Voice over IP Technology

Technorati Profile
Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network. Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network - see attached image - to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that's borne by the VoIP user.

There are two types of PSTN to VoIP services: -Direct Inward Dialing (DID) and access numbers. DID will connect the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user.

Cisco Systems Teams with Red Bull Cheever Racing to Bring Wireless and VoIP Technology to the Indianapolis 500
INDIANAPOLIS, May 27, 2005 - Cisco Systems, Inc. and Red Bull Cheever Racing today announced that they have teamed-up to bring wireless and Voice over IP (VoIP) technology to the 89th annual Indianapolis 500, the biggest car racing event in the world. For the first time, a racing team will be utilizing this technology to give engineers, pit crews and ultimately the race car drivers, immediate access to real time information and statistics during the race.

Red Bull Cheever Racing has two race cars that will be competing in this year's Indianapolis 500 race—No. 51 Toyota-powered Dallara driven by Alex Barron and the No. 83 Toyota/Dallara driven by Patrick Carpentier. Both drivers and racing crews will be utilizing this innovative technology solution, which includes Cisco Aironet 1100 and 1300 Series Access Points, the Cisco Mobile Access Router and Cisco 7920 Wireless IP Phones with VoIP capabilities, to take a more hi-tech approach to car racing.

"Cisco is changing the way our team communicates and analyzes data," said Eddie Cheever Jr., owner, Red Bull Cheever Racing. "For example, using the Cisco solution, at the Indianapolis Motor Speedway, in a single-lap around the track, we get nearly double the amount of coverage, in the transmission of data from car to engineers, than any other team. This complete footprint of the track means we get crucial information when it happens, without delay. This is the wave of the future and we are extremely excited to partner with Cisco in this project."

"Cisco is excited to be teaming with Red Bull Cheever Racing to bring this first-of-its-kind technology solution to the Indianapolis 500, the biggest spectator sporting event in the world," said Greg Akers, SVP and CTO of Cisco's Global Government Solutions Group. "By utilizing Cisco wireless and VoIP technology, Red Bull Cheever Racing can be more mobile, have real-time access to data, and finally, optimize its race car performance."

IP telephony in Japan/Technical details

In Japan, IP telephony is regarded as a service applied VoIP technology to whole or a part of the telephone line. As from 2003, IP telephony service assigned telephone numbers has been provided. There are not voice only services, but also videophone service. According to the Telecommunication Business Law, the service category for IP telephony also implies the service provided via Internet, which is not assigned any telephone number. IP telephony is basically regulated by Ministry of Internal Affairs and Communications (MIC), as a telecommunication service. The operators have to disclose necessary information on its quality, etc, prior to making contract with customers, and have obligation to respond to their complaints cordially.

Many Internet service providers (ISP) are providing IP telephony services. The provider, which provides IP telephony service, is so-called "ITSP (Internet Telephony Service Provider)". Recently, the competition among ITSPs has been activated, by option or set sales, connected with ADSL or FTTH services.

The tariff system normally applied for Japanese IP telephony tends to be described as below;

* The call between IP telephony subscribers, limited to the same group, is mostly free of charge.
* The call from IP telephony subscribers to fixed line or PHS is mostly fixed rate, uniformly, all over the country.

Between ITSP, the interconnection is mostly maintained at VoIP level.

* As for the IP telephony assigned normal telephone number (0AB-J), the condition for its interconnection is considered same as normal telephony.
* As for the IP telephony assigned specific telephone number (050), the condition for its interconnection tends to be described as below;
o Interconnection is sometimes charged. (Sometimes, it's free of charge.) In case of free of charge, mostly, the traffics are exchanged via P2P connection with the same VoIP standard. Otherwise, certain conversion is needed at the point of VoIP gateway, which needs running costs.
>>>read more....

Telephone number for IP telephony in Japan

Wireless Watch Japan

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that the service falls into certain required categories of quality. Highly qualified IP telephony is assigned a telephone number. Normally the number starts with 050. But, when its quality is so high that customer almost could not tell the difference between it and a normal telephone and when the provider relates its number with a location and provides the connection with emergency call capabilities, the provider is allowed to assign a normal telephone number, which is a so-called "0AB-J" number.

The two major competing standards for VoIP are the IETF standard SIP and the ITU standard H.323. Initially H.323 was the most popular protocol, though in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage.[citation needed]

However, in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones[citation needed], and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP.

Where VoIP travels through multiple providers' softswitches the concepts of Full Media Proxy and Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2) H.245; 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms.

One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically, to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimization techniques used, such as silence suppression and header compression. This can typically save 35% on bandwidth usage.

VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by multiplexing multiple conversations that are heading to the same destination and wrapping them up inside the same packets. Because the packet header overhead is shared between many simultaneous streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only about 1 kbit/s.

* List of commercial voice over IP network providers
* SIP
* IP Multimedia Subsystem
* Mobile VoIP
* Comparison of VoIP software
* PATS
* Computer conferencing
* ROIP
* Differentiated services
* Integrated services
* Predictive dialers
* Secure telephone
* SIP Telephony
* VoIP recording
* Capillary routing
* VoiceXML

New ways of communication: The VoIP technology

A simple solution for people’s desire for communica Technology is in a continuous change, it evolves every day. With the help of the Internet, various new and innovating technologies are now accessible to everyone and provide new solutions to the old problems. One of them is the desire to communicate more, cheaper and over longer distances than usual. The solution for this need is the VoIP technology which replaces the traditional telephone landlines. This technology works best on a high-speed connection to the Internet. The dial-up connection is too slow for a real-time voice or data transfer, so it is not recommended. The program used in this type of communication resembles a telephone keypad and works in the same way. It is called ‘Softphone’, which means ‘Software Phone’.

The main advantage of this brilliant technology is its low costs. Every VoIP provider has its own rates, but generally, all rates are lower than most of the traditional telecommunication systems. The VoIP technology transforms analog signals into digital data packets, which are then compressed and sent over the Internet at high speed. When the data reaches the destination, it goes through a reverse process of decompression and conversion to an analog signal. It Even if it seems complicated, it is a real-time process, and that means there will be no delay between the two speakers. There are two ways of calling through VoIP: PC to PC or PC to regular phone. The first alternative is completely free, while the second is charged. The rates of any VoIP provider are displayed on its website, generally at a section called ‘call shop rates’.

Every call shop customer can take advantage of a free online billing system, which detects all the booths that have made calls in the last three hours. The call in progress and the calls that have been completed can be accessed just by clicking on the booth link and the "end-of-day" cash accounting is assisted by daily reports. For this system there are available two levels of access, which means that only the administrator can configure the system, and the day to day billing system procedures can only be attended by the staff in the call shop. The billing procedure is completely self-configuring. You just need to sign up, set the currency that your callers will be charged and in the end set the rates that you wish to charge your customers. For the special call shop rates, you must visit the VoIP provider’s website.

You can use VoIP applications with a simple microphone and computer speakers, but many VoIP manufacturers are designing phones which are specially created to work with this technology, called SIP phones. If you don’t have a SIP-phone you can use the softphone which can be found on the VoIP provider’s website and which perfectly emulates a standalone VoIP SIP-phone and runs on any windows operating system. It is free, easy to install and to use.

Through this new technology, people from all over the world can talk and change data almost for free, no matter the distance between them. Another advantage is the great mobility offered by this service. You can travel all over the world and still make and receive phone calls, because your number remains the same whether you are in Europe or in USA.