SIP Trunking: Ready for Prime Time

Now, thanks to SIP trunking, companies can extend IP communications beyond the enterprise to reach customers, partners and suppliers in the most cost-effective manner.


Today, many businesses are leveraging Voice over Internet Protocol (VoIP) to reduce costs, increase flexibility, enable collaboration, and gain a competitive advantage in an increasingly global and dispersed marketplace. VoIP reduces the cost of intra-company calls by running voice traffic over a company’s own network, and by making it easier to centralize management and perform tasks such as moves, adds and changes (MACs).

In Frost & Sullivan’s 2011 survey of 205 C-level executives in North America, 28 percent of respondents say their organizations have deployed IP PBXs, primarily to reduce costs and support employee mobility. But until recently, most organizations had to connect to the Public Switched Telephone Network (PSTN) over traditional phone lines, which leads to higher costs and limits the benefits of IP communications to inside the firewall, thereby reducing its overall value.

Now, thanks to SIP trunking, companies can extend IP communications beyond the enterprise to reach customers, partners and suppliers in the most cost-effective manner. Based on Session Initiation Protocol (SIP), the clear standard for enterprise communications in the 21st century, these services can help lower costs and boost productivity for the companies that use them. No wonder Frost & Sullivan research finds that more than 7 million North American users had adopted such services by 2010 and that the number of North American SIP trunking users is expected to reach 59 million by 2017.

Simply put, SIP trunking connects a company’s internal phone system to the Internet, allowing voice traffic to travel over the providing carrier’s data network. That results in significantly lower costs for the provider, which it can then pass on to the customer. In the process, SIP trunking eliminates the need for PSTN and media gateways as well as BRI/PRI connections.

For a flat monthly fee, most SIP trunking providers offer local dial tone, long-distance calling, centralized call-management and control features, and four-digit extension dialing within an organization. More robust calling features, such as hosted auto attendant, voicemail, unified messaging, contact centre capabilities, mobile extensions and data services, may also be made available by a SIP trunking provider.

By enabling centralized application management, SIP trunking drives operational efficiencies and makes it easier to support advanced unified communications and collaboration applications, including presence/chat, web and video conferencing, and advanced social networking capabilities.

In 2010, IP line licenses accounted for approximately 74.3 percent of total telephony licenses shipped globally and are expected to account for close to 93 percent in 2017. By 2011, around 40 percent of North American businesses had deployed IP telephony systems with IP endpoints. So companies that deploy SIP trunking sooner rather than later can expect to get a jump on their competition by lowering costs, improving productivity and giving their employees access to the next generation of communications and collaboration services.

SIP Trunking  by the Numbers

  • 7 million North American users had adopted SIP trunking by 2010.
  • 59 million North American SIP trunking users are expected by 2017.
  • 143% growth in SIP trunking revenue in 2010 in the U.S.
  • 220% growth in adoption of SIP trunking services globally in 2010.
  • 75% of Fortune 100 companies use SIP trunking.
  • 74.3% of total global telephony licenses shipped in 2010 were IP line licenses.
  • 93% of those licenses are expected to be IP-based in 2017.
  • 40% of North American businesses had deployed IP telephony systems with IP endpoints as of 2011.
  • 28% of North American C-level executives surveyed said that their organizations deployed IP PBXs, primarily to reduce costs and support employee mobility.

(figures courtesy of Frost & Sullivan of Infonetics Research)


2 Comments

  1. I still have some difficulty with SIP Trunking and its primarily due to the issue with 911and its limited feature set and reliance on the IP network. You back up PRI with analog and I would back up SIP with PRI.

    When you call 911 on copper lines the local PSAP answers the call and will dispatch emergency services to the address assigned to the copper line even if the caller cannot speak. However, I understand on SIP there is a call center that answers the call before you even get to PSAP that will want to determine your physical address. If you cannot speak then how do they know where to send help?

    Brian Lemish / 7 years ago
  2. The situation you are describing is very important for all forms of VoIP, not just SIP Trunking. In circuit-switched networks, the service location is fixed, so that when a 911 call is made, the address information pre-loaded at the PSAP is displayed to the agent based on the ANI digits delivered with the call. In VoIP, the user may be accessing the service from anywhere using the internet. This is referred to as ‘nomadic’ VoIP when an endpoint is not fixed at a single location. For that reason, VoIP service in Canada routes to a 911 triage centre where the operator asks where the caller is calling from. If the caller cannot speak, then emergency services are dispatched to the default location on file for that subscriber. This is why it is very important to always ensure that default address information is always kept up to date. However, even if the default information is up to date, it may not reflect the true location of the call’s origin. This is a limitation of VoIP in Canada that is recognized by the CRTC. Currently there is a CRTC industry working group working on an implementation of Enhanced 911 for VoIP that will include address information of nomadic endpoints. This has been in the works for some time, but expect to see new information coming out on this within the coming year.

    Grant Bykowy / 7 years ago