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江南大學(xué)視頻會議系統(tǒng)的設(shè)計(jì)與開發(fā)畢業(yè)設(shè)計(jì)開題報(bào)告書(編輯修改稿)

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【文章內(nèi)容簡介】 prediction algorithms. Capacity Planning Before you place video traffic on a work, ensure that adequate bandwidth exists for all necessary applications. First, calculate the minimum bandwidth requirements for each major application, for example, voice, video, and data. The sum represents the minimum bandwidth requirement for any specific link. This amount should consume no more than 75 percent of the total bandwidth available on that link. This 75 percent rule assumes that some bandwidth is necessary for overhead traffic. Examples of overhead traffic include routing protocol updates and Layer 2 keepalives, as well as additional applications, such as and HTTP traffic. Have voice and video traffic occupy no more than 33 percent of link capacity. This Example Scenario explains capacity planning on a converged work. 11 Example Scenario A site has a link capacity of Mbps and contains two video terminals that support a maximum data rate of 256 kbps each. Although the rate of the two video calls equals 512 kbps, add 20 percent to the data rate of the call to account for overhead. Twenty percent is a conservative percentage that ensures proper capacity planning in most environments. You can start with an extra 20 percent for overhead and then adjust this value, higher or lower, with the results of your monitor as a basis. Provision the priority queue for sufficient bandwidth to allow both video terminals to have an active call across the WAN simultaneously without the possibility of an overrun of the priority queue. In this example scenario, if you add a third video terminal, you need to implement some form of CAC. Determine PerCall Bandwidth Consumption With capacity planning, one of the most critical concepts to understand is how much bandwidth you use for each call. This section lists the bandwidth that each coderdecoder (codec) uses. Refer to Voice over IP Per Call Bandwidth Consumption for more information. Audio Audio signals contain digitized, pressed sound (usually speech). supports proven ITUstandard audio codec algorithms. The algorithms with support include: — kilohertz (kHz) at 48, 56, and 64 kbps (normal telephony) — 7 kHz at 48, 56, and 64 kbps — kHz at 16 kbps — and kbps modes Selection of the right codec reflects tradeoffs between speech quality, bit rate, pute power, and signal delay. Video According to the standard, video capabilities in terminals are optional. However, when you implement the terminals, the terminals must support the codec, with optional support for the standard. — Video codec for audiovisual services at multiples of 64 kbps. devices fully encode initial frames. The devices then code only the differences between the initial and subsequent frames for minimal packet transmissions. Optional motion pensation improves image quality. — Video codec for video plain old telephone service (POTS). The standard is a backwardpatible update to the standard. significantly enhances picture quality with a halfpixel motion estimation technique, which is a requirement. Enhancements also e from predicted frames and a Huffman code table, with optimization for low bitrate transmissions. The standard defines five standard picture formats, as Table 1 shows in the document White Paper: Deploying Applications in Cisco Networks. 12 Classification To provide the appropriate QoS guarantees to video traffic, work devices need to be able to identify such traffic. The differentiated services (DiffServ) model of QoS uses DiffServ code point (DSCP) values to separate traffic into classes. DiffServ defines these two sets of DSCP values: Expedited Forwarding (EF)— Provides a single DSCP value (101110) that gives marked packets the highest level of service from the work. Cisco implements EF service via low latency queueing (LLQ). Generally, EF keeps the highpriority queue very
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