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.

A variety of methods, systems, devices and arrangements are implemented for assessing and/or controlling call routing for Internet-based (e.g., VoIP/VioIP) calls. According to one such method, endpoint devices are used to monitor and/or assess the call-quality. The assessment is sent to a centralized server arrangement and call-routing is controlled therefrom. Endpoint devices employ a decentralized testing mechanism to further monitor and assess call quality including the use of test connections. Aspects of call quality are analyzed and attributed to endpoint devices and/or local connections or networks to distinguish intermediate routing issues from local/endpoint issues.

A voice over Internet protocol processing method and a related network device are disclosed. The method includes: detecting, by a calling access network element, an off-hook event; encapsulating, by the calling access network element, the off-hook event into first signaling; sending, by the calling access network element, the first signaling to a cloud server; receiving, by the calling access network element, second signaling sent by the cloud server, where the second signaling is used to establish a first transmission path between the calling access network element and the cloud server, and the first transmission path is used to transmit data between the calling access network element and the cloud server; and establishing, by the calling access network element, the first transmission path based on the second signaling. Overall efficiency of the voice over Internet protocol system can be improved, and the construction costs can be lowered.

A technique for acquiring and correlating session-related information from an Internet Protocol Multimedia Subsystem, IMS, is described. The technique comprises the acquisition of control plane information from control plane signalling tapped at an IMS control entity, the acquisition of user plane information from user plane traffic tapped at an IMS user plane entity, and the acquisition of context information from tapped signalling between the IMS control entity and the IMS user plane entity. The acquired context information permits to correlate the control plane information and the user plane information acquired for a particular session.

An impact of a delay time on a conversation is reduced. A voice packet is transmitted to a distribution server. A user is notified of being in a wait time from when transmission of the voice packet stops until when voice packet transmission becomes available next. The wait time is calculated on the basis of a first delay time that is a delay time between a local terminal and the distribution server and a second delay time that is the maximum delay time among delay times between a plurality of terminals including the local terminal and the distribution server. The user can easily know the speaking timing from the local terminal, and speech from the local terminal and speech from another terminal can be made not to mix.

Embodiments described herein provide a dynamic voice over Internet Protocol (VoIP) audio quality management mechanism in real time, e.g., when a VoIP call is ongoing. Specifically, when a VoIP call has unsatisfactory audio quality, e.g., due to packet loss, jitter, etc., the dynamic VoIP audio quality management mechanism may redirect the VoIP traffic from the previous endpoint that initiates the VoIP session to a different endpoint within the same carrier. Upon the endpoint redirection, a new call leg is established, allowing re-negotiation or re-configuration of VoIP parameters. The re-negotiated or re-configured VoIP parameters may then be used to conduct the remainder of the VoIP call to improve the audio quality.

In various examples, data communications are routed as calls by a set for servers, and the calls are processed in various ways including generating a set of data metrics including communications summary metrics which may related to communications event messages. At least one processing circuit is communicatively coupled to the server set which route incoming calls (e.g., for a plurality of agents in a communications/call center). The processing circuit is configured to receive communications event messages from the server set for communications routed by the server set, generate, during a communication to a first agent of the plurality of agents, a set of data metrics including communications summary metrics based on the communications event messages; and redirect, during the communication to the first agent, the communication to a second agent of the plurality of agents in response to the set of data metrics satisfying a set of criteria indicated in a policy.

Systems and methods for communications are disclosed. The systems and methods can monitor a cloud-based voice over internet protocol (VoIP) calling system to determine an active call. The systems and methods can also analyze the active call to determine an indication of call quality, the analyzing occurring during the active call. Additionally, the systems and methods can compare the indication of call quality to a quality threshold. The compare can occur during the active call to determine when the active call has a poor call quality. The systems and methods can also report the poor call quality based on the comparing the indication of call quality to the quality threshold.

Various example implementations are directed to circuits, apparatuses, and methods for providing virtual computing services. According to an example embodiment, an apparatus includes a set of computing servers configured to provide a respective set of virtual servers for each of a plurality of accounts. The set of virtual servers for at least one of the accounts includes a voice-over-IP (VoIP) server. The apparatus also includes a processing circuit communicatively coupled to the set of computing servers. For each of the plurality of accounts, the processing circuit provides a graphical user interface (GUI) including a mechanism for an authorized user of the account to select options for reporting usage of the respective set of virtual servers. The processing circuit also monitors use of the respective set of virtual servers for the account and generates invoices for use of the respective set of virtual servers according to the selected options.

A method for detecting user triggered call drops includes identifying one or more user terminated calls from a plurality of monitored calls. Signaling information associated with the identified user terminated calls is correlated with media channel information associated with the identified user terminated calls. A determination is made if termination of the one or more of the identified user terminated calls is related to quality of media across corresponding media channels. A predefined cause code is assigned to the one or more of the identified user terminated calls, in response to determining that the termination of the one or more of the identified user terminated calls is related to the quality of media across the corresponding media channels.