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v1.02
Admin and Installation Guide M700, M300 Firmware Update
Setting Description
NAT Adaptation Default: Disabled. Ensures that all SIP messages go directly to the NAT
gateway. NOTE:
Enabled: When the base station receives a SIP response to a REGISTER
request with a “Via” header that includes the “received” parameter (e.g.:
Via: SIP/2.0/UDP 10.1.1.1:4540;received=68.44.20.1), it will adapt its contact
information to the IP address from the "received" parameter. The base
station will then send another REGISTER request with the updated contact
information.
Disabled: The base station will ignore the "received" parameter.
Registrar SIP server proxy DNS or IP address. Specifying the port number is optional.
Outbound proxy Session border controller DNS or IP address. Alternatively SIP server
outbound proxy address. Set the outbound proxy to the address and port
of the private NAT gateway so that SIP messages are sent via this gateway.
Format: addr:port
Default value: Blank
Conference server TBA
Reregistration time (s) The “expires” value signaled in SIP REGISTER requests. This value indicates
how long the current SIP registration is valid, and hence it specifies the
maximum time between SIP registrations for the SIP account.
Permitted value(s): Maximum 65636 seconds. A value below 60 seconds is
not recommended.
SIP Session Timers RFC 4028. A “keep-alive” mechanism for calls. The session timer value
specifies the maximum time between “keep-alive” or more correctly session
refresh signals. If no session refresh is received when the timer expires, the
call will be terminated.
Default: Enabled.
Session timer value (s) Default value: 1800 seconds.
Permitted values: Minimum 90 sec, maximum 65636 sec.
If the setting is disabled, session timers will not be used.
SIP Transport Default: UDP. Other options: TCP, TLS 1.0
Signal TCP Source Port When SIP Transport is set to TCP or TLS, respectively, a TCP or TLS connection
will be established for each SIP extension. The source port of the connection
will be chosen by the TCP stack, and the local SIP port parameter specified
in the SIP/RTP settings will not be used. The “Signal TCP Source Port”
parameter specifies whether the used source port shall be signaled explicitly
in the SIP messages.
Default: Enabled.
Use one TCP connection per
SIP extension
When using TCP or TLS as SIP transport, choose whether the base station is
to establish one TCL/TLS each connection for each SIP extension or one TCL/
TCS connection that all SIP extensions will use. NOTE: If TLS is used and the
SIP server requires client authentication (i.e., requests a client certificate), this
setting must be disabled.
Settings: Disabled (one TCP/TLS connection for all SIP extensions);
Enabled (one TCP/TLS per SIP extension)
Default: Disabled
RTP from own base station If disabled, the RTP stream will be sent from the base station where the
handset is located. I enabled, the RTP stream will always be sent from the
base station where the handset is registered.
Default: Enabled
Keepalive Default: Enabled. When enabled, a new keepalive message will be sent out to
the Registrar/Proxy port after 30 seconds in order to have the port stay open
and the phone remain reachable.
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