NAT. On modern systems it causes the problems it was built to prevent.

Turn it off. This is the single most common configuration fix in small-office VoIP, and the symptoms are distinctive:

• One-way audio, where you hear them and they hear nothing.
• Calls that connect and drop at a consistent interval.
• Phones that register, then lose registration repeatedly.
• Transfers or hold that fail in a way nothing else explains.

ALG implementations rewrite SIP headers with incomplete knowledge of the session, while modern SIP stacks handle NAT traversal correctly alone. A router "helping" corrupts the negotiation.

It hides under different names — SIP Transformations, SIP Helper, VoIP Passthrough, SIP Inspection. VoIP network requirements lists where to find it on common gear.

While you are in the router, check the firewall's UDP timeout. Phones re-register on an interval, and a NAT entry that expires sooner than that interval leaves the server with a stale path — inbound calls stop arriving even though the phone looks fine. SIP registration reference covers the timing.

So what is genuinely the provider's problem?

Three categories, and naming them keeps the split honest.

Carrier route quality. Calls to one area code sounding poor while everything else is fine points upstream. That is the provider's to fix.

Server-side congestion. On a shared platform, another tenant's traffic can affect yours. Signal gives each client an isolated phone server, so your call quality is not a function of what a stranger's call center did this morning.

Codec handling and trunk capacity. Both are platform choices, and insufficient capacity to the carrier shows up as failed calls and degraded audio at peak.

One test separates the two sides cleanly. If calls between two of your own extensions in the same building also sound bad, the problem is your LAN — that audio may never leave your network. If internal calls are clean and external calls are not, the path beyond your router is implicated.

What to check, in order

Do these before opening a ticket. They resolve most cases, and they make the ticket resolvable in one touch if they do not.

1. Note whether the problem affects one phone, one location, or every call. This one fact eliminates most of the possibilities.
2. Test internal extension to extension, then an external call. Compare.
3. Confirm the affected phone is on Ethernet rather than Wi-Fi.
4. Turn off SIP ALG and every SIP-inspecting feature on the router.
5. Check whether anything is saturating the upload — backups, syncs, video.
6. Swap the patch cable and try a different switch port on the affected desk.
7. Run a continuous ping during a bad call and watch for latency spikes and loss.

If it survives all seven, hand over the specifics: date and time of a bad call, the number, the extension, and which direction the audio failed. Choppy, robotic, or echoing VoIP calls is the full step-by-step, and one-way audio has its own path.

Whether you need a separate circuit for voice is answered by arithmetic, in do I need a dedicated internet line for VoIP. Home internet is its own case, covered in home internet and work calls.

Frequently asked questions

What causes choppy or robotic VoIP calls? Jitter — variation in packet arrival time — produces robotic or warbling audio, while packet loss produces clipped words and dropped syllables. Both usually come from a saturated upload link, Wi-Fi, or a router that buffers heavily under load. Phones absorb 30 to 50 ms of jitter; past that, you hear it.

How much packet loss is acceptable for VoIP? Below 1% is generally imperceptible. Between 1% and 3% you hear occasional clipped words. Above 3%, conversation becomes difficult. Voice runs over UDP with no retransmission, so lost packets are simply gone and the codec conceals what it can.

What is SIP ALG and why should I turn it off? SIP ALG is a router feature that rewrites SIP signaling to help it traverse NAT. Modern phone systems handle NAT correctly without it, and the rewriting corrupts sessions — producing one-way audio, calls that drop at consistent intervals, and repeated registration loss. Disable it, along with any SIP inspection setting.

Is bad call quality my internet or my phone provider? Test a call between two extensions in your own building. If that also sounds bad, the problem is on your LAN, because that audio may never leave your network. If internal calls are clean and outside calls are not, the path beyond your router is implicated.

Why do I only hear audio in one direction? One-way audio is almost always SIP ALG or a firewall configuration blocking the return media stream. Turn off SIP ALG first, then confirm the firewall is not restricting the RTP port range. It is rarely a codec or provider issue, which is why it has its own diagnosis path.

If your calls are unreliable and you cannot tell which side owns it, send us a bad call's time and number and we will trace it rather than trade guesses.