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.

Exemplary aspects concern apparatuses and methods for display-related analysis of call data in an IPBX. In an example embodiment, an apparatus is configured to route data/VoIP calls via a data-communications server. An interface circuit is configured to selected parameters of interest based on capabilities of a set of devices and generate subscription requests to subscribe the devices to the parameters of interest. A processing circuit is configured to generate call summary metrics from call event messages for calls routed by the server and with subscription requests being associated with the parameters of interest. The call summary metrics are evaluated in connection with the parameters of interest as subscribed to by the devices and results of the evaluation are provided to the communications devices.

In some embodiments, the present invention provides for a computer-implemented method, including: causing, by a specifically programmed computer call management communication system, to transform, over a computer network, computing devices of users, into corresponding specialized call management devices, by having each computing device to execute a specialized call management client software application being in electronic communication with the specifically programmed computer call management communication system over the computer network by utilizing SIP; where the specialized call management client software application generates specialized graphical user interfaces configured to allow each user to concurrently initiate and maintain, over the computer network, a plurality of voice communications of distinct types with other users, by, for example, allowing each user to independently and dynamically divert, in real-time, any voice communication of any type to any audio device associated with a corresponding specialized call management device of such user.

Techniques are described for determining when and how to re-route toll-free calls around a core telephony network experiencing significant disruptions. A core telephony network of a telephony service provider is monitored for anomalies. Upon detecting an anomaly, it is determined whether the anomaly is significant enough to warrant re-routing inbound toll-free calls around the core telephony network. The re-routing process comprises accessing an industry registry database including a plurality of toll-free numbers (TFNs) associated with a routing template comprising the instructions for routing toll-free calls. The routing template instructions may be switched to re-direct toll-free calls associated with one or more specific TFNs to a particular carrier. Within that particular carrier, the routing instructions may be switched to re-direct the toll-free calls away from the service provider's core telephony network to a network bypass cloud platform. The network bypass cloud platform may be configured to receive toll-free calls, determine a customer telephony network to which to route the received toll-free calls based on the TFN of the toll-free call, and route the toll-free call to the customer telephony network.

A method, computer system, and a computer program product for interpreting conference call interruptions is provided. The present invention may include analyzing a conference call including at least two participants. The present invention may also include identifying at least one interrupted segment in the analyzed conference call. The present invention may further include reconstructing the identified at least one interrupted segment of the analyzed conference call and integrating it with the uninterrupted (heard) portion of the conference call.

Methods and systems for replaying buffered audio of a telephone call are provided herein. In some embodiments, a method for replaying buffered audio of a telephone call, comprises buffering audio data associated with a telephone call when a quality of an established data connection decreases below a first threshold quality measure; playing the buffered audio data from the buffer; determining the quality of the established data connection exceeds a second threshold quality measure; and replaying the audio data from the buffer, wherein replaying the audio comprises modifying the playback rate of the buffered audio data during playback.

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.

Various example implementations are directed to systems, circuits, apparatuses, and methods for providing virtual computing services. According to an example embodiment, a system includes a set of computing servers configured to provide a respective set of virtual servers for each of a plurality of accounts. The system 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 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.

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.

Various example implementations are directed to circuits, apparatuses, and methods for providing virtual computing services. According to an example embodiment, an apparatus includes a data storage device and a set of computing servers communicatively coupled to the data storage device. The set of computer servers are configured to provide a respective virtual data center for each of a plurality of accounts. The respective virtual data center for each account is configured to provide virtual services specified in a respective settings file for the account stored in the data storage device. The virtual data center for at least one of the accounts includes a set of virtual desktops and a set virtual data-communications servers providing data communications services which may include Voice-over-IP (VoIP) services, and support services for each account as specified in the settings file for the account.