What is SIP phone?

SIP Phone is the most popular Voice over IP (VoIP) standard combining all the elements necessary to provide a simple, super-inexpensive and high quality experience.

SIP is short for Session Initiation Protocol (SIP) which is a signaling protocol, widely used for voice and video calls over the Internet. SIP Phone enables two or more people to make phone calls to each other using the Internet to carry the call.

By using the Internet, you gain some distinct advantages over traditional telephone calls or PSTN (Public Switched Telephone Network):

• Call quality: Technology today means that SIP Phone calls on broadband are digital quality calls across the street or around the globe.
• Cost: SIP Phone to SIP Phone calls are always free and calls to old PSTN phones can also be relatively inexpensive compared to conventional PSTN to PSTN.
• Features: Because SIP Phone calls are part of the Internet you can get great features like free voicemail to email and connection to phone numbers in many places in the world no matter where you live.

SIP phones can be operated in a number of ways either as a program running on a computer, an adaptor that lets you connect an ordinary PSTN phone to the internet or a phone that can make internet calls.

A SIP phone provides call functions such as dial, answer, reject, hold/unhold, and call transfer.

What Do You Need To Use SIP Phone?

To use SIP phone you will need the following as a minimum:

• A hardware adapter, headset and microphone or softphone.
• A broadband connection to the Internet
• To make free in-network (SIP Phone to SIP Phone) calls, both sides of the call must have SIP Phone connection software.
• The same software can also be used to make inexpensive calls to any of the billions of non-SIP Phone (PSTN) world
• To receive calls on your SIP Phone from non-SIP (PSTN) phones, you need at least one Virtual Number.

Well their tune is good and it gives an idea of how SIP works.

The Technical Aspect of a SIP Phone System

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. In November 2000, SIP was accepted as a 3GPP signaling protocol and as a permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems.

As such the SIP Phone protocol is a TCP/IP-based Application Layer protocol. Within the OSI model it is sometimes placed in the session layer. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. It is a text-based protocol, sharing many elements of the Hypertext Transfer Protocol (HTTP), upon which it is based.

SIP Phones are the phones which are specifically designed to work as VOIP phones. SIP Phones can be considered as a network endpoint routing voice via the endpoints IP address. Usually SIP phones are configured outside of the usual PBX using vendor specific options.

SIP Phone systems are distinguished from a number of other VoIP signaling protocols by having roots in the IP community rather than the telecom industry.

SIP Phone clients typically use TCP or UDP to connect to SIP servers and other SIP endpoints. SIP is primarily used for voice or video calls. However, it can be used in any application where session initiation is a requirement including Event Subscriptions and Notification, Terminal mobility and so on.

SIP phones are typically programmed in esoteric configuration with software being able to be downloaded to each phone. When updating SIP phone software, the usual process is to edit the software files and then reboot the phone.