SIP trunking and the next generation of enterprise voice services

By: Baiju Mandalia

Most of us have taken advantage of voice and video calling over the internet at no additional cost, while the typical cost for an analog phone line for business can be as high as $50 with taxes. Similarly, the standard go-to for business phone systems has been Time-Division Multiplexing (TDM), and trunks for 23 sessions can be as high as $500 with taxes. This is where Session Initiation Protocol (SIP) trunking comes into play, providing flexible delivery from carrier to customer and at a lower cost.

What is SIP trunking?  

SIP is a handshake mechanism between two points in a voice network to establish a phone call; SIP trunking is how voice over internet and the phone jacks over copper come to meet. SIP became an IETF standard in 2002 as rfc3261; but before SIP, there was another standard that played a role in its evolution.

Voice over IP as we know it today began in 1996 as a standard (called H.323) to make calls over the internet. Services we now know popularly as Skype or IBM Sametime voice had early versions of voice over IP and used H.323 to bypass the traditional phone lines and enable voice conversations over the internet.

Voice codec continued to evolve during the 1990s – voice codecs encode, decode and digitize voice for transmission as packets over the network with low latency and jitter while still achieving acceptable quality voice. G.711 and G.729 are just some of the many standards that have evolved to support packetized voice over IP.

Why SIP trunking?

It wasn’t long before carriers realized the value of network-based voice and took steps to make it part of their core infrastructure, delivering voice over their backbone data networks. Most enterprises connected to carriers using TDM voice trunks, termed as T1 or E1 based on country-specific standards and Integrated Services Digital Networks (ISDN) Primary Rate Interface (PRI), where more advanced signaling standard is adopted. SIP trunking delivers packetized voice sessions over data network circuits between the carrier and the customer as an alternative to the more expensive TDM trunks.

Why IBM Network Services for SIP trunking?   

SIP trunking has matured greatly since 2010 and the market is continuing to grow. SIP trunking–related network services can provide differentiation for infrastructure services customers. To take advantage of the SIP price structure, there are business requirements and architecture decisions that needs to be evaluated to optimize the transition to SIP trunking. IBM developed a service offering in 2010 to provide a vendor-agnostic view of these services – both for the carrier services and network equipment technology. Since then, IBM has enabled SIP trunking for a number of customers.

One aspect of SIP trunking and its flexibility is the centralization of call traffic that actually reduces the number of total concurrent voice trunks needed by a customer at multiple sites. In some cases, carriers are not motivated to move to SIP because it means lost revenue. This is where IBM can make a difference. IBM offers telecom expense management and SIP assessment services to assist customers in choosing a carrier that is right for their business. Additionally, technology vendors have hundreds of features related to SIP enablement, which the customer needs to select to optimize their overall cost. IBM can provide neutral guidance to help customers make the right choice based on their business requirements.

Please join in this discussion and share your experiences with SIP, the cost tradeoff decisions, the benefits and reference customers who can showcase their experiences with SIP.

Ready to take the next step? Schedule a consultation with an IBM Network Services expert.

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