Unified Communications

September 3, 2015, 11:10 am

≪ Previous: Ecosmob Technologies: VoIP Consultancy and Software Development Company

Unified Communications, often abbreviated as UC, is the combination of different types of communication mediums including telephone service (VoIP), chat, video and web conferencing, messaging, email, fax, and other types of communications. Unified Communications systems can be sold and implemented with a select amount of individual communication mediums, or many integrated into a single unified system.

Unified communications consist of integrating various real-time and non-real time communication services into a convenient package for the user. Examples of real-time services commonly used include instant messaging, telephony and data sharing. Non-real time communication services focus mainly on messaging services like e-mail, SMS, voicemail and fax.

With a unified communications solution, the user can receive a message through one communication medium and use another one to access it. One common example of this would be receiving a voicemail message on an office landline phone and using a mobile phone to retrieve it. ...

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Virtual PBX providers

September 3, 2015, 11:24 am

≫ Next: VOIP PBX and Servers

≪ Previous: Unified Communications

Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.

What Is a Virtual PBX?

A PBX, short for private branch exchange, is a telephone system with the capacity to switch calls between different users on local lines while still relying on the same number of external phone lines. With a virtual PBX system, the system is posted and software based without all of the traditional hardware of a physical PBX.

Virtual PBX Primary Function

A virtual PBX is used by businesses in a variety of ways. Primarily, companies utilize the system as an auto-attendant to establish preset call transfer options without needing an operator or receptionist. This type of system is capable of performing tasks that include auto-attendant settings, time of day or day of week functions, or even find or follow me sequences.

One of the most important functions of a virtual PBX system for companies is the software’s ability to establish pre-determined sequences. For example, in some businesses it may be appropriate for the phone to ring to a receptionist or operator first. If the receptionist does not answer in a predetermined number of rings, however, the call is then transferred to a secretary. Again, if the call is unanswered, it can be set to forward to an assistant. Left unanswered by these two individuals, the call can be forward to a manager or even an owner. These call settings are completely customizable and can be based on any number of sequences.

This type of software is also able to facilitate customized answering menus and sub-menus. The system can be modified to establish appropriate dial prompts leading to a number of different departments within the business, including different sequences on different days. PBXs are used by the vast majority of businesses to establish advanced call routing services.

Virtual PBX Cost

A virtual PBX is a complex service; however, that doesn’t mean that it is expensive. In fact, a virtual PBX is typically more cost effective than a physical PBX. The main reason that a virtual system saves on cost is that it does not require the same investment in capital to establish or set-up the call system. Because a virtual PBX is a software or hosted system, it is typically an operational cost, or a low monthly payment rather than a large upfront investment. This aspect alone generally makes a virtual or hosted PBX a less expensive, or at least more cost effective, option compared to the traditional PBX.

Virtual PBX Benefits

Aside from offering an effective call system, a virtual PBX presents a number of added benefits for users. As a whole, virtual PBXs lead the industry in business communication choices. This type of system seamlessly integrates the call management system with any existing phones to affordably and effectively deliver better call management. These systems also feature several innovative call features to meet the needs of any business. These systems offer various functions including call routing, follow and find me call forwarding, voicemail notifications, call recording, and more.

The benefits aren’t limited to the features, though. Virtual PBXs offer virtually limitless application for one or hundreds and even thousands of employees. Likewise, there is not hardware to maintain or constantly upgrade. Considering that benefit, the system is also more cost effective and generally provides for a variety of flexible billing options. The limited maintenance, web-based management, and hassle-free setup alone are often enough to convince a company to switch over to this option.

PBXs are an important tool in any business that makes and receives nearly any volume of calls. A virtual PBX can dramatically increase the efficiency of a business by effectively managing calls. This efficiency combined with the other numerous benefits of a virtual PBX can virtually transfer the communication capabilities of any company.

List of Virtual PBX Providers

See also Virtual PBX

OnePipe

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VOIP PBX and Servers

September 3, 2015, 11:33 am

≫ Next: voip-info.org

≪ Previous: Virtual PBX providers

Please list information about VoIP PBX and Servers on this page. Please keep VoIP PBX and server provider information in alphabetical order, and below any other relevant information.

Page Contents

Numeric

1comms.co.uk: Asterisk-based converged telephone system for UK Businesses
1GATE VoIP PBX by Wangate: Cheap VoIP PBX with hardware DSP. Optional internal gateways for Analog/ISDN/PRI/GSM. VoIP resellers welcome.
2daydirect: Brand NEW Small Business VoIP phones. Free 2 day shipping anywhere in the United States
2N NETSTAR PBX, virtual PBX: VoIP PBX system
2N Omega IP PBX: VoIP PBX system
2N VoiceBlue Enterprise: Simple VoIP SIP PBX
3CX: Windows IP PBX / VOIP Phone system
4PSA VoipNow: Hosted PBX software for service providers and enterprises, accelerating SaaS deployment. It runs on Linux environments (RHEL, SuSE Linux, CentOS, Fedora) on x86 and Power PC architecture based servers.
8ix Zenith: 8ix Zenith spells an Asterisk derived IP Telephony application with the most advanced calling and communication features.

A

ALLO PSTN-IPPBX for SOHO with 30 IP extension, upto 6 Analog Extension & upto 4 PSTN trunk
ActivePBX™ | Turn-Key Business Phone System $149/mo.

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voip-info.org

September 3, 2015, 12:43 pm

≫ Next: Old News

≪ Previous: VOIP PBX and Servers

Welcome to the VOIP Wiki - a reference guide to all things VOIP.

This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

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

NEWS

2015-09-03 - ICT Innovations launched SaaS / hosted software service offering re-known open source telephony projects
2015-09-03 - 2N® Indoor Touch communicator controls home IP audio systems
2015-09-02 - Alaris Labs announces release of the new module for its Alaris inVoice for blended routing management
2015-09-02 - Can 4G Propel VoIP Subscription Growth?

News Resources

Software Releases - Check here for recent VoIP-related software releases.
VoIP Services - Check here for news on Voip Services.
VOIP Event Calendar - Check here for news on VOIP Events, Trade shows, Training and Conferences
Older News - Check out older news articles here

Getting Started

What is VOIP? The very basics.
Free VOIP Publications: Magazines and Newsletters free to qualified subscribers about VOIP related products
Training: Seminars, tutorials, on-line classes. Check here for upcoming Training schedule.
VOIP Consultants: Finding help.
VOIP Service Providers - VOIP service providers.
VOIP Server Monitoring - VOIP server monitoring providers.
DID Service Providers - DID VOIP call origination service providers. ...

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Old News

September 3, 2015, 12:43 pm

≫ Next: IVR

≪ Previous: voip-info.org

This page lists all the old VoIP news stories from the home page.

Page Contents

August 2015

2015-08-26 - GET SBO: A New Home of SBO VoIP Bandwidth Solution developed by Synchronous ICT
2015-08-21 - Communicate with your neighbours using 2N® Indoor Touch
2015-08-21 - UC&C Hits Mass Adoption in North America, Vendors Now Focused On Quality
2015-08-21 - Kamailio SIP Server Advanced Training - Sep 28-30, 2015, in Berlin, Germany
2015-08-18 - Fear Is The #1 Obstacle In Deploying UC&C
2015-08-18 - Vonage Grows Its Indirect Channel with The Alliance Partners
2015-08-18 - 5 Ways a Good Softswitch Can Improve Your VoIP Service
2015-08-14 - How to become Internet Telecom Service Provider
2015-08-14 - Finding a wholesale Voip Provider for your business requirements
2015-08-13 - VoicePulse launches CallCode, a Javascript based call control feature
2015-08-13 - AccuCore ERP System for VoIP now available as a SaaS
2015-08-12 - UCaaS Beats Perpetual Licensing Any Day Of The Week
2015-08-10 - Speedflow launches VoIP outstaffing services
2015-08-06 - Comfortable door communication from 2N
2015-08-05 -

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IVR

September 4, 2015, 12:27 pm

≫ Next: VoIP Termination

≪ Previous: Old News

What Is IVR?

IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component

IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone key-presses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration

IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.

Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.

One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting

IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchtone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Here is an example of IVR implementation in Voice broadcasting

Graphical Design Tool for IVR Applications

Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

VoIP Termination

September 5, 2015, 8:25 pm

≫ Next: VoIP Origination

≪ Previous: IVR

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 5, 2015, 8:28 pm

≫ Next: Sip Trunking Providers

≪ Previous: VoIP Termination

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 5, 2015, 8:31 pm

≫ Next: Asterisk cmd GotoIf

≪ Previous: VoIP Origination

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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Asterisk cmd GotoIf

September 6, 2015, 1:21 am

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≪ Previous: Sip Trunking Providers

Synopsis

Conditional goto

Description


   GotoIf(condition?[label1][:label2])
   GotoIf(condition?[[[context1,]extension1,]priority1]:[[[context2,]extension2,]priority2])

Either label1 or label2 may be omitted (in that case, we just don't take the particular branch), but not both.

   GotoIf(condition?label1)

If "condition" is true, go to "label1". If condition is false, then simply proceed to the next instruction.

   GotoIf(condition?label1:label2)

If "condition" is true, go to "label1". If condition is false, go to "label2".

    GotoIf(condition?:label2)

If "condition" is true, go to the next instruction. If "condition" is false, go to "label2".

condition is just a string. If the string is empty or "0", the condition is considered to be false. If it is anything else, the condition is true. This is designed to be used together with expression syntax.

Labels take the form '[context,]extension,]priority', so they can be (a) a priority, (b) an extension and a priority, or (c) a context, an extension and a priority. This is the same syntax as for the Goto command.

Example 1


  exten => 206,1,GotoIf($["${CALLERID(num)}" = "303"]?dial1)
  exten => 206,n,GotoIf($["${CALLERID(num)}" != "304"]?moh:dial2)
  exten => 206,n(dial1),Dial(${SPHONE1},15,rt)
  exten => 206,n,Hangup()
  exten => 206,n(dial2),Dial(${PHONE2},15,rt)
  exten => 206,n,Hangup()
  exten => 206,n(moh),MusicOnHold(default)

20050427 - Note that ${CALLERID(num)} can contain spaces (e.g. "No CID available" from some Zap channels), and the example above has been corrected to cope with this situation - without the quotes around ${CALLERID(num)} it doesn't work!

Example 2


 ; Are the first three digits of the returned value of ${ENUM} are
 ;  either SIP or IAX?  If not,  we ignore and pass to normal
 ;  dialing path at priority 4.
 ;
 exten => _011X.,2,GotoIf($[$["${ENUM:0:3}" = "SIP"] | $["${ENUM:0:3}" = "IAX"]]?3:4)

Example 3


  ; This example checks for blank caller ID or 800 numbers.
  ; Callers from 800 numbers (usually telemarketers) or those 
  ; with no caller ID are asked to press 1 to speak with me.
  exten => s,1,NoOp(${CALLERID}) ; log callerID string
  ; check for callerID. If none,  make them hit 1.
  exten => s,n,GotoIf($["${CALLERID(num)}" = ""]?1000)
  ; If 800 number, make them hit 1.
  exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "877"]?1000)
  exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "800"]?1000)
  ; OK, we have valid caller ID and it's not an 800 number.
  ; Let's ring our phones now:
  exten => s,n,Dial(SIP/604&SIP/602,25,tr)
  exten => s,1000,Background(press1tospeaktome)

Example 4 (by Tikal Networks )


Important!!!
If from any reason gotoif dose not work with floating numbers try to cut the floating number and compare it with the first argument
exten => charge_set_up_fee,1,Noop(=== NOW STARTING 'set up fee'  ===)
exten => charge_set_up_fee,n,Set(fee_credit=20.36)
exten => charge_set_up_fee,n,ExecIf($[${fee_credit} < 0.50],Playback,custom/credit_less_then_0_5)
exten => charge_set_up_fee,n,ExecIf($[${fee_credit} < 0.50],Macro,hangupcall,EXIT); play msg and exit if user have lass the 0.5 cent
;exten => charge_set_up_fee,n,GotoIf($[${fee_credit} > 20]?continue) ; this will not work 
exten => charge_set_up_fee,n,GotoIf($[${CUT(fee_credit,\. ...

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Training and Conferences

September 6, 2015, 2:36 am

≫ Next: Synchronous ICT-SBO, SyncSwitch & CoolDialer.

≪ Previous: Asterisk cmd GotoIf

Check here for the latest Trainings and Conferences on VoIP and related. Please order by event date instead of posting date. This makes it easier to scan for upcoming events.

Sep 2015

Sep 28-30 - Kamailio SIP Server Advanced Training, in Berlin, Germany

Mar 2015

Mar 20th - WebRTC training Orlando, Florida: FREE!! One Day special overview class when you register to attend SIP Training March 18-19th. contact Info@TrainingCity.com for details!
Mar 18-19 - SIP training & WebRTC training hands on workshop course in Orlando, FL Join TrainingCity's expert SIP & WebRTC instructors in Orlando March 18 - 20th for this great new program. We will be running our ever popular Advanced SIP Training hands on workshop March 18 - 19th and then a special one day overview version of our new WebRTC training course on March 20th.

Feb 2015

Feb 23 - he first 2015 OpenSIPS eBootcamp training session to start on 23rf of February, covering 1.11 version - a seven weeks e-Training program providing in depth coverage of OpenSIPS.
Feb dates TBD: - SIP Training We're hoping to add an extra delivery of our ever popular SIP class in San Diego in Feb. Call or email for details and special pricing Info@TrainingCity.com

Jan 2015

Jan 26-28 - VoIP Course in Las Vegas

Dec 2014

Dec 22-23 - SIP Training Hawaii Join us for Christmas in Hawaii and our ever popular SIP class in Honolulu!
Dec 8-12 - VoIP Training New York City Special 5 day class with two extra days of intense TCP/IP Training

Nov 2014

Nov 18-19 -TrainingCity SIP Training Class New Orleans, LA
Nov 17-21 -Digium Asterisk Advanced - Madrid, España

Oct 2014

Oct 13-17 -

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Synchronous ICT-SBO, SyncSwitch & CoolDialer.

September 6, 2015, 3:24 am

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↧

Synchronous ICT

September 6, 2015, 3:25 am

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VOIP and VPN

September 6, 2015, 11:51 pm

≫ Next: Bandwidth consumption

≪ Previous: Synchronous ICT

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

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 - Test shows VoIP call quality can improve with SSL VPN links
O'Reilly Emerging Telephony

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Bandwidth consumption

September 7, 2015, 12:13 am

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VOIP Bandwidth consumption naturally depends on the codec used.

When calculating bandwidth, one can't assume that every channel is used all the time. Normal conversation includes a lot of silence, which often means no packets are sent at all. So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet (which adds overhead), the full bandwidth is not used at all times.

A codec that sends a 64kb stream results in a much larger IP network stream. The main cause of the extra bandwidth usage is IP and UDP headers. VoIP sends small packets and so, many times, the headers are actually much larger than the data part of the packet.

IAX2 trunking helps with the IP overhead, but only when you are sending more than 2 or so calls between the same Asterisk servers. John Todd has done some useful practical testing, named IAX2 trunking: codec bandwidth comparison notes and results.

The bandwidth used depends also on the datalink (layer2) protocols. Several things influence the bandwidth used, payload size, ATM cell headers, VPN headers, use of header compression and IAX2 Trunked. You can see the influence of some of this factors using the Asteriskguide bandwidth calculator.

Teracall has the table which shows how the codec's theoretical bandwidth usage expands with UDP/IP headers:

Codec BR NEB
G.711 64 Kbps 87.2 Kbps
G.729 8 Kbps 31.2 Kbps
G.723.1 6.4 Kbps 21.9 Kbps
G.723.1 5.3 Kbps 20.8 Kbps
G.726 32 Kbps 55.2 Kbps
G.726 24 Kbps 47.2 Kbps
G.728 16 Kbps 31.5 Kbps
iLBC 15 Kbps 27.7 Kbps

BR = Bit rate
NEB = Nominal Ethernet Bandwidth (one direction)

SBO- Synchronous Bandwidth Optimizer reduce bandwidth consumption and operation but, increase voice quality, a VoIP Software develop by Synchronous ICT
Asterisk guide bandwidth calculator, it calculates 32xCodecs, 13xL2-Protocols, IAX Trunked, ATM, cRTP, VPN
Asteriskguru's bandwidth calculator
Bidirectional Throughput VoIP Test - Utility to test if your network is ready for VoIP Service - Throughput, Capacity, Latency, and SIP & MGCP Port Tests
Cisco.com whitepaper: Per call bandwidth consumption This Cisco whitepaper explains how you can calculate bandwidth with VOIP.
Erlang.com online bandwidth calculators: http://www.erlang.com/calculator/
IAX2 trunking: codec bandwidth comparison notes and results
IPhone App: VoIP Bandwidth Calc and app store link

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SBO

September 7, 2015, 12:22 am

≫ Next: SBO Bandwidth Saver

≪ Previous: Bandwidth consumption

What is SBO?

How it works?

What makes SBO so popular?

Where to apply?

Key Features

Prerequisites to run SBO Solution?

List of Companies Providing SBO Services

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SBO Bandwidth Saver

September 7, 2015, 1:58 am

≫ Next: Synchronous Bandwidth Optimizer... A Bandwidth Solution!!

≪ Previous: SBO

Page Contents

What is SBO?

SBO is the short form of Synchronous Bandwidth Optimizer, it is a complete software for VoIP service provider and it's developed by Synchronous ICT, a world famous VoIP software provider. SBO mainly work to reduce the internet cost, beside this, it also has some other cool features.

How Does SBO work?

There are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO has its own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

Why Bandwidth Saver is Important for VoIP?

Minimizing bandwidth consumption is the main purpose of SBO Solution, but, at the same time it help to reduce business operation cost. As a perfect bandwidth saver SBO cutoff more than 80% internet cost directly to protect it's clients from over paid. But, it never compromised with service quality. In VoIP telecommunication, main expenditure is Bandwidth cost. For a SIP call with G.729 codec it consumes more than 32 kbps of bandwidth. So, per 32 ports bandwidth consumption with G.729 Codec is about 936 kbps and with G.723 its about 657 kbps, while the payload size is only 8 Kbps. Rest of the bandwidth is consumed by RTP and other headers which is not important for voice call. So, a perfect bandwidth solution has capacity to reduce it without degrading the voice quality at all.
To run a quality VoIP business, required a well maintained, congestion free, telco grade bandwidth connection. Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, bandwidth saver with multipath facilities make it possible to run IP business with low cost share internet connection by maintaining supreme service quality. A perfect bandwidth saver should have the following key features:

Key Features of SBO Bandwidth Saver

Reduce Bandwidth Cost for More Than 80% Without Degrading The Service Quality;
Increase Service Quality Dramatically;
Increase Route Performance;
Increase Average Call Duration by 1-2 Minutes. That means ensure good ACD and ASR;
Work with any types of internet firewall to ensure interruption free, stable VoIP business.
Work with all sorts of Internet Connection- 3G, 4G, WiFi, WMAX, EDGE, GPRS and so on;
It allows using more multiple internet connection at a time to balance load among available networks;
Work with Low Bandwidth, and manage congested network to ensure high performance.
It is not important to arrange Static or Real IP to operate your IP business rather it work with normal/ public shared internet connections and support SIP (Session Initiation Protocol);
SBO generally work with all Commonly use Codec, such as G.729, G. ...

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Synchronous Bandwidth Optimizer... A Bandwidth Solution!!

September 7, 2015, 1:59 am

≫ Next: Call Quality Metrics

≪ Previous: SBO Bandwidth Saver

Synchronous Bandwidth Optimizer is generally known as SBO. It is a bandwidth solution for VoIP call termination. There are a plenty of good features in SBO. Among these, the most important and highly recognized one is bandwidth optimization. http://www.getsbo.com/what-is-sbo/ reduce operation cost directly by 80% without compromising with service quality.
The most updated and best features of SBO is SBO Multipath V.3. It include a wide range of facilities for termination business. Nowadays, you can easily plug a number of internet connection at a time to develop your service quality.

Page Contents

Description

In VoIP, we consider transmission cost as one of the major expenditure. Considering such situation, SBO Multipath concentrates on operation cost and it become success to reduces operating expenditure up to 80% by applying it's highly technical features.

SBO works with any types of NAT and firewall internet,So, there a good change to run business even in highly restricted network. SBO Multipath also increases route performance by increasing Average Call Duration by two to three minutes. Multipath feature also give you an opportunity to plug more than one internet connections at the same time irrespective of types of networks. For example, 3g, 4g, EDGE, Wi-Fi, YMAX etc. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks.

key features of SBO Multipath.

1. The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
2. SBO works with any type of firewall and has capacity to avoid blockage issue. That means, it has anti-block feature.
3. Work with all commonly used codes such as G.729, G.723 etc.
4. Multipath: SBO multipath allow you to use number of internet connection same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
5. Highly secure and linux based software.
6. Full featured billing platform SyncSwitch is integrated with SBO. It help you to check CDR, ACD, add or remove clients.
7. Work as third party software.
8. Increase Average Call Duration (ACD) by 2-3 minutes;
9. Works with both real IP and local IP. That means it is not mandatory to arrange real ip. Public or share ip is enough to run your business.
10. Technical support is available 24/7 (265). Our support team is highly experienced and skilled.

News Update:

SBO Multipath: most updated version of SBO solution

References

Official Website:Synchronous ICT
CoolDialer
SyncSwitch
Basis Membership:

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Call Quality Metrics

September 7, 2015, 3:36 am

≫ Next: Automatic Call Distributor

≪ Previous: Synchronous Bandwidth Optimizer... A Bandwidth Solution!!

Several standards are available for measuring telephone call quality.

MOS measures subjective call quality for a call. MOS scores range from 1 for unacceptable to 5 for excellent.
VOIP calls often are in the 3.5 to 4.2 range.

The ITU P.861 (PSQM) and P.862 standards explain how to calculate MOS scores.
The methods used require a test call in order to make the measurements. Kurittu and Mattila debate the practicality of P.862 for VoIP in an AES paper titled "Practical Issues in Objective Speech Quality Assessment with ITU-T P.862". I can't vouch for it because I couldn't find it online for free, but there may be a few similarities with Takahashi, Hideaki, and Kitawaki's Perceptual QoS Assessment Technologies for VoIP.

ITU P.563 calculates call quality passively and calculates an R Factor that can be used to estimate a MOS score.

Using the PESQ and PAMS call quality measurement methods requires a license from Psytechnics a spinoff from British Telecom.

Newer versions of Cisco IOS has built-in tools for measuring call quality metrics including estimated MOS scores for test calls. A detailed description of their methods is in this document: Service Assurance Agent (SAA) VoIP Proactive Monitoring

For products to measure call quality see: How To Debug and Troubleshoot VOIP

Automatic Call Distributor

September 7, 2015, 11:11 am

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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 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. ...