VOIPSURF.COM

Voip Solution for Education

Voip Surf — Tue, 09 Jun 2009 13:14:00 GMT

Pingtel is well established in serving the educational community with open source voice over IP (VoIP) offerings. Having publicly documented SIPxchange implementations at Houghton College, Hall-Dale Schools, and The University of the Arts–among other educational facilities–Pingtel offers state-of-the-art IP communications in the area it matters most: our schools. Pingtel is successful with educational facility implementations because we understand the unique needs of educational institutions, including the difference between faculty and administration, and between employees and students. It is because of this understanding that Pingtel offers a special program with pricing that reflects how phones are used at different levels of academia.

VOiP terminology

Voip Surf — Thu, 26 Mar 2009 10:24:00 GMT

Asynchronous communications

Transmission method whereby bits are sent without synchronizing via a clock signal, but instead using start and stop bits to identity the beginning and ending of each block of data.

ATA

Analog Telephone Adapter; a device by which you can connect a regular analog phone (wired or cordless) to the Internet to make and receive VoIP calls. It converts the analog signal from the phone to digital and is available from VoIP providers such as Vonage or Lingo.

Call processor

VoIP providers' equipment that receives the phone number you dial, checks it for format validity, and maps it to an IP address.

Circuit switching

Older, less efficient but reliable technology used by the regular public switched telephone network (also see PSTN and POTS). A connection called a circuit is established for the duration of the call.

Codec

Coder-decoder software that converts audio signals into compressed digital signals so they can be transmitted across a digital network. It then converts them back to analog at the other end.

Conference bridge

A device for connecting several parties in a phone call so that all participants can hear one another.

Data compression

Methods of reducing the number of bits in a set of data so it can be transmitted more quickly over the network and then expanded to its original size when it reaches the destination.

Endpoint

A phone or computer associated with a phone number and temporarily or permanently assigned an IP address.

Full duplex

The ability of devices at both ends of a communications to send and receive information simultaneously.

H 323

A set of protocol standards established by the International Telecommunications Unions (ITU) originally designed for video conferencing and also used for VoIP.

Half duplex

The ability to send data in two directions, but only one direction at a time.

High availability

Methods of ensuring rapid recovery from hardware or software failure employing redundancy and failover to backup components.

IP PBX

Internet Protocol-based Private Branch Exchange internal telephone switching system that supports convergence of voice and data networks for routing calls within a building or organization.

IP phone

Looks like an ordinary phone but connects to an IP router with an RJ-45 Ethernet connector. These phones run software that allows them to handle VoIP calls without going through an ATA.

IP telephony

All telephone type services that work over TCP/IP, including VoIP, text messaging, and IP-based faxing.

Internet Protocol; the network layer protocol by which computers on the Internet and other TCP/IP networks communicate with one another via unique binary addresses (32-bit addresses represented as "dotted quad" decimal addresses in IPv4 or 128-bit addresses represented as hexadecimal addresses in IPv6).

IVR

Interactive Voice Response; an application that allows users to access computerized information over the phone using keypad touchtones or voice commands. The commands are translated into digital queries and the results are returned from the computer hosting the information database. The results are then translated into computerized voice messages spoken to the caller.

Jitter

Variations in arrival time of data packets.

Latency

The amount of time it takes for a data packet to be transmitted from one endpoint to another.

Mapping

The process of determining to what IP address a VoIP call is to be routed, based on the phone number that is dialed.

Media Gateway Control Protocol

Protocol used to control telephony gateways.

North American Numbering Plan (NANP)

The system that the traditional phone networks use for routing calls based on the telephone number dialed.

Packet switching

Newer, more efficient technology used for IP communications on the Internet, by which data is broken into parts called packets. Different packets can take different routes to the destination, arriving out of order. They are reassembled into the original order at the destination.

Packet

A unit or "manageable chunk" of data into which complete messages are divided to be routed across the Internet or other TCP/IP network.

PoE

Power over Ethernet; a method of sending electrical power over Ethernet cable to alleviate the requirement to plug equipment into an electrical outlet or other power source.

POTS

Plain old telephone network; a telephone industry colloquial nickname for PSTN.

PSTN

Public switched telephone network; the traditional circuit switching network used for transmitting voice conversations. Also see POTS.

QoS

Quality of Service; a guaranteed or predictable level of bandwidth, transmission speed, and freedom from dropped packets, delay, jitter, and error that is necessary to ensure adequate performance of particular applications.

SCCP/Skinny

Skinny Client Control Protocol; IP telephony protocol developed by Cisco whereby the telephone can communicate with an H.323 proxy.

Simple Gateway Control Protocol (SGCP)

Protocol used to control telephony gateways.

Simplex

The ability to send data in only one direction.

SIP

Session Initiation Protocol; a small and efficient application layer protocol specifically designed for VoIP communications.

Soft switch

Programmable switch that processes signaling for packet protocols and can be used to integrate telephone signaling with packet switching networks.

Softkeys

Buttons on a telephone handset or software keypad display that can be programmed by the user to activate a specific action, such as speed dialing a particular phone number.

Softphone

VoIP software that runs on your desktop, laptop or handheld computer and provides an onscreen telephone interface to allow you to make phone calls through your computer using its speakers or headset and microphone without a traditional telephone handset.

Synchronous communications

Transmission method whereby a fixed frequency synchronizing clock signal is used to synchronize data sent between a sending and receiving device.

TAPI

Telephony Application Programming Interface; programming interface for allowing Windows client applications to communicate with server-based voice communications services.

Telephony gateway

The network device by which analog signals on telephone circuits are converted to digital data packets to enable calls between VoIP phone lines and standard PSTN phone lines.

Voice messaging

Application whereby voice messages are recorded, stored, and retrieved for later playback. A private access code is usually required for remote retrieval. Some systems can notify the recipient of the message via pager, outdialing, or e-mail.

VoIP session

A connection between two computers or VoIP phones using the same protocols and sending data across two channels, one for transmission of packets in each direction.

VoIP

Voice over Internet Protocol; technology for transmitting voice calls over a TCP/IP packet switching network such as the Internet, thereby avoiding long distance charges associated with the traditional public switched telephone network.

Open source VOIP Client - SIP client

Voip Surf — Wed, 18 Mar 2009 11:26:00 GMT

SIP Clients (User Agents)

SIP (Session Initiation Protocol) is currently described by the rfc2543 SIP is a popular open standard replacement from IETF (Internet Engineering TasForce) for H.323 signaling standard for managing multimedia session initiation. SIP can be used to initiate voice, video and multimedia sessions, for both interactive applications (e.g. an IP phone call or a videoconference) and not interactive ones (e.g. a Video Streaming). It is the more promising candidate as call setup signaling for the present day and future IP based telephony services, as it has been also proposed for session initiation related uses, such as for messaging, gaming, etc.SIP needs two ports, one for the command exchange and one for the RTP stream which contains the voice. It’s easier to work with firewalls than H.323, but you still need to have a proxy running. The following SIP UAs are divided into two groups for Multiplatform and Linux only:

Multi-Platform

SFLphone - A nifty little default skin (Metal Gear) for SFLphone holds a multi-protocol (SIP/IAX) multi-GUI desktop VoIP phone for use in Desktop environments. The project is being developed on Linux, but should (”and must”) be portable to various flavors of BSD operating systems (and maybe win32) with some involvement.
Linphone - With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone is stable under Linux, but FreeBSD and OpenBSD are reported to work.
Minisip - Minisip was developed by Ph.D and Master students at the Royal Institute of Technology (KTH, Stockholm, Sweden). It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network. Runs on multiple Operating Systems (Linux PC, Linux familiar IPAQ PDA, Windows XP and soon Windows Mobile 2003 SE).
OpenWengo - The flagship product of the OpenWengo project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place. Through their partnership with Wengo, they also offer very cheap PC to telephone and SMS rates. Available for Linux, MacOSX, and Windows.
PhoneGaim - Make phone calls to your friends and family directly from your Linspire computer with the latest software from Linspire. PhoneGaim is built right into Gaim.
sipXtapi - sipXtapi is a comprehensive client library and software development kit (SDK) for SIP-based user agents. It includes SIP signaling support as well as a media framework. A complete and very feature rich softphone can be built easily by adding a graphical user interface on top of sipXtapi. Alternatively, sipXtapi was engineered to be embedded into existing applications adding real-time communications to such applications. sipXtapi is primarily developed under WIN32; however, sipXtapi can be built and used under Linux and MacOs X. WinCE support is in development.
OpenZoep - OpenZoep (pronounced “open soup”), developed by Voipster, is a client-side telephony and instant messaging (IM) communications engine. It supports computer-to-computer (peer-to-peer) VoIP calls, instant messaging, and outbound PSTN and SIP calls to free and premium SIP providers.

Linux

Cockatoo - Cockatoo is a project that focuses on implementing SIP/SIMPLE as an extension for Thunderbird (XPCOM component/XUL interface) that enables users to phone contacts from an address book and see their presence state. Functionalities are included into Thunderbird as an XPCOM component.
YeaPhone - The goal of the YeaPhone project is to bring VoIP-Software together with the Yealink USB handset (USB-P1K) and at the same time make a PC keyboard and monitor unnecessary. This makes YeaPhone ideal for “Embedded Devices” as these do typically need extra devices for user interaction (in this case the handset) while working very energy efficient.
Twinkle - Twinkle is a soft phone for your voice over IP communications using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls.

Windows

1videoConference - 1VideoConference allows its Web, Audio/ Video phone, Skype, Msn and Yahoo users to instantly participate in live web conferences without the need for lengthy downloads or complicated installations. Simply drop a small piece of code onto your website and instantly create an online video conference room. All you need is a web cam and an internet connection and seconds later you can show presentations, share applications or users’ desktops, hold live webinars, discuss new strategies face to face with business partners, and more…

SIP Proxies

Open Source SIP - Open Source SIP was created in March 2006 as a project to foster the development of commercially viable SIP applications. The Open Source SIP project is sponsored by Solegy, and draws on over six years of research and development.
Partysip - Partysip is a modular application where some capabilities are added and removed through GPL plugins. Depending on the list of included plugins, partysip can be used as a SIP registrar, a SIP redirect server or statefull server, or a SIP service provider (game session, answering machine, etc.).
MjSip - MjSip is a complete java-based implementation of a SIP stack that provides API and implementation bound together into one package. The MjSip stack has been used in research activities by Dpt. of Information Engineering at University of Parma and by DIE - University of Roma “Tor Vergata”. MjSip includes all classes and methods for creating SIP-based applications.
OpenSER - OpenSER is an open source GPL project that aims to develop a robust and scalable SIP server. Spawned from FhG FOKUS SIP Express Router (SER) by two core developers and one main contributor of SER, OpenSER promotes a development strategy open for contributions.
SIP Express Router - SIP Express Router (ser) is a high-performance, configurable, free SIP server. It can act as registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.
Siproxd - Siprox is an proxy/masquerading daemon for the SIP protocol that handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections possible via an masquerading firewall. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router.

Open Source VOIP agent - H.323

Voip Surf — Wed, 18 Mar 2009 11:21:00 GMT

H.323 Clients (User Agents)

VoIP traditionally uses H.323, a rather complicated protocol that uses multiple ports and a binary code for data. But apps like FreeSWITCH make H.323 seem like a piece of cake with its all-in-one application. The following H.323 clients are broken down into Multiplatform, Linux, MacOS X, and Windows.

Multiplatform

FreeSWITCH - FreeSWITCH is a telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple switching engine, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow. FreeSWITCH runs on several operating systems including Windows, Max OS X, Linux, BSD, and Solaris on both 32- and 64- bit platforms. Note: FreeSWITCH is also multiprotocol, as it works with SIP, IAX2 and GoogleTalk to make it easy to interface with other open source PBX systems.
YATE - Yate (Yet Another Telephony Engine) is a next-generation telephony engine that is the first open source telephony application capable of handling 600 H323 calls; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate’s flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. YATE can be used for anything from a VoIP server to an IVR engine. The software is written in C++ and it supports scripting in various programming languages (such as those supported by the currently implemented embedded PHP, Python and Perl interpreters) and even any Unix shell. Note: YATE is multiprotocol, as it works with SIP and IAX, and H.323 protocol is stable supported just by Yate. The most used application of Yate is as a SIP-H323 translator because is the only open source stable translator.

Linux

Ekiga - Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Note: Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting.

MacOS X

XMeeting - XMeeting is the first H.323 compatible video conferencing client for Mac OS X.

Windows

OpenH323 Project - The OpenH323 project aims to create a full featured, interoperable implementation of the ITU-T H.323 teleconferencing protocol that can be used by personal developers and by commercial users without charge.

H.323 Gatekeeper

OpenH323 Gatekeeper - The GNU Gatekeeper (GnuGk) is a full featured cross-platform H.323 gatekeeper, available freely under GPL license.

H.232 Radius Platform

BSDRadius - While there are quite large number of Radius servers (including free) available in the world, little number of them (if any) are developed with VoIP specific needs in mind. BSDRadius is a RADIUS - compliant AAA (Authentication, Authorization, Accounting) server with CHAP-password authentication for H.323. Platform-independent, but has not been tested on Windows.

Introduction - Voip how to

Voip Surf — Wed, 18 Mar 2009 11:13:00 GMT

How Does VoIP Work?

It is very easy to get into a discussion that is very technical and confusing to most readers. The purpose of this section will be to provide a very high-level overview of Voice over IP (▲VoIP) aimed at those who do not consider themselves experts in the subject and hopefully with enough clarity that it serves as a good introduction to most readers.

Many people have used a computer and a microphone to record a human voice or other sounds. The process involves sampling the sound that is heard by the computer at a very high rate (at least 8,000 times per second or more) and storing those "samples" in memory or in a file on the computer. Each sample of sound is just a very tiny bit of the person's voice or other sound recorded by the computer. The computer has the wherewithal to take all of those samples and play them, so that the listener can hear what was recorded.

VoIP is based on the same idea, but the difference is that the audio samples are not stored locally. Instead, they are sent over the IP network to another computer and played there.

Of course, there is much more required in order to make VoIP work. When recording the sound samples, the computer might compress those sounds so that they require less space and will certainly record only a limited frequency range. There are a number of ways to compress audio, the algorithm for which is referred to as a "compressor/de-compressor", or simply ▲CODEC. Many CODECs exist for a variety of applications (e.g., movies and sound recordings) and, for VoIP, the CODECs are optimized for compressing voice, which significantly reduce the bandwidth used compared to an uncompressed audio stream. Speech CODECs are optimized to improve spoken words at the expense of sounds outside the frequency range of human speech. Recorded music and other sounds do not generally sound very good when passed through a speech CODEC, but that is perfectly OK for the task at hand.

Once the sound is recorded by the computer and compressed into very small samples, the samples are collected together into larger chunks and placed into data packets for transmission over the IP network. This process is referred to packetization. Generally, a single IP packet will contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being most common.

Vint Cerf, who is often called the Father of the Internet, once explained packets in a way that is very easy to understand. Paraphrasing his description, he suggested to think of a packet as a postcards sent via postal mail. A postcard contains just a limited amount of information. To deliver a very long message, one must send a lot of postcards. Of course, the post office might lose one or more postcards. One also has to assemble the received postcards in order, so some kind of mechanism must be used to properly order to postcards, such as placing a sequence number on the bottom right corner. One can think of data packets in an IP network as postcards.

Just like postcards sent via the postal system, some IP data packets get lost and the CODECs must compensate for lost packets by "filling in the gaps" with audio that is acceptable to the human ear. This process is referred to as ▲packet-loss concealment (PLC). In some cases, packets are sent multiple times in order to overcome packet loss. This method is called, appropriately enough, redundancy. Another method to address packet loss, known as forward-error correction (FEC), is to include some information from previously transmitted packets in subsequent packets. By performing mathematical operations in a particular FEC scheme, it is possible to reconstruct a lost packet from information bits in neighboring packets.

Packets are also sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable, as the same PLC algorithms can smooth the audio to provide good audio quality.

Computers generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay (called jitter) is the most frustrating for IP devices. Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices must implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce "mouth to ear" delay. Such "adaptive jitter buffer" schemes are also used by CD recorders and other types of devices that deal with variable delay.

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to ▲video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets. We will describe some of the popular VoIP protocols in the next section.

Through this section, we have focused on computers that communicate with each other. However, VoIP is certainly not limited to desktop computers. VoIP is implemented in a variety of hardware devices, including IP phones, ▲analog terminal adapters (ATAs), and ▲gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks: one does not have to throw out existing equipment to migrate to VoIP.

Voip surf

Voip Surf — Tue, 10 Feb 2009 01:49:00 GMT

This blog about latest news of VOIP.

Welcome

Voip Surf — Mon, 09 Feb 2009 07:42:06 GMT

Welcome to my blog. Please check back soon for new entries.