VoIP

For business-side detail, see VoIP Business Phone System.

How VoIP works

A voice call has two pieces:

• Signaling — setting up, modifying, and tearing down the call. SIP (Session Initiation Protocol) is the dominant signaling protocol.
• Media — the actual audio. RTP (Real-time Transport Protocol) carries encoded audio packets between phones, typically peer-to-peer once SIP has established the call.

SIP runs over UDP or TCP on port 5060 (5061 for SIP-TLS). RTP runs over UDP on a dynamic port range (10000-20000 typical). Both endpoints (or their respective SIP servers) negotiate the call setup, then audio flows directly.

Codecs

Voice is digitized using a codec. Common choices:

What determines VoIP quality

Bandwidth is rarely the binding constraint. The metrics that matter:

• Latency: One-way under 150 ms keeps conversation smooth. Above 400 ms causes the "talk-over-each-other" problem.
• Jitter: Variation in arrival time. Under 30 ms is good; above 50 ms produces choppy audio.
• Packet loss: Under 0.5% is good; above 1% produces audible dropouts.
• Available bandwidth: Each call needs ~100 Kbps; 10 simultaneous calls need ~1 Mbps. Easy on modern internet.

QoS for VoIP

QoS (Quality of Service) prioritizes voice packets over other traffic. Without QoS, a large file upload on the same connection can starve voice packets, causing dropouts. Standard configuration:

• Voice packets marked with DSCP EF (decimal 46) — high-priority queue.
• Voice VLAN separation so switches can prioritize.
• Bandwidth shaping on WAN to guarantee voice has capacity.
• WMM (WiFi Multimedia) on access points for wireless voice.

VoIP deployment models

Hosted (cloud) PBX

The PBX runs in the cloud. Phones register over the internet to the provider (RingCentral, 8x8, Vonage, Dialpad, Microsoft Teams Phone, Zoom Phone). Standard for SMBs in 2026 — no on-site hardware beyond phones, $20-40/user/month all-in.

On-premises PBX

You run the PBX yourself on a local server (FreePBX, 3CX, Asterisk, Avaya). Connects to a SIP trunk provider for outside calls. Lower per-user cost at scale but requires operational expertise.

Carrier landline (POTS replacement)

Traditional landline service delivered as VoIP via an ATA (Analog Telephone Adapter) plugged into your modem. From the user's perspective, it's an old phone with a dial tone; from the network's perspective, it's VoIP.

The 911 problem

Traditional landlines automatically reported the calling address to emergency services. VoIP can be used from anywhere, so the address must be explicitly registered. RAY BAUM's Act (US) and Kari's Law require:

• Direct 911 dialing (no access code).
• Dispatchable location automatically sent.
• Notification to designated office contacts when 911 is dialed.

Hosted PBX providers handle this automatically; on-prem deployments must configure it. Fines for non-compliance start at $10,000.

Frequently Asked Questions

How much bandwidth does a VoIP call use?

Typically 80-100 Kbps per direction with G.711 codec, or 30-40 Kbps with compressed codecs (G.729, Opus). For 10 simultaneous calls: roughly 1 Mbps total with G.711. Bandwidth is rarely the binding constraint for modern internet connections; latency and jitter matter much more for voice quality.

What is the difference between VoIP and SIP?

VoIP is the general category — voice transmitted over IP. SIP (Session Initiation Protocol) is the specific signaling protocol used by most VoIP systems to set up and tear down calls. SIP handles "ring this number, accept the call, hang up"; RTP carries the actual audio. VoIP includes SIP plus the audio codecs (G.711, Opus, etc.) plus the broader infrastructure (PBX, SIP trunks).

What is acceptable VoIP quality?

For good voice: latency under 100 ms one-way, jitter under 30 ms, packet loss under 0.5%. For excellent voice: latency under 50 ms, jitter under 10 ms, packet loss under 0.1%. Modern fiber and well-configured business networks deliver excellent quality reliably; congested cable or WiFi connections often deliver acceptable quality with occasional artifacts.

Does VoIP work during a power outage?

Not without UPS. Unlike traditional landline phones that were powered by the phone company's central office battery, VoIP requires powered equipment at every point: modem, router, switch, phone. A power outage takes the office offline. Battery-backed UPS on network equipment, mobile-app failover (calls forward to cell phones), and cellular-based emergency phones address this. Critical for 911 access — make sure at least one phone in the building works without grid power.