Transcript Title of Presentation

Desmond Lee
Principal Consultant
BT Switzerland
www.leedesmond.com
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
Telephone 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
Telephone 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
Telephony
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
Signaling 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, 3661
SIP = RFC 2543, 3261, 3665
Audio 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: enables
bit
gradual voice quality degradation (iLBC) variable
rate codecs
RTAudio: Microsoft’s dynamic codec
Other ITU G-Series audio codecs: G.726, G.728,
G.723, GSM Full Rate Codec (GSMFRC)
G.711 = PCM analog scheme at 8KHz sample rate with 8 bits per sample
Media 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).
Media 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
Legacy PBX to VoIP
TDM PBX
User workspace
PBX phone
x99999
PC
Hybrid PBX
User workspace
IP Phone
x99999
PC
IP PBX
User workspace
IP Phone
x99999
PC
IP
IP
IP
IP PBX
TDM PBX
Hybrid
TDM PBX
+1 425 70xxxxx
+1 425 70xxxxx
+1 425 70xxxxx
PSTN
IP
PSTN
PSTN
UC endpoints
QoE
Monitoring
Archiving
CDR
Network
Perimeter
Data
Audio/
Video
SIP
Inbound
Routing
Outbound
Routing
Remote
Users
Voice Mail
Routing
Access
Server
Front-End Server(s)
(IM, Presence)
Conferencing
Server(s)
Exchange
Server 2007 UM
Mediation Server
Federated
Businesses
(SIP-PSTN GW)
PSTN
PRI
Backend
SQL server
Voicemail
PBX
Active
Directory
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
http://technet.microsoft.com/en-us/office/bb735838.aspx
Standalone
Gateway
Co-Existence
Direct SIP
Dual Forking
Dual Forking
with RCC
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
OCS 2007 R2 End-Points
PSTN Media
PSTN Signaling
PSTN
G.711/
TCP
QSIG
(media)
G.711/
TCP
RTAudio/
TLS
RTAudio/
TLS
OCS 2007 R2
Mediation
Server
SIP/TLS
Inbound Routing
Existing PBX
Or
IP-PBX
Outbound Routing
SIP/PSTN
Gateway
SIP/
H.323
Voicemail Routing
PSTN/SIP
Gateway
QSIG
(signal)
IM, Presence,
Audio, Video, Conferencing, IVR
SIP/
TCP
SIP/
TLS
Exchange Server
2007 SP1
Unified
Messaging
PBX 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
Configurations
Basic Media Gateway
Separate MGW
Basic GW
appliance and Mediation
Appliance
Server roles
TCP to TLS, G.711 to RTAudio
Apply SRTP to media on UC side
UC Mediation Server
Hybrid Media Gateway
MGW appliance running
Mediation Server
Rich GW appliance
hosting RTC (compatible)
UC Mediation Server
Media Server
runs Windows Server
2003 SP1
Native support: SIP over TLS,
SRTP, RTAudio
Functionality
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
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
ITU Recommendation
Universally accepted,
globally routable
unique number
Example:
41221234567
33169861234
12039876543
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 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
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
OCS 2007 R2 End-Points
PSTN Media
PSTN Signaling
PSTN
G.711/
TCP
RTAudio/
TLS
RTAudio/
TLS
OCS 2007 R2
Mediation
Server
SIP/TLS
Inbound Routing
Existing PBX
Or
IP-PBX
Outbound Routing
Voicemail Routing
SIP/
TCP
IM, Presence,
Audio, Video, Conferencing, IVR
SIP/
TLS
Exchange Server
2007 SP1
Unified
Messaging
Specific 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
Specific versions tested or supported
Versions tested and supported by Microsoft:
IP-PBX Vendor
Cisco
Cisco
Product
CUCM 6.1
CUCM 5.1
Cisco
CUCM 4.2
Versions tested
6.1.1.3000-2
5.1.3.1000-12
5.1.3.3000-5
4.2(3)SR3a
Versions successfully tested by customers:
IP-PBX Vendor
Product
Versions tested
Cisco
CUCM 4.2
4.2(1)
Cisco
CUCM 4.1
4.1(3)SR7
Other IP-PBX are being tested by customers
and/or partners
Quick Review
Normalization
Rules
• Convert numbers in various formats to
standard E.164 format
Location
Profiles
• Set of normalization rules that applies to
a particular location
Phone Usage
Records
• Call permissions and restrictions – used
in both Policies and Routing
Voice Policies
• Collections of phone usage records that
are assigned to one or more users
Routes
• Routing logic for calls to PBX and PSTN
Quick Review
Partition
Calling Search
Space
• Facilitates call routing by dividing
route plans into logical subnets
(applies route & translation patterns)*
• An ordered list of route partitions that
will be searched to complete a call.
Translation
Patterns
• Manipulate dial strings prior to
routing the call. Used for inbound
calls to CUCM (from OCS).
Routes
• Routing logic for calls to PBX and
PSTN (outbound traffic).
* Based on organization, location and call type
Examples
http://www.leedesmond.com/weblog/?p=507
Direct SIP with Cisco Unified
Call Manager 5
Step 1: Create a Partition
Step 2: Create a Calling
Search Space
Step 3: Create Translation
Patterns for a Partition
(inbound from OCS to CCUM)
Outbound OCS to International PSTN call (TP#1)
PSTN
.fr
To:
+14255551212
From: +33169864567
OCS 2007 R2 End-Points
From: 33169864567
To:
14255551212
Strips + sign and presents dial
string in a format that can be
interpreted by IP-PBX.
From:
169864567
To: 00014255551212
Mediation
Server
Existing PBX
Or
IP-PBX
4567
OCS 2007 R2
Inbound Routing
Outbound Routing
Voicemail Routing
IM, Presence,
Audio, Video, Conferencing, IVR
Translation Pattern
Prefix Digits (outgoing calls)
Called Party Transform Mask
Discard Digits
: [^33]!
: 000
:
: <None>
Calling Party Transform Mask* : XXXXXXXXX
* applies to FROM field
Outbound OCS to National PSTN call (TP#2)
PSTN
.fr
To:
+33155551111
From: +33169864567
OCS 2007 R2 End-Points
From: 33169864567
To:
33155551111
Strips + sign and presents dial
string in a format that can be
interpreted by IP-PBX.
From:
169864567
To:
00155551111
Mediation
Server
Existing PBX
Or
IP-PBX
4567
OCS 2007 R2
Inbound Routing
Outbound Routing
Voicemail Routing
IM, Presence,
Audio, Video, Conferencing, IVR
Translation Pattern
Prefix Digits (outgoing calls)
Called Party Transform Mask
Discard Digits
.
: 33 XXXXXXXXX
: 00
:
: PreDot
Calling Party Transform Mask* : XXXXXXXXX
* applies to FROM field
Outbound OCS to internal IP-PBX call (TP#3)
OCS 2007 R2 End-Points
PSTN
.fr
From: 33169864567
To:
33169861234
Strips + sign and presents dial
string in a format that can be
interpreted by IP-PBX.
From: 4567
To:
1234
Mediation
Server
Existing PBX
Or
IP-PBX
4567
OCS 2007 R2
Inbound Routing
Outbound Routing
Voicemail Routing
IM, Presence,
Audio, Video, Conferencing, IVR
Translation Pattern
Prefix Digits (outgoing calls)
Called Party Transform Mask
Discard Digits
1234
: 3316986XXXX
:
: XXXX
: <None>
Calling Party Transform Mask* : XXXX
* applies to FROM field
Step 4: Provision a SIP trunk
Step 5: Setup a Route Pattern
(outbound CUCM to OCS)
Outbound IP-PBX to internal OCS call (RP#1)
OCS 2007 R2 End-Points
PSTN
.fr
From: +33169861234
To:
+33169864567
Normalization rules to insert + sign
and manipulate digits.
4567
From: 1234
To:
4567
Mediation
Server
Existing PBX
Or
IP-PBX
OCS 2007 R2
Inbound Routing
Outbound Routing
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** :
1234
Calling Party Transform Mask
** Outbound calls (TO field)
:
Step 6: Configure OCS for
Direct SIP
Update Packages OCS 2007/MOC*
Server Roles \ Patch Name
Standard Edition Server
(Unique Front-end pool)
Enterprise Edition Server
(Front-end)
Proxy Server & Forwarding
Proxy Server
Director Server
Edge Server
(Access, A/V, Web
Conferencing)
Mediation Server
MOC
Server.
msp
Mediation
Server.msp
UCMARedist
.msp
Communicator
.msp
X
X
X
X
X
X
* OCS 2007 (RTM 6362.0) - KB 952783, 952780, 953659, 957707
X
X
Modifications 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)
RemovePlusFromRequestURI toYes
Step-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
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
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
14 – 15 avril 2010, CICG
Premium Sponsoring Partners
Classic Sponsoring Partners
[email protected]
Principal Consultant
BT Switzerland
www.leedesmond.com
Telephony
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