A distributed voice applications system includes a voice application rendering agent and at least one voice applications agent that is configured to provide voice applications to an individual user. A management system may control and direct the voice applications rendering agent to create voice applications that are personalized for individual users based on user characteristics, information about the environment in which the voice applications will be performed, prior user interactions and other information. The voice applications agent and components of customized voice applications may be resident on a local user device which includes a voice browser and speech recognition capabilities. The local device, voice applications rendering agent and management system may be interconnected via a communications network.

A multi-port VOID telecommunications system that allows the user to gain access to telephone connectivity through the Internet by connecting directly to the Internet or by connecting to the Internet through the existing Internet connection of a computer or cell phone device. The present system includes an Ethernet port, a Wi-Fi receiver to facilitate the transmission and receipt of Internet protocol signals wirelessly, a USB plug connectable to the ATA, connectivity to a home monitoring network and connectivity to Bluetooth devices.

A dual-mode voice communications system includes a processor, and a first communication connection module, a second communication connection module, and a wireless transceiver module that are separately connected to the processor, where the first communication connection module is configured to realize a communication connection between the dual-mode voice communications system and an Internet Protocol (IP) base station, the second communication connection module is configured to realize a communication connection between the dual-mode voice communications system and an external communications device, the wireless transceiver module is configured to realize a communication connection between the dual-mode voice communications system and a wireless terminal device, and the processor is configured to process data transmitted by the IP base station, the external communications device, and the wireless terminal device.

Described herein is a system and method for controlling a remote computer over a remote session. In an exemplary aspect, the method comprises receiving, by an intelligent personal assistant (IPA), a speech-based instruction from a user of a mobile device, the speech-based instruction comprising a request to execute a task; in response to determining, based on analysis of the speech-based instruction, an intent of the user to initiate the remote session with the remote computer: identifying, by the IPA, a remote client on the mobile device for creating the remote session with the remote computer via the cloud infrastructure; generating, by the IPA, a command object that causes a software agent on the remote computer to execute the task; and transmitting, by the IPA, the command object through the remote session to the remote computer; and presenting results for execution of the task in an interface of the remote client.

A multi-port VoIP telecommunications system that allows the user to gain access to telephone connectivity through the Internet by connecting directly to the Internet or by connecting to the Internet through the existing Internet connection of a computer or cell phone device. The present system includes an Ethernet port, a Wi-Fi receiver to facilitate the transmission and receipt of Internet protocol signals wirelessly, a USB plug connectable to the ATA, connectivity to a home monitoring network and connectivity to Bluetooth devices.

A call control entity communicates with Internet Protocol (IP) phones. The call control entity creates a mapping between a headset identifier of a headset connected to a registered first IP phone, and an identifier associated with the first IP phone. Upon receiving a request from a second IP phone that is not registered, the call control entity determines there is no existing IP phone record for the second IP phone. Responsive to the determining, the call control entity acquires from the second IP phone the headset identifier, and accesses an first IP phone record of the first IP phone based on the headset identifier and the mapping. The call control entity creates a second IP phone record for the second IP phone, copies existing configuration information from the first IP phone record to the second IP phone record, and configures the second IP phone with the existing configuration information.

Voice data transmission with adaptive redundancy creates a voice data packet by packetizing the voice data payload and a number of redundant payloads selected from a set of previous voice data payloads. The voice data from the voice data payload is analysed to determine whether it is a critical or non-critical payload by classifying the received voice data as voiced or unvoiced. If at least a portion of the voice data is classified as unvoiced, the voice data payload is determined to be a critical payload. If it is a critical payload, then the voice data payload is added to the set of previous voice data payloads for inclusion as a redundant payload in subsequent voice data packets. The voice data packet is then forwarded for transmission over the network.

A method of identity authentication for VoIP call, used for a first communication device supporting SIP and blockchain transmission technology, the method comprising: requesting an identity signature from a second communication device during a SIP call establishment, receiving the identity signature transmitted by the second communication device, wherein the identity signature is generated by the second communication device according to a private key, querying a public key corresponding to the identity signature from a blockchain server, utilizing the public key to verify the identity signature when obtaining the public key from the blockchain server, so as to generate a verification result, and determining to continue or terminate the SIP call establishment based on the verification result.

A method for computer control of online social interactions based on conversation processing. A portion of a conversation with a given user is recorded, stored in memory, and processed so as to influence subsequent interactions. These may include audio or other contextualized annunciations of subsequent telephone calls. Other modes addressing conversational processing for social purposes can result in interwoven conversation guided among threads of interest, advertising and incented participation of conversational content and placement. The invention is capable of implementation in telecommunications systems such as cellular, local exchange, and VOIP, and in combination with other forms of internet-based telecommunication, including smart phones and adaptive forums chat rooms.

According to an aspect, there is provided a method for monitoring a communication network, the method comprising: monitoring (701) user plane information and control plane information for a first session provided by the communication network for a first wireless device; and correlating (703) the user plane information and the control plane information to determine if the first session was ended due to a soft drop, wherein a soft drop corresponds to a user of the wireless device intentionally terminating the first session due to service quality issues.