Voice over IP (VoIP)/Internet Telephony

November 1st 1999

Sandro Grech
Department of Electrical and Communications Engineering
Helsinki University of Technology
sandro@cc.hut.fi

Abstract

Internet Telephony is a powerful and economical communication option by combination of the telephone and data networks. The ability to use IP network to carry traditional telephone traffic brings both challenges and opportunities to all the long-distance telephone service companies and their resellers. Although a lot of difficulties exist, from technology to social issues, it will undoubtedly bring a great change in communication field and bring a new huge market. 


Table of Contents

1 What is IP Telephony?

In contrast to traditional circuit switched telephony, where an end-to-end circuit is set up between two telephones, IP telephony uses the Internet protocol to transmit voice as packets over an IP network (Internet, Intranets, LANs, etc…). In circuit switched connections, the connection is established for the whole duration of every telephone call, with a fixed bandwidth (64 kbit/s) reserved even during silent periods. In an IP telephony connection, the voice signal is digitized, compressed and converted into IP packets, which are transmitted over the IP network and shared with other IP traffic. Not only is a packet-based shared network more efficient than a fixed 64 kbit/s circuit switched connection, but it also compresses the voice signal. To accomplish this, a gateway provides the connection between the telephone network and the IP network. 
 
 


Figure 1 - Network Architecture for Internet Telephony [4]


IP telephony isconnected to the PSTN by a voice gateway that is placed between the PSTN and the IP network (figure 1). This provides the physical interface between the telephone network and the IP network. The voice gateway performs the following functions:

  • signaling to and from the telephone network, 
  • reception of telephone numbers, 
  • conversion between telephone numbers and IP addressing in the IP network, 
  • voice processing  - reception of the voice signal, compression and packetisation, echo cancellation, silence suppression, etc. 
The gateway compresses the voice signal for two reasons 
  • to reduce the amount of bandwidth required in order to reduce cost 
  • to reduce the delay impact from the network. 
Generally, users dial a telephone number of the gateway, the gateway will respond with an audio request for the user’s destination telephone number, and a routing table will identify which gateway is located closest to the destination telephone network. The IP address of that gateway is then used to route the telephone call as datagrams through the IP network. The gateway also reverses the operation for packets coming in from the network and going out of the phone. Both operations (coming from and going to the phone network) can take place at the same time, allowing a full duplex conversation. [2] Gives a more detailed description of a typical VoIP call setup procedure. IP telephony encompasses a number of services including phone-to-phone, PC-to-phone, phone-to-PC, PC-to-PC and fax-to-fax, as well as video conferencing (figure 1). 

In an IP telephony solution, it is possible to have a combination of PC-based telephony applications and PSTN connected telephones. In a phone-to-phone scenario, the gateway will, in real time, perform the functionality needed in order to send and receive voice over the IP network. In a PC scenario an IP telephony client is needed. The client digitizes, compresses and packetizes the voice signal and transmits it over the IP network. Standard telephone calls are connected to a voice gateway and IP telephony calls connect to a telephone or a PC. IP telephony client software may allow users with multimedia PCs to video and audio conference, share documents and use a white board, enabling a more efficient work environment
 

2 How does an IP Gateway work? 

Conceptually, Internet telephone gateways work like this. 

  • On one side, the gateway connects to the telephone world. It can communicate with any phone in the world. A phone line plugs into the gateway on this end. 
  • On the other side, the gateway connects to the Internet world. It can communicate with any computer in the world. A computer network plugs into the gateway on this end. 
  • The gateway takes the standard telephone signal, digitizes it (if it is not already digital), significantly compresses it, packetizes it for the Internet using Internet Protocol (IP), and routes it to a destination over the Internet. 
  • The gateway reverses the operation for packets coming in from the network and going out the phone. 
  • Both operations (coming from and going to the phone network) take place at the same time, allowing a full-duplex (two-way) conversation. 

Figure 2 -  Internet Telephony Gateway


3 How Well Does It Work? 

There are two main factors contributing to quality: voice quality and round-trip time, or latency. Voice quality has improved greatly from early versions of the technology due to the following factors:

  • Improved gateways. 
  • Deployment over private networks. 
  • Internet development - Today's Internet was not designed with real-time communication in mind. The Internet Engineering Task Force (IETF), together with Internet backbone equipment providers, is addressing this with technologies like Reservation Protocol (RSVP), which will let bandwidth be reserved.  

4 The factors making Internet Telephony possible

  • Voice quality is increasing, thanks to new codec technology; 
  • There are ongoing improvements in compression techniques; 
  • Full-duplex PC sound cards enable two-way simultaneous calls; 
  • The typical PC is getting more and more powerful, making it possible to perform processor-intensive functions without specialized hardware. 

5 Ideal Internet Telephony should be:

  • high-volume call processing within and between public and private networks 
  • high-volume, real-time translation between IP and circuit-switched networks 
  • economic scaleability 
  • broad acceptance and implementation of standards 
6 What users can get now?
  • point-to-multipoint voice 
  • data conferencing 
  • application sharing 
  • long-distance telephone savings 

7 Three factors to success:

  • Ease of connectivity in terms of dialing another party directly, whether or not the party is on the Internet 
  • An open set of standards that operates independently of which product is used 
  • The addition of telephony features that provide a competitive differentiation, other than price, from conventional long-distance services 
8 Current Problems and Solutions

8.1 Current problems [12]

  • 8.1.1 Standards [2]
    The biggest difficulty that VoIP technology is facing is the interoperability between Internet telephony products and interworking with PSTN-based systems and services. Currently, no two products are compatible. Users who want to make Internet phone call have to have the same kind of software. Standard development and adoption are the key to ensuring interoperability. At this time, the primary specification issues that have to be resolved are related to the codec format, the transport protocol, and directory services. 
     
  • 8.1.2 Quality [1]


    • Voice performance is measured by delay. Calls on the public switched telephone network usually exhibit 50- to 70-millisecond delay. That latency increases substantially on the Internet, where it typically ranges from 500 milliseconds. Humans can tolerate about 250msec of latency before it has a noticeable effect. Latency affects the pace of the conversation. Today's Internet telephony products exceed this latency, so most connections sound like traditional calls routed over a satellite circuit. However today's products are well suited to many applications. 
     

  • 8.1.3 Capacity


    • The Internet is an open network of many different ISPs' networks. Consequently, there is no way to get network bandwidth, packet sequence and latency guarantees. One of the main parameters affecting the quality of service on the Internet is lost packets. Packet loss is a persistent problem, particularly with the increasing popularity, and therefore increasing load, of the Internet. Packet loss can occur for a number of reasons. Network congestion due to bandwidth limitation or traffic overload is the main reason. Inadequate network access links, especially local ISP connections to the Internet backbone, are already causing chronic bandwidth congestion. Current Internet telephony applications repair lost packets with silence, which leads to the speech clipping effects and comparatively large packets are used, even the loss of individual packets has a serious impact on the intelligibility of speech. 

  • 8.1.4 Bandwidth Usage


    • WAN bandwidth is still expensive, and most IT managers are reluctant to add yet another service to their already congested Intranets. 
     

  • 8.1.5 Service Reliability 
Voice is a mission critical application. The platform and software creating the IP Telephony service must be able to run error free and continuously for months, not hours. 
  • 8.1.6 Network Reliability 
Many networks have difficulty providing reliable file transfers, let alone real-time traffic. Since voice is mission critical, the IP network supporting it must be at least as reliable as the traditional phone network. 
  • 8.1.7 Network Performance 
Voice is real-time and the quality can suffer from such things as latency, jitter, and packet loss. Networks delivering digital voice must reduce these effects in order to preserve acceptable speech quality. 
  • 8.1.8 Voice Quality 
Codecs and compression algorithms must reconstruct voice with high readability and speaker recognition. Toll quality is the minimum accepted. 
  • 8.1.9 Interoperability
In order to connect to legacy analog voice systems, a successful IP Telephony solution must support interfaces to T1/E1, ISDN, etc. Signaling protocols such as SS7, Q.931 must also be supported. In addition, it has to be ‘future-proofed’ against changes in industry direction and the introduction of new applications. 
 
  • 8.1.10 Social issues


    • Regulation of Internet telephony is still largely a question mark. Traditionally, telephone service has been heavily regulated. In most countries, governments or government-sanctioned entities retain monopolies for provisioning telephone service. Moreover, even without the Internet, telephony service is deregulating in many countries around the world, although the deregulation process is time consuming and heavily political. ACTA argues that the major reason Internet telephone calls are cheaper than traditional circuit-switched calls is the access charge exemption ISPs enjoy. Hence, they argue, the Internet-based providers have an "unfair" advantage in offering cut-rate long distance phone service. Fortunately for the Internet telephone industry, and the Internet industry itself, the FCC does not seem to agree. 
    • 8.1.11 User Acceptance 
    Apart from voice quality, the users must be presented the same features (call waiting, call forward etc.) and ease of use as current voice systems. 
    • 8.1.12 Voice-band Data
    Fax and modem tones are routinely carried on analog voice lines and must be incorporated into any serious IP Telephony scheme. 
    • 8.1.13 Network and Services Management 
    Comprehensive network management capability must be integral, including configuration, account management and billing. 
    • 8.1.14 Scalability 
    The underlying architecture of any successful IP Telephony solution also had to be scaleable from hundreds to tens of thousands of users.  

    8.2 Solutions to current problems

    • 8.2.1 Standards [5,6]

      •  

       

      The ITU H.323 recommendation, defines the core technology for VoIP applications. The initial objective of the recommendation was to identify a voice compression algorithm that could transport voice with quality equivalent to 32 kbps ADPCM at only one-fourth of the bandwidth. The ratified standard compresses signals to 8 kbps while delivering 4 kHz speech bandwidth with toll quality. In fact, the algorithm delivers an exceptionally high level of voice quality with minimal delay and hopes are high that this state-of-the-art vocoder will be adopted by major vendors [9,14].

    H.323 also calls out T.120 for data conferencing. T.120 enables products from different vendors to interoperate without terminals assuming prior knowledge of the other systems. It specifies the network interfaces and wire formats, along with a data transmissions facility.    RTP is finding acceptance as the standard transport protocol for time related applications over the Internet. It provides a time-stamp and control mechanisms for synchronising different streams with timing properties. RTP has been introduced as a new protocol layer to provide support for applications with real-time properties including timing reconstruction, loss detection, security and content identification.    Since RTP does not address the issue of resource reservation or QoS control, it relies on the resource reservation protocol (RSVP) to provide these capabilities [7]. Currently a draft standard protocol, the RSVP is part of various efforts to enhance the current Internet architecture with support for QoS flows to be able to handle real-time traffic more reliably. RSVP is a new signalling protocol to be implemented in Internet routers to provide for new classes of services by reserving paths for sessions on an end-to-end basis. This is achieved through three main functions, admission control, packet classification and packet scheduling. Each vendor has, however, its own strategy for performing these functions to provide the controlled load service at a first stage and the ultimate guaranteed service later. Although some vendors have already announced plans for providing telephony gateways with RSVP support, the reality is that after 13 consecutive revisions, RSVP specifications are still only a draft and testing of products and services for interoperability in public networks is still in the planning stages. 
    • 8.2.2 Quality [8]
    Serveral ways are used to improve the quality: 
      • improvements in protocols: RSVP 
      • dedicated service lines with managed traffic loads 
      • co-locating telephone access with backbone nodes 
      • bigger routers 
      • new network architechtures 
      • capacity
    • 8.2.3 Social issues


      • Although there are a lot of society issues, people always encourage new technology. The trend of technology development can not be prevented by anything. More than 100 companies are involed in the develpment of Internet Telephony, including AT&T, MCI and Sprint. These long distance service giants do not regard the Internet Telephony technology as their threats but opportunities. Thinking about the supports from industry and keeping promises that encourage competitions in 1996, FCC deny the petition from ACTA and provides the 
      offical support to Internet Telephony. 
       

    9 Future Developments

    The future is exciting. There are two dimensions in which the Internet telephony can be improved. One dimension is technology itself. It will go to much more high quality, big capacity, multi-function, etc.. The other dimension is the combination with other technology, such as Intranet, to confirm a new pruduct or tool to meet the increasing needs to the communication. 
     
     

    List of References
     
     
    [1] MICOM Communications Corp., Voice/Fax Over IP: Internet, Intranet and Extranet, 16.04.1997 [referred 14.09.1999]
    <http://www.micom.com/WhitePapers/whtpaper.pdf> 
    [2] The International Engineering Consortium, Internet Telephony Tutorial, 13.06.1999 [referred 14.09.1999] 
    <http://www.webproforum.com/int_tele/index.html>
    [3] Microlegend Telecom Systems, Microlegend IP Telephony Tutorial, 1998 [referred 15.09.1999]
    <http://www.microlegend.com/about-it.htm>
    [4] Nokia IP Telephony Business Unit, IP Telephony Deployment Roadmap for Multiple System Operators (MSO's), 02.1999 [referred 15.09.19999
    <http://www.nokiaiptel.com/public/mso.html>
    [5] Schulzrinne, H. & Rosenberg, J., The IETF Internet Telephony Architecture and Protocols, May-June 1999 [referred 15.09.1999]
    <http://computer.org/internet/telephony/w3schrosen.htm/>
    [6] Schulzrinne, H. & Rosenberg, J., Internet Telephony: Architecture and Protocols - an IETF Perspective, 02.07.1999 [referred 15.09.1999]
    <http://members.xoom.com/woohaah/Schu9902_Internet.pdf>
    [7] Metz, J., IP QoS: Traveling in the First Class on the Internet, March-April 1999 [referred 15.09.1999]
    <http://www.computer.org/internet/v3n2/w2onwire.htm>
    [8] Kostas, T.J. & Borella, M.S. & Sidhu, I. & Schuster, G.M. & Grabiec, J. & Mahler, J., Real-Time Voice Over Packet-Switched Networks, Jan.-Feb. 1998 [referred 15.09.1999]
    <IEEE Network Magazine>
    [9] Rizzetto, D. & Catania, C., A Voice over IP Service Architecture for Integrated Communications, May-June 1999 [referred 18.09.1999]
    <IEEE Internet Computing Magazine>
    [10] Hewitt M., IP Telephony, 05.04.1999 [referred 18.09.1999]
    <http://www.telephony.net/telephony_home.htm>
    [11] Jain R., Voice over IP: References, 31.07.1999 [referred 18.09.1999]
    <http://www.cis.ohio-state.edu/~jain/refs/ref_voip.htm>
    [12] Udall J., Extending the PSTN: Voice over IP Gateways, 02.1998 [referred 18.09.1999]
    <http://www.tmcnet.com/articles/ctimag/0298/nettel001.htm>
    [13] V.A., Internet Telephony Magazine, Jan. 1998- Aug. 1999 [referred 10.09.1999]
    <http://www.ctiexpo.com/it/itpast.htm>
    [14] Databeam Corp., A Primer on the H.323 Series Standard, 14.09.1999 [referred 18.09.1999]
    <http://www.databeam.com/h323/h323primer.html>
    [15] Gareiss R., Voice over the Internet - Today it’s a toy - Tomorrow it may be a crucial part of corporate telephony, Sept. 1996 [referred 18.09.1999]
    <http://www.data.com/roundups/internet_voice.html> 

    Further Information
    Research Groups
     
    LBNL Network Research Group
    WWW: http://ee.lbl.gov/
    Cambridge University Engineering Department SVR Group
    WWW: http://svr-www.eng.cam.ac.uk/
    Tarifica
    WWW: http://www.tarifica.com/
    MICE Multimedia Index
    WWW: http://www.cs.ucl.ac.uk/mice/
    Telemedia, Networks, and Systems Group
    WWW: http://www.tns.lcs.mit.edu/
    MIT Advanced Network Architecture Group
    WWW: http://ana-www.lcs.mit.edu/anaweb/ANA_Home.html

     

    A few of the sources in the List of References were from IEEE papers available online from the main 
    library at HUT. To be able to view these papers you will require Adobe's Acrobat Reader. You can 
    Download Adobe's Acrobat Reader Here.

    Back to the Table of Contents

     
     

    Created:            28th Oct. 1999
    Last Modified: 1st Nov. 1999 @ 23:30 (GMT +02) - Layout modifications