UNIVERGE SV8100
Issue 5.0
General Description Manual
2 - 37
ITL-12D-1 (BK) TEL/ITL-12D-1 (WH) TEL
ITL-12PA-1 (BK) TEL
ITL-24D-1 (BK) TEL/ITL-24D-1 (WH) TEL
ITL-32D-1 (BK) TEL/ITL-32D-1 (WH) TEL
ITL-32OC-1 (BK) TEL
IP Single Line Telephone (SIP)
SIP (Session Initiation Protocol) is used for Voice over Internet Protocol. It is defined
by the IETF (Internet Engineering Task Force) RFC3261. Other RFC designations,
such as RFC 3842, refer to a later implementation of SIP and may be supported by
the UNIVERGE SV8100. Commonly called SIP Station, this feature is used for IP
Stations using SIP.
SIP analyzes requests from clients and retrieves responses from servers, then sets
call parameters at either end of the communication, handles call transfer, and
terminates. Typically, such features, including but not limited to Voice over IP
services, are available from an SIP service provider.
IP Trunk – (SIP) Session Initiation Protocol
The UNIVERGE SV8100 IP Trunk SIP package sends the real time voice over the
corporate LAN or WAN. The voice from the telephone is digitized and then put in
frames to be sent over a network using Internet protocol.
Using VoIP equipment at a gateway (a network point that acts as an entrance to
another network), the packetized voice transmissions from users in the company are
received and routed to other parts of the company intranet (local area or wide area
network) or they can be sent over the Internet using CO lines to another gateway.
The PZ-32IPLA/PZ-64IPLA/PZ-128IPLA is
an
interface that can provide IP trunks and
Tie Lines. It can operate in the following modes:
COI
COID
DID
TLI
DTI
IP Trunk – H.323
H.323 is an International Telecommunication Union (ITU) standard for Packet Based
Multimedia Communication Systems. The UNIVERGE SV8100 can use H.323 to
connect to another UNIVERGE SV8100 system or a third-party product.
Summary of Contents for DTL-12BT-1 - DT330 - Plus BCH
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