carriers-sip
NAT traversal
NAT traversal is the set of tricks that lets a phone behind a home or office router send and receive VoIP audio across the wider internet.
NAT stands for Network Address Translation, the everyday feature in routers that lets many devices share one public internet address. It works fine for web browsing, but it causes headaches for voice calls because SIP (Session Initiation Protocol) (the protocol that sets up calls) and RTP (the protocol that carries the audio) embed internal addresses that the outside world cannot reach. NAT traversal is the collection of techniques that work around this so the call still connects.
The classic symptom of a NAT problem is One-way audio: the call connects, but only one side can hear the other. That happens when the signaling negotiates a private address that the return audio cannot find its way back to. Because a Softphone on an agent's laptop almost always sits behind a router, NAT handling is something every contact center has to get right.
How it gets solved
There are a few common approaches. Some setups rewrite the addresses on the way out so the public side is correct. Others use helper servers, often grouped under the labels STUN and TURN, that let a phone discover its real public address or relay audio when nothing else works. In a VICIdial install, the server side is usually configured to expect phones behind NAT, so each SIP peer is told to learn the public address from where packets actually arrive.
For most teams running a hosted dialer, the good news is that NAT traversal is handled at the server, so agents just need a stable VoIP connection. If you do hit one-way audio after a network change, NAT is the first place to look.
A few real-world habits keep NAT problems rare. Give agents wired connections where you can, since a stable address changes less often than one bouncing across Wi-Fi access points. Avoid stacking multiple routers between an agent and the internet, because each extra layer of address translation is another place for the audio path to break. And when you do troubleshoot, test the audio in both directions on purpose: have one side count out loud while the other listens, then swap. That quickly tells you whether you have a genuine two-way connection or a NAT issue letting sound through only one way. None of this is exotic, but skipping it is the most common reason a brand-new agent setup connects yet nobody can hear anything.
Related terms
One-way audio
One-way audio is when one person on a call can be heard but cannot hear the other, almost always a problem with how the audio stream is routed.
RTP
Real-time Transport Protocol, the stream of small packets that carries the actual voice audio of a VoIP call between the two endpoints.
SIP (Session Initiation Protocol)
The standard signaling protocol that sets up, manages, and ends internet phone calls — how VICIdial talks to phones and carriers.
SIP peer
A SIP peer is any single phone or trunk the server knows how to talk to, with its own name, credentials, and connection settings.
Softphone
A softphone is phone software running on a computer or mobile device that makes and receives calls over the internet instead of using physical desk-phone hardware.
VoIP
Voice over Internet Protocol, the technology that sends phone calls as data packets over the internet instead of over traditional copper phone lines.