SIP Proxy – is an IP network server responsible for call routing (call transfer to
another entity closer to the destination). There can be one or more SIP Proxy
units between the users.
SIP Registrar – is an IP network server responsible for user registration in a
certain network section. As a rule, SIP device registration is necessary for a user
to be accessible to the others on a certain phone number. SIP Registrar and SIP
Proxy are often installed on one and the same server.
RTP – is a protocol defining the standard packet (Real-Time Transport Protocol)
format for audio and video transmission in IP networks. intercom uses the 2N IP
RTP for audio and video stream transmission during a call. The stream
parameters (port numbers, protocols and codecs) are defined and negotiated
via the SDP (Session Description Protocol).
The intercoms support three ways of SIP signalling:2N IP
via the ( ), which is the most frequently used User Datagram Protocol UDP
unsecured signalling method
via the , which is less frequent, yet Transmission Control Protocol ( TCP)
recommended unsecured signalling method
via the protocol, where SIP messages are Transaction Layer Security (TLS)
secured against third party monitoring and modification
List of Parameters
The intercom Phone settings are arranged in five tabs:2N IP
SIP 1 and SIP 2 – complete SIP terminal settings
Calls – incoming and outgoing call settings
Audio – audio codec, DTMF transmission and other audio stream transmission
settings
Video – video codec, video resolution and other video stream transmission
settings
SIP 1 and SIP 2
The allow two independent SIP accounts (SIP 1 and SIP 2 tabs) to be 2N IP intercoms
configured. Thus, the intercom can be registered under two phone numbers, with two
different SIP exchanges and so on. Both the SIP accounts process incoming calls
equivalently. Outgoing calls are primarily processed by account 1, or, if account 1 is not
by account 2. Select the account number registered (due to SIP exchange error, e.g.),
for the phone numbers included in the phone directory to specify the account to be
used for outgoing calls (example: 2568/1 - calls to number 2568 go via account 1, sip:
1234@192.168.1.1 calls to sip uri via account 2).
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