Tech NEWS and REVIEWS: July 2011

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:

What is SIP

gPlex Mobile Dialer

iTel Mobile Dialer

iTel Call Through Dialer

iTel Call Back Dialer

Best Mobile Dialer for KSA, UAE

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:

What is SIP Dialer

gPlex Mobile Dialer

iTel Mobile Dialer

iTel Call Through Dialer

iTel Call Back Dialer

Best Mobile Dialer for KSA, UAE

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.

Requirements:

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.

Features:

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.

Go to http://mobile.jajah.com/

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.