A method, device and medium for determining a coding format are provided. The method includes: receiving one or more data packets forwarded by a call center during a VoLTE communication, in which the one or more data packets carry a first coding format; detecting whether the first coding format is same with a negotiated second coding format; and modifying the coding format used during the VoLTE communication from the second coding format to the first coding format, if the first coding format is not same with the negotiated second coding format.

In various embodiments, a corpus of codecs may be correlated with different, partially overlapping ranges of transmission characteristics. As channel conditions degrade or improve, the system may select a new codec with which to continue the connection based upon the corresponding overlapping range. Codecs may not be switched immediately when the transmission characteristics enter overlapping ranges, to avoid degrading the user's experience. If the characteristics remain in the overlap, or manifest a likely progression toward another region, then the transition may be effected.

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.

Aspects of the subject disclosure may include, for example, including a processing system for performing operations for determining service requirements of a call session at first user equipment associated with a communication network, determining a first codec to facilitate the call session at the first user equipment according to the service requirements of the call session, searching a session border controller table according to the first codec to obtain a first resource identifier associated with a first session border controller type to facilitate the call session at the user equipment, receiving a first address of a first session border controller associated with the communication network from a domain name server associated with the communication network responsive to a first query including the first resource identifier, and sending a first transport protocol message to the first session border controller according to the first address. Other embodiments are disclosed.

A network device determines radio frequency (RF) conditions at a first endpoint and a second endpoint of a call or session. The network device determines optimum first codec parameters for the determined RF conditions at the first endpoint of the call or session, and determines optimum second codec parameters for the determined RF conditions at the second endpoint of the call or session. The network device sends the first codec parameters to the first endpoint for altering operation of a first codec at a first device at the first endpoint. The network device sends the second codec parameters to the second endpoint for altering operation of a second codec at a second device at the second endpoint.

A communication management system includes: a storage unit configured to store destination information of a first communication terminal that establishes a first session with a relay device that relays communication data, destination information of a conversion system that performs mutual conversion between communication schemes of communication data transmitted from the first communication terminal and a second communication terminal and establishes a second session with the relay device, and destination information of the second communication terminal that establishes a third session with the conversion system; a receiving unit configured to receive start request information to start communication between the communication terminals from the first communication terminal; an extracting unit configured to extract destination information of each communication terminal and the conversion system stored in the storage unit, based on the received start request information; and a transmitting unit configured to transmit the extracted destination information to the relay device.

The disclosure generally relates to various methods to increase network coverage for a Voice-over-Internet Protocol (VoIP) session between a first user equipment (UE) and a second UE. In an aspect, a first and second UEs negotiate a codec configuration to use in the VoIP session, transmits, to the second UE, a maximum end-to-end packet loss rate (PLR) that the first UE can tolerate for received media given the negotiated codec configuration, receives, from the second UE, a maximum end-to-end PLR that the second UE can tolerate for received media given the negotiated codec configuration, and determines a distribution of the maximum end-to-end PLRs among respective uplinks and downlinks at the first UE and the second UE.

A method and system for establishing optimized data streams in a network can be configured for crafting resource-draining Session Description Protocol (SDP) bodies on a call queue.

Embodiments provide systems, methods, apparatus, and computer program products for selecting codec parameters to satisfy one or more operating constraints. An example method performed by a network component that facilitates a communication session set-up process among endpoints in a communication network, the method includes: during the communication session set-up process, determining a utilization factor of the network component; selecting a value for sampling frequency associated with a first codec in response to determining the utilization factor; negotiating use of the first codec and the value for sampling frequency with a first endpoint and negotiating use of a second codec with a second endpoint; and transcoding a media stream of a communication session between the first endpoint and the second endpoint according to the first codec and the value for sampling frequency and the second codec.

Connecting agent to customers is the cornerstone of contact center operations. When a customer is connected to an agent for real-time communications (e.g., voice), they can be particular sensitive to any delay, such as the delay to establish a media channel between the agent's endpoint and the customer's device. By establishing a media channel between the agent's device and a server having initially received the communication, the communication may be attached to the existing media channel without requiring the time and resources otherwise necessary to establish the media channel. Additionally, if a problem or failure occurs, a server utilized for the connection may seek to establish the media channel with a second network, such as a telephone network, or a second agent endpoint or endpoint component.