Converged VoIP and data networks are costing enterprises a lot of money, but still they’re left with one question: “Is it secure?”

For Alphonse Edouard, IT vice president for Dune Capital Management, an investment firm, VoIP has become a cornerstone of business. So ensuring its security is imperative.

“For a great deal of what we do, voice is very important,” Edouard said.

Dune Capital Management started by deploying VoIP. “Then the ‘work anywhere’ concept came into play,” he said.

Dune needed a way to ensure call quality and to monitor the network to guarantee that it’s secure.

“We all know VoIP is very susceptible to hackers,” Edouard said. In the past, he has used QRadar from Q1 Labs to monitor flow data and network traffic. Eventually, he started to monitor VoIP quality of service (QoS). But as Dune Capital became more and more dependent on VoIP, the company needed to ensure that enough bandwidth was allotted and also had to find a way to monitor VoIP traffic separately from data traffic, though the two share a network.

A new QRadar module specifically designed for monitoring VoIP networks fit the bill, Edouard said. The VoIP…

Calculating the bandwidth for a VoIP call is not difficult once you know the method and the factors to include. The chart below, “Calculating one-way voice bandwidth,” demonstrates the overhead calculation for 20 and 40 byte compressed voice (G.729) being transmitted over a Frame Relay WAN connection. Twenty bytes of G.729 compressed voice is equal to 20 ms of a word. Forty bytes of G.729 compressed voice is equal to 40 ms of a word.

The results of this method of calculation are contained in the next table, “Packet voice transmission requirements.” The table demonstrates these points:

* Bandwidth requirements reduce with compression, G.711 vs. G.729.
* Bandwidth requirements reduce when longer packets are used, thereby reducing overhead.
* Even though the voice compression is an 8 to 1 ratio, the bandwidth reduction is about 3 or 4 to 1. The overhead negates some of the voice compression bandwidth savings.
* Compressing the RTP, UDP and IP headers (cRTP) is most valuable when the packet also carries compressed voice.

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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…