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

KX-TDE100/200 System (Version 1.0). Panasonic Communications Co., Ltd. Office Network Company Edition 1.0 18 JUN., 2007. Chapter 7.1 VSIPGW. Chapter 7.1 SVIPGW. 1. NAT Environment 2. DNS Settings 3. SIP Trunk Settings 4. SIP Provider Settings 5. SIP Trunk Channel Attribute

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

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  1. KX-TDE100/200 System(Version 1.0) Panasonic Communications Co., Ltd. Office Network Company Edition 1.0 18 JUN., 2007 Chapter 7.1VSIPGW

  2. Chapter 7.1 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

  3. SIP Service Provider SIP Server STUN Server 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 Equipment Local Area Network TDE SBC: Session Boarder Controller

  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 =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 NAT Router setting: SRC port No. SRC Port No. NAT Router setting: DEST port No. IPCMPR IP address DEST Port No.

  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 =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 RTP Port No. =YYYYY Global IP address =77.77.77.77 TDE NAT Router setting: SRC port No.  SRC Port No. NAT Router setting: SRC port No. VoIP DSP IP address SRC Port No.

  6. 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 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.

  7. 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

  8. 2. DNS Settings

  9. CO-0016 CO-001 CO-032 CO-017 Port 32 -ch.32 Port 1 -ch.1 Port 17 -ch.17 Port 16 -ch.16 3. SIP Trunk Setting 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)

  10. 4. SIP Provider settings(1) You can initialize some settings by SIP Provider pre-assigned data by Pressing “Select Provider”. In the next page, there is a list of SIP Provider pre-assigned data. You can select SIP provider by port by port.

  11. 4. SIP Provider settings(2) PCC will provide SIP Provider pre-assigned data. You can import the data and use this data. You can also edit SIP Provider pre-assigned data by “Excel” and import the data.

  12. CO-001 CO-0016 CO-017 Port 1 -ch.1 Port 16 -ch.16 Port 17 -ch.17 5. SIP Trunk Channel Attribute VSIPGW16-1 TDE Side : : Registration to SIP Server Basic Channel Additional Channel VSIPGW16-2

  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 Namedoes 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

  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:496123899850 <sip:496123899850@sipgw01.bmcag.com;user=phon>;tag=809498643 This fields are used for Caller ID indication Ex. Privacy Header) P-Asserted-Identity: "Cullen Jennings" <sip:+14085264000@cisco.com> Caller ID will be modified as explained in the Next Page.

  15. 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 Step 2 In case of International In case of Unknown 12 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

  16. 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

  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 3 digits remove Add “5” PBX Main Unit Remove Digit Additional Dial 4 digits remove Add “1” Ext. CLIP 123aaa Ext. CLIP 123bbb

  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 Mu-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) In-band (RFC2833 method) 2. Hold/Transfer by SIP server Hold/Transfer feature which is activated by TDE100/200 work. But Hold /Transfer feature which is activated by SIP server does not work (i.e. REFER /ReINVITE message will be ignored.)

  19. 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, RTP port start No. is as same as other Virtual IP cards. When ”NAT Traversal” setting is Fixed Global IP Address

  20. 1.Slot - VSIPGW – Card Property You can select the method to get IP address of DNS server. (1) Manual (2) DHCP Explained in “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.

  21. 1.Slot - VSIPGW – Port Property - Main Initialized by Select Provider Anything is OK. If you program IP address, IP address is used. Name is not solved by DNS .

  22. 1.Slot - VSIPGW – Port Property - Account 123456780 123456780 Initialized by Select Provider

  23. 1.Slot - VSIPGW – Port Property - Register Initialized by Select Provider VSIPGW can select to send Register Message to SIP server or not. (1) Enable : Send Register Message (2) Disable : Not Send Register Message If SIP server and Registrar server are same one, you don’t need to program Registrar server setting. If you program IP address, IP address is used. Name is not solved by DNS .

  24. 1.Slot - VSIPGW – Port Property - Nat Initialized by Select Provider If you program IP address, IP address is used. Name is not solved by DNS .

  25. 1.Slot - VSIPGW – Port Property - Option Initialized by Select Provider 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

  26. 1.Slot - VSIPGW – Port Property – Calling Party Initialized by Select Provider 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 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”.

  27. 1.Slot - VSIPGW – Port Property – Called Party Initialized by Select Provider 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”.

  28. 1.Slot - VSIPGW – Port Property – Voice/FAX Initialized by Select Provider

  29. 1.Slot - VSIPGW – Port Property – RTP/RTCP Initialized by Select Provider

  30. 1.Slot - VSIPGW – Port Property - DSP Initialized by Select Provider

  31. 1.Slot - VSIPGW – Port Property – Supplementary Service Initialized by Select Provider 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.

  32. Chapter 7.1VSIPGW Thank you very much !

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