Analog Telephony Part I
About this course
Prepare for your VoIP project like a pro
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.
This 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).
Here 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.
Here we will present the basics of VoIP.
How to select the best codec for your Job.
Channels and Bandwidth Dimensioning
In 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 Quality
In 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.
Only 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.
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 Precendence
Fragmentation, Interleaving and Compressed RTP
How to handle slow circuits using fragmentation and interleaving and how to enable cRTP in PPP circuits.
Queueing and Tail Drop
Most 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 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.
What you’ll learn
- How to dimension VoIP networks
- How to Implement QoS
- How to choose a Codec
- How to analyze the voice quality
- How to integrate with the legacy
Are there any course requirements or prerequisites?
- Network, Ethernet, TCP/IP
Who this course is for:
- Analysts responsible for designing VoIP networks
- Technical and Management staff starting new VoIP projects
- Open Source Telephony people who wants to increase their knowledge on telephony and networks