KX-TDA200/100 System (Ver.1.0)
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Transcript KX-TDA200/100 System (Ver.1.0)
KX-TDE100/200 System
(Version 1.0)
Chapter 10
VSIPGW
Panasonic Communications Co., Ltd.
Office Network Company
Edition 1.0
18 JUN., 2007
Confidential
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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)
(A) STUN Method
(B) Fixed Global IP address Method
(c) SBC Method
SIP Service Provider
SIP Service Provider
SIP Service Provider
SIP Server
STUN Server
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
IP Address
SIP Server
NAT Router
IPCMPR
IP Address
=50.60.70.80
SIP Port No.
=5060
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Global IP address
=77.77.77.77
=192.168.0.101
SIP Port No.
=35060
VoIP-DSP
IP Address
=192.168.0.102
RTP Port No.
=1600016063
TDE
DEST.
SRC.
DEST.
SRC.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.101
5060
35060/XXXXX
5060
35060
SRC.
DEST.
SRC.
DEST.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.101
5060
35060/XXXXX
5060
35060
SRC.
DEST.
SRC.
DEST.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.101
5060
35060
5060
35060
NAT Router setting:
SRC port No.
SRC Port No.
NAT Router setting:
DEST port No.
IPCMPR IP address
DEST Port No.
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1.NAT Environment (3)
In the case of (A) STUN Method and (B) Fixed Global IP address Method
SIP Server
IP Address
=50.60.70.80
SIP RTP Port No.
=YYYYY
NAT Router
IP Address
=192.168.0.101
SIP Port No.
=35060
Global IP address
=77.77.77.77
IPCMPR
DEST
SRC
DEST
SRC
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.102
YYYYY
1600016063
YYYYY
1600016063
VoIP-DSP
IP Address
=192.168.0.102
RTP Port No.
=1600016063
TDE
NAT Router setting:
SRC port No.
SRC Port No.
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SRC.
DEST.
SRC.
DEST.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.102
YYYYY
1600016063
YYYYY
1600016063
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
IP Address
=50.60.70.80
SIP Port No.
=5060
NAT Router
IP Address
=192.168.0.101
SIP Port No.
=35060
Global IP address
=77.77.77.77
IPCMPR
DEST.
SRC.
DEST.
SRC.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.101
5060
XXXXX
5060
35060
Keep Alive packet
IP Address
=192.168.0.102
RTP Port No.
=1200012511
VoIP-DSP
(Blank UDP or Register)
Keep Alive packet
Keep Alive packet
(Blank UDP or Register)
Sending Interval
Interval time depends on
NAT Router setting.
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SRC.
DEST.
SRC.
DEST.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.101
5060
XXXXX
5060
35060
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1.NAT Environment (5)
In the case (C)SBC method.
SIP Server
IP Address
=50.60.70.80
SIP RTP Port No.
=YYYYY
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NAT Router
IP Address
=192.168.0.101
SIP Port No.
=35060
Global IP address
=77.77.77.77
IPCMPR
DEST.
SRC.
DEST.
SRC.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.102
YYYYY
XXXXX
YYYYY
SRC.
DEST.
SRC.
DEST.
50.60.70.80
77.77.77.77
50.60.70.80
192.168.0.102
YYYYY
XXXXX
YYYYY
1200012511
IP Address
=192.168.0.102
RTP Port No.
=1200012511
VoIP-DSP
1200012511
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2. DNS Settings
DHCP
Client
DNS Server IP Address Method
In SVIPGW Card Property
Enable
DHCP
Operate by DNS server which DHCP server assign the IP address
Manual
DNS server IP address is assigned by manually
DHCP
PCMC informs warning
Manual
DNS server IP address is assigned by manually
Disable
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Operation
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3. SIP Trunk Setting
VSLOT
No.
1
Port
No.
Trunk
Group
1
1
2
1
CO Line
Name
VSIPGW16-1
----
TDE
Side
CO-001
:
Port 1 -ch.1
:
:
CO-0016
2
16
2
1
2
2
3
:
16
3
Port 16 -ch.16
Ch
VSIPGW16-2
CO-017
:
CO-032
Port 17 -ch.17
:
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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4. SIP Provider settings(1)
Select Provider
You can initialize some settings to the pre-assigned data.
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
Main
Provider Data
Provider Name
PCMC TAB
Option
SIP Server Name
SIP Server Port Number
Register
Session Expire Timer
Session Refresh
Calling Party
Header Type
SIP Service Domain
User Part of From Header
Register Ability
User Part of P-Preferred-Identity Header
Register Sending Interval
Called Party
Register Server Name
NAT
Provider Data
Called Party Number Format
Called Party Type
Register Port Number
Voice/FAX
DTMF
STUN Server Name
Supplement
ary 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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5. SIP Trunk Channel Attribute
VSIPGW16-1
CO-001
TDE
Side
Port 1 -ch.1
:
:
CO-0016
Registration to
Port 16 -ch.16
VSIPGW16-2
CO-017
SIP Server
Basic Channel
Additional Channel
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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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
SIP Server
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Outgoing Destination =
Registered URI +DDI
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 > P-Preferred-identity > From
Ex. From Header)
From:496123899850 <sip:[email protected];user=phon>;tag=809498643
This fields are used for Caller ID indication
Ex. Privacy Header)
P-Asserted-Identity: "Cullen Jennings" <sip:[email protected] >
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 CLIP
6.2 Caller ID Modification
Step 2
In case of International
Step 2
In case of Unknown
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6.2 Leading Digits
Step3
Modification by Leading digits
00
0
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
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7. SIP Trunk Outgoing Feature(1)
Calling numbers are modified as follows.
Provider A
[aaa.com]
Provider B
[bbb.org]
Provider C
[ccc.net]
SIP
Server
SIP
Server
SIP
Server
0924771660
810924771660
81924771660
6.7 Dialing Plan
When Trunk
Dialing type
is En-Bloc.
Add
- Removed
Number of
digits
-Added Number
For SIP trunk call
CO Dial
4771660
Virtual SIP Trunk Card
ISDN/IP-GW (En-bloc)
LCO Card
ISDN/IPGW
Card (Overlap)
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4771660
ISDN/
PSTN
Every Provider may support
different Numbering format,
So we added above settings
for SIP trunk call.
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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-Preferred=ID Header”.
Virtual SIP GW – Port Property – Calling Party
[SIP Edit Example]
123bbb
123aaa
5bbb
5aaa
BRI
PBX Main Unit
123aaa
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1bb
1aa
Virtual SIP
Trunk
SIP Edit
3 digits remove
Add “5”
SIP Edit
4 digits remove
Add “1”
Remove Digit
Additional Dial
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
Anything is OK.
Initialized by Select Provider
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If you input IP address then
Name is not solved by DNS .
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1.Slot - VSIPGW – Port Property - Account
123456780
123456780
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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)
Initialized by Select Provider
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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 .
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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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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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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.
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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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.
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1.Slot - VSIPGW – Port Property – Called Party
When you make a Outgoing call, you can selects
to add
(1) “+”
(2) or not
in Called party dials format.
Initialized by Select Provider
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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”.
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1.Slot - VSIPGW – Port Property – Voice/FAX
Initialized by Select Provider
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1.Slot - VSIPGW – Port Property – RTP/RTCP
Initialized by Select Provider
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1.Slot - VSIPGW – Port Property - DSP
Initialized by Select Provider
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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).
Initialized by Select Provider
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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 10
VSIPGW
Thank you very much !
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