[posted By Mae Kowalke on TMCnet] That’s the expected global growth rate of Session Initiation Protocol (SIP) trunking over the next five years, according to a recent article by Hamid Qayyum, VP of MSO Sales at Metaswitch.
SIP trunking, which delivers Voice-over-IP (VoIP) and streaming media services to customers with unified communications software applications or SIP-based private branch exchange system (IP-PBX), is expected to grow from 2.6 million SIP trunks in the first half of 2011 to nearly 22.3 million by 2015.
“SIP trunking is now the fastest-growing area of the VoIP marketplace, and all indications point to sustained growth for the foreseeable future, making it a very attractive business for both telco and cable providers,” Qayyum said in the article.
Providers must be ready for the opportunity, however; SIP trunking requires a solid infrastructure if providers plan on carrier-grade SIP products.
“There are many moving parts in a carrier-grade VoIP network, and service providers must carefully plan and implement their SIP trunking offering with an end-to-end view of how they will provision, deliver and support their service,” Qayyum noted.
Key SIP trunking deployment components include SIP trunking application feature server and media server, PSTN-connected media gateway or softswitch, session border controllers, backbone IP network, backend provisioning systems and network management server, he noted.
There also are challenges operators must overcome if they are to offer SIP trunking, including quality of service (QoS), interoperability and the ability to support and troubleshoot the solution from end to end.
Unlike TDM-based networks, “where each call is dedicated the required resources to facilitate the call on a channel basis,” Qayyum noted, SIP trunking has the potential for QoS issues when VoIP data packets exceed the capacity of the network and become delayed or dropped, hurting call quality.
The two common approaches operators usually employ is putting VoIP and data traffic on separate Internet connections to ensure bandwidth for VoIP, and the use of QoS routers that throttle back TCP data flows when needed to allow more inbound VoIP packets, Qayyum said.
Interoperability is a second challenge. Although SIP is a standard, early IPPBX systems implemented it differently. The SIPconnect 1.1 standard has largely solved the problem going forward, he noted, but there’s no guarantee that all features will work even on SIPconnect-certified devices.
The leading solution, he notes, are SIP normalization device deployed between a customer’s premises and the IPPBX IAD and the service provider’s SIP network. These normalize the transmissions by using SIP header customizations.
Operators also must be able to troubleshoot the full system to offer quality SIP trunking products, from the core application server through the core and access networks to the demarc router or gateway and into the customer’s local area network, Qayyum stressed: “The specific product selected does not matter quite so much, so long as it can provide the crucial detailed historical data for all calls, all the time.
Other issues to consider include whether to provision SIP trunking by concurrent calls or offer burstable SIP trunking groups to let customers scale their usage during peak times, as well as whether to offer high-value add-on applications such as hosted voicemail and conferencing.
“While it is impossible to predict the future,” he explained, two things are certain. First, “the majority of business customers are still using legacy TDM trunking to connect to the public network.” Second, “the vast majority of business customers understands the value proposition and plan to move to SIP trunking in the next five years.”
The SIP trunking explosion is under way.