Troubleshooting audio issues (echo, latency, dropouts) Print

VoIP audio issues — Diagnosis and resolution

VoIP audio quality depends on multiple factors: network, codec, NAT configuration, equipment quality. Here is how to diagnose and resolve the most common issues.

1. Echo (acoustic echo)

Symptom: You hear your own voice delayed during the call.

Causes and solutions:

• Microphone and speaker too close together: Use a headset with a built-in microphone
• Volume too high: Lower the speaker volume or enable echo cancellation (AEC) in the softphone
• Network-side echo: Enable echo processing in your PBX/SIP equipment (Asterisk: rxgain, txgain)

# Asterisk — Enable the echo canceller
[general]
echocancel=yes
echocancelwhenbridged=yes

2. Excessive latency (audio delay)

Symptom: Lag between your voice and its reception by the other party (> 150ms noticeable, > 400ms unacceptable).

Diagnosis:

ping sip.webtic.online   # Latency < 50ms : excellent, 50-150ms : acceptable
traceroute sip.webtic.online   # Identify problematic hops

Solutions:

• Use G.711 (no compression = less processing latency)
• Configure QoS (Quality of Service) on your router: prioritize UDP traffic on ports 10000-20000
• Reduce the RTP packet size (from 20ms to 10ms)

3. Dropouts and jitter

Symptom: The audio cuts in and out intermittently, "robotic" voice.

Cause: Variation in transmission delay (jitter). Acceptable < 30ms, problematic > 50ms.

# Test network quality
mtr --report sip.webtic.online
# Look for: latency variation between packets (jitter)

Solutions:

• Increase the jitter buffer: Asterisk jitterbuffer=yes and maxjitterbuffer=200
• Switch to a wired connection (Ethernet > WiFi for VoIP)
• QoS: DSCP EF (Expedited Forwarding) for RTP traffic

4. One-way audio

Symptom: You can hear the other party but they cannot hear you (or vice versa).

Cause: NAT issue — the RTP stream cannot cross the firewall.

Solution: See the article NAT and Firewall configuration for SIP.

5. RTCP test command

# Capture SIP and RTP traffic
tcpdump -i eth0 -w capture_voip.pcap "port 5060 or portrange 10000-20000"

# Analyse with Wireshark:
Telephony → RTP → Streams → analyze jitter, packet loss

Quick checklist

• ☐ Network latency < 150ms (ping sip.webtic.online)
• ☐ Jitter < 30ms (mtr sip.webtic.online)
• ☐ No packet loss (> 1% = network problem)
• ☐ RTP ports open (10000-20000 UDP)
• ☐ QoS enabled on the router
• ☐ G.711 codec configured as priority