Hardware of a stand alone IP phone
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.
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.
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 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.
DHCP client may be used to configure the TCP/IP parameters and server details if network segment uses dynamic IP address configuration. DHCP client then provides central and automatic management of IP phones configuration.