VoIP Network Fundamentals

A fast, visual guide to the network concepts that make or break call quality.

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QoS DMZ Latency, Jitter, Loss MTR SIP Ports SIP ALG WebRTC SIP Trunks STUN/TURN Wireshark

QoS (Quality of Service)

QoS prioritizes voice packets over bulk data so calls stay clear even when the network is busy.

Set WAN upload/download shapers to ~85-90% of real speeds to keep the router in control.
Reserve voice bandwidth based on max concurrent calls + ~20% overhead.
Typical starting point: allocate 20-30% of the uplink for VoIP, then adjust.

DMZ (Demilitarized Zone)

A DMZ isolates publicly reachable systems from internal devices.

Use a DMZ for any VoIP system that must accept inbound internet traffic.
Keep phones on the internal LAN or a dedicated voice VLAN.
Only open the required SIP and RTP ports through the firewall.

Note: Enabling a DMZ forwards all inbound internet traffic to the selected device, which effectively bypasses firewall protection. This can expose the device to security risks, so use it only as a last resort.

Latency, Jitter, Packet Loss

Voice is sensitive to delay variation. Keep these in the green for stable calls.

Metric	Excellent	Acceptable	Poor
Latency	< 70 ms	70-150 ms	> 150 ms
Jitter	< 10 ms	10-30 ms	> 30 ms
Packet Loss	< 0.5%	0.5-1%	> 1%

Jitter is variation in packet arrival time. Too much variation causes choppy audio.

What Is MTR?

MTR combines ping and traceroute so you can see loss and latency at every hop.

Run it toward the provider when calls are choppy.
Loss that starts mid-path is often ISP or transit related.
Look for jitter spikes near the end of the path.

Ports 5060 and 5061

SIP is the signaling protocol that sets up calls. Media flows separately over RTP.

5060: SIP over UDP or TCP.
5061: SIP over TLS (encrypted signaling).
RTP uses a provider-specific port range.

What Is SIP ALG?

SIP ALG is a router feature that edits SIP packets and often breaks VoIP.

Best practice: disable SIP ALG for stability.
Symptoms include one-way audio and dropped calls.
Document the setting in your network checklist.

Customer version: SIP ALG is a “helper” setting in some routers that rewrites call traffic. It often changes the wrong parts of the call, which leads to one-way audio, dropped calls, or phones that won’t register. Turning it off is a common fix.

What Is WebRTC?

WebRTC delivers real-time voice and video directly in the browser without plugins.

Used for softphones, web call widgets, and contact centers.
Relies on ICE, STUN, and TURN for NAT traversal.
Security is built-in with DTLS and SRTP.

What Are VoIP Trunks?

A SIP trunk is a shared connection between your phone system and the carrier.

One trunk provides multiple simultaneous call paths.
Scales without installing physical phone lines.
Common for PBX-to-provider integrations.

STUN, TURN, ICE (NAT Traversal)

Remote users and WebRTC need NAT traversal to avoid one-way audio.

Protocol	Purpose	When to Use
STUN	Discover public IP/port	Most NAT scenarios
TURN	Relay media traffic	Restrictive firewalls
ICE	STUN first, TURN fallback	Default in WebRTC

Wireshark & Packet Capture

Packet captures reveal SIP errors and RTP quality problems.

tcpdump -i eth0 -w voip.pcap port 5060 or portrange 10000-20000

sip || rtp || sip.Status-Code >= 400

Check Telephony → RTP → Stream Analysis for jitter and loss.
Filter by SIP error codes to see failed calls.

Quick Troubleshooting Checklist

Run MTR to the provider and look for loss or jitter.
Verify QoS is enabled and voice is prioritized.
Disable SIP ALG in the router or firewall.
Confirm SIP + RTP ports are allowed.
Test on wired Ethernet to rule out Wi-Fi jitter.