VoIP Phone - Wikipedia
VoIP Phone - Wikipedia
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting
telephone calls over an IP network, such as the Internet.[1] This is in contrast to a standard
phone which uses the traditional public switched telephone network (PSTN).
Digital IP-based telephone service uses control protocols such as the Session Initiation
Protocol (SIP), Skinny Client Control Protocol (SCCP) or various other proprietary protocols.[2]
Types
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 and other PC-based phone services, which have richer
feature sets but may experience latency-related problems because they rely on mainstream
operating systems' IP and audio support.
A VoIP telephone consist of the hardware and software components. The software requires
standard networking components such as a TCP/IP network stack, client implementation for
DHCP, and the Domain Name System (DNS).
In addition, a VoIP signalling protocol stack,
such as for the Session Initiation Protocol (SIP), H.323, Skinny Client Control Protocol
(Cisco), and Skype, is needed.
For media streams, the Real-time Transport Protocol (RTP) is
used in most VoIP systems. For voice and media encoding, a variety of coders are available,
such as for audio: G.711, GSM, iLBC, Speex, G.729, G.722, G.722.2 (AMR-WB), other audio
codecs, and for video H.263, H.263+, H.264. User interface software controls the operation of
the hardware components, and may respond to user actions with messages to a display
screen.
STUN client
To enable the VoIP communications, the SIP/RTP packets should be utilised and STUN client
would be the key component for VoIP communications with management of the SIP/RTP
packets. A Session Traversal Utilities for NAT (STUN) 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.
DHCP client
Hardware
The overall hardware may look like a telephone or mobile phone. A VoIP phone has the
following hardware components
Keypad or touchpad 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)
A voice engine or a digital signal processor (DSP) to process RTP messages. Some IC
manufacturers provides GPP and DSP in single chip
Ethernet or wireless network hardware to send and receive messages on data network
Power source - a battery or DC/AC source; some VoIP phones receive electricity from
Power over Ethernet
Some VoIP phones include an RJ-11 port to connect the phone to the PSTN
Other devices
There are several Wi-Fi enabled mobile phones and PDAs that have pre-installed SIP client
software, or are capable of running IP telephony clients, including most smartphones.
Analog telephone adapters provide an interface for traditional analog telephones to a voice-
over-IP network. They connect to the Internet or local area network using an Ethernet port
and have jacks that provide a standard RJ11interface for an analog local loop.
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.
Caller ID display
Dialing using name/ID (differs from speed dial in that no number is stored on the client)
Call park
Support for multiple VoIP accounts – the phone may register with more than one VoIP
server/provider.
Accounts are usually set and memorized on the phone itself. A more sophisticated
feature is dynamic download of account settings, also known as "extension mobility".
This feature allows settings stored on a server to be downloaded to the phone, based
on user login. The user logs into the phone and that phone becomes the user's
extension. This feature requires both a client (phone) and a server, usually in the
context of unif[3]ied communications systems.
See also
Mobile VoIP
References
1. Chris Yackulic. "Why Your Small Business Needs a VoIP Phone System" (https://www.androidheadline
s.com/2018/06/why-your-small-business-needs-a-voip-phone-system.html) . Android Headlines.
2. SBWire. "Internet Protocol (IP) Telephony Market to 2022: Top Companies, Trends, Deployment,
Growth Factors Details for Business Development" (http://www.digitaljournal.com/pr/3812941) .
Digital Journal.
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Last edited 1 month ago by ThisIsNotABetter
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