Transcription

Cisco BTS 10200 Softswitch SIP Guide,Release 6.0.3October 30, 2012Americas HeadquartersCisco Systems, Inc.170 West Tasman DriveSan Jose, CA 95134-1706USAhttp://www.cisco.comTel: 408 526-4000800 553-NETS (6387)Fax: 408 527-0883Text Part Number: OL-25004-02

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALLSTATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUTWARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THATSHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSEOR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCB’s publicdomain version of the UNIX operating system. All rights reserved. Copyright 1981, Regents of the University of California.NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS” WITHALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUTLIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OFDEALING, USAGE, OR TRADE PRACTICE.IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING,WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCOOR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to thisURL: www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnershiprelationship between Cisco and any other company. (1110R)Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in thedocument are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3Copyright 2011-2012 Cisco Systems, Inc. All rights reserved.

CONTENTSPreface5Organization5Obtaining Documentation and Submitting a Service RequestDocument Change HistoryCHAPTER16SIP Network Overview1-1General SIP Overview1-1Compliance51-2SIP Functions Performed by the BTS 10200 1-2Interworking 1-3SIP Cause Codes 1-4SIP Registrar 1-4User Agent Client and User Agent Server 1-4Back-to-Back User Agent 1-5SIP xGCP SDP Interworking FeatureLimitations 1-8Industry Standards 1-8CHAPTER2SIP Subscribers1-72-1SIP Phone Initialization2-2Provisioning a SIP Subscriber2-2SIP Registration and Security 2-2Enhanced SIP Registration 2-3Operations 2-6Measurements 2-8Events and Alarms 2-8SIP User AuthenticationSIP Subscriber Calls2-92-10Provisioning Session Timers for SIP SubscribersSIP Timer Values for SIP SubscribersDiversion Indication for SIP SubscribersSIP Privacy Header 2-12SIP Signaling Details2-112-112-122-13Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-021

ContentsPRIVACY Token 2-14Feature Interactions 2-15Prerequisites 2-15Limitations 2-15Feature Considerations 2-15Provisioning 2-15Comparison of SIP-Based Features and MGCP-Based Features2-16Cisco BTS 10200 Softswitch-Based Features 2-24Summary 2-24Call Forwarding 2-26Call Park and Directed Call Pickup Features 2-27Calling Name and Number Delivery 2-29Caller ID Delivery Suppression 2-29Customer Access Treatment 2-30Direct Inward Dialing 2-30Direct Outward Dialing 2-30Do Not Disturb 2-31E.164 and Centrex Dialing Plan (Extension Dialing) 2-31Operator Services (0-, 0 , 01 , and 00 Calls) 2-32User-Level Privacy 2-32Vertical Service Code Features 2-32Voice Mail 2-33Jointly Provided Features 2-37Call Transfer (Blind and Attended) with REFER 2-38Distinctive Ringing 2-38Distinctive Ringing for Centrex DID Calls 2-38Phone-Based FeaturesCHAPTER3SIP Trunks2-383-1General Characteristics and Usage of SIP TrunksSIP Trunk Provisioning ExampleCall Processing on SIP Trunks3-23-23-3Validation of Source IP Address for Incoming SIP MessagesLoop Detection3-43-4Locating SIP Servers Through DNS QueriesReliable Provisional Responses3-10Provisioning Session Timers for SIP TrunksSIP Timer Values for SIP Trunks3-53-123-13Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.32OL-25004-02

ContentsSIP Route Advance3-14SIP Status Monitoring and SIP Element Audit 3-14Status Monitoring of SIP Elements 3-14SIP Trunk Group States 3-18Internal SIP Audit 3-19SIP Element Audit 3-20SIP Triggers3-22Call Redirection3-22Support for Sending 302 on Call ForwardingDiversion Indication for SIP Trunks3-243-26Number Portability Information and Carrier Identification CodeSIP Trunk Subgroups3-29SIP-T, ISUP Version, ISUP Transparency, and GTDDTMF SIP Signaling3-333-35Asserted Identity and User-Level PrivacyThird-Party Call Control3-373-40ANI-Based Routing 3-40ANI Screening on Incoming Calls3-41T.38 Fax Relay CA Controlled Mode Across SIP Trunk InterfaceSIP Call Transfer with REFER and SIP INVITE with ReplacesSIP Trunk to Voice-Mail ServerCluster Routing3-273-423-433-483-49CMS-to-MGC Routing3-49SIP Server Groups 3-50Purpose of the SIP Server Groups Feature 3-50Provisionable Parameters Affecting SIP Server Groups 3-50Understanding SIP Server Group Operations 3-51Outbound SIP Messages That Apply to SIP Server Groups 3-54SIP Element Selection Algorithm 3-60Applications and Use Cases for SIP Server Groups 3-65Limitations on SIP Server Groups 3-69Provisioning SIP Server Groups 3-71Troubleshooting SIP Server Groups 3-72SIP Trunk Call Admission Control3-72Restrictions and Limitations 3-74Configuring SIP Trunk Call Admission ControlSIP Trunk Group Authentication and Registration3-743-75Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-023

ContentsLimitations 3-78Interoperability 3-78Provisioning 3-78Measurements 3-80Troubleshooting 3-81SIP Trunking for PBX ConnectionCHAPTER4SIP System Features3-824-1SIP Timer Values 4-1Rules for Configuring the SIP Timers 4-1Detailed Description of Timers 4-2Computation of Default Timer Values A Through J from Timers T1 and T4Calculation of Timer Retransmission Count 4-5SIP Session Timers 4-7Session Timers Description 4-8Upgrades and SIP Session TimersUsing the EXPIRES Header 4-94-9Limitations on Number of URLs, Parameters, and HeadersDifferentiated Services Codepoint4-54-94-12Message Handling Based On Content-Length HeaderLimitation On Transient Calls During Switchover4-13Automatic DNS Monitoring and Congestion ControlAutomatic Fault Monitoring and Self-Healing4-124-134-13SIP Enhancements 4-14Prerequisites 4-14Limitations 4-14SIP Traffic Measurement EnhancementsSummary Report Changes 4-15Trunk Group Usage Counters 4-16Call Processing Counters 4-164-14Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.34OL-25004-02

PrefaceRevised: October 30, 2012, OL-25004-02This document describes the Cisco BTS 10200 Softswitch features applicable to Session InitiationProtocol (SIP) subscribers and trunks. It also provides the procedures necessary to provision thesefeatures.OrganizationThis SIP Guide contains the following chapters: Chapter 1, “SIP Network Overview”—Provides an overview of the BTS 10200 functions in the SIPnetwork. Chapter 2, “SIP Subscribers”—Explains how to provision and use the features applicable to SIPsubscribers. Chapter 3, “SIP Trunks”—Explains how to provision and use the features applicable to SIP trunks. Chapter 4, “SIP System Features”—Explains how to provision and use features applicable to all SIPsystem operations.Obtaining Documentation and Submitting a Service RequestFor information on obtaining documentation, submitting a service request, and gathering additionalinformation, see the monthly What’s New in Cisco Product Documentation, which also lists all new andrevised Cisco technical documentation, w/whatsnew.htmlSubscribe to the What’s New in Cisco Product Documentation as a Really Simple Syndication (RSS) feedand set content to be delivered directly to your desktop using a reader application. The RSS feeds are a freeservice and Cisco currently supports RSS version 2.0.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-025

PrefaceDocument Change HistoryThe following table lists the revision history for the Cisco BTS 10200 Softswitch SIP Guide,Release 6.0.3.Version NumberIssue DateStatusReason for ChangeOL-25004-02October 30,2012RevisedUpdated the “Provisioning Session Timers for SIPSubscribers” section on page 11.OL-25004-01August 10,2011InitialInitial document for Release 6.0.3.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.36OL-25004-02

CH A P T E R1SIP Network OverviewRevised: October 30, 2012, OL-25004-02This guide describes the Session Initiation Protocol (SIP) signaling features supported in Release 6.0.3of the Cisco BTS 10200 Softswitch, and explains how to provision them.NoteIn this document, the term “SIP devices” includes SIP phones and softclients that act as a SIP user agent(UA) to originate and terminate calls for an address of record (AOR) identity.This chapter contains an overview of the SIP network and includes the following sections: General SIP Overview, page 1-1 Compliance, page 1-2 SIP Functions Performed by the BTS 10200, page 1-2General SIP OverviewThe SIP support features are built on the existing BTS 10200 software and hardware platform. TheBTS 10200 includes a Call Agent (CA), Feature Server (FS), Element Management System (EMS), andBulk Data Management System (BDMS). In this book, use of the term “BTS 10200” indicates the CallAgent unless otherwise specified.The BTS 10200 uses SIP and SIP for telephones (SIP-T) signaling to communicate with other SIP-basednetwork elements. The implementation is based on the evolving industry standards for SIP, includingIETF document RFC 3261, SIP: Session Initiation Protocol. The BTS 10200 supports both SIP trunksand SIP-based subscriber lines (SIP devices), and provides the following SIP-related functions: Protocol conversion between SIP and several other protocols, including Signaling System 7 (SS7),primary rate interface (PRI) Integrated Services Digital Network (ISDN), H.323, Media GatewayControl Protocol (MGCP), and Channel Associated Signaling (CAS). Tandem back-to-back user agent for direct SIP-to-SIP calls (trunk to trunk, phone to phone, andtrunk to/from phone), and SIP-to-SIP-T calls. SS7 bridging between softswitches using SIP-T methods. Native support of SIP endpoints such as SIP phones, including authentication and registrationmanagement. (For example, the BTS 10200 maintains the current location of SIP subscribers.)Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-021-1

Chapter 1SIP Network OverviewComplianceThe BTS 10200 provides billing data for SIP calls. Specific fields are supported in the call detail recordsfor calls that originate or terminate on a SIP trunk or subscriber. For detailed information on these fields,including billing management and data, refer to the Cisco BTS 10200 Softswitch Billing Interface Guide.ComplianceThe BTS 10200 SIP implementation is based on the evolving standards in the Internet Engineering TaskForce (IETF) Request for Comments (RFC) publications, including the documents in the following list,and may not be fully compliant in all cases. The BTS 10200 is largely compliant with RFC 3261. Forthe level of compliance with all other RFC publications and drafts referenced in this document, see thespecific feature descriptions. RFC 2617, HTTP Authentication RFC 2976, SIP INFO Method RFC 3261, SIP: Session Initiation Protocol RFC 3262, Reliability of Provisional Responses in the Session Initiation Protocol (SIP) RFC 3263, Session Initiation Protocol (SIP): Locating SIP Servers RFC 3265, Session Initiation Protocol (SIP)-Specific Event Notification RFC 3311, The Session Initiation Protocol (SIP) UPDATE Method RFC 3372, Session Initiation Protocol for Telephones (SIP-T): Context and Architectures RFC 3398, Integrated Services Digital Network (ISDN) User Part (ISUP) to Session InitiationProtocol (SIP) Mapping RFC 3515, The Session Initiation Protocol (SIP) Refer Method RFC 3891, The Session Initiation Protocol (SIP) Replaces Header RFC 3892, The Session Initiation Protocol (SIP) Referred-By Mechanism RFC 4028, Session Timers in the Session Initiation Protocol (SIP)SIP Functions Performed by the BTS 10200The BTS 10200 supports call processing for SIP trunks and phone users. As a result of native SIPsubscriber support, SIP subscribers can use features similar to those available to MGCP subscribers.NoteFor a comparison of the MGCP and SIP feature support, see the “Comparison of SIP-Based Features andMGCP-Based Features” section on page 2-16.Figure 1-1 shows a network architecture example in which the BTS 10200 provides native support forSIP subscribers and SIP trunks. As shown in this drawing, the BTS 10200 can establish calls betweennetworks with various protocols, including calls between SIP trunks and SIP subscribers. In the SIPnetwork, the BTS 10200 provides Registrar services with SIP user authentication.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.31-2OL-25004-02

Chapter 1SIP Network OverviewSIP Functions Performed by the BTS 10200Example of Network Architecture with the BTS 10200Cisco BTS10200 H.323 TGTGWCisco 5350H.323 networkH.323ETSI PRICASSIPCisco 2600SIP TrunkgroupSageSIP TGSIPMGCPIPSIPSIP7905SIP7905IPCisco2421IPSIP VMCSPSSIP7905RGWIPIPIPVMIPSIP7960IPSIP7960SIPIP 79607960ATAIP7905ATA87897Figure 1-1SIP functions performed by the BTS 10200 include:Note User agent server (UAS) User agent client (UAC) Registrar SIP subscriber authenticationThe Cisco BTS 10200, as part of the back-to-back functionality, plays the role of the UAS and UAC.Most features provided by the SIP phones comply with Local Access and Transport Area (LATA)Switching Systems Generic Requirements (LSSGR), depending on the phone implementation andcapabilities. Due to the nature of the SIP protocol, however, your experience with a feature might differfrom what is documented in the LSSGR for that feature.InterworkingThe system supports interworking combinations between SIP subscribers and the following entities: H.323 trunks SIP trunks Public switched telephone network (PSTN)—SS7 and ISDN user part (ISUP) ISDNCisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-021-3

Chapter 1SIP Network OverviewSIP Functions Performed by the BTS 10200 MGCP subscribersSIP Cause CodesFor information on SIP cause codes and their relation to ITU-T standard Q.850 cause codes, see the “SIPCause Code Mapping” section in the Provisioning Guide.SIP RegistrarSIP Registrar support enables SIP subscribers to be served by the BTS 10200 directly. The BTS 10200acts as a Registrar and authenticates the SIP request. SIP subscribers register with the BTS 10200 andoriginate calls through the BTS 10200.To initiate a session with a SIP subscriber, the BTS 10200 must know the current address of thesubscriber. Registration creates bindings in a location service for a particular domain. The bindingsassociate an address-of-record Uniform Resource Identifier (URI) with one or more contact addresses.A SIP subscriber notifies the BTS 10200 of its availability at the address provided in the contact for thespecified duration. The BTS 10200 uses the challenge-based Digest Access Authentication toauthenticate the SIP subscriber. (Digest Access Authentication is described in RFC 2617.)The SIP subscriber registers with the BTS 10200, setting up a binding between the AOR and its contactaddress. The registration is valid for a period of time specified by the SIP phone in the REGISTERmessage, after which the registration expires. If the SIP phone does not specify a time period forexpiration, the BTS 10200 applies a default timer, SIA REGISTER DEFAULT EXPIRES, which isprovisionable in the Call Agent Configuration (ca-config) table. The BTS 10200 also requires that theduration specified by the phone be in a range between the values provisioned forSIA REG MIN EXPIRES SECS and SIA REG MAX EXPIRES SECS in the ca-config table. Toprovision these parameters, see the procedure in the “Provisioning a SIP Subscriber” section onpage 2-2.Figure 1-2 demonstrates the SIP phone Registrar function.Figure 1-2SIP Phone Register FunctionCisco BTS 10200Softswitch RegistrarIPRegisterRegistered Contact dataAuthentication data87893SIP PhoneUser Agent Client and User Agent ServerThe user agent is a software application running on a SIP system.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.31-4OL-25004-02

Chapter 1SIP Network OverviewSIP Functions Performed by the BTS 10200The user agent can work either as a client or server. When a call is placed, the UAC places the request,and the UAS services the request and sends a suitable response. The roles change continually, however;for example, with call hold, either user can put the other user on hold.Figure 1-3 shows the BTS 10200 working as a UAC, sending out a call request.Figure 1-3User Agent ClientCisco BTS 10200User Agent ClientSS7InviteMGCPH.323SIP Phone87894IPRegisteredContact dataFigure 1-4 shows the BTS 10200 working as a UAS, accepting a call request.Figure 1-4User Agent ServerCisco BTS 10200User Agent ServerSS7MGCPH.323SIP PhoneSubscriber dataAuthentication data87895IPInviteBack-to-Back User AgentThe back-to-back user agent acts as a UAC and UAS for a single call. It keeps the two call segmentsseparate on the BTS 10200. Typically, a proxy routes a call, but does not act as a user agent. TheBTS 10200 acts as a user agent. In a call between two SIP endpoints (such as SIP phone or SIP trunk),the BTS 10200 terminates the originating half of the call, playing the UAS role, and then sets up theterminating half of the call as a UAC.NoteThere is no provisioning associated with the back-to-back functionality. The BTS 10200 automaticallyacts as a back-to-back user agent for a SIP-to-SIP call.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-021-5

Chapter 1SIP Network OverviewSIP Functions Performed by the BTS 10200Figure 1-5 shows the BTS 10200 working as a back-to-back user agent.Figure 1-5Back-to-Back User Agent ServerCisco BTS 10200User Agent Server/User Agent ClientIPInviteInviteIPSIP PhoneSIP Phone87896Registered Contact dataSubscriber dataAuthentication dataFigure 1-6 shows the call flow for registration.Figure 1-6Call Flow for RegistrationCisco BTS 10200SoftswitchSIP PhoneIPRegister401Register104308200Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.31-6OL-25004-02

Chapter 1SIP Network OverviewSIP xGCP SDP Interworking FeatureFigure 1-7 shows the call flow for a back-to-back user agent.Figure 1-7Back-to-Back User Agent Call Flow with AuthenticationCisco BTS 10200SoftswitchSIP Phone 1SIP Phone 98BYEBYE200200SIP xGCP SDP Interworking FeatureSIP and the protocols represented by the term xGCP use the Session Description Protocol (SDP).NoteIn this document, the term xGCP refers to the following protocols:Media Gateway Control Protocol (MGCP)Network-based Call Signaling (NCSTrunking Gateway Control Protocol (TGCP)—TGCP meets requirements for the Media GatewayController-to-Media Gateway interfaceSIP and xGCP use SDP differently. SIP and xGCP exchange SDP transparently using the Call Agent(Cable Management System, Media Gateway Controller). However, the exchange of SDP data createsinterworking problems because xGCP might ignore or reject SIP SDP syntax. Using this feature, theCisco BTS 10200 translates SIP SDP syntax into equivalent xGCP connection-handling syntax.To enable the Cisco BTS 10200 to operate SIP xGCP SDP Interworking, configure the following tokensin the MGW PROFILE table to suit the specific protocol interworking required between theCisco BTS 10200 and media gateways in your network:SDP XGCP SIP IWF SUPPSDP BANDWIDTH AS ONLYSDP MULTIPLE MEDIA DESC SUPPCisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-021-7

Chapter 1SIP Network OverviewSIP xGCP SDP Interworking FeatureSDP PTIME ADD FROM MPTIMESDP PTIME ADD FROM LCOSDP CALL SETUP IWF SUPPSDP PORT ZERO SUPPSDP IP ZERO SUPPNoteFor complete CLI information, see the Cisco BTS 10200 Softswitch CLI Database.LimitationsWhen the Cisco BTS 10200 operates the SIP xGCP SDP Interworking feature it requires additionalprocessing time. To minimize the additional time required to process this feature, consider the followingtwo criteria when you configure the MGW PROFILE table: Capacity— Set only the relevant MGW PROFILE table tokens to Y when interworking is required. Calls Per Second (CPS)—This feature scans SDP for each call. Set only the MGW PROFILE token,SDP XGCP SIP IWF SUPP to Y when interworking is required.Industry StandardsThis feature implements SIP-xGCP SDP interworking requirements defined in PacketCable EC(Engineering Change) CMSS1.5-N-07.0407-5.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.31-8OL-25004-02

CH A P T E R2SIP SubscribersRevised: October 30, 2012, OL-25004-02The Cisco BTS 10200 Softswitch supports SIP subscribers on SIP phones that are compliant withRFC 3261 or RFC 2543. This section describes the support for SIP subscribers and how to provision SIPsubscriber features.In this document:Note SIP subscriber means a SIP phone that is registered directly to the BTS 10200 and for which theBTS 10200 maintains subscriber information. SIP Automatic Number Identification (ANI)-based subscriber means a SIP phone thatcommunicates with the BTS 10200 over a SIP trunk.For quick-reference tables listing the subscriber features, see the “Comparison of SIP-Based Featuresand MGCP-Based Features” section on page 2-16.This section covers the following topics: SIP Phone Initialization, page 2-2 Provisioning a SIP Subscriber, page 2-2 SIP Registration and Security, page 2-2 SIP User Authentication, page 2-9 SIP Subscriber Calls, page 2-10 Provisioning Session Timers for SIP Subscribers, page 2-11 SIP Timer Values for SIP Subscribers, page 2-11 Diversion Indication for SIP Subscribers, page 2-12 Comparison of SIP-Based Features and MGCP-Based Features, page 2-16 Cisco BTS 10200 Softswitch-Based Features, page 2-24 Jointly Provided Features, page 2-37 Phone-Based Features, page 2-38Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-022-1

Chapter 2SIP SubscribersSIP Phone InitializationSIP Phone InitializationFigure 2-1 shows an example of SIP phone initialization on bootup, that is, how a typical phone mightinitialize itself and establish its identity with the BTS 10200. (The image shows actions that occurexternal to the BTS 10200—it does not show how the BTS 10200 controls SIP initialization.) The circlednumbers in the image indicate the numerical order in which the sequence occurs.Figure 2-1Example of SIP Phone InitializationDNSTFTPDHCPIP Addr,Gateway,TFTP Srv & FilesConfig File,Image, SIP Info42Cisco BTS10200'sIP Addr6Help mebootWho am I?Cisco BTS10200 IPAddress?53Cisco BTS 10200Softswitch8200 OKREGISTER7IP878991Provisioning a SIP SubscriberTo provision a SIP subscriber, see the “SIP Subscribers” section in the Provisioning Guide.SIP Registration and SecuritySIP subscribers use the SIP REGISTER method to record their current locations with the BTS 10200.Registering clients can specify an expiration time for the contacts being registered. However, theBTS 10200 has a minimum and maximum acceptable duration, both of which are configurable.NoteThird-party registration is not supported.It is possible to register multiple contacts for a single AOR; however, if multiple contacts are registeredfor a single subscriber, the BTS 10200 uses only the most recently registered contact to deliver the callto that subscriber. For this reason, multiple contacts are not supported.NoteOnly one contact should be registered for an AOR.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.32-2OL-25004-02

Chapter 2SIP SubscribersSIP Registration and SecurityWhen a SIP user attempts to register or set up a call, the BTS 10200 challenges the SIP subscriber basedon provisioning in the serving-domain-name table. If the serving-domain-name table indicates thatauthentication is required, the BTS 10200 challenges the SIP request (Register/INVITE) according tothe authentication procedures specified in SIP Protocol RFC 3261. If the BTS 10200 receives validcredentials, the authenticated AOR from the user-auth table identifies the subscriber based on theaor2sub table. (For specific provisioning parameters, see the applicable tables in theCisco BTS 10200 Softswitch CLI Database.)Registration creates bindings in the BTS 10200 that associate an AOR with one or more contactaddresses.The registration data is replicated on the standby BTS 10200. The BTS 10200 imposes a minimumregistration interval as a provisionable value. If the expiration duration of the incoming registrationrequest is lower than the provisioned minimum, a 423 (Interval Too Brief) response is sent to theregistering SIP endpoint.The BTS 10200 generates a warning event when a request from a client fails authentication. This canindicate a provisioning error or an attempt by an unauthorized client to communicate with theBTS 10200.The contacts registered for an AOR can be looked up using the status command, as demonstrated by thefollowing example.CLI status sip-reg-contact AOR ID [email protected] ID - [email protected] - 4695550184HOST - 10.88.11.237PORT - 5060USER TYPE - USER PHONE TYPEEXPIRES - 3600EXPIRETIME - Thu Jan 22 14:33:36 2004STATUS - REGISTERED CONTACTReply :Success:Enhanced SIP RegistrationSIP Registration ensures that a SIP REGISTER message to the BTS 10200 is from a provisionedendpoint, that is, an endpoint with a provisioned secure fully qualified domain name (FQDN) or IPaddress. The feature also ensures that the source IP address and contact parameter for all originating callsare from the provisioned SIP endpoint, and that no calls can originate from an unregistered endpoint.DescriptionPrior to Release 4.5.1, SIP endpoint registration was based on AOR, user ID, and password; there wasno verification of the origination of the REGISTER message. Certain service providers may prefer thatthe source IP address of SIP requests be verified against a provisioned FQDN of the endpoint to addressthe possibility of theft of VoIP service.The BTS 10200 can indicate SECURE FQDN provisioning for specified SIP term-type subscribers.This indication consists of specifying an FQDN with the subscriber AOR. The FQDN is theaddress/location of the SIP endpoint and is added to the AOR table. The FQDN does not have a serviceport.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3OL-25004-022-3

Chapter 2SIP SubscribersSIP Registration and SecurityTo enable or disable SECURE FQDN on a successful registered subscriber:1.Take AOR out of service to remove all registered contacts.2.Enable or disable SECURE FQDN for the subscriber.3.Bring AOR back in service (INS).4.Reboot the analog terminal adapter (ATA).A subscriber with the secure FQDN feature enabled has the following characteristics: One and only one AOR is associated with the endpoint. Does not have any static-contact associated with it. User ID and Password Authentication are supported. One FQDN (specified without service port). The DNS lookup of the FQDN should result in one and only one IP address. Cannot place or receive a call unless successfully registered.ExampleThis example presents a case in which a VoIP subscriber (Subscriber 1) uses the following options forthe user ID, password, and phone number: user-id-1 password-1 phone-no-1Without security, another VoIP subscriber, Subscriber 2, could access Subscriber 1’s information(perhaps by getting a Cisco ATA configuration file with the encryption key in clear text, and then gettingthe full configuration file with all the data). Subscriber 2 could then register with the BTS 10200 withSubscriber 1’s combination of user-id-1, password-1, and phone-no-1, as well as Subscriber 2’s own IPaddress. Without the secure FQDN feature, the Cisco BTS 10200 would accept this information unlessspecific measures were taken, and Subscriber 2 could steal service and make calls on behalf ofSubscriber 1.Provisioning CommandsThis section shows the CLI commands you need to provision a secure FQDN of a SIP endpoint.NoteUse this procedure to provision subscribers on the BTS 10200. The procedure does not cover the securityof configuration files provisioned on the SIP adapter (for example, an ATA), which are the responsibilityof the service provider.The SECURE FQDN token is present in both the subscriber and aor2sub tables. A non-null value in thefield indicates that the SECURE FQDN validations apply to all SIP messages received from theendpoint associated with that AOR. The SECURE FQDN value can be specified on a subscriber only if the AOR for the subscriber isout of service (OOS). When an AOR is taken administratively OOS, its registered contacts aredeleted. A static contact cannot be specified for a SECURE FQDN subscriber. Any existing static contactrecord for an AOR must be deleted before the subscriber can be made a SECURE FQDN SIPendpoint.Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.32-4OL-25004-02

Chapter 2SIP SubscribersSIP Registration and Security Th

Contents 3 Cisco BTS 10200 Softswitch SIP Guide, Release 6.0.3 OL-25004-02 SIP Route Advance 3-14 SIP Status Monitoring and SIP Element Audit 3-14 Status Monitoring of SIP Elements 3-14 SIP Trunk Group States 3-18 Internal SIP Audit 3-19 SIP Element Audit 3-20 SIP Triggers 3-22 Call Redirection 3-22 Support for Sending 302 on Call Forwarding 3-24 Diversion Indication for SIP Trunks 3-26