Global Crossing Enhances Its Suite of VoIP Services

Global Crossing, a  global IP solutions provider, today announced the addition of the Session Initiation Protocol (SIP) Refer feature to its Voice over Internet Protocol (VoIP) services portfolio. This feature enables enterprises and IP-based application service providers to use their IP networks more efficiently while transferring incoming calls to the location or party best able to serve the caller’s needs.

SIP Refer is designed to allow IP equipment to perform a call transfer automatically, and without attendant intervention, to any on-net or off-net phone number while removing the company’s equipment from the call path. Once a call is transferred, a port on the customer equipment is released, as well as the capacity being used on the IP network. This capacity is free to be used for other calls.
 
"By providing a migration path away from legacy features and traditional TDM equipment, we’re helping companies move to more cost-effective IP-based communications," said Gary Breauninger, Global Crossing’s chief marketing officer. "Our customers have expressed strong interest in these types of VoIP applications."
 
Global Crossing has released SIP Refer as a feature of its suite of VoIP services, including Global Crossing VoIP Outbound, which processes traditional long distance and international long distance; Global Crossing Toll-Free; Global Crossing VoIP Local Service; and Global Crossing VoIP On-Net Plus, which provides customers on-net to on-net calling capabilities that support cross-protocol routing schemes and network-defined private dialing plans. In addition, Global Crossing VoIP Consulting Service provides customers access to professional VoIP engineers and Global Crossing’s state-of-the-art VoIP Interoperability Labs for equipment, call flow and feature testing; security, protocol and IP Signaling compatibility; and specialized customer requirements.
 
Global Crossing's global, fully meshed MPLS-based network ensures that VoIP calls are delivered with minimal latency, packet loss, and jitter – a consistent and predictable call quality not possible with voice services based on public Internet transport.