Automatic Call Distributor

September 11, 2015, 6:38 am

Automatic Call Distributors

Automatic Call Distribution or ACD, is a tool commonly used in the telephony industry. ACD systems are commonly found in any office that handles a large volume of inbound calls. The primary purpose of an Automatic Call Distributor is to disperse incoming calls to contact center agents or employees with specific skills.

The ACD system utilizes a rule based routing strategy, based on a set of instructions that dictates how inbound calls are handled and directed. These rules are often simply based on guiding a caller to any agent as fast as possible, but commonly multiple variables are added, all with the end goal of finding out why the customer is calling. Matching and routing literally thousands of calls to the correct agent is a difficult task, and is often done in concert with Interactive Voice Response and Computer Telephony systems. ACD servers can cost anywhere between a few thousand dollars to close to millions of dollars for a very large call center handling thousands of calls per day.

Automatic Call Distributor Vendors

3CLogic Cloud-Based Contact Center Software 3CLogic is a leading provider of cloud contact center solutions based on an innovative approach, designed to deliver modern-day contact center features to meet the challenges of a modern world. With 3CLogic's ACD functionality, you can set, manage, and adjust call priorities to automatically ensure the most urgent inquiries are always answered first.
ICTBroadcast Automatic Call Distributor: Is a Unified automatic call distribution software solution from ICT Innovations . Feature- unifed Auto Dialing, Custom IVR Designer ,Survey Campaign , SMS blasting & marketing , Fax blasting , Voice blasting ,AMD supported, Email marketing and appointment reminder software solution.
Virtual Phone Number IVR GURU providing ivr service for call center to automatic distribute call to multiple number and we louche new DND software to filter data.
Vocalcom Intelligent distribution of calls is something that Vocalcom has been re-inventing for many years, refining and perfecting to ensure the optimum solution to connect customer and agent.
Voicent ACD Software is designed to be configurable to the user. We offer default 'round robbin' call distributions, to the more advanced 'rule & skill based' transfers. Voicent is the leading provider of the Managed Call Center Software.
DooxSwitch DooxSwitch provides one of the most comprehensive and state-of-the-art cloud-based call centre software solutions.
Five9 ACD Software is designed so that any business user can configure it, yet it has all the sophisticated routing features any enterprise requires. Five9 is the leading provider of cloud contact center software.
Foehn - We are the experts in IP Communications with over 12 years of successful deployment of Asterisk and open source technology solutions. ...

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CNAM

September 12, 2015, 7:41 am

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CNAM is an acronym which stands for Caller ID Name.

When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can be used to display the calling party's name alongside the phone number, to help users easily identify a caller.

There are numerous CNAM lookup services which allow you to pay a small fee to lookup the CNAM of a specified caller (by phone number).

CNAM Lookup Services List:

http://www.bulkcnam.com/ Cost: Only $0.005 per query for carriers or $0.009 for hobbyists! No catch, guaranteed with easy paypal integration. Sign Up for a FREE Account and we will credit you 30 FREE CNAM queries to try http://www.bulkcnam.com/.

http://www.calleridservice.com No monthly fees or account minimums and 20 free queries to test our service when you open an account ( instant setup ). Simple HTTP API or Fast AGI that can be placed in your Asterisk dial plan. Also native support for Switchvox PBX systems. Results are never cached so you get up to the minute real-time results. Retail prices are $.006 per query and bulk pricing is available with a volume commitment of at least 25,000 queries per month. Free support and installation assistance is available.

www.callwithus.com offers both CNAM ($0.006) and LRN ($0.0003) look ups. No minimums and monthly charges. Simple HTTP API, easy to integrate to Asterisk dial plan.

CID(name) Professional CNAM (Caller name) delivery

EVERY LOOKUP IS LIVE FROM THE SS7 (direct from the carrier owning the number)
NO CACHING... EVER!
NO 3rd party data sources
NO monthly fees
NEVER pay full price for unavailable results
Carrier grade, multi-redundant platform
Simple to integrate HTTP API
99.7% caller id name accuracy
Lightning fast query responses (under 500ms)
Volume pricing as low as $0.002 per query
Try before you buy, 100 free dips with every new account
You choose the output, TEXT/JSON/XML
Track sub-accounts
Easy integration with Freeswitch, Asterisk, OpenSIPS, and other open source voip platforms
Easy access and daily downloads to your account activity
Thousands of happy customers

Get CARRIER GRADE CNAM at http://www.cidname.com

www.cnam.info offers both CNAM and a pseudo-CNAM service at a fraction of the cost. Integration with asterisk is as easy as downloading the AGI and adding a single line to your dial plan.

http://www.data24-7.com CNAM service for $12 per month (membership fee) and $0.005 per query ($0.004 for over 500,000 transactions per month). Simple HTTPS/XML-based API with examples, plus batch file upload and manual entry available via website. Asterisk support. We also support SIP protocol so your switch/dialer can communicate with our service directly. Members have full access to Data24-7's other services, such as carrier lookups and email-to-SMS gateway address lookups. Free trial!

http://www.multitel.net/ We have multiple SS7 interconnects and are able to provide you with some of the most accurate and up to date results. Pricing is $0.004 per query for our Free tier and it goes to $0.003USD and $0.0025USD for our Standard and Professional tiers. We do provide free testing - limited to 60 queries per hour. No commitment , no hassle. ...

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AVOXI Reviews

September 12, 2015, 1:09 pm

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Read reviews about AVOXI Cloud PBX here. If you have experience with AVOXI cloud PBX, feel free to write your own.

VoIP Wholesale

September 12, 2015, 2:22 pm

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Wholesale VoIP Market:

There is no doubt today that VoIP is taking over the telecom market, and every month increases penetration into services and industries. Competitive carriers are looking at the numerous ways to make money from this exploding technology, but there's a lingering question as to whether it is profitable to deliver VoIP in a wholesale model? Their customers, typically Service Providers, are looking for their ‘competitive advantage' into this ‘lowest price' race, leveraging within three key alternatives for packet telephony : “build” , “buy” or “rent”. Business aspect, there’s no need to invest tens of millions of dollars in wholesale VoIP to join in. Many Telecom Companies have done the work for you. They offer a complete, turnkey VoIP business service and equipment. Now you can start wholesale VoIP business with virtually no investment and yet reap great dividends.

Wholesale VoIP Resellers:

In today’s world, Service providers seeking to deliver VoIP to as wide a customer base as possible may find that becoming a wholesale VoIP reseller is the way to go. Wholesale VoIP may be sold to both other service providers and to enterprises or residential customers.

Reselling IP telephony as a wholesale VoIP company is becoming an increasingly popular business model. For many companies, becoming a wholesale VoIP provider hits the sweet spot between profit and market control. Any firm with a well-established customer base is a good candidate for reselling wholesale VoIP.
Becoming a wholesale VoIP reseller is not a decision that should be taken lightly. It does, however, offer the potential of being very lucrative if done right.

Wholesale Consumer Demand:

An important characteristic of the industry is the complex segmentation of consumer demand and rapid change in the characteristics that are being demanded, both at the end customer and in the intermediate ones (wholesale customers).
Demand coming from ‘packed customers'? will be significantly different of the conventional telecommunications one, were telephony was the unique service to provide and differentiation was based on tariff-distance paradigm, being today's service offerings closer to data applications rather than telephony. Voice communication (and not old POT telephony) becomes the common feature into several communications applications and devices, but not the unique one.
Messaging, conference, collaboration, web contact centres, etc … requires a common communication format between parties, which is voice, implemented through VoIP technologies. Heterogeneous and rapidly changing customer demands and products are important dynamic influences on the evolving structure of the telecom industry, resulting into a new value-chain.
Telecommunication markets evolution will be driven by ‘packed customers' demand rather than networks, technology or finance, changing many decades rules into this industry.

Finance in Telecommunication Industry:

Finance institutions had been influencing Telecomm Industry since the beginning, due the business itself was characterized by huge investments, big market shares and bigger capitalization, influencing in many cases top management, who addressed their strategy towards ‘stock' opportunities rather long term and solid business models. WorldCom crash has been an example of this ‘financial market' pressure and wrong business management.
Today, the networks has been deployed. New scenario in Telecoms enable new players to deploy services over broadband without proprietary network and this new generation business will not be anymore capital intensive, let's say these will be innovation intensive.

U.S. VoIP Market:

The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants most lacking network facilities enter the market and drive demand for telephony features and applications, the high-tech market research firm says.
As retail VoIP expands, wholesale VoIP will accelerate quickly, says Bryan Van Dussen, In-Stat analyst. The largest segment remains international VoIP, but we expect the market for local services to surge from 12% of all revenues to 27% by 2010.
Recent research by In-Stat found the following:

Consumer VoIP adoption will drive wholesale VoIP revenues to $3.8 billion by 2010 from $1.1 billion in 2006.
In-Stat finds small businesses are driving the growth of hosted services in the U.S. Hosted VoIP seats in the U.S. ...

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VoiceXML

September 13, 2015, 2:33 am

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VoiceXML is used to create IVR applications in PBX solutions and other applications.

From the W3C Candidate Recommendation document:

VoiceXML is designed for creating audio dialogs that feature synthesized speech, digitized audio, recognition of spoken and DTMF key input, recording of spoken input, telephony, and mixed initiative conversations. Its major goal is to bring the advantages of web-based development and content delivery to interactive voice response applications.

Hello World example :
<?xml version="1.0"?>
<vxml version = "2.0" xmlns="http://www.w3.org/2001/vxml">
<form>
<block>Hello world!</block>
</form>
</vxml>

New Software Releases

September 13, 2015, 2:48 am

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This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

September 2015

2015-09-13 - IVRPowers - VoiceXML Platform 1.0 - a new IVR software base platform for any business.
2015-09-08 - SBO Multipath - a new invention for VoIP service provider.
2015-09-02 - Alaris Labs announces release of the new module for its Alaris inVoice for

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i6net

September 13, 2015, 6:28 am

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VoiceXML for everybusiness | www.i6net.com

I6NET provides a VoiceXML browser for Asterisk :

Products

VoiceXML Browser for Asterisk VXI*

The VXI* VoiceXML browser for Asterisk® gives operators and solution providers the ability to rapidly develop and deploy innovative voice and video applications via IP, PSTN, and 3G-324M networks. VXI* is fully compliant with the W3C's VoiceXML 2.0+ specification and is integrated with automatic speech recognition (ASR) and text-to-speech (TTS) software to enable advanced voice and video solutions, and real-time video calling applications. VXI* can be installed in common hardware configurations, providing a highly scalable base system to meet all customers' business and technical VoiceXML requirements.

VXI* VoiceXML interpreter works directly with the Asterisk® PBX software supported by Digium®. Not only can users of the open source PBX run VoiceXML applications in the same server, they can now offer these powerful, scalable IVR / IVVR solutions at an affordable cost.

Written in C, Linux Operating System, License Commercial

Key features and benefits

VoiceXML V2.0/V2.1 compliant
CCXML replaced by Call Control functions of Asterisk
Base on OpenVXI voice browser template
Application for Asterisk PBX (app_vxml)
VoiceXML accounts managment configuration (for hosting services)
Support plugging objects with the VoiceXML tag, SDK with API available
Fax support (send and receive)
Video silence parameter with VoiceXML syntax
Text-to-Speech (TTS) connector included
Automatic-Speech-Recognition (ASR) connector included
Text-to-Video (TTV) with HTTP connector (option)
Voice-Silent Detection (VSD)
MRCP v1 and v2 thru uniMRCP for TTS and ASR.

Xtras* Software Extensions

Xtras* are extended modules for VXI* and Asterisk that VoiceXML is able to control. These extensions provide special features and tools to get or post information thru Asterisk with the interpreter. All these packages can be deployed in a same VXI* server according to each system's configuration requirements.

Xtras* Outbound Dialer/API (Answering machine detection and full call reports)
Xtras* Call Recorder MP4 (h264 and h263)
Xtras* MRCP Client (uniMRCP with some improvments for VoiceXML integration)
Xtras* Flash/RTMP Server Channel (Allows call from WebBrowser using FlashPlayer)
Xtras* Video IP/3G and Tools

Written in C, Linux Operating System

Text-to-Speech engines supported:

Flite - free (TTS) from CMU Speech Group
Loquendo (TTS)
Ivona (TTS)
eSpeak MBROLA - free (TTS)
Voxygen (TTS)
Cepstral (TTS)
Verbio (TTS)
Nuance Scansoft (TTS)
Acapela (TTS)
VoiceInteraction (TTS)
Yantra Software (TTS)
GoogleTTS
Festival (Asterisk application)
Universal HTTP connector for any TTS engines
Universal MRCP connector for any TTS engines (uniMRCP)

Automatic-Speech-Recognition engines supported:

Lumenvox Speech Engine (ASR)
Verbio Speech Server (ASR)
VoiceInteraction Speech Engine (ASR)
Vestec Speech Engine (ASR)
Loquendo Speech (ASR-MRCP)
Nuance Speech (ASR-MRCP)
PocketSphinx (SpeechToText)
universal MRCP v1 and v2 (ASR) (uniMRCP modified with extra features)

Asterisk requirements:

Asterisk 1.4 / 1.6 / 1. ...

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VoIP Termination

September 14, 2015, 11:30 am

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Please add information to this page about VoIP call termination.

What is VoIP Termination?

VoIP call termination is used to refer to the procedures that are used for routing telephone calls from one provider to the next provider until the call has been routed to the last telephone company and has been received by the recipient. Voice termination is another term that is used for call termination. The telephone companies are also referred to as providers or carriers.

Called Party

The called party is the person who has received the telephone call. The end point of the route may be on the Internet or may be at a point that was reached by routing the call through the public switched telephone network. The procedures for routing the call stop when the call has been received by the recipient. The process may seem simple to individuals who do not experience problems with making telephone calls but is not so simple to individuals who make calls that will not connect such as calls to a different country.

Calling Party

The calling party is the person who has initiated the call and who wants to be connected to the called party. There may be problems with the telephone calls that begin on the Internet and end at a cellular phone. The sound of the voices may seem to be amplified with echoes and have a scratchy quality. The calls that are initiated with Google Voice may never connect or could be delayed.

VoIP

Voice over Internet protocol (VoIP) is a term that is used to describe a call that was initiated on the Internet such as Skype calls or calls made with Google Voice. The calls that were initiated on the Internet usually end at a point that is not on the Internet. Most of the recipients of telephone calls receive the calls with a landline phone or with a cellular phone. The route may begin on the Internet but will end at a point that was reached by routing the call from the public switched telephone network (PSTN), which is the common description for call termination.

Internet Networks

A tier-one operator is licensed and registered to operate an Internet protocol (IP) network for Internet telephony services. Tier-one operators can handle call origination and call termination. A tier-two operator can lease services from a tier-one operator. The tier-three operators can lease services from either a tier-one or two-two operator. There are also resellers of VoIP services and wholesalers in the market. The quality of the services is not very high because of the inconsistencies in the market such as fluctuations with demand, fraud and problems with doing business on an international level.

Call Origination

Call origination is used to refer to telephone calls that originate from the public switched telephone network and end the route on the Internet. Call termination is considered to be the opposite of call origination because the direction of the paths are reversed. The terms are associated with the starting point of the calls, path of the route of the calls and termination point of the calls. The operators of IP telephony services can handle calls that originate or terminate on the Internet.

Fees

The fees for services are subject to the regulations of several countries because the routes of the calls will cross over more than one country. Those countries can use legislation to control the fees for the services. Termination rates are usually very high for the countries in the Middle East and Asia. The rates are intentional high because there are more incoming calls than outgoing calls from the countries, which is caused by the diaspora effect of a migrating population.

VoIP Termination Providers

Please list VoIP Termination providers here in alphabetical order.

373K REAL Wholesale.

Local/LD Origination - $0.25/DID and No MOU or port charges--free inbound!
US and Canada Termination - Over 70% for less than $0.005, no commitments, or port charges/limits!
Toll-Free Origination - FREE Toll-Free DIDs and usage rates starting at $0.0002!
Toll-Free Termination - Get compensated for your toll-free termination.
Support - Live humans answering the phone 24/7. Engineers available for free assistance.
There's a reason service providers entrust the traffic of over 60 million users to us everyday. ...

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VoIP Origination

September 14, 2015, 11:31 am

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Please add information to this page about VoIP Origination.

What is VoIP Call Origination?

One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

What is Call Origination?

VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

Required Hardware

The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

How Call Origination Works

Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

Types of VoIP services

There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

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Sip Trunking Providers

September 14, 2015, 11:32 am

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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specific pages:

What Is SIP Trunking?

Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

The Benefits of Using SIP Trunking Services

Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.

It offers very low cost calling.
It's much easier to scale than other options, making it very future proof.
SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
Network outages are much less impactful, as incoming calls can easily be routed to other locations.
It's ideal for any sized business with at least 25 physical phones.
It's a fantastic choice for any business that has an international location.
It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

How SIP Trunking Can Take Your Business To The Next Level

It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level. ...

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WebRTC vs VoIP

September 14, 2015, 5:25 pm

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Theres is a bit of confusion in the telecommunication industry as to whether or not WebRTC is compatible with or runs against VoIP. WebRTC is a viable Internet Protocol (IP) communications system that parallels and runs alongside the internet-based phone system VoIP. VoIP services and WebRTC solutions are both heavily promoted in the business and residential fields. So the confusion lays here: are VoIP and WebRTC providers friends or foes? Can the two systems coexist, do they overlap, and how does the client benefit from these?

Page Contents

The similarities

WebRTC and V.VoIP are similar in that both aim to enhance the user experience and enable any consumer device (whether it be mobile phone, fax, internet etc.) to effortlessly connect from anywhere and on any network internationally.

The differences

The primary difference between the two services is that VoIP uses a multitude of variants such as VoIP over DSL/cable modem, voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS), while WebRTC is solely focused on browser-based communications.

VoIP

VoIP is an online telecommunications system which offers simpler and more efficient technology than traditional phone service. VoIP uses advance phone technology in order to make phone calls from the office or home more cost effective and with more features. Standard telephone systems uses telephone lines to transmit phone calls, using physical circuits for connection. Since VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices. Millions of people and businesses have switched to VoIP in order to save money as well as to be able to access the same lines from any place and any device.

Benefits that most VoIP providers include are: around the clock customer service; reduced costs compared to traditional phones; no installation or service fees; free ad-ons including unlimited calling to the US and Canada, unlimited extensions, 1,000 free toll-free minutes, high-definition video-conferencing, desktop integration with popular CRMs, online PBX controls, virtual extensions, remote access, auto-attendants, and unlimited extensions for multiple office locations.

WebRTC

WebRTC (Web Real-Time Communication) is an API being drafted by the World Wide Web Consortium (W3C). Put simply, its a software intermediary that makes it possible for application programs to interact with each other and share data. WebRTC is used to enable browser-to-browser applications for voice calling, P2P file sharing, and video chat without plugins. WebRTC is an emerging technology that are accessed with JavaScript APIs and currently in development are an audio and video data stream as well as API which allow for two or more users to communicate browser-to-browser, real-time gaming, text chat, file transfer and other online based sharing.

Connecting vs. clashing

WebRTC makes it feasible for web developers to enable VoIP into their Web-based applications. Since WebRTC is in its early stages of development, it does not include any signaling protocol which leaves this choice and development and integration to the developer. By integrating a signaling protocol into WebRTC, a developer can create a full VoIP soft client on a browser.
One nice example of such VoIP soft client is CryptoVoIP SIP WebRTCDialer which uses SIP Protocol for signalling. The good part of CryptoVoIP web dialer is that it does not require web-sockets or webrtc support in SIP Servers or softswitch. ...

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VOIP and VPN

September 14, 2015, 9:33 pm

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Surprisingly, using VOIP across an SSL-based VPN can actually improve the call quality (as measured by MOS scores). The improvement seems to be due to encapsulating the UDP VOIP packets ( SIP and RTP ) in TCP/IP. NB Datagram-based VPNs, such as IPSec's ESP are still bad.

According to a study by Sirrix VPN has no negative influence on latency, jitter and packet loss; in the case of the g7.11 codec and compressed VPN it is even possible to gain 10% bandwidth compared to non-VPN traffic. Apart from that, different common VPN solutions have big difference on the available throughput, which is due to the rather small packet sizes and greatly increased overhead:

With enabling authentication, encryption, HMAC, anti-replay attack, and initialization vector, and use small RTP size for Codec, the vpn overhead is high:
g723 with 30ms RTP size and using VPN tunneling: approx. 85% overhead;
g729a with 20ms RTP size and using VPN tunneling: approx. 80% overhead;

But when making some adjustments on the encryption/authentication settings and double the RTP size, the overhead can go down to about 20%-30%, which is affordable for most of cases.

Comparing to SRTP as encryption method for VoIP: approx. 5% additional overhead.

There is an OpenVPN-based service available on the net which resolves the excessive traffic consumption issue. Several voice packets are placed in the buffer before encapsulation. This minimizes VPN impact and traffic usage doesn't grow with VPN service. This can also help to prevent VoIP traffic detection by packet size, since the size of a single packet is comparable with MTU size (usually 1500 or less).

VoIP and VPN Forums:

VoIP Security Forum

VoIP Tunneling methods

CryptoVoIP EP Tunnel VoIP Anti Block Solution with compression works on both UDP and TCP. Make all VoIP calls in VoIP blocked areas work with all ISPs.
SBO Tunneling Method for VoIP
Zebedee: Can tunnel UDP via TCP (HowTo for Asterisk in German)
Stunnel: Uses SSL, can do both UDP and TCP
OpenVPN: Can do both UDP and TCP
Mizutech VoIP tunneling solution: A complete solution (both server and client side) for encrypted voip

Articles

VoIP-sip.org - VoIP call quality over VPN
Network World -

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WebRTC

September 14, 2015, 9:37 pm

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Synopsis

The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

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The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.

Overview

WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.

Architecture

The architecture of WebRTC, as described by W3C looks something like this:

Design

Major components of WebRTC include:

getUserMedia, which allows a web browser to access the camera and microphone
PeerConnection, which sets up the audio/video calls
DataChannels, which allow browsers to share data via peer-to-peer

Support

Chrome WebRTC Development Team

Discussion List: https://groups.google.com/group/discuss-webrtc
Google Plus Page: https://plus.google.com/113817074606039822053
Chrome WebRTC Issue Tracker: http://code.google. ...

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Paging Management System

September 14, 2015, 10:19 pm

≫ Next: CooCall App

≪ Previous: WebRTC

Paging Management System is an innovative software which is primarily used to distribute public address via sip protocol over SIP Speakers. According to the simple interconnection with CooVox IP PBX, multiple SIP Speakers will be centrally managed, monitored and configured through paging management system.

Paging Management System allows you to make announcement with specific voice file at the specific time/date. And all the announcement logs are available to check from the paging management system. Moreover, Paging Management System is easily expandable to add more SIP Speakers to the network.

The SIP based audio system iSpeaker utilizes the built-in intercom and paging capability already inherent in most modern IPPBX systems and enhances this to improve end user experience by providing a dedicated high performance digital amplifier on which to distribute announcements or play background music. There are two models now: iSpeaker B20 and iSpeaker C20.

Key Features

Volume Control
Group announcement
Real-time announcement
Public Address for specified area
Public Address in specified time
Remote settings /management/ upgrade
Centralized management of audio files
Centralized configuration of SIP speakers
Real-time and remote monitor speakers status
Friendly Web interface and easy to use
Paging to up to 500 SIP Speakers
Custom service is available (e.g.: emergency announcement)

Paging Management System Application
Applicable in mall, factory, office, school...

Contact us:

ZYCOO China
Web: www.zycoo.com
Tel: +86 (28) 85337096
Address: 7F, B7, Tianfu Software Park, Chengdu, China.

ZYCOO UAE
Web: www.zycoo.ae
Tel: +971 (4) 3798839
Address: #305, Al Riffa Plaza Bldg., Khalid Bin Alwaleed RD, DUBAI, UAE

ZYCOO UK
LS21, Armstrong House, First Avenue, Robin Hood Airport, Doncaster, UK (DN93GA)

ALL THE RIGHTS ARE RESERVED BY ZYCOO Co.,Ltd

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CooCall App

September 14, 2015, 10:20 pm

≫ Next: Automatic Call Distributor

≪ Previous: Paging Management System

CooCall App is the free softphone based Android and iOS to be integrated with the ZYCOO's PBX platform. You can take your extension along with you from anywhere with internet connctivity for mobile office.

Mobile working from anywhere

No cost for making or receiving calls via CooCall
Call recording to allow recording any calls you need
Call waiting display for alerting user about new call
Call transfer to another contact
Three-way conference call
VoiceMail is able to be saved in your cell phone

Friendly using and configuration

Primary IP configuration for office network connection
Secondary IP configuration for mobile workforce (failover IP of VoIP server)
Auto integration of phone contact book and PBX contact to Contact list
Busy Lamp Field to allow user to view their contacts current status

Contact us:

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Automatic Call Distributor

September 15, 2015, 9:56 am

≫ Next: VOIP Consultants

≪ Previous: CooCall App

Automatic Call Distributors

Automatic Call Distributor Vendors

3CLogic Cloud-Based Contact Center Software 3CLogic is a leading provider of cloud contact center solutions based on an innovative approach, designed to deliver modern-day contact center features to meet the challenges of a modern world. With 3CLogic's ACD functionality, you can set, manage, and adjust call priorities to automatically ensure the most urgent inquiries are always answered first.
ICTBroadcast Automatic Call Distributor: Is a Unified automatic call distribution software solution from ICT Innovations . Feature- unifed Auto Dialing, Custom IVR Designer ,Survey Campaign , SMS blasting & marketing , Fax blasting , Voice blasting ,AMD supported, Email marketing and appointment reminder software solution.
Virtual Phone Number IVR GURU providing ivr service for call center to automatic distribute call to multiple number and we louche new DND software to filter data.
Vocalcom Intelligent distribution of calls is something that Vocalcom has been re-inventing for many years, refining and perfecting to ensure the optimum solution to connect customer and agent.
Voicent ACD Software is designed to be configurable to the user. We offer default 'round robbin' call distributions, to the more advanced 'rule & skill based' transfers. Voicent is the leading provider of the Managed Call Center Software.
DooxSwitch DooxSwitch provides one of the most comprehensive and state-of-the-art cloud-based call centre software solutions.
Five9 ACD Software is designed so that any business user can configure it, yet it has all the sophisticated routing features any enterprise requires. Five9 is the leading provider of cloud contact center software.
Foehn - We are the experts in IP Communications with over 12 years of successful deployment of Asterisk and open source technology solutions. ...

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VOIP Consultants

September 15, 2015, 10:19 am

≫ Next: PySWITCH

≪ Previous: Automatic Call Distributor

Please find information about VoIP Consultants below. Feel free to add VoIP consultant information, but be sure to please review the Posting Guidelines for Promoting Products and Services and the HIGHLIGHTED text in the edit screen before posing your consulting company.

Please sort your service by your company name and don't use marketing tricks to get to the front of the list (eg. Putting AAA in front of your name, unless it actually is your business name).
Editors are encouraged to delete entries not following the guidelines

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PySWITCH

September 16, 2015, 3:10 am

≫ Next: BULK SMS

≪ Previous: VOIP Consultants

Twisted Protocols for communication with FreeSWITCH. This is similar to StarPy for Asterisk

PySWITCH allows you to communicate with FreeSWITCH using inbound and outbound EventSocket connections.

The protocols are designed to be included in applications that want to allow for multi-protocol communication using the Twisted protocol. Their integration with FreeSWITCH does not require any modification to the FreeSWITCH source code (though an eventsocket account is obviously required for the Inbound connections, and you have to actually call the Outbound server from the dialplan).

Documentation of PySWITCH - http://pyswitch.sourceforge.net/docs/index.html

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BULK SMS

September 16, 2015, 5:58 am

≫ Next: FoIP

≪ Previous: PySWITCH

Bulk SMS, also known as bulk messaging or bulk text messaging is the act of disseminating SMS messages in large numbers so they are able to be delivered to various mobile phone terminals. This form of messaging is typically utilized by consumer brands, banks, enterprises and media companies. The messages are generally used for mobile marketing, enterprise and entertainment. However, banks often use them for fraud control. For example, if criminals are circulating a fake email that is asking people who have accounts at a certain bank to provide their social security numbers or other confidential information, these text messages can alert people to the scam so they do not fall victim to it. Bulk messaging is often utilized for reminders and alerts. However, it is more frequently used to send communications and information between customers and staff of various companies. Bulk messaging enables the delivery of SMS messages to large numbers of mobile phones that are located all around the world.

Bulk messaging software

In order to receive and send bulk messages, software is needed. There are many types of software packages specifically designed for this task that are available. These packages give their users the ability to send messages to as many phone numbers as they want. There are many different ways in which these phone numbers can be managed.

The vast majority of software applications that are designed to be used with SMS enable the user to upload mobile phone number lists with the use of a CSV or TXT file. Systems that are more advanced are capable of automatically deleting any numbers that are repeated. There are also systems that can be programed to validate all of the mobile phone numbers before the messages are sent to them.

Enhanced software features are also currently available that allow users to schedule messages to be delivered at certain days and/or times. Bulk messages are also able to be sent on mobile networks that are international or national, assuming that the provider of the bulk messaging software sends internationally.

Bulk messaging portal

Bulk messaging features can be added to websites through the use of this specific online script. Unlimited mobile phone numbers can be added to the list of numbers to send messages to. There are a wide variety of ways that can be used to manage these numbers.

Bulk messaging API

The majority of services that handle bulk messaging use the API's (Application Programming Interface) listed below. These enable the addition of functionality to programs by their programmers:

Email
HTTP
SMPP (Short Message Peer to Peer)
FTP (File Transfer Protocol)

Immediate benefits of using bulk SMS messaging

When a particular business is not doing well financially, they need to utilize various tools that can help them gain a competitive advantage in their specific industry. One of the main reasons that bulk SMS is so popular is its ability to lower operational costs while also generating revenue at the same time. Bulk SMS might be the only medium that is able to show a return on investment that is able to be measured. Wholesale SMS messaging is targeted, which makes it extremely effective at getting people to respond and generate revenue.

Reduces operational costs

Bulk SMS message transmission is more effective than email and less expensive than voice calls. There are thousands of businesses located all around the globe that utilize wholesale SMS as a way to communicate with their suppliers, employees and customers. There is a significant cost savings as a result of time being saved because actual voice calls to suppliers, employees and customers do not need to be made. A single message can instantly be sent to many people at the same time, as long as the person is located in an area with mobile coverage. The ability to disseminate information so quickly to a large target audience reduces communication costs while also generating revenue if used for marketing purposes.

Allows customers to be accessed easily

More people have mobile phones than have access to email or landline phones. Every mobile phone supports the use of text messaging. All mobile phone users are comfortable using this technology because it is simple and easy to understand. This makes wholesale SMS the perfect medium to use for communication with customers. There are also no demographical or geographical restrictions. ...

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FoIP

September 16, 2015, 2:35 pm

≫ Next: T.38

≪ Previous: BULK SMS

Fax over IP - FoIP

Fax over IP can be done in many ways, either realtime or store-and-forward

Realtime

Transmission of a fax audio stream over a VoIP protocol, like SIP, IAX, H.323. This requires an uncompressed 64 kbit codec to have any chance of working, resulting in the need of around 80 kbit bandwidth. For a number of reasons it may or may not work even then.
Fax over IP with a special protocol designed for this application. The key standardised protocol for this is T.38.

Store and forward

A FAX, from a FAX machine, is decoded and stored by a FoIP gateway. It is then sent, using SMTP or a similar transport, to another FoIP gateway where it may be sent out to a destination FAX machine. The key standardised protocol for this is T.37.

FoIP Resources

All about FoIP As IP networks and VoIP adoption expands, and traditional Public-Switched Telephone Network (PSTN) phone services change, organizations with paper-intensive communication processes increasingly look to Fax over IP (FoIP) as part of their solution. AllAboutFoIP.com is a bold initiative which provides VoIP providers, integration specialists and their clients with the most instrumental offerings of FoIP components and solutions, including Audio Codes, Alcatel, Avaya, Biscom, Cisco, Commetrex, Dialogic (Brooktrout), EtherFax, Faxback, Faxcore, GFi, Open Text (RightFax), SagemCom (XmediusFax), ShoreTel, and TargetFax.

FoIP Software

T.38 with Fax Voip T38 Fax and Voice

FaxVoip Software develops solutions for the transmission of a fax via the Internet Telephony (FOIP). The main emphasis has been placed on the transfer T.38 fax using SIP and H.323.
Fax Voip application operates with T.38 faxes via standart COM port interface.
Fax Voip T38 Fax & Voice - Fax and Answering Machine for your SIP/H.323/ISDN CAPI 2.0 line. Multiple SIP Registrations. Call Routing. Voice Fax Modem for your Fax & Voice software. Color faxes over VOIP and ISDN. Incoming Routing Methods. Fax-On-Demand.
What is Voip? For your fax or voice application, it's a Voice Fax Modem. You can setup your Fax & Voice program to operate with Fax Voip Virtual COM ports or virtual Fax Voip 14.4K TAPI Voice-Fax Modems. From the perspective of your VOIP internet network, it’s a SIP/H.323 client with T.38 and G.711 Fax support.From the perspective of your ISDN line, it’s CAPI 2.0 client with audio fax support. You can send/receive T.38 and audio (color/black-and-white) faxes and voice messages without any hardware, using your favourite Fax & Voice program. Fax Voip is the ideal solution for the implementation of Fax and Voice Mailbox into SIP/H.323/ISDN network. You can use Fax Voip with your VOIP or ISDN-based PBX or with your SIP/H.323/ISDN Provider. Up to 100 virtual modems can be used simultaneously.
Multiple SIP Registrations and Call Routing functions make your system the most flexible as well as allow you to work with different SIP and H.323 providers simultaneously.
Caller ID and Dial a Phone Number Extension features are supported.
You can send T.38, audio and CAPI faxes via Fax Voip Virtual Printer and receive faxes directly in TIFF, PDF or SFF files. You can manage faxes with Fax Voip Console.
Fax Voip allow to send faxes via e-mail (Mail to Fax) and receive faxes to e-mail (Fax to Mail).
Incoming Routing Methods (Route through e-mail, Store in a folder, Print) allow you to route incoming faxes to recipients on the network.
Fax on Demand function allow callers to retrieve information via fax on the same call.
Fax Voip T38 Fax & Voice has been successfully tested with Microsoft Fax, CallStation,

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