Abstract:
Congestion in the control plane of a network may be handled by an individual node. The individual node may, upon determining that congestion has been detected on a link to another node, commence implementing a compression scheme on control packets outgoing on the link. The individual node may also transmit a notification of congestion on the link to further nodes farther away from a controller. The further nodes, responsive to receiving such notifications, may commence implementing a compression scheme on control packets outgoing on a control channel that includes the link. The various nodes may also transmit notifications of congestion to a controller. The controller may, upon determining that a count of cases of congestion in the network exceeds a threshold, transmit control packets to the nodes, where the control packet contains instructions to the nodes to implement compression on outgoing control packets.
Abstract:
Various communication systems may benefit from improved bandwidth compression techniques. For example, certain communication systems may benefit from a radio fronthaul traffic compression on a frequency domain data. A method can include identifying a composite waveform corresponding to a real component and an imaginary component of a frequency domain data at a first device. The method may also include causing a transmission of a value that represents the composite waveform to a second device from the first device.
Abstract:
In some examples, the method and apparatus may comprise dynamically scaling the packet compression procedures based on available system resources. For example, as the available resource capacity (e.g., processing power, bus bandwidth and/or memory) decreases, aspects of the present disclosure may dynamically adjust the usage of the packet compression procedures on one or more data packets to maximize available resources and achieve optimal compression gains.
Abstract:
Systems and methods in accordance with various embodiments of the invention enable quality based streaming. A content player in accordance with an embodiment of the invention includes: a processor; a network interface; and memory containing a content player application. The content player application can direct the processor to: receive quality metadata describing a plurality of streams, where: the plurality of streams are encoded at different maximum bitrates; each stream is divided into content segments; and the quality varies between content segments in each stream. Furthermore, the content player application directs the processor to measure available bandwidth; request content segments from the plurality of streams based upon the available network bandwidth and the quality metadata, where the requested content segments include content segments encoded at a maximum bitrate and having quality that is the lowest maximum bitrate that achieves a target quality level.
Abstract:
Methods and computing systems for dynamic rate adaption during real-time Long Term Evolution (LTE) communication are described. A real-time LTE communication session with another mobile device is established over an LTE connection. The real-time LTE communication session is established with codec rate. A monitor component receives data indicating a performance of the real-time LTE communication session, and causes the real-time LTE communication component to perform, during the real-time LTE communication session, a renegotiation of the codec rate based at least on the performance of the real-time LTE communication session.
Abstract:
A method, processor and system for controlling packet flow in a network provides for selecting a transmission scheme based on a payload rate for payloads arriving at a buffer. The transmission scheme incudes data blocks comprising packets each having an associated payload, the data blocks have a variable packet transmission rate and a variable payload per packet ratio. The product of the variable packet transmission rate and the variable payload per packet ratio remains constant. The data blocks are sent to a network switch that deterministically routes the data blocks.
Abstract:
Aspects of the subject technology relate to systems, methods, and machine-readable media for communicating using adaptive data compression. A system is configured to compare operation of an encrypted communications channel to at least one operational threshold for the encrypted communications channel and select a level of compression for the encrypted communications channel based on the comparing. The system is further configured to compress data packets to be transmitted over the encrypted communications channel in accordance with the level of compression, encrypt the data packets, and transmit, via a network, the encrypted and compressed data packets to a destination.
Abstract:
본 발명은 미디어 스트리밍 서비스 상에서 트랜스코딩 연산의 설정 값을 네트워크 환경에 맞게 적응적으로 변화하여, 모바일 기기와 같은 환경에서 영상의 끊김을 최소로 줄여 서비스 질을 향상시키기 위한 적응적 실시간 트랜스코딩 방법 및 이를 위한 스트리밍 서버에 관한 것이다. 본 발명의 적응적 실시간 트랜스코딩 방법은, 스트리밍 서버에서 분할 및 트랜스코딩 된 미디어 데이터의 세그먼트를 네트워크를 통해 클라이언트 단으로 순차 전송하기 위한 실시간 트랜스코딩 방법이며, 클라이언트 단의 플레이-백 버퍼 충진 정도를 판별하는 단계; 플레이-백 버퍼 충진 정도에 따라 클라이언트 단으로 전송되지 않은 미디어 데이터 세그먼트의 화질을 결정하는 단계; 및 결정된 화질에 따라 전송되지 않은 미디어 데이터 세그먼트를 트랜스코딩하는 단계를 포함한다.
Abstract:
A method for initiating a codec rate change during a VoIP call by a wireless communication device is disclosed. The method can include the wireless communication device establishing a first codec rate for use in the VoIP call during a call establishment phase; using the first codec rate to encode voice data for transmission during a first portion of the VoIP call; determining a channel quality while using the first codec rate; determining that the channel quality satisfies a threshold for requesting a codec rate change; requesting a codec rate change from the first codec rate to a second codec rate in response to the channel quality satisfying the threshold; and using the second codec rate to encode voice data for transmission during a second portion of the VoIP call.