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SIP Trunking is an excellent addition to unified communications. It connects a phone over a public internet connection, which leaves out the old analog phone lines. SIP Trunking uses Voice over Internet Protocol to connect an on-premises phone system and the PSTN (public switched telephone network). SIP Trunking allows for better, more flexible communication services. It can send telephone calls, video conferencing, instant messaging, and images to a variety of devices, like business and mobile devices.

Voxtelesys recently launched SIP Trunk Registrar, a registration with your Trunking Provider. With SIP Trunk Registrar, we can link the customer's IP Address to their account. We also have IP Authentication, which requires a public static IP Address. If the static IP changes, manual changes are needed for the customer to allow outbound calls and to point inbound calls to the new IP. With IP Authentication, we need to know what PBX Phone System to pass along the phone call so the customer can answer it. SIP Trunk Registrar improves the quality of service for the customer.

How It Works

With SIP Trunk Registrar, the customer is given a domain (link), username, and password to set in their PBX. Then, their PBX will contact us every few minutes and "register" to our SIP services. When they register, we know which IP contacts us with those specific credentials and where to pass inbound calls.

SIP Trunk credentials live in all of our data centers, and if a problem exists in one DC, the domain will resolve to another DC automatically. There is no one-hour lag time. We also have extended potential security options like only allowing whitelisted IPs to register with Registrar credentials.

What to Know about SIP Trunk Registrar

• Requires username / password.
• Can restrict to a customer-provided access control list (ACL) for tighter security.
• In almost 99.999% of cases, Registrar is coupled with SIP REGISTERs. The REGISTER request tells our SIP servers where the UAC can be reached for inbound calls. This is handy because it makes inbound call routing dynamic.
• Requires a domain provided by SIP provider (Voxtelesys). The domain can be unique (i.e., 190761-1111.sip.voxtelesys.net) or shared (i.e., 3cx.pstn.voxtelesys.net).
• Our domain records resolve A, NAPTR, and SRV records. These records allow client failover if the SIP servers are unreachable if supported by the UAC. They can also be used to steer protocol-specific traffic (i.e., TLS, TCP).

SIP Trunking enhances voice services for the customer, and our SIP Trunk Registrar makes it even easier. View our Tutorial or Video to find out how to add IP addresses / register in our Portal. If you have any further questions about our Trunking Services, contact us at Voxtelesys.