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Desmond Lee Principal Consultant BT Switzerland www.leedesmond.com.

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Presentation on theme: "Desmond Lee Principal Consultant BT Switzerland www.leedesmond.com."— Presentation transcript:

1 Desmond Lee Principal Consultant BT Switzerland

2 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 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 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 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, Mbps (USA) E1: 30(B)*64kbps + 1(D)*64kbps channels, 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 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 Ciscos 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, 3661 SIP = RFC 2543, 3261, 3665

7 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: enables gradual voice quality degradation (iLBC) RTAudio: Microsofts 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 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 In a Cisco environment, UDP ports in the 16,384 to 32,767 range are used (RTP odd, RTCP even).

9 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 TDM PBX PSTN User workspace PBX phone x99999 PC xxxxx IP TDM PBX IP PBX User workspace IP Phone x99999 PC xxxxx PSTN IP IP PBX IP Hybrid PBX User workspace IP Phone x99999 PC xxxxx PSTN Hybrid TDM PBX IP

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

12 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 R2

13 Slide Objective: Quickly review OCS Dial Plan concepts and components StandaloneCo-Existence GatewayDirect SIPDual ForkingDual Forking with RCC Available & Supported Consult TechNet site for the latest info:

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

15 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 Media Gateway Separate MGW appliance and Mediation Server roles TCP to TLS, G.711 to RTAudio Apply SRTP to media on UC side Hybrid Media Gateway MGW appliance running Mediation Server UC Mediation Server runs Windows Server 2003 SP1 Native support: SIP over TLS, SRTP, RTAudio UC Mediation Server Basic GW Appliance Rich GW appliance hosting RTC (compatible) Media Server

17 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 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 3 rd 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 ITU Recommendation Universally accepted, globally routable unique number Example:

20 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

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

23 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 Versions tested and supported by Microsoft: Versions successfully tested by customers: Other IP-PBX are being tested by customers and/or partners

25 Convert numbers in various formats to standard E.164 format Normalization Rules Set of normalization rules that applies to a particular location Location Profiles Call permissions and restrictions – used in both Policies and Routing Phone Usage Records Collections of phone usage records that are assigned to one or more users Voice Policies Routing logic for calls to PBX and PSTN Routes

26 Facilitates call routing by dividing route plans into logical subnets (applies route & translation patterns) * Partition An ordered list of route partitions that will be searched to complete a call. Calling Search Space Manipulate dial strings prior to routing the call. Used for inbound calls to CUCM (from OCS). Translation Patterns Routing logic for calls to PBX and PSTN (outbound traffic). Routes * Based on organization, location and call type

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28 Direct SIP with Cisco Unified Call Manager 5

29 Step 1: Create a Partition

30 Step 2: Create a Calling Search Space

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

32 OCS 2007 R2 IM, Presence, Audio, Video, Conferencing, IVR Inbound Routing Outbound Routing Voic Routing OCS 2007 R2 End-Points Mediation Server Mediation Server Existing PBX Or IP-PBX Existing PBX Or IP-PBX PSTN.fr To: From: From: To: Strips + sign and presents dial string in a format that can be interpreted by IP-PBX. From: To: Translation Pattern : [^33]! Prefix Digits (outgoing calls) : 000 Called Party Transform Mask : Discard Digits : Calling Party Transform Mask* : XXXXXXXXX * applies to FROM field

33 OCS 2007 R2 IM, Presence, Audio, Video, Conferencing, IVR Inbound Routing Outbound Routing Voic Routing OCS 2007 R2 End-Points Mediation Server Mediation Server Existing PBX Or IP-PBX Existing PBX Or IP-PBX PSTN.fr To: From: From: To: Strips + sign and presents dial string in a format that can be interpreted by IP-PBX. From: To: 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 OCS 2007 R2 IM, Presence, Audio, Video, Conferencing, IVR Inbound Routing Outbound Routing Voic Routing OCS 2007 R2 End-Points Mediation Server Mediation Server Existing PBX Or IP-PBX Existing PBX Or IP-PBX PSTN.fr 1234 From: 4567 To: Strips + sign and presents dial string in a format that can be interpreted by IP-PBX. From: To: Translation Pattern : XXXX Prefix Digits (outgoing calls) : Called Party Transform Mask : XXXX Discard Digits : Calling Party Transform Mask* : XXXX * applies to FROM field

35 Step 4: Provision a SIP trunk

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

37 OCS 2007 R2 IM, Presence, Audio, Video, Conferencing, IVR Inbound Routing Outbound Routing Voic Routing OCS 2007 R2 End-Points Mediation Server Mediation Server Existing PBX Or IP-PBX Existing PBX Or IP-PBX PSTN.fr 1234 From: 1234 To: Normalization rules to insert + sign and manipulate digits. From: To: 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)

38 Step 6: Configure OCS for Direct SIP

39 * OCS 2007 (RTM ) - KB , , ,

40 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) RemovePlusFromRequestURI toYes

41 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 CUCM

42 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 with interface for SIP connectivity ITSP or SIP Trunk Provider to connect to PSTN (mobile, analog devices, etc.) ITSP = Internet Telephone Service Provider

43 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

45 14 – 15 avril 2010, CICG

46 Classic Sponsoring Partners Premium Sponsoring Partners

47

48 Principal Consultant BT Switzerland

49 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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