Patent classifications
H04M7/0057
Method, apparatus, and system for handover to circuit switched domain
Embodiments of the present invention provide a method, apparatus, and system for a handover to a circuit switched domain. The method includes: selecting an MSC capable of both CSFB and SRVCC for a UE; and selecting, if a location service of a CS domain triggers an SRVCC procedure for the UE when the UE has a VOIP service in progress, the MSC capable of both CSFB and SRVCC in the SRVCC procedure, so that the UE, after being handed over to the CS domain, accesses the MSC and implements the location service of the CS domain. The apparatus includes: a first selecting module and a second selecting module. The system includes a UE and the apparatus. The embodiments of the present invention enable the UE with the VOIP service in progress to be correctly handed over to the CS domain and implement the location service.
Dual-Mode Device for Voice Communication
In one or more implementations, a request is received at a client device to initiate a communication session with a selected contact using a communication service. One of a first communication network or a second communication network for the communication session with the selected contact is selected at the client device. The selection is based on the selected contact and user preferences. Next, the communication session is established with the selected contact using the selected first communication network or second communication network.
Internet protocol telephony with variable-length carrier systems
First call identification information is stored at a network-connected device. The first call identification information identifies a call between an initiating device connected to an Internet Protocol (IP)-based network and a called device connected to the IP-based network. The call identified by the first call identification information is established through a telephone carrier network. Second call identification information is stored in the network-connected device. A determination is made that the first call identification information and the second call identification information identify the call from the initiating device to the called device. The call is connected through the IP-based network so as to avoid the telephone carrier network based upon the determining that the first call identification information and the second call identification information identify the call.
SYSTEM AND METHOD FOR CONCURRENTLY JOINING VOICE AND WEB CHANNELS
Systems, methods, and other embodiments associated with concurrently joining voice channels and web channels are described. In one embodiment, a method includes establishing a voice session to communicate over an audio channel, wherein a live agent communicates audio voice signals with a user. In response to identifying an issue from the user, transmitting a navigation link wherein the navigation link, when activated, navigates a browser to a web page associated with the issue. A web session is established to communicate between the browser and the web page. The voice session and the web session associated with the user are linked together. A call controller may then communicate simultaneously with both channels since they are connected allowing a live agent to disconnect from the audio channel.
Programmatical PSTN trunking for cloud hosted applications
Novel tools and techniques are provided for implementing programmatical public switched telephone network (PSTN) trunking for cloud hosted applications. In various embodiments, a computing system may determine one or more first network interconnection characteristics associated with a first entity service provider within a call service network operated by a call network service provider. Based on the determined one or more first network interconnection characteristics associated with the first entity service provider, the computing system may cause a network provisioning application layer to establish one or more network interconnections between a first network associated with the first entity service provider and the call service network, in some cases, by establishing shared peering connections between the first network and the call service network. The shared peering connections may enable a plurality of customers of the first entity service provider to establish call service connections that are shared over the shared peering connections.
SYSTEM AND METHOD OF ENABLING AUDIO CONFERENCING IN LIEU OF VIDEOCONFERENCING
An apparatus or method implemented in a computer system. In one embodiment the method includes receiving, from a device, a request to join a video call, and determining, in response to receiving the request, whether the device meets predetermined requirements to join the video call via videoconferencing. The method further includes sending, in response to determining the device does not meet the predetermined requirements to join the video call via videoconferencing, a message to the device, wherein the message comprises a link to join the call via audio conferencing.
Managing Voice over Internet Protocol (VoIP) Communications
The disclosed embodiments include a computer implemented method for managing network communications. In one embodiment, the method includes gathering, using performance information packet (PIP) data packets, network performance information from a communications network that includes network performance information from a set of egress points between the communications network and an outside network. The method selects a network connection including an egress point and an egress packet path within the communications network to the egress point offering the best quality of service between the communications network and an outside network based on the network performance information. The method then establishes the network connection between the communications network and the outside network for routing communications.
Communication System
A method of controlling a call between first and second user terminals, the method comprising: during the call, detecting a failure of a connection between the first user terminal and a packet-switched network; and in response, causing the call to be conducted at least part way via a PSTN network, via a connection between the first user terminal and the PSTN network. The call may be initiated by the first user terminal (such that the first user terminal is the caller and the second is the callee). The call may be initially conducted, prior to the failure, via a packet-switched connection between the first user terminal and a packet-switched network. The method may be implemented by a client application run on the first user terminal. Alternatively, the method may be implemented by a server.
Limiting failure rate by serving through multiple channels
Systems, methods, and devices use a wireless device's capability to transmit and/or receive data over multiple communication pathways to improve data transmission quality. In the various embodiments, the same continuous data stream may be transmitted and/or received via different communication pathways. Different communication pathways may be established using different antennas of a wireless device, different wireless networks, different wireless communications protocols, and/or additional wireless devices. The continuous data stream may be transmitted and/or received via different communication pathways in a manner that enables the continuous data stream to be reconstructed from one or more of the different communication pathways. Additional communication pathways may be established based on user input indicating a voice call is high priority and/or approving the expenditure of additional resources.
Systems and methods for emergency communications
Described herein are methods, devices, media, and systems for managing emergency communications and providing seamless data extraction from a communication device by an emergency service.