AudioCodes Mediant 1000 User Manual page 66

Table 5-3: Protocol Definition, General Parameters (continues on pages 64 to 70)
Parameter
SIP Transport Type
[SIPTransportType]
SIP UDP Local Port
[LocalSIPPort]
SIP TCP Local Port
[TCPLocalSIPPort]
SIP TLS Local Port
[TLSLocalSIPPort]
Enable SIPS
[EnableSIPS]
Enable TCP Connection
Reuse
[EnableTCPConnectionRe
use]
SIP Destination Port
[SIPDestinationPort]
Use "user=phone" in SIP
URL
[IsUserPhone]
Use "user=phone" in From
Header
[IsUserPhoneInFrom]
Use Tel URI for Asserted
Identity
[UseTelURIForAssertedID]
Tel to IP No Answer
Timeout
[IPAlertTimeout]
Enable Remote Party ID
[EnableRPIheader]
Add Number Plan and Type
to Remote Party ID Header
[AddTON2RPI]
SIP User's Manual
Description
Determines the default transport layer used for outgoing SIP calls initiated by the
gateway.
UDP [0] (default).
TCP [1].
TLS [2] (SIPS).
Note: It is recommended to use TLS to communicate with a SIP Proxy and not for
direct gateway-gateway communication.
Local UDP port used to receive SIP messages.
The valid range is 1 to 65534. The default value is 5060.
Local TCP port used to receive SIP messages.
The default value is 5060.
Local TLS port used to receive SIP messages.
The default value is 5061.
Note: The value of 'TLSLocalSIPPort' must be different to the value of
'TCPLocalSIPPort'.
Enables secured SIP (SIPS) connections over multiple hops.
Disable [0] (default).
Enable [1].
When SIPTransportType = 2 (TLS) and EnableSIPS is disabled, TLS is used for
the next network hop only.
When SIPTransportType = 2 (TLS) or 1 (TCP) and EnableSIPS is enabled, TLS is
used through the entire connection (over multiple hops).
Note: If SIPS is enabled and SIPTransportType = UDP, the connection fails.
Enables the reuse of the same TCP connection for all calls to the same
destination.
Valid options include:
0 = Use a separate TCP connection for each call (default)
1 = Use the same TCP connection for all calls
SIP destination port for sending initial SIP requests.
The valid range is 1 to 65534. The default port is 5060.
Note: SIP responses are sent to the port specified in the Via header.
No [0] = 'user=phone' string isn't used in SIP URI.
Yes [1] = 'user=phone' string is part of the SIP URI (default).
No [0] = Doesn't use ';user=phone' string in From header (default).
Yes [1] = ';user=phone' string is part of the From header.
Determines the format of the URI in the P-Asserted and P-Preferred headers.
0 = 'sip:' (default).
1 = 'tel:'.
Defines the time (in seconds) the gateway waits for a 200 OK response from the
called party (IP side) after sending an INVITE message. If the timer expires, the
call is released.
The valid range is 0 to 3600. The default value is 180.
Enable Remote-Party-ID (RPI) headers for calling and called numbers for Tel IP
calls.
Disable [0] (default).
Enable [1] = RPI headers are generated in SIP INVITE messages for both
called and calling numbers.
No [0] = TON/PLAN parameters aren't included in the RPID header.
Yes [1] = TON/PLAN parameters are included in the RPID header (default).
If RPID header is enabled (EnableRPIHeader = 1) and 'AddTON2RPI=1', it is
possible to configure the calling and called number type and number plan using
the Number Manipulation tables for Tel IP calls.
66
Mediant 1000
Document #: LTRT-83301