The SSCA SIP Training Program The SSCA SIP Training Program

Transcription

The SSCA SIPtraining programThe SSCA SIP training programOverviewThe SIP School is ‘the’ place to learn all about the Session Initiation Protocol alsoknown as SIP. There is so much information on the internet about SIP that is bothhard to read and poorly presented making it difficult for people to learn about this mostimportant protocol. So, The SIP School with its lively, clear and fully animatedeLearning program has become the only place you need to learn about SIP.Who would benefit from the SSCA SIP training program?Everyone ! This training is designed to suit anyone working with SIP such as:Manufacturers of IP PBX and IP Phone equipment, SIP Security equipmentmanufacturers, SIP Trunk service providers, Hosted/Cloud service providers, Carriers,Mobile Network Operators, Network Design specialists, Sales and Marketing personnelworking with Voice and Video over IP equipment and services; all of these will benefitfrom this program.What is in the SSCA SIP training program?Once you have enrolled, you’ll see 14 modules. You can work through the modules inorder or simply choose the ones you are most interested in. The modules are listedhere but for more detail, please look further into this document by clicking on themodule names in the list below.1. Core SIP2. Wireshark3. SIP and the PSTN4. SIP, VVoIP and QoS5. SIP and Media Security6. STIR/SHAKEN and the ‘identity’ problem7. Firewalls, NAT and Session Border Controllers8. SIP Trunking9. Testing, Troubleshooting and Interoperability10. ENUM, Peering and Interconnect11. SIP in the Cloud12. SIP in Cellular networks13. SIP and Fax over IP14. SIP in UC, UCaaS and CPaaSNOTE:This program was last updated on March 22nd 2021. All new / edited sections areshown in a bold, blue font. Copyright Vocale Ltd and The SIP School , All rights reservedPage 2

The SSCA SIPtraining programHow long will it take to work through?Running times for this program are approximate as the time will vary based on thestudent’s own experience and of course, how much time they want to spend on thematerial and if they want to replay some modules.NOTE: The time it takes to ‘Play’ all the slides and Videos (also known as the ‘running’time) plus complete all of the quizzes is 16hours 15mins.The TOTAL time will be more than this and dependent on factors such asslides being replayed, note taking, working on Labs (some of which can take afew hours), also doing some ‘extra’ work with the Software tools provided forthe labs which we believe is a great idea as it increases student skills.Further study time for the SSCA and the taking of the SSCA final test itselfshould also be accounted for.Is there a Pre-requisite to this program?This program assumes the student has a ‘good’ understanding of Data networkingtechnologies along with the ‘basics’ of Voice and Video over IP. This could be gainedthrough long term working experience, other certifications such as Cisco’sCCENT/CCNA/CCNP, even The SIP School’s own ‘Networking for VVoIP program’also available via the website. Please check carefully as having the skills required willmake the SIP learning experience a more productive one.Become a ‘SIP School Certified Associate’ or SSCA You can gain access to the certification test separately or with a ‘bundle’ license –check license ‘purchase’ options carefully.The SSCA certification is recognized in the Telecommunications world as the onlycertification on SIP to strive for ‘Globally’. It is endorsed and supported by USTelecom,Incompas, the ITSPA along with BICSI and an extensive number of Manufacturers,Service providers, Cloud providers, Carriers and Mobile Network OperatorsTo prepare for the certification test, each SIP training module has its own ‘mini’ quiz atthe end to help delegates ‘gauge’ how well they are doing.NOTE: An access license for any training course and certification test is for 12 monthsfrom the date of purchase. Copyright Vocale Ltd and The SIP School . All rights reservedPage 3

SSCA SIPCore SIPDescriptionSIP (The SessionInitiation Protocol)is described inthis module alongwith the manyother componentsand Services thatwill beencountered on aSIP basednetworkModule timesRunning time87 minutesQuizzes10 minutesTotal97 minutes1Core SIPtopicsSIP Why SIP?What is SIP?SIP ‘from the RFC’What are ‘Requests for Comments’ – RFCs?More than just 3261New RFCsIETF Working groupsBased on HTTPWhere does SIP fit in?SIP Clients and ServersSIP User AgentsSIP Dialog - INVITESIP System ArchitectureThe URI - Unique Resource IdentifierSIP AddressingSIP Addressing ExamplesSIP Servers and Operation RegistrationRe-RegistrationSIP Proxy servers and why we need themProxy Server ‘State’ typesDHCP and SIPSIP Proxy – Trapezoid ModelSIP Server – Proxy ModeSIP Server – Re-Direct ModeLocation ServicesSIP Server in Proxy ModeSIP Server in Proxy Redirect ModeStateful and Stateless ProxiesLocation ServeroComponentsoInformation SourcesoExampleSIP Client Configuration Configuration scenariosSome basic elements needed to configure a clientSIP Messaging Request MethodsResponse CodesSIP HeadersINVITE – ExampleRESPONSE (200 OK) – ExampleMore on HeadersSupport and Require HeadersoTimer (Session Times)o100rel (PRACK)Short form ‘compact’ Headers Copyright Vocale Ltd and The SIP School . All rights reservedPage 4

SSCA SIPCore SIPSDP – the Session Description Protocol SDP – The Session Description ProtocolSDP in a SIP MessageAn SDP ExampleExtending SDPMultiple ‘m’ linesChanging Session ParametersSDP Example - Put a call on HoldSDP Example - Call Hold TraceCall Hold – Old and New MethodsMusic on Hold exampleINVITE and reINVITESIP Mobility SIP MobilitySIP Call Forking - ParallelSIP Call Forking - SequentialCall legs, dialogs and Call IDsDialog trace exampleDialogs and TransactionsBranch IdsCall Forward to VoicemailCall Forward - No AnswerReplaces headerDiversion headersHistory-infoMore on Proxies and SIP Routing Stateless ProxyStateful ProxyMore Proxy informationVIA and Record RouteVIA DetailsRecord-Route DefinedRecord Route ExampleLoose and Strict RoutingSession PoliciesMIME MIMEMultiple MIME partsSIP and B2BUA B2BUA - Back to Back User AgentB2BUA ExampleB2BUA Benefits and FeaturesSIP ‘Call Process’ Summary The Call Process Copyright Vocale Ltd and The SIP School . All rights reservedPage 5

SSCA SIPWiresharkDescriptionThis module onWireshark is anintroduction that isintended to getstudents setupquickly so thatthey can capturetraffic to analyzeduring the CoreSIP module andthe rest of thecourse. MoreadvancedWireshark trainingcan be found intheTroubleshooting,Testing andInteroperabilitymodule of thiscourse.2topicsWireshark Module timesRunning time44 minutesQuizzes1 minuteLabs equate toexercisessuggested withinthe moduleWireshark What is Wireshark?Initial SetupFree SIP Account optionsFree @thesipschool.com SIP account / addressTest NumbersDesktop clientsoJitsi client for testingoBlink client for testingoBria Solo client for testingoPhonerLite client for testingMobile clientsoBria Solo for testingoMizuPhone for testingoLinphone for testingoWeePhone SIP for testingSIP phone in a BrowserSIP Browser clientsFree DID and CreditSecurity and SIP in WiresharkSocial Study directorySecurity and SIP in WiresharkDownload WiresharkWiresharkoIntroductionoMenus, Screens and ViewsoCapturing trafficoProfilesoDisplay FiltersoCapture FiltersoSIP Packet AnalysisoSIP ladders and Audio PlaybackoOther Menu optionsoSIP INVITE AnalysisoFollow a UDP StreamoFrame RelationshipsoColouring RulesoRTP StreamsUse the CloudPCAPs from ‘other’ placesLAB ExercisesWhat are the codes?Approx. lab time80 minutesTotal125 minutes Copyright Vocale Ltd and The SIP School . All rights reservedPage 6

SSCA SIPSIP and the PSTNDescriptionSIP Networks willof course have toallow connectionsto and from thePSTN. Thismodule worksthrough SIP andPSTNconnectivityModule timesRunning time24 minutesQuizzes7 minutesTotal31 minutes3SIP and the PSTNtopicsSIP and the PSTN SIP to PSTN OverviewSIP to PSTN Call FlowSIP to PSTN DetailPSTN to SIP Call FlowSIP to PSTN Call FailureSIP Codes and the PSTNEarly Media Early Media explainedEarly Media - SIP to PSTN CallEarly Offer and Delayed Offer Early Offer / Delayed OfferGateways Default Gateway?Gateways and expectationsSIP-T and PSTN Bridging SIP-T and SIP-ISS7, ISDN and SIPISUP and SIP MessagesISDN User Part (ISUP) to SIP CodesPSTN to PSTN via SIPISUP EncapsulationISUP Encapsulation / SDPAddressing NotesSIP and DTMF DTMF - Quick Re-CapWhat is DTMF?Inband vs Out-of-bandRFC 2833 ‘Trace’ exampleRFC 4733 replaces 2833RFC 4734SIP INFO 6086RFC 2833 ‘Trace’ exampleSIP INFO ‘Trace’ example Copyright Vocale Ltd and The SIP School . All rights reservedPage 7

SSCA SIPSIP, VVoIP andQoSDescriptionThis modulestarts as a‘refresher’ moduleon the basics ofVoice over IPbefore diggingdeeper. It thenmoves on tocover Video overIP and throughoutthe module thereis a big focus onthe componentsand (good) QoSpractices that areimportant to a SIPbased Network.Module timesRunning time82 minutesQuizzes7 minutesLabs equate toexercisessuggested withinthe moduleApprox. lab time10 minutesTotal99 minutes4SIP, VVoIP and QoStopicsWhat is VoIP or Voice over IP? What is VoIP?What is Voice over IP?VoIP – ‘A Basic Call’VoIP and TCP / UDPVoIP over the InternetBranch to Branch VoIPSignaling pathsSpeech pathsIP PBXVoice Sampling and Codecs EncodingCodecs for VoiceDynamic [RTP Payload type]The ‘Codec Test’MOS, R-Factor and High Definition (HD) VoiceSound testsCodecs and BandwidthPacket Rate / Packets per secondVariable bit rate / Constant bit rate codecsWideband (HD) codecsOpus codecOpus audio examplesThe Real Time Protocol or RTP RTP IntroRTP EncapsulationRTP Header TraceReal Time Control Protocol (RTCP)RTCP-XR (Extended Reports)RTP / RTCP and UDP PortsQuality of Service QoS describedQoS IssuesMeasuring DelayJitter and Packet LossGeneral VoIP Acceptance CriteriaQoS across all Networks802.1Q – VLANs802.1Q/P Tagging802.1P - L2 ClassificationTOS and DiffServLayer 3 ClassificationDSCP with Assured forwarding (AF)Bandwidth decisionsLink options – Symmetric DSL (SDSL)Bandwidth (kbps) vs. Packet per Second (pps)Network Behavior AnalysisIssues that can affect QoSQoS SummaryTesting your link Copyright Vocale Ltd and The SIP School . All rights reservedPage 8

SSCA SIPSIP, VVoIP andQoSSIP, SDP and VoIP SIP in the TCP/IP ModelSIP and SDP Messages (e.g. Invite and 200OK)SIP and SDP Codec mappingVideo over IP What is Video over IP?Streaming Voice and Video – 1 Way TransmissionTwo-way Conferencing with RTPCodec and Bandwidth ConsiderationsVideo bitrate CalculatorSetting Video Codecs on DevicesAudio and Video in the SDP bodyAssured SIP Services Assured SIP introService Provider ArchitectureProxy and Access Router functionsResource-PriorityVideo ‘example’Reason Header for Pre-emption EventsMore Proxy detailsMulti-Level Pre-emption and Precedence (MLPP)Summary Copyright Vocale Ltd and The SIP School . All rights reservedPage 9

SSCA SIPSIP and MediaSecurityDescriptionSIP and MediaSecurity is acomplex subjectto address yet isimperative forsafecommunicationstoday. Thismodule coversmany Securityproblems andchallenges alongwith possiblesolutions.Module timesRunning time55 minutesQuizzes5 minutesLabs equate tovarious exercisessuggested withinthe moduleApprox. lab time60 minutesTotal120 minutes5SIP and MediaSecuritytopicsAuthentication and Authorization SIP Proxy Authentication – in detail401 and 407 AuthorizationSIP AuthorizationPROXY AuthenticationHashing Algorithms [MD5, SHA etc.]Encryption Why Encrypt SIP?Encryption types (Symmetric / Asymmetric)Keying and HashingCertificate AuthoritiesCertificate ExampleThe Certificate application processInstalling your new CertificateBackup your Private keySelf-Signed CertificatesPublic Key Infrastructure - PKITLS – Transport Layer Security TLS in ActionTLS 1.2 Capture exampleTLS 1.3SSL/TLS checkingSecuring SIP signaling Securing SIP Signalling and then the media‘SIPS’ addressingTLS and SIP in actionCombinations of what you may see Securing the Media Stream Secure RTP (SRTP)Setting SRTP on SIP DevicesSecure RTP (SRTP) - ExampleSRTP and SRTCPsdes and the Crypto attributeCrypto attribute exampleSRTP Call example ‘showing’ CryptoCrypto – multiple streamsDTLS/SRTPSRTP with ZRTPEncryption summarySIP trunks and Security SIP trunks and SecurityEnhancing SIP Trunk Security Copyright Vocale Ltd and The SIP School . All rights reservedPage 10

SSCA SIPSIP and MediaSecurityAttacks and Responses Types of Attack on a VoIP/SIP NetworkFBI network examplesResponses and ProtectionResponse Identity – A Problem!Rogue SIP ProxyPhishing and SIP exploitMore Examples RFC 4475Try for yourself with ‘example’ software toolsNIST Recommendations NIST Recommendations on securing VoIP3rd party training to extend your knowledge The SANS institute Copyright Vocale Ltd and The SIP School . All rights reservedPage 11

SSCA SIPSTIR/SHAKEN andthe ‘identity’problemDescriptionThere is anexisting (andgrowing) issuewith illegalrobocalls, andspam calls andthis module aimsto teach thestudent what isbeing doneinitially in NorthAmerica with aneye on the rest ofthe world.STIR/SHAKEN isthe FCCmandatedsolution forproving identity ona call and it ishappening now.Module timesRunning time109Quizzes10 minutesTotal119 minutesSTIR/SHAKEN and the6‘identity problem’topicsSTIR/SHAKEN Introduction and topicsWho’s calling? The PSTN Caller ID Spoofing problemThe ‘scale’ of the problem (USA)Caller Identity Caller IdentityEnterprise IdentitiesP-Preferred and P-AssertedCNAM/eCNAM Spoofing a number - VideoSpoofingSTIR/SHAKEN Robocalling and moreWhy this is a problemA First Step: STIR/SHAKENSTIR/SHAKEN in a NutshellWhat is a PASSporT?Haven’t I Heard of SIP Identity Already?STIR/SHAKEN ArchitectureSigned INVITE ExamplePASSporT Token from ExamplePASSporT Token in JSONPASSporT Token Protected HeaderPASSporT Token PayloadThe ‘digital signature’Fetching CertificateSuccess Call FlowFailure Call Flow – Missing Identity HeaderFailure Call Flow – Bad Identity HeaderCertificate management for STIR/SHAKENPartner systemSTI Certificate for AuthenticationAttestationThe SIP School ‘test system’VerstatSTIR/SHAKEN in actionVideo - Authentication to VerificationService providers with SHAKENEnterprises and the ‘A’ The ‘Attestation gap’How to ‘fix’ the gap – some optionsDelegate CertificatesDelegate Certificates base PASSporTDelegate Certificates for OTT providersEnterprise CertificatesTN DatabasesDistributed LedgerTrustGetting ‘Creative’Which option is best? Copyright Vocale Ltd and The SIP School . All rights reservedPage 12

SSCA SIPSTIR/SHAKEN andthe ‘identity’problemRich Call Data What is Rich Call Data?Rich Call data locationAdding Rich Call DataRich Call Data in the tokenRCD jCard / rcdiRCD and Delegated certsInternational STIR/SHAKEN International AttestationATIS and International calls – BilateralATIS and International calls – Central RegistryOut of Band STIR/SHAKEN Why is this a problem?Out of Band (OOB) STIR with TDMAnother OOB exampleCall Diversion Diverted call flow“div” in a SIP INVITE“div-o”Call Analytics An overviewWhat’s happening now The Traced ActWhere are we now?‘Other Services and TechniquesBringing it all togetherPossible extensionsFCC mandateRobocall mitigationFind the call originatorIndustry Traceback Group (ITG)Resources ‘Some’ other companies offering STIR/SHAKENATIS testbedSTIR and SHAKEN referencesSTIR/SHAKEN conferenceBest practices. Copyright Vocale Ltd and The SIP School . All rights reservedPage 13

SSCA SIPFirewalls, NAT andSession BorderControllersDescriptionInevitably, all IPtraffic traverses aFirewall / NATdevice and in thecase of SIP thesedevices can stopthe flow of SIPmessages. Thismodule looks atthe problems andthe solutionsincluding a focuson SessionBorderControllers.Module timesRunning time64 minutesQuizzes10 minutesTotal74 minutesFirewalls, NAT and Session7Border ControllerstopicsOverview Issues to addressFirewalls What does a Firewall do?Are Firewalls effective?NAT or Network Address Translation What is NAT?NAT RequestNAT ResponseUDP Hole punchingNAT HairpinningMedia Hairpinning/TromboningMultiple NATsNAT in more detail Types of NATNAT – Full ConeNAT – Restricted ConeNAT – Port Restricted ConeNAT – SymmetricNew TerminologiesoMapping and FilteringEndpoint Independent MappingAddress Dependent MappingAddress and Port Dependent MappingNAT Filtering RulesThe NAT & Firewall ‘problem’ The NAT problemThe NAPT or (PAT) ProblemThe Firewall ProblemThe Solutions Interactive Connectivity Establishment (ICE)‘Classic STUN’ (Session Traversal Utilities for NAT)VIA received parameterVIA rport parameterProblems with ‘Classic’ STUNSymmetric RTPSTUN RFC 8489Request and Response exampleTURN (Traversal Using Relays around NAT)ICE ‘In Theory’Candidate information and other ‘ICE stuff’.ICE ‘In action’ICE tagsICE-Lite and Trickle-ICEICE Client settingsMore on ICEMedia Proxy Copyright Vocale Ltd and The SIP School . All rights reservedPage 14

SSCA SIPFirewalls, NAT andSession BorderControllersThe Solutions (continued) Application Level GatewaySIP Aware Firewalls - IncomingSIP Aware Firewalls - OutgoingUniversal Plug and Play (UPnP)‘Near end’ NAT‘Far end’ NATGRUU (Globally Routable User Agent)Session Border Controllers SBC for the Enterprise and SBC for the ITSPRecommended Session Border Controller featuresSBCs in Action!SBCs and message manipulation / normalizationSIP ‘Refer’ problemsSBC ‘Interop’ exampleSBC Manufacturers – examplesSBCs in the Cloud / as a Service Copyright Vocale Ltd and The SIP School . All rights reservedPage 15

SSCA SIPSIP TrunkingDescriptionThis moduleteaches thetheory ofconnecting a SIPbased PBX intoan ITSPs ownnetwork and alsofocuses onNetworktechnologies,Security,Troubleshootingas well as offeringadvice on how toselect an ITSP foryour company orclients.Module timesRunning time84 minutesQuizzes7 minutesLabs equate tomultiple exercises‘suggested’ withinthe module suchas SIP PBX andtrunkconfiguration.Approx. lab time120 minutesTotal211 minutes8SIP TrunkingtopicsSIP Trunks What is a SIP TrunkAlternative to TDMSeparate Data and Voice connectionsConverging the networkSIP Trunks and CodecsSIP Trunk BenefitsSIP Trunking – In More Depth SIP Trunk CapabilitiesSIP Trunking Network ExamplesSIP PeeringPeering problems?Least Cost routing (LCR)Disaster RecoveryDisaster Recovery ‘Expanded detail’Disaster Recovery – Last resort?Number ConsolidationVirtual PresencesTrunking Variations Single Site, No ‘Forklift’Single Site, TDM PBXSingle Site, ConvergedConverged – SIP/IP PBXMultiple Site, ‘Converged’Multiple Site, ‘Converged’ central SBCMultiple Site, ‘Converged’ Multiple SBCsMedia Gateways SIP PBX to Non-SIP PBXSIP PBX to Non-SIP PBX, Call FlowSIP Trunk Performance Connection typesThe ADSL issueCodecs, Voice and DataSymmetric DSL (SDSL)Bandwidth CalculatorTesting your linkADSL DevelopmentsFibre OptionsTrunk ‘bursting’Elastic SIP Copyright Vocale Ltd and The SIP School . All rights reservedPage 16

SSCA SIPSIP TrunkingSIP Trunks, MPLS and SD-WAN MPLS, basic explanationMPLS Label formatMPLS in a MAC frameMPLS example networkMPLS benefitsYour own private WANbut ‘Not the only client’Separate MPLS networksVPLS explainedWAN Optimization, Hybrids and SD-WANSoftware Defined WANs explainedoOrchestratoroPoliciesoSD-WAN device capabilitiesSetting up a SIP Trunk SIP trunk configuration on ‘sample’ PBXOutbound ‘Dialling’ RuleCalling across the trunkCall analysis with WiresharkoCall FlowoSIP ladderModes of Operation Registration ModeStatic ModeSecurity and SIP Trunks SIP Trunk Security - OverviewMicrosoft (a little) Skype for Business and SIP TrunksServers and ProtocolsMicrosoft Teams and Calling plansMicrosoft Teams and Direct RoutingTroubleshooting and Interops SIP Trunks and Common ProblemsThe SIP ForumSIP ConnectSIP Connect 1.1 onto 2.0Interoperability testingChoosing an ITSP Understanding ITSP Offerings'Sticking points’?What you may need in the futureSIP trunk ‘connectivity’oThings to watch out for when connecting to your ITSP‘Finding’ an ITSPSIP trunking Checklist for ITSP evaluation Copyright Vocale Ltd and The SIP School . All rights reservedPage 17

SSCA SIPTesting,troubleshootingand Interop.DescriptionLearn how to‘monitor’ and TestSIP devices andservices usingWireshark. Thistool enablesdelegates toanalyze callcontrol messagesto establish wherea fault may lie in aSIP infrastructure.Full examples areprovided, anddelegates areencouraged tofollow theexercises to tryfor themselves.Module timesRunning time55 minutesQuizzes7 minutesLabs equate tovarious exercisessuggested withinthe moduleApprox. lab time240 minutesTotal302 minutesTesting, Troubleshooting9and InteroperabilitytopicsSetting up your test environment Your SetupUsing SIP IP Phones and SoftphonesJitsi, Blink, Bria Solo and PhonerLite setup – reminder.Choosing a ‘Trial/Test’ ITSPGet ‘another’ SIP accountSIP2SIP accountConfigure Blink and Jitsi on the same PC for testingUsing ‘Test Numbers’Wireshark Where to ‘capture’More options for Packet CapturingWireshark ‘Revisited’Colours and the Intelligent ScrollbarPacket ‘Marking’ and ‘Comments’New Packet WindowExporting ‘Specified’ FramesRTP StreamsTShark (Terminal-based Wireshark)PCAP-ng and PCAP formatsAlternatives to WiresharkYou try!Interoperability Testing Interop Testing and why Interop can be toughDifferent interpretations in the RFC 3261Interop Test ScenarioInterop Test OperationsSample Interop Traces with WiresharkWireshark example videos to help understand interop issuesMore Sample capturesVideo call testingVideo tests with Wireshark trace analysis‘Basic’ Interop Test ListSIPIT eventsCommon SIP problems Will it ever work?Where can you start checking?What else can you do?Common SIP/VoIP ProblemsTroubleshooting SIP Trunks4xx — Client Failure Responses5xx — Server Failure Responses6xx — Global Failure ResponsesMore SIP Testing Tools SIP WorkbenchSIP ScanVisualware for testingHoverIPNSLookupVoip-info for more tools!Using the NET to find answersOther SIP Resources Copyright Vocale Ltd and The SIP School . All rights reservedPage 18

SSCA SIPENUM, Peeringand InterconnectDescriptionENUM (along withDNS) isdeveloping into anessential protocolon SIP networksand its purpose isto assist in findingdestination SIPdevices from asingle SIPaddress. Peeringis also discussedas more and moreservices providersare ‘connecting’together to allowa full IP to IPexperience.Inclusion of theIP-NNIrecommendationbuilds on‘Peering’ toenable ITSPs to‘Peer’ in a moreeffective manner.Module timesRunning time62 minutesQuizzes7 minutesTotal69 minutes10ENUM, Peering andInterconnecttopicsENUM Explained What is E.164?What is ENUM?Why ENUM?Call Routing and ENUM - ExampleEnum, DNS and Domains Why are we using DNS?DNS OperationDNS Root Server ‘Mirrors’‘Finding’ Domain name servers using NSLookupThe e164.arpa DomainApproved ENUM Delegations (RIPE)TIERS 0, 1, 2 and 3e164.arpa Domain ‘in action‘ENUM DelegationsAddress of RecordPSTN to SIP UA – ExampleThe ENUM QueryDNS Response to an ENUM queryNAPTR and DNS recordsFinding SIP servers using the tool - DIGIP to PSTN (Simplified)RFC 6140Types of ENUM Different ‘Types’ of ENUMThe Problems with ‘Public’ ENUMExample – ‘Private’ ENUM‘Carrier’ ENUM and e164enum.netPeering and Interconnect (for VoIP and Video) Stay ‘On-NetFrom ITSP to PSTN and Back !Loss of features with the PSTNPeering Profiles and AgreementsBi-lateral PeeringMulti-lateral PeeringBack to ENUMA complete ‘infrastructure’Who’s involved?IP-NNI Network-to-Network interface [NNI]ATIS and the SIP Forum for NNIBenefits of SIP NNIHistory of IP NNI EffortLayers of InterconnectionoIP Interconnection ProfileoIP Interconnection RoutingIP NNI ProfileIP NNI Trust Model Copyright Vocale Ltd and The SIP School . All rights reservedPage 19

SSCA SIPENUM, Peeringand InterconnectIP-NNI (continued) IdentitiesCodecsDTMF and FaxFault Isolation and TroubleshootingQoSSIP-Specific Details of IP NNIIP Interconnection RoutingAggregate ApproachPer-Telephone Number (TN) ApproachWhat’s Next for NNI Copyright Vocale Ltd and The SIP School . All rights reservedPage 20

SSCA SIPSIP in the CloudDescriptionSIP is critical tophones andservers involvedin a hosted VoIPand Videoservice. Thismodule aims toshow the studentwhat ‘the cloud’ isall about alongwith its manydifferentdeployment types.Videos will takethe studentthrough ‘cloud’deployment of anSBC and PBXalong with SIPtrunk connectivityfor a full ‘cloudbased’ VoIPservice.11Quizzes7 minutesLabs in thismodule areoptional and donot count towardscourse runningtime. Labs couldtake from ½ dayupwards - alldepending onwhat the studentwould like toattempt.Total71 minutestopics‘Types' of ‘Cloud’ Public, Private and HybridHosted SIP What Hosted SIP service isHosted functions and featuresExample Network including ‘failover’‘Hosted’ clients in actionWhy Hosted – Benefits and things to considerWhy on-site PBX – Benefits and things to considerThe Cloud and ‘Anything as a Service’ Module timesRunning time64 minutesSIP in the CloudPizza as a ServiceIaaS / PaaS / SaaSSaaS in ‘reality’oWhat is Virtualization?oVirtual MachinesoEmulationoVirtual Machines (contd.)oNetwork Functions Virtualization (and VNF)oSBCs in the Cloud / as a ServiceVirtualization of the PBXOur own Network examplesMoving to the CloudoExample with - AWS / Azure and TwiliooCall flow in the example ‘Cloud based’ systemVideo demos of ‘Cloud systems’ Visualising the migration to the cloudCloud marketplacesAzure – Anyone VM SBCRDP connection to the SBCAnynode configurationSBC and TwilioAWS instancesAWS and 3CX (PBX)SBC, PBX and TwilioCapturing the ‘Cloud call’Auto Provisioning Auto Provisioning ExampleBoot ServerClient ConfigClient boot sequenceClient config downloadRFC 6011Zero-Touch ProvisioningZero-touch exampleBenefits of Hosted SIP ServiceBenefits of Onsite PBX and SIP trunksTroubleshooting Troubleshooting a cloud serviceWhat to look for – DashboardsWhat to look for – PCAP filesMonitoring across cloud-based services Copyright Vocale Ltd and The SIP School . All rights reservedPage 21

SSCA SIPSIP in Cellularnetworks12DescriptionSIP is a criticalpart of VoLTE andVoNR callingacross Cellularnetworks. Thismodule aims tomake studentsaware of SIPsrole in all theseenvironments.Quizzes5 minutesTotal57 minutestopicsSIP in Cellular networks Module timesRunning time52 minutesSIP in Cellularnetworks Network OverviewRAN, eNodeB, EPC, IP Core and 3GPP4G, LTE, LTE Advanced LTA-Pro, WiMAX2The RAN and EPCDefault Bearer SetupIntroduction to the Servers and Functions in the IMSoCSCFoS-CSCFoP-CSCFoI-CSCFoHome Subscriber Server HSSoApplication ServeroTASoPSCFoDNS and ENUMDevice Registration (with SIP)SIP Registration packet exampleSIP in the IMS – Call Flow explainedIntroduction to VoLTE and the threat of OTT servicesMaking VoLTE workoSIP Preconditions in ActionoWith Codec examples within SDPSIP Call flow for VoLTEQuality settings ‘recap’VoLTE media flowMore on VoLTEThe IMSLayers architectureoApplicationoIMS / Session ControloAccess and Transporto3GPPMultiple access devicesoRCS and OTTWho provides IMS solutions?IPX and Peering for Security, QoS and SLAsGSMA and IR.92HD Voice NewsVoLTE media flowMore on VoLTEThe IMSLayers architectureoApplicationoIMS / Session ControloAccess and Transporto3GPPMultiple access devicesoRCS and OTTWho provides IMS solutions?IPX and Peering for Security, QoS and SLAsGSMA and IR.92 Copyright Vocale Ltd and The SIP School . All rights reservedPage 22

SSCA SIPSIP in Cellularnetworks5G Benefits of 5G5G service examplesVoice over 5G5G – NSA Option 3x (and more)Mandatory CodecsSIP in 5GSummarizing the state of 5GResourcesCoverage Checker Copyright Vocale Ltd and The SIP School . All rights reservedPage 23

SSCA SIPSIP and Fax overIPDescriptionA lot ofcompanies arenow trying to runFax servicesacross SIP trunksand finding it’s notan easy service toget workingsuccessfully. Thismodule intends todescribe thevarious flavors ofFax over IP alongwhat should befocused on inorder totroubleshoot anyissues.Module timesRunning time38 minutesQuizzes7 minutesTotal45 minutes13SIP and Fax over IPtopicsFaxing Basics Faxing backgroundT.30 Fax signalingAssociated tones and protocolsThe ITU and TIA standardsFax over IP Fax over IP benefitsFrom the old to the newIntro to FoIPFoIP and SIP trunksProtocol conversionsFax Protocols G.711 Pass-throughT.37 Store and ForwardT.38 RelayWhere does SIP fit in?UDPTLProtocol options for the futureFoIP in action SIP in FoIP – Call FlowSIP INVITEINVITE for T.38The INVITE SDP bodyWireshark FoIP exampleSIP T.38 Call flows – IETF draft documentBandwidth T.38 and G.711 network trafficTroubleshooting The basicsMore complex issues to watch out forOngoing Efforts RFC 6913 and sip.fax tagUse DTMF events instead? Copyright Vocale Ltd and The SIP School . All rights reservedPage 24

SSCA SIPSIP in UC, UCaaSand CPaaSDescriptionSIP in UC,UCaaS andCPaaS showsyou how SIPunderpins all theelements ofUnifiedCommunicationsservices andCPaaSapplications torealizeefficiencies that asuccessfulimplementationpromises tobusiness.Module timesRunning time58 minutesQuizzes7 minutesTotal65 minutes12SIP in UC, UCaaS andCPaaStopicsCommunication Breakdown Playing Voicemail tagCan’t find peopleAvailable but not Available.!More Examples of communication problemsIM Clients IM Client Examples and FeaturesClients and UC providersMore IM Cl

4. SIP, VVoIP and QoS 5. SIP and Media Security 6. STIR/SHAKEN and the 'identity' problem 7. Firewalls, NAT and Session Border Controllers 8. SIP Trunking 9. Testing, Troubleshooting and Interoperability 10. ENUM, Peering and Interconnect 11. SIP in the Cloud 12. SIP in Cellular networks 13. SIP and Fax over IP 14. SIP in UC, UCaaS and .