Technology VoIP Applications Description A Route Group is a list of one or more gateways or ports on gateways that are seen as equal access. It is analogous to a trunk group in traditional PBX terminology. For instance, you may have two PRI circuits to the same carrier that can be used arbitrarily. A gateway (or a particular port on a gateway) can only be added to one Route Group. ... View more

Introduction A site-to-site VPN allows offices in multiple fixed locations to establish secure connections with each other over a public network such as the Internet. Site-to-site VPN extends the company's network, making computer resources from one location available to employees at other locations. An example of a company that needs a site-to-site VPN is a growing corporation with dozens of branch offices around the world. There are two types of site-to-site VPNs: Intranet-based -- If a company has one or more remote locations that they wish to join in a single private network, they can create an intranet VPN to connect each separate LAN to a single WAN. Extranet-based -- When a company has a close relationship with another company (such as a partner, supplier or customer), it can build an extranet VPN that connects those companies' LANs. This extranet VPN allows the companies to work together in a secure, shared network environment while preventing access to their separate intranets. Even though the purpose of a site-to-site VPN is different from that of a remote-access VPN, it could use some of the same software and equipment. Ideally, though, a site-to-site VPN should eliminate the need for each computer to run VPN client software as if it were on a remote-access VPN. Dedicated VPN client equipment, described later in this article, can accomplish this goal in a site-to-site VPN. ... View more

What is VPN? A VPN or Virtual Private Network is a method used to add security and privacy to private and public networks, like WiFi Hotspots and the Internet. VPNs are most often used by corporations to protect sensitive data. However, using a personal VPN is increasingly becoming more popular as more interactions that were previously face-to-face transition to the Internet. Privacy is increased with a VPN because the user’s initial IP address is replaced with one from the VPN provider. This method allows subscribers to attain an IP address from any gateway city the VPN service provides. For instance, you may live in San Francisco, but with a VPN, you can appear to live in Amsterdam, New York, or any number of gateway cities. VPN Security: Security is the main reason why corporations have used VPNs for years. There are increasingly simple methods to intercept data traveling to a network. WiFi spoofing and Firesheep are two easy ways to hack information. A useful analogy is that a firewall protects your data while on the computer and a VPN protects your data on the web. VPNs use advanced encryption protocols and secure tunneling techniques to encapsulate all online data transfers. Most savvy computer users wouldn’t dream of connecting to the Internet without a firewall and up-to-date antivirus. Evolving security threats and ever increasing reliance on the Internet make a VPN an essential part of well-rounded security. Integrity checks ensure that no data is lost and that the connection has not been hijacked. Since all traffic is protected, this method is preferred to proxies. Types of VPN: Site-to-site VPN Often abbreviated to S2SVPN. It’s a connection between two sites and encrypts all traffic between two (or multiple) subnets. There are two types of S2SVPN:     Policy-based: interesting traffic triggers an ACL and is encrypted and sent to the remote VPN peer.     Routed: traffic is routed into an encrypted tunnel to the remote VPN peer. DMVPN A dynamic multipoint VPN is not a protocol but more a technique using different protocols. One or more central hub routers are required, but the remote (spoke) routers can have dynamic IPs and more can be added without having to modify the configuration on the hub router(s), or any other spoke routers. The routers use a next-hop resolution protocol, combined with a dynamic routing protocol to discover remote peers and subnets. The VPN itself is a mGRE tunnel (GRE with multiple endpoints) which is encrypted. This way, traffic between spoke routers does not have to go through the hub router but can be sent directly from spoke to spoke. Client VPN A client VPN is an encrypted connection from one device towards a VPN router. It makes that one remote device appear as a member of a local subnet behind the VPN router. Traffic is tunneled from the device (usually a computer or laptop of a teleworker) towards the VPN router so that user has access to resources inside the company. It requires client software that needs to be installed and configured. SSLVPN This type of VPN works like a client VPN. The difference is that the remote client does not need preconfigured software, but instead the browser acts as VPN software. The browser needs to support active content, which every modern browser supports, either directly or through a plug-in. Traffic is tunneled over SSL (or TLS) to the SSLVPN router. From a networking perspective, traffic is tunneled over layer 4 instead of layer 3. The benefit is that the remote user does not need to configure anything and can simply log in to a web page to start the tunnel. The drawback that you’ll likely need a dedicated device as SSLVPN endpoint because this is not a standard feature. ... View more

Introduction The IEEE 802.3af is more commonly known as Power over Ethernet. This standard states that power may be delivered by an end-point power source, using either the active data wires of an Ethernet port or the spare wires, to a powered device. It was originally intended to support power to IP phones however, rapid development of Power over Ethernet capable devices has extended its usage to other relevant applications such as wireless access points, Video surveillance cameras, digital signage and electronic badge readers. For more information also visit What is Power over Ethernet (PoE)? References Power over Ethernet Solutions ... View more

Description Adaptive Differential Pulse Code Modulation (A DPCM ) Complete  Definition Adaptive DPCM (A DPCM ) is a  variant of DPCM which in  term is a  form of PCM, DPCM varies the size of  the quantization  step, to allow further reduction of the required bandwidth for  a given  signal-to-noise ratio. Delta modulation (differential PCM), another variant, uses one bit per   sample. In telephony, a standard audio signal for a single phone call is encoded as   8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital   signal known as DS0. The default signal compression encoding on a DS0 is either   μ-law (mu-law) PCM (North America and Japan) or a-law PCM (Europe and  most of the rest of the world).  These are logarithmic compression systems where  a 12 or 13 bit linear PCM sample number is mapped into an 8  bit value.  This system is described by international standard G.711. An  alternative proposal for a floating  point representation, with 5 bit mantissa  and 3 bit radix, was abandoned. Where circuit costs are high and loss of voice quality is acceptable, it   sometimes makes sense to compress the voice signal even further. An ADPCM   algorithm is used to map a series of 8 bit u-law (or a-law) PCM samples into a series of 4 bit ADPCM  samples.  In this way, the capacity of the line is doubled. The technique is  detailed in  the G.726 standard. Later it was found that even further compression was possible and additional   standards were published. Some of these international standards describe  systems  and ideas which are covered by privately owned patents and thus use of  these  standards requires payments to the patent holders. Some ADPCM techniques are used in Voice over IP communications. ADPCM  Wikipedia Definition Also See: PCM DPCM codec ... View more

Description ITU-T companding standard used in the conversion  between analog and  digital signals in PCM systems. A-law is used primarily in  European telephone  networks and is similar to the North American mu-law standard. Complete  Definition An A-law algorithm is a regular companding algorithm. For  optimizing or  modifying the dynamic range of an analog signal for digitizing,  it is utilized  in European digital communications systems. It is fairly akin to  the mu-law  algorithm used in Japan and North America. The wide-ranging active  range of  speech does not lend itself well to proficient linear digital  encoding. A-law algorithm encoding efficiently lessens the dynamic range of the  signal.  Thus, the coding effectiveness gets increased and results in a   signal-to-distortion ratio which is advanced to that obtained by linear  encoding  for a particular number of bits. A-law algorithm is an audio  compression scheme  defined by Consultative Committee for International  Telephony and Telegraphy or  CCITT G.711 which constrict  16-bit linear PCM data down to  8  bits of logarithmic data. By restricting the linear sample values to 12   magnitude bits, the A-law compression is defined by its equation. In the   equation, A is the compression parameter and x is the normalized integer to be   compressed. It is a linear estimation of logarithmic input/output relationship.   It is put into practice by using 8-bit code words. 8-bit code words permit for  a  bit rate of 64 kilobits per second. By multiplying the sampling rate by the size of the code word, this is   calculated.A-law algorithm breaks a vibrant range into a total of 16 segments.   There are 8 positive and 8 negative segments. Each segment is twice the length   of the preceding one. Within each segment, homogeneous quantization is used.  For  coding the 8-bit word, 1st (MSB) identifies polarity. Then, bits 2, 3 and  4  recognize segment. The final 4 bits quantize the segment.A-law algorithm has   certain edges over the mu-law algorithm. The A-law algorithm provides a  somewhat  superior dynamic range than the mu-law algorithm, at the cost of  substandard  relative alteration for small signals. For an unvarying PCM, A-law  algorithm calls for 13-bits. For using  A-law algorithm, a worldwide link is  mandatory.The application of A-law  algorithm has been observed extensively Both a-law and u-law are  companders for the G.711 voice  codec A-law Compander formula (from Cisco's  Waveform Coding Techniques ) A-Law Wikipedia   Definition Also useful is the : ITU  website and  search on a-law Also See: mu-law u-law G.711 codec PCM RTP ... View more

Description: Companding technique commonly used in North America and Japan.  U-law is standardized as a  64- kbps CODEC in ITU-T G.711. This is a  CCITT-ratified audio encoding  and compression technique supported by Windows 95  and Web phones. Among other  implementations, u-law was originally intended as a  phone-communications  standard. Complete  Definition: The µ-law algorithm (pronounced mew-law) is a companding algorithm, primarily used in  the digital telecommunication systems of North  America and Japan. As with other  companding algorithms,  its  purpose is to reduce the dynamic range of an audio signal. In the analog   domain, this can increase the signal-to-noise ratio (SNR) achieved during  transmission, and in the  digital domain, it can reduce the quantization error  (hence increasing signal to  quantization noise ratio). These signal-to-noise  ratio (SNR) increases can be traded instead for reduced bandwidth for equivalent  SNR . Both a-law and u-law are  companders for the G.711 voice  codec U-law Compander formula (from Cisco's Waveform Coding Techniques ) u-Law  Wikipedia  Definition Also useful is the : ITU  website and  search on u-law Also See: mu-law a-law G.711 codec PCM RTP ... View more

Description Consultative Committee for International Telegraph and Telephone (CCITT) Complete  Definition A former group that has been renamed to the ITU-T body. The standardization work of ITU dates back to 1865, with the birth of the   International Telegraph Union. It became a United Nations specialized agency in   1947, and the International Telegraph and Telephone Consultative Committee   (CCITT), (from the French name "Comité Consultatif International Téléphonique  et  Télégraphique") was created in 1956. It was renamed ITU-T in 1993 Wikipedia CCITT  Definition See Also ITU-T ... View more

Description mu-law (optional typing of u-law) Complete  Definition mu-law is a more phonetic typing of the term u-law (pronounced mew-law) , which is a companding algorithm. Also See: u-law a-law G.711 codec PCM RTP ... View more

Differential Pulse Code Modulation (DPCM) Complete Definition DPCM is and advanced form of PCM. Differential (or Delta) pulse-code modulation (DPCM) encodes the PCM values as  differences between the current and  the previous value. For audio this type of  encoding reduces the number of bits  required per sample by about 25% compared  to PCM. Predict  the next sample based on the last  few decoded samples. Wikipedia  Definition of  Delta Modulation Wikipedia  Definition of DPCM Also See: PCM ADPCM codec u-law ... View more

Description G.711 Voice codec standard Complete  Definition There are two main compression algorithms defined in the standard, the u-law algorithm  (used in North  America & Japan) and a-law algorithm (used in Europe and the rest of  the world). Both are logarithmic, but  a-law was specifically designed  to be simpler  for a computer to process. The standard also defines a sequence of  repeating  code values which defines the power level of 0 dB. The u-law and A-law algorithms encode 14-bit and 13-bit signed linear PCM   samples (respectively) to logarithmic 8-bit samples. Thus, the G.711 encoder   will create a 64 kbps bitstream  for a signal sampled at 8 kHz. G.711, also known as Pulse Code Modulation (PCM), is a very commonly used waveform codec.   G.711 uses a sampling rate of 8,000 samples per second, with the tolerance on   that rate 50 parts per million (ppm). Non-uniform quantization with 8 bits is   used to represent each sample, resulting in a 64 kbit/s bit rate. There are two   slightly different versions; μ-law, which is used primarily in North America,   and a-law,  which is in use in  most other countries outside North America. G.711 u-law tends to  give more resolution to higher  range signals while G.711 a-law provides  more quantization levels at lower signal levels. When using u-law G.711 in  networks where  suppression of the all 0 character signal is required, the  character signal  corresponding to negative input values between decision values  numbers 127 and  128 should be 00000010 and the value at the decoder output is  -7519. The  corresponding decoder output value number is 125. Wikipedia G.711  Definition ITU  G.711 Status Also See a-law U-law PCM G.703/G.704 G723.1 G.726 G.728 G.729 ... View more

Description G.726 Voice codec audio  standard Complete  Definition G.726 is an ITU-T ADPCM speech codec standard covering the transmission of   voice at rates of 16, 24, 32, and 40 kbps. It was introduced to supersede both   G.721, which covered ADPCM at 32  kbps, and G.723, which  described  ADPCM for 24 and 40 kbps. G.726  also  introduced a new 16 kbps rate. The four bit rates associated with G.726 are   often referred to by the bit size of a sample, which are 2-bits, 3-bits,  4-bits,  and 5-bits respectively. The most commonly used mode is 32 kbps, since this is half the rate of  G.711,  thus increasing the usable network capacity by 100%. It is primarily  used on  international trunks in the phone network. It also is the standard  codec used in  DECT wireless phone systems and is used on some Canon  cameras. ITU  G.726 Status Wikipedia Definition for   G.726 Also See G.703/G.704 G.711 G.723.1 G.728 G.729 ... View more

Description International Telecommunication Union Telecommunication standardization sector (ITU-T) Complete  Definition International body that develops worldwide standards for telecommunications   technologies. The ITU-T carries out the functions of the former CCITT. . Wikipedia ITU-T  definition See also: CCITT Also useful is the : ITU  website search. ... View more

Description G.723.1 is an audio codec for voice audio. Complete  Definition G.723.1 is an audio codec for voice that compresses voice audio in 30 ms   frames. An algorithmic look-ahead of 7.5 ms duration means that total   algorithmic delay is 37.5 ms. ITU  G.723.1 Status Wikipedia Definition  for  G.723.1 Also See G.703/G.704 G.711 G.726 G.728 G.729 ... View more

Description G.729 Voice [[codec] audio standard Complete  Definition G.729 is an ITU-T algorithm  for voice encoding that  produces an 80-bit voice sample every 10 msec (bit rate  of 8 kbps). The codec  works in blocks of 10 msec and so it is possible to  generate frames of multiple  10 msec duration. G.729 is an audio data compression algorithm for voice that compresses   voice audio in chunks of 10 milliseconds. Music or tones such as DTMF or fax  tones cannot be transported reliably  with this codec, and thus use G.711 or  out-of-band methods to transport these  signals. G.729 is mostly used in Voice over IP applications for its low bandwidth   requirement. Standard G.729 operates at 8 kbit/s, but there are extensions,   which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and   better speech quality respectively. Also very common is G.729a which is   compatible with G.729, but requires less computation. This lower complexity is   not free since speech quality is marginally worsened. The annex B of G.729 is a silence compression scheme, which has a VAD module which  is used to detect  voice activity, speech or non speech. It also includes a DTX  module which  decides on updating the background noise parameters for non speech  (noisy  frames). These frames which are transmitted to update the background  noise  parameters are called SID frames. A comfort noise generator (CNG) is also  there  because in a communication channel, if transmission is stopped, because  it's not  speech, then the other side may assume that link has been cut. This is  also  taken care of by the annex B standard. Recently, G.729 has been extended to provide support for wideband speech and   audio coding, i.e., the transmitted acoustic frequency range is extended to 50   Hz - 7 kHz. The respective extension to G.729 is referred to as G.729.1. The   G.729.1 coder is hierarchically organized: Its bit rate and the obtained  quality  are adjustable by simple bitstream truncation. ITU  G.729 Status Wikipedia Definition for   G.729 Also See G.703/G.704 G.711 G.723.1 G.726 G.728 ... View more