Guide

VoIP Troubleshooting Tools: Call Quality, Ports and Audio Problems

Updated August 16, 2026

Use this page when a VoIP call sounds poor or fails to connect. Start with the call-quality scorer, then check the port reference and follow the symptom guide. These tools provide practical starting points; your provider’s documentation and call records remain authoritative.

Call-quality scorer

Enter the approximate worst-case values measured during a call. Lower latency, jitter and packet loss are better.

Round-trip latency (ms)

Jitter (ms)

Packet loss (%)

Likely acceptable
Continue monitoring and investigate if users report a problem.

Common VoIP ports

Purpose	Typical ports	Notes
SIP over UDP/TCP	5060	Signalling; confirm the provider’s transport requirements
SIP over TLS	5061	Encrypted signalling; certificate and DNS configuration matter
RTP/RTCP	Provider-defined range	Allow the exact media range required by the PBX or carrier
STUN	3478	Helps endpoints discover public addressing; implementation varies
TURN over TLS	5349	Relays media when direct connectivity is unavailable
HTTPS provisioning	443	Used by many phones for configuration and firmware retrieval

Symptom guide

No audio

Confirm the call connects and inspect SDP for private or unreachable media addresses.
Check that RTP ports are allowed in both directions where required.
Review NAT behaviour and disable or correctly configure SIP ALG.
Confirm the phone, PBX and provider agree on at least one codec.

One-way audio

Compare the working and missing RTP directions in a packet capture or provider trace.
Check firewall state, NAT pinholes and advertised contact/media addresses.
Test from another network to separate endpoint issues from site issues.

Calls drop after about 30 seconds

Check whether the BYE or timeout is generated locally, by the provider or by a firewall.
Review SIP session timers and refresh messages.
Confirm that return signalling can reach the registered endpoint.

Registration fails

Verify the username, secret, realm, registrar and transport.
Check DNS resolution and whether the provider restricts source IPs.
Use the SIP response code and provider logs to distinguish authentication, routing and policy failures.