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Design & Implementation of SIPTrunking using Cisco’s SessionBorder ControllersGraham Francis – CEO, The SIP SchoolDarryl Sladden– Technical Marketing Manager, CiscoPashmeen Mistry – Technical Marketing Engineer, CiscoOctober 27th 2011 2011 Cisco and/or its affiliates. All rights reserved.1

Founded in April 2000 5300 Students Provide the Industry recognised SSCA SIP Certificationprogram, endorsed by the TIA more. eLearning in modular format Unique as content evolves as SIP evolves Connected with Cisco to provide SIP foundation training http://cisco.thesipschool.com / Discount codes later. Now, let’s talk about why we’re all here today and we’ll start withSIP 2011 Cisco and/or its affiliates. All rights reserved.3

Enabling Business-to-Business CollaborationEnterprise Domain 1AEnterprise Domain 2IPIP Changing Landscapes –Narrowband voiceto Rich-mediaInterconnectVoIP Islands to VoIPInterconnectsA Extend rich-mediacollaboration to vendors,partners and customers A Cisco Unified BorderElement (CUBE) providesb2b interconnectivity forsecure rich-media services Unified communications SIPTrunks to destinationsbeyond the EnterpriseEnterpriseDomain 1AIP 2011 Cisco and/or its affiliates. All rights reserved.SP VoIPCUBESBCSBCCUBEIPAEnterpriseDomain 216

1. TDM Trunking – YesterdayCVPAACampusContact Center2. TDM and IP Trunking – TodayBranchOfficesSP SIPACVPA3. IP Trunking – TomorrowCampusCVPASP SIPCampusContact CenterBranchOfficesAContact CenterBranchOffices 2011 Cisco and/or its affiliates. All rights reserved.18

Impact of an SBCChallenge I have multiple PBXs that all need tohave SIP Trunking enabled in order toget the best Return on Investment(ROI). Allows you to have a singleinterconnect point to your ServiceProvider across multiple disparatesystems. I would like to centralize all of my SIPTrunking in a single location. Allows you to scale your SIP Trunksolution while only connecting toone device. SIP Trunking is complex newtechnology, how do I make Troubleshooting easier. Allows a single point oftroubleshooting for your SIP Trunks.A device that is supported by Ciscoallows you to have one vendorsupport your entire solution. How can I ensure that I am compliantwith my company’s security policieswhen I implement SIP Trunking ? SBC’s ensures security on SIPTrunks. An SBC from a trustedvendor such as Cisco incorporatessecurity in all aspects from anembedded firewall to administrativecontrol on changes. 2011 Cisco and/or its affiliates. All rights reserved.Features of aCisco SBC19

Slide 19mrf4New oneMike Fratesi, 14/05/2008

An Integrated Network Infrastructure ServiceTDM GatewayCisco Unified Border Element Address Hiding Voice and Video TDMInterconnect H.323 and SIP interworking PSTN Backup DTMF interworking SIP security TranscodingRouting, FW,IPS, QoSCUBENote: An SBC appliance wouldhave only these featuresUnified CMConferencing andTranscodingWAN InterfacesRSVPAgent 2011 Cisco and/or its affiliates. All rights reserved.SRSTVXMLGKNote: Somefeatures/componentsmay require additional21licensing

ASR 1004/6 RP250-150ASR 100150-1003900E ISR G2End of Life platformsLast IOS Release: 15.1.4M20-35ASR 1002Introducedin Nov ‘1017CPS3900 ISR G2AS5000XM8-123800 ISR2900 ISR G22800 ISR 52801 ISR800/1861 ISRIntroduced in Mar ‘114 50 2011 Cisco and/or its affiliates. All rights reserved.500-600600-800900-1000 1500-1700 2000-2500 10-12KActive Voice Call (Session) Capacity12-16K 22

CUBE Session Capacity SummaryPlatformCUBE SessionsC880/C890 80039459503925E21003945E2500ASR1002/1004/1006 RP11750ASR100110000ASR1004/1006 RP216000 2011 Cisco and/or its affiliates. All rights reserved.ReferenceIntroduced inMarch 2011End of Life PlatformsLast IOS Release:15.1.4MASR1001 introduced inRLS 3.2 in Nov 201023

Reduced Pricing for redundancyPlatformCisco 2901, 2911, 2921 ISR G2Cisco 2951, 3925 ISR G2Cisco 3945, 3925E, 3945E ISR G2Cisco ASR1000Single-Use -CUBEE-4KPFLASR1-CUBEE-16KPActive-Standby B2BRedundancy CUBEE-4K-RFLASR1-CUBEE-16KRMore info in the CUBE Ordering oicesw/ps6790/gatecont/ps5640/order guide c07 462222.html 2011 Cisco and/or its affiliates. All rights reserved.24

TDMSIP Trunksfor PSTNAccessNetworkbasedMediaRecordingSolutionSIP TrunkSIPH.323SP VOIPSBC ServivcesCUBEPartner APIIVRIntegrationfor ContactCentersBusiness toBusinessTelepresence 2011 Cisco and/or its affiliates. All rights reserved.MediaSenseRTPSIPSIPCUBERTPCVPvXML ServerSP IPNetworkSBCMediaServerSIPCUBESIPACUBESBCSP IPNetworkSP IPNetworkSBCSIPACUBE26

Characteristics of CentralizedOperational Benefits Central Site is the only locationwith SIP session connectivity toIP PSTN Centralizes PhysicalOperations Centralizes Dial-PeerManagement Centralizes SIP TrunkCapacity Voice services delivered toBranch Offices over theEnterprise IP WAN (usuallyMPLS) Media traffic hairpins throughcentral site between SP andbranchesChallenges Increased campus and branchbandwidth, CAC, latency; mediaoptimization HA in campus (single point offailure) Survivability (backup branch callprocessing) Emergency services Legal/Regulatory, GeographicalCentralizedIP PSTNACUBEEnterpriseIP WANSite-SP Media 2011 Cisco and/or its affiliates. All rights reserved.28

Characteristics ofDistributedOperational BenefitsChallenges Each site has direct connectionfor SIP sessions to SP Leverages existing branchrouters Takes advantage of SP sessionpooling, if offered by SP No media hair-pinning thru anysite. Media traffic goes direct fromeach branch site to the SP Lower latency on voice or video Built-in Redundancy strategy Quickest transition from existingTDM Distributed dial-peer management Distributed operational overheadDistributedIP PSTNAEnterpriseIP WANCUBECUBECUBECUBECUBESite-SP Media 2011 Cisco and/or its affiliates. All rights reserved.30

Characteristics of HybridBenefitsConnection to SP SIP service is determined on a siteby site basis to be either direct or routed through aregional site. Adaptable to site specific requirements Optimizes BW use on Enterprise WAN Decision to route call direct or indirect based onvarious criteria Adaptable to regional SP issues Built-in redundancy strategy Media traffic goes direct from site to SP or hairpinsthrough another site, depending on branchconfiguration. CUBEHybridIP PSTNAAEnterpriseIP WANCUBECUBECUBESite-SP Media 2011 Cisco and/or its affiliates. All rights reserved.31

Validated with serviceproviders world-wide Tested with 3rd partyPBXs Standards basedCisco Interoperability Portal:www.cisco.com/go/interoperability 2011 Cisco and/or its affiliates. All rights reserved.32

Re-purpose your existing Cisco Voice Gateway’s as Cisco’s Session BorderController – Cisco Unified Border Element (CUBE)Digital/AnalogTrunksSIP/H.323Trunksx 1001dial-peer voice 1 voipdestination-pattern 1.session protocol sipv2session target ipv4: PBX IP Addr codec g711ulawBuy CUBE LicenseOnlySIP TrunksSIP/H.323Trunksx 1001dial-peer voice 1 voipdestination-pattern 9Tsession protocol sipv2session target ipv4: PBX IP Addr 2011 Ciscoand/or its affiliates. All rights reserved.codecg711ulawdial-peer voice 2 potsdestination-pattern 9Tport 0/0/0:23SIP SPCUBESBCChange POTSCall Leg toVoIP Call Legdial-peer voice 2 voipdestination-pattern 9Tsession protocol sipv2session target ipv4: SIP Trunk Provider IP Addr 34codec g711ulaw

Actively involved in the call treatment,signaling and media streams SIP B2B User Agent Signaling is terminated, interpretedand re-originated Provides full inspection of signaling, andprotection against malformed andmalicious packetsCUBEIPMedia Flow-Through Signaling and media terminated by theCisco Unified Border Element Transcoding and complete IP addresshiding require this model Media is handled in two differentmodes: Media Flow-Through Media Flow-Around Digital Signal Processors (DSPs) arerequired for transcoding (calls withdissimilar codecs) 2011 Cisco and/or its affiliates. All rights reserved.CUBEIPMedia Flow-Around Signaling and media terminated by theCisco Unified Border Element Media bypasses the Cisco UnifiedBorder Element35

InternalNetworkSIP 120.1.1.2 1 408-526-6855B2B UserAgentx1001192.168.1.50INVITE /w SDPINVITE /w SDPc 192.168.1.50m audio abc RTP/AVP 0c 20.1.1.1m audio xxx RTP/AVP 0100 TRYING100 TRYING180 RINGING180 RINGING200 OK200 OKc 20.1.1.2m audio uvw RTP/AVP 0c 10.1.1.1m audio xyz RTP/AVP 0ACKACKRTP (Audio)192.168.1.50 2011 Cisco and/or its affiliates. All rights reserved.10.1.1.120.1.1.120.1.1.236

SignalingPacketsMediaPacketsVoice Application CodeVoice ApplicationCodeL7 Protocol-independent memory structures holding callstate and attributes (CLID, Called #, CodecM)Dial-PeerDial-peerSIP/H323Protocol StackDial-peerSIP/H.323Protocol StackTCP/UDP/TLS VoicestackDTMF xlationCodec FilteringXcoding ControlSIP/H.323Protocol StackRTP LibraryRTP LibraryDSP APIIOS InfrastructureTCP UDP TLSPhysicalInterfacesIngress I/FDSP Hardware 2011 Cisco and/or its affiliates. All rights reserved.RTP LibraryIOSInfrastructureTCP UDP TLSIOS Infrastructure (ACLs, FW, IPS, VPN)HW LAN/WAN InterfacesSignalingDSP (Ifinvoked)PhysicalInterfacesEgress I/FMedia37

Step 0 – Configure IP PBX to route calls to the edge SBC Step 1 – Get SIP Trunk details from the Provider Step 2 – Turn CUBE Application ON on Cisco routers Step 3 – Configure Call routing on CUBE (Incoming & Outgoing Dial-Peers) Step 4 – Normalize SIP messages to meet SIP Trunk Provider’srequirements Step 5 – Execute the Test Plan 2010 Cisco and/or its affiliates. All rights reserved.Cisco Confidential38

SIPSIPCUBESBCSP IPNetworkSIP Trunk pointing to CUBE Configure CUCM to routecalls to CUBE via aSIP/H323 Trunk Make sure all differentpatterns of calls – local,long distance,international, emergency,informational etc. arepointing to CUBE 2010 Cisco and/or its affiliates. All rights reserved.Cisco Confidential39

ItemSIP Trunk service provider requirementSampleResponse1SIP Trunk IP Address (Destination IP Address for INVITES)20.1.1.22SIP Trunk Port number (Destination port number forINVITES)50603SIP Trunk Transport Layer (UDP or TCP)UDP4Codecs supportedG711,G7295Fax protocol supportT.386DTMF signaling mechanismRFC28337Does the provider require SDP information in initial INVITE(Early offer required)Yes8SBC’s external IP address that is required for the SP toaccept/authenticate calls (Source IP Address for INVITES)20.1.1.19Does SP require SIP Trunk registration for each DID? If yes,what is the username & passwordNo10Does SP require Digest Authentication? If yes, what is theusername & password 2010 Cisco and/or its affiliates. All rights reserved.NoCisco Confidential40

1. Turn CUBE Application “ON”voice service voipmode border-element license capacity 200allow-connections sip to sip3. Create a trusted list of IPaddresses to prevent toll-fraudvoice service voipip address trusted listipv4 10.1.1.50ipv4 20.20.20.202. Global settings to meet SP’s requirements and SIPTrunk towards SP if neededvoice service voipsipearly-offer forcedheader-passing error-passthrumidcall-signaling passthru 2010 Cisco and/or its affiliates. All rights reserved.Cisco Confidential41

Dial-peer – “static routing” table mapping phone numbers to interfacesor IP addressesINBOUND & OUTBOUND CALLSSIPH.323 or SIPCUBELAN DialPeersSP IPNetworkSBCWAN Dial-Peers LAN Dial-Peers – Dial-Peers that are facing towards the IP PBX forsending & receiving calls to & from the PBX WAN Dial-Peers – Dial-Peers that are facing towards the SIP TrunkProvider for sending & receiving calls to & from the provider 2010 Cisco and/or its affiliates. All rights reserved.Cisco Confidential42

INBOUND DP FOR CALL FROM CUCM TO CUBEOUTBOUND DP FOR CALLS FROM CUBE TO CUCMdial-peer voice 100 voipdescription *** LAN side dial-peer ***incoming called-number 9Tsession protocol sipv2destination-pattern [2-9].voice-class sip bind control source gig0/0voice-class sip bind media source gig0/0session target ipv4: CUCM Address codec g711ulawdtmf-relay rtp-nteCUCM sending 9 All digits dialedSP will besending 10 digitsinboundNote: If more than 1 CUCM exists, you will haveto create multiple such LAN dial-peers with“preference CLI” for CUCM redundancy 2011 Cisco and/or its affiliates. All rights reserved.43

INCOMING WAN DIAL-PEER FOR CALLS FROM SP TO CUBEdial-peer voice 200 voipdescription *** WAN side Incoming DP ***incoming called-number [2-9].session protocol sipv2dtmf-relay rtp-nteCatch-all for allSP inbound callsOUTGOING WAN DIAL-PEER FOR CALLS TO SP FROM CUBEdial-peer voice 201 voipdescription *** WAN side dial-peer Long distance***translation-profile outgoing Digitstrip 9DP for sendingdestination-pattern 91[2-9].long distancesession protocol sipv2calls to SPvoice-class sip bind control source gig0/1voice-class sip bind media source gig0/1session target ipv4: SIP Trunk Provider IP address dtmf-relay rtp-ntecodec g729r8Note: Separate outgoing DP to be createdfor Local, International, Emergency,Informational calls etc. Thus, for WANInbound & Outbound DP are separate 2011 Cisco and/or its affiliates. All rights reserved.44

SIP ProviderRequirement1. For Call Forward & Transfer scenarios back to PSTN, the Diversion header should match theregistered DID of your network2. The User-Agent field in all SIP messages should state the version of PBX and of SBC that is beingusedConfigureSIP Profilesvoice class sip-profiles 400request INVITE sip-header Diversion modify “sip:(.* )” “sip:4085266855@sip.abc.com ”request REINVITE sip-header Diversion modify “sip:(.* )” “sip:4085266855@sip.abc.com ”request ANY sip-header User-Agent modify “User-Agent:(.*)” “User-Agent: Cisco CUCM8.5/IOS-15.1-3”response ANY sip-header Server modify “Server:(.*)” “Server: Cisco CUCM8.5/IOS-15.1-3”Apply toDial-peer orGloballySee thedifferencedial-peer voice 4000 voipdescription Incoming/outgoing SPvoice-class sip profiles 400voice service voipsipsip profiles 1000Received:INVITE sip:2000@9.44.44.71:5060 SIP/2.0SSSUser-Agent: Cisco-CUCM8.5SSSDiversion: sip:3000@9.44.44.4 ;privacy off;reason unconditional;screen yesSS.m audio 6001 RTP/AVP 0 8 18 101a rtpmap:0 PCMU/8000SS.Sent:INVITE sip:2000@9.44.44.4:5060 SIP/2.0SSS.User-Agent: Cisco CUCM8.5/IOS-15.1-3SSS.Diversion: sip:4085266855@sip.abc.com ;privacy off;reason unconditional;screen yesSSS.m audio 32278 RTP/AVP 18 8 101a rtpmap:0 PCMU/8000SSS. 2010 Cisco and/or its affiliates. All rights reserved.Cisco Confidential45

Inbound and outbound Local, Long distance, International calls for G711 &G729 codecs Outbound calls to information and emergency services Caller ID and Calling Name Presentation Supplementary services like Call Hold, Resume, Call Forward & Transfer DTMF Tests Fax calls – T.38 and fallback to pass-through (if option available) 2011 Cisco and/or its affiliates. All rights reserved.46

CUBE# show call active voice brief121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 2040 pid:2 Answer 2000 activedur 00:00:14 tx:0/0 rx:0/0IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/a121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 2020 pid:1 Originate 1000 activedur 00:00:14 tx:0/0 rx:0/0IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: offmedia inactive detected:n media contrl rcvd:n/a timestamp:n/along duration call detected:n long duration call duration:n/a timestamp:n/aTelephony call-legs: 0SIP call-legs: 2H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 2CUBE# show voip rtp connectionsVoIP RTP active connections :No. CallIddstCallId 0.10.10.1020.20.20.20RemoteIP1.1.1.12.2.2.2Found 2 active RTP connections 2011 Cisco and/or its affiliates. All rights reserved.47

Is CUBE Active ?show cube statusCUBE-Version : 9.0SW-Version : 15.2.1T, Platform 2911HA-Type : noneLicensed-Capacity : 200debug voip ccapi inoutIs the call matchingright Dial-peers ?Are we sending theright SIP call to SP basedon their requirements ? 2011 Cisco and/or its affiliates. All rights reserved.Oct 26 18:59:01.146: //-1/66A6B1BF8013/CCAPIcc api call setup ind common:.Incoming Dial-peer 1, Progress Indication NULL(0), Calling IEPresent TRUE,.Outgoing Dial-peer 100, Params 0x26E8574, ProgressIndication NULL(0)debug ccsip messagesReceived:INVITE sip:912025552000@14.128.101.24:5060 SIP/2.0Date: Wed, 26 Oct 2011 18:59:01 GMTAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK,UPDATE, REFER, SUBSCRIBE, NOTIFYFrom: "Paul Hewson" sip:1500@10.88.156.166 ;tag 90d94d926ee4-45aa-9f18-2d09025c1ee4-27352390.48

Network Managem

VXML RSVP SRST Agent Cisco Unified Border Element Address Hiding H.323 and SIP interworking DTMF interworking SIP security Transcoding Unified CM Conferencing and . Re-purpose your existing Cisco Voice Gateway’s as Cisco’s Session Border Controller –