Teletraffic engineering/How does Internet telephony traffic differ

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Author: Chilekwa Bwalya

Summary[edit | edit source]

The growth of the Internet has seen the demand and introduction of new services beyond those originally envisaged by the early designers of what we now call the Internet. One of the new services that has received a lot of attention is the transmission of voice calls on the Internet. Applications have been developed that enable Internet users talk to other users located anywhere in the world. We examine the characteristics of the traffic generated by Internet Telephony and compare it with the traditional PSTN traffic.

What is Internet Telephony?[edit | edit source]

Definition[edit | edit source]

Internet Telephony
Internet telephony refers to communications services- voice, facsmile and voice-messaging applications that are transported via the Internet, rather than the public switched telephone network(PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol(IP) packets for transmission over the Internet; the process is reversed at the recieving end.[1] The other term used for Internet Telephonyy is VOIP or Voice Over Internet Protocal, a term we shall use interchangeably with Internet Telephony.

How does Internet Telephony traffic differ(VOIP) from PSTN traffic?[edit | edit source]

  • Transmission Mode:
VOIP traffic is packetised and transported as UDP datagrams. The network does not guarantee the availability of sufficient transmission bandwidth for the call.
Figure 1: Packet-switched communications between two systems. Adapted from Gorry Fairhust
PSTN traffic is transported as streams of data over a dedicated transmission link. The network guarantees the availiabilty of sufficient transmission bandwidth for the duration of the call.
Figure 2: Circuit-switched communications between two nodes. Adapted from Gorry Fairhust
  • Transmission Delay:
Internet is designed to be a packet switched network. Thus VOIP packets are subject to varying amounts of transmission delay(Jitter) which results in periods of silence during the conversation or breaks in some words. The propagation delay between any two links is constant. The variation in delay is caused by the non-constant amount of time a given packet spends at a node awaiting service in a queue. The amount of time a packet spends at a given node is a function of the traffic load on that node or the average queue length.
The PSTN network is circuit switched in which the traffic channel is reserved for the duration of the call. The delay experienced in a purely circuit switched network is constant is only limited to propagation delay. The reserving of resources along the transmission path ensures that data arriving at any given node is immediately routed onto the next hop towards the destination.
  • Transmission Path:
Because the Internet is a packet switched network the paths taken by any two packets for a given source and destination node are not necessarily the same. This results in an increase in the amount of variations of transmission delay experienced by packets.
In a circuit switch network, a transmission path between this source and destination node is first established prior to the transmission of data. This path is maintained for the duration of the call. Thus the data stream flows in the same path during the call.
  • Intermediate Node Failure(Partial Network Failure):
Packet switched data has a high degree of immunity from network failure than circuit switched data. A single node failure would result in packets taking a different route avoiding the failed element. The packets for an on-going call would still reach the destination node and thus the call would be maintained.[3]
For circuit switched data, a node failure along the path would disrupt an on-going call and would result in a Call drop.
  • User Identity
The user in Internet telephony is identified using an IP address.
User identity in PSTN traffic is through the E.164 Numbering system [4]
  • Signalling
Signalling in Internet telephony is a best-effort inband system.
PSTN employs common channel signalling which uses dedicated signalling channels. An example of such a signalling channel is Signalling System No. 7 or SS7[4]

Example[edit | edit source]

An example of Internet telephony applications on the Internet is Skype which operates on a peer-to-peer model, rather than the more traditional server-client model. The Skype user directory is entirely decentralised and distributed among the nodes in the network, which means the network can scale very easily to large sizes (currently over 100 million users) without a complex and costly centralised infrastructure. [2]

Exercises[edit | edit source]

Assuming the following topology:

  • Two hosts, A and B, connected by a single link of rate R.
  • Links are un-congested thus there is no queuing delay within each switch
  • Host A sends a packet to Host B. Packet size is L bits
  • The length of each link is M meters and the propagation speed is S meters/sec

Express the following:

a) Propagation delay, dprop in terms of M and S
b) Transmission delay dtrans in terms of L and R
c) End-to-end delay (ignoring queuing and processing delays)
d) Suppose dprop is greater than dtrans. At time t= dtrans where is the first bit of the packet?
e) Suppose dprop is less than dtrans. At time t= dtrans where is the first bit of the packet?


References[edit | edit source]

[1] International Engineering Consortium,Voice Over Internet Protocol tutorial

[2] Wikipeadia , last accessed 11 March 2007.

[3] Kennedy I. G.,Grade of Service. Teletraffic Engineering-ELEN7015 Full Lecturing Notes, School of Electrical and Information Engineering, University Of Witwatersrand, Johannesburg,2007

[4] Hanharan H., Integrated Digital Communications. School of Electrical and Information Engineering, University of the Witwatersrand, Johannesburg, 2006.

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