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NAT Traversal Panasonic Communications Co.,Ltd Office Network Company Network SE Team 2008 Feb 25 th.

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Presentation on theme: "NAT Traversal Panasonic Communications Co.,Ltd Office Network Company Network SE Team 2008 Feb 25 th."— Presentation transcript:

1 NAT Traversal Panasonic Communications Co.,Ltd Office Network Company Network SE Team 2008 Feb 25 th

2 Account info VAD : Voice Activity Detection If it work correctly then voice packet traffic become reduced so it is good. But it rarely cause a trouble. Bad voice quality or call cut off. STUN / SBC. Please refer the “Programming Manual for SIP”. Management site: It is used for checking the charge or choosing the phone services. CLIP: Calling Line Identification Presentation CLIR: Calling Line Identification Restriction Star code is a special code which provider support. In case of Bellshare (German provider) *67 is used for CLIR. When you call *67+area code + phone number then it become a CLIR call. There are other way to stop Caller ID sending if provider support. Logging into the provider web site (using 05) and CLIP off. (Some provider permit user to change some setting on their management site.) [ITSP Essential information] 1Provider name 2Provider type ISP or Carrier 3Type of the networkClosed / Open 4Provider URIhttp:// 5Management sitehttp:// > Login ID > Login Password 6Address (Country) 7Area code 8The method of an international call [Service interface functionalities] 10CLIP* Support 11CLIR* Support Star Code for CLIP/CLIR WEB setting 12FAX relay (T.38) Support 13DTMF (RFC2833) Support 14Hold Support 15Transfer Support 16Conference Support 17DDI Support 18VAD* Support 19STUN Support > If yes then Server name 20SBC (Session Border Controller) Support [SIP Telephone Settings Essential information] 30Proxy Server 31Proxy Server port 32Registrar Server 33Domain Name 40-1SIP Account 41-1Authentication ID 42-1Authentication Password 43-1Incoming Number 44-1DDI Number 40-2SIP Account 41-2Authentication ID 42-2Authentication Password 43-2Incoming Number 44-2DDI Number 50Codec 51Packet Interval 52VAD 53DTMF relay

3 *NAT Traversal type SIP Service Provider SIP ServerSTUN Server SIP Service Provider SIP Server SIP Service Provider SBC SIP Server *NAT: Network Address Translation *SBC: Session Boarder Controller STUN MethodFixed Global IP address Method SBC* Method TDE NAT Router Local Area Network

4 Media Server Proxy Server Nat Off (SBC method) SIP Phone Wireshark L2 Switch TDE Router Router Media Server TDE Proxy Server TDE SBC * SBC: Session Border Controller DSL modem

5 Nat Off (SBC method): outgoing call TDE Router TDE NAPT LAN side WAN side ① ④ ③ ② ⑤ Router Router Proxy Server Media Server NAPT

6 Nat Off (SBC method): incoming call TDE TDE Router Proxy Server LAN side WAN side NAPT ① ④ ③ ② Media Server NAPT Router Router

7 STUN SNOM phone Proxy Server TDE TDE STUN Server Proxy Server SIP Phone Wireshark L2 Switch TDE Router Router DSL modem

8 TDE TDE SNOM phone Router Proxy Server LAN side WAN side STUN: outgoing call ① ④ ③ ② Router Router

9 TDE Proxy Server TDE Router Proxy Server LAN side WAN side STUN: incoming call SNOM phone Router ① ④ ③ ②

10 SBC / STUN TDERouter TDE Server RouterServer SBC STUN REGISTER : : REGISTER : : :32844 IP SIP IP SIP

11 SBC / STUN INVITE SIP/2.0 Via Via: SIP/2.0/UDP :35060;branch=z9hG4bK00007d74;rport Max-Forwards: 70 To: From: Call-ID: CSeq: 1 INVITE Contact Contact: Supported: timer,100rel Session-Expires: 180 Allow: INVITE,ACK,CANCEL,BYE,PRACK,REGISTER,UPDATE Content-Type: application/sdp User-Agent: V1.000i Content-Length: 270 v=0 o=- 1 1 IN IP s=- c=IN IP t=0 0 m=audio RTP/AVP a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv a=ptime:20 a=rtcp:16001 ■IP-Address: Global IP address ( ) ■RTP port: ~ ■* ポート番号は変更しない INVITE SIP/2.0 Via Via: SIP/2.0/UDP :35060;branch=z9hG4bK00007d74;rport Max-Forwards: 70 To: From: Call-ID: CSeq: 1 INVITE Contact Contact: Supported: timer,100rel Session-Expires: 180 Allow: INVITE,ACK,CANCEL,BYE,PRACK,REGISTER,UPDATE Content-Type: application/sdp User-Agent: V1.000i Content-Length: 270 v=0 o=- 1 1 IN IP s=- c=IN IP t=0 0 m=audio RTP/AVP a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv a=ptime:20 a=rtcp:12001 ■IP-Address: Local IP address ■RTP Port: ~ ■ Signaling IP address is different from SDP IP address < STUN > < SBC >

12 a-line before m-line

13 Audio type change

14 Session (IP address) change TDE ITSP ITSP

15 MAC address error: one way audio

16


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