Network Architecture in Voice Over IP (IP Telephony) Networks

01.11.1999

Janne Martola
Department of Industrial Manegement
Helsinki University of Technology
Janne.Martola@iki.fi


Summary

Voice over IP is a key topic in both established telecommunications industry and the fast expanding Internet sector.  The technlogy promises substantially cheaper voice and fax services through the use of public or private IP  infrastructure for the trunk connection of calls. Voice is becoming just another form of data.

There are a few different scenarios how a Voice Over IP service could be applied. In the first scenario the voice connection is made between two terminals (e.g. between two PC clients, or IP Telephones) that are connected to the IP network. The terminals have required software and hardware to convert the analog voice signals to digital IP  packets with the destination IP address. In the second scenario a connection is made between a terminal connected to an IP network and a traditional telephone in public telephone network. The other possibilities include solutions where the connection is between two telephones in public telephone networks, but the connection goes through an IP network. In the last two scenarios a connection between IP networks and the traditional telephone network is done with an IP telephony gateway and gatekeeper, which are among the most important elements in VoIP technology.

During a typical IP Telephony call the sender speaks into a telephone handset that generates a voltage analogue of the speech. This voltage is converted to digital form, then processed with voice compression software, devised to exploit key voice characteristics. The resulting data is divided into packets for transport over an IP network. At the receiving end, the process is reversed and the sender’s voice is synthesized in the telephone earpiece of the receiver.

Most used standard or group of standards for VoIP applications is H.323. It defines a group of standards for use with multimedia communications over local area networks (LANs) that do not provide a guaranteed quality of service (QoS).


1 Introduction

2 Voice Over IP technology

3 Voice Over IP network elements

4 Network architecture in VoIP networks

References

For more information...


1 Introduction

The current development in telecommunications industry is affecting strongly telecom operators. People are using their telephone lines more and more to transfer data (like surfing in the Internet) and not just for making voice calls.

This is affecting strongly telecom operators and their networks. Public networks seem to be shifting from traditional circuit switching (PSTN) to packet switching networks. Voice networks and IP networks are converging into a IP based networks. This means that both data traffic and voice traffic will be transported in the same IP based networks.

IP Telephony, in other words Voice Over IP (VoIP), is the technology that allows IP data networks to carry also voice and fax traffic. Virtually unknown 5 years ago, it is promising to revolutionize the way voice traffic is delivered and tariffed.

IP telephony is being commercialized for good reason. From a user’s perspective it enables the delivery of services that a conventional switched network could never provide to the masses. And from a service provider’s view the very nature of the packet environment leads to such a substantial gain in efficiency and economy that it cannot be ignored.

This document focuses on giving an overview of VoIP networks and their architecture.
 
 

2 Voice Over IP technology

During a typical IP Telephony call the sender speaks into a telephone handset that generates a voltage analogue of the speech. This voltage is converted to digital form, then processed with voice compression software, devised to exploit key voice characteristics. The resulting data is divided into packets for transport over an IP network. At the receiving end, the process is reversed and the sender’s voice is synthesized in the telephone earpiece of the receiver. [1]

The advantage of transferring voice in packet switched networks is huge. When making a phone call in a traditional circuit switched networks, a large capacity of the network is allocated even though the line is quiet. In packet switching networks data (i.e. voice) is transmitted only when there is something to transmit. This means that the network capacity can be exploited much more efficiently with packet switching VoIP technologies than with traditional circuit switching technologies.
 
 

3 Voice Over IP network elements

In order to make the phone calls through IP networks, such as Internet a set of different equipments or network building blocks are needed. These building blocks and different connection methods are illustrated in the figure 1 below.
 

Figure 1: Main network elements in VoIP networks. [2]
Terminal Equipment

One scenario for making IP Telephony calls is that the voice connection is made between two terminals (e.g. between two PC clients) that are connected to the IP network. The terminals have required software and hardware to convert the analog voice signals to digital IP packets with the destination IP address. One possible example of a suitable software is Microsoft Netmeeting software.

The terminal can be also an IP telephone, which is directly connected to the IP network. In this case all the required analog to digital transformation, and IP packet construction is made inside the telephone.

In other scenarios a connection between IP networks and the traditional telephone network is needed. This is done with an IP telephony gateway and gatekeeper,  which are among the most important elements in VoIP technology.
 

IP Telephony Gateway

The purpose of an IP Telephony gateway is to connect the PSTN and an IP network together. The PSTN can then be used for the first and last legs of the link, with an IP network used for the trunk "connection". Gateways  accept connections from telephones or fax machines, determine the termination point of the call and decide the cheapest way to route the call. Gateways can be PC-based servers, router based voice modules or access concentrator based voice modules. Most of the gateways interoperate through H.323 standard. [2] [3] [4]
 

Example: Sonera IP Communicator

Sonera's IP Communicator is a good example of future networks for Voice over IP applications. Sonera's IP Communicator is a genuine multimedia Virtual Private Network service offering voice, video, data and other services to the customer. All traffic is carried in IP packets from the end user into the corporate intranet and then through Sonera's IP telephony backbone to servers at central sites, where individual PBXs are dedicated to each customer. The picture of Sonera’s network can be seen in the figure 2 below. [3] [5]

Figure 2: Sonera's network for VoIP services. [5]
In IP Communicator the phones as well as other terminals are directly connected to the corporate Local Area Network (LAN) and not to the corporate PBX. In all terminals H.323 client software is installed. When a user wants to make a call, he or she logs into the IP Communicator and clicks with a mouse the person to call to from a phone book. The information is sent from the web server to the IP PBX, which then infroms the master gatekeeper to find the best route to the destination gateway. The call is then assigned to that gateway. The IP PBX rings the destination user, and when the destination user picks up the phone, the call is established. [3] [5]
 

4 Network architecture in VoIP networks

The IP telephony architecture can be over overlaid on the TCP/IP model. Figure 3 shows a typical IP telephony application built on the Transport Layer of the TCP/IP stack.[2]

Figure 3: IP Telephony and TCP/IP [2]
A key element in enabling a wide spread use of voice over IP will be to get standards that make it possible to combine equipment of different vendors and in different countries. Before this is implemented, VoIP cannot offer the same level of ease and certainty that the regular POTS offers. One of the most important standards created for IP telephony is the H.323 standard.

The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. By complying to H.323, multimedia products and applications from multiple vendors can interoperate, allowing users to communicate without concern for compatibility. The H.323 standard defines a group of standards for use with multimedia communications over local area networks (LANs) that do not provide a guaranteed quality of service (QoS). Although it was originally designed for use with a single LAN, modifications have been introduced to make it suitable also for wide area networks (WAN) and enable wide spread implementations. The H.323 structure is presented in figure 4 below. [2] [6]

Figure 4: H.323 structure. [6]
The standards included in H.323 can be seen in Figure 4. On the lowest level H.225.0 defines media packetization, stream synchronization, control message packetization and control message formats. Below it is the layer 4 of theTCP/IP stack which makes the standard network independent. H.323 uses both UDP and TCP protocols to achieve optimal performance. The real time nature of video and audio make reliable transport not possible and UDP is used for these. For control signals and data, reliable transport is required and TCP is used. In addition to H.225.0 control signals are defined by H.245 that defines negotiation of channel usage and capabilities exchanges. The registration, admission and status (RAS) component defines communication with gatekeepers.[6]

The media standards included in H.323 contain the audio codecs (G.711, G.722, G.723, G.728, G.729) and video codecs (H.261, H.263), used to digitize the voice and optionally the video. Also optionally included is the T.120 recommendation that defines the use of data channels like application sharing and text messages. [6]
 
 

References

[1] Whatis.com Inc. Reference: VoIP [browsed in 27.09.1999].
URL: http://whatis.com/
[2] Nokia. Nokia IP Telephony - Architecture white paper. [browsed in 01.11.1999]
URL: http://www.nokiaiptel.com/
[3] Heywood, Peter. The Future is calling: Sonera is revolutionizing telecom services with its IP Communicator, DataComm Magazine, January 1999. [browsed in 27.09.1999]
URL: http://www.data.com/issue/990107/services2.html 
[4] Whatis.com Inc. Reference: gateway [browsed in 27.09.1999].
URL: http://whatis.com/
[5] Heywood, Peter. The Future is calling: VPNs: The Next Generation, DataComm Magazine, January 1999. [browsed in 27.09.1999]
URL: http://www.data.com/issue/990107/services2_figure.html
[6] Cisco Systems. Voice Over IP Network Design, Cisco Presentation at Networkers 98’ Conference December 14-16, 1998, Cannes, France

For more information...

Whatis.com provides a large amount of descriptions about Internet related topics:

Whatis.com Inc. What Is.... URL: http://whatis.com/
 

Cisco Systems is one of the major manufacturers of VoIP network solutions and network elements. A look at the Cisco's products gives a good view about the current development:

Cisco Systems. IP Telephony. URL: http://www.cisco.com/warp/public/cc/cisco/mkt/iptel/
Cisco Systems. Cisco CallManager. URL: http://www.cisco.com/warp/public/cc/cisco/mkt/iptel/callmgr/index.shtml
Cisco Systems. Cisco Access Gateways. URL: http://www.cisco.com/warp/public/cc/cisco/mkt/iptel/gateway/index.shtml
Cisco Systems. Cisco IP Telephones. URL: http://www.cisco.com/warp/public/cc/cisco/mkt/iptel/ipphone/index.shtml
 

Excellent pages from Finnish IP Telephony solutions provider Nokia. Contains lots of pictures about the VoIP network architecture and descriptions about the functionalities of different network elements:

Nokia. Nokia IP Telephony Corporation - New Business Opportunities for Internet Service Providers. URL: http://www.nokiaiptel.com/public/isp.html
Nokia. Nokia IP Telephony - New Business Opportunities for Start-up Companies. URL: http://www.nokiaiptel.com/public/startup.html