A data processing device comprising: a jitter buffer for receiving data packets; a media decoder configured to decode the data packets so as to form a stream of media frames, each frame comprising a plurality of samples; a media consumer having an input buffer for receiving the stream of media frames and being configured to play media frames from the input buffer according to a first frame rate; a buffer interface configured to monitor the input buffer so as to detect when the number of samples at the input buffer of the media consumer falls below a predetermined level and, in response, generate a play-out request; and a media controller configured to, responsive to each of the generated play-out requests, play-out one or more data packets to the media decoder so as to cause media frames of the stream to be delivered into the input buffer at a rate commensurate with the first frame rate.

The present invention relates to systems and methods for improving transmission of voice packets over a network are provided. The systems and methods include a few central internet data centers (IDCs) which include a routing controller and an access controller. The system also includes a number of edge IDCs. Each edge IDC includes a last mile optimizer and a relay server. The last mile optimizer operates along with an application located on the users' devices and the access controller in the central IDCs to identify the best edge server for the particular device to connect to. The edge servers continually monitor pathway performance once the call is in progress. If an error is detected, then the server may automatically transition to back-up pathways rapidly to minimize call performance disruption.

A heartbeat packet timer identification method and a device, where the identification method is performed by a device that sets a timer, and the method includes determining, in a data packet transmitted by the device, each associated data packet corresponding to each timing end moment of the timer and determining, according to each associated data packet and each timing end moment of the timer, whether the timer is a heartbeat packet timer set by the device for transmitting a heartbeat packet. With reference to a data packet transmitted by a device and each associated data packet that corresponds to each timing end moment of a timer, the heartbeat packet timer identification method and the device may determine with relatively high accuracy whether the timer is a heartbeat packet timer.

According to one example, a method includes processing a communication session with a first virtual machine of a plurality of virtual machines associated with a network node and monitoring packet loss on a leg of the communication session between a first endpoint and a second endpoint. The method further includes, in response to determining that the packet loss exceeds a first threshold, toggling on an enhanced mode for a codec associated with the communication session, the enhanced mode providing increased error resilience. The method further includes, in response to determining that the toggling on the enhanced mode causes the first virtual machine to exceed a processing capacity threshold, moving the communication session to a second virtual machine of the plurality of virtual machines.

Systems and methods are described herein for determining dropped call rates (DCR) for various communications networks, such as IP Multimedia Networks (IMS), which include Voice over LTE (VoLTE) networks. For example, the systems and methods utilize data (e.g. abnormal cause codes) generated by layers of the IMS networks, such as a Session Initiation Protocol (SIP) layer of the IMS network, when determining dropped call rates for IMS networks.

The disclosed system provides a Real-time Telephony (or Call) Monitor, Analyzer and Decision SIP Server (RTMADS) for mitigating attacks on emergency telephone systems. The RTMADS works in conjunction with an ingress node to fork incoming calls to an IMS network and the RTMADS. Within the RTMADS, forked telephone calls undergo data collection and mining, and parametric analysis. A decision matrix in the RTMADS uses the results of the data collection, mining, and parametric analysis, and other information, to make a decision with respect to incoming calls. For example, the RTMADS may decide to perform call setup on an incoming call using a dedicated or backup Public Safety Answering Point (PSAP), alert an Operations and Management (OAM) team regarding the incoming call, or accept and then terminate the incoming call.

A value of a quality of service parameter is monitored based on one or more communication sessions with an agent communication endpoint of a contact center agent. For example, a jitter parameter is monitored in an audio communication session with the contact center agent. A determination is made if the value of the quality of service parameter does not meet a threshold level. For example, a minimum amount of jitter threshold. In response to determining that the value of the quality of service parameter does not meet the threshold level, the agent communication endpoint is prevented from receiving new communication sessions in one or more media types originally supported by the contact center agent. For example, the agent communication endpoint is prevented from receiving any new voice communication sessions while still being allowed to receive communication sessions in other mediums, such as Instant Messaging and email.

A failover system includes a plurality of configuration controllers and a plurality of switches. The switches include a site proxy, a failover module and a routing table. A first site proxy of a first switch is designated as the active proxy and controls failover in the event a switch fails or otherwise becomes unavailable. A second site proxy of a second switch is designated as the backup proxy and controls failover in the event the first switch fails or otherwise becomes unavailable. In the event a switch fails or otherwise becomes available, the active proxy interacts with the configuration controller and reassigns the extensions associated with the failed switch to a different switch of the plurality of switches. If a configuration controller fails or otherwise becomes unavailable, another configuration controller becomes active and establishes communication with the plurality of switches to maintain the operational status of the IP telephony system.

Systems and methods are described herein for determining dropped call rates (DCR) for various communications networks, such as IP Multimedia Networks (IMS), which include Voice over LTE (VoLTE) networks. For example, the systems and methods utilize data (e.g. abnormal cause codes) generated by layers of the IMS networks, such as a Session Initiation Protocol (SIP) layer of the IMS network, when determining dropped call rates for IMS networks.

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.