The top 8 VoIP courses you need to take

1. VoIP Design and Telephony Fundamentals

4.3

(232)

In the last year I have thought this course twice to companies who were starting a new VoIP project. It is an interesting training for those who have little knowledge of telephony and voice over IP networks.  We start with conventional telephony, covering analog and digital telephony. In the second section we move to VoIP fundamentals covering dimensioning of circuits and bandwidth, codec selection and legacy integration. Then we cover QoS for routers and switches, in this chapter we have used Cisco routers and switches for the examples and demos, but the concepts can be applied to any vendor. When I started with open source PBXs in 2004, one of the things I've noticed is, most people working with open source telephony do not have a background on network design. One of the key factors for the success of proprietary vendors is the enforcement of best practices thru their channel partners. You probably won't see a Cisco VoIP project without QoS enabled, but still in these days, most open source projects do not enforce best practices on dimensioning, Quality of Service and integration with the legacy. That's the gap I want to fill with this course. I believe this training will never be ready, there will always new devices and new technologies to cover. So I plan to update it regularly. As you have lifetime access, you will always receive the updates. Section 1: Analog TelephonyThis section is key. It is very important to understand the concepts of analog telephony, FXS, FXO and E+M interfaces.  How to integrate a VoIP network with the PSTN (Public Switched Telephone Network). Digital TelephonyHere we cover T1/E1 networks, CAS (Channel Associated Signaling) and CCS (Common Channel Signaling). Our focus will be in the standard ISDN (Integrated Services Digital Network) used by almost all countries in the world and also MFC/R2 for CAS, often found in Latin America, Africa and China.  Section 2: VoIP FundamentalsHere we will present the basics of VoIP. Codec SelectionHow to select the best codec for your Job. Channels and Bandwidth DimensioningIn this chapter you will learn how to calculate the number of channels required given the amount of minutes in the busiest hour. Once you have the number of channels we will use the VoIP calculator to define the bandwidth required. How to design for Voice QualityIn this section you will learn how to design and measure the performance of your network. You will understand important concepts related to voice quality such as the R-Factor and MOS (Mean Opinion Score). I will point you some utilities capable to measure the voice quality in your network. Legacy IntegrationOnly in rare situations you will implement a completely new project. Most of the big projects will require you to integrate with the legacy. Legacy large PBXs, PBXs in the branches, survivability and others.  Section 3: In the section 3 we are going to cover QoS (Quality of Service). Classification and Marking Covering CoS (Class of Service), DSCP (Differentiated Services Code Point) and PrecendenceFragmentation, Interleaving and Compressed RTPHow to handle slow circuits using fragmentation and interleaving and how to enable cRTP in PPP circuits. Queueing and Tail DropMost common congestion management and avoidance techniques. We will do a demo for the implementation of Low Latency Queuing on Cisco and AutoQoS on switches. Voice VLANsVoice VLANs are essential for security and broadcast storm protection. One of the best security features for any network were the phones are open to the public such as Hotels and Universities...

2. Asterisk Administrators Guide to VoIP Polycom IP SIP Phones

4.3

(75)

In this course you will learn how to create a configuration for seamless interoperability between a Polycom IP phone and an Asterisk based PBX. Since building an Asterisk server is only half of the phone system you need to ensure the physical phone endpoints are configured correctly. The phones are what all the end users interact with daily and gives them the tool to communicate with internal and external users. You will learn how to configure Polycom IP phonesI will show you the different Polycom IP phone models to choose fromIn the course resources for each lesson you will get copies of the config files I demonstrateIt should only take you a couple hours to get a working configuration file for your PBXIf you plan on using Polycom phone you need to take this course and learn how to configure them using the different methodsMy courses are always structured in an easy to follow format giving you a little bit of knowledge at a time to implement then building on top of that knowledge as the course progresses. I give you real world examples and show you how to do fuctions that are commonly asked for of phone systems...

3. FreePBX Asterisk VoIP PBX Server Administration Step by Step

4.7

(431)

Do you need a cost-effective phone system for business?Are you tired of paying the licensing costs to the proprietary phone system?Do you want to expand your business and require a phone system but legacy systems don't allow you?You are interested to learn asterisk but like to avoid the command line and Linux shell at the start?FreePBX is the world's most popular and widely adopted open-source IP telephony software. The core VoIP communication is based on Asterisk 18 - The most powerful IP telephony platform. This course is designed for the newbies, small & medium businesses that like to use the IP telephony - PBX or even the solution providers that like to gear up for telephony services to the end-users. The course starts with initial telephony concepts and terms used in the phone service industry. No prior technical knowledge about telecommunication is required to take this course. The course has a project depicting a real-world scenario. This helps the students to have activity-based learning and can apply learned knowledge to the real world. The course clearly explained the phone system features with lab practice for configurations. Proprietary systems invite a license fee for most Business-critical features and UC Features like mobility and third-party integration. To avoid the expensive add-ons go the open source way. Use a single system to manage multiple tools and platforms, thereby having "greater employee productivity, reduced costs and a means to improve customer engagement. The course covers the latest release of FreePBX 16...

4. Setup Own Asterisk VoIP Server with Android, iOS & Win Apps

4.3

(104)

VoIP for Dummies - Asterisk VoIP Server setup with Android, iOS, Win Apps - Using Fully Open Source Server and ClientsIn this course, we will setup a VoIP server and the client devices and the clients can make calls in between them using the VoIP server. And no prior experience is required. VoIP or Voice over Internet Protocol is a technology that allows you to make phone calls across devices without using the normal analogue phone connection. So your calls will be placed across internet and the normal phone lines are not required. VoIP allows you to make calls from a computer, a mobile phone which is connected to internet, or a normal phone which is connected to a specific adapter called the 'VoIP Adapter'. The major benefit is that since this call is placed over internet, you don't need a separate line or a dedicated line in-order to make the call. Just an internet connection is more than enough to make a call. If you are a business owner trying to get down the cost of communication at your office, or you are trying to setup a vast call based call centre operation or you are a technical enthusiast who  wants to host your own VoIP server and provide this service to your client users, then this is the course exactly for you. Let me now give you a brief overview of what are the topics that we are going to cover in this course. In the first session, which is basically a theory session, we will be covering the technology behind VoIP. And we will be covering the architecture and the working of VoIP technology compared to the normal PSTN (Public Switching Telephone Network) that has been there from the beginning. And in the next session we will be setting up own own ubuntu based VPS server in an Amazon (AWS) ec2 instance. I am preferring Amazon , because they have a very flexible plan and they have an option called 'Free Tier', which we can use free for one year with a very minimal cost if you are keeping your usage in an optimal state. And after setting up the VPS server, we will installing Asterisk, which is a very popular open source VoIP server software available, and we will be installing it into our VPS server and we will also configure the ports, the specific number of ports that are required for the client devices to communicate with other client devices through the server. And in the next session, which is an important session, in which we will be configuring the dial plan and the extensions that we are going to use with our server. We will also make configuration to accept audio calls as well as video calls through our VoIP server. For our VoIP client applications we will be using an application called Linphone. It is also a completely open source application and the advantage is that it is available for all the platforms. For windows, Linux, Mac, Android and iOS device and the source code is completely open. You can download it and customize it as per your needs. The same configuration settings that you are going to make in this softphone, do the same configuration if you have a physical, hard wired, IP Phone with you. You can enter all these configuration into that and it will also work in the same way as we configure the soft phone clients. And in the next session, we will be configuring a windows based softphone, which is our linphone. We will be installing it in our windows and we will configure the option so that it can register with the server and make video and audio calls in between devices. And later, we will configuring it for Android. The same configurations. We will be installing it from the play store in an android device and we will make the configuration so that video and audio calls can be placed. And after that, we will have it for iPhone We will directly download it from the iPhone App Store and we will be installing it into our iPhone and then we will try to make calls from iPhone to other devices and we will be testing the video calls as well as audio calls across the devices. Learning and becoming an expert in VoIP technology is actually a very rewarding career because VoIP technology is very extensive so far and there is some time in future where we will discard all those analogue telephone lines and we will rely completely on VoIP based IP telephones. Because we need only a single internet channel , rather than having multiple channels. So world is evolving into that kind of a technology and VoIP experts are very much required in the market. And by the end of this course, we will be providing with you an experience certificate (Course Completion Certificate), which you will have great benefits, if you are trying for a VoIP based career. So see you soon in the class room. Let's make our first call! Have a great day!!...

5. Amazon Connect AWS Contact Center VOIP Bootcamp zero to hero

4.2

(127)

Become an AWS Call Center Telecom Engineer and learn one of the employers hard to find requested skills of 2021. This is the most comprehensive, yet straight-forward course for the AWS Call Center Bootcamp from zero to hero on Udemy! Whether you have never programmed a Call Center solution before, already know the basics of AWS Connect or Call Center administration, or want to learn about the advanced Artificial Intelligence (AI) features in AWS Connect, this course is for you! In this course, we will teach you AWS Connect which would help you kick start your career in the AWS cloud. This would also help small, mid, and even large-size companies to retire on-prem call centers and potentially make a move to the cloud call center. Just for a side note Amazon one of the largest organizations on the planet runs their entire call center on AWS connect so if Amazon could do then other companies as well. AWS Scripts and ChatBot Files are included with this course under the Udemy Resource section. We cover a wide variety of topics, including: By the end of this course, you will be confidently implementing AWS Call center solution as an alternative to Cisco, Asterisk, Avaya, Five9, NiceInContact, Aspect, Talkdesk, Genesys, 8x8, Serenova, EvolveIP, Telecom System, and many more. You'll Also Get: Lifetime Access to course updatesFast & Friendly Support in the Q & A section...

6. Learn to Write Your Own Asterisk VOIP Dialplans and Queues

4.4

(244)

Do you want to learn how to read and write your own Asterisk dialplans?Have you been trying to build your own Asterisk auto attendant but don't know where to start?Do you want to create your own custom recordings for auto attendant greetings?Do you want to setup a call center queue with dynamic agents?Are you looking to configure ACD (automatic call distribution) for your inbound phone calls?Learn how to tackle all of these configuration items and more by taking this advanced Asterisk course. During the course you will learn details about how create your own auto attendant and adding different digit options to route calls within your Asterisk phone system. Go from an Asterisk beginner to an expert and write all your own unique dialplan code. I will show you how to write dialplans and make test phones calls showing you the result of the changes made. If you want to learn by doing then you need to take this course to learn how to use the different Asterisk applications to create a truly unique dial plan for you or your clients. To take the course you should have a working Asterisk or AstLinux system so you can go in and write your own dialplan to follow along with the lessons...

7. VoIP PBX & Call Center on Asterisk 16 Issabel [Master Class]

4.5

(606)

Course Package: This course contains the best of both worlds. Asterisk is open source with GUI (Issabel) and then Asterisk from scratch (Vanilla Asterisk) using source code compilation and CLI based. UPDATED! Cover Asterisk 16 [Command Line/Web GUI] & Cloud Issabel Installation, Security  WebRTC Video Conferencing & Call Center.  Do you need a cost-effective phone system or full features rich Call centers for business?Are you tired of paying the licensing costs to the proprietary phone system?Do you want to expand your business and require a phone system, but legacy systems don't allow you? You are interested to learn asterisk but like to avoid the command line and Linux shell at the start?Issabel (previously known as 'ELASTIX') is the world's most popular and widely adopted open-source IP telephony software. The core VoIP communication is based on Asterisk - The most powerful IP telephony platform. This course is designed for the newbies, small & medium businesses that like to use IP telephony - PBX, or even the solution providers that like to gear up for telephony services to the end-users. The course starts with initial telephony concepts and terms used in the phone service industry-no prior technical knowledge about telecommunication is required to take this course. The course has a project depicting a real-world scenario. This helps the students to have activity-based learning and can apply learned knowledge to the real world. The course clearly explained the phone system features with lab practice for configurations. Proprietary systems invite a license fee for most Business-critical features and UC Features like mobility and third-party integration. To avoid the expensive add-ons, go the open-source way. Use a single system to manage multiple tools and platforms, thereby having "greater employee productivity, reduced costs, and a means to improve customer engagement...