+86-755-2564 3785info@pearlelectronics.com
Home  FAQ VoIP

Voice Over Internet Protocol - VoIP


What is VOIP

Introduction: VOIP is an acronym for Voice Over Internet Protocol, or in more common terms phone service over the Internet.

If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company. Some people use VOIP in addition to their traditional phone service, since VOIP service providers usually offer lower rates than traditional phone companies.


How does VOIP work? What happens when you make a VoIP call?


When a VoIP call is made, your voice goes through the following process:
1. Your voice (analog) is sent from your regular telephone to a device called an Analog Telephone Adapter (ATA). The ATA converts your analog voice into digital samples through the use of an Analog-to-Digital Converter (ADC). The ATAs are usually provided by your VoIP service provider when you sign up for service.
Note: If you have one of the new digital IP telephones that are now available on the market, there is no need for the ATA device since the ADC function is performed inside the IP telephone.

2. The digital bits must now be compressed into a standard format which can be transmitted faster and more efficiently. In VoIP, digital signal processors (DSPs) perform this compression using codecs which segment the voice signal into frames and store them in voice packets.

Some compression standards and associated bandwidths are listed as follows:
o PCM, Pulse Code Modulation, Standard ITU-T G.711, 64Kbps
o CS-ACELP, Standard ITU-T G.729 and G.729a, 8Kbps
o ADPCM, Adaptive differential PCM, Standard ITU-T G.726, up to 40Kbps
o LD-CELP, Standard ITU-T G.728, 16Kbps
o MP-MLQ, Standard ITU-T G.723.1, 6.3Kbps, Truespeech
o ACELP, Standard ITU-T G.723.1, 5.3Kbps, Truespeech
o LPC-10, able to reach 2.5 Kbps
While standard phones utilize the G711 codec, the G723 codec is emerging as the popular codec choice for IP Telephony applications. This codec is preferred due to its smaller size and higher compression which allows for easier transport over the internet.

3. The compressed data must then be encapsulated within IP packets. VoIP is a Layer 3 network protocol that uses various Layer 2 point-to-point protocols such as PPP for its transport. VoIP protocols typically use Real-time Transport Protocol (RTP) for the media stream or speech path. RTP uses User Datagram Protocol (UDP) as its transport protocol. For IP networks, the reliable service of TCP is not appropriate for real-time applications because TCP uses retransmission to ensure reliability. The IP layer provides routing and network-level addressing; the data-link layer protocols control and direct the transmission of the information over the physical medium.

4. The packets are then transmitted across the internet in compliance with a voice communications protocol or standard such as H.323, Media Gateway Control Protocol (MGCP), or Session Initiation Protocol (SIP).

5. When your IP packet (which contains your speech) arrives at the destination (the telephone that you called) it must go through a similar process mentioned in 1-4, but in reverse. As such the IP packets are de-capsulated or disassembled to retrieve the compressed voice data, which can then be decompressed using the same codec that performed the compression. After the decompression, the original digital data is left which can then go through a digital to analog converter and be returned to its original analog voice format and be clearly heard and understood by your called party.

This entire process is completed in real time such that telephone users do not detect a delay in the speech. The diagram below shows a high level view of how a basic VoIP call is made and the path that the packets travel to reach their destination. 


The CO or Central Office connects the local loop from the demarcation point at the VoIP subscriber's residence. The CO then makes the decision where to send the call. An expanded view of the CO and the PSTN (of which the CO is a part of) is shown in the diagram below. This diagram shows how a typical DSL line is integrated into the network. The topology will be slightly different for other types of broadband connection but the general path of the data packets will be the same when it reaches the CO.


This diagram has expanded the view of the CO and shown some potential destinations for circuit switched voice that goes through the PSTN. This is obviously not where the VoIP packets are destined and as such it is necessary to show an expanded view of the Internet Service Provider (ISP) network since this is where the VoIP packets will be sent to. The diagram below indicates the path of a typical call through the ISP chain.


Hopefully this advanced guide has given you a deeper insight into the inner workings of VoIP. Below you will find related articles for further reading, including an in-depth article on the actual digitization process.


Why use VOIP?
There are two major reasons to use VOIP
• Lower Cost
• Increased functionality


Lower Cost: In general phone service via VOIP costs less than equivalent service from traditional sources. This is largely a function of traditional phone services either being monopolies or government entities. There are also some cost savings due to using a single network to carry voice and data. This is especially true when users have existing under-utilized network capacity that they can use for VOIP without any additional costs.


In the most extreme case, users see VOIP phone calls (even international) as FREE. While there is a cost for their Internet service, using VOIP over this service may not involve any extra charges, so the users view the calls as free. There are a number of services that have sprung up to facilitate this type of "free" VOIP call. Examples are: Free World Dialup and Skype for a more complete list see: VOIP Service Providers


Increased Functionality: VOIP makes easy some things that are difficult to impossible with traditional phone networks.
• Incoming phone calls are automatically routed to your VOIP phone where ever you plug it into the network. Take your VOIP phone with you on a trip, and anywhere you connect it to the Internet, you can receive your incoming calls.
• Call center agents using VOIP phones can easily work from anywhere with a good Internet connection.






Contact Us



Unit 1311, 13/F, Building C, Jufu Garden, 41 Guowei Road, Luohu District, Shenzhen China 518004