Learn More; Voice Over Ip (Voip) - Axis I8016-LVE User Manual

AXIS I8016-LVE Network Video Intercom

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Voice over IP (VoIP)

Voice over IP (VoIP) is a group of technologies that enables voice communication and multimedia sessions over IP networks, such as
the internet. In traditional phone calls, analog signals are sent through circuit transmissions over the Public Switched Telephone
Network (PSTN). In a VoIP call, analog signals are turned into digital signals to make it possible to send them in data packets
across local IP networks or the internet.
In the Axis product, VoIP is enabled through the Session Initiation Protocol (SIP) and Dual-Tone Multi-Frequency (DTMF) signaling.
Example
When you press the call button on an Axis intercom, a call is initiated to one or more predefined recipients. When a recipient replies,
a call is established. The voice and video is transferred through VoIP technologies.
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
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