Softswitch Interworking Protocol

Transcription

1 Softswitch Interworking Protocol Chang Sup Keum Network Research Lab. Softswitch Team E T R I -1-

2 Contents Softswitch SIP-T (Session Initiation Protocol for Telephones) BICC (Bearer Independent Call Control) Conclusion -2-

3 What is a Softswitch? A Softswitch (a.k.a call agent, call server or media gateway controller) is a device that provides, at a minimum: Intelligence that controls connection services for a media gateway, and/or native IP endpoints. The ability to select processes that can be applied to a call Routing for a call within the network based on signaling and customer database information. The ability to transfer control of the call to another network element. Interfaces to and supports management functions such as provisioning, fault, billing, etc. -3-

4 Softswitch Objectives Flexible distribution of switching functionality Interoperability among functional elements Carrier selection of best-in-class components Rapid introduction of innovative new services -4-

5 Global MSF Interoperability (GMI) 2002 Date: November 4 th - 15 th Scope Interoperability trial to prove MSF Release 1 Architecture utilizing MEGACO/H.248, and SIP as the main control and signaling protocols. Shows that competitors, both carriers and vendors, can work together to a ccelerate the development of equipment and services within the MSF s co llaborative framework. Assist carriers to achieve their goal: to deploy flexible, best of breed prod ucts. Assist vendors to achieve their goal: to market products more cost effectively. 15 participants including carriers, system suppliers and test equipment manufacturers -5-

6 The GMI2002 Test Sites Asia BTexact Adastral Park, UK. Europe BT ignite NTT Tokyo, Japan. Abilene Qwest UNH Interop Lab -6-

7 Definition SIP-T Protocol The usage of the SIP protocol for the purpose of voice over packet network signaling interconnection with the PSTN Standardization IETF RFC3261, " SIP: Session Initiation Protocol " (Obsolete: 2543) IETF RFC2976, "The SIP INFO Method IETF RFC3204, "MIME media types for ISUP and QSIG Objects" IETF RFC3372, "SIP for Telephones (SIP-T): Context and Architectures IETF RFC3398, "ISUP to SIP Mapping " IETF RFC3262, "Reliability of Provisional Responses in SIP MSF-IA-SIP-T.001, "Implementation Agreement for SIP-T Profile for Media Gateway Controller " -7-

8 IP-PSTN Interconnection SIP-T 적용분야 MGC MGC SS7 based IN ISUP STP SG SIGTRAN H.248/ Megaco SIP-T SIP-T SIP SIP 서버 SG ISUP STP SS7 based IN PSTN LEX IP Network LEX PSTN TG SIP RTP TG -8-

9 IDC IDC SIP-T Characteristics Originator MGC1 SS7 SS7 ISUP message LEX1 PSTN origination IP(SIP) origination SIP message SS7 ISUP message encapsulation SS7 ISUP message header translation Terminator MGC2 SS7 SS7 ISUP message PSTN LEX2 PSTN termination IP(SIP) termination -9-

10 Encapsulation ISUP messages are composed of arbitrary binary data that is transparent to SIP processing. The best way to encode these is to use binary encoding. This is in conformance with the restrictions imposed on the use of binary data for MIME (RFC 2045). This media type is defined by the following information: Media type name: application Media subtype name: ISUP Required parameters: version Optional parameters: base Encoding scheme: binary -10-

11 Examples of ISUP Encapsulation INVITE message = SIP Header + originating SDP information + encapsulated ISUP IAM INVITE sip: @den1.level3.com SIP/2.0 Via: SIP/2.0/UDP den3.level3.com From: sip: @den3.level3.com To: sip: @den1.level3.com Call-ID: DEN @Den1.level3.com CSeq: 8348 INVITE Contact: <sip:jpeterson@level3.com> Content-Length: 436 Content-Type: multipart/mixed; boundary=unique-boundary-1 MIME-Version: unique-boundary-1 Content-Type: application/sdp; charset=iso v=0 o=jpeterson IN IP s=sdp seminar c=in IP4 MG122.level3.com t= m=audio 9092 RTP/AVP unique-boundary-1 Content-Type: application/isup; version=itu; Content-Disposition: signal; handling=optional d a a b 0e 95 1e 1e 1e d f unique-boundary

12 Translation ISUP Message IAM, SAM INVITE SIP Message REL BYE, STATUS ANM 200 ACM, CPG SUS, RES BLO(H/W failure) BLO(Maintenance) RSC, GRS All Others 18x INVITE (INFO) BYE, CANCEL Not affect any ongoing call BYE, CANCEL INFO ISUP Parameter Calling Party Number Called Party Number From SIP Header To, Request-URI -12-

13 SIP to ISUP Mapping (1/2) 18X received If no ACM has been sent yet and no ISUP is present in the 18x Response received Message sent by the MGC 180 Ringing ACM 181 Call is being forwarded Early ACM and CPG, event=6 182 Queued ACM 183 Session progress ACM -13-

14 SIP to ISUP Mapping (2/2) 18X received If an ACM has been sent and no ISUP is present in the 18x Response received Message sent by the MGC 180 Ringing CPG, event=1(alerting) 181 Call is being forwarded CPG, event=6(forwarding) 182 Queued CPG, event=2(progress) 183 Session progress CPG, event=2(progress) -14-

15 ISUP to SIP Mapping CPG received ISUP event code SIP response Alerting Progress In-band information Call forward; line busy Call forward; no reply Call forward; unconditional 180 Ringing 183 Call progress 183 Call progress 181 Call is being forwarded 181 Call is being forwarded 181 Call is being forwarded - (no event code present) 183 Call progress -15-

16 Support for mid-call Signaling SIP INFO Method (RFC2976) An extension add the INFO method to SIP protocol Used for the carrying of mid-call signaling information along the session signaling path For example, SUS/RES/INF/INR in ISUP Used to allow for the carrying of session related control information that is generated during a session. Don t change in the state of SIP calls or parameters of the sessions that SIP initiates -16-

17 Support for mid-call Signaling SIP INFO Method (RFC2976) An extension add the INFO method to SIP protocol Used for the carrying of mid-call signaling information along the session signaling path For example, SUS/RES/INF/INR in ISUP Used to allow for the carrying of session related control information that is generated during a session. Don t change in the state of SIP calls or parameters of the sessions that SIP initiates -17-

18 SIP Bridging Call Flow PSTN MGC#1 MGC#2 PSTN IAM ACM ANM REL RLC INVITE 100 Trying 180 Ringing 200 OK ACK Conversation BYE 200 OK IAM ACM ANM REL RLC -18-

19 PSTN Origination IP Termination PSTN MGC Proxy SIP-phone IAM INVITE 100 Trying INVITE 180 Ringing 180 ACM 200 OK 200 ANM ACK ACK REL RLC Conversation BYE 200 OK BYE

20 IP Origination Termination PSTN SIP-phone Proxy MGC PSTN INVITE ACK INVITE 100 Trying IAM 180 Ringing ACM 200 OK ANM ACK BYE 200 Conversation BYE 200 OK REL RLC -20-

21 ISUP to SIP mapping(originator MGC) Idle REL/F7 Trying IAM/F1 400+/F6 REL/F7 T11/F8 Progressing 18x/F3 200/F4 400+/F6 200/F4 REL/F7 BYE/F9 18x/F3 Alerting 200/F4 Connected 400+/F6 REL/F9-21-

22 SIP to ISUP mapping(terminator MGC) Idle Not alerting Trying INVITE/F1 CANCEL/F3 E.ACM/F5 ACM/F6 T9/F8 T7/F2 CON/F7 REL/F4 CANCEL/F3 CPG/F9 T9/F8 CPG/F9 Alerting REL/F4 CANCEL/F3 ANM/F7 Waiting for ACK REL/F9 BYE/F9 Connected ACK/F10 REL/F9-22-

23 High Level Signaling Flow (IP based) LEX MGC1 TGW1 IP Network TGW2 MGC2 LEX IAM ADD REPLY INVITE with IAM (TWG1 SDP) 100 Trying IAM (COT Prev) ADD REPLY 180 Ringing with ACM (TWG2 SDP) PRACK COT ACM ACM MODIFY REPLY 200 OK Ring-back Tone ANM MODIFY REPLY 200 OK with ANM ACK MODIFY REPLY ANM -23-

24 Software Structure for MG and MGC -24-

25 Development Environment Softswitch H/W Platform : SUN OS : Solaris CASE Tool : Telelogic Tau Compiler : SUN Workshop 6.0 (Forte C/C++) Language: SDL, MSC, C/C++ Principles Design-Oriented Development Scalability, Reusability, Portability, Easy to Maintain -25-

26 Working with the SDL Suite MSC Specification Simulator Validator SDL Specification Analyzer Cadv/Cmicro C Code Generator External C Code C Code Compiler / Linker Master/Cmicro Library Application -26-

27 SIP-T 구현구조 SIP-T Call Control API Management Library ISUP Mapping Transaction Layer Message Handling Supplem entary High Availability Library API UDP/TCP/SCTP -27-

28 BICC 적용분야 Softswitch BICC Softswitch SG SIGTRAN SIGTRAN SG ISUP MEGACO MEGACO ISUP PSTN E1 TG RTP IP/ATM 기반 Packet network RTP TG E1 ISDN -28-

29 Developed in ITU-T SG11 Based on SS7 ISUP Characteristics Quicker to define and to implement, easier ISUP-BICC inter-working Multiple Capability Sets, easing phase deployment -29-

30 Characteristics Architecture Provides a means of supporting narrowband(pstn, ISDN) services across a Packet-based backbone network without impackting the existing network interfaces and end-to-end services Call Control Un-aware of the actual bearer transport being employed. Binding information identifies the bearer used for each call and bearer instance Bearer Control Depends on the underlying bearer technology used -30-

31 Capability Sets BICC protocols have been defined to apply over Packet (ATM or IP) based transport network CS1 CS2 CS3 Applicable to ATM transport with AAL1 or AAL2 Applicable to ATM and IP transport Various enhancements including interworking with SIP (currently under development) -31-

32 Specific capabilities Forward & Backward connection setup Reuse of idle bearer Codec negotiation Codec modification -32-

33 Q.761~Q.764 ISUP Protocols ISUP Q.765 APM Mechanism + other Q APP for BICC TRQ.2140 BICC Architecture Q.2931 DSS2 Protocol Q GIT Mechanism + other Q AAL2 Protocol + other DSS2 AAL2 Q.2764 B-ISUP Protocol Q AGI Mechanism + other B-ISUP Bearer Networks TRQ.3000 DSS2 Bearer TRQ.3010 AAL2 Bearer TRQ.3020 B-ISUP Bearer Q.1901 BICC Protocol (CS1) BICC Signalling Transport Q.1901AnnexC MTP3 & MTP3b Q.1901AnnexD SSCOP + other -33-

34 Q.1901 BICC Protocol (CS1) Bearer Control Q.1990 BICC BCTP Q.1970 BICC IP BCP CS2 Requirement TRQ CS2 TRQ CS2 Signalling Flows Q ~6 BICC Protocol (CS2) TRQ.2500 CS2 CBC Requirement TRQ.2410 CS2 IP BCP, BICC CS2 CS2 Mapping TRQ.303 BICC CS2 with IP BCP (CS1) Signalling Transport Converter Q STC on MTP3 & MTP3b Q STC on SSCOP & SSCOPMCE -34-

35 Control Mechanism The BICC needs transport bearer related information between call control instances. Application Transport Mechanism (APM) used. Q APM for BICC The Bearer setup is requested to the BCF Call Control BICC_Msg Call Control Bearer_Information Bearer_Information Bearer Control Bearer Control -35-

36 APP in BICC Message BICC Message APP Parameter Bearer Information Message Header Parameters (Mandatory, Optional) Parameter #1 Parameter #2 APP Parameter APP Header EAI Q Action indicator BNC-ID Codec List Single Codec BNC Characteristics T-BIWF Address Q

37 Application Transport Mechanism Parameter APP Parameter Represented EAI(Encapsulated Application Information) format within APP Parameter : Q Contain the bearer related informations Action indicator (forward/backwrad/reuse setup ) Backbone Network Connection Identifier (BNC-ID) Codec List or Single Codec (G.7xx) Bearer Network Connection Characteristics (AAL1/AAL2) T-BIWF Address(NSAP format) -37-

38 Network Architecture Serving Node Serving Node SS7 ISUP Call & Service Functions BICC Call Mediation Node (Opt) BICC Call & Service Functions SS7 ISUP Bearer Control Function Bearer Signalling Bearer Signalling Bearer Control Function TDM Trunk Bearer Function Packet (ATM/IP) Transport Network Bearer Function TDM Trunk -38-

39 Protocol Stack -39-

40 High Level Message Flow (ATM based) LEX MGC MGC LEX IAM ADD REPLY IAM SETUP ADD IAM CONNECT CONNECT ACK REPLY COT ACM MDFY REPLY ACM ANM MDFY REPLY ACM ANM MDFY ANM REPLY -40-

41 Conclusion SIP-T IETF SIPPING WG An application of SIP 1 st Candidate of inter-mgc communication for VoIP SIP is a perfect complement to today s Next Generation Network BICC ITU-T SG11 Based on SS7 ISUP 1 st Candidate of inter-mgc communication for VoATM Good Performance and Easy to Implement 기술이전 NGN 프로토콜 : SIP-T, BICC, MEGACO, SIGTRAN -41-