VoIP phone

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, instead of the traditional public switched telephone network (PSTN). 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, instead of 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. VoIP phones can be simple software-based softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Traditional PSTN phones are used as VoIP phones with analog telephone adapters (ATA). 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 (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. 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 Call 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. 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.