H04M7/006

Method and apparatus for supporting internet call sessions in a communication network

Aspects of the subject disclosure may include, for example, including a processing system for performing operations for determining service requirements of a call session at first user equipment associated with a communication network, determining a first codec to facilitate the call session at the first user equipment according to the service requirements of the call session, searching a session border controller table according to the first codec to obtain a first resource identifier associated with a first session border controller type to facilitate the call session at the user equipment, receiving a first address of a first session border controller associated with the communication network from a domain name server associated with the communication network responsive to a first query including the first resource identifier, and sending a first transport protocol message to the first session border controller according to the first address. Other embodiments are disclosed.

SITE LINK TESTER VIA UNIQUE PHONE EMULATION
20230208967 · 2023-06-29 · ·

Remote on-demand site link testing is provided. A site link tester (SLT) system includes an SLT connected to a customer's VoIP phone system. The SLT is configured to communicate with a front end client application operating remotely on a user's computing device. The packet-capture application receives instructions from the client application to perform a packet capture in association with the SLT's network interface and/or to operate as an emulated VoIP endpoint and conduct a test call (e.g., to confirm the customer's VoIP system's compliance with 911-associated legislation or to troubleshoot a VoIP issue). Results of the packet capture may be sent to the client application and analyzed for remotely diagnosing and troubleshooting VoIP-related problems. Using the SLT system, the technician is enabled to perform 911-associated legislation compliance and diagnose VoIP issues on-demand from a remote location, which can reduce or eliminate the need for a technician to be on-site.

Detecting, verifying, and preventing unauthorized use of a voice over internet protocol service

A computer-implemented method, a computer program product, and a computer system for detecting, verifying and preventing unauthorized use of a Voice over Internet Protocol (VoIP) service. A computer rates a VoIP call based on a database including information of the caller number, in response to determining that no record of a caller number exists in a database including the information of unauthorized uses. The computer sets a predetermined time period for the VoIP call based on a rating of the VoIP call, adds the predetermined time period to a session initiation protocol (SIP) invite, and connects the VoIP call to a called party. In response to that the predetermined time period is reached, the computer interrupts the VoIP call and prompts the caller to conduct user verification. In response to that the caller is successfully verified, the computer reconnects the VoIP call to the called party.

INMATE CALLING SYSTEM WITH GEOGRAPHIC REDUNDANCY
20170366674 · 2017-12-21 ·

A geographically redundant inmate calling system is described. The geographically redundant inmate calling system includes two or more session border controllers that share session state information and provide for automatic failover in the event of the loss of availability of one session border controller. The geographically redundant inmate calling system eliminates any single point of failure such that communication services are highly available to inmates.

CONCURRENT COLLABORATION CONFERENCE PORT MANAGEMENT

Aspects of the present disclosure involve systems and methods for a collaboration conferencing system to track a total number of concurrently utilized ports across any number of conferencing bridges of the network for a particular customer and one or more billing actions may occur based on this tracking. This may result in an alternate billing option for the customer's use of the system. Further, a telecommunications network administrator may provide access to the collaboration conferencing system based on a total number of concurrently utilized ports rather than on a per conference or per minute basis. With the information of the number of purchased ports by the customer, the administrator may more accurately predict an available capacity for the collaboration conferencing system needed to support all of the users of the system and the potential collaboration conferences.

IDENTIFYING THE SOURCE AND DESTINATION SITES FOR A VOIP CALL WITH DYNAMIC-IP ADDRESS END POINTS

In a voice-over-IP communications network, call data records include dynamically assigned IP signaling addresses such as IPv6 signaling addresses used in provisioning communications sessions. Those dynamically assigned IP signaling addresses are computed from customer site identification codes using a reversible algorithm. The algorithm can then be reversed to compute a customer site identification code from an IP signaling address contained in a call data record, allowing the communications network provider to perform quality monitoring and diagnostics based on call data records.

Content delivery system

The audio outputs 29 of each of a plurality of telecommunications terminals in a packet switched system are synchronized by having each terminal transmit a stream of packets whose rate, and therefore duration is determined according to a clock generator 25, and is thus indicative of the rate at which the terminal is generating its audio output. The signals from each terminal are transmitted to a common server. For each terminal, the server uses a master dock to compare the duration of the packet stream with an expected duration, and calculates an offset value which is returned to the respective terminal. Each terminal stores the offset value it receives (20) and uses it to adjust the output of its clock generator 25 so that its operations can be synchronized to the server. This allows all the terminals' digital-to-analog conversion processes to be synchronized such that all their analog outputs are coordinated, allowing co-located acoustic outputs to be synchronous. A second embodiment maintains synchronization by maintaining the volume of the data buffer 23 serving the audio output 29 within predetermined limits.

Method and apparatus for selecting domain service in wireless communication system

According to an embodiment, a network access method of a user equipment (UE) in a communication system comprises the steps of: receiving information including an access order from a configuration server; storing the received information; and transmitting an access request to the network according to an access method determined on the basis of the stored information. According to another embodiment, a UE for accessing a wireless communication system comprises: a transceiving unit which receives information including an access order from a configuration server; and a control unit for storing the received information and controlling the transceiving unit to transmit an access request to the network according to an access method determined on the basis of the stored information. According to an embodiment of the present disclosure, when a UE for supporting a PS network and a CS network accesses a network, the UE receives a configuration of each domain registration method in advance, and thus can more easily access each network. In addition, according to another embodiment of the present disclosure, message transmission can be more easily achieved between a system in which a long SMS message can be transmitted and a system in which a SMS message of conventional-length is transmitted. Furthermore, according to another embodiment of the present disclosure, when a UE, which has performed a CS fallback, returns to a 4G network due to the end of a CS service, the UE can be prevented from returning to another provider network since the UE has existing return information.

REAL TIME ANIMATION GENERATOR FOR VOICE CONTENT REPRESENTATION

During a voice conversation, conversationally continuous voice input may be received from at least a first participant of the voice conversation. During the voice conversation, the conversationally continuous voice input may be converted into text, and the text may be analyzed to characterize content thereof. Using a graphical user interface displayed to at least a second participant of the voice conversation, a voice content animation may be rendered that visually represents the characterized content and is repeatedly updated as new content is characterized during the voice conversation.

System and method for managing access point functionality and configuration

A system for managing access point functionally and configuration includes a server that is coupled to a computer network and configured to communicate with an access point via the computer network. The access point is configured to couple a mobile device to the computer network by providing a wireless link between the mobile device and the access point. The access point is further configured to produce a status point regarding the access point and the server is configured to receive the status report from the access point following a trigger event at the access point. In other examples, the server is further configured to transmit a response message and/or a configuration file to the access point in response to the status report that is received at the server. Other features and systems are also disclosed.