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SS7 Protocol Converters and Gateways

July 13, 2015, 5:38 am
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Convert SS7 connections to other connection types (e.g. PRI) to make interconnecting SS7 networks and non-SS7 equipment possible.

Products and Services

  • SS7 TDM E1 ISDN R2 T1 Signaling/SIGTRAN gateways converters. www.acecomnet.com mervinteo@acecomnet.com
  • Aculab - GroomerII media and signalling gateways for IP and TDM protocol conversion
  • Aculab - ApplianX SIP to TDM gateways
  • AudioCodes Enhanced SS7 Termination Solutions
  • Cosmos SS7-SIP Gateway Turn-key, high performance SS7/C7 solution from Cosmact. Scales from 1E1 to 64E1 per chassis. Supports single point code across multiple chassis. Flexible built in routing/digit manipulation rules for SS7/SIP conversion.
  • Dialogic IMG1010 Any-to-Any gateway that supports both media, signaling and transcoding in a single chassis
  • Dialogic Integrated Media Gateway, IMG1010.
  • Cisco PGW-2200 SS7/C7 Solution
  • Excel Switching (now Dialogic ) SS7 Gateways and Service Development platforms
  • Janus Systems High-density SS7/ISDN/CAS/SIP Signaling Converter
  • OpenCloud IP to IN translator: provides protocol conversion of IMS SIP to an SS7 protocol, such as INAP and CAMEL Phase 3.
  • PATTON Electronics - SmartNode 10K TDM + VoIP SmartMedia Gateway facilitate carrier-paced transition to all-IP architecture. Supporting concurrent TDM POTS/ISDN and NGN VoIP protocols, Patton’s commercial-grade SN10K gateways connect traditional SS7 edge, core and peer zones with new-generation MEGACO and SIP networks. FMC - By supporting such mobile and IP vocoders as AMR, AMR-WB (G.722.2), GSM-FR/GSM-EFR, EVRC/QCELP, G.728, G.729eg, and iLBC, with independent dynamic codec selection per channel, the SN10K enables fixed-mobile convergence (FMC). VAS - Scaling from four T1/E1 trunks up to a full STM-1, SN10K solutions enable customized ringback, background audio, number spoofing, ads-for-minutes, and other value-added services (VAS).
  • RR Managed Telecom Services -
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SS7

July 13, 2015, 5:44 am
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Common Channel Signaling System No. 7 (aka SS7 or C7) is a global standard for telecommunications defined by the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T). The standard defines the procedures and protocol by which network elements in the public switched telephone network PSTN exchange information over a digital signaling network to effect wireless (cellular) and wireline call setup, routing and control. The ITU definition of SS7 allows for national variants such as the American National Standards Institute (ANSI) and Bell Communications (Telcordia Technologies) standards used in North America and the European Telecommunications Standards Institute (ETSI) standard used in Europe.



Reference pages


SS7 Forums

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VoIP Origination

July 13, 2015, 12:04 pm
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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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Hosted PBX Vs On Premise PBX

July 13, 2015, 12:16 pm
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There are pros and cons of both hosted PBX as well as on-premise PBX. There are some fundamental differences to each of the systems and they feature advantages that should be known prior to making a decision on one or the other. The move to an IP-PBX business phone system is beneficial regardless of which system is ultimately chosen. However, there are differences and knowing those leads to a better VoIP phone system and a higher level of satisfaction for the company, employees and even callers.

Open source on-premise systems, such as Asterisk, are responsible for driving down costs for VoIP providers and manufacturers. This provides users with the ability to get the latest technology with many more features at a lower cost than what was traditionally available.

What is Hosted PBX?


Hosted PBX or hosted VoIP, otherwise known as an Internet phone system is one where the provider is responsible for housing the IP-PBX as well as handling the technology required to provide the services to the phone system. The desk sets will plug into a router and the calls, signaling, and features are handled through an IP-PBX server at the provider’s location. The provider of the hosted PBX charges a monthly fee that is inclusive of a minutes package and potentially certain features. Charges can also be at a per minute calling cost. Either one can be affordable depending on the rates. A company that knows the amount of minutes spent on the phone in a given month can make effective cost comparisons. Extended features may come with additional cost.

What is On Premise PBX?


On-premise PBX is also known as an IP-PBX phone system. It is similar to a traditional PBX system that resides at a location, such as a computer equipment room or phone closet. The main difference is that IP routing is done with more current technology. The signaling is done with an IP phone to the IP-PBX server using a LAN. Calls can go through a traditional phone company as well as voice over Internet (VoIP) using SIP trunking. Gateway cards are used to connect the system to the traditional phone company provider. The provider can be the one that already provides service, though a SIP trunk can be configured for use with an Internet service telephone provider (ISTP). An Asterisk based system is the most affordable option for on-premise PBX due to the flexibility that is offered with open source software.

Hosted IP-PBX versus On-premise IP-PBX


There are some differences between the two options. Understanding benefits and limitations makes it easier to determine the best option for any particular organization. Cost, expansion, and other considerations are laid out to make it possible to compare the hosted IP-PBX and the on-premise IP-PBX within the same categories to learn of the greatest differences.

Costs


Purchasing an on premise IP-PBX phone system involves buying hardware, which includes a server with the proper number of interface cards (if needed) to be able to connect the telephone company with the IP phones. Hosted IP-PBX only involves purchasing IP phones, though a router and network switch may be needed to ensure there is one specifically dedicated to VoIP.

Hosted PBX:

  • Lower initial equipment cost and set-up cost
  • Network qualification is performed by the customer. ...
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VoIP Routes

July 13, 2015, 12:16 pm
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VoIP Routes are similar to regular telecom routes, except they are designed for use with VoIP telephone systems. VoIP packets can travel over the Internet over VoIP routes, which essentially lets the VoIP call bypass the traditional telephone routes.

Check out the table on the right side of this page to find providers of VoIP routes. You can also browse our VoIP routes forum to buy and sell VoIP routes and minutes.

VoIP Routes Providers


VoIP Routes testing services


See also


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VOIP Service Providers

July 13, 2015, 12:19 pm
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For a list of VOIP to PSTN service providers, indexed by country, please see:


VoIP and VoIP Service Providers

What is VoIP?

VoIP (which stands for "voice over internet protocol" and is commonly referred to simply as an internet phone) is a highly cost effective and reliable way for businesses or even homeowners to make calls across the world or even just across town. The majority of major cable companies that offer bundled internet, television, and phone services already utilize this newer technology, but there are tons of other independent companies that specialize in providing this service to their customers at reasonable rates and with tons of extra features.

More Than Just Computers

When people think of VoIP, they generally think of computers due to the popularity of the numerous free communication services like FaceTime and Skype, but this is truly just one aspect of what VoIP can truly offer. It is true that VoIP technology transmits voice communication that's been converted into digital data across a packet-switched network or the internet (what this means, in essence, is that a user making phone calls over high speed internet lines rather than phone lines). With that in mind, users are not confined to only using it on a computer. VoIP technology can connect through the internet using traditional telephone equipment just like a regular line. The phone itself is connected to the internet using an adaptor that's plugged straight into a home or business's internet network. Most major services offer a softphone option as well, which allows the user to use their computer directly as a telephone service. In addition to all that, VoIP providers will generally also offer mobile or tablet apps that allow their customers to make calls on the device (assuming it's connected to Wi-Fi at the time). ...
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DID Service Providers

July 13, 2015, 12:24 pm
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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

SMS enabled DID Providers

  • MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.
  • SIPMarket is providing DIDs in more than 150 countries. Direct numbers with incoming SMS available. Incoming SMS is free.


Algeria

  • Incorpus TeleNetworks Incorpus provides DID of 50+ countries. Just visit out website and live chat with us for details. Cheap DIDs available at low costs and discounts for bulk orders. No per minute charge. Only monthly and go on
  • CarryMyNumber.comAlgeria DID /Virtual Phone Numbers at _wholesale rate@$ 4/month with free PBX with the Free forwarding to SIP, Softswitches ,Trixbox ,Asterisk or VOIP. Phone Numbers from over 70 countries available. Free PBX . Unlimited Channel numbers for call centers /Calling Card Providers__. Largest FootPrint worldwide. No Per Minute charges.
  • BuyDDINumbers.com Provides Cheapest Algeria DID /Virtual Phone Numbers/DDI Numbers @_€ 6.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , iTalkWorld, Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.
  • DIDx.net Algerian | Algeria Virtual DID numbers whole sale pricing check out today.
  • DomesticNumbers.com Algerian Virtual Phone Numbers € 7.50/month including free PBX. Forwarding to Skype, Google Hangouts, FreePBX, Asterisk, Voipbuster or other VoIP provider, PSTN, or with free sip account.
  • BuyDIDNumber We Provide Algeria Virtual Phone Numbers@ $ 7.99 / Month NO SETUP FEE , UNLIMITED CHANNELS available with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk , voipbuster , iTalkWorld ,any Betamax Delmont Voip or any other ITSP . Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments. ...
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voip-info.org

July 13, 2015, 10:00 pm
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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


News Resources


Getting Started

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Session Border Controller

July 14, 2015, 9:04 am
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A Session Border Controller is a device used in select VoIP networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The SBC enforces security, quality of service and admission control mechanism over the VoIP sessions.

The Session Border Controller is often installed in a point of demarcation between one part of a network and another. Most Session Border controllers will be installed between peering service provider networks, between the enterprise network and the service provider network, or between the service provider network and residential users.

A Session Border Controller is like a Firewall for VOIP.
They are often configured as a SIP Back-To-Back User Agent (See SIP RFC).

In addition to firewall functions they also may provide services like NAT traversal.

See: NAT and VOIP for more information and watch the video on The Anatomy of Session Border Controllers and To Couple or Decouple Routing Intelligence from SBC.

Martyn Davies of Dialogic discusses Session Border Controllers as the often misunderstood "black magic" application for VoIP networks.

Although carriers often use Session Border Controllers for signal translation and security, most do not include the hardware-based signal processing needed for media transcoding. For all-IP environments, new elements are required that can mediate signaling, transcode among different media formats, and handle basic security issues. The concept of Multimedia Border Element (MMBE) meets these needs.

Products


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VoIP Gateways

July 14, 2015, 9:41 am
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Information about VoIP gateways, including VoIP media gateways, FXO gateways, and other VoIP gateways can be found on this page. If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

Media Gateways

Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone networks and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.

Page Contents


Analog FXO gateways


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PRI Protocol Converters

July 14, 2015, 9:49 am
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If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

PRI Protocol Converters

As the name suggests, PRI protocol converters convert from one PRI protocol to another and back, for example, between T1 and E1, or between J1 and T1/E1.

J1 (INS1500) protocol converters

in alphabetical order


T1/E1 protocol converters

in alphabetical order
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Call Center Software

July 14, 2015, 2:57 pm
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Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

Types of Call Center Software


ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBXauto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

Contact Center Software

For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

Call Center Software Providers

This is a list of call center software providers and developers. Please keep this list in alphabetical order.

  • Ameyo Contact Center Software is an all-in-one software based communication solution that manages end-to-end customer journeys and consistently delivers exceptional customer experiences. It is a powerful and highly flexible IP-based Call Center Software platform that lets you have a personalized interaction with every customer across multiple channels, thereby driving customer engagement to a level par excellence. ...
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Predictive dialer

July 14, 2015, 3:43 pm
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What Is A Predictive Dialer?


  • "A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers." - Wikipedia

"Definitions of Predictive dialer on the Web:

  • A predictive dialer is an outbound call processing system designed to maintain a high level of utilization and cost efficiency in the contact center. The dialer automatically calls a list of telephone numbers, screens the unnecessary calls such as answering machines and busy signals, and then connects a waiting representative with the customer.

note: above text may be copyrighted by it's respective owners)

A VOIP Predictive Dialer, a.k.a. soft predictive dialer, is a software product capable of predictive dialing using VOIP service directly. Besides computer and internet connection, there is no equipment needed in order to use VOIP predictive dialer.

Software Only Predictive Dialer

New predictive dialing technology, together with faster computers and bigger broadband bandwidth, enables software only predictive dialers to work as good as or even better than hardware based dialers. Software based solution avoids expensive telephony board and associated hardware maintenance cost. It is easy to install and configure. For example, it is very easy to setup remote agent (at home agent).

SOFTWARE ONLY PREDICTIVE DIALER

  • Unique Call center Suite Predictive/ Progressive/Auto Dialer and Web CRM Solution. For more details contact: avil@uniqueperform.com

  • Ameyo Predictive Dialer Ameyo Proactive Outbound is equipped with capabilities that power multi-channel campaigns and proactive outbound communications across multiple channels (voice, SMS, social, and web). Ameyo dialer is the heart of proactive outbound communications, that offers outbound capabilities that ensures communications compliance with legal regulations and harbours loyal and long-term relationship with customers.

  • Texo.cc auto dialer is an Agile Customer Interaction hub on cloud, enabling you to build Personalized engagements across the customer journey. Texo.cc empowers users to quickly innovate with minimal technical Intervention, thereby challenging the status quo of traditional inbound, outbound, or blended contact centers. Contact Texo.cc for the best voip predictive dialer. The Texo.cc auto dialer has been trusted by leading outbound sales teams worldwide, checkout Texo.cc predictive dialer reviews for more.


Aavyukta Intel e Call Dialer and VoIP Solutions to Call Centers: Predictive Dialer (Unlimited Seats) + VoIP (US/UKLL/Canada) + Hosted/Cloud Server @ 1 US cent/Min, Reach us on www.dialerphilippines.com or catch us on skype on id avyukaindia +91-9549999916

  • 3CLogic Cloud-Based Contact Center Solutions - No hardware is required. Agents can be working from home or multiple locations. Automatically initiate contact with the next prospect before a rep finalizes a call, reducing call center queue times and operational costs. With 3CLogic, you will have complete visibility into call center operations including advanced scripting and reporting.
  • Astral Predective Dialer Solutions from
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Free VoIP Networks

July 15, 2015, 5:33 am
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Whats a Free VoIP Network?
Free VoIP Networks are based on the SIP.EDU project, which means:

  • It's a 100% SIP compliant network that allows you to connect with SIP softphones/IP phones/WiFi Phones...
  • It's a network where you dont pay to register and get a SIP URI.
  • Allows you to freely dial people inside your domain.
  • Allows you to freely dial people from other Free VoIP Networks.
  • Allows you to freely receive calls from other Free VoIP Networks.
  • Does ENUM lookups to Arpa, E164.org and nrenum.net before dialing E164 numbers.

Free VoIP Networks available

  • BBee-free calls between BBee-Users, free mobile applications, Free text and pictures Messaging, free calls between iPhone and Android users.
  • MO-Call-free calls between MO-Callers, free mobile applications, Instant Messaging, free calls between iPhone users, callback calls,access numbers...
  • StarTel - StarTel.pt - does ENUM lookups, Instant Messaging, Video, Presence and free PSTN calls to 30 countries
  • VoipGate - IAX2, SIP, Enum, various option......
  • OnSIP - Free SIP to SIP calls, free extension to extension dialing. Unlimited free users and and extensions. Loads of free features including time based routing. Built on open standards means OnSIP works with any SIP compliant phones and networks. WebRTC and SIP over WebSocket support.
  • TalkBD - SIP only. Free SIP phones.Voice mail,Conference
  • Free SIP providers - Free SIP providers. Review of free sip services.
  • CloudNumbers - Free calls between CloudNumbers customers and to other users on the TeleWare platform (deposit required for outbound calls). 0844 number allocated for inbound calls. Other numbers available.
  • http://www.vpn3000.com free calls for sip or iax2 between users - Free calls between users using sip or iax2, also allows the use of sipbroker shared dids, use your own device or soft phone. no monthly fees.
  • FreelyCall provides free internet voice and video calls, in addition to low rate local and international calls to regular telephones and mobiles
  • Sipmobile Free calls to USA and Canada. Free US number. Free audio and video calls between Sipmobile users, free calls to Sipbroker networks. Instant Messaging. PSTN access numbers. Cheap international calls to regular telephones and mobiles. ENUM lookups. WebRTC support. Free WebRTC client Webphone. ...
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Optima Saver: Bandwidth Optimization Service for VOIP

July 15, 2015, 5:44 am
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Overview


Optima Saver is a popular bandwidth optimization service for VOIP traders. This service is introduced by ImproLabs Pte Ltd, a Singapore based IT & communications solutions Company. Within a short time, Improlabs has developed a substantial client base all over the world. This was possible due to the continuous effort from the experienced and skilled RnD, development, support and business development teams.

Bandwidth costs money. So the VOIP operators are always looking for the services/software by which they can reduce the bandwidth usage and can terminate more calls in that specific bandwidth limit. This is where the Optima Saver can help the VoIP operators. Using this service, they can reduce upto 80% of their overall bandwidth usage, hence bandwidth cost. As a result they can ensure better performance with much lower costs. They can use this service to provide termination service even on shared internet connections.

Key Features


o Reduces bandwidth usage for VOIP call termination upto 80%.
o Improves phone conversation experience as the voice quality improves.
o Supports major audio codecs like G723.1, G729, AMR, GSM, G711 etc in pass through mode.
o Works with all major SIP termination end points like Quintum, Addpac, Dinstar, GoIP, Cisco, Eurotech gateways etc.
o Users can use Optima from all kinds of removable storages like USBHDD, USB flash drive etc.
o Full featured web based platform is integrated. Users can view CDR, manage capacity and end points.
o Compatible with different kinds of internet networks like broadband, WiFi,WiMAX, 3G etc.
o More reduction in bandwidth usage per call with more connected calls.
o Compatible with public, private, static, DHCP networks.
o It helps to by-pass voice port blocking problem.
o Hides original IP address of the termination provider.
o Works with both real IP and local IP addresses.
o Works from behind the NAT and firewall.
o Reduces Post Dialing Delay (PDD).
o Works with any SIP based softswitch/gatekeeper.
o Flexible and competitive pricing plans.


For more information, please visit www.optimasaver.com and www.improlabs.com.
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Optima Dialer: Revolution in mobile VoIP business

July 15, 2015, 5:44 am
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Optima Dialer is a Mobile VoIP product of Improlabs Pte Ltd. Mobile VoIP refers to the technology that helps the users to make VoIP calls from their mobile communication devices like Smart Phones and Tabs. Optima Dialer is an state-of-art Mobile VoIP App by Improlabs which enables smart mobile devices to make VoIP calls with ease.

It is an app that runs not only on both ARM and X86 architectures, but also in a wide range of mobile handsets. Optima dialer uses SIP signaling protocol and is compatible with all the major SIP supported softswitches. Optima Dialer has the capability to pass through complex network firewalls while used with BytePlex; a revolutionary tunneling product developed by Improlabs.

Optima Dialer ensures enhanced performance and best possible voice quality even on low bandwidth. Only 6 kbps bandwidth is enough for Optima Dialer to deliver crystal clear voice quality. It overcomes the major limitations of the other dialer products in the market. Implementation of latest technology and a lot of useful features make it a must-to-have dialer product for the VoIP calling operators in the today's market.

Key Features


    • Crystal clear voice only @ 6 kbps speed
    • Low bandwidth consumption
    • Most user friendly graphical interface.
    • Faster registration and call connectivity.
    • Display account balance on the dialer screen
    • Supports any SIP softswitch.
    • Ability to bypass network blockage and firewalls while being used with Byteplex module.
    • Integrated with native contact list.
    • Customized branding facility.
    • Ability to work from behind NAT.

Tech Specification

    • Signaling Protocol: SIP
    • Compatibility: All major SIP softswitch.
    • Supported Mobile Platforms: Android, IOS, Windows Mobile

For more information, please visit www.optimadialer.com and www.improlabs.com.
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BrightPattern

July 15, 2015, 1:12 pm
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Our goal at Bright Pattern is to help enterprises to bridge the growing expectation gap between organizations and their connected customers by leveraging customers’ digital lifestyle, while supporting existing processes and driving the costs down.

We designed, architected, and built a cloud based customer engagement solution aimed from inception at satisfying the needs of today’s enterprises, its customers, and its employees.

It features omni-channel interactions (e-mail, text, voice and video), social networking, mobile and context-aware capabilities for next-generation consumer engagement and customer service.

The solution is specifically designed for rapid innovation delivery in a cloud to reflect the ever-evolving customer journey.

In-App Customer Service
Call Center Solution Overview
About Us


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BlackBerry VOIP

July 15, 2015, 7:27 pm
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Mobile VoIP

July 15, 2015, 7:29 pm
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Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

How Does Mobile VoIP Work?


Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

Page Contents


By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

  • Anytime minutes
  • Night or weekend minutes
  • Rollover minutes
  • Roaming charges
  • Incoming call charges
  • Messaging limits
  • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

Accessing Mobile VoIP

Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. (The SIM card contains all the information needed to identify network subscribers. ...
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ZRTP

July 15, 2015, 7:34 pm
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ZRTP is key exchange protocol designed to enable VoIP devices to agree keys for encrypting media streams (voice or video) using SRTP. ZRTP is defined in an Internet draft http://tools.ietf.org/html/draft-zimmermann-avt-zrtp.

The authors of ZRTP describe it as "Media Path Key Agreement for Secure RTP". This means that the ZRTP end points use the media stream rather than the signaling stream to establish the SRTP encryption keys. Many other key exchange protocols use the signaling stream (for example SIP or H.323) for media key exchange. The disadvantage of this approach is that the key exchange is visible to any intermediate device that processes the signaling stream.

ZRTP’s use of the media path for key agreement ensures that media keys are agreed directly between the caller and call recipient and those keys are not visible to any intermediate signalling device. This makes ZRTP an ideal choice for use on networks where signalling is processed by intermediate devices and where it is important to ensure call confidentiality.

Key Exchange

ZRTP is designed to provide a secure method for two VoIP end-point to securely agree encryption keys that are subsequently used to encrypt media streams (voice or video) using SRTP. ZRTP uses the Diffie-Hellman algorithm which enables secure key agreement and avoids the overhead of certificate management or any other prior setup. ZRTP supports two Diffie-Hellman variants, finite field and elliptic curve. The keys agreed by ZRTP are ephemeral which means that they are discarded at the end of a call, avoiding the need for key management.

Man-in-the-Middle protection

ZRTP includes features for both detecting and preventing MitM attacks. MitM is a classic method of eavesdropping on encrypted communications. An attacker intercepts the communication and relays messages between the two end-points making each believe they have a secure channel to the other. ZRTP’s MitM defences include the use of a Short Authentication String (SAS), and Key Continuity.

The SAS is a cryptographic hash of some of the Diffie-Hellman values which is displayed as a word-pair on the user interface of each ZRTP device. The words are selected from the PGP word-list . This list generates 65,356 different SAS values. Users compare the displayed strings by reading them to each other. To remain undetected a MitM attacker would have to guess the correct SAS, there is only a 1 in 65,536 chance of a correct guess. Key commitment adds further defences by re-using some key material in subsequent key agreements. This feature means that a MitM would need to be present on the very first call between any pair of callers.

End-User reassurance

The SAS provides useful reassurance to end-users that they have a secure line. By reading and comparing a word pair, users can be certain that the key exchange has completed.

ZRTP on Mobile Networks

ZRTP’s use of the media stream for key agreement makes it a good choice for use on mobile networks where the network operators process the signaling protocol. A number of implementations are available for Symbian and Windows mobile cell phones.

ZRTP/S for traditional telephony (GSM / ISDN)

ZRTP has been extended by PrivateWave in partnership with Philip Zimmermann to work on traditional telephony data communications (GSM CSD, UMTS CSD, ISDN Data call, SAT CSD, etc) narrowband channels (from 4800bps).

ZRTP Implementations

There are multiple ZRTP protocol implementation such as
  • GNU ZRTP opensource (GPL) c++ implementation of ZRTP with DH key exchange
  • iCall Open ZRTP opensource (LGPL) c++ implementation of ZRTP with DH key exchange
  • M5T ZRTP SAFE is a ZRTP stack implemented independently.
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