An IP phone uses Voice over IP 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, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. There also exist the possibility to reuse ordinary PSTN phones as IP phones, with analog telephony adapters (ATA).

It 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.
Elements of an IP phone

1. Hardware
2. DNS client
3. STUN client
4. DHCP client (not commonly used)
5. Signalling stack (SIP, H323, Skinny, Skype, or others)
6. RTP Stack
7. Codecs (Audio codecs such as G.711, GSM, iLBC, Speex, G.729 ,G.722, G.722.2 (AMR-WB) etc. Video codecs such as H.263, H.263+ and H.264)
8. User interface
The overall hardware may look like a telephone or mobile phone. An IP phone has the following hardware components.

- Speaker/ear phone and microphone
- Key pad/touch pad to enter phone number and text (not used for ATAs).
- Display hardware to feedback user input and show caller-id/messages (not used for ATAs).
- General purpose processor (GPP) to process application messages.
- A voice engine or a Digital signal processor to process RTP messages. Some IC manufacturers provides GPP and DSP in single chip.
- ADC and DAC converters: To convert voice to digital data and vice versa.
- Ethernet or wireless network hardware to send and receive messages on data network.
- Power source might be a battery or DC source. Some IP phones receive electricity from Power over ethernet.

For wireless IP phones

- battery
- Wireless network interface controller

Other devices

There are several WiFi enabled mobile phones and PDAs that come pre-loaded with SIP clients or are at least capable of running IP telephony clients. Some IP phones may also support PSTN phone lines directly.
Gateway devices

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 IP 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.