MediaPack SIP 
MediaPack SIP User’s Manual   54  Document #: LTRT-65405 
Table  5-1: Protocol Definition, General Parameters (continues on pages 52 to 55) 
Parameter  Description 
Enable SIPS 
[EnableSIPS] 
Enables secured SIP (SIPS) connections over multiple hops (MP-11x only). 
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. 
SIP Destination Port 
[SIPDestinationPort] 
SIP UDP destination port for sending SIP messages. 
The default value is 5060. 
Use “user=phone” in SIP URL
[IsUserPhone] 
No  [0] = ‘user=phone’ string isn’t used in SIP URL. 
Yes [1] = ‘user=phone’ string is part of the SIP URL (default). 
Use “user=phone” in From 
header 
[IsUserPhoneInFrom] 
No  [0] = Doesn’t use ‘;user=phone’ string in From header (default). 
Yes [1] = ‘;user=phone’ string is part of the From header. 
Tel to IP No Answer Timeout
[IPAlertTimeout] 
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 
[EnableRPIheader] 
Enable Remote-Party-ID (RPI) headers for calling and called numbers for TelÆIP calls.
Disable  [0] (default). 
Enable  [1] = RPI headers are generated in SIP INVITE messages for both called and 
calling numbers. 
Add Number Plan and Type to 
Remote Party ID Header 
[AddTON2RPI] 
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 TelÆIP calls. 
Use Source Number as 
Display Name 
[UseSourceNumberAsDispl
ayName] 
No  [0] = Interworks the Tel calling name to SIP Display Name (default). 
Yes [1] = Set Display Name to Calling Number if not configured. 
 
Applicable to TelÆIP calls. If enabled and calling party name is not defined 
(CallerDisplayInfoX = <name> is not specified per gateway’s x port), the calling number 
is used instead. 
Use Display Name as Source 
Number 
[UseDisplayNameAsSource
Number] 
No  [0] = Interworks the IP Source Number to the Tel Source Number (default). 
Yes [1] = Sets the Tel Source Number to IP Display Name. 
Applicable to IPÆTel calls. 
If enabled, the outgoing Source Number is set to the IP Display Name and Presentation 
is set to Allowed. If there isn’t a Display Name, the user part of the SIP URI is used as 
the Source Number, and the Presentation is set to Restricted. 
For example: 
When the following is received ’from: 100 <sip:200@201.202.203.204>’, the outgoing 
Source Number is set to ’100’, the Display Name is set to ’100’ and the Presentation is 
set to Allowed (0). 
When the following is received ‘from: <sip:100@101.102.103.104>’, the outgoing 
Source Number is set to ‘100’ and the Presentation is set to Restricted (1). 
Play Ringback Tone to IP 
[PlayRBTone2IP] 
Don’t Play  [0] = Ringback tone isn’t played to the IP side of the call (default). 
Play     [1] = Ringback tone is played to the IP side of the call after SIP 183 
session progress response is sent. 
Note 1: To enable the gateway to send a 183 response, set ‘EnableEarlyMedia’ to 1. 
Note 2: If ‘EnableDigitDelivery = 1’, the gateway doesn’t play a Ringback tone to IP and 
doesn’t send a 183 response. 
Play Ringback Tone to Tel 
[PlayRBTone2Tel] 
Don’t Play  [0] = Ringback Tone isn’t played. 
Always Play  [1] = Ringback Tone is played to the Tel side of the call when 180/183 
response is received. 
Play According to PI [3] = N/A. 
Play According to 180/183 [2] = Ringback Tone is played to the Tel side of the call if no 
SDP is received in 180/183 responses. If 180/183 with SDP message is received, the 
gateway cuts through the voice channel and doesn’t play Ringback tone (default).