Alvarion SIP R2J Server User Manual


 
Telephone Menu
Voice Gateways System Manual 49
Encode SIP URI with
user parameters
User=Phone will be inserted in the Contact field of SIP uniform resource
identifier (URI).
Encode default port in
SIP URI
Include default port in SIP uniform resource identifier (URI) even though it
is not mandatory according to standard.
Include default port in
INVITE
Include default port in the INVITE even though it is not mandatory
according to standard
Send INVITE with timer
header value
If the called user agents (UA) or the SIP Proxy Server (SPS) requires a
session timer for a requested session and the calling UA does not include
the Session-Expires header in the INVITE message, then the called UA or
the SPS may reject the request with a 487-request failure message. If the
use of a session timer is desirable but optional for the session and the
calling UA does not include the Session-Expires header in the INVITE then
the called UA or SPS may add a Session-Expires header to the next
session setup message. In this case, the SPS shall add the
Session-Expires header to the INVITE message and the called UA shall
add the Session-Expires header to the 200 OK response message. The
range for the timer header value is from 1 to 999.
SIP Session timer value The SIP Session Timer Support feature adds the capability to periodically
refresh Session Initiation Protocol (SIP) sessions by sending repeated
INVITE requests. The repeated INVITE requests, or re-INVITEs, are sent
during an active call leg to allow user agents (UA) or proxies to determine
the status of a SIP session. Without this keep alive mechanism, proxies
that remember incoming and outgoing requests (stateful proxies) may
continue to retain call state needlessly. If a UA fails to send a BYE
message at the end of a session or if the BYE message is lost because of
network problems, a stateful proxy does not know that the session has
ended. The re-INVITES ensure that active sessions stay active and
completed sessions are terminated. The range for the timer value is from 1
to 999 seconds.
Use NOTIFY message to
keep alive the session
with SIP proxy every X
seconds
The gateway will send a SIP NOTIFY message to the SIP proxy at the
configured interval. These messages can keep the connection with SIP
proxy alive, as well as the NAT port mapping when the Voice Gateway is
behind NAT.
The range is: 0-99999
Default interval: 15 seconds
Table 3-9: SIP Extensions Page Parameters
Parameter Description