Introduction to audio Acoustics, speakers and audio terminology OCTOBER

24 Jul 2018

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Introducing SIP

Session Initiation Protocol (SIP) is used to initiate, maintain and terminate multimedia sessions between different parties. Usually these sessions consist of audio, but sometimes they consist of video. SIP is the standard protocol used in Voice over IP (VoIP) applications and unified communication platforms. SIP is for example supported by AXIS C3003-E Network Horn Speaker and AXIS A8004-VE Network Video Door Station and constitute a new way to connect, integrate and control your Axis network products.

How does it work?

In order to communicate using SIP at least two SIP clients are needed. A SIP client can be a SIP hardphone, softphone, mobile client or SIP enabled Axis product.

Each SIP client is assigned its own SIP address. A SIP address looks like an email address with a “sip:” prefix, for example, sip:[email protected] [sip:@]. This identifier can be used across a number of devices and is analogous to the use of a telephone number linked to a SIM card that can be used in a number of devices.

Peer-to-peer setup – the simple way

A SIP system can take many forms. In its simplest form, the system consists of two or more SIP User Agents (UA) communicating directly with each other. This can be called a peer-to-peer setup, a direct call setup or local setup. A typical SIP address in this case would look like sip:, for example, sip:192.168.0.90

In order to make a peer-to-peer call from one UA to another on a local network all that is needed is the SIP address containing the unit’s IP address. Note that not all SIP clients support peer-to-peer calls.

Using a SIP server (PBX) – adding more possibilities

A SIP-based VoIP infrastructure scales very well. The next step up in size is to use a registrar, Private Branch Exchange (PBX) or SIP server, as a central hub. The SIP UAs register with the registrar and can then reach each other’s SIP UAs simply by dialling an extension on the PBX. A typical SIP address in this case would look like sip:@ or sip:@ such as sip:6007@ mysipserver.net. A PBX works like a traditional switchboard, showing the clients’ current status, allowing call transfers, voicemail, redirections and much more.

A SIP server usually includes proxy, registrar and redirect functionality. Proxies route calls and provide additional logic to incoming calls. Registrars accept register requests and act as a location service for the domain that it handles. Redirect servers redirect the client to contact an alternative SIP address.

The SIP server can be set up as a local entity or it can be located offsite. It can be hosted on an intranet or it can be hosted by a third party provider. When making SIP calls across sites, calls are normally initially routed through a set of SIP proxies. These proxies query the location of the SIP address to be reached.

In order to use your device with a SIP server you need to create an account on the server with a specified user ID and password. To register your device to the server, you need to set up an account on the device entering the server address, user ID and password.

Using SIP trunks – assigning a telephone number

Using a SIP trunk, SIP UAs can even be switched to the traditional analog telephone network (PSTN). This way you can assign a traditional telephone number to the SIP UA.

When used with a SIP trunk the device connects to the server in the way described above. The service provider will usually charge extra for external numbers.

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