Browse Prior Art Database

METHOD SYSTEM AND LOGIC FOR CLOUD CONFERENCING PSTN VIA VOIP

IP.com Disclosure Number: IPCOM000243669D
Publication Date: 2015-Oct-09
Document File: 5 page(s) / 352K

Publishing Venue

The IP.com Prior Art Database

Related People

Shawn Rolin: AUTHOR [+3]

Abstract

A mechanism is provided to leverage a new audio connector for a cloud conferencing system when a participant selects a service to callback via audio The new approach preserves the experience without requiring audio connectivity The mechanism uses Direct Inward Dialing DID to send the audio call via a native Voice over IP VoIP codec Over The Top OTT to the associated audio connector which then forwards the audio call to Unified Communications Manager UCM for routing to a Unified Communications UC endpoint

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METHOD, SYSTEM AND LOGIC FOR CLOUD CONFERENCING PSTN VIA VOIP

 AUTHORS: Shawn Rolin Chris Barwick Graeme Geddes

CISCO SYSTEMS, INC.

ABSTRACT

    A mechanism is provided to leverage a new audio connector for a cloud conferencing system when a participant selects a service to "callback" via audio. The new approach preserves the experience without requiring audio connectivity. The mechanism uses Direct Inward Dialing (DID) to send the audio call via a native Voice over IP (VoIP) codec "Over The Top" (OTT) to the associated audio connector which then forwards the audio call to Unified Communications Manager (UCM) for routing to a Unified Communications (UC) endpoint.

DETAILED DESCRIPTION

    Cloud conferencing systems provide for a great customer experience but may create a significant increase in cost due to the shift of audio connectivity from on-net to off-net. Customers want to consume the service without paying charges for the large majority of participants that are on their own network.

    In a cloud conferencing system such as Cisco's WebEx┬«/Spark Audio Fusion, when a cloud conference is started and a participant selects the service to "callback" via audio the system checks whether DID is available and registered by any audio connector. DID is a feature used in private branch exchange (PBX) systems where a range of telephone numbers is associated with one or more telephone lines. DID allows a personal number to be assigned to each participant, without requiring a separate physical telephone line for each participant to connect to the PBX. If DID is available, the cloud conferencing system sends the audio call via a native VoIP codec OTT to the associated audio connector which then forwards the call to a UCM such as Cisco's Unified

Copyright 2015 Cisco Systems, Inc.
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Communications Manager (CUCM) for routing to a UC endpoint. Once the audio call has been routed it rings the associated UC endpoint (for example a 7800 or 8800 series Cisco UC endpoint).

    Should DID not be found registered to any audio connector, the cloud conferencing system sends the audio via the native VoIP codec OTT to the associated audio connector of the hosting members subscribing company, i.e., audio is delivered over the Internet without the involvement of an operator of the cloud conferencing service in the control or distribution of the content. Once there, the call is routed via the UCM leveraging any/all Tail End Hop-Off and/or cost based routing to egress at the closest geography to the requesting call back number. The audio call egresses via a SBC (Session Border Controller) which, can if required, transcode the audio from the native VoIP codec into the codec required for accessing the public switched telephone network (PSTN).

FIG. 1 illustrates an existing cloud conferencing system as it is used today.

FIG. 1

    In existing cloud conferencing systems, hard endpoints still provide a better audio experience than PC based software due to high-quality p...