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OCS Direct SIP: Interoperability with IP-PBX

OCS Direct SIP: Interoperability with IP-PBX. Desmond Lee Principal Consultant BT Switzerland www.leedesmond.com. Agenda. Terminology Review Legacy PBX to VoIP UC Voice Components in OCS 2007 R2 Voice Deployment Scenarios Interoperability –Today and Beyond Direct SIP with IP-PBX Demo

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OCS Direct SIP: Interoperability with IP-PBX

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  1. OCS Direct SIP: Interoperability with IP-PBX Desmond Lee Principal Consultant BT Switzerland www.leedesmond.com

  2. Agenda • Terminology Review • Legacy PBX to VoIP • UC Voice Components in OCS 2007 R2 • Voice Deployment Scenarios • Interoperability –Today and Beyond • Direct SIP with IP-PBX • Demo • SIP Trunking • Q&A

  3. Terminology ReviewTelephone System • PBX: Private Branch Exchange • POTS: Plain Old Telephone Services • Switch: PBX • Node: specific PBX in a network • Trunk: interconnects PBX or gateway to other PBX system, gateway or PSTN

  4. Terminology ReviewTelephone System • IP-PBX: IP based PBX • Hybrid: IP-PBX supporting VoIP & analog (TDM) • Gateway: connects and translates between different network types • DTMF: tone generated from touchtone phone that is transported in RTP stream by default • PSTN: Public Switched Telephone Network

  5. Terminology ReviewTelephony • Digital Voice Circuits • ISDN Basic Rate Interface (BRI) • 2(B)*64kbps + 1(D)*64kbps channels, 128kbps • ISDN Primary Rate Interface (PRI) • T1: 24(B)*64kbps + 1(D)*64kbps channels, 1.544 Mbps (USA) • E1: 30(B)*64kbps + 1(D)*64kbps channels, 2.048 Mbps (Europe) • Signaling • Channel Associated Signaling (CAS): takes place within the voice channel itself • Common Channel Signaling (CCS): out-of-band, separate dedicated channel

  6. Terminology ReviewSignaling Protocols • SS7: used in PSTN to connect central offices (CO) • Integrated Services Digital Network (ISDN) • QSIG: ISDN-based signaling protocol used to connect different PBXs from multi-vendors • Cisco’s Skinny Client Control Protocol (SCCP) • Media Gateway Control Protocol (MGCP) • H.323: ITU H.32x standard protocol suite (H.225, H.245) • SIP: Session Initiated Protocol(IETF Multi-party Multimedia Session Control) MGCP = RFC 2705, 3660, 3435, 3661SIP = RFC 2543, 3261, 3665

  7. Terminology ReviewAudio Codecs • G.711: ITU standard voice codec 64kbps • a-law in Europe and ROTW • mu-law in North America and Japan • G.729: compresses voice stream down to 8kbps • Internet Low Bit Rate Codec: enablesgradual voice quality degradation (iLBC) • RTAudio: Microsoft’s dynamic codec • Other ITU G-Series audio codecs: G.726, G.728, G.723, GSM Full Rate Codec (GSMFRC) variable bit rate codecs G.711 = PCM analog scheme at 8KHz sample rate with 8 bits per sample

  8. Terminology ReviewMedia Transmission Protocols • Real-time Transport Protocol (RTP) • defines a standardized packet format to deliver audio and video over data network directly between endpoints • no defined standard TCP or UDP port to communicate • RTP Control Protocol (RTCP) • primary function is to report back on the QoS provided by RTP e.g. lost packets, jitter, latency, etc. • also delivers control information for individual RTP streams RTP and RTCP were built on top of UDP. Both are described in IETF RFC1889 and 3550. In a Cisco environment, UDP ports in the 16,384 to 32,767 range are used (RTP odd, RTCP even).

  9. Terminology ReviewMedia Transmission Protocols • Compressed Real-time Transport Protocol (cRTP) • suppresses sending of redundant header information in every packet in a VoIP stream (“compression”) • reduces overhead for RTP traffic = reduces delay • Secure Real-time Transport Protocol (sRTP) • provides encryption, message authentication and integrity, and replay protection to RTP • likewise, Secure RTCP (sRTCP) protects RTCP cRTP = RFC 2508, 2509 and 3545 sRTP = RFC 3711

  10. Overview of PBXsLegacy PBX to VoIP TDM PBX Hybrid PBX IP PBX User workspace User workspace User workspace IP Phone x99999 IP Phone x99999 PBX phone x99999 PC PC PC IP IP IP IP PBX TDM PBX HybridTDM PBX IP +1 425 70xxxxx +1 425 70xxxxx +1 425 70xxxxx PSTN PSTN PSTN

  11. UC Voice Components UC endpoints QoE Monitoring Archiving CDR Network Perimeter Data Audio/Video Inbound Routing Outbound Routing SIP Remote Users Voice Mail Routing Active Directory Front-End Server(s) (IM, Presence) Conferencing Server(s) Backend SQL server Access Server Exchange Server 2007 UM Mediation Server Federated Businesses (SIP-PSTN GW) Voicemail PRI PSTN PBX

  12. UC Open Interoperability Program • Microsoft Unified Communications Open Interoperability Program (OIP) for enterprise telephony infrastructure • Program to qualify 3rd party SIP-PSTN gateways, IP-PBXs and SIP Trunking services for interoperability with OCS 2007 R2http://technet.microsoft.com/en-us/office/bb735838.aspx

  13. Voice Deployment Scenarios Slide Objective: Quickly review OCS Dial Plan concepts and components Available & Supported Consult TechNet site for the latest info:http://technet.microsoft.com/en-us/office/bb735838.aspx

  14. Back-to-back IP/PSTN Gateway PSTN OCS 2007 R2 End-Points G.711/TCP QSIG(media) G.711/TCP RTAudio/TLS RTAudio/TLS PSTN Signaling PSTN Media OCS 2007 R2 SIP/TLS Inbound Routing Mediation Server Existing PBX Or IP-PBX Outbound Routing PSTN/SIP Gateway SIP/PSTN Gateway Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR SIP/H.323 QSIG(signal) SIP/TCP SIP/TLS Exchange Server 2007 SP1 Unified Messaging

  15. Media GatewaysPBX Connectivity • Connect VoIP and PSTN or PBX • Translate TDM (circuit-switched based) protocols such as QSIG into packet-based protocols used in VoIP (such as SIP) • Types of Media Gateway • Basic • Hybrid (Collocated) • Works in conjunction with Mediation server

  16. Basic GW Appliance Rich GW appliance hosting RTC (compatible) Media Server UC Mediation Server Media GatewaysConfigurations • Basic Media Gateway • Separate MGWappliance and MediationServer roles • TCP to TLS, G.711 to RTAudio • Apply SRTP to media on UC side • Hybrid Media Gateway • MGW appliance runningMediation Server • UC Mediation Serverruns Windows Server2003 SP1 • Native support: SIP over TLS,SRTP, RTAudio

  17. Mediation ServerFunctionality • Connects OCS 2007 and SIP/PSTN Gateway or IP-PBX to provide IP telephony capability • Translates SIP/TCP (gateway) to SIP/MTLS (OCS) • Encodes/decodes RTP (gateway) to SRTP (OCS) • Transcoding of media from G.711 (gateway) to RTAudio and SIREN • 1:1 ratio between Mediation Server and Media Gateway

  18. Interoperability Issue • Traditional PBX phone systems and commonly deployed IP-PBX do not understand or are not designed to process the plus sign • Not all so-called SIP solutions are Standard SIP • 3rd party IP-PBX or SIP/PSTN solutions do not qualify for Direct SIP interoperability with OCS in OIP primarily due to lack of RFC3966 standard compliance

  19. E.164 Numbering Plan • ITU Recommendation • Universally accepted,globally routableunique number • Example:412212345673316986123412039876543 http://www.itu.int/rec/T-REC-E.164/en

  20. RFC 3966 • Defines the tel: URI and was created to enable numbering in the new world of SIP • Encompasses E.164 covering both public and private numbering plan (phone-context) • The plus + prefix is mandatory for global numbers to substitute the international dialing prefix • All SIP compliant IP-PBX should conform to the RFC 3966 standard http://www.ietf.org/rfc/rfc3966.txt

  21. Direct SIP • Enables OCS 2007 to communicate directly with qualified OIP IP/PBX and SIP/PSTN devices • An intermediary device in the form of a separate Media Gateway is not required • Both ends of the SIP trunk converse using standard protocols like SIP over TCP, G.711 and RTP • Does not require changes or an upgrade of existing non-RFC3966 conforming IP/PBX

  22. Direct SIP PSTN OCS 2007 R2 End-Points G.711/TCP RTAudio/TLS RTAudio/TLS PSTN Signaling PSTN Media OCS 2007 R2 SIP/TLS Inbound Routing Mediation Server Existing PBX Or IP-PBX Outbound Routing Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR SIP/TCP SIP/TLS Exchange Server 2007 SP1 Unified Messaging

  23. Direct SIP with IP-PBXSpecific versions tested or supported • Microsoft adapted R2 to support Direct SIP interop with IP-PBX, starting with CCM/CUCM* • OCS R2 now supported in Direct SIP interoperability with CUCM (back ported to OCS 2007 RTM) * extend to more IP-PBX planned

  24. Direct SIP with IP-PBXSpecific versions tested or supported • Versions tested and supported by Microsoft: • Versions successfully tested by customers: • Other IP-PBX are being tested by customers and/or partners

  25. Dial Plan – OCSQuick Review

  26. Cisco TerminologyQuick Review * Based on organization, location and call type

  27. Phone Number NormalizationExamples http://www.leedesmond.com/weblog/?p=507

  28. demo Direct SIP with Cisco Unified Call Manager 5

  29. demo Step 1: Create a Partition

  30. demo Step 2: Create a Calling Search Space

  31. demo Step 3: Create Translation Patterns for a Partition(inbound from OCS to CCUM)

  32. Direct SIP with IP-PBXOutbound OCS to International PSTN call (TP#1) PSTN.fr OCS 2007 R2 End-Points To: +14255551212 From: +33169864567 From: 33169864567 To: 14255551212 Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX. 4567 OCS 2007 R2 From: 169864567 To: 00014255551212 Inbound Routing Mediation Server Outbound Routing Existing PBX Or IP-PBX Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR Translation Pattern : [^33]! Prefix Digits (outgoing calls) : 000 Called Party Transform Mask : Discard Digits : <None> Calling Party Transform Mask* : XXXXXXXXX * applies to FROM field

  33. Direct SIP with IP-PBXOutbound OCS to National PSTN call (TP#2) PSTN.fr OCS 2007 R2 End-Points To: +33155551111 From: +33169864567 From: 33169864567 To: 33155551111 Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX. 4567 OCS 2007 R2 From: 169864567 To: 00155551111 Inbound Routing Mediation Server Outbound Routing Existing PBX Or IP-PBX Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR Translation Pattern : 33.XXXXXXXXX Prefix Digits (outgoing calls) : 00 Called Party Transform Mask : Discard Digits : PreDot Calling Party Transform Mask* : XXXXXXXXX * applies to FROM field

  34. Direct SIP with IP-PBXOutbound OCS to internal IP-PBX call (TP#3) PSTN.fr OCS 2007 R2 End-Points From: 33169864567 To: 33169861234 Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX. 4567 OCS 2007 R2 From: 4567 To: 1234 Inbound Routing Mediation Server Outbound Routing Existing PBX Or IP-PBX Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR Translation Pattern : 3316986XXXX Prefix Digits (outgoing calls) : Called Party Transform Mask : XXXX Discard Digits : <None> Calling Party Transform Mask* : XXXX * applies to FROM field 1234

  35. demo Step 4: Provision a SIP trunk

  36. demo Step 5: Setup a Route Pattern (outbound CUCM to OCS)

  37. Direct SIP with IP-PBXOutbound IP-PBX to internal OCS call (RP#1) PSTN.fr OCS 2007 R2 End-Points From: +33169861234 To: +33169864567 Normalization rules to insert + signand manipulate digits. 4567 OCS 2007 R2 From: 1234 To: 4567 Inbound Routing Mediation Server Outbound Routing Existing PBX Or IP-PBX Voicemail Routing IM, Presence, Audio, Video, Conferencing, IVR Route Pattern** : [4-5]XXX Gateway or Route List** : Trunk_to_OCS (SIP Trunk) Called Party Transform Mask** : Calling Party Transform Mask : ** Outbound calls (TO field) 1234

  38. demo Step 6: Configure OCS for Direct SIP

  39. Direct SIP with IP-PBXUpdate Packages OCS 2007/MOC* * OCS 2007 (RTM 6362.0) - KB 952783, 952780, 953659, 957707

  40. Direct SIP with IP-PBXModifications on Mediation Server • Create %programfiles%\Microsoft Office Communications Server 2007\Mediation Server\MediationServerSvc.exe.config if not exist • Set RemovePlusFromRequestURI to Yes and restart machine • For R2, modify the WMI setting (default No) RemovePlusFromRequestURItoYes

  41. Direct SIP with IP-PBXStep-by-Step Summary CUCM • Step 1: Create a Partition • Step 2: Create a Calling Search Space • Step 3: Create Translation Patterns for a Partition (inbound from OCS to CUCM) • Step 4: Provision a SIP Trunk • Step 5: Setup a Route Pattern (outbound CUCM to OCS) • Step 6: Configure OCS for Direct SIP

  42. SIP Trunking • Routes speech using VoIP technology over the IP backbone of a worldwide, enterprise-class carrier • Eliminates investment (and maintenance) in costly legacy, PBX switches or TDM-based voice circuits that are often limited in scalability • Key components • IP-PBX or PBX withinterface for SIP connectivity • ITSP or SIP Trunk Provider toconnect to PSTN (mobile, analog devices, etc.) ITSP = Internet Telephone Service Provider

  43. SIP Trunking • BT Partnership with Microsoft in the global TAP Program (BPOS)* • BT OneVoice – global voice platform anchored on strong heritage of voice services (in/out bound) • Planned availability 2009/2010 * Business Productivity Online Services on Microsoft Hosted services platform; one of only two worldwide enterprise partners

  44. Save the date for tech·days nextyear! 14 – 15 avril 2010, CICG

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  46. OCS Direct SIP: Interoperability with IP-PBX Desmond.Lee@swissitpro.com Principal Consultant BT Switzerland www.leedesmond.com

  47. Cisco TerminologyTelephony • Media Termination Point (MTP) • bridges 2 voice streams using the same codec or different packetization periods • enables both to be separately setup and torn down • transcodes a-law to mu-law (vice-versa) • On-net calls • both endpoints communicate on same data network • Off-net calls • phone – VoIP router or PBX via Foreign Exchange Office or T1/E1 – PSTN – phone

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