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

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Panasonic Communications Co., Ltd.Panasonic Communications Co., Ltd.

Office Network CompanyOffice Network Company

Edition 1.0 18 JUN., 2007Edition 1.0 18 JUN., 2007

Chapter 10VSIPGW

KX-TDE100/200 SystemKX-TDE100/200 System(Version 1.0)(Version 1.0)

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Chapter 10

SVIPGW

1. NAT Environment1. NAT Environment

2. DNS Settings2. DNS Settings

3. SIP Trunk Settings3. SIP Trunk Settings

4. SIP Provider Settings 4. SIP Provider Settings

5. SIP Trunk Channel Attribute5. SIP Trunk Channel Attribute

6. SIP Trunk Incoming Feature6. SIP Trunk Incoming Feature

7. SIP Trunk Outgoing Feature7. SIP Trunk Outgoing Feature

8. Others8. Others

9. PCMC Programming9. PCMC Programming

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1.NAT Environment (1)

SIP Service Provider

SIP Server STUN Server


SIP Server


SBC

SIP Server

SBC: Session Boarder Controller

(A) STUN Method (B) Fixed Global IP address Method (c) SBC Method

TDE

NAT Router

Local Area Network

Example sipgate.co.uk sipnet.ru

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1.NAT Environment (2)In the case of (A) STUN Method and (B) Fixed Global IP address Method

NAT Router

TDE

IPCMPR VoIP-DSPSIP ServerIP Address =50.60.70.80SIP Port No. =5060

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

SRC. DEST.

50.60.70.80 77.77.77.77

5060 35060/XXXXX

IP Address =192.168.0.101SIP Port No. =35060

Global IP address =77.77.77.77

SRC. DEST.

50.60.70.80 192.168.0.101

5060 35060

IP Address =192.168.0.102RTP Port No. =1600016063

SRC. DEST.

50.60.70.80 77.77.77.77

5060 35060

SRC. DEST.

50.60.70.80 192.168.0.101

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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In the case of (A) STUN Method and (B) Fixed Global IP address Method

NAT Router

TDE

IPCMPR VoIP-DSPSIP Server

IP Address =50.60.70.80SIP RTP Port No. =YYYYY

DEST SRC

50.60.70.80 77.77.77.77

YYYYY 1600016063

DEST SRC

50.60.70.80 192.168.0.102

YYYYY 1600016063




NAT Router setting:

SRC port No.

SRC Port No.


SRC. DEST.

50.60.70.80 77.77.77.77

YYYYY 1600016063

SRC. DEST.

50.60.70.80 192.168.0.102

YYYYY 1600016063

NAT Router setting:

SRC port No.

VoIP DSP IP address

SRC Port No.

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NAT RouterIPCMPR VoIP-DSP

SIP Server

Keep Alive packet (Blank UDP or Register)

Keep Alive packet

Sending Interval

Interval time depends on NAT Router setting.

In the case (C)SBC method. To Keep port NAT table in NAT Router IPCMPR sends Keep Alive packets

DEST. SRC.

50.60.70.80 77.77.77.77

5060 XXXXX

DEST. SRC.

50.60.70.80 192.168.0.101

5060 35060

SRC. DEST.

50.60.70.80 77.77.77.77

5060 XXXXX

SRC. DEST.

50.60.70.80 192.168.0.101

5060 35060





Keep Alive packet (Blank UDP or Register)

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NAT RouterIPCMPR

SIP ServerIn the case (C)SBC method.

DEST. SRC.

50.60.70.80 77.77.77.77

YYYYY XXXXX

DEST. SRC.

50.60.70.80 192.168.0.102

YYYYY 1200012511

SRC. DEST.

50.60.70.80 77.77.77.77

YYYYY XXXXX

SRC. DEST.

50.60.70.80 192.168.0.102

YYYYY 1200012511

IP Address =50.60.70.80SIP RTP Port No. =YYYYY




VoIP-DSP

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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

Manual DNS server IP address is assigned by manually

Disable DHCP PCMC informs warning

Manual DNS server IP address is assigned by manually

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3. SIP Trunk SettingVSLOT

No.Port No.

Trunk Group

CO Line Name

- - - -

1 1 1

2 1

:

16 2

2 1 2

2 3

:

16 3

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)

CO-001 Port 1 -ch.1TDE Side

VSIPGW16-1

: :

CO-0016 Port 16 -ch.16

CO-017 Port 17 -ch.17

CO-032 Port 32 -ch.32

::

VSIPGW16-2

Ch

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4. SIP Provider settings(1)

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.

Select ProviderSelect Provider

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4. SIP Provider settings(2)

PCMC TAB Provider Data 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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5. SIP Trunk Channel Attribute

CO-001 Port 1 -ch.1

TDE Side

VSIPGW16-1

: :

CO-0016 Port 16 -ch.16

CO-017 Port 17 -ch.17

VSIPGW16-2

Basic Channel

Additional Channel

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.

Registration to

SIP Server

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6. SIP Trunk Incoming Feature(1)

2. DDI (Direct Dial In)

TDE supports the DDI service like ISDN.

SIP Server TDE

Incoming by DDIIncoming Destination =

Registered URI +DDI

Outgoing Destination = Registered URI +DDI

1. Incoming processTDE 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.

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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.

In case of Unknown

6.2 Caller ID Modification6.2 CLIP

6.2 Leading Digits

In case of InternationalStep 2Step 2

Step3Modification by Leading digits

128

000

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)

Provider A

[aaa.com]

SIP Server

CO Dial

4771660Virtual SIP Trunk CardISDN/IP-GW (En-bloc)

81924771660 0924771660 810924771660

LCO CardISDN/PSTN

4771660

Provider B

[bbb.org]

SIP Server

Provider C

[ccc.net]

ISDN/IPGW Card (Overlap)

SIP Server

Add- Removed Number of digits

-Added Number

For SIP trunk call

Every Provider may support different Numbering format,So we added above settings for SIP trunk call.

Calling numbers are modified as follows.6.7 Dialing Plan

When Trunk Dialing type is En-Bloc.

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PBX Main Unit

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

Trunk

BRI

123aaa

123bbb

５ aaa

５ bbb

1aa

1bb[SIP Edit Example]

4 digits removeAdd “1”

3 digits removeAdd “5”

SIP Edit SIP Edit

Virtual SIP GW – Port Property – Calling Party

Remove Digit

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.729ATDE100/200 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)

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.

2. Hold/Transfer by SIP server

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When ”NAT Traversal” setting is Fixed Global IP Address

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.)

Used for via Header rport (INVITE, 100 Trying, …)

“Enavle Active” is not used.

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You can select the method to get IP address of DNS server. (1) Manual (2) DHCPExplained 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.

1.Slot - VSIPGW – Card Property

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If you input IP address thenName is not solved by DNS .

Initialized by Select Provider

1.Slot - VSIPGW – Port Property - Main

Anything is OK.

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1.Slot - VSIPGW – Port Property - Account

123456780

123456780

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1.Slot - VSIPGW – Port Property - Register

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 .

In case your provider does not require the Registration then change this setting to Disable. (Not Send Register Message)

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If you input IP address then

Name is not solved by DNS .

Initialized by Select Provider

1.Slot - VSIPGW – Port Property - Nat


Session Refresh feature is used for verifying

the Normality of the (speech) communication.

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

VSIPGW can select the Session Refresh Method from following. 1. “re-INVITE Message” 2.”UPDATE Message”.

1.Slot - VSIPGW – Port Property - Option

★


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”.

Explained in “7.SIP Trunk Outgoing Feature (2)

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 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 Outgoingcall, you can add “+” or not if User part is selected as PBX-CLIP. (1) “+International” (2) or not in Calling party dials format.

1.Slot - VSIPGW – Port Property – Calling Party

If you write SIP-URI, then this setting data is use for SIP-URI. (High Priority)

★


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.

1.Slot - VSIPGW – Port Property – Called Party


1.Slot - VSIPGW – Port Property – Voice/FAX


1.Slot - VSIPGW – Port Property – RTP/RTCP


1.Slot - VSIPGW – Port Property - DSP


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 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 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.

1.Slot - VSIPGW – Port Property – Supplementary Service

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Thank you very much !Thank you very much !

Chapter 10VSIPGW

Documents

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