1. KX-TDE100/200 System (Version 1.0)
Chapter 10 VSIPGW
Panasonic Communications Co., Ltd.
Office Network Company
Edition JUN., 2007
Confidential 1

2. Chapter 10 SVIPGW 1. NAT Environment 2. DNS Settings
3. SIP Trunk Settings
4. SIP Provider Settings
5. SIP Trunk Channel Attribute
6. SIP Trunk Incoming Feature
7. SIP Trunk Outgoing Feature
8. Others
9. PCMC Programming
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3. (B) Fixed Global IP address Method SBC: Session Boarder Controller
1.NAT Environment (1)
(c) SBC Method
(A) STUN Method
(B) Fixed Global IP address Method
SIP Service Provider
SIP Server
STUN Server
SIP Service Provider
SIP Service Provider
SIP Server
SIP Server
SBC
Example
sipgate.co.uk
sipnet.ru
NAT Router
Local Area Network
TDE
SBC: Session Boarder Controller
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4. 1.NAT Environment (2)
In the case of (A) STUN Method and (B) Fixed Global IP address Method
SIP Server
NAT Router
IPCMPR
VoIP-DSP
IP Address =
SIP Port No.
=35060
IP Address =
RTP Port No.
=1600016063
IP Address =
SIP Port No.
=5060
Global IP address =
TDE
DEST.
SRC.
5060
35060/XXXXX
DEST.
SRC.
5060
35060
NAT Router setting:
SRC port No.
SRC Port No.
SRC.
DEST.
5060
35060/XXXXX
SRC.
DEST.
5060
35060
NAT Router setting:
DEST port No.
IPCMPR IP address
DEST Port No.
SRC.
DEST.
5060
35060
SRC.
DEST.
5060
35060
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5. 1.NAT Environment (3)
In the case of (A) STUN Method and (B) Fixed Global IP address Method
SIP Server
NAT Router
VoIP-DSP
IPCMPR
IP Address =
SIP Port No.
=35060
IP Address =
RTP Port No.
=1600016063
IP Address =
SIP RTP Port No.
=YYYYY
Global IP address =
TDE
DEST
SRC
YYYYY
1600016063
DEST
SRC
YYYYY
1600016063
NAT Router setting:
SRC port No.
 SRC Port No.
SRC.
DEST.
YYYYY
1600016063
SRC.
DEST.
YYYYY
1600016063
NAT Router setting:
SRC port No.
VoIP DSP IP address
SRC Port No.
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6. 1.NAT Environment (4) SIP Server NAT Router IPCMPR VoIP-DSP
In the case (C)SBC method. To Keep port NAT table in NAT Router IPCMPR sends Keep Alive packets
SIP Server
NAT Router
IPCMPR
VoIP-DSP
IP Address =
SIP Port No.
=35060
IP Address =
SIP Port No.
=5060
Global IP address =
IP Address =
RTP Port No.
=1200012511
DEST.
SRC.
5060
XXXXX
DEST.
SRC.
5060
35060
Keep Alive packet (Blank UDP or Register)
Keep Alive packet
Sending Interval
Keep Alive packet (Blank UDP or Register)
Interval time depends on
NAT Router setting.
SRC.
DEST.
5060
XXXXX
SRC.
DEST.
5060
35060
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7. 1.NAT Environment (5) SIP Server NAT Router IPCMPR VoIP-DSP
In the case (C)SBC method.
SIP Server
NAT Router
IPCMPR
VoIP-DSP
IP Address =
SIP Port No.
=35060
IP Address =
SIP RTP Port No.
=YYYYY
Global IP address =
IP Address =
RTP Port No.
=1200012511
DEST.
SRC.
YYYYY
XXXXX
DEST.
SRC.
YYYYY
1200012511
SRC.
DEST.
YYYYY
XXXXX
SRC.
DEST.
YYYYY
1200012511
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8. 2. DNS Settings DHCP Client DNS Server IP Address Method
In SVIPGW Card Property
Operation
Enable
DHCP
Operate by DNS server which DHCP server assign the IP address
Manual
DNS server IP address is assigned by manually
Disable
PCMC informs warning
Setting combination.
If you assign “DNS Server IP Address Method” manualy then it becomes the best advantage. (effective)
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9. 3. SIP Trunk Setting : : : : VSIPGW16-1 TDE Side VSIPGW16-2 VSLOT No.
Port
Trunk
Group
CO Line Name 1 2 : 16 3
VSIPGW16-1
TDE Side
CO-001
Port 1 -ch.1 : : Ch
CO-0016
Port 16 -ch.16
VSIPGW16-2
CO-017
Port 17 -ch.17 : :
CO-032
Port 32 -ch.32
A SIP trunk has to belong one of Trunk Groups.
There are 3 types of Trunk (Public, Private, VPN). SIP trunk work as Public line as same as Analog CO line.
You can program SIP trunk as same as traditional CO line by PCMC.
(Example.10.1 CO Line Setting)
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10. 4. SIP Provider settings(1)
You can initialize some settings to the pre-assigned data.
Select Provider
In the next page, there is a list of SIP Provider pre-assigned data. You can select SIP provider port by port.
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11. 4. SIP Provider settings(2)
PCMC TAB
Provider Data
Main
Provider Name
Option
Session Expire Timer
SIP Server Name
Session Refresh
SIP Server Port Number
Calling Party
Header Type
SIP Service Domain
User Part of From Header
Register
Register Ability
User Part of P-Preferred-Identity Header
Register Sending Interval
Called Party
Called Party Number Format
Register Server Name
Called Party Type
Register Port Number
Voice/FAX
DTMF
NAT
STUN Server Name
Supplementary Service
CNIP (Send)
STUN Server Port Number
CNIP (Receive)
You can make SIP Provider pre-assigned data by “Excel” and import to TDE.
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12. 5. SIP Trunk Channel Attribute
VSIPGW16-1
CO-001
Port 1 -ch.1
TDE Side : :
Registration to
SIP Server
CO-0016
Port 16 -ch.16
Basic Channel
Additional Channel
VSIPGW16-2
CO-017
Port 17 -ch.17
Channel Attribute
meaning
Basic Channel
Contract account channel.
Register operation is activated by this channel data.
Additional channel of Ch xx
Bundle channel contract with Basic channel xx.
Registration operation is not activated by this channel data.
Not Used
No contract channel
Registration operation is not activated.
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13. 6. SIP Trunk Incoming Feature(1)
1. Incoming process
TDE decides “Called Party Number” by “To Header” or “Request-URI Header”.
TDE analyzes SIP-URI in “To Header” or “Request-URI Header” and decides the ringing destination
by followings.
1) Basic channel of “User Name = User part of SIP-URI”.
2) Basic channel of “User Name = User part of SIP-URI + DDI”
3) Any Basic channel of “User Name does not hit with User part of SIP-URI” and,
if “SIP Called Party Number Check Ability” is Enable, Incoming call is rejected by 404 message.
if “SIP Called Party Number Check Ability” is Disable (High  Low), TDE selects the incoming
destination from idle biggest number channel.
if “SIP Called Party Number Check Ability” is Disable (Low  High), TDE selects the incoming
destination from idle smallest number channel.
2. DDI (Direct Dial In)
TDE supports the DDI service like ISDN.
Incoming by DDI
Incoming Destination = Registered URI +DDI
Outgoing Destination = Registered URI +DDI
SIP Server
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TDE

14. 6. SIP Trunk Incoming Feature(2)
3. Caller ID
TDE100/200 VSIPGW supports CLIP/CNIP feature in ISDN . Caller information is stored in “From Header” or
Privacy Header. Display priority is as follows.
P-Asserted-Identity > P-Preferred-identity > From
Ex. From Header)　
From:
This fields are used for Caller ID indication
Ex. Privacy Header)　
　　　 P-Asserted-Identity: "Cullen Jennings"
Caller ID will be modified as explained in the Next Page.
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15. 6. SIP Trunk Incoming Feature(3) In case of International
Caller ID modification for call back
Step If there is “+” in received digits, “+” is removed and received digits are treated as “International” dials.
If not, received digits are treated as “Unknown” dials.
6.2 CLIP
6.2 Caller ID Modification
Step 2
Step 2
In case of International
In case of Unknown 12 00 8
6.2 Leading Digits
In case of “Unknown”, received digits are modified in Step 2 as
1) (7 digits)  No modification
2) (8 digits)  Add “0”
3) (11 digits)  add ”0”
4) (13 digits)  add ”00”
Step3
Modification by Leading digits
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16. 7. SIP Trunk Outgoing Feature(1) When Trunk Dialing type is En-Bloc.
Calling numbers are modified as follows.
6.7 Dialing Plan
Provider C
[ccc.net]
Provider A
[aaa.com]
Provider B
[bbb.org]
SIP Server
SIP Server
SIP Server
Add
- Removed
Number of
digits
-Added Number
For SIP trunk call
When Trunk Dialing type is En-Bloc.
Virtual SIP Trunk Card
ISDN/IP-GW (En-bloc)
CO Dial
LCO Card
ISDN/
PSTN
Every Provider may support
different Numbering format,
So we added above settings
for SIP trunk call.
ISDN/IPGW
Card (Overlap)
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17. 7. SIP Trunk Outgoing Feature(2)
Caller numbers are edited as follows when PBX-CLIP is selected in User Part of “From Header” or “P-Preferred=ID Header”.
Virtual SIP GW – Port Property – Calling Party
[SIP Edit Example]
1bb
５bbb
123bbb
1aa
５aaa
123aaa
BRI
Virtual SIP
Trunk
SIP Edit
SIP Edit
PBX Main Unit
3 digits remove
Add “5”
Remove Digit
4 digits remove
Add “1”
Additional Dial
123aaa
123bbb
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18. 8. Others Information 1. Voice / Fax / DTMF communication Ability
TDE100/200 support the following Voice communication ability.
1) G.711 a-Law
2) G.711 u-Law
3) G.729A
TDE100/200 support the following FAX communication ability.
1) In-band (G.711 communication) only
2) T.38 (not supported with TDE V1.0)
TDE100/200 support the following DTMF communication ability.
1) In-band (G.711 communication)
2) Out-band (RFC2833 method)
DTMF by INFO is not supported
2. Hold/Transfer by SIP server
Hold /Transfer feature which is prepared by SIP server does not work (i.e. REFER /ReINVITE
message will be ignored.)
Hold/Transfer feature which is terminated by TDE100/200 works.
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19. Confidential 1.Slot - VSIPGW – Shelf Property
In the case of STUN or Fixed IP address NAT-Traversal
Method, RTP port No. start No is set here.
In the case of SBC(NAT Traversal: Off), RTP port start No. is same with the other Virtual IP cards. (Voice UDP Port No. becomes ineffective.)
When ”NAT Traversal” setting is Fixed Global IP Address
Used for via Header rport (INVITE, 100 Trying, …)
“Enavle Active” is not used.
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20. Confidential 1.Slot - VSIPGW – Card Property
You can select the method to get IP address
of DNS server.
(1) Manual
(2) DHCP
Explained on “2.DNS Setting”
DNS Server IP address by DHCP server.
DNS Server IP address by Manual entry.
Activate “DNS SRV record resolve ability” or not.
Page 8
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21. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property - Main
Page 12
Anything is OK.
If you input IP address then
Name is not solved by DNS .
Initialized by Select Provider
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22. 1.Slot - VSIPGW – Port Property - Account
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23. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property - Register
In case your provider does not require the Registration then change this setting to Disable. (Not Send Register Message)
If SIP server and Registrar server are same then you don’t need to input Registrar server setting.
If you input IP address then
Name is not solved by DNS .
Initialized by Select Provider
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24. If you input IP address then Name is not solved by DNS .
1.Slot - VSIPGW – Port Property - Nat
Initialized by Select Provider
If you input IP address then
Name is not solved by DNS .
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25. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property - Option
Session Refresh feature is used for verifying
the Normality of the (speech) communication. ★
VSIPGW can select the Session
Refresh Method from following.
1. “re-INVITE Message”
2.”UPDATE Message”.
VSIPGW can select Session Timer ability from following.
1. “Enable (Passive)” : When other party requests session refresh, session refresh will be activated.
2. “Enable (Active)” : When other party supports session refresh, session refresh will be activated.
3. “Not Used” : Does not activate this feature
Initialized by Select Provider
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26. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property – Calling Party
If you write SIP-URI, then this setting data is use for SIP-URI. (High Priority)
Explained in
“7.SIP Trunk Outgoing Feature (2)
When you make a Outgoing
call, you can add “+” or not if
User part is selected as PBX-CLIP.
(1) “+International”
(2) or not
in Calling party dials format. ★
When you make a Outgoing call,
you can select to send CLIP in
User Part of “P-Preferred
ID Header” from followings.
(1) User Name
(2) Authentication ID
(3) PBX-CLIP.
if ext. select CO number,
“subscriber number” is used.
if ext. select ext. setting CO
number, ISDN CLIP is used.
Page 16, 17
When you make a Outgoing call, you can select to send CLIP
in User Part of “From Header” from followings.
(1) User Name
(2) Authentication ID
(3) PBX-CLIP.
if ext. select CO number, “subscriber number” is used.
if ext. select ext. setting CO number, ISDN CLIP is used.
When you make a Outgoing call,
you can select to send CLIP Message by
1. “only From Header”
2. or “From Header and P-Preferred-ID Header”.
Initialized by Select Provider
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27. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property – Called Party
Page 13, 14, 15
When you make a Outgoing call, you can selects
to add (1) “+”
(2) or not
in Called party dials format.
When you receive an Incoming call, you can select
to get called party number from
(1)“User Part in To Header”
(2) or “User Part in Request-URI”.
Initialized by Select Provider
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28. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property – Voice/FAX
Page 18
Initialized by Select Provider
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29. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property – RTP/RTCP
Initialized by Select Provider
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30. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property - DSP
Initialized by Select Provider
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31. Initialized by Select Provider Confidential
1.Slot - VSIPGW – Port Property – Supplementary Service
When you make a Outgoing call, you can select to send Name
from main unit.
(1) if “No”, VSIPGW does not send Name information to
SIP server.
(2) if “Yes”, VSIPGW sends Name information to SIP server.
When you make a Outgoing call,
(1) if “Yes”, you can select Send-CLIP or Non-CLIP
(Anonymous) to destination.
(2) if “No”, You cannot select Non-CLIP to destination
(CLIP is basically sent; if there is no CLIP, Anonymous
will be sent).
When you receive an Incoming call, you can select to send
Name to main unit.
(1) if “No”, VSIPGW does not send Name information
to main unit.
(2) if “Yes”, VSIPGW sends Name information to main unit.
Initialized by Select Provider
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32. Chapter 10 VSIPGW
Thank you very much !
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