VoIP phones or IP phones use voice over IP technology to locate and transmit phone calls over IP networks, such as the Internet, instead of traditional public switched telephone networks (PSTN).
Digital IP based telephone services use control protocols such as Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP) or other proprietary protocols.
Video VoIP phone
Jenis
VoIP phones can be either simple software-based software or custom-made hardware devices that look like regular phones or cordless phones. The traditional PSTN phone is used as a VoIP phone with an analog phone adapter (ATA).
VoIP phones or apps may have many features that are not supported by analog phones, such as IDs such as e-mail for contacts that may be easier to remember than names or phone numbers, or easy sharing of contact lists between multiple accounts. Generally VoIP phone features follow Skype and other PC-based phone services, which have richer features but (because they rely on mainstream operating system 'IP support) audio problems associated with latency.
A competing view is that as the main operating system gets better on voice applications with the appropriate Quality of Service (QoS) warranty and 5G handoff (IEEE 802.21 etc.) becomes available from outdoor wireless carriers, netbooks and smartphones will only becomes the dominant interface. iPhone, Android, and QNX OS used on BlackBerry phones 2012-and-later are generally able to perform VoIP performance even on small devices that charge the battery. They also typically support USB but not Ethernet or Power over Ethernet interfaces, at least at the end of 2011. According to this view, smartphones become dominant VoIP phones because they function both indoors and outdoors and shift basic stations/protocols easily to trade. off access fee and call clarity and other personal factors to the user, and the PoE/USB VoIP phone is thus a transition device.
Maps VoIP phone
Components and software
VoIP phones consist of hardware and software components. The software requires standard network components such as TCP/IP network stack, client implementation for DHCP, and Domain Name System (DNS). In addition, the arrangement of VoIP signaling protocols, such as for Session Initiation Protocol (SIP), H.323, Call Control Control Protocol (Cisco), and Skype, are required. For media streams, Real-Time Transport Protocol (RTP) is used in most VoIP systems. For voice and media encoding, various 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. The user interface software controls the operation of hardware components, and can respond to user actions with messages to the display screen.
STUN client
To enable VoIP communication, SIP/RTP packets must be used and STUN clients will be a key component for VoIP communications with SIP/RTP package management. A Session Traversal Utilities for NAT (STUN) client is used on some SIP-based VoIP phones because the firewall on the network interface sometimes blocks SIP/RTP packets. Some special mechanisms are needed in this case to enable routing SIP packets from one network to another. STUN is used in some sip phones to enable SIP/RTP packets to cross the boundaries of two different IP networks. A packet becomes unroutable between two sip elements if one of the networks uses private and other IP address ranges within the public IP address range. Stun is a mechanism for activating this border traversal. There is an alternative mechanism for NAT traversal, STUN is just one of them. STUN or other NAT traversal mechanisms are not required when two SIP phone connections can be routed to each other and there is no firewall in between.
DHCP Client
The DHCP client software simplifies the device connection to the IP network. The software automatically configures VoIP network and service parameters.
Hardware
The overall hardware can look like a phone or a mobile phone. VoIP phones have the following hardware components.
- Mobile speaker and microphone
- Keypad or touchpad to enter phone number and text (not used for ATA).
- Show hardware to provide user input feedback and display caller-id/message (not used for ATA).
- General purpose macro processor (GPP) for processing messaging applications.
- Voice engine or digital signal processor (DSP) for processing RTP messages. Some IC manufacturers provide GPP and DSP in one chip.
- AD and DA converter: To convert sound to digital data and vice versa.
- Ethernet or wireless network hardware to send and receive messages on the data network.
- Power source - 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 PSTN.
Other devices
There are several phones and PDAs that support Wi-Fi that have pre-installed SIP client software, or are capable of running IP phone clients, including most smartphones.
The analog phone adapter provides an interface for traditional analog phones to voice-over-IP networks. They connect to the Internet or local area network using an Ethernet port and have jacks that provide a standard RJ11 interface for local analog loops.
Another type of gateway device serves as a simple GSM base station and a regular cell phone can connect to this and make VoIP calls. While the license required to run one of these in most of these countries can be useful on boats or remote areas where low-powered gateways emitting at unused frequencies are likely to escape the attention.
Some VoIP phones also support PSTN phone line directly.
General functions and features
- Call ID Display
- Call transfer and call suspension
- Call using name/ID (different from speed dialing because there is no number stored on client)
- The stored directory and local network based
- Conference calls and multiparty calls
- Call park
- Call blocking feature.
- Support for multiple VoIP accounts - phones can register with more than one VoIP server/provider.
- Accounts are usually organized and stored on the phone itself. A more sophisticated feature is the dynamic download of account settings, also known as "mobility extensions". This feature allows settings stored on the server to be downloaded to the phone, based on user login. The user enters the phone and the phone becomes a user extension. This feature requires the client (phone) and server, usually in the context of an integrated communication system.
- Secure encrypted communication
Technology issues
- Requires Internet access to call outside the local area network (LAN) unless a compatible local private branch (PBX) exchange is available to handle calls to and from off the beaten path.
- VoIP phones and routers rely on primary electricity for power, unlike PSTN phones, which are supplied with power from telephone switches. However, this can be reduced by installing the UPS. The Power over Ethernet interface simplifies this greatly because power can be "injected" on each connector (especially in passive mode where all devices are drawing the same voltage) or on the router. This is the main reason that dominant call centers and VoIP PBX systems rely on PoE exclusively, but UPS and PoE only help if upstream Internet providers also have a reliable backup power.
- The IP network, especially the residential Internet connection is very easy to jam. This may cause a worse sound quality or call to be completely removed.
- VoIP phones, like other network devices, may be subject to denial-of-service attacks and other attacks especially if the device is assigned a public IP address. This is very important as a problem with wireless devices that use the 802.11 protocol.
- Due to latency caused by protocol overhead and other factors, they do not work well on satellite Internet, analog cells ("edge" networks) and other high latency Internet connections. Applications that are very latency sensitive (music, remote device control) in 2012 can not exploit the VoIP protocol.
- Exclusive vendors like Skype and Google Voice focus on improving call quality among their own users to develop their user base, which to some degree compete and contradict the purpose of a better connection from Skype to Google Voice, or from both to PSTN and existing mobile network. The best codecs tend to be exclusive and unlicensed to competitors, slowing industry growth and causing incompatibility.
- A variety of schemes exist to allow one Internet phone user to talk to another completely through the Internet and without paying PSTN calls. Some are based on SIP addresses, some on proprietary protocols such as webcam or Internet chat apps. While it is not uncommon for two clients from the same voice provider over IP to talk to each other online for free, various Internet phone applications often do not speak directly to each other - requiring calls to the PSTN gateway and returning at full toll rates.
- Some Internet-to-Internet calling schemes use non-numeric names for user names, gateways or providers. Any valid characters in the e-mail address can be used in SIP addresses, for example, but VoIP phones with standard phone keys can only dial numbers. Various work methods (such as e164.arpa directory or SIP Broker) exist to associate names with numbers.
See also
- List of SIP software
- IP Multimedia Subsystem (IMS)
- Media phone
- Mobile VoIP
References
Source of the article : Wikipedia
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