How VoIP Works

WHAT IS VoIP?

Voice over Internet Protocol (VoIP) converts your voice into digital packets and sends them over the internet instead of traditional phone lines. When you make a call through SpoofGlobal, your voice travels as data packets from your device, through the internet, through our servers, and out to the phone network (PSTN) where it reaches the recipient's phone.

THE TWO PARTS OF A VoIP CALL

1. Signaling (SIP)

SIP (Session Initiation Protocol) handles the "phone call management" — setting up the call, ringing, answering, and hanging up. When you press Call, your SIP client sends an INVITE message to the SIP server. The server routes this to the destination. When the other party answers, a 200 OK response comes back, and the call is established.

The caller ID is transmitted in the SIP signaling — specifically in the From header of the INVITE message. This is where caller ID spoofing happens: by setting a different number in the From header.

2. Media (RTP)

RTP (Real-time Transport Protocol) carries the actual audio. Once the SIP signaling establishes the call, RTP streams flow directly between your device and the media server. Audio is encoded using a codec (like G.711 or Opus), packetized, and sent as UDP packets typically every 20 milliseconds.

CODECS

Codecs compress and decompress audio. Common VoIP codecs:

G.711 (PCMU/PCMA) — uncompressed, highest quality, 64 kbps. Best for local/broadband connections.
G.729 — compressed, good quality, 8 kbps. Good for limited bandwidth.
Opus — modern, excellent quality at variable bitrates. Used by WebRTC (Web Dialer).
GSM — highly compressed, 13 kbps. Decent quality, low bandwidth.

NAT AND FIREWALLS

Most home and office networks use NAT (Network Address Translation), which can interfere with VoIP. SIP and RTP use different ports, and NAT can prevent the audio stream from reaching your device. Solutions include:

STUN servers — help your client discover its public IP
Port forwarding — configure your router to forward SIP/RTP ports
Using a VPN with a static IP
WebRTC — handles NAT automatically using ICE

HOW SpoofGlobal ROUTES CALLS

Your SIP client or Web Dialer sends a SIP INVITE to our server
Our server authenticates you (by credentials or IP)
The dialed number is matched against available routes
Your chosen caller ID is set in the outgoing SIP headers
The call is forwarded through our carrier trunks to the PSTN
The recipient's phone rings with your spoofed caller ID displayed

Does VoIP quality match regular phone calls?+

Yes, often better. With a stable internet connection and the G.711 or Opus codec, VoIP calls can be higher quality than traditional phone calls.

How much bandwidth does VoIP use?+

A single VoIP call uses 30-100 kbps depending on the codec. G.711 uses about 85 kbps, G.729 uses about 30 kbps. Any broadband connection easily handles multiple concurrent calls.

Why does my VoIP call sometimes have echo?+

Echo is usually caused by audio from your speaker feeding back into your microphone. Use headphones to eliminate it. Most modern clients have built-in echo cancellation.

Can VoIP calls be intercepted?+

Without encryption, VoIP packets can theoretically be captured on the network. SpoofGlobal's Web Dialer uses WebRTC with mandatory encryption (DTLS-SRTP). For SIP clients, look for SRTP support.