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
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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