A Voice Over Ip Network - Document

A Voice over IP Network Alexandru Juncu

1.Introduction Voice over IP or IP Telephony or most often referred as VoIP is the attempt to unite the old telephony system with the data network in what is know as a converged network. It is a very daring attempt because of the implications that are technological, financial and even sociological. It is a rather new concept and still in it’s development infancy, but it has a huge potential for the future. Why would we need it? Although we all are used to the old telephone system, voice over IP, will make things much simple. The cost for a telephone call will be reduced dramatically but the quality will increase, the administration will be much simple and we will have a new way perspective on a simple phone.

2.Components of a VoIP network When thinking of a VoIP network we need to consider a large number of components. These components can be • • •

Hardware components ( end devices, interconnecting devices, core devices) Applications (server software, client software, administration software ) Protocols (the way components communicate with each other)

All these components have to be taken into consideration to make the network work. There is a large number options to choose from for each component so it best fits the need of the end users.

The end devices are the phones. They can either be pure IP Phones that are physical devices similar to plain phones (but with more advances features ) that understated the VoIP protocols and directly connect to the VoIP network or plain old phones that only understand analog signals and connect to the IP network through an interconnecting device (called ATA ). The difference is that IP Phones use protocols like IP, TCP, UDP and VoIP protocols (like SIP, H323, RTP) while normal phones only understand analog signals. IP Phone (Cisco 7960) The end devices can also be so Softphones that are applications that run inside a PC’s operating system. Because this type of device only needs the software part the choices are vast. They are the cheapest option of an end device (most of the time is free). The interconnecting devices are the usually the same as in a typical network (routers, switches, access points) to witch we can add devices such as ATAs. Softphone (XLite for Windows)

An important part of the VoIP network are the voice servers. They connect and manage the other devices and control all of the voice traffic. They are actually applications (and here too we have a large number of choices) that run inside of dedicated computer server or routers or dedicated devices. The job of the voice server is to be aware of the whole network so they can route calls that the end devices make. Here is where all the call management takes place ( filtering, accounting, traking). The critical components of the voice network is the protocol suite. All VoIP network run on top of the Internet Protocol that assures communication at the third layer of the OSI stack. This means that packet are routed inside the network(s) form sender to receiver. The way packets are used is the job of the upper layers and the protocols of those layers. On top of IP, we have UDP (user datagram protocol). UDP is used and not TCP (that is much more reliable) because of it’s speed and low overhead. To fix the

shortcomings of UDP, RTP (Real-time Transport Protocol) make the transmission of packets (this time voice or video packets) as reliable as TCP. Finally, on top of RTP we have all the VoIP protocols. Examples are SIP (Session Initiation Protocol), H323, SCCP, MGCP. These protocols are responsible for establishing, controlling, administering and ending phone calls. Aside from these, we have Codecs that represent the voice or video format. The quality of the audio stream and the bandwidth consumption depends on what and how codecs are used. G.711, G.722, G.726, G.729 are G.722 are the most used.

3.Designing and implementing a VoIP network When designing a VoIP solution there are some things that you need to take into consideration. Current and needed infrastructure, number of users, budget, need features and even legal issues. First of all you need an IP infrastructure. This means that you have to have a network that can route IP packets. Most of this infrastructure represents equipment: switches(or hubs) for LAN connectivity, routers for WAN connectivity, access points for wireless capabilities and cables to connect these components. Also you also need to consider things like firewalls (for security) NAT servers and DHCP servers (for addressing management) and user management servers (such as LDAP or Active Directory). It is important to have a “VoIP ready” infrastructure because this has an important impact on QoS (quality of service) of the voice network QoS in a voice network means that you have to take care of some factors • Bandwidth – the speed between devices • Delay – the time a packet needs to reach the destination • Jitter – the time between two packets • Packet loss – number of packets that are lost in the transmission The bandwidth is proportional with the number of users and the codec used. If you have a large number of users you need a high bandwidth or use a codec that uses less bandwidth (but the quality of the sound will decrease). Delay is important because voice packets need to arrive in real-time. You don’t want to have moments of silence on your phone until you get a response from the other side. Rules say that you need to have a delay of maximum 150ms. Because of the buffer delay in equipments, the packets arrive with a delay to each other. This is called jitter. You don’t want jitter because, like in the case of delay, you have real-time transmission and you don want words to arrive at the right time ( with no silence time or overlapping streams). The worse case is that you have packets dropped on the way to the destination. This would mean that whole words would be lost from the conversation. This is unacceptable so the rule is that the maximum number of packets lost is 2% (2 for every 100 packets sent). This is because a very small number of lost packets can be corrected using some advanced predictions and correction methods.

The migration from POTS (plain old telephony system) to VoIP (in other words from a circuit switched network to a packet switched network) depends mostly on the current data network. If you have a network that was properly designed, the migrations would take place easily, but if the network wasn’t made with QoS in mind, the voice network would still work but quality would be reduced. If you have the chance to design and implement the voice and data network from zero it would give you the opportunity to optimize the network to the maximum. Things that would assure that are: the correct use of the proper equipment, proper cabling, correct addressing, use of VLANs, implementation of QoS policies. Once the packet network is operational you need to decide of the deployment of voice equipment. You need to start at core level and decide what you need. At this point you need to decide if you want an independent VoIP network or one that interacts with the POTS. If you need to interact with the Public Switched Telephone Network (PSTN) you will need beside data networking knowledge, a good understanding of how the PSTN works. This could double your design and implementation work. You need to have equipment that interact with analog devices called gateways. These devices have analog ports (like FXO for connection to the POTS or FXS to connect to normal phones and E&M ports to connect to PBXs)

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In core of the voice network you have the Call Agent, the brain of the design. The Call Agent and the Gateway can sometimes be the same equipment. This is either a router equipped with the voice ports and VoIP software (like Cisco Call Manager Express) or a dedicated server with software like Asterisk, SER, openSER or Cisco Call Manager. You have the choice of a centralized (one point of control and administration) or a distributed design(several point of management). Each option has advantages and disadvantages. You need to balance factors like: size of the network, performance of the equipments, ease of administration and configuration to pick the best choice for your design. Last, you need to choose the end devices. These can be IP Phones, plain phones with ATA adapters or Softphones. You need to think about what the end users need and want. An IP Phone could be more expensive, but offers the most features. A plain telephone is cheap, but has the lest features and needs extra equipment. The softphone is also cheap and offers the features that an IP phone but it’s an application on a PC.

4.Conclusion The idea of VoIP is new in both data and telephone networking. It is still under development, but it has reached the point where enough equipment and protocols exist and are competitive with PSTN technology that has more then 100 years “of experience”. IP Telephony is already being implemented and we can see the results. Phone call costs for the home user have dropped considerably mostly because the migration from circuit switched to packet switched networks. The number of extra features (like voice mail) increased. But the implementation of a voice network requires a big investment from the one implementing it. Also it requires people with considerable knowledge in networking, telephony, quality of service and troubleshooting to design and implement such a network. Without any doubt, IP Telephony will replace the old telephone system. It is just a matter of time. We will change the way we see a phone call because of the things VoIP will offer us. It will be cost effective, it will reduce the number of equipment because we won’t have two networks (one for data and one for voice) bun one converged network. Already here, but not so far in the development process is Video over IP. This will overtake yet another network: the TV network. We will have Internet (data) Voice and Video in one single network.