An IP Phone is a telephone that operates on a data network instead of traditional telecoms networks. It is designed as a networking device, running over the TCP/IP suite of protocols, where the speech is digitized and encapsulated in a series of IP (Internet Protocol) packets for transmission over copper or fibre transmission lines. An IP Phone is more traditionally referred to as a VoIP (Voice over Internet Protocol) phone because it is because it uses technology that come under the general heading of VoIP.
VoIP telephone calls can be made over the Internet or private IP networks run by companies and organisations. Just like traditional telephony, VoIP needs to use signalling and control protocols for the setup, control and clearing of the calls. SIP (Session Control Protocol) is the most popular control protocol in use today, and most commercial IP Phones support SIP. On the Internet, Skype is the most popular Voice over IP system, but it is proprietary and doesn’t support Industry standard protocols.
Voice over IP technologies also support multimedia as well as voice, and there are many video applications that are supported by VoIP.
In order for the IP phone to operate in a digital network then the analogue speech input via the microphone must be digitised so that it can be packetized ready for transmission on the IP network. There are many variations of the processes for digitization and they generally come under the heading of a Codec (Coder / Decoder). One codec that is also used in the digital telephone networks is the ITU-T G.711 codec, which digitises the analogue speech through a process of sampling and quantization. The first part of the process is PCM (Pulse Code Modulation), where the analogue speech is sampled at a constant rate of 8khz to produce 8 bit binary words which represent the original analogue speech. It must be noted that when the binary data is reconverted to an analogue form at the receiver, there is a drop in quality due to what we call quantization error. Other popular codecs include G.729, G.726 and G.723, and a lot of IP phones support multiple codecs.
Traditional telephones can be used on VoIP systems provided an intermediate device is used to provide conversion and connection to the data network. These devices come under the heading of Analogue Telephone Adapters or ATA.
VoIP Phones can communicate directly with each other over a data network, but they are often used with an IP PBX which is a hardware or software device that emulates the operation of a traditional PABX as used in the Telecommunications industry. The IP PBX will provide such functionality as registration services for the phones, proxy services and supplementary services such as Call Divert, Call Forwarding, Ring Groups and Voice mail. Indeed an IP PBX should be able to provide all the services currently supplied by traditional PABXs.
Just like any other IP networking device such as a PC or Server, an IP Phone needs some basic configuration parameters such as a valid IP Address, Network Mask an MAC Address. The phones can be manually configured with IP Addresses or can be supplied with IP Addresses from a DHCP (Dynamic Host Configuration Protocol) server like most network devices in our IP networks today.
VoIP Phones need a power supply, just like any other network device and in recent years technology has been developed to supply these phones with DC power direct from network connection with the local switch. The power is passed down the Ethernet patch cables using the previously unused copper pairs within the cable. A standard known as IEEE 802.3af was released in 2003 and it can provide up to 15.4 watts of DC power to each device direct from the switch port. The standard was updated in 2009 as IEEE 802.3at to provide up to 25.5 watts of power per device.
IP Phones have to compete with other network protocols in our packet based data networks, but because they are conveying Real-Time information in the form of speech, they must be afforded some kind of special handling or prioritisation in the network. The methods of affording some network protocols prioritisation over others is known as QoS (Quality of Service). In other words, we must minimise the amount of delay across the network for the IP Packets containing the digitised speech. Too much delay across the network will result in a two way conversation being difficult to manage, rather like talking over a Satellite phone where the delay to and from the satellite makes the conversation difficult unless the users are disciplined and understand the problems. Variable delay, often referred to as packet switching delay, is where the delay between individual packets varies in sympathy with network conditions. This type of delay normally comes under the heading of jitter, and there are a number of techniques that can be employed to alleviate the effects of jitter.
There are many manufacturers of IP Phones, and if you are purchasing a VoIP Phone then you must ensure that it supports the standards supported elsewhere in your network such as the correct codecs, inline power standard and of course the features that you require.