Voice over IP is a telecommunication technology. The primary function of VoIP is to control the transmission of human voice data over Internet. The usage of traditional public switched telephone network (PSTN) for voice transmission is limited by the development of VoIP. The signal transmission of VoIP involves conversion of analog signal to digital signal and transmitting the digital signal as discrete data packets to the destination. The most important benefit of using internet telephony and VoIP is that it eliminates the tariff charged by typical telephone service. Voice over Internet Protocol (VoIP) is a communication technology that enables one to compose voice calls using a broadband Internet connection instead of a normal telephone line. This technology initially converts the voice signal into a digital signal which can travel through internet. If a call is made to a regular telephone line then the digital signal is converted into voice/analog signal before it reaches the sound card of the receiver. VOIP is a very complex, large, and rapidly sprouting technology. The rabid development of VoIP enables the communication between two terminals to be continuous, flexible, scalable, constant and most importantly reliable.
2VOIP Security Terminology:
1.2.1Tunneling and Encryption:
The secure facility in VoIP is offered by two important factors known as Tunneling and Encryption. Employing these security measures provides a reliable environment for the personal data of VoIP user. In order to keep the internet hackers at pay, the providers of VoIP uses Layer 2 tunneling and Secure Sockets Layer or SSL encryption methods.
The advantage with this strategy is that it allows an organization to logically split and secure separate voice and data networks in front of individual voice and data components and between interactive points within the network. A system (like the passwords and firewalls), encryption, an audit trail of calls, and facilities.
1.2.2Secure Real Time Protocol (SRTP)
It is defined as a protocol which is the profile of the RTP "Real-time Transport Protocol". It offers confidentiality, message authentication and replay protection for the RTP traffic as well as for the Real-time Transport Control Protocol "RTCP".
1.2.3Compression of Packet Size
Compression of Packet Size results in considerably less jitter, latency, and better crypto-engine performance. The crypto-engine performance also improves.
1.3VOIP Communication Factors
Cost :
Speed and Quality:
Privacy and Legal Issues with VOIP:
VOIP Security Issues :
1.4VoIP QOS Requirement Parameters
Latency
It indicates the time take by the voice signal to travel from its source to destination.
Jitter
Jitter in VoIP indicates the difference in the transmission of data packets that are occurred due to the diversion in the signal route and/or congestion in the network. in general Jitter is controlled by the usage of a Jitter buffer. This factor has a huge impact on quality of service (QOS).
Packet Loss
It occurs most often in VoIP due to large latency. Packet loss occurs when a set of packets reaches late and is eliminated in favor of newer data packets.
Bandwidth & Effective Bandwidth
Effective bandwidth is defined by Beriberi et al. as "the percentage of bandwidth carrying actual data with regard to the total bandwidth used."
The Need for Speed
The key to conquering QoS issues like latency and bandwidth congestion is speed. By definition, faster throughput means reduced latency and probabilistically reduces the chances of severe bandwidth congestion. Thus every facet of network traversal must be completed quickly in VOIP.
1.5VoIP Diagram:
1.6 Some VOIP Security System:
1.6.1SecureLogix:
The highly scalable and cost efficient VoIP security system is developed by a corporation called SecureLogix. A set of integrated management and security applications called Enterprise Telephony Management (ETM) System that identify and charge off telephony-based attacks and abuse, while simplifying management of the VoIP system. It is combined package of robust management appliances and platform security applications.
1.7 Objectives for VoIP Project
Integration of IP-based internet applications, such as email and unified messaging, with voice applications
The expansion of VoIP services so as to support several multimedia applications and providing a chance for cost effective video streaming, video conferencing, gamming, and other media applications.
1.8 Aim of VoIP
The flexibility of next generation VoIP platforms allows for the rapid development of new services and development cycles are typically greater than for ATM or TDM.
In addition the consolidation of voice and data in one network can significantly reduce cost.
VoIP leverages data network capacity removing the requirement to operate separate voice and data networks.
VoIP equipment is typically faster and cheaper than ATM or TDM-based equipment - a gap that is increasing rapidly every few months.
Re-routing of IP networks is much cheaper than existing systems.
1.9 Document Scope and Purpose
The purpose of this document is to provide guidance for establishing secure VOIP networks. VoIP are encouraged to tailor the recommended guidelines and solutions to meet their specific security or business requirements. VOIP security considerations for the public switched telephone network are largely outside the scope of this document. Although legal issues regarding VOIP are beyond the scope of this document, readers should be aware that laws and rulings governing interception or monitoring of VOIP lines, and retention of call records, may be different from those for conventional telephone systems.
.
CHAPTER-III
3.1 SYSTEM SPECIFICATION
3.1.1HARDWARE REQUIREMENTS:
PROCESSOR : PENTIUM III 766 MHz
RAM : 128 MD SD RAM
MONITOR : 15" COLOR
HARD DISK : 120 GB
CDDRIVE : LG 52X
KEYBOARD : STANDARD 102 KEYS
MOUSE : 3 BUTTONS
Head Phone : Any one
Webcam : Any one
The hardware used for the development of the project is:
3.1.2.SOFTWARE REQUIREMENTS
OPERATING SYSTEM : Windows Xp Professional
ENVIRONMENT : Visual Studio .NET 2008
.NET FRAMEWORK : Version 3.5
LANGUAGE : C#.NET
BACK END : SQL SERVER 2005
DLL : Microsoft DirectX
The software used for the development of the project is:
3.1.3 SYSTEM ARCHITECTURE
Business Solutions
Data base
Security & Privacy
VoIP Applications
Video meeting with call
Voice Chatting
Online Text Chatting
VoIP Processing
User Authentication
VoIP User Login
VoIP Voice Processing
Voice Calls
END USER Terminal VoIP
INTERNET
UDP
Secure Real-Time Protocol
Data Compression
Analog Digital input
CHAPTER -IV
4.1Development and Implementation
4.1.1 VIOP
VoIP is the abbreviation for 'Voice over Internet Protocol'. It transfers the human voice signal in the form of individual IP packets through the internet. In order to generate these IP packets VoIP uses a set of accelerating hardware devices that are normally found in a PC.
It was discovered the signal transition to a remote location can also be done in digital method. The analog signal to be transmitted is converted to digital format using an Analog to Digital converter (ADC). The signal is then transmitted. At the receiver end a Digital to Analog convertor (DAC) is used to decode the digital signals into analog signals. This principle is used in VoIP to digitize the voice signals as data packets, transmit them and decode them into analog signals at the receiver end. Employing a Digital signal enhances the transmission process in several ways. It is easy to control, simple compression, simple to route, easier to convert and so on. Digital signals are more noise tolerant than analog signals. The data packets employed in TCP/IP network contains header and pay load information. This data is used by VoIP to transmit and receive signals across the network.
4.1.2Technical facts of VoIP
4.1.2.1 VoIP connection an overview
In order to establish a successful VoIP connection,
1. Initially an Analog to Digital converter (ADC) is required to convert the analog signals into digital signals. The converted signals are then compressed into a suitable format for transmission. There are different protocols employed in compression technique which will be discussed later in this chapter.
2. The next process is to insert voice packets in data packets. This is done using a real time protocol namely RTP on UDP on IP.
3. A signaling protocol is required to initiate a user call. This is achieved by a protocol called ITU-T H323. At receiver end all the packets are disassembled and converted into analog signals. It is then transmitted to sound card.
4.1.3Analog to Digital Conversion
This conversion is done by a card integrated ADC hardware. In contemporary technical world, all the sound cards enables one with 16 bit a band of 22050 Hz (minimum sampling frequency is 44100 Hz for Nyquist Principle) achieving a throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/s, 176.4 kBytes/s for stereo stream. For VoIP it requires even lesser throughput (176kBytes/s) to transmit a voice packet.
4.1.4Compression Algorithms
The digital signal is then compressed into a standard format for a smooth and efficient transmission.
In PCM technique only the difference between the present and previous data packets is converted which requires 32 kbps.
These are most significant protocol since it requires a very low minimal band using source coding. G.723.1 protocols have a very high MOS (Mean Opinion Score, used to measure voice fidelity) but attention to elaboration performance required by them, up to 26 MIPS!
4.1.5 RTP Real Time Transport Protocol
In order to encapsulate the raw data into TCP/IP stack a standard structure is followed, VoIP data packets
The data packets of VoIP are inserted into Real−Time Transport Protocol packets (RTP), which is present inside UDP−IP packets. Initially, TCP is not used in VoIP because it is too profound for real time applications; hence UDP is employed. But, UDP is not able to control the time period at which the data arrives or order in which packets appear at the RX. These two phenomenons are significant for over all voice and conversation quality. Employing RTP solves the problem by allowing the RX to rearrange the packets into correct format and not wait too long for data packets that are taking too long to arrive or have lost their way.
4.1.6 Quality of Service (QoS)
In case of an interactive voice data exchange, a real time data streaming is required by the VoIP connection. Unfortunately, a TCP/IP protocol does not assure this kind of idea, it just attempts to do it. Hence, a better queuing technique is required to manage the flow of data packets across every router.
Queuing packets methods:
First in First out (FIFO), the most commonly used technique that helps the data packets to arrive in correct order at RX. Weighted Fair Queuing (WFQ), this method is used in a reasonable transmission of packets; it sends one packet for UDP and one packet for TCP in a reasonable manner.
Custom Queuing (CQ), this method allows the users to choose the priority of packets. Priority Queuing (PQ), this method uses a number of queues each one containing different priority levels. The packets with highest priority are sent initially and when the first queue becomes empty it starts transmitting the second packet.
Class Based Weighted Fair Queuing (CB−WFQ), it is similar to WFQ but, with additional classes and bandwidth for every packet. The shaping capability of this method allows source to limit at a fixed bandwidth in terms upload, download, and elimination of congestion.
4.1.7 Signaling Protocol H323
The H323 protocol is widely employed in making of VoIP calls. Apart from VoIP connection, H323 also allows data and video communications. This protocol enables different elements to communicate with each other. The VoIP connection is normally initiated by either at terminals or by clients. Despite the fact that terminals can communicate with each other without help, it requires some other elements called Gate keepers to obtain a scalable image.
The essential functions of Gate keepers are,
To use names instead IP addresses for addressing a transmission.
To provide appropriate admission control by allowing or denying the users or hosts.
To provide efficient bandwidth management
In PSTN gateways are used as points of reference for TCP/IP conversion.
Gateways are used in Multipoint Control Units (MCUs) to enable conference calls.
It is also used in Proxy servers.
4.1.8Project description
Videoconferencing, Voice chat, File Sharing, and Text-chat. Conference Sessions with multi-party live video, file sharing, Private Chat, Presence, VoIP and Instant Messaging. Compatible with popular firewall products.
4.1.9 Module Description:
Login Module:
In this module if you are already register you have to give user name and password. If you are not having login id you have to register first then you have use and enjoy.
Already you have a login id but if you forget password or user name means you have use forget password to retrieve your user and password. In this forget password they will ask security question and Mail id.
Register Module
If you are new to this application first you have to register your details and some security question. The security questions are very import to future if you loss your user name or password you can retrieve that form this one only.
Forget Password
Text chat:
In this application two type of chatting using public and private chat.
Features of Text chat
Chat with one person or with up to multiple people at once.
Easily invite your contacts to join a text chat session.
Send chat messages to offline contacts.
All your text chats are saved automatically will be saved.
Public Chat:
Private Chat:
Voice Chat Module:
In case of a voice chat application the audio signal is from the micro phone is received using direct sound and then it is send in UDP packets. In this module I have used G711 voice coder to compress the audio data before transmitting it. In order to know more about audio signal capturing, look into the references given at the end of this article. The architecture of this application is discussed in the following section.
Collapse
| Invite |
| --------------------------------> |
| OK |
| <-------------------------------- |
| |
| --------------------------------> |
| Audio flow |
| <-------------------------------- |
| Bye |
| --------------------------------> |
A B
If a user needs to make call, he/she has to send an open message and should wait for an OK response. When an OK is received, the user will start to transmit/receive an audio signal captured from the microphone. If the user at destination rejects the call then a busy response will be sent call initiator. If the user wants to end a call a simple bye message is sent to receiver. The application will synchronously receive/send audio data on port 1550 and asynchronously receive/send call messages on port 1450. However, the application simultaneously attends on two ports: one for call messages and the other for audio data.
In real time application, two variants called mu−law (North America) and a−law (Europe) are used that logarithmically scales the analog signal code by using 12 or 13 bits instead of 8 bits (G.711).But we are also using mu-law and a-law in this application.
Video chat Module:
Videoconferencing uses telecommunications of audio and video to bring people at different sites together for a meeting. This can be as simple as a conversation between two people in private offices (point-to-point) or involve several sites (multi-point) with more than one person in large rooms at different sites. Besides the audio and visual transmission of meeting activities, videoconferencing can be used to share documents, computer-displayed information, and whiteboards.
Simple analog videoconferences could be established as early as the invention of the television. Such videoconferencing systems usually consisted of two closed-circuit television systems connected via cable.
File transfer Module:
In this file transfer module if you want to send file one user to another means first you chose the file then you have ask permission then only you can send file to that person. In case if that person not accept your process means you cant send file to that person. We are done large size file transfer application and also very good performance.
CHAPTER -V
5.1 Evaluation and Testing Criteria
The VoIP can be ensured effective deployment and continuing service quality by relying on passive and active test solutions for provisioning, monitoring and service level fault management. Passive and active telephony testing techniques were developed to measure the quality of digital PSTN voice networks.The passive test systems are used to measure signaling performance, while active bit-error-rate (BER) testing is used to monitor PSTN call quality.Passive testing involves either collecting traffic statistics from soft switches, media gateways, and other network elements, or requires placing test-probes at strategic network locations to "sniff" live IP traffic and filter out VoIP packets.
VoIP use a combination of active and passive testing techniques for monitoring and fault management. Whereas passive testing offers good visibility into call signaling performance, active testing offers operators the best possible technique to analyze end-to-end media quality, the true end-to-end customer experience of service quality. In active testing, probes place short test calls that are recorded then analyzed by industry-standard algorithms to produce a detailed, ear-to-mouth service quality assessment. Active testing addresses these various requirements using different test layouts, optimized for provisioning, troubleshooting, and monitoring. Test layouts include innovative test methods that can be combined to provide comprehensive coverage from subscriber to central office/head-end, and all points in between.
Single Ended test Layout
The single-end test layout measures voice, signaling, fax, internet and modem service quality without using a far-end probe.
A test probe records call progress and content for playback and analysis.
A location database provides public telephone, fax, modem, or IP destinations for testing on or off the service provider's network.
Tests calls are designed to assess service quality quickly - minimizing long-distance billing charges by hanging up just after the connection is established.
The test server typically reports results by area code, city, network provider or country.
Single-ended testing is typically used by long-distance service providers to verify call quality over partner carrier networks, and to automate least-cost-routing processes.
It can also be used for troubleshooting to specific phone numbers, and to routinely spot-check quality on geographically-large networks.
Probe to Probe Test Layout
In probe-to-probe testing, test calls are placed from one probe to another, providing complete control over test stream content and quality measurements in both directions.
Tests can be initiated and received on any standard signaling interface over Next-Generation and legacy networks.
Tests record call progress as well as the actual voice/data traffic for playback and signal analysis.
Probe-to-probe testing is typically used to monitor VoIP, PSTN, video, internet, and fax/modem service quality over large-scale IP, TDM and mobile networks.
Test automation is used to schedule network-wide test calling patterns between all probes in the network.
When a service quality issue is identified, the operator can choose to focus test calls over the problematic route to quickly isolate and troubleshoot the problem.
Responder Test Layout
Responders replace far-end test probes when portability and cost are primary concerns.
Responders are ideal tools for performing day-of-install testing and troubleshooting, and validation.
Active test responders can receive and transmit test traffic, and they are connected by 2-wire analog or IP ports.
Analog connections allow technicians to replace a telephone handset with a responder to measure user-perceived quality.
VoIP installation involves by using an analog responder to quickly test call quality from all the phone jacks in a home, measuring MOS, distortion, etc.
IP responders are usually used to test RTP traffic streams for VoIP and video quality assessment.
Loop Back Test Layout
Integration with soft switches and network components allows active test systems to conduct analog and RTP/IP loopback tests from test probes
Loopback tests reduce the need to deploy technicians to resolve network and service performance issues, and require no investment in portable test equipment.
Loopback testing is often used by Cable MSOs to streamline operations and reduce support costs, and to accelerate problem resolution.
Loopback testing can quickly isolate a service issue by testing to a subscriber's home, neighboring homes, and nearby transponders.
Data from RF-level transponder tests, IP and analog loopback tests to the MTA can be correlated to remotely determine on what network layer the problem originates and what steps are required to correct the problem.
Interactive Test Agent Layout
Voice-guided testing is the recent active test technology in which the test probe hosts an interactive voice response agent that uses natural speech to guide technicians or subscribers through a series of simple steps leading to detailed speech quality, DTMF, echo and noise tests without any far-end responder.
Designed to reduce installation, troubleshooting and trouble-ticket resolution time, interactive testing is ideally integrated into customer support systems to provide immediate test data when customers call with service issues.
Test results can be used by customer service to guide the user through corrective actions, by operations staff to remotely correct the problem or to dispatch a field technician if field repair is required.
Integrating Test layouts
A test automation platform is used to manage and integrate the results from passive and active test layouts and applications, centralizing all aspects of VoIP service delivery and maintenance.
An integrated test platform enforces centralized standards and test plans, provides event correlation and reporting, and is easily integrated into existing operational support systems - allowing all departments to work and communicate within a single, quality-focused framework.
An integrated, comprehensive test strategy that encompasses VoIP delivery from day-of-install to proactive service monitoring can be addressed using a combination of automated active and passive testing
An effective service level test automation platform can integrate results from different test layouts, allowing all departments to work towards their respective goals within a common framework.
With a test platform, a service quality focus can be integrated into existing operational support systems, fault management systems, and existing business processes - giving the visibility required competing in today's competitive telephony landscape.
The VoIP Hop Test
This test is widely performed by VoIP Administrators and Network Engineers. It is performed to identify security risks on their network. The security validation test can be executed using three fundamental procedures.
Unplug Ethernet cable of IP Phone and directly terminate into laptop
Initially, the PC/Laptop must be directly disconnected by the Ethernet cable form the network jack on the wall, instead of being disconnected by Ethernet port on IP phone.
The IP phone which supports three-port switch function has two Ethernet ports. These ports act a doorway for incoming and outgoing Ethernet frames of the originating station.
The initial procedure is directly depended on the ability to unassumingly unplug the IP Phone and attacker having physical access to the cabling and terminate their laptop directly into the wall.
The security risks of VoIP networks can be reduced if the IP Phone Ethernet cabling is physically protected against tampering, and by monitoring the IP phone using CCTV.
Sniff for the Voice VLAN ID
Create the Voice VLAN interface on PC
The last step in this verification procedure is to enable the Linux PC for 802.1q VLAN tagging in the Ethernet frame headers.
By default, PCs are not enabled for this feature or functionality. In this method, the tester enables a Voice VLAN interface on the Linux PC.
Download the "802.1Q VLAN Implementation for Linux":
Unpack the compressed file:
Verify that the correct interface used for VoIP Hopping is up
Run the 'vconfig' utility to add the VLAN interface based on the discovered VLAN ID
Verify that new Interface is created
Send a DHCP client request for an IP address on the Voice VLAN
If the DHCP server returns a DHCP lease for an IP address, then any PC can successfully VoIP Hop onto the Voice VLAN, simulating the behavior of an IP Phone - the Layer two networks allows a successful VoIP Hop.
Depending on the VoIP call scenarios and network design, this can represent a critical vulnerability in the configuration. This raises questions on how to prevent or mitigate attacks based on the VoIP Hop.
5.2 Planning for VoIP Deployment
VoIP is still an emerging technology, which develop a complete picture of what a mature worldwide VoIP network will one day look like. Competing protocols and designs for the infrastructure of the net flourished at the time, but as the purpose of the Internet became more defined with the emergence of the world wide web and other staples of today's net, the structure and protocols became standardized and interoperability became much easier. Although there are currently many different architectures and protocols to choose from, eventually a dominant standard will emerge. The most obvious of these competing standards are SIP and H.323.
SIP is a fast growing protocol with similarities to current Internet standards such as HTTP, but it has yet to reach the level of deployment of H.323. The opinion of many academics [16, 18] seems to favor SIP, and we have seen that some of the security issues associated with VoIP becomes simpler with the SIP scheme. In fact, non-standardized VoIP environment, organizations looking to integrate several VoIP networks ought to support both protocols. Several companies have developed infrastructure elements to enable multi-protocol telephony. As voice and data networks converge, support for both protocols is essential for a robust and forward-looking network.
Although the future will probably see the emergence of one of these protocols as the defined standard in the field the present disorganization makes support for both protocols in a VoIP network a pertinent issue. Deploying a VoIP network in today's non-standardized world requires support for both protocols. If organizations moving to VoIP should seek out gateways and other network elements that can support both H.323 and SIP. This strategy helps to ensure a stable and robust VoIP network in the years that come, no matter which protocol prevails.
VoIP security today is the choice of end-to-end VPNs versus firewall-based VPNs i.e. VoIP traffic must traverse firewalls one way or the other. The use of VPNs has been touted by many industry articles as the definitive solution to the tribulations posed by firewall and NAT traversal in tunnel mode. However, much of their research has focused on small-scale operations where VoIP phones are not used in the volume needed to overwhelm the crypto-engines or congest the network enough to cause a significant downturn in QoS. The ability to use firewalls for analyzing VoIP traffic for malicious or suspicious patterns would be lost. This being said, the implementation of VoIP aware firewalls and proxies incurs a significant cost now and in the future.
Such protocol specific hardware would need to be upgraded each time standards evolve. A third solution that has not been fully developed yet is a hybrid system, where call setup information (H.323 or SIP) is sent through a VoIP aware gateway/firewall but the RTP traffic itself is encrypted and tunneled over VPN. The call setup protocols could be secured using their proprietary authentication mechanisms in place of the IPsec tunnel. This combines the network protection of the firewall and the data security/protection of IPsec. Also, the robust authentication mechanisms and abstraction of the voice network from the data network accomplished by SIP proxies and H.323 gateways/gatekeepers would be preserved. However, no expanded study has been done on the ramifications of this hybrid approach.
CHAPTER-VI
CONCLUSION
6.1 Conclusion and Future work
VoIP is internet telephony ubiquitous and cost-effective. By carrying to a VoIP network a mover can:
Deploy new converged voice and data services
Remove the need to manager separate voice and data networks
Utilize cheaper IP-based backbone equipment to carry voice
Reap the benefits of standards-based and highly flexible network design, giving a competitive market between equipment vendors and encompassing a wide range of equipment for different market niches.
Now a days there are a number of VoIP solutions available today, most of these have limitations of one kind or another. In some cases the solutions are built on early versions of standards and provide restricted interoperability with other vendors.
In some cases the solutions do not provide the scalability, robustness, security or features required for PSTN equivalency. The proposed VoIP is committed to providing a next generation network that provides both full multi-vendor interoperability, and support for a full featured, secure VoIP service.
6.3 Related Software
6.3.1SKYPE
Skype is a software application that allows users to make voice calls over the Internet. Calls to other users within the Skype service are free, while calls to both traditional landline telephones and mobile phones can be made for a fee using a debit-based user account system. Skype has also become popular for its additional features which include instant messaging, file transfer, and video conferencing. The network is operated by a company called Skype Limited, headquartered in Luxembourg and partly owned by eBay.
Unlike other VoIP services, the Skype Company does not run servers, but makes use of background processing on computers running Skype software-the original name proposed, Sky peer-to-peer (see below) reflects this.
Features of skype
Registered users are identified by the unique skype name
It allows instant messaging and voice calls by using skype online number
It supports for panel discussions, lectures, and town hall forums.
It is safe ansd secure for communications.
Security and privacy
It provides a secure communication and the encryption cannot be deactivated and invisible to user
It provides an uncontrolled registration system for users with no proof of identity
It hides the traffic and route the data
The saved messages contain personally identifiable information about the messages senders and recipients
Disadvantages
The new version of Skype (3.6.32.244) creates this problem. Upon the initial install, everything worked as expected. No problems connecting to or using the service.
After a reboot however Skype will no longer connect. Additionally, the process will use 100% of the cpu and continue to grow in memory usage until the process is killed. This happens some other program clash with skype or anti-virus or a program called Total Recorder.
Skype credit cards are at higher rates at some places.
6.3.2 Gizmo5
Gizmo5 have accquired by google.It is similar to skype and available all over the country
It is an open-source VoIP softphone that enables free or minimally expensive calls worldwide. Its spinoff, SIPphone.com, is a startup offering more complex, business-oriented tools that use Gizmo's protocols, including PSTN (public switched telephone network) gateways, voice mail and SIP (Session Initiation Protocol)/PSTN network peering.
Gizmo features Call In and Call Out (similar to SkypeIn and SkypeOut). typically for 1.9 cents per minute, instant messaging, and the ability to record conversations or map where caller and receiver are located. Pro: Available for Windows, Linux and Mac platforms.
Advantages
Cheap Computer Calling
Make cheap calls to anywhere
No Per Call Connection Fee
No hidden charges when calling
Works with Google Voice
Use it with your Google Voice account
6.3.3 Google Talk
Google talk comes in two categories and they are1. A widget to use from a Google site (with Flash 8.0), 2.a 1.5MB download.
It integrates directly with a user's Google Personalized Homepage, and the Google Talk application automatically loads your contacts from a Gmail account, speeding setup time.
Disadvantages
Unfortunately, the Google Talk Client currently works only with Windows and BlackBerry devices
Although Google promises Linux and Mac OS X versions in the future. Pro: Interfaces available in a dozen languages, including British English.
Condition: Google Talk does not connect with landline phones or mobiles.
6.3.4 Jajah
It offers the chance to make entirely free PC-to-PC VoIP calls and a limited number of free calls to and from landline phones.
You visit Jajah's Web site, then enter your phone number and the number you want to call.
Your phone rings and you answer, after which your recipient's phone Rings and jajah completes the call.Jajah operates on an honor system that allows you about 1,000 minutes per month of free calls because other Jajah users pay to use the company's premium services (such as business accounts, calls to non-Jajah users and some foreign calls).
Go over that limit, and Jajah will ask that you purchase some premium services; if you don't, you may be cut off (and remember, they have your phone number).
DisAdvantages
Pro: Mac-friendly, with a plug-in that connects Jajah calling with the OS X Address Book, plus an Outlook plug-in that is in beta.
Con: Ceiling on free calls limits usefulness for high-volume callers.
If your company has some IT experts lying around, consider WengoPhone an open-source application being developed through OpenWengo.com. WengoPhone Classic provides voice and video over IP, while the WengoPhone NG project currently underway completely rewrites Classic to be more modular, extensible and VoIP-provider agnostic.
Pro: Open-source.
Con: Probably unusable by the nonprogrammer.
6.3.5 ooVoo
It lets you to videoconference with up to six people at once and sends out video messages rather than emails if your fingers are numb from typing.
It's easy to get set up -all you need to get started video chatting is a webcam. You'll look and sound your best with ooVoo's high quality audio and video, unsurpassed by the competition.
ooVoo sells several compatible third-party cameras, headsets and speakerphones and runs its own servers, making a Skype-like outage less likely.
If your friend doesn't have ooVoo, you can have a free browser-based chat with them by sending your unique Web Video Call link through email, IM or through your favorite social networking
Feature
Up to 6 participants in a call
Import your contacts from IM, social network, email, or more
Record and send video messages
Alter your view in Full Screen Mode
Record and store audio and video calls
Text chat and share files instantly while on a call
DisAdvantage
Although it's currently available only for Windows users, ooVoo's conferencing setup looks strikingly like iChat's, and the company is now developing Mac-friendly software.
Pro: Makes video-spamming friends easy.
Con: Friends can spam you back.
6.3.6 SightSpeed
It emphasizes free PC-to-PC voice calling and videoconferencing, and its $4.95-per- month Pro service adds video-mail recording and multiparty videoconferencing.
The patented video technology ensures you get the best possible video and sound quality every time, unlike other online video services. SightSpeed offers the highest-quality, full-motion 30 frames per second video with clear audio and no annoying delay.
Available for Windows and Mac.
Pro: MySightSpeed feature allows voice and videoconferencing via browser with non-SightSpeed members
Con: ooVoo does most of the same things for free.
Features
Connect with everyone in your network, face to face, anyplace
Save time, hassle, and money spent on travel
Help save our environment by minimizing your carbon footprint!
SightSpeed users are able to have free video calls with each other.
Video Mail messages can be sent to any e-mail address.
The SightSpeed software is able to host video conferenceswith up to nine participants.
"Phone Out" is the name of the service that allows users to make phone calls to landlines and mobile phones, paying with pre-paid credit. "Phone In" allows SightSpeed users to receive phone calls from regular telephones
Limitation
Lack of true privacy features such as encryption.
SightSpeed "Phone Out" does not support outbound calleridin the United States, where people commonly reject calls from unrecognized numbers. The recipient sees "unknown" or a blank field instead of the caller ID number.
Though it uses the SIP standard it is not interoperable with SIP networks and can only be used with a SightSpeed account.
6.3.7 VBuzzer
It allows PC-to-PC voice VoIP-based calling, faxing and videoconferencing for free, with nominal rates for completion to landlines and mobiles worldwide (typically 1.5 cents per minute to the U.S., Canada and China and, 1.7 cents per minute to other countries).
It is a voice over IP and instant messaging software and service, based on open protocols of XMPP
Pro: Works with relatively old computers (minimum 500 MHz chip speed).
Con: Currently available only for Windows 2000 and XP.
Features
Vbuzzer allow users to exchange text messaging, call/fax to the PSTN and reveive incoming PSTN calls/faxes.
Vbuzzer also offers free Voicemail, Call forward, Caller ID, Chat history logs and RSS Feeds.
Version 2.0.282 and up can work with Windows Live Messenger (WLM)
Receive a US or Canada phone number for unlimited incoming calls.
Free Faxing within US, Canada and China
Works with Nokia E61,E51,N95 cell phone SIP client
6.3.8 Voip Buster
It provides free calls - for a low price.
It is a mistranslation from the original German, VoipBuster advertises that if you buy credits, (called "Freedays") you can then make up to 300 minutes per week of "free" calls to landlines in three dozen countries, plus mobiles in the U.S., New Zealand, Puerto Rico and Hong Kong. Go over that 300-minute limit, and you pay VoipBuster's normal rates, which are still quite cheap at 1.2 eurocents per minute including VAT.
VoipBuster is a free program that uses the latest technology to bring free and high-quality voice communications to people all over the world.
When you use the free VoipBuster software, you can call regular phones in various popular destinations for free or call at an incredible low rate to any other phone on the planet.
You can also call all your online friends (pc-to-pc calls) as long as you like, for free.
The program also enables you to import contacts from MSN, Skype, and Outlook, send contacts to other users, search for a contact and archive contacts.All in all,
VoipBuster could be a good alternative when choosing a voip application
Pro: Very cheap rates for toll calls.
Con: Confusing idea of "free."
Features
It provides at low cost than other software
It has high quality voice communications
It is a handy VoIP application
It is simple and intuitive
DisAdvantage
No video integration.
Non-existent customer support.
The interface is not really attractive.
