VICIdial Carrier Protocol: SIP vs IAX2 (and PJSIP)

Every carrier you add in VICIdial has a Protocol field, and it only accepts a short list of values: IAX and SIP, plus PJSIP on modern builds. This field tells your server how to speak to the trunk on the wire. Pick the wrong one and the account entry and dialplan you wrote for the other protocol simply will not load.

For nearly every new VICIdial deployment the answer is SIP or PJSIP. IAX still works and has its niche, but the protocol your provider hands you is almost always SIP. This post explains the difference so you can match the field to what your carrier actually supports.

What each protocol is

SIP, the Session Initiation Protocol, is the standard signaling language of voice over IP. It sets up, manages, and tears down calls, while the audio rides separately over RTP. It is what every commercial provider speaks, so a SIP trunk is the default path into the phone network.

PJSIP is the newer Asterisk SIP stack. It speaks the same SIP protocol on the wire but uses a different, more flexible config format inside Asterisk. VICIdial supports it alongside legacy SIP, and many providers now ship PJSIP example configs.

IAX, the Inter-Asterisk eXchange protocol, was built to connect Asterisk boxes to each other. It carries signaling and audio on a single UDP port, which makes it friendlier to firewalls and NAT. Its drawback is reach: outside the Asterisk world, almost nobody offers an IAX2 trunk.

Choosing the right value

flowchart TD
  A[New carrier] --> B{Provider gives SIP config?}
  B -->|Yes, classic format| C[Protocol SIP]
  B -->|Yes, pjsip format| D[Protocol PJSIP]
  B -->|No, Asterisk to Asterisk| E[Protocol IAX]
  C --> F[Save and check peers]
  D --> F
  E --> F

Let your carrier decide. Whatever account format they hand you, set the Protocol to match. A SIP-format account entry under an IAX protocol setting will not register, and the reverse fails the same way. The protocol and the account syntax must agree.

Verifying the peer is up

After you save a carrier, wait about a minute for the conf to regenerate, then confirm the peer registered. From the Asterisk CLI you run sip show peers for a SIP trunk or iax2 show peers for an IAX trunk. PJSIP trunks use pjsip show endpoints. A peer that shows up with a status means your Registration string and account settings are correct and the protocol matches.

If the peer is missing, the usual suspects are a protocol mismatch, a wrong secret, or the carrier record left inactive. The protocol field is the first thing to recheck because it silently invalidates an otherwise perfect Conf file entry. Also confirm your chosen Codec is one the provider allows, since a codec mismatch can let a peer register yet still fail audio.

For the full sequence of carrier fields, see our carrier integration guide. If you are still deciding which provider to use at all, our post on choosing a SIP carrier covers the trade-offs, and the SIP trunk vs registration post explains how the trunk authenticates.

Wrap-up

Set Protocol to whatever your carrier hands you: SIP or PJSIP for almost everyone, IAX only for box-to-box links. Then verify the peer from the CLI before you load leads. On VICIfast your dialer is provisioned and ready in under 40 seconds, so you can paste a carrier config and run sip show peers immediately. See our pricing to get started.