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前摄VOIP安全监视

一致VoIP 并且数据网花费企业很多金钱,但他们仍然留给一个问题: “是它安全?”

为Alphonse Edouard,它副总统为 沙丘资本管理投资公司, VoIP成为了事务基石。 如此保证它的安全是必要的。

“为很多什么我们做,声音是非常重要的”, Edouard说。

沙丘资本管理 通过部署开始 VoIP. “然后`工作’任何地方概念开始了活动”,他说。

沙丘需要方式保证电话质量和监测网络保证它是安全的。

“我们全部知道VoIP是非常易受黑客”, Edouard说。 从前,他使用QRadar从Q1实验室监测流程数据和网络信息流通量。 最终,他开始监测VoIP服务质量(QoS)。 但是,当沙丘资本在VoIP越来越适合受抚养者,公司需要保证定量了足够的带宽并且必须发现方式分开地监测VoIP交通与数据量,虽然二份额网络。

一个新的QRadar模块为监视VoIP网络具体地设计了适合了票据, Edouard说。 VoIP…

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VoIP带宽基本部分II

一旦您知道方法和因素包括,计算带宽为VoIP电话不是困难的。 图如下, “计算的单程声音带宽”,展示顶上的演算为20个和40个字节被传送在帧中继WAN连接的被压缩的声音(G.729)。 G.729被压缩的声音二十个字节与词的20女士是相等的。 G.729被压缩的声音四十个字节与词的40女士是相等的。

这种计算方法的结果在下张桌里包含, “小包声音传递要求”。 桌展示这些点:

*带宽要求减少以压缩, G.711对 G.729.
*带宽要求减少,当使用时更长的小包,从而减少在头顶上。
*即使声音压缩是一个8个到1个比率,带宽减少约为3或4到1。 天花板否定某些声音压缩带宽储款。
*压缩RTP, UDP和IP倒栽跳水(cRTP)是最可贵的,当小包也运载压缩的声音时。

Click to continue reading "VoIP bandwidth fundamentals Part II"

Written by Lovely on August 4th, 2007 with no comments.
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VoIP bandwidth fundamentals Part I

Bandwidth requirements for Voice over IP can be a tricky beast to tame until you look at the method and factors involved. This guide investigates what bandwidth means for VoIP, how to calculate bandwidth consumption for a VoIP network and how bandwidth can be saved by using voice compression.
After this some questions may arise in your mind like:

What about bandwidth for VoIP?
Voice over IP (VoIP) is the descriptor for the technology used to carry digitized voice over an IP data network. VoIP requires two classes of protocols: a signaling protocol such as SIP, H.323 or MGCP that is used to set up, disconnect and control the calls and telephony features; and a protocol to carry speech packets. The Real-Time Transport Protocol (RTP) carries speech transmission. RTP is an IETF standard introduced in 1995 when H.323 was standardized. RTP will work with any signaling protocol. It is the commonly used protocol among IP PBX vendors.

An IP phone or softphone generates a voice packet every 10, 20, 30 or 40ms, depending on the vendor’s implementation. The 10 to 40ms of digitized speech can be uncompressed, compressed and even encrypted. This does not matter to the RTP…

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Written by Lovely on August 4th, 2007 with no comments.
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VOIP Performance Testing Fundamentals Part II

3. Analyze call quality with technical metrics
Once VoIP traffic is running in an accurately emulated virtual environment, the team can apply metrics such as mean opinion score (MOS) to pinpoint any specific places or times where voice quality is unacceptable. Typically, these trouble spots will be associated with observable network impairments — such as delay, jitter and packet loss — which can then be addressed with appropriate remedies.

4. Validate call quality by listening to live calls
Technical metrics alone can be misleading, since the perception of call quality by actual end users is the ultimate test of VoIP success. So the virtual environment should be used to enable the team to validate firsthand the audio quality on calls between any two points on the network under all projected network conditions. Again, a call generator can be used so that testers can act as the “nth” caller at any location.

5. Repeat as necessary to validate quality remedies
A major advantage of a virtual environment is that various fixes can be tried and tested without disrupting the production network. Testing in the virtual environment should therefore be an iterative process, so that all bugs can be fully…

Click to continue reading "VOIP Performance Testing Fundamentals Part II"

Written by Lovely on August 2nd, 2007 with no comments.
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VOIP Performance Testing Fundamentals Part I

VoIP network performance testing can mean the difference between a VoIP system working at a high level QoS and a weak system that runs so poorly customers could take their business elsewhere. This guide discusses why it is important to run regular performance testing and some of the ways it can be done.
Voice over IP (VoIP) technology offers a wide range of benefits — including reduction of telecom costs, management of one network instead of two, simplified provisioning of services to remote locations, and the ability to deploy a new generation of converged applications. But no business can afford to have its voice services compromised. Revenue, relationships and reputation all depend on people being able to speak to each other on the phone with five 9’s reliability.

Thus, every company pursuing the benefits of VoIP must take steps to ensure that their converged network delivers acceptable call quality and non-stop availability.

A virtual network test bed is particularly useful for taking risk out of both initial VoIP deployment and long-term VoIP ownership. Essentially, such a test bed enables application developers, QA specialists, network managers and other IT staff to observe and analyze the behavior of network…

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Written by Lovely on August 1st, 2007 with no comments.
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Improving VOIP Call Quality

If the problem is echo and it is happening within your building – return loss in your cable plant is the likely culprit – you can buy an expensive echo canceller or have your cables reterminated and tested (generally FAR cheaper). If the problem is jitter and delay, the solution could be as simple as putting your voice equipment on a VLAN and increasing the priority at which those packets are transmitted. You will also want to look at a few other things like making sure that your subnet is not too large, that all of your devices handle QoS and that they are layer 3 devices.

How can a business ensure VoIP quality of service (QoS)?

The first step is to be sure that your routers and switches support QoS, because many of the older ones do not. Second you will want to be sure that your network is healthy. You should turn SNMP (V3) on and monitor your network for a period of time (ideally 30 days as this covers end of month processes and other high traffic times). Look for bit errors, retransmissions, discards, etc. Correct any problems there. Make sure that your cabling is…

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Written by Lovely on July 31st, 2007 with no comments.
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