Nearly five years ago, we implemented Voice-over-IP (VoIP) at Compugen based on Cisco technology, converging our voice network into our IP data network and issuing Cisco IP phones to roughly 800 Compugen staff across Canada. Our configuration is based on a centralized model, with a single instance of Cisco Call Manager (plus Cisco Unity for voice mail) in Richmond Hill serving all Compugen branch locations across the country. Until recently, connection to the Public Switched Telephone Network (PSTN) was done locally at each branch through local routers and Primary Rate Interface (PRI) connections to local Allstream switching offices (COs). Call management (call initiation/termination, etc.) is handled centrally by Call Manager, by way of signals over Compugen's WAN to the local router in each branch office, while the call itself takes place over the local PRI to the PSTN. Although we had positive end-user feedback as a result of our migration to the new unified messaging system, we found that managing the infrastructure was not as efficient as we would have liked. For example, we were renting three T1 PRIs to connect the head office to the CO, each with 23 channels feeding our phone system. When we needed additional capacity, we had to acquire another complete 23-channel PRI, even if we only needed one more channel. In addition, we could not easily re-assign phone numbers to end users across PRIs. So when we opened our new head office building in Richmond Hill, we incorporated advanced VoIP technology in the form of SIP Trunking. In effect, the SIP (Session Initiation Protocol) trunk connects Call Manager to the CO and thus to the PSTN via our WAN service connection rather than through PRIs–by allocating a portion of our WAN bandwidth just to voice traffic. As a result of no longer needing PRIs at head office, we are realizing significant cost savings. In addition, we can now acquire additional capacity from our service provider in any increment needed; and, by virtualizing the connections to the CO, assign phone numbers to end users without reference to PRIs, thus easing end-user management–any phone number can be assigned to any user in any location. When we're ready to roll out SIP Trunking to our branch offices, thus eliminating all PRIs, we can implement it on the existing WAN service connection to the local CO at each location (as we did in head office) by simply expanding the WAN bandwidth to include SIP traffic. In the meantime, we've just successfully eliminated some Quality of Service (QoS) glitches we initially experienced with our voice traffic by working with Allstream to tune up our CO connection. Feedback has been positive and we're pleased with the transition.
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Posted By Dean Reid December 10, 2012 in
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