Call Center Services

We heard so many things about VoIP since this technology brought into call center industry, VoIP is new generation’s communication, VoIP is superior and cheaper then conventional PSTN service and so many other things. You can explore all the benifits and feature which are described by the variuos bloggers and writers (Call Center Technology Blog) in their writting. In today blog, we will discuss something about how voip works, what are the equipment an individual or corporate required, what are basic infrascture requirment for using voip and other not so important thing.

What is VoIP

Short for Voice over Internet Protocol, a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls by sending voice data in packets using IP rather than by traditional circuit transmissions of the PSTN. more details…

VoIP services convert your voice into a digital signal that travels over the Internet. VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless “hot spots” in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable you to use VoIP service wirelessly.

What we need for VoIP

Broadband connection high speed connection, which can be cable, modem, DSL, ISDN or from your local network. A computer, adaptor, or specialized phone is required. Some VoIP services only work over your computer or a special VoIP phone, while other services allow you to use a traditional phone connected to a VoIP adapter. Software is also required and headset with microphone, normal headphone with mic will also work.

Special VoIP phones plug directly into your broadband connection and operate largely like a traditional telephone. If you use a telephone with a VoIP adapter, you’ll be able to dial just as you always have, and the service provider may also provide a dial tone.

Technical details

The two major competing standards for VoIP are the ITU standard H.323 and the IETF standard SIP. 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.

H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK, NetMeeting and X-Meeting, and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks.

The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. It can be used to create two-party, multiparty, or multicast sessions that include Internet telephone calls, multimedia distribution, and multimedia conferences.

Implementation

VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary satellite). Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network.

Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there are long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.

The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed for fax transmission. An effort is underway to remedy this by defining an alternate IP-based solution for delivering Fax-over-IP, namely the T.38 protocol.

VoIP over mobile phones: “dual mode” telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular. Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.

References:

http://www.fcc.gov/voip/
http://en.wikipedia.org/wiki/VoIP
http://www.protocols.com/pbook/VoIPFamily.htm

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