SIP trunking is a method of delivering telephone and other unified communications services over the Internet to customers that have SIP enabled private branch exchange (IP-PBX) solutions. SIP utilizes both Voice over Internet Protocol (VoIP) and Session Initiation Protocol (SIP) and it replaces traditional telephone lines or PRIs (Primary Rate Interface).
Traditional business phone systems consist of two key components. The PBX, which provides call management and features such as Auto Attendants and voicemail, and the PRI lines which connect calls to the PSTN (Public Switched Telephone Network) where they are routed to the destination telephone. When SIP trunks are utilized, the IP enabled PBX connects to the data network instead of the PRI lines and the voice traffic travels the Internet to connect to the PSTN.
While there are many advantages to the VoIP SIP trunk approach, the primary drivers are cost and flexibility. SIP trunking eliminates the need for PRI lines and the associated cost. Unlike PRI lines, which contain 30 channels, SIP trunks can be purchased in increments as low as one channel, or one concurrent call. This gives businesses the ability to purchase and pay for only what they need and to easily scale as capacity requirements change. However with our pay per minute products we offer unlimited channels at ZERO cost.
We use only Tier-1 upstream providers to rout traffic for our customers. This means that our clients get the best quality voice along with a redundant platform that ensures performance and reliability. We employ multiple gateways across the World to eliminate any single point of failure.
Choose from cable, DSL, T-1 or Metro Ethernet, whichever works best for you. Each G.711 or ULAW/ALAW IP phone call will consume approximately 85kbps up/down across your network. We also allow the GSM compressed codec which cuts this number down to around 35kbps, but only recommend it in areas where bandwidth is at a premium.
Paying for your SIP VoIP trunk is as easy as setting it up. Our service is prepaid with no-contract and costs we offer a no contract rate of ksh3 per minute. You can add channels or DID anytime.
All plans come with one Kenyan DID but extra DIDs are available on request.
Moving to AfriQ SIP trunk is done without any disruption of service or existing infrastructure.
Our SIP Trunking service is a perfect fit for open source systems such as, Asterisk, FreeSwitch, Elastix, PBX in a Flash and other popular Graphical User Interfaces to configure and control Asterisk. We provide detailed ‘cut-and-paste’ trunk configuration settings enabling you to be up and running on our service in a matter of minutes.
We’re so confident that you’ll love our service, we offer 5 days POC account which include ksh 500 talk time of outbound calling to test our service.