Certain embodiments of the invention generally relate to lawful interception of communications for IP Multimedia System-based (IMS-based) voice over IP (VoIP). For example, some embodiments relate to interception of communications including interception for encrypted, transcoded media. A method may include identifying a network node that provides call content interception. The method may also include determining a codec type used at a location of the call content interception at the network node. The method may further include sending a matched codec of the codec type used at the location of the call content interception at the network node to a delivery function in the communications network.

Systems and methods are provided for operating an intercom system using a wireless access point. A codec is selected (501) from a plurality of available codecs. In some implementations, the available codecs present a tradeoff between audio quality and intercom device capacity. The access point operates (505) using the selected codec and, in response to detecting a new intercom device connecting to the intercom system (507) through the access point, the access point transmits a signal to the new intercom device identifying the selected codec. In response to a determination that the new intercom device does not have the selected codec stored in its memory, the access point automatically uploads (515) the codec to the new intercom device and transmits communications with the intercom deviceincluding, for example, audio stream datausing the selected codec.

Technology for transcoding avoidance is disclosed. A mobile switching center (MSC) server can decode a single radio voice call continuity (SRVCC) packet switch (PS) to circuit switched (CS) request message received from a mobility management entity (MME) that includes selected CODEC information for a selected CODEC used for a user equipment (UE) in an internet protocol (IP) Multimedia Subsystem (IMS) over long term evolution (LTE) system. The MSC server can encode the selected CODEC information for transmission to a target MSC to enable the target MSC to identify the selected CODEC for the UE to allow the selected CODEC to be used in the CS domain.

Concepts and technologies provided herein can provide high-definition voice for fixed mobile convergence on mobile and wireline networks. A processor executing instructions can detect a call request associated with a called telephone number to setup a call session, where the call request is initiated from a calling device associated with an originating network. The processor can determine a call path for the call session from the originating network to a receiving network. The processor can create a fixed mobile convergence request to alert an electronic number mapping system of the call path for the call session from the originating network to the receiving network. The electronic number mapping system can provide a network presence map identifying a plurality of call receiving devices associated with the called telephone number that are available to participate in the call session via the receiving network.

A videoconferencing system includes a codec configured to generate one or more acknowledgement signals, a remote control for controlling the codec, a system controller in communication with the codec, and at least one other videoconferencing component. The system controller is configured to send one or more commands to the other videoconferencing component in response to the one or more acknowledgement signals generated by the codec. Other videoconferencing systems and presentation systems and related methods are also disclosed.

Aspects of the subject disclosure may include, for example, determining from a transport protocol message, capability information associated with user equipment associated with a call session, determining a first codec to provide a first media service to the user equipment associated with the call session according to the capability information, and responsive to determining the first codec, searching a codec table to obtain a first resource identifier associated with a first media resource function to provide the first media service according to the first codec. If the first resource identifier associated with the first codec is not available at the codec table, accessing the first resource identifier associated with the first media resource function from a domain name server and, in turn, transmitting a first request to the first media resource function to provide the first media service to the user equipment associated with the call session. Other embodiments are disclosed.

The disclosure generally relates to various methods to increase network coverage with respect to Voice-over-Internet Protocol (VoIP) sessions and/or other voice-based multimedia services. In particular, when establishing a voice session, two or more terminals may perform a codec negotiation to exchange information related to supported multimedia codecs and/or codec modes, jitter buffer management (JBM), and packet loss concealment (PLC) capabilities. Based on the exchanged information, a message may be sent to a base station to indicate the maximum packet loss rate (PLR) for each terminal. Additional techniques may ensure that the terminals use the most robust codecs or codec modes that are available when nearing the edge of coverage to help avoid unnecessary and/or excessive handovers to different radio access technologies.

A method of controlling a codec negotiation of a gateway (1) providing a codec capability offer, e.g. OLC according to the standard H.245, for establishing a data connection (1A, NA, 5T, 12T) to a communication device (10), comprises the gateway (1) providing a DSP resource (5A, 5B) for encoding (C1E) and/or decoding (C1D) of data to be transmitted using a codec which is negotiated between the gateway (1) and the communication device (10), and controlling the gateway (1) to limit (4B1, 4B2, CA2, CA14) a selection of codecs available at the gateway (1) from at least two codecs reserving different amounts of the DSP resource (5A, 5B) respectively, to be included in the codec capability offer to a codec which reserves the least amount of the DSP resource (5A, 5B) such that the communication device (10) is forced (CA4, CA15) to select the codec which reserves the least amount of the DSP resource (5A, 5B).

A system and method for diagnosing a problem on a packet network. Network performance information associated with data packet communications over a packet network may be collected. A network performance information parameter may be monitored and a determination that the network performance information parameter crosses a threshold value may be made. In response to determining that the network performance information parameter crossed the threshold value, diagnostics may be initiated to determine a cause of the network performance information parameter crossing the threshold value.

An information communication device arranged between a transmission-side device and a reception-side device according to one embodiment includes a reception unit that receives transmission data to be transmitted from the transmission-side device to the reception-side device, a control unit that receives a notification indicating deterioration in quality the transmission data, determines conversion object data for converting a data type of the transmission data or reducing an information amount of the transmission data and non-conversion object data for not converting a data type based on the notification, and generates post-conversion object data by converting a type of the conversion object data based on the notification, a transmission queue that synthesizes the non-conversion object data and the post-conversion object data and generates converted transmission data, and a transmission unit that transmits the converted transmission data to the reception-side device.