First, let’s address SIP: Session Initiation Protocol is the de facto standard protocol for establishing, conducting, and ending a VoIP call. If you are really interested in technical details, check our dedicated blogs on SIP, VoIP, or how SIP vs. VoIP compares.
SIP is to VoIP as SMTP is to email. Just like the standard protocol, SMTP, which allows two email servers to exchange email data, SIP allows two endpoints (IP phones) to connect to one another using a standard protocol. Without such a standard, two phones would not share a common set of instructions and guidelines in order to properly exchange packetized voice.
The great thing about SIP is that it promises to help do away with phone numbers all together. How? Imagine your email address could be “dialed” from a phone and it would kick off a call. Further, imagine everyone at your company was reachable by phone using email addresses.
Think of SIP as the protocol for yet another service capability of a domain:
- Email: SMTP
- Web: HTTP
- Voice: SIP
- Video: SIP
The term “SIP provider” means different things depending on who is using it. It has its roots in “SIP trunking,” which is a service provided to businesses that allows them to connect their in-house IP PBX systems to the outside world—the public switched telephone network, or PSTN.
Without SIP trunking, devices within a PBX can only talk to other devices on the same network. The Internet Telephony Service Provider (ITSP) gives its clients a pipe, or a “trunk,” that makes the connection to the PSTN, thereby exponentially increasing the reach and usefulness of their PBXs. One example of a SIP trunking service is our own, OnSIP trunking. Most SIP trunking providers charge by the “trunk,” where each trunk has a simultaneous call capacity. OnSIP does not charge per trunk.
This is the most common definition of a SIP provider today. However, there are now lots of new entrants to the field offering innovative services making use of the SIP protocol as a backbone. These nimble ITSPs are able to build on the traditional SIP trunk service and create entirely new business models; these new SIP providers are changing the face of the VoIP industry.
Hosted SIP (PBX)
While a SIP trunk only caters to firms who already have their own IP PBX system built and are ready to hook up to the PSTN, the latest breed of SIP providers don’t require their clients to have anything other than phones. These ITSPs have their own PBXs hosted on their servers and allow businesses to make use of them, delivering VoIP services at extremely low prices. Hosted PBX services come with built-in connections to the PSTN network, eliminating their clients' need to purchase trunking separately.
There are also many SIP services that are targeted toward private individuals, such as sip2sip.info and getonsip.com. These offerings allow people to get their own SIP accounts for free. Sip2sip.info offers a PSTN connection, while getonsip is currently SIP only, meaning you can only make calls to SIP addresses.
Additional Benefits for Hosted SIP Customers
Here at OnSIP, we offer SIP hosting, which puts the full benefits of the technology within the reach of firms who have no SIP infrastructure of their own. Along with the ability to make VoIP calls, businesses can get SIP accounts to distribute to their employees, which allows anyone using a SIP device to call them for free as long as they know the SIP address associated with that account.
A SIP address looks just like an email ID. By default, the domain of the email—the part just before the “.com”—belongs to the ITSP providing the service. When we host your VoIP communications, your SIP address will look something like “email@example.com.” However, we also give you the ability to change the SIP address domain to match the corporate email ID that your employees use.
OnSIP is a SIP service provider. With an OnSIP account, you can open your domain to SIP traffic. This will allow you and your team members to be reached by phone using email addresses too!
You can make and receive SIP calls to and from any SIP address. Calls to users on our network are all free, since they are all SIP calls. We also offer free SIP accounts.
Check out our Features section for more info.
SIP Services for Developers
Many application services providers have leveraged the SIP protocol to build real-time communications (RTC) applications as SIP has become the de facto VoIP standard. In the past, building an RTC application has been a hefty undertaking. However, as it requires building voice, video, and transport engines in the application so that it is able to send and receive communications. It also requires a SIP platform (server-side) that can locate application endpoints and set up communications between them. Recently, there has been a breakthrough in browser technology that has made SIP application building easier for web and mobile developers: web real-time communications (WebRTC).
WebRTC enables web and mobile browsers to send and receive real-time voice, video, messaging, and data. That is, WebRTC-compliant browsers contain voice, video, and transport engines so that developers can use both SIP and WebRTC to build RTC features within any application. Using WebRTC, web and mobile developers can not only create new communications applications, but they can build RTC features right into existing applications. (Think of a taxi app that now allows you to call the cab driver right inside the application—no phone number exchange or dialing necessary.)
However, developers still need a SIP platform to locate application endpoints and set up communications between them. That is where a SIP provider comes in. OnSIP offers hosted SIP services for developers.