Wednesday, October 5, 2011

Basics of VOS 3000 : Part -1

In VOS3000, most data managements can be completed through sheets. Sheets can be opened by double-clicking corresponding nodes in (navigation). The following operations are supported:
  • Filter: Get current configuration from server.
  • Copy: Copy the currently selected sheet line into the clipboard.
  • Paste: Paste the line in the clipboard into a sheet with the same type.
  • Insert: Insert new lines.
  • Delete: Delete sheet lines. If the data are at the server, the selected lines will be marked as “to be deleted”.
  • Apply: Send currently specified operations (such as insertion, deletion and modification) to the server to carry out. (*** Before clicking “apply”, all the operations of data are saved only at the client end and will not affect the server-end data; closing the management page would discard these operations)
  • Export: Export the current sheet into local files.
  • Import: Import data from local files into the sheet (supported by a few types of sheets) Batch data operations can be fulfilled by “copy”, “paste”, and column “fill-down” functions supported by spreadsheets in VOS3000. See the illustration below:

After downloading the VOS3000 install the software in your computer.
Upon running VOS3000 client, the login dialogue box will be shown.
  • Server IP: IP address of the remote server.
  • User Name: User names allowed by the platform.
  • Password: User password allowed by the platform.

The system will record IP addresses typed by users for later use. Users can also delete this historical server IP. The initial User Name and Password are both admin.

After Login:
After successful logged in the opening screen will appear (Click the below picture to enlarge).

At the left side navigation menu you will see the following options:
  • Rate Management
  • Package Management
  • Account Management
  • Operation Management
  • Audio Management
  • Data Query
  • CDR Analysis
  • Cards Management
  • System Management
  • Number Management

At the top you will see the Menu Bar. The functions in the Menu Bar can also be found in the Navigation Menu. These Are:

Operation Management
Audio Management
Data Query
CDR Analysis
Cards Management
System Management
Number Management

On the Tools Bar you will also find the following options:

Related Story:

Friday, September 30, 2011

Wholesale VoIP Solution

The birth of Voice-over-IP technology in late 1990s and its rapid growth in early 2000s, have resulted in the proliferation of many alternative telecom providers. Those providers have introduced a number of innovative communications services such as calling cards and callback which relied on call origination and termination over the Internet.

The growing number of new telecom providers led to increased demand for traffic aggregation services and as a result the Wholesale VoIP business was born. Wholesale VoIP providers act as traffic aggregates and/or traffic exchanges for their customers.

They sign up contracts with both retail and wholesale VoIP providers and act as a middle man for call origination and termination services. When partners send traffic to the Wholesale VoIP provider, he/she reroutes it to other partners for termination and makes profit from the difference in negotiated rates.

Business Solution
PROVIDER Softswitch will offer end-to-end, cost-effective and scalable Wholesale VoIP solution. That solution will feature powerful billing, flexible routing, and proven interoperability with equipment from other leading VoIP vendors.

Because all solution components will be developed by PROVIDER and Softswitch manufacturer we can eliminate integration issues and for that we can quickly start and/or expand our Wholesale VoIP business while enjoying high return on investment.

How will the solution work?

PROVIDER Experience
  • PROVIDER will receive VOIP Traffic request from Partner 1 for termination.
  • Then PROVIDER Softswitch will accept the traffic and will send authorization request to PROVIDER VoIP Billing server. The VoIP billing server will verify the account balance of Partner 1 and will authorize call termination.
  • PROVIDER Softswitch will re-route the traffic to Partner 2, hiding traffic source information.
  • Upon completing the call, the PROVIDER VoIP Billing server records will debit the account of Partner 1 and will record CDR record of the call.

Solution components are required
Core elements of PROVIDER Wholesale VoIP Solution include:
  • PROVIDER VOIP Billing Server is a robust billing server that provides our customers with all necessary tools to successfully implement a wide spectrum of VoIP business models.
  • PROVIDER Softswitch is an advanced VoIP softswitch that provides our customer with secure and reliable peering between their own networks and the VoIP networks of their business partners.
Other Posts:
CallShop Solution

Sunday, September 25, 2011

Callshop Solution

The rapid development of Voice-over-IP technology in the early 2000s has lead to fast innovation in the telecom sector. New telecom providers have introduced many innovative services utilizing VoIP technology, such as calling cards, call shops, broadband VoIP and others. Because such services are delivered over broadband lines, countries with well established last mile Internet infrastructure tend to adopt faster VoIP services where end-users install VoIP equipment at their premises (e.g. IP Phones), such as broadband VoIP. 

Alternatively, countries with limited or expensive last mile Internet infrastructure tend to adopt VoIP services that do not require end-users to install equipment, such as calling cards and call shops.

The Callshop business model appeals to providers in countries with limited bandwidth or expensive access to the Internet. Call shop services also offer lucrative opportunity in countries with limited number of telecom providers and heavy telecom regulation.

Consumers, in such countries, typically do not have access to low cost long distance and international telephony services. Finally, Callshops tend to flourish in areas which attract international tourism. In any of the above cases, Call shop operators can capitalize on VoIP technology and offer competitively priced calling services to any part of the world.

Business Solution
PROVIDER will offer end-to-end, cost-effective and scalable Callshop solution. The solution will feature powerful billing, flexible routing, and proven interoperability with VoIP equipment from other leading voip provider. Because PROVIDER will develop all solution components, callshop owners benefit from reduced integration costs and improved return on investment.

How will the solution work?

User experience
  • When a customer visit the Callshop. Customer will chooses a vacant telephony booth, will enters and place a call by dialing a destination number.
  • The call will get connected.
  • Upon call completion, the callshop operator will present the customer with an invoice for accumulated call charges.
  • Then the customer will pay the callshop operator.

Provider experience
  • When the user picks up the phone and dials a destination number, VoIP Gateway will send an authorization request to PROVIDER VoIP Billing Server.
  • Billing Server will verify whether a call can be placed from that particular booth and will authorize the call.
  • Then VOIP Gateway will pass the destination number to PROVIDER Softswitch and will requests routing information.
  • PROVIDER Softswitch will return VOIP Gateway the IP address of the remote (termination) gateway and VOIP Gateway will connect to it. Then the remote gateway will terminate the call to the destination party.
  • Upon call completion, PROVIDER VOIP Billing Server will records complete CDR information for the call and will make it available to the operator for billing, reporting and monitoring purposes.

Solution components are required
Core elements of SysMaster's Callshop Solution include:
  • PROVIDER VOIP Billing Server that supports a wide spectrum of prepaid post-paid VoIP services.
  • VOIP Gateway will be a carrier grade gateway that provides universal IP-PSTN switching, high flexibility and remote feature upgrade ability.
  • PROVIDER Softswitch is a flexible VoIP Softswitch that offers routing and reliable peering between VoIP networks and capable for high volume callshop implementations.

Other Posts:

Wednesday, September 14, 2011

Calling Card Business - Using VoIP Technology

Traditionally, the telecom sector has been one of the most lucrative market segments in both emerging and developed economies. But also traditionally, that sector has been protected by heavy local regulation. Nowadays, however, many countries have either started or are considering deregulation of their telecom sectors. Such strong trend towards liberalization of the sector, coupled with the emergence of new technologies for cost effective transport of voice data, and namely Voice-over-IP, opens up new revenue generating opportunities for entrepreneurs all over the world.

The calling cards business model offers a relatively low cost entry into the lucrative telecom market segment. The essence of the calling cards business model is to creatively segment customers by various demographic and/or behavioral characteristics and to design calling cards offerings to meet their specific calling needs. The calling cards business typically attracts entrepreneurs who want to enter the VoIP market, businesses with established retail distribution channels, and service providers who want to diversify their revenue streams.

Business Solution
Aiming PROVIDER as a leading VoIP service provider, PROVIDER has to offer integrated, scalable, and cost-effective calling cards solutions. Such solutions will feature powerful billing capabilities, intuitive CRM (Customer Relation Management) web portals, and proven interoperability with equipment from other VoIP providers. Because all solution components are developed by the engineers of PROVIDER and Softswitch Billing company, that reduces the need for integration costs at the PROVIDER VOIP service level which saves money for PROVIDER and improves our return on investment.

How will the solution work?
User experience
  • The user will purchase a calling card from a retail or online store.
  • The user will dial the local or toll-free access number printed on the card.
  • The user will hear a voice prompt, asking him to enter his PIN number.
  • The user will then enter the PIN number printed on the calling card using the phone dial pad.
  • He will hear his account balance and is invited to make a call.
  • The user will dial the number and will get connected.

Provider experience
  • When the user calls, PROVIDER Gateway, which is connected to the access number, will answer the call and send IVR prompts to the caller, inviting him to enter his PIN number.
  • PROVIDER Gateway will read the PIN and will send an authorization request to PROVIDER VoIP Billing
  • PROVIDER VoIP Billing will verify account information and will authorize the call if the user has sufficient balance.
  • Then PROVIDER Gateway will announce to the caller his account balance and will invite him to dial a destination number.
  • Then PROVIDER Gateway will pass the destination number to PROVIDER Softswitch and will request routing information.
  • PROVIDER Softswitch will return to PROVIDER Getaway the IP address of the remote gateway and Getaway will connect the call.
  • During the conversation, Getaway will convert voice signal to data packets and will route them to the terminating gateway (owned by the call termination provider) and vice versa.
  • Upon call completion, PROVIDER VoIP Billing will record complete CDR information for the call and then debit the user account with accumulated service charges.

Solution components are required
Core elements of PROVIDER's Calling Cards Solution include:
  • PROVIDER VoIP Billing, will be a robust billing server that provides us with all necessary tools to successfully implement a wide spectrum of VoIP business models.
  • VoIP Gateway, will be a flexible switching device that provides us with universal IP-PSTN switching, high flexibility and remote feature upgrade ability.
  • PROVIDER Softswitch will be an advanced VoIP softswitch that provides secure and reliable peering between our own networks and the VoIP networks of our business partners.

Related Topic:

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.

Other article can be read:

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.

Related Other Topics:

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.

Other related topics:

Friday, July 29, 2011

What is SIP Dialer - a brief discussion

A SIP Dialer is called a softphone dialer or Mobile dialer or PC dialer which is used to make phone calls directly to any land phone number or mobile number in the world using VoIP (Voice over Internet Protocol). It is one kind of call center management software which is use in many Call Centers for their regular communication with their client and customers. SIP stands for Session Initiation Protocol.

There are some special advantages of using this kind of call center management software or SIP dialer. This are able to work in both Unicast and Multicast server. Through Unicast the conversation take place between two parties and for Multicast the conversation take place more than two parties which is called teleconferencing. There are some other special features for which the SIP dialer is going to be popular day by day.

SIP dialer or softphone has a number of advantages like:
  • Availability: It is available to download for free on the internet. Many provider provide free version of a SIP dialer to download, user can make calls using this free version directly to any land line or mobile phone number in the world.

  • Cost effectiveness: Using a SIP dialer you can make calls via VoIP (Voice over Internet Protocol), which is the most cost effective way to make phone calls. Traditional phone line or mobile phone can be used to make calls by paying a huge amount of bill.

  • Time saving ability: Some of its special features like teleconferencing facility, broadcasting recorded messages or SMS to a large group of people simultaneously save time when using SIP dialer services.

  • A SIP Dialer is very much predictive, Call centers use this kind of Softphone regularly for their business effectiveness, simple to use functions etc. Using this dialer communication with potential customer and clients is getting more easy and reliable.

  • After downloading the software in the computer it becomes easy to send single SMS or a recorded message to a client or multiple recorded messages or SMS to thousands of recipients.

  • A SIP dialer can easily installed in computer or standard mobile handset. Users can make VoIP phone calls directly from their computer or mobile using SIP dialer. At present people around Middle East countries like KSA, UAE, Qatar, Oman etc. are using this service to make overseas calls. Voice and video calls can also be making from a SIP dialer. The main requirement to make call is internet connection while using in both computer and mobile handset. The quality of voice and video conversation is depend on some other things like voice carrier route, softswitch, byte saver etc. If all these requirements got fulfilled than this kind of calling method is one of the best in the world for cost effectiveness.

Some other use of SIP Dialer:
  • Corporate business offices use SIP dialer service due to its good quality video and voice (single and multiple) services. SIP can sometimes eliminate the need of overnight travel to a business meeting.
  • Doctors who are giving patients prescription around the world using teleconference are using this service regularly.
  • Many university / college / school use this service to taught courses in the classroom or outside the classroom.

Related topics:

Saturday, July 23, 2011

What is SIP - Session Initiation Protocol

To provide advanced telephony service over the Internet (VoIP) a Protocol named Session Initiation Protocol (SIP) has been developed by the IETF (Internet Engineering Task Force). Session Initiation Protocol or SIP is signaling protocol used to established sessions in an IP network like two-way call (IP-2-IP call, Net-2-Phone call etc), a collaborative multi-media conference system. SIP calls may be terminal to terminal, or they may require a server to intercede. If a server is to be involved, it is only required to locate the called party. For inter-working with non-IP networks, Megaco and H.323 are required. Often vendors of VoIP equipment integrate all three protocols on a single platform.

Various telecommunication services like Mobile dialer or PC Dialer service, Calling card services, Click and dial from Web page, Instant messaging, IP Centrex service and many other Voice-enriched E-Commerce services has become possible with the help of Session Initiation Protocol or SIP.

SIP has been designed upon some other protocol like HTTP (Hypertext Transfer protocol), SMTP (Simple Mail Transfer Protocol). SIP borrows most of its syntax and semantics from the familiar HTTP. In an IP-based network it is used to setup, change or end calls between two or more users.

In two method calls can be setup using SIP called, Redirect and Proxy and the server are designed to handle these modes. Both modes issue an “invite” message for another user to participate in a call. The redirect server is used to supply the address (URL) of an unknown called addressee. In this case the “invite” message is sent to the redirect server, which consults the location server for address information. Once this address information is sent to the calling user, a second “invite” message is issued, now with the correct address.

The following features of SIP are playing a major role in the field of VoIP (Voice over Internet Protocol):

Media negotiation: The inbuilt SIP mechanism that allow concession of the media used in a call, enable selection of the proper codec system to making a call between various devices. Thus, simple devices can make VoIP phone call, using the selected codec system.

Feature Negotiation: This feature allows the group involved in a call to agree on the features supported. This may be a multi-party call. All the parties can support the same level of features. For some codec video may or may not be supported; as any form of MIME type is supported by SIP. During making a call one user can bring other users onto the call or cancel the call or may placed or hold the call.

User location and name translation: It ensure that the participation of the second user in the call wherever he is located. Carrying out any mapping of expressive information to location information. Ensuring that details of the nature of the call (Session) are supported.

Changes of Call features: Using SIP a user can setup Voice-Only mode, but during making a call the caller need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call.

Some special features of Session Initiation Protocol:
  • SIP messages are text based and hence are easy to read and debug. Programming new services is easier and more intuitive for designers.
  • SIP re-uses MIME type explanation in the similar way that email clients do, so applications related with sessions can be launched automatically.
  • SIP re-uses a number of existing and mature internet services and protocols such as RTP, DNS, RSVP etc. No new services have to be introduced to support the SIP infrastructure, as much of it is already in place or available off the shelf.
  • SIP extensions are defined clearly, enabling VoIP service providers to add them for new applications without damaging their own networks. Older SIP-based equipment in the network will not hamper newer SIP-based services. An older SIP implementation that does not support technique / title utilized by a newer SIP application would simply ignore it. SIP is transport layer independent. Therefore, the underlying transport could be IP over ATM.
  • SIP uses UDP - User Datagram Protocol, as well as the TCP - Transmission Control Protocol, lithely connecting users independent of the primary communications.
  • SIP supports multi-device feature leveling and negotiation. If a service or session initiates video and voice, voice can still be transmitted to non-video enabled devices, or other device features can be used such as one way video streaming.

SIP sessions use up to four major components:
  • SIP User Agents
  • SIP Registrar Servers
  • SIP Proxy Servers and
  • SIP Redirect Servers.

Together, these systems deliver messages embedded with the SDP protocol defining their content and characteristics to complete a SIP session.

Further reading about SIP.

Other topic:

Monday, July 4, 2011

JAJAH Mobile Web - Direct Call from Mobile Web Browser

Make calls directly from your web-enabled mobile handset and save money when talking. Using the browser of your web-enabled Smart phone (Blackberry, Treo, iPAD, Smart phones by Motorola, Nokia, SonyEricsson, Samsung etc..) you can make one-click free global call around the world. No download is required to use JAJAH Mobile Web. User can make calls directly by log in with their username and password.


  • Run on special phone which runs on Microsoft Windows Mobile, Microsoft Windows CE, Pocket PC, Symbian Operating System.
  • JAJAH Mobile Web service is compatible with any standard mobile handset with internet connection that has WAP 2.0 or XHTML web browser.
  • It works any mobile network in any countries that has GSM and CDMA network which includes 99% of all with a minimum of 2.0 G dada service.
  • For GSM Network GPRS or EDGE is required and for CDMA network 1X(RTT) and EVDO is required.
  • For both type of networks use the term 3G and HSPDA to talk about high speed mobile data networks. All of this network type allows the end-user to use JAJAH mobile web.

  • User can make free calls to other JAJAH users and to make calls all over the world (any Landline or Mobile number) the JAJAH users can make calls with a competitive and low price. JAJAH mobile web is different if compare with other VoIP service provider because some of its special features. User can directly make calls all over the world simply using their mobile web browser. For details rate plan Click here.

  • In fall 2006 JAJAH launched a special JAJAH Mobile plug-in, a downloadable application for mobile phones, it attracted thousands of customers across the globe. Using these service users was able to make calls by sending a SMS. But today the new JAJAH Mobile Web is specially designed fro Smart Phones give opportunity to its users to make low-cost phone calls simply from their mobile web browser.

  • JAJAH Mobile web service can be use when you are in roaming. Roaming is very high-cost service any provider. But JAJAH gives you the opportunity to save this cost. TO do so you need to change your registered mobile number. During travel you can buy a local SIM card to use in your handset. Now using JAJAH mobile web you can update your registered mobile number by simply pressing the “Change” link at the top of the page. Now in the new page change your registered mobile number entering in the text box. Alternatively, you can choose one of your home, office or mobile numbers as your source number. Finally press the "Save" button and you'll return to the previous page with your updated source number. This system helps you to save your roaming fees while you are in travel.

How to make calls using JAJAH Mobile Web:
  • Open your mobile web browser.
  • Log into your JAJAH account using your username and password if you are already JAJAH user. If you are a new user than you need to get registered with JAJAH service, sign up is absolutely free.
  • For faster access next time you can bookmark this page.
  • Type the number you want to make call or you can select your saved number from your address book. To make calls check your current balance at the top of the page.
  • Now press the “Call” button and start talking.
JAJAH.Mobile Web connects you.

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Wednesday, June 29, 2011

Make direct call from your browser - Using JAJAH Web

No headset, no need to download, no need to install any software if you want to use JAJAH Web. Start talking now directly from JAJAH Web and save your money.

How to use JAJAH Web:
Open your favorite web browser and go to to make a call. After get registered you can make calls directly from your web browser. Sign up is absolutely free; before get registered you can make a trial call using JAJAH Web. To make VoIP Phone calls directly from JAJAH Website simply follow the following instructions:
  • Go to JAJAH website
  • Log into your own account using your Username and Password
  • Select the destination country where you want to make call
  • Enter the number in the destination field
  • Press the CALL button and start talking.
The process is really easy. After pressing the CALL button your phone will ring, pick up the phone call and your call will be forwarded to your destination number. JAJAH will forward your call to your destination number automatically. Just wait for the ring back tone and when your friend receives your call start talking. SIMPLE !

Special Features of JAJAH Web:

Low Calling Cost:
JAJAH Web is one of the lowest rated calling systems from web browser around the world. You can save a lot of money using JAJAH various special offers. Go to for further details about their rate plan and other promotion.

Conference Calls using JAJAH:
JAJAH give you the facility to make conference calls around the world and it is one of the lowest cost conferences calling system in the world. To make a conference call no password or no call-in-number is needed. The JAJAH system first calls to your regular phone and start the conference calling automatically by calling the other participants.

JAJAH Scheduled Calls:
This is one of the special attractions of JAJAH Web. It offers you to make scheduled call any time. This scheduled call is great for business, for sales, families. You can fix up your business call to your clients; you can fix up call to make birthday wishes to your friends and family. And even you will get an SMS direct from JAJAH which will remind you about your scheduled call. This special function can be applied also for conference call.

Text SMS using JAJAH:
JAJAH offers you to send special rate for SMS around the world. Like scheduled call you can also scheduled an SMS to your friends or business clients etc. JAJAH lets you schedule calls, minutes, hours, days ahead of time. Great for business. Great for sales. Great for families. (Never miss a birthday). JAJAH will even send you a SMS / text "Reminder" minutes before the call is made. You can schedule regular calls and conference calls.

For further details about JAJAH Web:

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