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SIP Trunking FAQ's

Welcome to our comprehensive list of SIP Trunking FAQs. Whether you’re a seasoned IT professional or a business owner exploring the world of voice and data communication, SIP Trunking plays a crucial role in modern telecommunications. 

In this collection of frequently asked questions, we aim to demystify SIP Trunking, offering insights, explanations, and practical advice. From the basics of what SIP Trunking is to the more intricate details of implementation and best practices at VirtualPBX, these FAQs will serve as your go-to resource for understanding and optimizing your organization’s voice and data connectivity.

If you don’t find your answer in our SIP Trunking FAQs, fill out the form below and we’ll be happy to answer any questions you may have!

SIP (Session Initiation Protocol) Trunking provides a path on which companies with an on-premise telephone system can access the PSTN (Public Switched Telephone Network) via an internet connection, therefore replacing traditional telephone lines or PRIs (Primary Rate Interface) at a much lower cost per month.

SIP Trunking offers a much lower overall cost per minute for voice traffic and can often allow companies to consolidate communications providers. But perhaps the most attractive advantage is the ability to scale up or down with ease whenever your business needs require.

Gartner reports that upgrading to SIP Trunking from a legacy system can save as much as 50% in telephony costs. To see what your savings could be, get a quote from our team today.

SIP channels can be likened to telephone lines where each line (or channel) can support a single call. VirtualPBX SIP Trunking Plans start with 10 channels, allowing you 10 concurrent calls. Need more? No problem. With VirtualPBX you may add or remove channels at any time. Each plan also includes 5 additional burstable channels, allowing you some extra margin during a burst of traffic.

Historically, a business could only make one call per business phone number, (sometimes referred to as lines) they had – when they needed to expand their use to two concurrent calls, the business would have to buy two lines, and so on. It was only in the 80’s when analog lines were replaced with Primary Rate Interface (PRI) lines. These PRI lines, which still serve traditional hardwired phone systems today, allow 23 calls to be made concurrently on each line. Like analog lines, PRI lines can be purchased in multiples, though they are only available in groups of 23 lines. Conversely, SIP channels can be purchased individually as needed – allowing you to pay only for what you need.

SIP Trunking and VoIP are the same in that they both facilitate voice traffic over an internet connection. A typical VoIP service plan will include features and configuration options in a web portal as well as provide a connection to the PSTN. The key difference is that SIP Trunking connects a second service (typically an on-site PBX system) to the PSTN to deliver voice traffic.

SIP Trunking is designed specifically to be the bridge from an on-site system to the cloud that is totally lossless in terms of sound quality. For crystal clear calls via SIP Trunking or VoIP, it is essential to maintain a robust internet connection with enough bandwidth to support the call traffic. Companies outgrowing their network may consider a separate internet connection for VoIP, enabling QoS (Quality of Service) on their router(s), or a more holistic approach like VPN for VoIP Service. Need help with your network? A Network Health Check from VirtualPBX is a great place to start and provides a seven day analysis into your network’s activity with customized suggestions from our network experts.

No! Our online interface makes it easy to connect your SIP-enabled PBX so you can deploy SIP Trunking in an instant. Simply choose your provider and enter your static IP or SIP URI to get started.

Any SIP-enabled PBX like Avaya, Cisco, ShoreTel, Mitel, Digium, Blue Box, Freeswitch, 3com, Allworx, AltiGen, Asterisk, Epygi, FreePBX, ObjectWorld, Pingtel, Responsive Point, Samsung, Sutus, TalkSwitch, and Taridium. We support a number of other SIP-enabled PBX systems as well. If your provider was not listed, let us know and we can check for you.

SIP Trunks use UDP on port 5060 by default.

Yes, TLS is supported as an option on all SIP Trunks.

The maximum number of concurrent calls is determined by the capacity you order from VirtualPBX, the capabilities of your internet connection, and the remote endpoint processing the calls.

We only support Normal Media.

We do not require P-Asserted Identity. You must pass the Caller ID using either P-Asserted Identity or Remote-Party-ID headers on every call.

No. But we do reserve the right to block traffic based on source to provide a secure service.

No. We do not support SIP Diversion Headers. Attempting to use these headers may interfere with normal operation of your trunk.

SIP Trunking is a cost-effective alternative to utilizing traditional PRI lines to connect a PBX to the PSTN. If you do not currently use a PBX to manage your phone system’s configuration, you are not a candidate for SIP Trunking and alternatively we recommend a fully hosted PBX service that will be able to handle custom configurations as well as dial tone. If you employ a PBX that does not utilize SIP Protocol and you wish to convert your traffic from PRI to SIP with a gateway, we recommend discussing this custom layout with our team.

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