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IP Enabled Contact Center

Take your Contact Center to new levels

IP Enabled Contact Center (IPECC) is a portfolio of interaction services that includes IP Toll Free (IP TF) and IP Interactive Voice Response (IP IVR). This network-based portfolio provides Voice over IP (VoIP) conversion in the network for reliable termination to Session Initiated Protocol (SIP) devices. These products help customers attain efficiency gains associated with IP such as compression and dynamic bandwidth allocation.

A smooth migration extends the capabilities of your legacy Toll Free network. Your customers continue to dial an 8XX number (TDM) but the Network Gateway converts the Toll Free call into VoIP and allows for termination over standard access methods such as Internet Dedicated Access (IDA) and Private IP.

IP Enabled Contact Center

  • Replaces Take-back and Transfer (TnT) with SIP REFER and Interoperability assurances that emphasize SIP interfaces and certification of SIP-enabled vendors (ACD, PBX, VXML)
  • Controls transport/network cost savings by preserving ACD and Intelligent Call Routing (ICR) infrastructure
  • Leverages the benefits of VoIP

Unleash the power of IP technology

Under the IP Enabled Contact Center umbrella, the IP Toll Free (IP TF) and IP Interactive Voice Response (IP IVR) products provide foundational network-level services that are needed to transition your call center from a TDM to an IP infrastructure at your chosen pace. These IP Enabled Contact Center products are part of an evolving portfolio of new IP Contact Center products that help improve customer service, productivity, and business strategy, in addition to assisting you in controlling your costs.

  • Control costs with IP Networking, Application deployment and administration, and new Customer Premises Equipment (CPE).
  • Productivity enhancements may result from:
    • Integration of voice and data with multi-site, multimedia routing
    • Centralized management of distributed operations with virtual agents
    • Greater levels of redundancy for business continuity and disaster recovery
  • Customer Service improvements may be attained from:
    • Greater levels of integration for multiple customer touch points
    • The use of collaborative applications - advanced conferencing, presence, and interactive video
    • More consistent contact handling and prioritization across all media channels

IP Toll Free

  • Route incoming toll-free calls over IP for greater efficiency
  • Support global multiple-contact media simultaneously – phone calls, e-mail, and instant messaging
  • Use SIP capabilities to achieve real-time Internet communication

IP IVR

  • Process calls in a pure IP environment over a global network infrastructure
  • Choose from extensive call-routing and processing features
  • Terminate incoming calls to TDM and IP endpoints
  • Adopt IP technology at your own pace.
  • Leverage our VoIP

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Reap the benefits of IP Enabled Contact Center

IP Enabled Contact Center

FeatureDescriptionCustomer Benefit
Toll Free Origination Supports Traditional Toll Free/Freephone numbers around the globe. Leverages features, route plans and carrier-grade infrastructure of NANP and ITFS/UFIN networks.
IP (URI) and TDM Terminations Supports Traditional TDM (PSTN) and new IP terminations. Benefits of IP plus improved utilization of existing TDM Contact Center installations.
Network-based TDM to IP Conversion TDM-to-IP protocol conversion that is traditionally performed by a customer premises Gateway. Reduces the capital, scalability, and operational concerns associated with premises Gateways.
IP TF and IP IVR Service Continued support for the combined Toll Free and IVR services required by traditional contact centers. Full integration of Inbound TDM originations, IP conversion, IP call treatment and the capability to terminate calls to IP and TDM call centers – seamlessly and including agent transfers (IP IVR hybrid transfers as described below).
Session Initiation Protocol (SIP) signaling per IETF RFC 3261, etc. Open standard to set up, modify, and terminate voice calls which allows for interoperability between SIP-enabled devices. Greater compatibility and access to multiple types of termination systems (IP end-points) that are SIP compliant (i.e., SIP phones, PBX/ACD, IVR, etc.).
SIP Registration REGISTER is a type of SIP request that allows a device to identify its particular address with the network and allows IP to support personal mobility. Ability to support a SIP phone connectivity from any location (i.e., work from home agents).
SIP (REFER) Transfer A SIP REFER request enables the sender to instruct the receiver to contact a third party -with the contact details of the third party included in the request. IP CC supports SIP REFER transfers (blind transfers) between SIP devices. Note: SIP REFER Transfers must be initiated through SIP signaling (not DTMF), must terminate to another SIP device and cannot provide progress messages or error recovery.
IP IVR Hybrid Transfer Capability to transfer calls via DTMF and SIP (REFER) Transfer commands. Support agent transfers to IP or TDM terminations, where TDM terminations can initiate transfer via DTMF and IP terminations can initiate transfer via SIP (REFER) Transfer or DTMF.
Test Center Support Verizon Business live network allows you to test and get hands-on experience with proposed network, hardware and software solutions. Capability to test and verify critical IP functionality, actual IP TF and IP IVR terminations and transfers with the customer's associated CPE, before deployment in mission critical environments.
Service Controller Key system responsible for routing of the SIP calls, managing sessions, tracking bandwidth utilization, and registration of SIP phones and trusted entities (enterprise gateways, IP-ACDs, IP-PBXs, etc.). Provides common infrastructure that simplifies application deployment and will support future IP services such as "presence."
Session Border Controllers Systems which control real-time session traffic at the signaling, call control, and packet layers across network borders or between network segments. Enables the use of MPLS (Private IP) for access; will support other protocols (such as H.323) in the future.
SODEE (Service on Demand Execution Environment) Architecture Network capabilities that separate the application from the network and provide more choice for the customer. Provides future migration path to IP Multimedia Subsystem (IMS) for customer IP investments and new application solutions.

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IP Toll Free (IP TF)

FeatureDescriptionCustomer Benefit
Feature parity with Verizon Business Toll Free Service Advanced features allow you to tailor the origination, routing and termination of your IP Toll Free calls. Requires minimal migration redesign and allows the use of existing route plans to route calls via advanced features: TOD, DOW, Geographic, ANI, etc.
Toll Free Network Manager (TFNM) compatibility TFNM allows customers to modify their IP Toll Free routing structures from their desktops. Provides near real-time customer control over hybrid (TDM and IP) routing scenarios.
Voice over IP conversion in the network Network-based Gateways that convert TDM to VoIP. Eliminates the burden of owning and managing costly Gateway equipment.
Call transfers via SIP (REFER) Transfer A SIP (REFER) Transfer request enables the sender to instruct the receiver to contact a third party – with the contact details of the third party included in the request. Enables agents to make unattended transfers via IP without the use of DTMF tones and in a more cost effective manner.
Automatic Device Registration REGISTER is a SIP method used by a user-agent client to log in and identify its address with a SIP server, thereby letting the registrar know the address at which the user can be reached. Provides automatic resolution of SIP devices such as phones and ACD/PBX's (devices auto configure when connected to the network), allowing for simpler moves/adds/changes/deletes.
Compression – G.729 (Note: Music, DTMF or fax tones cannot be transported reliably with this codec) G.729 is a compression algorithm that compresses voice audio in chunks of 10 milliseconds, operates at 8 kbit/s (37 kbit/s including overhead) and is commonly used in VoIP. When G.729 is used in IP TF calls it increases data bandwidth efficiency (typically allows 41 concurrent calls vs. 23 utilizing TDM on the same T1).
Maximum bandwidth protection Bandwidth management tools to control the service, throughput and the efficiency of network facilities. Allows administrators to manage Quality of Service (QoS) and committed access rate (CAR).

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IP Interactive Voice Response (IP IVR)

FeatureDescriptionCustomer Benefit
Carrier-grade IP infrastructure for call treatment and routing Toll-Quality voice, fully redundant and managed network capacity. Provides high-capacity, native-IP IVR functionality that extends the benefits of IP CPE deployments without added protocol conversions.
Robust IVR call treatment and routing capabilities Support currently available network IVR features such as menu routing, transfer, message announcement and others. Allows the re-use of existing Network IVR (Hosted IVR-ECR) application call flow logic and minimizes migration design requirements.
Hybrid terminations (TDM and IP) and transfers Terminate to and transfer from both, TDM and IP end-points. Provides seamless continuity of service and allows customer to migrate to IP at their own pace.
Basic SIP Transfer, Full SIP Transfer, and Take Back and Transfer capabilities Agent-driven selection from available transfer types. Allows greater management control and more cost effective call transfers.
Simple system for migrating Hosted IVR - ECR applications Dedicated team and proven process to migrate network-IVR (ECR and other carrier) applications Allows existing customers to add IP functionality without rewriting complex applications and scripts
G.711 Codec Support (Recommended codec for Speech Recognition, Music, DTMF and fax tones) G.711 is a standard for audio "companding". It represents 8-bit compressed PCM samples for signals of voice frequencies, sampled at the rate of 8000 samples/second and 8 bits per sample. Provides Toll-Quality Voice (4.3 Mean Opinion Score) and leverages 64kbps bit rate (approximately 80kbps with overhead) to provide converged access with seamless DTMF and SIP transfers as well with reliable transmission for Speech Recognition and Fax applications.

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Services and/or features are not available in all countries/locations, and may be procured from in-country providers in select countries. We continue to aggressively expand our service availability around the world, please consult your Verizon Business representative for service availability. Contact us