摘要:
A media access control (MAC) apparatus and corresponding methods for guaranteeing quality-of-service in a wireless local area network (LAN) are presented. The MAC method includes extracting, performing, determining, a first transmitting step, and a second transmitting step. The extracting includes extracting a user priority from a frame received from an upper layer and separately storing a voice frame and a non-voice frame according to an access category (AC). The performing includes independently performing backoff operations for the voice frame and the non-voice frame. The determining includes determining whether the backoff operations for the voice frame and the non-voice frame have simultaneously ended. The first transmitting includes transmitting the voice frame having a higher priority first and performing the backoff operation for the non-voice frame if the backoff operations have simultaneously ended. The second transmitting includes transmitting a frame whose backoff operation ends if the backoff operations have not simultaneously ended.
摘要:
A method and apparatus for controlling a variable bit-rate voice codec are provided. The method of controlling the variable bit-rate voice codec may include: extracting calling capability of terminals that request a new call to be made; extracting network parameters from existing calls in the network through an exchanged packet; measuring voice quality of the existing calls based on the extracted network parameters; and determining whether to permit a new call to be made based on the measured voice quality and the calling capability. Accordingly, it is possible to secure QoS of a voice service between terminals by adjusting a transmission rate of the variable bit-rate codec based on transmission capability of a network.
摘要:
A media access control (MAC) apparatus and corresponding methods for guaranteeing quality-of-service in a wireless local area network (LAN) are presented. The MAC method includes the steps of extracting, performing, determining, a first transmitting step, and a second transmitting step. The extracting step includes extracting a user priority from a frame received from an upper layer and separately storing a voice frame and a non-voice frame according to an access category (AC). The performing step includes independently performing backoff operations for the voice frame and the non-voice frame. The determining step includes determining whether the backoff operations for the voice frame and the non-voice frame have simultaneously ended. The first transmitting step includes transmitting the voice frame having a higher priority first and performing the backoff operation for the non-voice frame if the backoff operations have simultaneously ended. The second transmitting step includes transmitting a frame whose backoff operation ends if the backoff operations have not simultaneously ended.
摘要:
A media access control (MAC) apparatus and corresponding methods for guaranteeing quality-of-service in a wireless local area network (LAN) are presented. The MAC method includes the steps of extracting, performing, determining, a first transmitting step, and a second transmitting step. The extracting step includes extracting a user priority from a frame received from an upper layer and separately storing a voice frame and a non-voice frame according to an access category (AC). The performing step includes independently performing backoff operations for the voice frame and the non-voice frame. The determining step includes determining whether the backoff operations for the voice frame and the non-voice frame have simultaneously ended. The first transmitting step includes transmitting the voice frame having a higher priority first and performing the backoff operation for the non-voice frame if the backoff operations have simultaneously ended. The second transmitting step includes transmitting a frame whose backoff operation ends if the backoff operations have not simultaneously ended.
摘要:
Provided is a method and apparatus for minimizing the number of transcodings between network devices in a multi-network multi-codec environment. The method includes: creating a received codec list by receiving a transmit codec comprised in a call setting message from a transmission device and the number of transcodings of the transmit codec, which has been performed from a codec of an initial transmission device; creating a codec Quality of Service (QoS) list containing total codecs of the multi-network and quality information of each of the total codecs; creating a transcodec list containing internally providable transcodecs and quality information of the transcodecs based on the codec QoS list; and creating an updated codec list by adding codecs of the transcodec list matching codecs of the received codec list to the received codec list and adjusting codec priority according to the number of transcodings. Accordingly, since the number of transcodings can be minimized, a quality decrease of original media can be minimized.
摘要:
The transmission delay of a voice frame can be reduced by performing internal collision resolution and frame aggregation according to the presence or absence of a voice frame awaiting transmission in a MAC layer, thereby reducing an end-to-end voice transmission delay time for a VoIP service.
摘要:
The present invention provides a method of controlling Wireless LAN (WLAN) medium access using Pseudo-Time Division Multiplexing (PTDM) to improve the Quality of Service (QoS) of the WLAN. The method includes the first step of a Mobile Terminal (MT) acquiring QoS information, which relates to a voice frame to be transmitted, from an upper layer; the second step of the MT exchanging a frame, including the QoS information, with the AP and being allocated QoS slot (QSLOT) information by the AP; the third step of dividing an RF link section between the MT and the AP by the creation period of the voice frame; and the fourth step of transmitting the voice frame in the creation period of the voice frame using the allocated QSLOT information.
摘要:
A method and apparatus for controlling a variable bit-rate voice codec are provided. The method of controlling the variable bit-rate voice codec may include: extracting calling capability of terminals that request a new call to be made; extracting network parameters from existing calls in the network through an exchanged packet; measuring voice quality of the existing calls based on the extracted network parameters; and determining whether to permit a new call to be made based on the measured voice quality and the calling capability. Accordingly, it is possible to secure QoS of a voice service between terminals by adjusting a transmission rate of the variable bit-rate codec based on transmission capability of a network.
摘要:
Provided is a transmission apparatus for matching sound quality measurement sections of a variable bandwidth multi-codec. The apparatus includes a measurement section setting unit setting a measurement section, which is to be measured for sound quality, in units of time; a first conversion unit converting the measurement section into a measurement section in units of samples; and an information synthesis unit synthesizing information regarding the measurement section in units of samples with a digital original sound and outputting the synthesis result. In addition, provided is a method of matching a measurement section of a reference sound, based on which the end-to-end sound quality measurement of the variable bandwidth multi-codec is performed, and a measurement section of a sound produced by the variable bandwidth multi-codec in a real-time Internet multimedia service. Therefore, distortion of measurement results due to un-matching measurement sections can be reduced.
摘要:
Provided are a method and apparatus for measuring sound quality in a variable band multi-codec. The sound quality measurement apparatus includes: a recording file receiving/generating unit receiving a first recording file in which a natural sound is recorded, and a second recording file obtained by converting the natural sound into digital data using the variable band multi-codec, receiving information obtained by encoding the natural sound using the variable band multi-codec, in the format of a Real Time Protocol (RTP) packet, unpacking the RTP packet, decoding the RTP packet using the variable band multi-codec, and generating a third recording file; a Mean Opinion Score (MOS) value calculating unit repeatedly selecting a file from among the first recording file, the second recording file, and the third recording file, or selecting two files from among the first recording file, the second recording file, and the third recording file, and calculating a MOS value by obtaining a difference between the selected results; and a MOS value comparison unit comparing a plurality of MOS values generated by the MOS value calculating unit, with each other, and detecting a cause of sound quality deterioration.