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Telerupted: Twilight for Telephone Networks

Source: gigaom.com

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Session initiation protocol-compatible VoIP devices already account for as much as 20 percent of landline telephone traffic, thanks to the efforts of companies like Cisco, which sells to enterprises, and Comcast (in the U.S.) or Free (in France), which target consumers. Mobile telephones will not remain a safe haven for long, however, as more companies like Fring and Truphone start to offer VoIP alternatives to operator voice plans.

Such plans involve downloaded SIP User Agent software that can also voice-enable gadgets like the Nintendo DS, Sony PSP or iPod Touch. Dan Borislow claims the marketing blitz for his SIP-based magicJack puts him on track to sell 500,000 of the devices by the end of this month. Yet the displacement of analog phones by VoIP devices has not displaced the telephone network itself.

The state of affairs is analogous to printing email before it reaches the destination in order to preserve a role for the post office. It will not last. The ubiquitous use of SIP makes it possible to configure VoIP traffic to peer directly via the Internet, but the business models of VoIP companies depend on the minutes-based charging enabled by the telephone network. Companies like Skype and Vonage, as well as the Web 2.0 voice plays Jajah or Jaxtr, provide for free calls between users, but they all generate revenue by sending off-Net calls to the telephone network.

The 20 percent VoIP penetration number implies that both called and calling parties have VoIP devices about 4 percent of the time. This leaves plenty of work for the telephone network, but the long-term utility of passing VoIP traffic through the telephone network does not look promising. At some point, the penetration of VoIP devices will cross a threshold that makes the minutes-based business model of telco and VoIP players alike untenable.

The strategy of using VoIP to make the telephone network more efficient has short-term merit in that it avoids the expensive process of changing end user behavior. The usual mode of price competition serves as a reasonable placeholder until VoIP devices get sufficiently mature and achieve critical mass. A regulatory environment that requires VoIP players to implement traditional functionality associated with E911 and CALEA contribute to preserving the status quo. In the meantime, patent litigation remains a drag on investment capital, and the availability of suitable Internet connectivity remains particularly weak in the mobile context — Internet access in any form remains an obstacle in many locations around the world. Reliability gives the telephone network an edge over the Internet for sales calls and other high-value communication.

Telephone network integration may provide a useful transition strategy en route to VoIP nirvana, or it may represent a unrecoverable dead end for the current crop of VoIP startups. Companies that depend on the telephone network inherit of a range of artificial constraints. VoIP devices connected via the telephone network lose the prospect of delivering high-quality audio. Traditional telephones do not support the use of domain names for routing or hyperlinking. Global flat-rate termination that serves as a driving force for applications of the Internet get sacrificed. Embracing the telephone network postpones the search for new forms of communication.

The dependence of VoIP plays on the telephone network does not erase the risks VoIP poses to the telco status quo. At this point it’s not even clear if there exists a role in a VoIP infocom ecosystem for traditional service providers. The metered usage charges that make the telephone network attractive from a business model perspective do not exist in an Internet context. Infocom seems likely to mirror the existing infotech bring-your-own-device ecosystem with hardware vendors, software companies and access providers. It will take at least another decade for the forces at work to play out, but this provides little consolation for an industry that traces its roots to 1876 and telco executives unable to retire before the music stops.

Published on June 25th, 2008 under , ,

Sipera Gets Upgraded SIP Security

Source: www.voip-news.com

Sipera System’s Sipera IPCS security appliances  now have advance security for SIP trunking. Sipera VIPER Engine also has upgraded security as well.

“Many enterprises today are embracing Unified Communications because they see it playing a key role in increasing the productivity of their organization. What some overlook, however, are the security issues that arise any time an enterprise application is connected to the Internet,” said Matthias Machowinski, Infonetics Research Directing Analyst, Enterprise Voice & Data. “In order to realize the benefits of UC without increasing security risks, enterprises need to add security to their infrastructure that protects against threats in real-time.”

Sipera execs agree.

“As companies extend Unified Communications beyond the enterprise perimeter to allow SIP trunking and mobility solutions, they require sophisticated and comprehensive security from a dedicated UC security provider,” said Eric Winsborrow, Chief Marketing Officer for Sipera Systems. “Sipera’s comprehensive UC security provides threat protection, policy enforcement, access control, and privacy measures, along with the ability to simplify the deployment of SIP trunks and mobile workspaces. Sipera IPCS threat protection is backed by the expertise of and ongoing signature updates from Sipera VIPER Lab to ensure complete protection in real-time.”

Published on June 24th, 2008 under , , , , , , , ,

Siphera’s New Partner Network Unveiled

Source: www.voip-news.com

Sipera System has unveiled Sipera Partner Network, which provides distributors, resellers and other partners to add VoIP and UC security to its implementations.

“As more enterprises deploy VoIP and UC, we anticipate an increase in both security awareness and emerging threats. With access to Sipera products and VIPER Lab, we can ensure our customers stay ahead of vulnerabilities inherent in these networks, and proactively safeguard critical business communications,” said Jeff Graham, CTO at Shared Technologies. “By providing this security as part of a UC deployment, we can better support our customers and generate new revenues in the process.”

Siphera’s UC security solutions work with major VoIP infrastructure from companies including Avaya, Cisco and Nortel.

“We’re seeing a lot of customer interest in securing VoIP networks and extending them outside the enterprise by deploying unified communications applications,” said Bruce Steele, VP at Paranet. “But a lot of enterprises are trusting the equipment vendors to provide UC security, and don’t realize they can deploy purpose built, real time UC security. The Sipera IPCS product family is ideal for helping our customers protect their brand, networks, applications and users.”

The partners are provided with specialized training as well, so they understand the technology.

“These partnerships are critical to customers’ networks so we developed a partner program that focused on their VoIP/UC needs beyond deployments and normal operations,” said Eric Winsborrow, Chief Marketing Officer for Sipera. “We expect partners’ customers will embrace our products along with partners’ support, and that new partners will likewise welcome adding this expertise to their portfolios.”

Published on June 12th, 2008 under , , ,

Skype To SIP is Already Here

Source: andyabramson.blogs.com

Over the weekend James Body of Truphone mentioned hearing something about a Skype To SIP gateway platform with Voxeo. Now PhoneBoy tips off the world.

All I know is the calls to the PSTN of late have been excellent from Europe.

Published on April 22nd, 2008 under , , , , ,

AT&T / Starbucks / Apple Combo Video - Log On. Sip. Buy.

Source: alanweinkrantz.typepad.com

This video illustrates the process of logging on to the AT&T WiFi network at Starbucks, logging on the browser, going to iTunes and seeing the slightly different user experience of buying iTunes music in a Starbucks, as well as seeing what is currently playing and having the option to buy on the spot if you like the song you are hearing.

In simple terms, AT&T does the infrastructure; Starbucks does the physical plant; Apple sells the content.

Media:  OK to call me over the weekend:

Media can call me this weekend on my cell at 210-410-3075. 

Please, no calls on Saturday or Sunday after 7:00 PM CST because of Passover. 

All day Saturday and Sunday is cool, but up until 7:00 PM.

Email is alan at weinkrantz dot com.

Published on April 18th, 2008 under , , , , , , , ,

Wednesday Links: Fring, Blackhat Europe and Sipcall

Source: www.voip-news.com

Voice Over IP Weblog reports that Fring and Boingo are teaming up. But why? What’s the benefit? Read it here.

Sipcall’s new Hipsip lets you make free VoIP calls from any mobile phone. Pretty nifty. Read about it on the UK’s VoIP-News.

SIPVicious is doing some more blogging from Blackhat Europe. Read it here.

Published on April 3rd, 2008 under , , , , , , ,

A free bridge from Skype to phone

Source: goebel.net

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Do you remember my blog post "A SIP address for Skype? Better the other way around!"? This mission has now been accomplished. As of yesterday you can call me on Skype and I will answer this call on my desk phone or cell phone using SIP VoIP telephony. As I always try to achieve, this is a totally free solution.

I have joined Voxeo’s developer program for their Evolution application, a visual design tool for interactive voice response (IVR) systems. Part of the deal is that you get a strange phone number with a +990 country code. There is no country associated with this code and Skype users can call these numbers for free. My Skype account is now being forwarded over Voxeo to a SIP address from Gizmo Project which I manage on Voxalot to make use of it’s call connection rules and voice mail.

Have a peek on my settings:

A better explanation can be found at the Voxeo support forum. I wonder what VOIPSA’s Dan York would say. In January he started a discussion with his blog post "Skype says "No" to VoIP interoperability - *because customers aren’t asking for it!* - Well, I am!". He is, by the way, working for Voxeo and this partly solution for his problem comes from his own company. So I guess he was always aware of this trick.

I am happy now that people can call me with Skype and I don’t have to keep me computer running or buy a special Skype phone for this purpose. That’s the reason why I nearly never used Skype. I don’t like applications which keep me tied to my computer in order to receive messages or phone calls, like Skype or the MagicJack normally do. Let’s see which other solutions I can develop with Voxeo. Their visual tool makes the design of VoiceXML fairly easy.

Published on March 30th, 2008 under , , , , , , , , ,

What Is SkySipTel?

Source: www.voip-news.com

So I saw a semi-coherant item on TMCnet about SkySipTel and was curious . . . what is this thing? The item - for lack of a better word - seemed to be somewhere between press release and ESL practice essay.

I did what any good reporter would do: Searched SkySipTel.

Apparently it’s a VoIP provider that offers softphone PC-to-phone calling for cheap rates (it seems) as well as other services like call forwarding and voicemail. And they have at least one misspelled word on the website. Hello, pet peeve.

Oh and they have a new softphone update available too . . . Click here for that.

Published on March 21st, 2008 under , , , , ,

Sipera Systems Growing VoIP/UC Security Business

Source: www.voip-news.com

Sipera Systems is hoping their recent appointment of John Lochow as president and CEO is going to do for that company what it did for his previous company, Syndesis Limited. Under Lochow’s leadership, Syndesis revenues grew tenfold from $6 million to $60 million. Lochow has experience in both small and large enterprise  and service provider technology solutions companies.

Sipera is growing and expanding its VoIP/UC security solutions business, which is crucial to the security of enterprise VoIP.

“With the growing awareness in the marketplace of the need for comprehensive VoIP/UC security, now is the time for Sipera to bring its VoIP security leadership to a new level.  John is the person to help Sipera accelerate its momentum,” said Ben Scott, chairman of the board for Sipera Systems.

Lochow said he’s excited to join the organization.

“Sipera is at an inflection point with the demand for comprehensive VoIP/UC security solutions expanding as enterprises and service providers seek to protect and control real-time unified communications,” said Lochow.

Published on February 25th, 2008 under , , , , , , , , , , ,

Tpad has cleaned out dormant accounts although they were in use

Source: goebel.net

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One of the most reliable VoIP services I know is Tpad. Not only that it worked flawlessly for more than one year, they even credited $10 to my account when I found an error this weekend. Needless to say that Tpad never got any money from me penny pincher, because I use their service only to receive calls.

Long before Jajah Direct, Wifimobile or Gizmocall started similar services, Tpad already had break-in numbers in 39 countries. It’s an entire callthrough system: You can dial whichever of these 79 numbers and the number of my Tpad account to reach me for the price of a local call. That’s much more reliable than the other services, which depend on the Caller ID to connect the call. In poor countries with bad networks this Caller ID often cannot be transmitted for technical glitches. I am permanently connected to Tpad with my SIP ATA so that my Peruvian friends in Lima can always call me for the price of a local call.

Today it’s more than one year that I started to write about Tpad and I have used it since then. But some days ago I realized that my SIP devices could not connect to the Tpad server anymore. Not from my ATA, not from Voxalot, not from a Nokia E61, not from a Nokia N810. Other German friends had the same problem. What was wrong? I asked in their forum and learned that Tpad had cancelled my account because they thought I didn’t use it anymore:

Tpad performed a cleanup of "dormant" accounts, without remembering that call records are only captured for calls that use the Tpad softphone. Since you use Tpad exclusively from an ATA or non-Tpad softphone, your call activity is not remembered. So, it is very likely that your account was improperly considered dormant and was suspended. Tpad should be able to restore it for you pretty quickly.

What really impressed me was that the forum admin immediately wrote "Send me a PM of your Tpad Number(s) and we will fix asap". What a difference to other VoIP services! His answer, apology and $10 to my account arrived the same Saturday. On Sunday they fixed the problem. What a great service!

I think I should charge some money to my Tpad account as a gesture of gratefulness. If only it was necessary! With $10 I can call for more than ten hours to Germany and this credit never expires. That’s another big difference of Tpad to other VoIP companies.

Published on February 25th, 2008 under , , , , , , , , ,

Wide Open VoIP: Top 50 Open Source VoIP Apps

Source: www.virtualhosting.com

For many businesses, open source VoIP programs and apps offer a great way to save thousands of dollars every year in telephony costs. Better yet, open source programs are fully customizable to a business’ specific needs, making them a popular solution that often just can’t be beat. This popularity hasn’t just helped business, but has also driven many open source programs to the forefront of their industry. In fact, it has been speculated that open source VoIP solutions could surpass the popularity of the ubiquitous desktop solution Linux. Here are a few of the open source programs and developers out there that have had loads of success as VoIP and open source solutions for it become more and more common in businesses around the world.

SIP Proxies

SIP Proxies enable service providers to build scalable and reliable VoIP networks that are based on the Session Initiation Protocol. This allows a full array of call routing capabilities that make the most of network performance. Here are some of the most popular and successful SIP proxies on the market.

  1. OpenSer: OpenSER has been described as a “mature and flexible” SIP server so it’s no surprise that it’s popular among users. OpenSER development began with FhG FOKUS SIP Express Router, but then diverged into its own feature-laden software package that was released in 2005. Since then it’s been exhibited around the world, and makes a great addition to Linux systems looking to employ VoIP technology.
  2. VOCAL: Open source VoIP developers can benefit from the software and tools found in VOCAL. Developed through the Cisco sponsored labs at Vovida, VOCAL is fully customizable to business needs and can provide call routing, billing information, call control and more in an easy to control and maintain Linux based system. It’s been successful due largely in part to its immense capability for adaptation and scalability, and likely will only see further integration into business systems in the future.
  3. partySIP: Developed back when VoIP was just starting to take off, partySIP can still be a relevant solution for businesses looking for VoIP today. This lies largely in the modular construction of partySIP, which relies on various plugins to add or remove capabilities. This flexibility allows its users to disable useless functions and enable new ones with very little development, making it easy to use and customize, which is likely the reason for users’ continued interest in the product.
  4. SIP Express Router: This high performance SIP product can act as registrar, proxy or redirect server depending on your needs. It’s been widely successful in the VoIP market due to its ability to deal easily with operational problems like broken network components. Another reason it’s loved is its scalability from small office environments to acting as a PBX replacement and can in many cases act as a replacement for the very popular Asterisk system.
  5. MjServer: One of the things that makes MjServer so important to the VoIP market is that it works on a variety of platforms, not just Linux, so those who aren’t quite ready to take the fully fledged open source route can ease into it. MjServer is a Java based application that is easily configurable and can act as a registrar, redirect or proxy in your VoIP setup, making it a versatile and useful tool for implementation.
  6. OpenSBC: OpenSBC has been in use for over 7 years in both low and high volume applications. In this way, it’s a very reliable system, but also still employs a great deal of possibility for expansion and modification based on personal needs for the program. In fact, like most open source VoIP applications, the developers actively encourage the changing and development of the program to make it better for all users.
  7. sipX: Developed by SIPFoundry, sipX is designed to be an incredibly feature rich and standards compliant infrastructure for businesses who want to employ VoIP technology. It is, in fact, one of the most widely used and well respected open source developments out there and feature wise is very similar to Asterisk.

SIP Clients

Session Initiation Protocol is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging, and is fast becoming one of the more popular protocols for VoIP in businesses and homes alike. Here are a few programs that have helped bring SIP to the forefront of the market.

  1. Linphone: Linphone is promoted as a solution to help users communicate more freely over the Internet using voice, video and text messaging. Recent updates to the program have made it even better, solving many compilation issues while adding improved interoperability and new features. While currently only stable on Linux systems, development is under way for a Windows version as well.
  2. PhoneGaim: If you haven’t heard of PhoneGaim you’ve likely heard of its proprietary counterpart Gizmo Project. While it doesn’t have the instant name recognition of its VoIP cousin Gizmo, PhoneGaim is still a product to take note of. Developed in an attempt to challenge Skype, the program is loaded with integrated features that help make the VoIP experience rewarding, even for those just using the software at home.
  3. OpenWengo: Started and developed by the French company, Wengo, OpenWengo is a great, and popular, open source choice for anyone looking for simple and easy-to-use VoIP software. This softphone program allows users to call between computers and phones, and has additional instant messaging and contact management capabilities. The recent development of a Firefox plugin that allows users to make calls quickly and simply from their browsers is just one example of the continued innovation and popularity of this multi-featured program.
  4. Cockatoo: Users of Thunderbird have Cockatoo to thank for simple VoIP integration with their email. The program allows users to make a call simply by clicking on entries in their address book. It’s simplicity and aim to make VoIP more fully integrated into business systems has made it a popular addition to business and personal computers.
  5. Minisip: Minsip is an Internet based phone that can be used to make phone calls, instant message and video call to anyone connected to the same SIP network. Developed by PhD and masters students at Royal Institute of Technology in Stockholm, Minisip is a simple by highly functional VoIP phone. Users can even make calls from PDAs or pocket PCs running Windows or Linux, making VoIP on the road easy and cheap.
  6. OpenZoep: Developed by Voipster, OpenZoep is a popular client-side VoIP choice, providing the ability to both make calls and send and receive instant messages. Since its release, developers have continually added new features, especially from users in Europe, where the product was first developed. Continued changes and a responsive market have made OpenZoep a popular solution both here and abroad.
  7. Shtoom: Shtoom is a open-source, cross-platform VoIP softphone, implemented in Python which also includes an application called doug which can be used to write and modify VoIP applications. This built-in framework for modification encourages customization, one of the reasons open source software is so popular.
  8. Twinkle: Linux users have embraced the softphone Twinkle for making VoIP calls through an SIP protocol. Twinkle is a great solution for many users as it provides many, if not more, of the features found in regular telephony including custom ring tones, voice mail, conference calling, and multiple lines. These features, in addition to its open source usability, make Twinkle a popular choice among Linux users.
  9. YeaPhone: YeaPhone is unique among open source VoIP systems in that it hopes to take the computer monitor and keyboard completely out of the picture when making VoIP calls, opting instead to use the Yealink USB headset. This makes it more similar to many commercially available phone systems, and a popular choice among users searching for an open source alternative to those systems.

H.323 Clients

H.323 is the traditional protocol for most VoIP systems which has been continually refined with new elements to help improve voice and video quality. These popular VoIP clients make the most of what H.323 is capable of.

  1. YATE: The YATE system relies on its ability to adapt to the conditions in which it’s being used. A flexible routing engine allows communications to be made efficiently and cheaply, both often big concerns to businesses when choosing VoIP platforms. It’s easily combined and expanded with other services making it an incredible versatile and successful tool in the VoIP market.
  2. FreeSWITCH: FreeSWITCH is “an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch.” This ability to work both as a soft phone and a soft switch depending on the needs of the users makes it an attractive and intriguing option for many searching for VoIP technology. It’s even been touted as a viable alternative to using Asterisk, as many plugins and modules are available that don’t require reworking the main program code.
  3. Eikga: Formerly known as GnomeMeeting, Ekiga is an open source VoIP and video conferencing program that was developed for the Linux GNOME platform. It has a relatively simple interface, but gains one major advantage with users in that it works with both the H.323 protocol as well as with SIP, giving it double the functionality for users.
  4. OpenPhone: The original goal of OpenPhone was to enable every computer on the planet with phone capabilities. While this lofty goal may still be a ways off from completion, the OpenPhone software is still making strides in that direction. With an active development community, OpenPhone is a great place to find open source inspiration and functionality.
  5. XMeeting: Mac users need not despair, there are great open source alternatives for you as well, like XMeeting. XMeeting is the first H.323 compatible video conferencing client for Mac OS X, and not only supports H.323 but SIP as well. This functional versatility has made it a great solution for businesses primarily using Macs but also wanting to make the most of VoIP technology.

IAX Clients

IAX stands for inter-Asterisk exchange and programs using this protocol are used to enable VoIP connections between servers as well as to facilitate client-server communication. Here are a few of the most popular of these applications.

  1. IAXComm: IAXComm is a multi-platform softphone that works with Asterisk to allow users to place and receive VoIP calls. With features like music on hold and speakerphone, it is a great addition to an Asterisk system for businesses looking for VoIP technology.
  2. Kiax: Kiax relies on the IAX protocol to help it make it friendlier to users behind a NAT, or a router system that rewrites the source and/or destination IP addresses of IP packets as they pass through. Kiax maintains a simple interface that allows users to make calls to an Asterisk server quickly and easily, helping speed the spread of VoIP implementation both in homes and businesses.
  3. YakaPhone: YakaPhone is a simple and skinnable softphone. It is often a great solution for businesses looking for VoIP technology that is relatively simple but also easy to customize and use for day to day business. Businesses can even customize the phone skins to reflect company logos and branding, making it a more personalized experience.
  4. SFLPhone: For those with busy or especially large offices, SFLPhone is often one of the best IAX options as it was designed to handle high loads of daily phone calls. New partnerships should take it even further into the enterprise sector, as it has been announced that video conferencing is in the works.

PBX and IVR Platforms

PBX, or private branch exchange refers to a the telephone operating systems of a business or office, rather than those established for public use. Part of those systems might include Interactive Voice Response, which allows the computer to detect voice and touch tones to route phone calls to the appropriate menus or locations. These VoIP programs have taken the lead in those technologies.

  1. Asterisk: Asterisk is perhaps the greatest open source VoIP success story of them all. It is the leading open source telephony engine and tool kit and is used in thousands of servers and VoIP setups all over the world. What makes it so great? The standard system supports many features available in proprietary PBX system like voice mail, conference calling, interactive voice response, and automatic call distribution but also has the versatility to be adapted and personalized based on business or individual needs.
  2. OpenPBX: Developed by Australian company Voicetronix, OpenPBX is a popular solution both with small offices and with large call centers. With features like unlimited voicemail, auto-attendant, automatic call distribution, music on hold and call parking it’s easy to see why. It also has the advantage of highly compact Perl code, meaning it’s very easy to customize and extend.
  3. GNU Bayonne: An integral part of GNU telephony, Bayonne offers users technology that is not only free but scalable and customizable as well. Working with the complete suite of GNU enterprise solutions, Bayonne can be an easy way for users to integrate with telephony and provides a great VoIP solution for many.
  4. CT Server: CT Server is based on the ccscript language developed by David Sugar for the IVR server Bayonne as well as Perl for other tasks like database lookup. CT Server has been great resource for developers looking for framework for customizing or creating their own PBX quickly and creatively.
  5. sipX PBX: One of the main competitors to Asterisk, the sipX PBX and Asterisk are often compared. In contrast with Asterisk’s complete open source approach, sipX has a bit more of a commercial flair, as additional support and plugins can be purchased from the developers website. But sipX, once installed on your system, can provide much of the same functionality and in some cases might even be easier to use.
  6. Trixbox: Fast becoming one of the most popular Asterisk based PBX phone systems, Trixbox has seen loads of success in the past few years from businesses and enterprises searching for a VoIP solution. Designed for businesses with anywhere from 2 to 500 employees, the product comes in a few different versions.
  7. Evolution PBX: Evolution is another, more commercial application based on the open source server Asterisk. Basic editions of the software are free, however, and can be downloaded from the developers site. Evolution has been instrumental in helping solve one of the major obstacles to many businesses implementing VoIP as it makes integrating existing phone systems with newer VoIP systems easier, making the change much more cost effective for businesses, a key selling point for any VoIP product.
  8. CallWeaver: Originally derived from Asterisk, CallWeaver works on many different platforms and with new versions being released regularly it has a growing list of features. CallWeaver was developed as an alternative form of Asterisk that encourages community involvement and employs multiple vendors who drive the project rather than just one working for a single interest. This open-minded approach to open source VoIP has won the program many fans who believe that it’s already better than other versions of Asterisk.

Stacks and Libraries

Stacks and libraries are an integral part of what makes open source such powerful technology. Using these resources, businesses or individuals can develop and refine VoIP systems that work best for their business. These are just a few of these such resources that have had a big impact on VoIP development.

  1. OpenSIPStack: OpenSIPStack provides developers with a platform agnostic stack implementation of RFC 3261 so that development can be done in Linux, Solaris, BSD, Darwin and Windows. This versatility has made it an ideal choice for developers wanting to work in a variety of platforms.
  2. The GNU oSIP Library: Developers wanting to work with SIP have found just about everything they need in this library. Described as having the aim to “provide multimedia and telecom software developers an easy and powerful interface to initiate and control SIP based sessions in their applications” the GNU oSIP Library can do just that as it includes not only a library but examples of programs that use the oSIP protocol to operate.
  3. Twisted: Twisted comes from Twisted Matrix Laboratories and is an “event driven networking engine written in Python.” It supports a variety of protocols ((including HTTP, NNTP, IMAP, SSH, IRC, and FTP) and also has support for SIP, making it ideal for VoIP development.
  4. Verona: The Verona Project is an open source VoIP toolkit based on a phone API called Phapi and a minimal user agent called aptly miniua. It is similar to the toolset used in the highly successful OpenWengo software but is modified to reduce dependence on certain libraries, allowing users reliable building blocks for constructing their own VoIP programs.
  5. PJSIP: Written in C, PJSIP is an open source protocol stack for SIP. Due to its small footprint, high portability, customizability, and loads of other features its become a popular choice among SIP developers.
  6. eXosip: The eXosip library is a common choice among those who want to take the complexity of using the SIP protocol for multimedia session establishment down a notch. eXosip hides it, and makes implementing SIP easier whether you’re using it for VoIP or for something like multiplayer gaming.
  7. Vovida SIP: Vovida is a hugely popular place to get VoIP software both to use as is and like this protocol stack, to be used more commonly in further development of VoIP programs. This SIP stack is popular with Linux based developers wanting to embrace this protocol.
  8. reSIProcate: Part of SIPFoundry, reSIProcate works in a variety of operating systems including Unix, Windows, Mac OS X and more. The application is well suited and widely used in companies wishing to implement phones, softphones, gateways, proxies, or instant messaging.

Developers

While anyone is able to edit and create parts of open source software, the original programming has to come from somewhere. These are a few developers that have had great success in creating and releasing many of the most popular and widely used VoIP technologies in the open source field today.

  1. SIPFoundry: SIPFoundry is a not for profit open source community that aims to support the development and adoption of the SIP protocol. It’s also the home of much of the development of the sipX PBX for Linux, an award winning open source PBX program. The success of the sipX project as well as the increasing popularity of SIP have brought the SIPFoundry to the forefront of the VoIP community.
  2. Pingtel: Pingtel’s unique approach to the VoIP market may have a lot to do with their success. Using a system that runs using Linux and the sipX, Pingtel hopes to give business more control over how VoIP is built and used within their communications, something that proprietary software often can’t offer. The company also prides itself on providing reliable support and service for their products, making many business more willing to use them as there is less risk if something goes wrong.
  3. Vovida: Vovida is home to numerous SIP protocol stacks to help developers create and innovate new VoIP technologies and programs. Acquired in late 2000 by Cisco systems, this company’s work is well funded and its VOCAL tools and software have helped push VoIP development forward.
  4. Sangoma: Sangoma is a Canadian based company that develops both hardware and software based on the open source model, especially that having to do with telephony. While popular in North America, Sangoma is capitalizing on the hotbed of tech activity in Asia by forming a partnership with Vietnamese telephone distributor Dinh Quang. Their extension of open source VoIP software into new and widely used markets made them one of the most successful VoIP open source developers of 2007.
  5. Digium: With over a million downloads, Digium is one of the leading providers of Asterisk’s open source PBX software and has been the recipient of several awards for best open source software. With continued growth, and the acquisition of smaller VoIP players like Switchvox, Digium continues to add to its VoIP arsenal and likely will remain at the forefront of VoIP developers in years to come.

Miscellaneous

VoIP provides an opportunity for many different types of open source development to improve and refine systems. Here are a few miscellaneous programs that aren’t directly providing VoIP service, but are having an impact on the technologynonetheless.

  1. SIP Thor: SIP Thor is based on P2PSIP technology, and is built so that there is no single point of failure despite a large amount of scalability. With these features as well as quick disaster recovery and reliable service, those looking to start a VoIP reselling venture have found SIP Thor to be a great choice.
  2. MobiCents: MobiCents is billed as “the most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.” MobiCents facilitates the creation of new services, enabling the development of a market oriented and cost effective platform, all the while encouraging developers to continue coming up with new and better ideas.
  3. Aradial: For business needing a means to bill minutes used with VoIP Aradial provides a viable open source solution. It’s easy to use servers are highly scalable and feature a plug-in architecture for quicker upgrades. Its low cost and easy adaptability make profit margins higher for businesses reselling VoIP and make it a popular solution.
  4. Lintad: Lintad is helping to make VoIP more than just a voice technology. The program provides both fax and voicemail support for VoIP phone systems. While voicemail is nothing new, the fax capabilities are nice addition and one that makes switching to VoIP much less painful for businesses.
Published on February 20th, 2008 under , , , , , ,

IPTEGO, Radware Partner To Deliver Scalability, Performance And Security In SIP-Based VoIP Service Delivery

Source: snapvoip.blogspot.com

Radware and IPTEGO, integrated application delivery solutions provider and a voice over IP, VoIP service provider based on SIP protocol, respectively have formed a joint venture based on OEM agreement to provide carrier grade, high availability paired with scalability and security for SIP based VoIP services.

MAHWAH, New Jersey, February 6 /PRNewswire-FirstCall/ — Radware , the leading provider of integrated application delivery solutions for business-smart networking, and IPTEGO, a voice over Internet Protocol (VoIP) provider of Session Initiation Protocol (SIP)-based services and solutions, announced today that the two companies have entered into a strategic original equipment manufacturer (OEM) agreement to ensure carrier-grade high availability, scalability and security for SIP-based VoIP networks.

The Radware and IPTEGO partnership has resulted in the creation of a more complete and robust SIP-based solution. Using Radware’s SIP Director, a fully SIP-aware intelligent ADC solution as an integral part of IPTEGO’s SIP-based VoIP framework, enables IPTEGO to offer a more cost-effective carrier-grade solution that meets carrier requirements for scalability and greater interoperability integrating service availability, performance and security.

"This OEM agreement with IPTEGO is another proof point for the carrier market need for intelligent layer-7 application delivery controllers", said Yossi Vardi, Vice President Business Development Radware. "IPTEGO’s expertise in deploying SIP-based VoIP network solutions combined with Radware’s expertise of providing intelligent and innovative ADC solutions for both application and network layers, will provide a more complete SIP-based solution to fulfill customer needs."

"Radware represents a critical partnership for us, as they help differentiate our product and solutions portfolio through their robust, state-of-the-art SIP capabilities," said Alex Hoffmann, CEO, IPTEGO. "This yields high significance for us, as we set out to service for millions of users across our SIP-based VoIP framework. Furthermore, this partnership allows us to bring SIP-based services faster-to-market and with minimal development costs."

Radware’s and IPTEGO’s partnership will enable the companies to deliver a complete, scalable solution with guaranteed carrier-grade SIP service delivery to serve millions of users, which is critical when addressing future Web/VoIP collaboration trends. By integrating Radware’s SIP Director, IPTEGO will offer service providers enhanced service delivery.

Published on February 6th, 2008 under , , ,

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