Glossary / Networking for VoIP
🌐 Networking for VoIP covers the network infrastructure that makes internet-based phone calls possible. From local networks and quality settings to firewalls and advanced routing, this section contains 26 terms that help you understand why a good network matters for clear voice calls.
On this page: LAN · WAN · VLAN · QoS · DSCP · Bandwidth · Latency · Jitter · Jitter Buffer · Packet Loss · NAT · STUN · TURN · ICE · Firewall · Port · UDP · TCP · DNS/DNS SRV Record · SIP ALG · MPLS · SD-WAN · PoE · Ethernet · Wi-Fi Calling · MTU
LAN (Local Area Network)
A network that connects devices within a small area such as an office, a floor, or a building. All your desk phones, computers, and printers typically share the same LAN. VoIP phones need a reliable LAN to deliver clear calls without interruption.
Related: VLAN · Ethernet · WAN
WAN (Wide Area Network)
A network that connects locations across a large geographic area. If your company has offices in Luxembourg City and Esch-sur-Alzette, a WAN links them together. Your internet connection is also a WAN link. SIP trunks and cloud PBX traffic travel over the WAN to reach your provider.
Related: LAN · MPLS · SD-WAN
VLAN (Virtual LAN)
A way to split one physical network into separate virtual networks. Businesses often put voice traffic on its own VLAN so that phone calls do not compete with file downloads, web browsing, or video streaming. This separation improves call quality and makes the network easier to manage.
Related: LAN · QoS · Ethernet
QoS (Quality of Service)
A set of techniques that prioritise certain types of network traffic over others. For VoIP, QoS rules tell routers and switches to send voice packets first, before less time-sensitive data like email. Without QoS, a large file download could cause your phone call to sound choppy.
Related: DSCP · VLAN · Jitter
DSCP (Differentiated Services Code Point)
A marking placed on each network packet that tells routers how important it is. Voice packets are usually marked with DSCP value EF (Expedited Forwarding), which means "deliver this first." Routers that support QoS read the DSCP value and prioritise accordingly.
Related: QoS · UDP
Bandwidth
The maximum amount of data your network connection can carry per second, measured in megabits per second (Mbps). Each VoIP call uses a small amount of bandwidth (roughly 80 to 100 kbps). If your internet connection does not have enough bandwidth for all your calls plus normal data use, call quality will suffer.
Related: Latency · Packet Loss
Latency
The time it takes for a voice packet to travel from your phone to the other person's phone, measured in milliseconds (ms). Low latency means the conversation feels natural. High latency (above 150 ms one way) causes noticeable delays where speakers talk over each other.
Related: Jitter · QoS · Bandwidth
Jitter
Variation in the arrival time of voice packets. Even if average latency is low, packets that arrive at uneven intervals cause choppy or robotic-sounding audio. A jitter buffer can smooth this out, but excessive jitter still degrades call quality.
Related: Jitter Buffer · Latency · Packet Loss
Jitter Buffer
A small memory area in your phone or PBX that collects incoming voice packets and releases them at evenly spaced intervals. This smooths out the effects of jitter. A larger buffer reduces choppiness but adds a slight delay. Most VoIP devices adjust their jitter buffer size automatically.
Related: Jitter · Latency
Packet Loss
When voice packets are lost during transmission and never reach the destination. Even 1% packet loss can make a call sound broken or cause words to disappear. Packet loss is usually caused by network congestion, faulty cables, or overloaded equipment.
Related: Jitter · QoS · Bandwidth
NAT (Network Address Translation)
A technique used by routers to share a single public IP address among many devices on a private network. NAT is standard in most offices, but it can cause problems for VoIP because SIP was designed for direct connections. Special techniques like STUN, TURN, and ICE help VoIP work through NAT.
Related: STUN · TURN · ICE · Firewall
STUN (Session Traversal Utilities for NAT)
A protocol that helps your phone discover its public IP address and port number when it is behind a NAT router. The phone contacts a STUN server on the internet, which tells it how it appears to the outside world. This information is then used to set up the call's audio path correctly.
Related: NAT · TURN · ICE
TURN (Traversal Using Relays around NAT)
A relay server that forwards voice traffic when a direct connection through NAT is not possible. TURN is used as a fallback when STUN alone cannot solve the NAT problem. It adds a small amount of latency because all audio passes through the relay server.
Related: STUN · NAT · ICE
ICE (Interactive Connectivity Establishment)
A framework that combines STUN and TURN to find the best way to connect two phones through NAT and firewalls. ICE tries multiple connection paths and picks the one that works most reliably. Most modern VoIP systems use ICE automatically in the background.
Related: STUN · TURN · NAT
Firewall
A security device or software that controls which network traffic is allowed in and out of your network. Firewalls can block VoIP traffic if not configured correctly. You need to open specific ports for SIP signalling and RTP audio, or use a SIP-aware firewall that understands VoIP protocols.
Related: Port · SIP ALG · NAT
Port
A numbered endpoint used by network protocols to direct traffic to the right application on a device. SIP typically uses port 5060 (unencrypted) or 5061 (encrypted with TLS). RTP audio streams use a range of ports, often 10000 to 20000. Firewalls must allow traffic on these ports for VoIP to work.
Related: Firewall · UDP · TCP
UDP (User Datagram Protocol)
A fast, lightweight method of sending data over a network. VoIP typically uses UDP for audio because speed matters more than guaranteed delivery. If a voice packet arrives late, it is useless, so UDP's "send and forget" approach is better than TCP's slower but more reliable method.
Related: TCP · RTP · Port
TCP (Transmission Control Protocol)
A reliable method of sending data that confirms every packet was received. TCP is slower than UDP because of this checking process. VoIP uses TCP mainly for SIP signalling (especially with TLS encryption) rather than for audio streams, where speed is more important than guaranteed delivery.
Related: UDP · Port · TLS
DNS / DNS SRV Record
DNS (Domain Name System) translates domain names into IP addresses. A DNS SRV record is a special type of DNS entry that tells your PBX which server to contact for SIP services and on which port. If your provider's SRV records are configured correctly, your PBX can automatically find the right connection point and fail over to backup servers.
Related: TCP · UDP · SIP
SIP ALG (Application Layer Gateway)
A feature built into many home and office routers that tries to "help" SIP traffic pass through NAT. In practice, SIP ALG often causes more problems than it solves by modifying SIP packets incorrectly. VoIP specialists usually recommend turning SIP ALG off on your router.
Related: NAT · Firewall · SIP
MPLS (Multiprotocol Label Switching)
A method of routing network traffic using short labels instead of long network addresses. MPLS creates predictable, low-latency paths through a provider's network. Businesses sometimes use MPLS connections for VoIP between offices to guarantee call quality, though it is more expensive than regular internet.
Related: SD-WAN · WAN · QoS
SD-WAN (Software-Defined Wide Area Network)
A technology that uses software to manage connections between office locations across multiple internet links. SD-WAN can automatically route VoIP traffic over the best available path, switching to a backup link if the primary one has too much latency or packet loss. It is a more flexible and often cheaper alternative to MPLS.
Related: MPLS · WAN · QoS
PoE (Power over Ethernet)
A technology that delivers electrical power through the same Ethernet cable that carries data. Most IP desk phones support PoE, which means you need only one cable per phone instead of a separate power adapter. You need a PoE-capable network switch to use this feature.
Related: Ethernet · IP Phone
Ethernet
The standard technology for wired local networks. Ethernet cables (Cat5e, Cat6, Cat6a) connect phones, computers, and switches. For VoIP, a wired Ethernet connection is almost always more reliable than Wi-Fi because it provides consistent speed and very low jitter.
Related: LAN · PoE · VLAN
Wi-Fi Calling
Making phone calls over a wireless network instead of a wired Ethernet connection or a mobile network. Many VoIP softphones and some desk phones support Wi-Fi. While convenient, Wi-Fi can introduce more jitter and packet loss than a wired connection, especially on busy networks. A dedicated VLAN and QoS settings help improve Wi-Fi call quality.
Related: Ethernet · VLAN · QoS
MTU (Maximum Transmission Unit)
The largest packet size (in bytes) that a network link can carry without splitting the packet into smaller pieces. The standard MTU for Ethernet is 1500 bytes. If MTU is set incorrectly, voice packets may be fragmented, which adds delay and can cause audio problems. Most networks work fine with the default setting.
Related: Ethernet · UDP · Packet Loss
Related Sections
🔗 SIP Protocol — The signalling protocol that controls VoIP calls
🎵 Audio, Media and Codecs — How voice is compressed and transmitted
🔒 Security — TLS, SRTP, and VoIP fraud prevention
🎧 Devices and Hardware — IP phones, headsets, switches, and gateways
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