SIP Trunking FAQs
The following are some of the most common questions asked about SIP Trunking. If you don’t find your answer here, contact us by calling 1-888-825-0800, Option 1 and we’ll be happy to answer any questions you may have.
Please click on a topic to expand or collapse the solution.
- What is SIP Trunking?
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.
- What are the main benefits of SIP Trunking?
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.
- What are SIP channels?
SIP channels can be likened to telephone lines where each line (or channel) can support a single call. VirtualPBX SIP Trunking Plans start with 20 channels, allowing you 20 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.
- We are using a PRI now, how is this different?
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.
- Are SIP and VoIP the same thing?
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.
- How is the quality of a call over a SIP Trunk?
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 VoIP Clear Fix 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.
- Is SIP Trunking very technical?
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.
- What PBX can you connect to?
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.
- I don’t have an IP PBX, can I still use a SIP 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.
- What is the Inbound Protocol and the Inbound Protocol Listen port?
SIP Trunks use UDP on port 5060 by default.
- Do you support TLS encryption?
Yes, TLS is supported as an option on all SIP Trunks.
- What is the maximum number of concurrent calls on a SIP Trunk?
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.
- Do you require Normal Media or Delayed Media?
We only support Normal Media.
- Do you require P-Asserted Identity if the connection type is public IP address?
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.
- Do you require Outbound Digest Authentication if the connection type is public IP address?
- Do you require the FROM Header to come from a specific address if the connection type is public IP address?
No. But we do reserve the right to block traffic based on source to provide a secure service.
- Do you support SIP Diversion Headers?
No. We do not support SIP Diversion Headers. Attempting to use these headers may interfere with normal operation of your trunk.