Abstract:
A request to authenticate a user is received at an authentication system. The request to authenticate the user requires one or more non-numeric metrics to authenticate the user. For example, the one or more non-numeric metrics may include a user name, a password, and a fingerprint scan. The request to authenticate the user that requires the one or more non-numeric metrics is validated. In response to validating the request to authenticate the user that requires the one or more non-numeric metrics, a first numeric code is generated (e.g., a password). The first numeric code is used to grant access to a telephone that requires authentication via a numeric keypad. The first numeric code is sent and displayed to the user. The user can then access the telephone by providing the numeric code at the telephone.
Abstract:
A provider system (e.g., a cloud based provider system) receives a workflow. For example, the workflow may be for handling a voice communication session in a contact center. The workflow comprises a plurality of workflow tasks. The workflow tasks comprise enterprise workflow tasks and provider workflow tasks. The identified provider workflow tasks are executed on the provider system according the workflow. The provider system initiates execution of the identified enterprise workflow tasks on the enterprise system according to the workflow. By allowing a split workflow between the provider system and the enterprise, exposure to sensitive information used by the provider system may be limited.
Abstract:
A call request is received, from a mobile device, to establish a communication with a contact center. For example, the call request may be to establish a voice call with the contact center. In response to the call request, the mobile device sends authentication factors to a cloud authentication service that the user/mobile device has previously registered with. For example, the authentication factors may include usage factors of the mobile device, such as a call history of the user on the mobile device. If the authentication factors are validated, a token is generated. The token is sent to the contact center along with the call request. The token is validated at the contact center. At this point, the contact center knows that the user/mobile device are authentic. A call is then established between the user and the contact center.
Abstract:
Communication servers hosting interactions between requesting devices and resources may require load sheading or maintenance and, as a result, be placed in Deny New Service (DNS) mode, whereby a subsequent session request is refused while existing sessions continue until concluded. However, an interaction comprising an existing session may be supplemented by a subsequent session, often utilizing a different application layer implementation. By providing the requesting device with an interaction identifier, a subsequent session made after the server has been placed in DNS mode, may be accepted and associated with the same resource associated with the existing session.
Abstract:
A communication system, method, and components are described. Specifically, a communication system having one or more Back-to-Back User Agents (B2BUAs) therein is described. The communication system also includes an RFC 4579 conference focus. Mechanisms are described which enable User Agents (UAs) to subscribe to conference state events and create ad-hoc conferences even though the conference focus is operating in a B2BUA environment.
Abstract:
Enhancing media characteristics during Web Real-Time Communications (WebRTC) interactive sessions by using Session Initiation Protocol (SIP) endpoints, and related methods, systems, and computer-readable media are disclosed herein. In one embodiment, a method comprises intercepting, by a media redirection agent of a WebRTC client executing on a computing device, a WebRTC initiation token. The method further comprises generating a SIP endpoint WebRTC token based on the WebRTC initiation token, and sending the SIP endpoint WebRTC token to a remote endpoint. The method also comprises establishing a WebRTC interactive session between the remote endpoint and a SIP endpoint based on the SIP endpoint WebRTC token. By leveraging the audio and/or video functionality of the SIP endpoint, the media characteristics of the WebRTC interactive session may be enhanced, resulting in an enhanced user experience.
Abstract:
A communication session is established between a first communication device and a second communication device. The communication session comprises a first dialog between an application server and the first communication device. The first communication device uses a first network address in the first dialog. In response to an event, such as a first network interface failing, a SIP INVITE with replaces header message is received by the application server with a second address of the first communication device. In response to receiving the SIP INVITE with replaces header message from the first communication device with a second address of the first communication device, the first dialog between the application server and the first device is reestablished using the second network address.
Abstract:
Providing Web Real-Time Communications (WebRTC) media services via WebRTC-enabled media servers, and related systems, methods, and computer-readable media is disclosed herein. In one embodiment, a system for providing WebRTC media services comprises a WebRTC-enabled media server including a scripting engine, a WebRTC functionality provider, and a control application programming interface (API). The WebRTC-enabled media server is configured to receive, from a WebRTC application server, a stream establishment application, and to establish, via the stream establishment application, a plurality of WebRTC interactive flows associated with a corresponding plurality of WebRTC clients. The WebRTC-enabled media server is also configured to apply a media service to one or more of the plurality of WebRTC interactive flows to generate one or more media server flows, and provide the media server flows to one or more of the plurality of WebRTC clients. The WebRTC-enabled media server may thus provide functionality via familiar WebRTC control interfaces.
Abstract:
Embodiments disclosed provide access to Traversal Using Relays around Network Address Translation (TURN) servers using trusted single-use credentials, and related methods, systems, and computer-readable media. In one embodiment, a method comprises receiving, by a TURN authentication agent, a request for a TURN server credential. Responsive to determining that the request is authorized, the agent generates a trusted single-use credential and transmits it to the requestor. Using this trusted single-use credential allows untrusted clients to access a TURN server without exposing a userid/password combination. In another embodiment, a method comprises receiving, by the TURN server, a request for a TURN service. The server challenges the request, and receives a userid and a password. Responsive to determining that the userid and the password constitute a trusted single-use credential and responsive to determining that the request is authorized, the server provides the TURN service for the requestor.
Abstract:
The system and method allow enhanced capabilities for Session Initiation Protocol (SIP) dialogs (communication sessions) between SIP devices. The SIP dialogs have applications that are inserted into the SIP dialog such as a Back-to-Back User Agent (B2BUA) or a proxy application. After the initial dialog is established and these applications fail or become unavailable, the system and method allow the applications to be bypassed or have the SIP dialog redirected to an alternative application. This provides a better user experience because the SIP dialog (e.g., a SIP telephone call) will not be dropped if the application fails mid-dialog.