Voice Over Internet Protocol

Eric Rice

Department of Software Engineering

University of Wisconsin – Platteville

Abstract

The Internet has drastically changed the way that we communicate today. It has changed the way we get music, share pictures and write each other. In recent years it has started to change something that has existed for over a century – telephony. Voice Over Internet Protocol (VoIP) is the technology which is supposed to replace the existing system. In order for VoIP to work it must, at the very least, meet the high standards the existing system has set. In other words it must have the same or better: voice quality, cost, features and reliability of the existing system. This paper will talk about the different problems and solutions that VoIP has encountered over the past decade.

Introduction

Why reinvent the wheel? Most people would argue that if something works and works well there is no sense in changing it. This is exactly what is happening with VoIP today. VoIP is said to eventually replace the current telephony network and change the way that we communicate. VoIP is technology that has been in development over the last decade. The way a VoIP phone works in comparison to the current telephone network is instead of sending communications over a network of switches it converts the sound into packets, and then sends the packet over the Internet to its destination. There are numerous reasons why there is desire to switch to VoIP. Some reasons include: lower cost, better features, more mobility, less taxes, no long distance and higher competition between companies. There are many standards which must be met in order for VoIP to replace or even compete with the existing network. In order to do this many flaws must be addressed involving the Internet. This paper will analyze the problems that this shift in technology has produced and what developers are doing to resolve these issues.

The Issues

There are a number of standards that the current telephony network has set, the major standards are as follows:

  1. Five-nine’s reliability: this is a standard which the major telephony providers have clamed to achieve. What this means is that 99.999% of the time a customer will be able to place a call. It also means that in any given year the network will have no less more than 5 minutes and 15 seconds of downtime [2].
  2. Voice Quality: In order to assure the same voice quality developers us a ranking system called the Mean Opinion Score (MOS) which is a 5 point scale. The goal of VoIP developers is to use a audio codec which is not only low on bandwidth but also has a MOS of higher than 4.
  3. Delay: It is recommended that for standard telephony the round trip time for a signal should be no greater than 300ms. If the round trip time is greater then 300ms talking with VoIP would be similar to using a satellite phone.
  4. Jitter: not only does the delay have to be minimized but it also has to stay constant. If the delay jumps between 50ms and 300ms the result would be a voice that would sound choppy.
  5. Features: The current telephony network has a standard called SS7 which allows for features like caller ID, call waiting, three-way calling, call Forwarding, voice mail, etc. VoIP would have to include these features. This is a category where VoIP will easily surpass the current network.
  6. Signaling Protocol: there are numerous protocols which are available for VoIP companies to use including SIP, and H.323 they all have their pluses and minuses but SIP seems to be one of the most popular.

There are numerous other issues which are associated with telephony, but these issues are the ones which VoIP has the most development.

Availability

The current standard in the telephony business is five-nine’s reliability. This standard is hard to meet when dealing with the Internet. The Internet is based on a best effort mentality, which means that there is no guarantee that a connection will be established every time or that every piece of data that is sent will reach its destination. If there is a high volume of traffic on a particular router,packets may be dropped resulting in a low quality of service. There are many things that are being considered to fix this problem, including prioritizing Internet traffic and bandwidth reservations. Before VoIP becomes a standard, this will have to be addressed and fixed.

911 Availability

VoIP poses new problems when interacting with 911 emergency services [4]. Currently with hard line phones, when a call is placed into 911 the address and information of the caller is automatically displayed on the operator’s computer screen, which allows for fast and accurate action. Since a VoIP phone is not restricted to any one location it makes it hard to locate the user. Since there are over eight-thousand 911 operators in the US, a decision needs to be made on which 911 operator to route the call to. In order to fix this problem VoIP providers allow for users to give a list of frequently visited call calling locations [3]. This solution in only temporary since it doesn’t provide the absolute accuracy that current wire line phones provide. There are numerous options which are being pursued. One of those options is to add on to the current port mapping technologies that exist in private networks. This would simply require the port mapping to include a physical location. The other option that is available is to add a GPS locator which is similar to what is found in wireless phones [3].

Voice Quality

Voice quality is one of the most important issues when considering VoIP. The goal in this matter is to keep the bandwidth needed to a minimum while at the same time maintain a quality that is comparable to the current system. For a standard call, the audio codec that is used is G.711, which operates at 64kbps [1]. This codec has a MOS of about 4.3, which is pretty good, but since it requires 64kbps it is not a desirable codec to use. Instead, VoIP developers are looking for a codec having a similar quality but requiring less bandwidth. To do this, new algorithms have been developed with bandwidth requirements ranging from 32kbps down to 6.3kbps. Currently the most popular codec is G.729 which had an MOS of about 4.0 and operated at 8kbps [1]. This codec uses silence suppression, which basically means that if an audible noise is not being made nothing will be sent. It will also send frames that vary in length depending on what noise is being made. In the cases where there is silence, the decoder will emit a comfort noise so that there isn’t an uncomfortable silence[1]. Here is a table of some of the codec’s that are being considered along with its corresponding MOS score:

Table 1: Speech Coder Quality [1]

Delay

Another major issue that has to be dealt with is delay. When sending information over the Internet, there is no guarantee on how fast it will get to a destination. Round trip time is used to measure this. Round trip time is the time that it takes for a bit to be sent to the destination plus the time that it takes to send the bit back to the source. An acceptable round trip time for telephony is anything less then 300ms[1]. If the round trip time is greater then 300ms, it can be annoying. The situation would be similar to what happens when making an international or satellite call. Consider a conversation between A and B, let say that A is talking and B decides to interrupt, because of the delay A will continue to talk because it will take time for A to hear the interruption. So it will seem to B that A has ignored the interruption. Since it will still seem like A is still talking B will stop. A will also stop because A will finally get the interruption, so the end result will be silence. A and B will realize what has just happened and they might both decide to start talking at the same time, as a result they will both stop again and the process could repeat itself.[1] There are many things that have to be considered to fix the problem of delay. One of which is reserving bandwidth on a router for VoIP. The problem with this is this would not be fair to the rest of the data that is being routed. Some people have suggested using different priorities for different types of data in this case the same problems arise. Another popular, but very costly solution to this problem is to simply add bandwidth to the Internet. This is popular simply because it doesn’t require the development of any new technology. The major problem with this is historically as soon as more bandwidth is available we tend to find way to use this extra bandwidth. If this were true, after adding the bandwidth we would soon be back in the same situation that we were in before[1].

Jitter

Jitter is a problem that is a result of delay changing. Delay is undesirable, but people can learn to adjust for delays. The problem of jitter happens when the delay is changing, for example one minute the round trip time is 200ms the next minute the round trip time is 400ms. This problem is a common occurrence in the Internet, there are spikes in Internet traffic all day and it is almost impossible to predict them. If jitter is not controlled conversations might seem choppy. To help measure this problem a new protocol was developed called Real-Time Transfer Protocol (RTP) [1]. This protocol runs over UDP. Similarly RTCP was developed which runs over TCP. What RTP packets include in comparison to UDP packets is a sequence number and a time stamp which can be used to calculate jitter and delay.

Features

Currently SS7 is the standard in the telephony industry [1]. SS7 is simply a signaling protocol in which enables different features. The features that SS7 enables are call waiting, caller ID, call screening, call forwarding, three-way calling, all of the star feature, voice mail etc. One of the major benefits of using VoIP is the wide range of features that it can provide. There are a number of different protocols that VoIP can use; one of the most popular protocols for its simplicity is SIP, or Session Initiation Protocol.

One of the protocol options gives intent for a call. So let’s say that an employer is calling one of his employees about vacation. The employee’s caller ID would display that his boss is calling regarding vacation 1]. There are also more advanced features of call screening, where instead of sending the person to voicemail or disconnecting, the phone can be set to a busy signal, or an invalid number message, or if wanted to a insult hotline [1]. Another advanced feature is if a person is waiting for someone to call and they have to leave for some reason, they would be able to set their phone to say that they would be back in 15 minutes or they would be back at 4pm instead of having the person call every 5 minutes until the person is home. VoIP allows the user to customize their phone to any situation might happen. Another good enhancement is being able to check voicemail with a GUI instead of an audio interface[5]. A user would be able to select a voice message, play, fast-forward, rewind etc. They would also be able to email a voice message or do anything else that they could do to a regular audio file. SIP simply allows a user to customize their phone to do what today’s providers would make you subscribe to.

Session Initiation Protocol

SIP is a very powerful protocol and over the last couple years has been gaining popularity. There are two main reasons for this, one is because it is so simple, and the other is the information that it can transfer in the headers as described in the last section. Some of the examples of different message headers are listed in Table 2. Since the headers of SIP protocol are so versatile, it allows the user to customize their service to their personal needs. This will be one of driving factors why VoIP will replace the current telephony system.

Table 2: SIP Protocol Headers [1]

Conclusion

There are many benefits of using VoIP as opposed to the current telephony network in place. With new and advanced features, lower cost, and more integration with the multimedia technology that already exists today,VoIP becomes an attractive option. The major problem that VoIP companies are left to deal with is the best effort mentality of the Internet. Once ISP’s get the same mentality on reliability that the current telephony providers today have, there is no doubt that VoIP will be the new standard. This will require ISP’s to allow resource reservations for Real Time Transmissions or priority queues depending on the type of data. It will also require more redundant systems and possibly even more bandwidth. Until then, VoIP is will be a good option for LAN type environments and for people who subscribe to the same VoIP provider.

Reference:

[1] Danial Collins. (2001). Carrier Grade Voice Over IP. New York. McGraw-Hill.

[2] John Shepler. (2005). The Holy Grail of five-nines reliability.

Retrieved April 1, 2005 from

[3] Tim Lorello, Rich Tehrani (2005). E-9-1-1. Internet Telephony, 8, (3), 40-41.

[4] FCC. (2004) VoIP: FCC Consumer Facts. Retrieved April 1, 2005 from

[5] Vonage. (2005) Features. Retrieved April 1, 2005 from