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What is SIP Trunking? How Does it Work?

 In Blog, General, Technology, Unified Communications, Voice

Session Initiation Protocol (SIP) trunking is a method of taking an organization’s complex phone infrastructure and connecting it to the outside world via the internet and VoIP technology rather than physical phone lines (either analog or digital).

“Trunk” is an old telephony term referring to a location’s connection to the wider grid. “SIP” refers to the software protocol used to begin and end communication connections between devices.

That said, SIP trunks can also carry video and messaging data – so think of them as the general communication bridge to the outside world. They’re a common fixture of modern Unified Communications systems.

A SIP trunk is essentially a more correct way to say “SIP phone line” or “business VoIP line,” although it can do more than those terms imply.


What does SIP stand for?

SIP stands for “Session Initiation Protocol.” SIP is a set of software protocols for initiating and ending connections between devices to send multimedia data (typically via RTP) – it’s key to much of modern organization-level VoIP data transfer, though it also has other uses.

Alternative data trunking systems exist that work similarly, but use a different (typically proprietary) protocol, so they’re technically not “SIP” trunks.


What is SIP calling?

“SIP calling” is a slang term referring to VoIP (Voice over IP) at the organizational level: the process of converting voice audio into data packets and sending them over the same data connections an organization uses for internet access. The system streams them to receiving VoIP hardware, software, or a PSTN gateway (which uses the data to make a traditional phone call closer to the receiving user, a local rather than long distance connection).


What’s the difference between SIP trunking and VoIP?

While the two terms are often used interchangeably in casual conversation, they aren’t the same thing. VoIP refers (broadly) to the systems for carrying audio over the internet. SIP is the most popular software protocol used for initiating and ending conversations using that audio. SIP trunking is the term used for connecting office phone systems (using SIP, which not all VoIP systems do) to the internet for VoIP purposes.


Why use SIP trunks?

The biggest and best reason organizations use SIP trunks in the short term is cost savings. SIP phone connections tend to be far cheaper (up to 60-80% less) than analog (PSTN) service, so switching can save your business a lot of cash.

Another good reason is future-proofing. Right now when you make a VoIP call, there’s a good chance that it passes over traditional phone lines at some point in its journey to the recipient. However, that won’t always be the case – analog phone lines are being phased out. In a few years the analog network won’t be there to connect to, so you might as well switch sooner and start saving money now.


How much bandwidth does a SIP call use?

This can vary from system to system, but a good estimate of an active SIP call’s bandwidth usage is 64k/s. Because that number is pretty low, the more likely limitation on the number of concurrent calls on a sip trunk is the number of channel licenses the organization has.


What is a SIP channel?

While people sometimes ask “how many SIP trunks do I need?” the real question is “how many SIP channels do I need?”

Each SIP trunk connects one location to the outside world, while the number of SIP channels within that trunk determine the number of concurrent phone calls that can be active at a time. You typically need far fewer channels than you have extensions, as most extensions won’t be in use at the same time.

The great thing about SIP channels is that increasing or decreasing them typically doesn’t require corresponding hardware changes. While you’d have to add actual cables to increase an organization’s phone lines, channels are more of a licensing and bandwidth concern.


What is a SIP PBX?

More properly called an “IP PBX”, these replace traditional analog PBXs with a more compact & powerful digital alternative. They facilitate the connection of calls to and from your company’s private phone network through a SIP trunk (possibly with a Session Border Controller in between) over the internet, to phones around the world.

What is a PBX?


Cost-Effectively Upgrade Your Telephony with CX-E

Taking your phone and fax systems into the modern era with VoIP and (its fax equivalent) FoIP can improve efficiency and drop costs, but you can take things even further. Integrating CX-E, XMedius’s premier solution, leverages your existing hardware to deliver a host of powerful applications, such as automated attendants with voice recognition, Interactive Voice Response (IVR) programs, informal call center functionality, unified messaging and a lot more.

Reach out to our team to discuss what XMedius’s Unified Communications, Secure File Exchange, and Fax over IP solutions can do for you.



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