Tuesday, August 16, 2011


VoIP (Voice over Internet Protocol) is going to be more popular around the world comparatively with PSTN (Public Switched Telephone Network). The main reason for growing market for VoIP over PSTN is availability, reliability and low cost etc. In this article I will show that there is no reason to limit the expectations to achieve only the same level of quality as in PSTN.

By deploying wideband codec, much better voice quality can be achieved. But using narrowband codec there are some standard solution that can be used in VoIP to get finest quality voice. Such as, traditional PSTN solution does not use the available spectral bandwidth, but these can be done easily in VoIP system. Implementing a wideband codec or expanding the bandwidth of narrowband codec does not automatically guarantee great quality. There are many potential pitfalls when deploying VoIP. There are some issues need to be discussed that effect the final voice quality over PSTN or VoIP. In this article I will discuss what level of quality can be achieved and describe how this can be implemented.

Speech Signals and Speech Coding Sampled digital signals can hold frequency content up to half the sampling frequency. As we know, a young adult has a hearing span from about 20 Hz to 20,000 Hz.
And the sampling frequency of CD audio is chosen to be 44.1 kHz, which is more than double that of the highest frequency perceivable by most humans. Legacy telephony solutions are narrowband, which seriously limits the achievable quality. Wideband codec could potentially be used in digital telephone systems, but this has never been practical enough to gain any real interest

In traditional telephony applications such as PSTN, the speech bandwidth is restricted much more than the inherent limitations of narrowband coding. Typical telephony speech is band limited to 300 Hz to 3400 Hz (listen to Sound Pure digital connections are found in enterprise environments.
Due to poor connections or old wires, significant distortion is often generated in the analog part of the phone connection, a type of distortion that is entirely absent from VoIP implementations. The cordless phones so popular today also generate significant amounts of analog distortion due to radio interference and other implementation issues.

Calls between two parties in a PSTN (Public Switched Telephone Network) are connect by a series of private or public switches. The resulting fixed communications link is dedicated for the duration of the call. When an individual makes a phone call over a circuit-switched network, a connection is made between the providers PBX and the local telephone company, also known as the PSTN. Depending on the destination, this connection might extend to the national or international exchange before reaching another local exchange, where it will be passed on to the PBX and the person who receives the call. This end-to-end link, established by a series of public and private switches, is 100% dedicated on a single, per-call basis and cannot be shared or used for another function as long as the call is in progress. For this reason, these dedicated circuits cannot be shared and the carrier bills the call on a time and distance rate.

But the Internet does not use switches to link calling parties. Instead, the analog voice signal is digitized by an Internet Protocol (IP) and broken up into thousands of small data packets by a router – the VoIP equivalent to a switch. These data packets are sent, or routed, over the public Internet to their destination, enabling calls to bypass the PSTN entirely.

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Thursday, August 11, 2011

What is H.323 Signaling Protocol - A basic

To provide some special services through the internet (IP based network) such as real-time audio / video communication, data communication H.323 is a standard signaling protocol which components, protocols and procedures. H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia communication services over a variety of networks.

H.323 was invented to provide various multimedia applications through LAN (Local Area Network). But due to its popularity and rapidly growing services it has evolved in needs of VoIP (Voice over Internet Protocol). One important strength of H.323 was the relatively early availability of a set of standards. It was not only used to define basic call models but in addition to supplementary services that can easily involved in Business Communications. The first VoIP standard protocol is H.323 to adopt IETF standard RTP to transport audio and video communications over the internet (IP network). H.323 standard specifies four kinds of components, when networked together, provide the "point-to-point" and "point-to-multipoint" multimedia-communication services. These are:
  • Terminals
  • Gateway
  • Gatekeepers
  • Multipoint Control Unit

Terminals: Used for real-time bi-directional multimedia communications, an H.323 terminal can either be a personal computer (PC) or a stand-alone device, running an H.323 and the multimedia applications. It supports audio communications and can optionally support video or data communications. Because the basic service provided by an H.323 terminal is audio communications, an H.323 terminal plays a key role in IP–telephony services. The primary goal of H.323 is to inter-work with other multimedia terminals. H.323 terminals are compatible with H.324 terminals on SCN and wireless networks, H.310 terminals on B– ISDN, H.320 terminals on ISDN, H.321 terminals on B–ISDN, and H.322 terminals on guaranteed QoS LANs. H.323 terminals may be used in multipoint conferences.

Gateways: A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non – H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and SCN networks ( SCN networks include all switched telephony networks, e.g., public switched telephone network [PSTN] ). This connectivity of dissimilar networks is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals on an H.323 network.

Gatekeepers: A gatekeeper can be considered the brain of the H.323 network. It is the focal point for all calls within the H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization and authentication of terminals and gateways; bandwidth management; accounting; billing; and charging. Gatekeepers may also provide call-routing services.

Multipoint Control Units: MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video coder/decoder (CODEC) to use, and may handle the media stream. The gatekeepers, gateways, and MCUs are logically separate components of the H.323 standard but can be implemented as a single physical device.

Key Benefits of H.323:

Codec Standards: 
H.323 establishes standards for compression and decompression of audio and video data streams, ensuring that equipment from different vendors will have some area of common support.

Users want to conference without worrying about compatibility at the receiving point. Besides ensuring that the receiver can decompress the information, H.323 establishes methods for receiving clients to communicate capabilities to the sender. The standard also establishes common call setup and control protocols.

Network Independence:
H.323 is designed to run on top of common network architectures. As network technology evolves, and as bandwidth management techniques improve, H.323-based solutions will be able to take advantage of those enhanced capabilities.

Platform and Application Independence:
H.323 is not tied to any hardware or operating system. H.323-compliant platforms will be available in many sizes and shapes, including video-enabled personal computers, dedicated platforms, IP-enabled telephone handsets, cable TV set-top boxes and turnkey boxes.

Bandwidth Management:
Video and audio traffic is bandwidth-intensive and could clog the corporate network. H.323 addresses this issue by providing bandwidth management. Network managers can limit the number of simultaneous H.323 connections within their network or the amount of bandwidth available to H.323 applications. These limits ensure that critical traffic will not be disrupted.

An H.323 conference can include endpoints with different capabilities. For example, a terminal with audio-only capabilities can participate in a conference with terminals that have video and/or data capabilities. Furthermore, an H.323 multimedia terminal can share the data portion of a video conference with a T.120 data-only terminal, while sharing voice, video, and data with other H.323 terminals.

Inter-Network Conferencing:
Many users want to conference from a LAN to a remote site. For example, H.323 establishes a means of linking LAN-based desktop systems with ISDN-based group systems. H.323 uses common codec technology from different videoconferencing standards to minimize transcoding delays and to provide optimum performance.

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Wednesday, August 3, 2011

Ezzy Dialer - Works in blocked firewall network

In previous some of my posts I have discussed about some Mobile Dialer which works fine in Middle east countries (KSA, UAE, Oman, Qatar etc). These are iTel Mobile Dialer, gPlex Mobile Dialer. In this post I am going to discuss another mobile dialer called Ezzy Dialer which has become a good VoIP solution in the world.

The Ezzy dialer is a highly scalable VoIP carrier solution that works fine in Symbian operating system (2nd, 3rd, 5th edition) enabled handset. VoIP calls can easily make from Symbian OS supported handset from anywhere around the world.

The basic requirements to run Ezzy Mobile Dialer:
  • Internet connection (GPRS / EDGE / 3G / Wi Fi) enabled handsets.
  • Symbian operating system supported handsets.

Some special features of Ezzy Dialer:
  • Users can get crystal clear voice quality while making VoIP phone calls using Ezzy dialer.
  • It works fine in Low Bandwidth for some of its special technology called CNG (Comfort Noise Generation), VAD (Voice Activity Detection), PLC (Packet Loss Concealment) etc.
  • Works with SIP (Session Initiation Protocol) based Soft Switches and any type of Firewall network.
  • Provide details call log history around 50 call records.
  • Incoming and Outgoing call facility (IP to IP), using which user can make free and low cost call.
  • User friendly or easy to use interface.
  • Supports various codec systems like: G-729, G-711, G-723, GSM, AMR-NB etc.
  • Works with various internet connectivity’s like: EDGE, GPRS, 3G, Wi-Fi, VPN Access points etc.
  • User can see their balance on the screen and can hear their remaining balance using the IVR facility.
  • Calls can be make finding numbers directly from handset address book or from the call log history.
  • Loudspeaker / Speaker phone facility.
  • User can switch emergency to other mobile application while talking to others using dialer.
  • GSM call alert while VoIP call is running, this is another special features of Ezzy dialer which most dialer can’t provide.
  • External or internal memory of your mobile handset can be used to install the Ezzy dialer application.
  • The Ezzy dialer if flexible to install and it never conflict with other applications.

Supported handset: Below is the list of all supported handsets with Ezzy Dialer.

N70, N71, N72, N73, N75, N76, N77, N82, N85, N91, N92, N93, N93i, N95, N95_8gb, N96, N97, N97 Mini, 3250, 5228, 5230, 5235, 5530, 5800, 5800 Music Express, 6120, 6121, 6124, 6210, 6220, 6290, 6650, 6700, 6710, 6720, 6730, 6760, 6788, 6790, E50, E51, E52, E55, E60, E61, E61i, E62, E63, E65, E66, E70, E71, E72, E75, E90, C6, X6, X6_8gb, X6_16gb

G810, SGH 1400, SGH 1450, SGH 1520, SGH 1550, SGH 1560, GT i7110, GT i8510 Innov8, 1870

For further details about Ezzy Dialer and list of latest supported handset go to their site.

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