User's Manual

SME VoIP System Guide, Version 2.6
Proprietary and Confidential
Chapter:
SME VoIP Administration Interface
37
SIP Transport
UDP
Select UDP, TCP, TLS 1.0
Signal TCP
Source Port
Disabled
When SIP Transport is set to TCP or TLS, 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 hence the local SIP port parameter,
specified within the SIP/RTP Settings (see 5.5.5) will not
be used. The “Signal TCP Source Port” parameter
specifies if the used source port shall be signaled
explicitly in the SIP messages.
Use One
TCP/TLS
Connection
per SIP
Extension:
Disabled
When using TCP or TLS as SIP transport, choose if a
TCL/TLS connection
shall be established for each SIP extension or if the base
station shall establish one connection which all SIP
extensions use. Please note that if TLS is used and SIP
server requires client authentication (and requests a
client certificate), this setting must be set to disabled.
0: Disabled. (Use one TCP/TLS connection for all SIP
extensions)
1: Enabled. (Use one TCP/TLS connection per SIP
extensions).
RTP from
own base
station:
Disabled
If disabled RTP stream will be send from the base, where
the handset is located. By enable the RTP stream will
always be send from the base, where the SIP registration
is made.
This setting is typically enabled for operation with Cisco.
Keep Alive
Enabled
This directive defines the window period (30 sec.) to
keep opening the port of relevant NAT-aware router(s),
etc.
Show
Extension on
Handset Idle
Screen
Enabled
If enabled extension will be shown on handset idle
screen.
Hold
Behaviour
RFC 3264
Specify the hold behaviour by handset hold feature.
RFC 3264: Hold is signalled according to RFC 3264, i.e. the
connection information part of the SDP contains the IP
Address of the endpoint, and the direction attribute is
sendonly, recvonly or inactive dependant of the context
RFC 2543: The ”old” way of signalling HOLD. The
connection information part of the SDP is set to 0.0.0.0,
and the direction attribute is sendonly, recvonly or
inactive dependant of the context
Attended
Transfer
Behaviour
Hold 2
nd
Call
When we have two calls, and one call is on hold, it is
possible to perform attended transfer. When the transfer
soft key is pressed in this situation, we have traditionally
also put the active call on hold before the SIP REFER
request is sent. However, we have experienced that some
PBXes do not expect that the 2nd call is put on hold, and
therefore attended transfer fails on these PBXes.
The "Attended Transfer Behavior" feature defines
whether or not the 2nd call shall be put on hold before
the REFER is sent.