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Bicom Systems

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Bicom Systems provides the Communicating World with the most Complete Turnkey Communication Systems available by Creating, Unifying and Supporting the Most Advanced of Current Technologies.

History


Bicom Systems Ltd. was founded in 2005 to exploit its PBXware product.
PBXware was the first Commercial Turnkey Telephony System to use Open Source software including Asterisk.

Among the first customers to use PBXware was Redhat. The business model of Bicom Systems does hold similarities to Redhat in the manner by which it wraps Open Source software in a professional and charge for model, warranted to work.

In 2008 Bicom Systems delivered a custom built conferencing solution to NASA to facilitate the holding of scientific study groups such as the Inter Planetary Conference.

In 2009 Bicom Systems launched its Multi-Tenant Edition of PBXware.

In 2010 Bicom Systems began a relationship with NEC to provide a hosted Telephony Platform to businesses across Australia.

Bicom Systems published How to Grow an ITSP.


Products



Hosted PBX

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What is Hosted PBX?

Unlike a traditional PBX, or Private Branch Exchange, which requires a large investment and ongoing maintenance and training, a hosted PBX is a cloud-based PBX system accessible via an IP network. Rather than being responsible for hardware, software, training, maintenance, and more, a hosted PBX provider takes care of it all. In addition to being completely managed off-site, resulting in no IT or installation costs, a hosted PBX system also provides businesses with the ability to manage their phone systems via a user-friendly control panel. For these reasons and more, hosted PBX systems are becoming increasingly popular solutions for today’s growing small to medium-sized business owners.

Features and Benefits of Hosted PBX

If you own a business with fewer than 300 lines, hosted PBX can provide you with a multitude of features and benefits. For starters, a hosted PBX system is much less expensive than a traditional on-site system. With hosted PBX, there is no need to buy expensive hardware and software, pay for installation, and manage the system. You simply pay a monthly fee and enjoy all of the benefits of the service.

While less costly than standard on-site PBX systems, hosted PBX solutions are teeming with valuable features, such as on-hold music, call waiting, call routing, transfers, and more. Moreover, as the popularity of hosted PBX continues to grow, additional features like auto attendants, extension dialing, and ACD queues are being introduced.

A hosted PBX can also be deployed immediately. In fact, most business owners are able to have their hosted PBX setup and running in under a day. Best of all, newer and more flexible hosted PBX applications are being deployed as well. Adding these applications and features can typically be done with a simple download or click of the mouse.

The ability to add new features as they roll out is just one example of the scalability of hosted PBX. Additional lines, phones, and even an entire new department can effortlessly be added, which would be much more complex with a traditional on-site PBX.

When you opt for a hosted PBX and outsource all the techie stuff, you will find your stress levels lowering as well. With a hosted PBX, a mountain of responsibility will be taken off your shoulders, allowing you to focus your attention on more important matters, like improving your bottom line.

Hosted PBX Tips and Considerations

Hosted PBX systems are incredibly easy to setup and utilize, but there are a few things you need to do when opting for one of these modern business phone solutions, such as:

  • Make Phone Number Arrangements – A temporary phone number is necessary for the porting process, but if you already have an existing number to port over, you must remain with your current provider until the number is ported to your new PBX provider. If you are getting a new number, it’s important to update others and let them know of the change through an email blast or similar means. While brief, all calls can be conveniently routed to cell phones during the transition period. Just remember, if you cancel your service with your previous carrier prior to porting your number, the number will no longer be yours to keep.
  • Arrange Your Dial Plan – When switching over to a hosted PBX, organizing the routing of your calls is one of the first things that must be done. In order to do so, you’ll need to define various rules for your calls, including:
    • The buttons used to activate voicemail and other features
    • The hours of operation for your desk phones
    • Setting up your directory
    • The handling of faxes
    • Sequential and simultaneous ringing
    • How off-hour calls are handled
    • Configure your emergency 911 settings

Review Your Bill

As you can see, setting up a hosted PBX is remarkably easy. However, after doing so, you should review the bills for the first and second month to ensure you’re receiving and utilizing all of the services you’re paying for. Depending on which company you choose as your hosted PBX provider, the first month’s bill may or may not reflect activation fees, setup fees, and number porting fees. The second month’s bill should be a typical bill without added charges for the setup process. By this point, you'll be well on your way to enjoying one of the most streamlined and hassle-free business phone systems available for small and medium-sized business owners. ...

softswitch

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Softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose system. Most landline calls are routed by purpose-built electronic hardware however, soft switches using general purpose servers and VoIP technology are becoming more popular.
Nowadays, many telecommunications networks make use of combinations of softswitches and more traditional purpose-built hardware.

A softswitch is also a VoIP server, providing a soft switch platform with full IP PBX call features. The most difference from IP PBX is its enormous numbers of users.

Typical application networking diagram
Typical softswitch application.jpg


See also


Softswitch Manufacturers and Providers




Sonus Softswitch

Sonus is one of the pioneers of Softswitch technology, delivering it's first version PSX6000 in 2001. Since then Sonus established itself as a major global supplier of carrier-grade, highly scalable solutions.
- Proven performance and scale even under the most adverse conditions
- Complete network protection against Denial of Service (DoS) and Distributed Denial of Service (DDoS) attack
- Robust features expected in a high-end softswitch, including media transcoding, encryption/authentication, centralized policy management and broad signaling interworking

Cloud Sonus Softswitch is offered by R&R Managed Telecom Service



Vox Switch
Vox Switch is carrier grade softswitch platform from Voxvalley Technologies that is soon becoming a favorite choice for service providers and carriers across the globe. Vox Switch facilitates service providers to carry out routing, billing, reporting & monitoring of their VoIP services.
Vox Switch platform is flexible to allow hassle free migration from other softswitches without any data loss.
Get a Vox Switch Free Quote

Vox Switch includes numerous advanced features
● Multi-level reseller user management.
● Multi-branding support.
● DID Management.
● Tariff plan, Balance, Payment and Voucher management.
● Advanced Routing.
● Invoicing & Billing.
● Real-time Reporting.
● IVR Routing.
Get a Vox Switch Free Quote




iTel Switch
iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services. ...

PHP Code for provisioning and rebooting Grandstreams

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Based on the current examples with Bash scripts, Perl and LUA code on the site, I decided to contribute some PHP snippets which provide various functions:

Update: According to my research, the reboot and provisioning methods used for the GXP2120 and GXP2140 should also apply to the GXP2100 and GXP1200 phones.

Reboot for GXP2120 and GXP2140:



	$pass = "admin";
	// generate URLs
	$login_url = "http://" . $ip . "/cgi-bin/dologin";
	$reboot_url = "http://" . $ip . "/cgi-bin/api-sys_operation";

	// curl object
	$ch = curl_init();

	// set curl to use cookie file
	$ckfile = tempnam("/tmp", "CURLCOOKIE");
	curl_setopt($ch, CURLOPT_COOKIEJAR, $ckfile);

	//set the url, number of POST vars, POST data
	curl_setopt($ch, CURLOPT_URL, $login_url);
	curl_setopt($ch, CURLOPT_POST, 1);
	curl_setopt($ch, CURLOPT_POSTFIELDS, "&password=" . $pass);
	curl_setopt ($ch, CURLOPT_RETURNTRANSFER, true);

	// get 'sid' as a string
	$response = curl_exec($ch);
	preg_match('/sid" : "([A-z0-9]*)/', $response, $sid);
	$sid = $sid[1];

	// now that we have the sid, send the reboot command
	curl_setopt($ch, CURLOPT_URL, $reboot_url);
	curl_setopt($ch, CURLOPT_POST, 1);
	curl_setopt($ch, CURLOPT_POSTFIELDS, "&request=REBOOT&sid=" . $sid);
	$response = curl_exec($ch);
	curl_close($ch);


UPDATE 2017/03: as of recent firmware builds, the following GET request appears to be enough


		$pass = "admin";
		// generate URLs
		$login_url = "http://" . $ip . "/cgi-bin/dologin";
		$reboot_url = "http://" . $ip . "/cgi-bin/api-sys_operation";

		// http://xx.xx.xx.xx/cgi-bin/api-sys_operation?password=XXXXX&request=REBOOT
		$reboot_url = "http://" . $ip . "/cgi-bin/api-sys_operation?password=$pass&request=REBOOT";

		// curl object
		$ch = curl_init();

		// Set some options - we are passing in a useragent too here
		curl_setopt($ch, CURLOPT_RETURNTRANSFER, 1);
		curl_setopt($ch, CURLOPT_URL, $reboot_url);
		curl_setopt($ch, CURLOPT_USERAGENT, 'GSProv cURL');

		// send GET request
		$response = curl_exec($ch);
		curl_close($ch);


Reboot for the GXP2000:



	$pass = "admin";
	// generate URLs
	$login_url = "http://" . $ip . "/dologin.htm";
	$reboot_url = "http://" . $ip . "/rs.htm";

	// curl object
	$ch = curl_init();

	// set curl to use cookie file
	$ckfile = tempnam("/tmp", "CURLCOOKIE");
	curl_setopt($ch, CURLOPT_COOKIEJAR, $ckfile);
	//set the URL, number of POST vars, POST data
	curl_setopt($ch, CURLOPT_URL, $login_url);
	curl_setopt($ch, CURLOPT_POST, 1);
	curl_setopt($ch, CURLOPT_HEADER, 1);
	curl_setopt($ch, CURLOPT_POSTFIELDS, "P2=" . $pass . "&Login=Login&gnkey=0b82");
	curl_setopt($ch, CURLOPT_RETURNTRANSFER, 0);
	curl_setopt($ch, CURLOPT_FOLLOWLOCATION, 1);
	curl_setopt($ch, CURLOPT_MAXREDIRS, 3);
	curl_setopt($ch, CURLOPT_REFERER, str_replace('dologin', 'login', $login_url));
	curl_setopt($ch, CURLOPT_USERAGENT, "Mozilla/5.0 (Windows; U; Windows NT 5.1; en-US; rv:1.9.2.13) Gecko/20101203 Firefox/3.6.13 ( .NET CLR 3.5.30729)");

	// get 'sid' cookie by logging in
	$response = curl_exec($ch);
	// now that we have the sid cookie, send the reboot command
	curl_setopt($ch, CURLOPT_URL, $reboot_url);
	curl_setopt($ch, CURLOPT_POST, false);
	curl_setopt($ch, CURLOPT_HEADER, 1);
	curl_setopt($ch, CURLOPT_COOKIEFILE, $ckfile);
	curl_setopt($ch, CURLOPT_RETURNTRANSFER, 0);
	curl_setopt($ch, CURLOPT_FOLLOWLOCATION, 1);
	curl_setopt($ch, CURLOPT_MAXREDIRS, 3);
	$response = curl_exec($ch);
	curl_close($ch);



Provisioning (plain XML) for GXP2120 and GXP2140:

-- This example assumes you have a database with tables named after the configuration parameters. If not, substitute the database part with reading a textfile of params instead or get creative...


	// clear any previous output
	ob_end_clean();

	// set headers for filetype
	header("Content-type: text/xml");
	header("Content-Disposition: \"inline; filename=cfg" . $mac . ". ...

WebRTC

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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.

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

customLogo.gif.png

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:
WebRTCpublicdiagramforwebsite (2).png


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. ...

VOIP Event Calendar

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May 2017


  • 2-5: AMSTERDAM Opensips three exciting days filled with VoIP and RTC presentations, workshops and design clinics bringing the latest updates from the OpenSIPS community
https://www.opensips.org/events/Summit-2017Amsterdam.html

  • 8-10: BERLIN - Kamailio World Conference & Exhibition - real time communications event with sessions covering SIP, VoIP, VoLTE/IMS and WebRTC - https://www.kamailioworld.com


  • 17-18: LONDON Discover the solutions designed to provide flexible ways of delivering, managing and supporting communication that include major UC&C technologies: Cloud, Mobile, Customer, Video, Networks, Collaboration.
http://www.ucexpo.co.uk

November 2016

  • 16-17: VoIP2DAY 2016 | Santiago Bernabéu (Madrid) - Discover the european meeting of Telephony and VoIP Communications that brings together each year the main international industry leaders. Enjoy two full days of papers and workshops featuring the most important professionals from international scene. More information at http://www.voip2day.com/en/2016

October 2016


September 2016

  • 26 – 27 TELECODAYS Dubai - TelecoDays Dubai is an annual global telecom event bringing together wholesale voice, mobile, data carriers, SMS companies, MVNOs and telecom solutions vendors from around the world.

Asterisk-based commercial PBX

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Here is a list of producers of ready-made, black box PBXs that are based on Asterisk (in no particular order):

Bicom Systems


Bicom Systems is the creator of Telco-in-a-Box, the most complete IP PBX Unified Communications solution for ITSPs, Service Providers, CLECs, and more. Telco-in-a-Box meets all of your telecom needs including telephony, billing, mobility, security, and more in one tidy package. With unmatched compatibility, stability, and reliability, Telco-in-a-Box is the tool to build and grow a telecom. The product suite includes:

  • SERVERware is a Cloud IP Services Delivery Platform
  • PBXware is an IP-PBX turnkey business communications platform
  • TELCOware is a provisioning platform that performs billing, accounts, and more
  • sipPROT is an advanced SIP security module
  • sipMON is a SIP monitoring module that analyzes and improves VoIP calls
  • gloCOM is a desktop and mobile Unified Communications app

A Zycoo Asterisk 13 System



A Zycoo Asterisk V.13 base VoIP phone systems is user friendly and license free IP PBX system, includes optional telephony interfaces: PSTN, GSM, WCDMA, E1/T1, ISDN BRI. UC Pro& CooCall Softphone expand user's mobility, security, productivity and collaboration.

Visit our website Zycoo to drop us an email or give us a call for more information.

  • VoIP Phone Systems
    • CooVox-U20 2FXO/FXO+FXS/2FXS/1GSM/2xISDN BRI IP PBX, 30 users & 15 concurrent calls.
    • CooVox-U50 PSTN, GSM, WCDMA IP PBX, 100 users & 30 concurrent calls.
    • CooVox-U80 PSTN, GSM, WCDMA, E1/T1, ISDN BRI IP PBX, 200 users & 60 concurrent calls.
    • CooVox-U100 PSTN, GSM, WCDMA, E1/T1, ISDN BRI IP PBX, 500 users & 100 concurrent calls. ...

Panamax Inc.

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Panamax Inc. is successfully scaling the international markets and helping telcos, carriers, MNOs & MVNOs to do business with advanced technologies. Our Business Support System (BSS) product BillCall extends converged & interconnect roaming solutions with complete reconciliation and settlement features. It also offers fraud management and revenue assurance solutions that are designed to overcome fraud and losses due to faulty billing systems and scams. The BSS retail solution Bqube is a carrier grade billing, customer care and provisioning platform that enables data, voice and video services over wired and wireless networks. Our Network Switching Subsystem (NSS) product iMax is a class 4 integrated carrier switch and session border controller while Bridge2Call is a class 5 enterprise retail switch with a communication platform to deliver VoBB, hosted PBX, Unified Communication, calling card, contact center, contact center solutions and DID retail services. Mobile financial solution (MFS)product MobiFin enables telcos to roll out and manage mobile money services, microcredit, bill payment, international remittance etc. Banks can also deploy MobiFin to offer services like mWallet, mobile banking, subsidy distribution and agent/merchant Payments. Panamax has earned recognition for our disciplined efforts in bringing innovative technology to the global platform. To know more visit https://www.panamaxil.com/



Plantronics MX500i 3-in-1 Headset

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Featuring hands-free versatility and professional-class performance, the MX500i 3-in-1 VoIP Headset lets you make Skype and other Internet calls through your computer, or make calls with your cordless phone or cell phone. Ideal for on-the-go home office professionals, the MX500i offers advanced sound technologies, such as WindSmart microphone technology for reduced wind and fast-motion noise, giving you the professional-class call clarity that your work demands.

See Also

VoIPSupply.com

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About VoIPSupply.com


VoIPSupply.com is the retail website of North America's leading VoIP solutions provider, VoIP Supply. VoIPSupply.com offers a fast and easy way to find the VoIP hardware you are looking for.

VoIPSupply.com has products for consumers, small medium businesses, enterprises, government/educational entities, resellers and service providers. You can visit VoIPSupply.com Here.

About VoIP Supply


VoIP Supply is North America's leading VoIP solutions provider. Since 2004, VoIP Supply has delivered valuable solutions for some 60,000 customers worldwide.

With over 30 passionate employees, 2,500 products, 40,000 square feet of office space and an unlimited number of VoIP solutions to meet your needs VoIP Supply has everything you need for VoIP, whether you are a consumer, business, service provider or reseller.

VoIPSupply.com Offerings


VoIPSupply.com specializes in standards based VoIP equipment and VoIP systems. In addition to complete VoIP systems, VoIP Supply offers:


Besides VoIP equipment and complete Business Phone Systems, VoIP Supply offers a number of value-added services to consumers, businesses, resellers and service providers:

  • Full online catalog at VoIPSupply.com
  • Network assessments
  • Solution design
  • Device configuration, set-up and installation
  • On-going maintenance and support
  • Device fulfillment (privately branded)
  • Extended no-questions asked warranties

Upcoming Events


Your Favorite VoIP Manufacturers


VoIP Supply makes it easy for you to find products from your favorite manufacturers.

For your convenience, some popular manufacturer categories include:

GSM Termination - SIM Server

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What's a SIM Server for GSM Termination?

A SIM Server solution separate the SIM cards from GSM terminals and allows a total control of the SIM cards and GSM channels.

Who offers VoIP SIM Server?


See especially (alphabetic order):

================================
2015-03-05_13-32-07.png

ANTRAX solution

SIM Server& SIM box

MAIN FEATURES:
SIM box includes the following elements:
Sub-rack with fifteen universal slots which provides power and cool the system
SIM-board that comprises 20 SIM cards
Each client has opportunity to choose individual structure of SIM box and use from 300 SIM cards during GSM termination. It is possible to combine GSM and SIM boards in one rack, so it is gsm sim box. ...

SD-WAN Vendors

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Getting the highest bandwidth possible is a network user's dream. SD-WAN works by delivering better connectivity, minimal complexity, and lower costs to the wide area network. Other than sellers producing SD-WAN products, there are resellers specializing in SD-WAN solutions who help customers select the platform that works best for business requirements, traffic mix, and application deployment model. SD-WAN is specifically used to link network enterprises that are located over great distances. Within these circumstances, SD-WAN is considered to be the most efficient solution to route traffic to distant areas.

The SD-WAN Solution

Diverse types of tangible WAN links can be implemented to transport traffic with little effort required by the network engineer. SD-WAN travels parallel above the tangible infrastructure which extracts the actual links. Read more about SD-WAN.

How the SD-WAN Process Works

Each physical link's performance is monitored by SD-WAN forwarders. Next, individual flows are transmitted to distant SD-WAN forwarders passing over the link that will efficiently manage that flow's specific SLA needs.

How Are SLA Requirements Defined?

SLA requirements are defined by a network operator in a central controller which is passed on as a principle of action to every SD-WAN forwarder. The benefit? A company can reduce expenses on non-public WAN links by tallying more affordable broadband links to the WAN conglomeration.

List of SD-WAN Vendors

Below is a list of SD-WAN Vendors. Feel free to add to it, but keep it in alphabetical order. ...

IP PBX

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IP PBX is a phone system that utilizes IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.

VoIP Softphones

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What is a VoIP Softphone?

A VoIP softphone is a program that installs to and runs from your computer. A VoIP softphone enables you to make calls with just your computer using a VoIP service. Skype, iChat, and GoogleTalk are some of the more popular services, but there are many different ones available for you to choose from.

A VoIP softphone is accessible wherever you have a computer with you and there are both paid and free VoIP softphones. Companies often offer their own proprietary softphones that are configured to work with their service like Cisco and Counterpath.


The VoIP softphones are designed to be intuitive to use and most resemble an actual phone handset. Or you can choose to have the layout show your contacts if that makes calls easier. You can either click the buttons on the interface to dial or use the number pad on your keyboard.

VoIP softphones are only a program on the computer, so a headset with a microphone or an internal microphone and speakers are also needed to make the calls. Headsets prices start at a very affordable $5-$10 for a standard configuration and can get up in the $100-$200 area for a high-quality wireless headset with a long battery life and interchangeable ear pieces.

Who Can Benefit From VoIP Softphones?

Softphones can be used by anybody with a computer. There are a few types of users who can really benefit from the features of a VoIP softphone:

  • VoIP beginners
  • Heavy travelers
  • Telecommuters
  • Call center employees
  • Small businesses
  • Frequent long-distance callers

Find out How Softphone can boost your business


VoIP beginners can quickly and cheaply explore how the service works by downloading a VoIP softphone to make free computer-to-computer or computer-to-phone calls. Heavy travelers can avoid racking up large bills on their mobile phones or at hotels by using a low-cost VoIP service with a VoIP softphone. Telecommuters can register a VoIP softphone with their office PBX system to enjoy the same call features available to them at the office while they are on the move. Call center employees and small businesses can save on costs by pairing a VoIP service with a softphone to avoid purchasing and maintaining desk phones. International and long-distance rates are much lower when using a VoIP softphone, so those making regular or frequent calls out-of-state or country can cut some major costs.

VoIP Softphone Features

VoIP softphones offer the same features that traditional phones offer and more:

  • Call forwarding
  • Call conferencing
  • Hold capabilities
  • Call transferring
  • Voicemail
  • Greeting capabilities
  • Text, IM, and video capabilities
  • Echo cancellation to improve sound quality
  • Contact list/address book


Softphones also use less energy than phone and phone system hardware, which saves on costs and is useful for green companies.

VoIP Softphone Protocols

VoIP service uses different protocols to determine how the data is processed and transferred over the network. ...

WebRTC

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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.

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

customLogo.gif.png

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:
WebRTCpublicdiagramforwebsite (2).png


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. ...

Mobile VoIP

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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.

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. ...

VoIP Termination

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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.



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Plantronics MX500i 3-in-1 Headset

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Featuring hands-free versatility and professional-class performance, the MX500i 3-in-1 VoIP Headset lets you make Skype and other Internet calls through your computer, or make calls with your cordless phone or cell phone. Ideal for on-the-go home office professionals, the MX500i offers advanced sound technologies, such as WindSmart microphone technology for reduced wind and fast-motion noise, giving you the professional-class call clarity that your work demands.

See Also

Mobile VoIP

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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.

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. ...
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