Reducing operational costs by moving from traditional telephony to Voice over IP (VoIP)
Voice over Internet Protocol (Voice over IP, VoIP) is a transmission technique that uses a family of protocols and methods to leverage the internet or any data communication bandwidth for voice call set up and transmission. This vastly reduces the traditional costs involved with maintaining and paying subscriptions for legacy circuits from age old telecom giants.
VoIP implementations use many protocols such as H323, SIP, MGCP, RTP, IAX, etc of which SIP (Session Initiation Protocol) has gained the widest market penetration.
Because of the bandwidth efficiency and low costs that VoIP technology can provide, since 2004 or even earlier, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new PBX lines installed internationally were VoIP!
SIP (Session Initiation Protocol)
SIP is a text-based protocol with syntax similar to that of HTTP, making it very robust and easy to troubleshoot. Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types and compression codec such as G711, G729, G723, GSM, iLBC
SIP Trunking / SIP Trunk is a Voice over Internet Protocol (VoIP) service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communication to customers equipped with private branch exchange (IP-PBX) facilities.
SIP Trunk with predictive dialer
A predictive dialer dials a list of telephone numbers and connects answered dials to people making calls, often referred to as agents. Predictive dialers use statistical algorithms to minimize the time that agents spend waiting between conversations, while minimizing the occurrence of someone answering when no agent is available.
The SIP Trunk short call surcharge problem when using predictive dialer
A recent article on Telecom Monthly reveals that Verizon noticed an “unprecedented” increase in calls from auto-dialers which have led to network congestion and call blockage. Rather than build, Verizon has drawn a line in the sand that it doesn’t want customers who make an unreasonably large number of short duration calls. Customers who fall into Verizon’s danger zone, risk being suspended.
Verizon is not alone in the war against short duration calls. In the past few months, AT&T, Qwest Communications, Global Crossing, and others have all targeted the unwanted side effects of the auto-dialer: in some cases refusing to accept dialer traffic and in others attempting to force the worst offenders off the network through surcharges and other penalties.
Surcharges can be levied at a rate of a penny or two per call attempt. A penny may not seem like much, but to call centers accustomed to getting free call attempts (when hanging up before an answering machine) or a fraction of a penny on a 6 second or less call, a penny is a huge amount of money: an effective 10 fold rate increase for some customers. Many call centers will face the very real threat of going out of business.
But if so many carriers all push the auto-dialer traffic off of their networks, doesn’t that make a great opportunity for someone? It might.
There is light at the end of the tunnel. A few carriers like VoIPInvite Inc. are offering products without short duration surcharges and this helps the contact center community harness the power of next generation technologies such as VoIP.
