Star Telecom is an industry leader in providing reliable, quality, outbound and inbound SIP Trunking for Contact Centers. We currently service clients dialing into Canada, US, China, UK and Australia. We are one of few providers to support both Caller ID and Caller Name for calls placed into Canada. As well, a number of SIP add-ons we provide such as SIP Call Recording and Enhanced Caller ID can be used in conjunction with outbound SIP service.
What is SIP Trunking?
Availability of carrier VoIP services has enabled customers to move away from traditional T1, PRI and DAL lines and instead order SIP Trunks or SIP PRIs, sometimes also referred to as 'Virtual PRIs'. The only real similarity of SIP Trunks to PRIs is that carriers often, for ease of communications and productization, bundle 23 or 24 channels into a SIP Trunk. There is no technical reason for this, since the trunk or channels are virtual an can be provisioned in any quantity. Star Telecom does not bundle channels in groups of 23 or 24, and since there are no charges for outbound channels, we make sure that there are enough channels to meet your outbound calling needs. For inbound SIP Trunking, our clients can order any number of channels which suits their needs.
A recent article on SIP Trunking concluded:
It [Moving to SIP Trunking] should be one of the easiest decisions that CIOs make, because it can dramatically cut costs and provide seamless interoperability with other network services
What are advantages of SIP Trunking?
SIP Trunking has several advantages of TDM PRIs:
SIP PRIs are virtual and not subject to access charges as is the case with traditional TDM telephony. With 'real' PRIs the local ILEC generally supplies the service, regardless of who resells the PRI to the end customer. The regulated pricing model prevents the market economy from taking its course. This is not the case with SIP Services. Data access is significantly more cost effective and for the price of a single PRI, it is possible to run 10 or more SIP trunks over the data connection which costs the same.
SIP Service is an IP service and therefore it is possible to architect it like other IP services, giving you more flexibility, redundancy, reliability and failover options. For inbound SIP Trunking, it is possible to share the channels across multiple call center sites or locations.
When you order one, two or ten PRIs the number of channels offered and paid for is fixed regardless of how much usage you actually have, and how efficient your utilization of the PRIs is. For example, most outbound contact centers, use their telecommunications infrastructure only in the afternoon and evening hours. This means the infrastructure is unused during most of the day. Regardless of volume fluctuations, traditional carriers always charge the same amount for the same number of fixed PRIs. With Star Telecom outbound SIP services, you are not charged for a fixed number of channels, trunks or PRIs. In fact, we do not charge for outbound channels at all. Star Telecom charges on per-minute usage basis - therefore eliminating the overhead and planning complexity associated with telecommunications services. Toll Free services are also not charged on per-channel basis, only the per-minute Toll Free rate competitive to what you pay today.
- Access to voice communications as a network based service opens the door to other voice-related functions and features such as call recording, call queuing and call routing to be used on a cloud-based model. This reduces the cost, time to deployment and total cost of ownership of these features. In other words, having access to SIP Trunking from Star Telecom makes it possible to use these features and functions on software-as-a-service (SaaS) basis which with traditional telephony usually take integration between vendors, installation, project management and are very expensive. As an example, Call Recording which can cost upwards of a thousand dollars per port (channel) can be accessed at a fraction of a penny per minute of recorded calls.
To sum up: cost savings, versatility, flexibility and alignment with other network services make SIP Trunking an obvious choice for call centers.
Why Focus on call centers?
Many carriers dislike the call center market due to aggressive dialing patterns produced by automated predictive dialers. Predictive dialers produce many short calls in very rapid succession, putting a significant strain on carriers' equipment and networks. This can results in Post Dial Delay or sometimes even more critical network failures. Star Telecom network is designed for exactly this type of traffic. In working with contact centers we have become experts in terms of integrating with various dialer and PBX equipment including Avaya, Cisco, Altitude, Asterisk, Vicidial, Aspect, Strata and others.
What equipment does the service work with?
SIP is a standard and should be pretty straightforward to deploy, however each developer/manufacturer may implement it slightly differently. Our service has been integrated with various releases of Avaya, Altitude, Cisco, Vicidial, Asterisk and a number of other platforms. In general, integration testing takes only a couple of hours. However nuances do exist with each manufacturer's implementation of SIP and we leverage our past experience to make the service work with your PBX or dialer.
What if my dialer / PBX does not support SIP?
If your PBX / Dialer does not support SIP, it is possible to SIP-enable it by placing a SIP gateway between legacy equipment and Star Telecom services. Most telephony and network manufacturers including Cisco, Dialogic and AudioCodes sell SIP Gateways which can be used to convert TDM to SIP. We can help you select the right gateway based on your needs or we can provide a managed gateway as a part of our services.
Contact us for a quote or to find out more about our services and solutions.
