Echo, Delays and Dropped VoIP Calls: Causes and Fixes
Practice4 minInnovialabs editorial team
In short
Most VoIP problems are network problems: packet loss and jitter cause robotic, choppy audio, SIP ALG and NAT timeouts cause one-way audio and drops, and echo usually comes from a headset or speakerphone. Disable SIP ALG, enable QoS and keep-alive, use cable.
The short answer
Voice over IP cares less about internet speed and more about stability. Match the symptom to the cause:
| Symptom | Common cause | First thing to do |
|---|---|---|
| Robotic voice, missing syllables | Packet loss, jitter | Check the link, enable QoS, move from Wi-Fi to cable |
| Delay, talking over each other | High latency, congested link | Fix bufferbloat, prioritize voice |
| One-way audio | NAT, SIP ALG, blocked RTP ports | Disable SIP ALG, check the phone's NAT settings |
| Call drops after about 30 seconds | Signaling packets lost through NAT/ALG | Disable SIP ALG, try TCP or TLS |
| Inbound calls stop arriving | NAT timeout, registration expired | Shorten the registration interval, enable keep-alive |
| Echo | Speakerphone, cheap headset, analog gateway | Headset, echo cancellation, gateway settings |
Step 1. Measure the network
Three metrics define call quality:
- Latency — how long a packet takes. ITU-T Recommendation G.114 treats one-way delay up to about 150 ms as acceptable for most conversations.
- Jitter — variation in delay. The phone smooths it with a jitter buffer, but large variation still causes stutter.
- Packet loss — missing pieces of voice. Even small losses are audible.
Check the path to your provider's SIP server while the problem is happening:
# Linux / macOS
mtr -rwc 100 sip.provider.example
# Windows
pathping sip.provider.example
Many IP phones and softphones show call statistics (jitter, loss) in a menu during the call — that is the most accurate source.
Step 2. Disable SIP ALG
SIP ALG is a router feature that "helps" SIP through NAT by rewriting packets. In practice it often breaks calls: one-way audio, drops, failed registration. If your provider or phone handles NAT itself, which is usually the case, turn ALG off. Look for "SIP ALG", "SIP Helper", "SIP Passthrough" or similar in the router settings.
Step 3. NAT timeouts and keep-alive
A router keeps a UDP mapping only for a limited time. If the phone re-registers less often than that mapping lives, inbound calls stop arriving and long calls drop.
What to check:
- Registration interval (registration expires) on the phone — make it shorter than the router's UDP timeout.
- NAT keep-alive on the phone — enable it.
- TCP or TLS transport instead of UDP, if the provider supports it.
- STUN or NAT settings on the phone — as your provider recommends.
Step 4. QoS and bufferbloat
When someone in the office downloads a large file, the router queue grows and voice waits in line with everything else. That is bufferbloat.
- Enable QoS and mark voice: RTP as DSCP EF (46), SIP signaling as CS3 or AF31.
- If the router supports SQM (fq_codel or CAKE), enable it and set the rate slightly below your real link speed.
- In larger offices, use a dedicated voice VLAN.
- Reserve bandwidth for voice instead of leaving everything at maximum.
Step 5. Codecs
| Codec | Characteristics | When to choose |
|---|---|---|
| G.711 | Uncompressed, good quality, more traffic | Stable links |
| G.722 | Wideband, "HD voice" | Internal calls and supporting carriers |
| G.729 | Compressed, less traffic | Narrow links, if the provider supports it |
| Opus | Adapts to loss and bandwidth | Softphones and unstable networks |
Put your preferred codec first in the list and remove unsupported ones to avoid unnecessary transcoding.
Step 6. Wi-Fi
- Connect desk phones by cable, ideally with PoE.
- For softphones on laptops, use 5 GHz and good coverage with no dead zones.
- Enable WMM (multimedia priority) on access points.
- Roaming between access points can interrupt calls — check for fast roaming support (802.11r/k/v).
- Disable aggressive Wi-Fi power saving on laptops during calls.
About echo
- Acoustic echo: speaker sound leaks into the microphone. Fix it with a headset, sensible volume and a quality speakerphone.
- Echo on analog gateways (FXS/FXO): adjust echo cancellation and gain levels on the gateway.
- If you hear your own echo, the source is usually on the other person's side.