Session Initiation Protocol

SIP (Session Initiation Protocol) is the standard signaling protocol used to initiate, maintain, and terminate voice calls over IP networks. In VICIdial, SIP is the primary protocol for connecting to VoIP carriers and trunks, handling call setup, teardown, and in-call signaling between the Asterisk PBX and external telephony providers.

How It Works in VICIdial

VICIdial runs on Asterisk, which supports SIP through two channel drivers: the legacy chan_sip module and the newer PJSIP stack. SIP trunks are configured in the Asterisk configuration files (typically sip.conf or pjsip.conf) and define the connection parameters to each carrier — including authentication credentials, codecs, registration settings, and NAT traversal options.

When VICIdial places an outbound call, Asterisk constructs a SIP INVITE message and sends it to the carrier. The carrier routes the call to the PSTN, and when the recipient answers, a RTP media stream is established for the audio. SIP also carries DTMF signals and handles call transfers within the IVR and call menu systems.

For agent connections, SIP softphones or WebRTC clients register to the Asterisk server, creating persistent connections that the dialer uses to bridge answered calls to agents.

Why It Matters

SIP is the backbone of VICIdial’s telephony infrastructure. Proper SIP configuration directly impacts call quality, connection reliability, and the number of concurrent calls your system can handle. Misconfigured SIP settings — particularly around NAT, codecs, and registration timers — are among the most common causes of one-way audio, dropped calls, and registration failures in VICIdial deployments.

Understanding SIP fundamentals is essential for troubleshooting call quality issues and optimizing carrier connections. See the VICIdial setup guide for SIP trunk configuration best practices.