We get asked this question a lot! At Yay, we support both SIP trunks, AND SIP users, so we thought we should explain some more.

One thing our customers love at Yay is how we let them mix SIP Trunks and SIP Users in our cloud switchboard. For example, you can make a call to a Yay number and send that call to both a SIP trunk connected to your on-premise PBX and at the same time send it to a SIP User;  the first to answer the call receiving it.

SIP Trunks, also know as SIP Channels, were the original solution to transition hardware based PBX’s (Switchboards) away from analogue phone lines to a digital VoIP solution.

A hardware based PBX was that box in the corner of your office that your phones used to connect to. It then connected all your calls to the analogue phone cables coming into your building and let you transfer/hold calls etc - features nowadays better handled by a cloud based PBX like Yay

Although technically not limited in number of concurrent calls, historically, SIP Trunks were sold with a limitation of one call per “channel” to make them easier to sell and compare against a traditional analogue phone line.

Modern hardware based PBX's will generally let you connect to your VoIP provider via a SIP Trunk or a SIP User (See below), but this is quite recent, so you will often still need a SIP Trunk to make your on premise PBX (Now both hardware and software in form) make and receive calls.

SIP Trunks:

• Authenticate outbound using an IP address instead of a Username and Password. This is a key differentiator between a SIP Trunk and SIP User.
• Receive all traffic to their configured IP or Domain Name regardless of whether the PBX is available to receive it or not. The PBX is always assumed to be alive and awaiting our calls.
• Have no concurrency limits with Yay. Lots of SIP trunk providers do impose a concurrency limit (EG: One call at a time). We think that’s old fashioned.
• Need a hardware or software based PBX to handle what to do with a call that is received. EG: Send it to a phone or let a phone call out.
  
  

SIP Users, also known as SIP End Points, are the accepted standard for connecting devices such as VoIP Phones, VoIP Intercoms and VoIP Apps to a cloud based telephone switchboard (Widely referred to as a Hosted PBX).

SIP Users:

• Registry with Yay using a username, password to tell us they are online and available. This then means outbound calls can be made and inbound calls received with no IP address or domain name restrictions.
• Won't be able to receive inbound calls or be able to make them if they aren’t “registered” with their VoIP Provider.
• Authenticate with a username and a password only and are therefore far more transportable than SIP Trunks. You can easily use them on a softphone or desktop anywhere in the world as a SIP User doesn’t care if the device has a different IP Address each time.

Take advantage of all the hosted PBX features at Yay.