AXIS C8210 Network Audio Amplifier
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Session Initiation Protocol (SIP)
The Session Initiation Protocol (SIP) is used to set up, maintain and terminate VoIP calls. You can make calls between two or more
parties, called SIP user agents. To make a SIP call you can use, for example, SIP phones, softphones or SIP-enabled Axis devices.
The actual audio or video is exchanged between the SIP user agents with a transport protocol, for example RTP (Real-Time
Transport Protocol).
You can make calls on local networks using a peer-to-peer setup, or across networks using a PBX.
Peer-to-peer SIP (P2PSIP)
The most basic type of SIP communication takes place directly between two or more SIP user agents. This is called peer-to-peer SIP
(P2PSIP). If it takes place on a local network, all that’s needed are the SIP addresses of the user agents. A typical SIP address in this
case would be
sip:<local-ip>
.
Example
SIP
sip:192.168.1.101
sip:192.168.1.100
You can set up a SIP-enabled phone to call an audio device on the same network using a peer-to-peer SIP setup.
Private Branch Exchange (PBX)
When you make SIP calls outside your local IP network, a Private Branch Exchange (PBX) can act as a central hub. The main
component of a PBX is a SIP server, which is also referred to as a SIP proxy or a registrar. A PBX works like a traditional switchboard,
showing the client's current status and allowing for example call transfers, voicemail, and redirections.
The PBX SIP server can be set up as a local entity or offsite. It can be hosted on an intranet or by a third party provider. When you
make SIP calls between networks, calls are routed through a set of PBXs, that query the location of the SIP address to be reached.
Each SIP user agent registers with the PBX, and can then reach the others by dialing the correct extension. A typical SIP address in
this case would be
sip:<user>@<domain>
or
sip:<user>@<registrar-ip>
. The SIP address is independent of its IP
address and the PBX makes the device accessible as long as it is registered to the PBX.
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