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VoIP is the technology for voice conversations over an internal or external data network using IP packets (digital form) without loss in functionality, reliability or quality and in agreement with the International Telecommunications Union specifications. The term is also used to set the hardware and software to point to carry such calls over the network.
Voice over Internet Protocol (VoIP) is a common term for a communication protocols, and communication technologies for sending of voice communications and multimedia sessions over Internet . It employ session control protocols to control the set-up and breakdown of calls, and encode audio codecs that enable speech transmission over an IP network, such as digital audio via an audio stream.
First, voice is transformed by an ATA (Analog Telephone Adapter), from an analog signal to a digital signal. It is then directed over the Internet in data packets to a place that will be close to the target and thereafter it will be transformed back to an analog signal for the remaining distance over a traditional circuit switch. Calls can be received by telephones worldwide, as well as other VoIP users. VoIP to VoIP calls can travel entirely over the Internet. As your voice is changed to digital (so that it can travel over the Internet),there are other features such as voice messages to email, call forwarding, caller ID, etc., can be included in your basic calling plan all for one low price. Many of these features are ideal for small business, based on their phone service to more of a information center rather than just a phone.
DELAY IN TRANSFERING
The sum of the store-and-forward delay, which is a packet in each router experiences or delay the transfer measurement of this packet over the network. Packet transmission delay is influenced by the level of network utilization and the number of routers on the path of transmission.
There are four sources of packet transfer delay:
1 Check bit errors,
2 Define output link
1 Time waiting at output link for transmission,
2 Depends on congestion level of router
1 R=Link bandwidth (bit/s),
2 L=Packet length (bits),
3 Time to send bits into link = L/R
1 d = Length of physical link,
2 s = Propagation speed in medium,
3 Propagation delay = d/s
Bandwidth is the amount of data that is transmitted across a network cable line. It is measured in bytes, but often the number of bytes is so large that it is in kilobytes (KB), which is to be measured in thousands of bytes, megabytes (MB), the millions of bytes, or gigabytes (GB), the billions of bytes. The final measure of bandwidth is bytes per second.The term is used to measure the amount of transmitted data (for billing, for example), or the amount available to be used by a computer system.
Determine the size of all files that are used in the transmission. For example, if you measure the bandwidth used by a particular Web page, add all the file sizes for all parts of the web page the html, the images and any other files that are included with the web page. To measure this for a website, take an average of the web page file sizes. This sum is the base file size. For example, an HTML file may be 2 KB and has 15 KB of images. This base file size is 17 KB.
Estimate the number of page views for the file. To regulate the bandwidth for a single page, use one for the number of page views. Use ancient data or assessed data for views in the future for this number. For example, if 3000 visitors are expected to view the web page in a month, use 3000 for this value.
Multiply the base file size with the number of page views. Continuing the example, this would be 17 multiplied by 3000, resulting in 51,000 KB, or 51 MB. The estimated bandwidth used in this example is, 51 MB
JITTER IN NETWORK
In very simple terms, when a device is expected to issue a message and when the message is actually transmitted, network jitter is simply the deviation between the time. For example, if a Device Net or Ethernet/IP device is expected to issue cyclic messages every 10msecs (very typical cycle time) and sends them between 9 and 11 msecs then we have 1 msec of network jitter.
So which is good? Anything less than 50% jitter is acceptable. In the previous example, a device could send the next cyclic message between 5 and 15msecs of sending the last one. Unless you have a very high speed, timing critical application network jitter like this is acceptable. Once the device gets over 50% jitter , the Client or Master device could begin to detect missing messages. If the jitter gets bad enough, the Master (Client) might even shut down the device and try to reconnect with causing a loss of data or worse.
What is the impact of the switch on network jitter in an Ethernet/IP network. It turns out that there is virtually nothing. As the switches receive a packet they first analyse the few bytes of the packet to identify the outgoing port and start to send message immediately out of that port, even before the end of the message is received. The switch adds almost nothing to network jitter at this speed.
The failure of one or more transmitted packets to arrive at their destination ispacket loss. This event can cause significant impact in all types of digital communication.
The effects of packet loss:
Packet loss produces errors in both text and data.
Packet loss can create jitter in video conference environments.
Packet loss can cause jitter in pure audio communications, such as VoIP and frequent gaps in received speech.
In the worst cases, packet loss can cause severe damage of received data, broken-up images, meaningless speech or even the complete absence of a received signal.
The reasons of packet loss include inadequate signal strength at the destination, natural or human- ade interference,excessivesystem noise, hardware failure,software corruption or overloaded network nodes. Often these factors are involved in more than one. In a case where the cause cannot be resolved, packet loss concealment may be used to minimize the effects of lost packets.
VOIP telephone systems are vulnerable to attacks as are any internet-enabled devices. This means that hackers who know about these vulnerabilities (like insecure passwords) can institute denial attacks, harvest customer data, record conversations and break into voice mailboxes. Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. For example, Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. The methods to traverse NATs involve using protocols such as STUN or ICE.Many consumer VoIP solutions do not support encryption, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN.
There are open source solutions, such as Wireshark, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications; however, such security through obscurity has not proven effective in other fields. Some vendors also use compression, which may make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol (SRTP) and the new ZRTP protocol are available on Analog Telephone Adapters (ATAs) as well as various softphones. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. In 2005, Skype invited a researcher.The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream.