한데 모아진 VoIP 그리고 데이터 통신망은 기업에게 많은 돈을 요하고 있다, 그러나 아직도 1개의 질문 남겨진다: "이다 안전한 그것은?"

Alphonse Edouard를 위해, 그것 부통령 를 위한 모래 언덕 자본 관리, 투자 회사, VoIP는 사업의 초석이 되었다. 이렇게 그것의 안전을 지키는 것은 긴급하다.

"많은, 음성을 위해 우리가 하는 무엇을 아주 중요하다," Edouard는 말했다.

모래 언덕 자본 관리 배치해서 시작하는 VoIP. "그 때 `일 어느 곳에서든지' 개념 활동하기 시작했다,"는 그는 말했다.

모래 언덕은 외침 질을 지키고 안전하다는 것을 보장하기 위하여 네트워크를 감시하는 방법을 필요로 했다.

"우리는 모두 VoIP가 해커에 아주 감염되기 쉽다는 것을 알고 있다," Edouard는 말했다. 과거에는, 그는 흐름 자료와 네트워크 소통량을 감시하기 위하여 Q1 실험실에서 QRadar를 이용했다. 최후에, 그는 VoIP 서비스 질 (QoS)를 감시하는 것을 시작했다. 그러나 모래 언덕 자본이 VoIP에 종에 점점 어울리는 때, 회사는 2 몫 네트워크 충분한 대역폭이 VoIP 소통량을 자료 소통량으로부터 분리되 감시하는 방법을 찾아내야 할당되고 또한 했다는 것을 보증할 필요가 있었다.

QRadar 새로운 단위는 감시 VoIP 네트워크를 위해 특히 적합했다 계산서를 디자인했다, Edouard는 말했다. VoIP…

VoIP 외침을 위한 대역폭을 산출하는 것은 일단 당신이 포함하기 위하여 방법 및 요인을 알고 있으면 곤란하지 않다. 아래에 도표는, "계산 일방통행 음성 대역폭," 20 그리고 40의 프레임 릴레이 WAN 연결에 전달되는 바이트에 의하여를 위한 머리 위 계산을 압축된 음성 (G.729) 설명한다. G.729에 의하여 압축된 음성의 20개의 바이트는 낱말의 20 Ms와 동등하다. G.729에 의하여 압축된 음성의 40개의 바이트는 낱말의 40 Ms와 동등하다.

이 계산법의 결과는 다음 테이블에서 포함된다, "소포 음성 전달 필요조건." 테이블은 이 점을 설명한다:

* 대역폭 필요조건은 압축, G.711로 대 감소시킨다 G.729.
* 대역폭 필요조건은 머리 위에 감소시키는 더 긴 소포가 이용될 때, 그로 인하여 감소시킨다.
* 비록 음성 압축이 8개 1개의 비율이더라도, 대역폭 감소는 대략 3 4에서 1이다. 간접비는 어떤을의 음성 압축 대역폭 저축 부정한다.
* 소포가 또한 압축 음성을 나를 때 RTP를 압축해서, 사용자 데이터그램 프로토콜과 IP 우두머리 (cRTP)는 가장 귀중하다.

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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…

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…

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…

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…