R&S 4200 Series Operating Manual page 72

R&S XU 4200
VoIP radio URI TX
VoIP PTT Summation
Mode
VoIP Jitter Buffer
Prefetch Value
Emergency VoIP URI
ACL
Normal VoIP URI ACL
Permit Only ACL URI
Call
Coupling PTT
Summation
Primary Domain Name
Server
Secondary Domain
Name Server
RTP Port Range Start
6166.5368.02.01
This is the unified identifier for VoIP communication of the TX module. This
identifier consists of
user@<IP Address>
user@muenchen.rohde-schwarz.de.
The URI can be up to 64 characters. Characters and other naming
conventions have to follow the RFC 3986 rules. Using the reserved
characters makes the URI invalid and thus the VoIP operation is not
possible.
The default value of URI TX module is tx@192.168.52.102.
This setting is used for enabling/disabling of PTT summation for multiple
RTP audio streams.
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.
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.
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.
This configuration parameter enables or disables acceptance of the VoIP
session requests which only have URIs matching the VoIP URI ACL lists.
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.
This parameter is used to setup an IP address of a Domain Name Server.
This parameter is used to setup an IP address of a Domain Name Server
which is used for backup purposes.
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.
3.20
Configuring with the R&S ZS 4200
two parts concluded
or user@<Full Qualified Domain Name>
with the "@" sign
eg.