VoIP Essentials: The World of Packetized Voice
Welcome to the new world of Packetized Speech! The voice infrastructure has been rapidly evolving during the last few years. Moving from traditional circuit switching to the packet switching technology has not been easy.
The Internet and IP networks had to cope with the reliability and excellence of service of the Traditional TDM networks. Lot’s of battles had to be fought in order to overcome all of the obstacles that packetized networks had introduced to services.
The world of Voice packetization or Voice over IP (VoIP) as it is lately known, is now proud to achieve:
- Controllable End-to-end Delay
- Nearly constant delay variation ( jitter)
- High levels of Quality of Service
It is only a matter of time before VoIP will ultimately take over the job of the traditional telephone networks.
This change will result in many benefits for both service companies and customers. The major benefits of this transition are:
- Large number of new services will become available resulting in greater customer satisfaction and also in new sources of revenue for the service companies.
- Less maintenance costs for companies and cheaper services for customers.
- Integration of Voice, Data and Video under a common infrastructure will reduce time to market, providing flexible and cost-effective solutions.
Fundamental VoIP Terms
The convergence of Public Switched Telephone Network (PSTN) with IP Network has introduced a lot of new terms and hardware components.
Let’s get a little bit familiar with these new concepts:
Signalling Gateway Controller (SGC)
Also known as “called agent” because of its call control function, SGC is commonly referred to as a “Media Gateway Controller” because of its Media Gateway control function. The SGC is responsible for controlling the call flow and also for setting up and tearing down media pin-holes for the media flow.
The packetization-digitization of Voice is carried out by what is known as a codec (Coder-Decoder). The codec determines the actual amount of bandwidth that the voice stream will occupy. The most widely used voice codecs are G.711 and G.729. The first one result’s in 64kbps transmission speed while the later one takes only 8 kbps.
Learn more about Voice encapsulation.
Session Initiation Protocol (SIP) and H.323
These are the two VoIP signaling protocols that stand out. H.323 is well standardized and it was the first call control signaling protocol adopted, well before Session Initiation Protocol. SIP, however is gaining ground lately due to its simplicity and improvement throughout the time. Eventually SIP will dominate as the next generation VoIP signaling protocol.
Different services require different service behavior. A network that carries Internet data as well as Voice packets has to be classified into different service profiles in order to be able to provide the appropriate level of “satisfaction” to individual services. The Quality of Voice speech depends mostly on transmission delay, delay variation (jitter) and packet loss.
QoS throughout the Voice path is required in order to control these parameters under sustainable values and provide high level of service quality.
Learn more about QoS using Differentiated Service Model.
Real-Time Transport Protocol (RTP)
The Real-Time Transport Protocol is the protocol used for carrying media packets. It provides timestamping for the determination of the jitter variable and the adjustment of jitter buffer adaptive software, sequencing for determination of lost packets and the adjustment of the packet loss concealment mechanism and finally, Marking, for the indication of special events, RTCP is used in conjunction with RTP and its main purpose is to provide statistics and QoS information.
Learn more about RTP and how to effectively encapsulate Voice in IP packets.
It’s Time To Go VoIP
Now is a good time to start thinking about moving into VoIP. I bet you want to know why … well I will give you 3 good reasons why moving to VoIP is a good idea:
- Cheaper calls — even long distance ones. Most VoIP providers offer various service package plans where most of them include a large amount of free domestic calls.
- Multiple Phone lines over a single physical connection. You may have 4 or even 10 phone numbers to call from.
- If you have offices in multiple locations you can connect them all and even use short dialing digits for intra-calls.
And if that’s not enough, how about the fact that VoIP serves mobile users. You can initiate and receive calls while you are on the move as if you were at home or work. This way you can be reached anywhere, anytime.
All you need is either a software installed on your laptop or a VoIP phone at your remote location and an Internet connection. Let the VoIP technology do the rest.
In Love With VoIP
With VoIP, a new world of services and applications is emerging. The benefit is tremendous to both service providers and customers.
As a customer, you will have the chance to exploit useful and interesting services such as Voicemail-to-email, Remote Office, Push-to-talk and many other upcoming services. Not to mention the enriched service quality.
VoIP is truly the future and it is here to stay. Are you ready for VoIP?
Stay tuned to find out how to prepare your network for VoIP and take advantage of its benefits.
In my upcoming articles I will show you how to build your own telephone exchange system and how to benefit from its added capabilities.