A VoIP phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP network such as that of a company. The phones use control protocols such as Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP) or one of various proprietary protocols such as that used by Skype.
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VoIP phones can be simple software-based softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as VoIP phones with analog telephone adapters (ATA).
Two combined signal-and-power wired cable interfaces are in common use to communicate between computer networks (or computers) and physically separate VoIP phones: USB and Power over Ethernet. The latter is preferred in industry such as large scale call centers or PBX replacement applications, because PoE has the following advantages over USB:
For these reasons, USB and softphone PC applications are considered transitional by many industrial users and makers of larger telephone switches, used only to build markets for VoIP that will eventually shift over to the more robust PoE technology shipped by Cisco, Siemens, Alcatel and other large switch makers.
A VoIP phone or application may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers, or easy sharing of contact lists among multiple accounts. Generally the features of VoIP phones follow those of Skype, Google Voice and other PC-based phone services, which have richer feature sets but (because they rely on mainstream operating systems' IP support) latency-related audio problems.
A competing view is that as mainstream operating systems become better at voice applications with appropriate Quality of Service (QoS) guarantees and 5G handoff (IEEE 802.21 etc.) becomes available from outdoor wireless carriers, netbooks and smartphones will simply become the dominant interfaces. iPhone, Android and the QNX OS used in 2012-and-later BlackBerry phones are generally capable of VoIP performance even on small battery-charged devices. They also typically support the USB but not Ethernet or Power over Ethernet interfaces, at least as of late 2011. According to this view, the smartphone becomes the dominant VoIP phone because it works both indoors and outdoors and shifts base stations/protocols easily to trade off access costs and call clarity and other factors personal to the user, and the PoE/USB VoIP phone is thus the transitional device.
[1] The overall hardware may look like a telephone or mobile phone. A VoIP phone has the following hardware components.
For wireless VoIP phones
There are several Wi-Fi enabled mobile phones and PDAs that come pre-loaded with SIP clients, or are capable of running IP telephony clients. Some VoIP phones also support PSTN phone lines directly.
Analog telephony adapters are connected to the internet or Local area network using an Ethernet port and have sockets to connect one or more PSTN phones. Such devices are sent out to customers who sign up with various commercial VoIP providers allowing them to continue using their existing PSTN based telephones.
Another type of gateway device acts as a simple GSM base station and regular mobile phones can connect to this and make VoIP calls. While a license is required to run one of these in most countries these can be useful on ships or remote areas where a low-powered gateway transmitting on unused frequencies is likely to go unnoticed.
A STUN (Session Traversal Utilities for NAT) client is used on some SIP-based VoIP phones as firewalls on network interface sometimes block SIP/RTP packets. Some special mechanism is required in this case to enable routing of SIP packets from one network to other. STUN is used in some of the sip phones to enable the SIP/RTP packets to cross boundaries of two different IP networks. A packet becomes unroutable between two sip elements if one of the networks uses private IP address range and other is in public IP address range. Stun is a mechanism to enable this border traversal. There are alternate mechanisms for traversal of NAT, STUN is just one of them. STUN or any other NAT traversal mechanism is not required when the two SIP phones connecting are routable from each other and no firewall exists in between.
A DHCP client may be used to configure the TCP/IP parameters and server details if a network segment uses dynamic IP address configuration. The DHCP client then provides central and automatic management of VoIP phones configuration.