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VoIP

Voice over Internet Protocol

VoIP (Voice over IP) is the technology of transmitting voice as packets across an IP network, including private networks and the Internet.

TopicIT knowledge → About 2 min read

Simple definition

VoIP stands for Voice over Internet Protocol. It enables phone calls by carrying digital voice packets over an IP network instead of relying exclusively on traditional switched telephone circuits.

Technical definition

VoIP endpoints convert voice to digital audio, encode it with a codec and exchange packets over the network. SIP is often used to establish the call; RTP or SRTP generally carries the voice media. The PBX, endpoints and intervening network all need suitable configuration for both participants to hear each other.

Concept diagram How voice travels over an IP network The codec turns sound into data; the network carries the packets.
  1. Speech Audio source
  2. Codec Audio encoding
  3. RTP / SRTP Media packets
  4. IP network Transmission
  5. Playback Audio decoding

SIP is commonly used to establish a call, while RTP/SRTP carries voice media.

How does a VoIP call work?

  1. An IP phone, desktop app or handset connected through a base station contacts its phone system.
  2. The destination is resolved and call signaling, often SIP, establishes the session.
  3. A codec encodes voice, then media packets are typically sent over UDP/RTP.
  4. When the conversation ends, the signaling session is terminated.

What is VoIP used for?

VoIP powers business calling, cloud and on-premise PBX systems, softphones, contact centers and communication between office sites. An IP PBX such as 3CX manages calling features; a SIP provider connects calls to public telephone numbers.

Practical example

A staff member calls a customer from the 3CX desktop app. The PBX applies the outbound rule and sends the call through a SIP trunk to the carrier. If SIP signaling works but RTP packets are blocked, the call may ring even though audio fails.

Common issues

  • Choppy sound or delays: inspect latency, jitter, packet loss and congestion.
  • One-way audio: check firewall, NAT and RTP routing.
  • Phone not registered: inspect user credentials, SIP, DNS and PBX reachability.
  • Rejected outbound calls: review permissions, dialed number format, carrier and outbound rules.
  • Codec mismatches: check endpoint and provider media compatibility.

VoIP quality and security

  • Use reliable network connectivity and plan capacity for concurrent calls.
  • Apply QoS and voice VLANs when required by the network design, while sizing the Internet connection appropriately.
  • Protect accounts, PBX access and SIP trunks; use SIP over TLS and SRTP when supported and correctly configured.
  • Minimize exposed services and watch for unusual call patterns or toll fraud.

VoIP vs SIP vs 3CX

VoIP is the transport of voice across IP networks. SIP is one signaling protocol commonly used for VoIP calls. 3CX is an IP PBX product that manages users, phones, SIP trunks and routing. VoIP is not synonymous with 3CX.

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