Networks Training

IP Telephony and VoIP Tutorial-Part 1

Although this is a Cisco networks dedicated blog, I decided to start a series of tutorial posts about a general technology which is not directly related to Cisco but it is a field in which Cisco is again a major player. This is IP Telephony and Voice over IP (VoIP). The two terms, IP Telephony and VoIP, are related around the same concept but in my opinion they are not exactly the same thing. Many people refer to these two terms interchangeably but they are not exactly the same. So, before moving on lets clarify the difference between IP Telephony and VoIP.

IP Telephony Vs VoIP

IP telephony has to do mainly with digital telephony systems (LAN based IP PBX systems) which use the IP protocol entirely for voice communication. All components of the IP telephony system use digitized voice which is transferred as IP packets through an IP network (usually the LAN network). The telephone handsets (VoIP phones) translate the analogue voice signal into digital voice (binary voice) which is transferred as IP packets from one phone to another. The call control system is usually a software based (softswitch) server which handles all call signaling, call routing, IP phone management etc, again using IP protocol for transport. So think about IP telephony as a bigger concept.

VoIP on the other hand is a subset of IP Telephony. Basically, VoIP is the technology which is used by IP Telephony as the vehicle to transport phone calls. VoIP is the technology in which the analogue voice signal is digitized (analog to digital conversion) and becomes binary numbers in order to be transferred by the IP protocol. VoIP is the basis for the implementation and functionality of an IP Telephony system. VoIP can also be used by legacy TDM based PBX systems to transport voice calls over an IP WAN network or even over the Internet. Special voice gateways are used to connect to the legacy PBX telephone system on one end and to the IP network on the other end in order to translate the TDM voice stream into IP voice packets.

MORE READING:  Basic IP Phone Configuration on Cisco Call Manager Express

So to summarize, IP Telephony is the overall concept of the modern form of voice communication which harnesses the power and features of VoIP technology in order to offer the overall experience of communicating effectively and with lots of extra features.

Now that we described the difference between IP Telephony and VoIP, let’s see more details about the two concepts:

1. More details about Voice over IP

The term VoIP or Voice over IP refers to the transfer of voice packets over networks based on Internet technology and, more specifically, the IP Protocol. The IP protocol on which the whole Internet is based on was created to implement the transmission of data in the form of data packets. This means that when a data document is transferred over the Internet is cut into small IP packets and sent over the network. When the document reaches its destination, the packets are joined again thus recreating the original document. The same logic applies if the data transferred corresponds to a voice conversation. The voice is digitized, chopped into packets of data transferred over the network via the IP protocol. At the destination the packets are rejoined to recreate the voice stream. Here we should make clear that VoIP refers to the transfer of voice over any IP network. Such a network is the Internet of course, but when considering VoIP it does not necessarily mean that we carry voice over the Internet only. It can be any IP-based network (such as a private corporate WAN network).

MORE READING:  IP Telephony and VoIP Tutorial-Part 2

2. Packet based (IP Telephony) Vs Circuit Switched Telephone Systems

IP Telephony systems are those using entirely IP packets for voice communication, as explained before. In contrast to packet switched telephone systems (those based on IP protocol), conventional telephone systems apply the logic of direct connection between the two communicating voice parties through a dedicated circuit reserved exclusively for each contact. Thus the term Circuit switched telephone systems. In packet switched systems, however, the same communication line can be used to simultaneously pass different kinds of packets. Thus, the voice packets of one or more conversations may travel through the same route as other packets transferring data, video etc. This is the main difference between traditional telephony which is implemented to the public switched telephone network (PSTN) and telephony implementation on IP networks (or more generally to packet switched networks).

See Parts 2 and 3 of IP telephony and VoIP Tutorials Below:

IP Telephony and VoIP Tutorial PART2

IP Telephony and VoIP Tutorial PART3

Filed Under: IP Telephony Tagged With: ,

Comments

  1. Making the distinction between business and residential VoIP is useful. The former is going gang busters, but it hasn’t been such a great success in winning over mums and dads. They generally don’t need all the integration stuff and residential VoIP has copped a bad press. It’s cheaper, but voice quality and other issues are the downside.

  2. Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.

    Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.

    Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.

    By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.

    How VoIP Works

    When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.

    What are speech codec’s and what role codec plays in VoIP?

    Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.

    Following is a list of VoIP codec’s along with how much data network bandwidth they consume.

    * AMR Codec
    * BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
    * GIPS Family – 13.3 Kbps and up
    * GSM – 13 Kbps (full rate), 20ms frame size
    * iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
    * ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
    * ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
    * ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
    * ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
    * ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
    * ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
    * ITU G.726 – 16/24/32/40 Kbps
    * ITU G.728 – 16 Kbps
    * ITU G.729 – 8 Kbps, 10ms frame size
    * Speex – 2.15 to 44.2 Kbps
    * LPC10 – 2.5 Kbps
    * DoD CELP – 4.8 Kbps

    Switch to VoIP Today and you will never want to use traditional PSTN ever again.

    Thanks

    -Imran

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About Networks Training

We Provide Technical Tutorials and Configuration Examples about TCP/IP Networks with focus on Cisco Products and Technologies. This blog entails my own thoughts and ideas, which may not represent the thoughts of Cisco Systems Inc. This blog is NOT affiliated or endorsed by Cisco Systems Inc. All product names, logos and artwork are copyrights/trademarks of their respective owners.

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