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 terminal supports two cards the primary card and the secondary card, the primary card supports a data service, and the secondary card supports transmission of a voice service using a data service, or the secondary card supports concurrent transmission of a voice service and a data service. The method includes, when the terminal performs the data service using the primary card, if the terminal performs the voice service using the secondary card, switching, by the terminal, the data service performed using the primary card to the secondary card, and performing the data service using the secondary card.

A method is disclosed for forking call audio channels. The method includes, during an active call, receiving a first audio stream of first call audio that originates from a far-end telephony device, and receiving a second audio stream of second call audio that originates from a microphone of a near-end headset. Further, the method includes generating, during the active call, a third audio stream containing the first call audio by processing the first audio stream, and generating a fourth audio stream containing the second call audio by processing the second audio stream. Still yet, the method includes, during the active call, generating a fifth audio stream containing the first call audio, and generating a sixth audio stream containing the second call audio.

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.

Apparatus and methods for guaranteeing a quality of experience (QoE) associated with data provision services in an enhanced data delivery network. In one embodiment, a network architecture having service delivery over at least portions of extant infrastructure (e.g., a hybrid fiber coax infrastructure) is disclosed, which includes standards-compliant ultra-low latency and high data rate services (e.g., 5G NR services) via a common service provider. In one exemplary implementation, “over-the-top” voice data services may enable exchange of voice traffic with client devices in the aforementioned network. A distribution node may use a detection rule to identify received packets as voice traffic, and cause a dedicated bearer to attach to the default bearer, thereby enabling delivery of high-quality voice traffic by at least prioritizing the identified packets thereafter and sustaining the delivery even in a congested network environment, and improving the quality of service (QoS) and QoE for the user(s).

The present application discloses a media interaction method in a Digital Enhanced Cordless Telecommunications (DECT) network cluster. Communication with a DECT host is established through a DECT network cluster constructed in advance, to obtain first account information allocated by the DECT host, and information of a first call request to call a second account is sent to the DECT host, so as to make a call to the second account in a first communication manner. If no Real-time Transport Protocol (RTP) from the second account is received, information of a second call request to call the second account is sent to the DECT host, so as to make a call to the second account in a second communication manner. Embodiments of the present disclosure can solve the technical problem of a communication failure and reduce a waiting time for communication, thereby achieving optimal Voice over Internet Protocol (VOIP) call experience.

A media appliance comprising: video apparatus for outputting signals to a screen; a control device allowing a user to control the output of signals from the video apparatus to the screen, thereby selecting a viewing activity; a network interface for accessing a packet-based network; a memory storing a communication client application; and processing apparatus, coupled to the memory, network interface and video apparatus, and arranged to execute the communication client application; wherein the client application is configured so as when executed to allow the user to conduct bidirectional communications with other users via the packet-based network, and output notifications to the user of incoming communication events received from other users over the packet-based network; and wherein the client application is further configured to defer one or more of said notifications of incoming communication events received during said viewing activity, determine a delineation in the viewing activity, and automatically output the one or more notifications to the user following said delineation.

Methods, systems, and apparatus can be used to provide handover of on-hold sessions in converged networks. In various example implementations, an on-hold session between a mobile communication device and a fixed packet network-connected peer CPE can be handed over from a fixed packet domain to a cellular domain, and vice versa. In various example implementations, an on-hold session between a mobile communication device and a PSTN-connected peer CPE can be handed over from a fixed packet domain to a cellular domain, and vice versa.

A system, topology, and methods to enable certain session functions to be conducted over many session devices or over a large geographical area via geographically distributed servers. Other embodiments may be described and claimed.

According to one aspect, a system and method of exchanging GRUUs (Globally Routed User Agent URI (Uniform Resource Identifier)) between a first telephony-enabled device and a second telephony enabled device using a circuit-switched message is provided. Once exchanged, the telephony enabled devices can exchange SIP (session initiated protocol) communications routed by the GRUUs. Any one of the telephony-enabled devices can add a media component to the SIP communications. According to another aspect, a system and method of generating GRUUs is provided. According to another aspect, a system and method of handing off communications to a packet switched network from a circuit switched network is provided.