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Archive for: November, 2005

Analogue and Digital Signalling

Every sound you hear, which of course includes human speech, is in the form of an analogue signal. Only until just a few decades ago our telephony networks were built on an analogue infrastructure. Whilst an analogue signal is perfect for human communication it is not resilient or competent at improving from line noise. Line noise is usually due to static being present over the voice network. At the beginning of telephony networks amplifiers were used to boost the analogue signals in order to make the signal more audible. The problem with this technique was that the entire signal was amplified, both voice and line noise being increased. The line noise often made the connection inoperative.

Line noise is much less of an issue with digital telephony networks due to the fact that repeaters on the lines do not just amplify the signal but also clean the digital signal back to it’s initial state. The reason this is achievable with digital signals is because these signals are based on 1s and 0s. The repeaters (or digital amplifiers) only need to determine whether or not to restore a 1 or a 0. This system results in a much cleaner sound being sustained even though signals are being repeated along the way.

When the advantages of digital communication signals were fully acknowledged the telephony networks moved over to pulse code modulation (PCM). PCM is the most frequently used process for encoding analogue voice signals into digital signals made up of 1s and 0s.

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What Is SIP (Session Initiation Protocol) & What’s It Got To Do With VoIP?

The Session Initiation Protocol (SIP) is the IETF basis used for the creation of multimedia sessions. The sessions may be utilized for video, audio, instant-messaging amongst many other real-time based data communication sessions.

Constructing a whole setup which uses the SIP protocol needs the creator to understand the base SIP specification & a whole host of protocol documents which are aimed for a specific application. For instance, SIP (Session Initiation Protocol) is defined separately from the SIP specification for Instant Messaging plus when SIP is used with VoIP (Voice Over IP) the individual setting it up needs to have a grasp of SDP (Session Description Protocol) and that is a completely separate RFC. The SIP protocol being ‘modular’ in this way is seen as a strength.

SIP (Session Initiation Protocol) has a relatively broad scope which includes the ability of pretty much any variety of session connecting two individuals. SIP is as well totally free from the core vehicle, however, TCP (Transmission Control Protocol) & UDP (User Datagram Protocol) are utilized practically solely.

SIP was originally available as an Internet Draft from the IETF during 1996, and the primary RFC was in 1999.

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IP-001 USB VoIP Phone Overview - Internet Phone and VoIP Phone

The IP-001 USB VoIP phone is a great new VoIP internet phone developed by Logos Asia Ltd in mainland China. The fantastic thing about this VoIP internet phone is that is utilizes the USB interface making it very easy to use and setup whatever computer skill level you may have. Plus because of the nature of USB, the IP-001 USB VoIP phone is also plug-and-play compatible making it easy to move the VoIP phone between computers should you need to. With the funky and well thought out design of the IP-001 USB VoIP phone it is not only pleasant to look at but also very simple and easy to use.

IP-001 USB VoIP Phone

IP-001 USB VoIP phone main features:

  • The IP-001 USB VoIP phone supports SIP, H.323, MGCP.
  • Also supported are the VoIP networks: Skype, MSN, X-pro, X-lite, Net2Phone, eysBeam, Firefly, SJphone and StanaPhone.
  • The IP-001 is easy to use in a similar fashion to a mobile phone.
  • Utilizes a 16bit sound card internally.
  • When using the IP-001 USB VoIP phone with Win98 SE, Win ME, Win 2k, Win XP or Mac OS no drivers are required for installation.
  • The IP-001 USB VoIP phone has built in echo cancellation, noise reduction and full duplex communication abilities.
  • The IP-001 can have its handset earpiece volume and ring tone volume adjusted to suit.
  • Both PC-toPC and PC-to-Phone operation is possible with the IP-001 USB VoIP phone.
  • The IP-001 USB VoIP phone requires no external power supply because it is USB bus-powered.
  • The IP-001 VoIP phone is perfectly suited for use as a Skype phone.
  • Plug-and-play compatible due to USB interface.
  • The IP-001 USB VoIP phone is simple to operate and configure lending itself well to home and office/SOHO users.
  • USB A-type connector.
  • Conforms to USB 12 Mbps spec. version 1.1
  • Conforms to USB audio device release 1.0
  • Operation temperature: 0 to 40 degrees Celsius.

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