Panasonic Communications Co., Ltd.

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KX-TDE100/200 System(Version 1.0)
Chapter 10VSIPGW

• Panasonic Communications Co., Ltd.
• Office Network Company
• Edition 1.0 18 JUN., 2007

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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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1.NAT Environment (1)
(c) SBC Method
(A) STUN Method
(B) Fixed Global IP address Method
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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1.NAT Environment (2)
In the case of (A) STUN Method and (B) Fixed
Global IP address Method
NAT Router
IPCMPR
VoIP-DSP
SIP Server
IP Address 192.168.0.101 SIP Port No.
35060
IP Address 192.168.0.102 RTP Port No.
16000?16063
IP Address 50.60.70.80 SIP Port No. 5060
Global IP address 77.77.77.77
TDE
DEST. SRC.
50.60.70.80 77.77.77.77
5060 35060/XXXXX
DEST. SRC.
50.60.70.80 192.168.0.101
5060 35060
NAT Router setting SRC port No. ?SRC
Port No.
SRC. DEST.
50.60.70.80 77.77.77.77
5060 35060/XXXXX
SRC. DEST.
50.60.70.80 192.168.0.101
5060 35060
NAT Router setting DEST port No. ?IPCMPR
IP address DEST Port No.
SRC. DEST.
50.60.70.80 77.77.77.77
5060 35060
SRC. DEST.
50.60.70.80 192.168.0.101
5060 35060
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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 192.168.0.102 RTP Port No.
16000?16063
IP Address 192.168.0.101 SIP Port No.
35060
IP Address 50.60.70.80 SIP RTP Port No.
YYYYY
Global IP address 77.77.77.77
TDE
DEST SRC
50.60.70.80 77.77.77.77
YYYYY 16000?16063
DEST SRC
50.60.70.80 192.168.0.102
YYYYY 16000?16063
NAT Router setting SRC port No. ? SRC
Port No.
SRC. DEST.
50.60.70.80 192.168.0.102
YYYYY 16000?16063
SRC. DEST.
50.60.70.80 77.77.77.77
YYYYY 16000?16063
NAT Router setting SRC port No. ?VoIP DSP
IP address SRC Port No.
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1.NAT Environment (4)
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 192.168.0.101 SIP Port No.
35060
IP Address 50.60.70.80 SIP Port No. 5060
Global IP address 77.77.77.77
IP Address 192.168.0.102 RTP Port No.
12000?12511
DEST. SRC.
50.60.70.80 77.77.77.77
5060 XXXXX
DEST. SRC.
50.60.70.80 192.168.0.101
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.
50.60.70.80 77.77.77.77
5060 XXXXX
SRC. DEST.
50.60.70.80 192.168.0.101
5060 35060
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1.NAT Environment (5)
In the case (C)SBC method.
SIP Server
NAT Router
IPCMPR
VoIP-DSP
IP Address 192.168.0.101 SIP Port No.
35060
IP Address 50.60.70.80 SIP RTP Port No.
YYYYY
Global IP address 77.77.77.77
IP Address 192.168.0.102 RTP Port No.
12000?12511
DEST. SRC.
50.60.70.80 77.77.77.77
YYYYY XXXXX
DEST. SRC.
50.60.70.80 192.168.0.102
YYYYY 12000?12511
SRC. DEST.
50.60.70.80 77.77.77.77
YYYYY XXXXX
SRC. DEST.
50.60.70.80 192.168.0.102
YYYYY 12000?12511
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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
Enable Manual DNS server IP address is assigned by manually
Disable DHCP PCMC informs warning
Disable Manual DNS server IP address is assigned by manually
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3. SIP Trunk Setting
VSLOT No. Port No. Trunk Group CO Line Name - - - -
1 1 1
1 2 1
1
1 16 2
2 1 2
2 2 3
2
2 16 3
VSIPGW16-1
TDE Side


Ch
VSIPGW16-2


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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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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4. SIP Provider settings(2)
PCMC TAB Provider Data PCMC TAB Provider Data
Main Provider Name Option Session Expire Timer
Main SIP Server Name Option Session Refresh
Main SIP Server Port Number Calling Party Header Type
Main SIP Service Domain Calling Party User Part of From Header
Register Register Ability Calling Party User Part of P-Preferred-Identity Header
Register Register Sending Interval Called Party Called Party Number Format
Register Register Server Name Called Party Called Party Type
Register Register Port Number Voice/FAX DTMF
NAT STUN Server Name Supplementary Service CNIP (Send)
NAT STUN Server Port Number Supplementary Service CNIP (Receive)
You can make SIP Provider pre-assigned data by
Excel and import to TDE.
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5. SIP Trunk Channel Attribute
VSIPGW16-1
TDE Side


Registration to SIP Server
Basic Channel
Additional Channel
VSIPGW16-2
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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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
TDE
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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 gt
P-Preferred-identity gt From Ex. From Header)
From496123899850 ltsip496123899850_at_sipg
w01.bmcag.comuserphongttag809498643
This
fields are used for Caller ID indication Ex.
Privacy Header) P-Asserted-Identity
"Cullen Jennings" ltsip14085264000_at_cisco.com gt
Caller ID will be modified as explained in the
Next Page.
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6. SIP Trunk Incoming Feature(3)
Caller ID modification for call back Step 1
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 Caller ID Modification
6.2 CLIP
Step 2
Step 2
In case of International
In case of Unknown
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00
8
0
6.2 Leading Digits
In case of Unknown, received digits are
modified in Step 2 as 1) 3848507 (7
digits) ? No modification 2) 38485078
(8 digits) ? Add 0 038485078 3)
38485078901(11 digits) ? add 0 038485078901
4)3848507890123(13 digits) ? add 00
003848507890123
Step3
Modification by Leading digits
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7. SIP Trunk Outgoing Feature(1)
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.
810924771660
81924771660
0924771660
Virtual SIP Trunk Card ISDN/IP-GW (En-bloc)
CO Dial 4771660
LCO Card
Every Provider may support different Numbering
format, So we added above settings for SIP trunk
call.
ISDN/ PSTN
ISDN/IPGW Card (Overlap)
4771660
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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-PreferredID Header.
Virtual SIP GW Port Property Calling Party
SIP Edit Example
1bb
5bbb
123bbb
1aa
5aaa
123aaa
Virtual SIP Trunk
BRI
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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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)
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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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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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.
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1.Slot - VSIPGW Port Property - Main
If you input IP address then Name is not solved
by DNS .
Anything is OK.
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(No Transcript)
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1.Slot - VSIPGW Port Property - Register
If SIP server and Registrar server are same then
you dont need to input Registrar server setting.
If you input IP address then Name is not solved
by DNS .
In case your provider does not require the
Registration then change this setting to Disable.
(Not Send Register Message)
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1.Slot - VSIPGW Port Property - Nat
If you input IP address then Name is not solved
by DNS .
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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
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1.Slot - VSIPGW Port Property Calling Party
Explained in 7.SIP Trunk Outgoing Feature (2)
If you write SIP-URI, then this setting data is
use for SIP-URI. (High Priority)
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.
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.
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1.Slot - VSIPGW Port Property Called Party
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.
When you make a Outgoing call, you can selects
to add (1) (2)
or not in Called party dials format.
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1.Slot - VSIPGW Port Property Voice/FAX
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1.Slot - VSIPGW Port Property RTP/RTCP
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1.Slot - VSIPGW Port Property - DSP
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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.
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Chapter 10VSIPGW
Thank you very much !