R&S 4200 Series, Руководство по эксплуатации

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R&S XU 4200 Configuring with the R&S ZS 4200
6166.5368.02.01 3.20
VoIP radio URI TX This is the unified identifier for VoIP communication of the TX module. This
identifier consists of two parts concluded with the “@” sign
user@<IP Address>
or user@<Full Qualified Domain Name> eg.
VoIP PTT Summation
Mode This setting is used for enabling/disabling of PTT summation for multiple
RTP audio streams.
VoIP Jitter Buffer
Prefetch Value In order to compensate network delays, the VoIP implementation of the
radio uses a so called Jitter Buffer. The adjustment of this buffer controls
the delay between sender and receiver.
Note: An inadequate value can cause interrupted audio flow. The optimal
value is system-specific and has to be found during the system-setup.
Note: This value influences the maximum confirmation delay. If the value is
greater than 20 ms, the maximum confirmation delay is not compliant to
ED-137-1.
Emergency VoIP URI
ACL The VoIP mode of the radio offers the possibility to configure the access for
VoIP connections. Each entry contained in the URI ACL grants access to
establish VoIP connections to the radio. In default configuration the URI
ACL is a whitelist. This means that accessing the radio via VoIP is not
restricted. The URI ACL can contain up to 20 entries with a maximum of 64
characters per entry.
Emergency VoIP URI ACL stores URI of the VoIP clients which are allowed
to access the radio with either normal or emergency call priority.
Normal VoIP URI ACL Compared to Emergency VoIP URI ACL the Normal VoIP URI ACL stores
URI of the VoIP clients which are allowed to access the radio with normal
call priority.
Permit Only ACL URI
Call This configuration parameter enables or disables acceptance of the VoIP
session requests which only have URIs matching the VoIP URI ACL lists.
Coupling PTT
Summation This parameter enables or disables additional summing of the VoIP RTP
stream of the SIP call-type “coupling” and PTT-type “coupling” together
with the RTP streams selected for the transmission.
Note: The setting of this parameter will end all active SIP sessions.
Primary Domain Name
Server This parameter is used to setup an IP address of a Domain Name Server.
Secondary Domain
Name Server This parameter is used to setup an IP address of a Domain Name Server
which is used for backup purposes.
RTP Port Range Start
The real time transport protocol uses several IP ports for communication
with VCS or the R&S GB4000V. This parameter sets the start port for the
port range which can be used for VoIP audio streams.