Introduction to audio Acoustics, speakers and audio terminology OCTOBER
24 Jul 2018
Axis offers intelligent security solutions that enable a smarter, safer world. As the market leaderin network video, Axis is driving the industry by continually launching innovative networkproducts based on an open platform - delivering high value to customers through a globalpartner network. Axis has long-term relationships with partners and provides them withknowledge and ground-breaking network products in existing and new markets.
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:bob@biloxi.ex.com [sip:
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:
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:
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.
Contd...
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Introduction to audio Acoustics, speakers and audio terminology OCTOBER
Axis Communications (UK Ltd)
Website
Email Us
T: 01582 283760
1 Capability Green
LU1 3LU
Luton, Bedfordshire
United Kingdom
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