Desmond Lee Principal Consultant BT Switzerland .

OCS Direct SIP: Interoperability with IP-PBX

Desmond LeePrincipal ConsultantBT Switzerland

www.leedesmond.com

Agenda

Terminology ReviewLegacy PBX to VoIPUC Voice Components in OCS 2007 R2Voice Deployment ScenariosInteroperability –Today and BeyondDirect SIP with IP-PBXDemoSIP TrunkingQ&A

Terminology ReviewTelephone System

PBX: Private Branch ExchangePOTS: Plain Old Telephone ServicesSwitch: PBXNode: specific PBX in a networkTrunk: interconnects PBX or gateway to other PBX system, gateway or PSTN

Terminology ReviewTelephone System

IP-PBX: IP based PBXHybrid: IP-PBX supporting VoIP & analog (TDM)Gateway: connects and translates between different network typesDTMF: tone generated from touchtone phone that is transported in RTP stream by default

PSTN: Public Switched Telephone Network

Terminology ReviewTelephony

Digital Voice CircuitsISDN 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)

SignalingChannel Associated Signaling (CAS): takes place within the voice channel itselfCommon Channel Signaling (CCS): out-of-band, separate dedicated channel

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-vendorsCisco’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, 3661

SIP = RFC 2543, 3261, 3665

Terminology ReviewAudio Codecs

G.711: ITU standard voice codec 64kbpsa-law in Europe and ROTWmu-law in North America and Japan

G.729: compresses voice stream down to 8kbpsInternet Low Bit Rate Codec: enablesgradual voice quality degradation (iLBC)

RTAudio: Microsoft’s dynamic codecOther 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

Terminology ReviewMedia Transmission Protocols

Real-time Transport Protocol (RTP)

defines a standardized packet format to deliver audio and video over data network directly between endpointsno 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).

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 RTPlikewise, Secure RTCP (sRTCP) protects RTCP

cRTP = RFC 2508, 2509 and 3545sRTP = RFC 3711

TDM PBX

PSTN

User workspace

PBX phonex99999 PC

+1 425 70xxxxx

IP

TDM PBX

IP PBX

User workspace

IP Phonex99999 PC

+1 425 70xxxxx

PSTN

IP

IP PBX

IP

Hybrid PBX

User workspace

IP Phonex99999 PC

+1 425 70xxxxx

PSTN

HybridTDM PBX

IP

Overview of PBXsLegacy PBX to VoIP

UC Voice ComponentsQoE

Monitoring Archiving

CDR

RemoteUsers

Network Perimete

r

FederatedBusinesses

Front-End Server(s)

(IM, Presence)

InboundRouting

OutboundRouting

PSTN

BackendSQL

server

ExchangeServer 2007 UM

Voicemail

UC endpoints

Active Directory

Voice MailRouting

Conferencing

Server(s)

PBX

(SIP-PSTN GW)

AccessServer

DataAudio/Video

SIP

Mediation Server

PRI

UC Open Interoperability Program

Microsoft Unified Communications Open Interoperability Program (OIP) for enterprise telephony infrastructureProgram 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

Voice Deployment Scenarios

Slide Objective: Quickly review OCS Dial Plan concepts and components

Standalone Co-Existence

Gateway Direct SIP Dual Forking Dual Forkingwith RCC

Available & Supported

Consult TechNet site for the latest info:http://technet.microsoft.com/en-us/office/bb735838.aspx

Back-to-back IP/PSTN Gateway

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

Unified Messaging

Exchange Server2007 SP1

PSTN

PSTN/SIPGateway

SIP/PSTNGateway

QSIG(signal)

SIP/TLS

SIP/TCP

SIP/H.323

PS

TN

S

ign

alin

g

QSIG(media)

RTAudio/TLS

G.711/TCP

G.711/TCP

SIP/TLS PS

TN

Med

ia

RTAudio/TLS

Connect VoIP and PSTN or PBXTranslate TDM (circuit-switched based) protocols such as QSIG into packet-based protocols used in VoIP (such as SIP)Types of Media Gateway

BasicHybrid (Collocated)

Works in conjunction with Mediation server

Media GatewaysPBX Connectivity

Basic Media GatewaySeparate MGWappliance and MediationServer rolesTCP to TLS, G.711 to RTAudioApply SRTP to media on UC side

Hybrid Media GatewayMGW appliance runningMediation ServerUC Mediation Serverruns Windows Server2003 SP1Native support: SIP over TLS,SRTP, RTAudio

Media GatewaysConfigurations

UC Mediation ServerBasic GW Appliance

Rich GW appliancehosting RTC (compatible)

Media Server

Connects OCS 2007 and SIP/PSTN Gateway or IP-PBX to provide IP telephony capabilityTranslates SIP/TCP (gateway) to SIP/MTLS (OCS)Encodes/decodes RTP (gateway) to SRTP (OCS)Transcoding of media from G.711 (gateway) to RTAudio and SIREN1:1 ratio between Mediation Server and Media Gateway

Mediation ServerFunctionality

Traditional PBX phone systems and commonly deployed IP-PBX do not understand or are not designed to process the plus signNot all so-called SIP solutions are Standard SIP3rd 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

Interoperability Issue

ITU RecommendationUniversally accepted,globally routableunique numberExample:412212345673316986123412039876543

E.164 Numbering Plan

http://www.itu.int/rec/T-REC-E.164/en

Defines the tel: URI and was created to enable numbering in the new world of SIPEncompasses E.164 covering both public and private numbering plan (phone-context)

The plus + prefix is mandatory for global numbers to substitute the international dialing prefixAll SIP compliant IP-PBX should conform to the RFC 3966 standard

RFC 3966

http://www.ietf.org/rfc/rfc3966.txt

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

Direct SIP

Direct SIP

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

Unified Messaging

Exchange Server2007 SP1

PSTN

SIP/TCP

SIP/TLS

PS

TN

S

ign

alin

g

G.711/TCP RTAudio/

TLS

SIP/TLSPS

TN

Med

ia

RTAudio/TLS

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)

Direct SIP with IP-PBXSpecific versions tested or supported

* extend to more IP-PBX planned

Versions tested and supported by Microsoft:

Versions successfully tested by customers:

Other IP-PBX are being tested by customers and/or partners

Direct SIP with IP-PBXSpecific versions tested or supported

IP-PBX Vendor Product Versions testedCisco CUCM 6.1 6.1.1.3000-2Cisco CUCM 5.1 5.1.3.1000-12

5.1.3.3000-5Cisco CUCM 4.2 4.2(3)SR3a

IP-PBX Vendor Product Versions tested

Cisco CUCM 4.2 4.2(1)

Cisco CUCM 4.1 4.1(3)SR7

Dial Plan – OCSQuick Review

•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 usersVoice Policies

•Routing logic for calls to PBX and PSTNRoutes

Cisco TerminologyQuick Review

• 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

Phone Number NormalizationExamples

http://www.leedesmond.com/weblog/?p=507

Direct SIP with Cisco Unified Call Manager 5

demo

Step 1: Create a Partition

demo

Step 2: Create a Calling Search Space

demo

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

demo

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

PSTN.fr

To: +14255551212From: +33169864567

From: 169864567To: 00014255551212

4567

Direct SIP with IP-PBXOutbound OCS to International PSTN call (TP#1)

Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX.

From: 33169864567To: 14255551212

Translation Pattern : [^33]!Prefix Digits (outgoing calls) : 000Called Party Transform Mask :Discard Digits : <None>

Calling Party Transform Mask* : XXXXXXXXX

* applies to FROM field

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

PSTN.fr

To: +33155551111From: +33169864567

From: 169864567To: 00155551111

4567

Direct SIP with IP-PBXOutbound OCS to National PSTN call (TP#2)

Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX.

From: 33169864567To: 33155551111

Translation Pattern : 33.XXXXXXXXXPrefix Digits (outgoing calls) : 00Called Party Transform Mask :Discard Digits : PreDot

Calling Party Transform Mask* : XXXXXXXXX

* applies to FROM field

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

PSTN.fr

1234

From: 4567To: 1234

4567

Direct SIP with IP-PBXOutbound OCS to internal IP-PBX call (TP#3)

Strips + sign and presents dialstring in a format that can beinterpreted by IP-PBX.

From: 33169864567To: 33169861234

Translation Pattern : 3316986XXXXPrefix Digits (outgoing calls) : Called Party Transform Mask : XXXXDiscard Digits : <None>

Calling Party Transform Mask* : XXXX

* applies to FROM field

Step 4: Provision a SIP trunk

demo

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

demo

OCS 2007 R2

IM, Presence,Audio, Video, Conferencing, IVR

Inbound Routing

Outbound Routing

Voicemail Routing

OCS 2007 R2 End-Points

MediationServer

Existing PBXOr

IP-PBX

PSTN.fr

1234

From: 1234To: 4567

4567

Direct SIP with IP-PBXOutbound IP-PBX to internal OCS call (RP#1)

Normalization rules to insert + signand manipulate digits.

From: +33169861234To: +33169864567

Route Pattern** : [4-5]XXXGateway or Route List** : Trunk_to_OCS(SIP Trunk)Called Party Transform Mask** :

Calling Party Transform Mask :

** Outbound calls (TO field)

Step 6: Configure OCS for Direct SIP

demo

Direct SIP with IP-PBXUpdate Packages OCS 2007/MOC*

Server Roles \ Patch Name

Server.msp

MediationServer.msp

UCMARedist.msp

Communicator.msp

Standard Edition Server(Unique Front-end pool)

X

Enterprise Edition Server(Front-end)

X

Proxy Server & ForwardingProxy Server

X

Director Server XEdge Server(Access, A/V, WebConferencing)

X

Mediation Server X XMOC X* OCS 2007 (RTM 6362.0) - KB 952783, 952780, 953659, 957707

Create %programfiles%\Microsoft Office Communications Server 2007\Mediation Server\MediationServerSvc.exe.config if not existSet RemovePlusFromRequestURI to Yes and restart machineFor R2, modify the WMI setting (default No) RemovePlusFromRequestURI toYes

Direct SIP with IP-PBXModifications on Mediation Server

Step 1: Create a PartitionStep 2: Create a Calling Search SpaceStep 3: Create Translation Patterns for a Partition (inbound from OCS to CUCM)

Step 4: Provision a SIP TrunkStep 5: Setup a Route Pattern (outbound CUCM to OCS)

Step 6: Configure OCS for Direct SIP

Direct SIP with IP-PBXStep-by-Step Summary

CUCM

SIP Trunking

Routes speech using VoIP technology over the IP backbone of a worldwide, enterprise-class carrierEliminates investment (and maintenance) in costly legacy, PBX switches or TDM-based voice circuits that are often limited in scalabilityKey components

IP-PBX or PBX withinterface for SIP connectivityITSP or SIP Trunk Provider toconnect to PSTN (mobile, analog devices, etc.)

ITSP = Internet Telephone Service Provider

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

Save the date for tech·days next year!

14 – 15 avril 2010, CICG

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OCS Direct SIP: Interoperability with IP-PBX

[email protected] ConsultantBT Switzerland

www.leedesmond.com

Cisco TerminologyTelephony

Media Termination Point (MTP)

bridges 2 voice streams using the same codec or different packetization periodsenables both to be separately setup and torn downtranscodes a-law to mu-law (vice-versa)

On-net callsboth endpoints communicate on same data network

Off-net callsphone – VoIP router or PBX via Foreign Exchange Office or T1/E1 – PSTN – phone