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OpenTouch SBC (Session Border Controller)

Design and Quotation Guidance R2.1.1 - MLE Offer H1-2016


Presales Presentation
Edition 03 May 2016
Presales Presentation | May 2016
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History
Edition 01: Creation
Edition 02: Update for product release 2.1.1
* support of WebRTC feature
* support of Transcoding feature
Edition 03: Update for MLE Offer H1-2016
* remove Server Edition for new configurations

Disclaimer
This documentation is provided for reference purposes only and does not fully describe the capabilities of each Product and related features. Therefore, ALE International
declines any liability for inaccuracies contained herein. For an exhaustive view on features list and product limits for the current product release please see the required
Feature List/Product Limits document available through the ALE eBusiness Portal web site.
In the interest of continued product development, ALE International reserves the right to make improvements or other changes to this document and the products it
describes at any time without prior notice.

Copyright
Copyright ALE International 2016. Distribution of substantively modified versions of this document is prohibited without the explicit permission of the copyright holder.
Distribution of the work or derivative of the work in any standard (paper) book form for commercial purposes is prohibited unless prior permission is obtained from the
copyright holder.

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Preamble
This presentation describes the OpenTouch Session Border Controller (SBC) product and
provides to the reader:
1) a high level view of the functional capabilities of the product
2) a description of the different use cases of the product in a global ALE
Communications solution
3) design guidance for the product
4) configuration/quotation guidance for the product
The product OT SBC is available in release 2.1.1 delivered in November 2015 and is part of
the OpenTouch Suite MLE Offer H1-2016

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Agenda
1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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Agenda
1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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COPYRIGHT 2016 ALE International. ALL RIGHTS RESERVED.

Key questions
What
It
It
It
It

is
is
is
is

is OpenTouch SBC ?

a Session Border Controller for ALE IP Communications solutions


the result of a partnership with Audiocodes company (product OEM)
a pure software solution, installed on a physical server or virtualized
used to address the usual concerns at the border of the corporate network:

Interoperability / Connectivity / QOS / Security

Where

is OpenTouch SBC used ?

It is used in some specific configurations with the OmniPCX Enterprise solution, the OpenTouch BE (Business
Edition) and OpenTouch MS (MultiMedia Services) solutions
It is used for a configuration based on the MLE Offer and also OT EC (Enterprise Cloud) offer (refer to dedicated
OT EC documentation for more details)

How

to make a quotation of OpenTouch SBC ?

It can be configured in the ACTIS tool

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OpenTouch SBC: an enabler for ALE solutions


Un-trusted areas
Trusted area

Carriers

Customer
Premises
OXE

DMZ
OTMS

VoIP Carriers
network

OT SBC

Public SIP Trunking

SIP Carrier
Gateway

with Carriers/SIP Providers

LAN
Firewall

OT SBC

Internet

Remote SIP user


with validated SIP clients

SRTP

Internet
access

Encryption 8082/8088 CV
users Peer to peer audio calls
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Use cases overview (1/3)


Public SIP Trunk
With OmniPCX Enterprise and OpenTouch BE (OTBE)

Interoperability with a Carrier network using Public SIP Trunk. The OpenTouch SBC should be part of the
configuration to be qualified. Refer to the specific PCS (Premium Customer Support) process description related to
Public SIP Trunk for more details (available in eBusiness Portal/Resources Center/Presales Documentations/PCS
Process Doc)

Premium Customer
Support

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Use cases overview (2/3)


Remote SIP Users
With OpenTouch BE and OpenTouch MS (OTMS) in release 2.2
- For an OpenTouch user with:
o

OTC for PC client with audio and video, and with encryption (SIP TLS&SRTP)

OTC for MAC OSx client with audio, and with encryption (SIP TLS&SRTP)

OTC for Tablet clients (Android and Apple) with audio and video (video only with Apple iPad), and with
encryption (SIP TLS&SRTP)

OTC for Smartphone clients (Android and Apple) with audio, and with encryption (SIP TLS&SRTP)

- For OpenTouch and external users with OTC Web using WebRTC API (with compatible web browser) for VoIP support

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Use cases overview (3/3)


Encryption for 8088/8082 phone sets
With OpenTouch BE (OTBE) and OpenTouch MS (OTMS) in release 2.2
- For peer to peer (IP phone to IP phone) and for audio only. Based on SIP TLS between the phone set and the OT SBC
and SRTP between the two phone sets (8088 or 80882). Video is not supported on the IP phone if encryption of the
voice is activated
- For an OTBE configuration this feature applies only for installed base where Conversation user profile was already
configured in a previous product release. It does not apply to new OTBE system in release 2.2 as the Conversation user
profile is deprecated for new configuration

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Cross Compatibility and Availability


OT SBC R2.1.x

Public SIP Trunk


*

Remote SIP user

Encryption
8088/8082 **

OmniPCX Enterprise
OmniPCX Enterprise from R11.0.x

OpenTouch BE/MS
OTBE / OTMS R1.x

OTBE / OTMS R2.x

(for OTMS the trunks are


managed by the OXE)

(for Conversation user profile only)

* Public SIP Trunk feature for OXE/OTMS and OTBE configuration is under PCS (Premium Customer Support)
** For peer to peer calls with audio only encryption (SIP TLS&SRTP) between two 8088/8082 sets for Conversation user profile only

Note: OT SBC release 2.1.x is not cross-compatible with former products OmniTouch ICS and ACS/Teamwork and cannot be used in a
configuration with these products. The products OmniTouch ICS and ACS/Teamwork are end of sale and the replacement solution is
OpenTouch Multimedia Services

- Refer to OT Suite MLE Offer Cross Compatibility document for detailed information Presales Presentation | May 2016
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AGENDA

1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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Use case 1a: Public SIP Trunk


IP VPN and SIP Trunk from the same provider
Branch
Office

OXE
Main Site

PSTN gateway

Priv@

Priv@

Priv@

VoIP Carriers network

Carrier A
NAT

IP VPN

Carrier SBC A
Softswitch A

Internet/
IP Managed

NAT

SIP Signaling
Private Telephony Signaling
Voice flows

In this configuration SBC on the customer side is in general not mandatory:


No issue with NAT/PNAT for VoIP: Managed by carrier
- Carrier SBC has an access in the customer network and IP@ routable in customer WAN

No security issue: Managed by carrier


- Protected by IP VPN from other customers. Carrier SBC is the unique external access
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Use case 1b: Public SIP Trunk


SIP Trunk over IP Managed or Internet
OXE
Main Site

Priv@

Branch
Office

Priv@
Carrier A or Customer
IP VPN

NAT

PSTN gateway

SBC
NAT5

Pub@
NAT

SIP Signaling
Private Telephony Signaling

Internet/
IP Managed

VoIP Carrier B network

Pub@

SIP Trunk

Carrier
SBC B

Voice flows

In this configuration SBC on the customer side is mandatory:


NAT/PNAT for VoIP: Managed on the customer side
-

A third-party IP network is used to connect enterprise to SIP Carrier B SBC (ISP)


SIP Carrier B provides 1 or 2 public addresses for the enterprise SIP proxy

Security issue: not managed by SIP Provider B


-

IP managed network is shared with other customers. Firewalling is necessary

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Softswitch B

Use case 1c: Public SIP Trunk


Multiple SIP Providers
OXE
Main Site

Priv@

Branch
Office

Priv@
Carrier A or Customer
IP VPN
NAT

SBC
NAT5
NAT

SIP Signaling
Private Telephony Signaling

Internet/
IP Managed

SIP Trunk 2

PSTN gateway
Pub@

SIP Trunk 1
Carrier
SBC B

Carrier
SBC C
VoIP Carrier C network

Voice flows

PSTN
gateway

Softswitch C

VoIP Carrier B network

Softswitch B

In a configuration with multiple SIP Trunk Providers a SBC is mandatory because NAT/PNAT and security issues
will always raise (similar to previous slide) and must be considered when designing such a topology for a
customer
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Use case 2: Remote SIP users


Secure remote worker with OTC client on OTBE/OTMS server

Trusted environment
Untrusted environment

Enterprise LAN

DMZ

OTC client
(PC/MAC/Smartphone/Tablet)

OTES
(optional)

External firewall

OXE Call
Server
Internal firewall

OTMS server

Reverse
Proxy

Priv@
HOME

interne
t box

Internet
Pub@

Pub@

OpenTouch SBC

Priv@

NAT

HTTPS
SRTP

RTP

SIP TLS

SIP

NAT5
Hosted NAT Traversal
(SIP)

Connectivity with the remote SIP client over the public Internet
The SBC manages NAT for SIP sessions from the central site (Hosted NAT)
The recommendation is to activate the encryption of the IP flows over Internet
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IP Phone

Use case 2: Remote SIP users


Secure remote worker with OTC client on OTBE/OTMS server

Summary of the OTC clients and OT SBC usage:


Solution

OT SBC

Remark

WEB

PC/MAC

Smartphone

Tablet

(for media over IP)

OTC iPad for Conversation user

Yes (audio and video)

OTC Android Tablet for Conversation user

Yes (audio only)

OTC iPhone for Conversation user

Yes (audio only)

OTC Android Smartphone for Conversation user

Yes (audio only)

OTC iPhone for Connection user

N/A

No VoIP stack available in the software


client

OTC Android Smartphone for Connection user

N/A

No VoIP stack available in the software


client

OTC for PC for Connection user and Conversation user

Yes (audio and video)

OTC for MAC OSx for Connection user and Conversation


user

Yes (audio only)

OTC Web

Yes (audio only)

Based on WebRTC with OTC Web client

N/A

No VoIP stack available in the software


client

OTC One

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Use case 2: Remote SIP users


Secure remote worker with OTC client on OTBE/OTMS server
The OTC clients support standard TLS for encryption of SIP protocol and standard SRTP for
encryption of the media over IP:

SIP TLS
RFC 3261 compliant (support for TLS_RSA_WITH_AES_128_CBC_SHA ciphersuite)
Server certificate with RSA-1024-2048 signed with SHA1 or SHA2 signature

SRTP
media keys negotiation based on RFC4568
AES128-CM encryption algorithm
Support for audio stream and video stream (for OTC clients compatible)

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Use case 2: Remote SIP users


WebRTC in OTC Web

WebRTC and its usage within OTC Web:

OTC Web is the OpenTouch conferencing client that gives access to collaboration services
(such as IM, active talker indication, document or desktop sharing) as well as audio media (as
of product release 2.1.1 based on WebRTC) and video media (in future product release)

OTC Web is a pure Web based client running into any (or quite) recent browser. OTC Web is
primarily targeted at OTs conference guest participants, who are generally people outside of
the enterprise and connect to the OpenTouch conference system via the Internet. OTC Web
can also be used by employees of the company, with Conversation or Connection user profiles

In the current product release WebRTC for audio media is supported into OTC Web when the
user (guest or corporate) is connected remotely from the public Internet through the OT
SBC installed in the DMZ

NB: Support of WebRTC for audio media when the user is connected from the enterprises LAN is
planned for a future product release
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Use case 2: Remote SIP users


WebRTC in OTC Web

OpenTouch Edge Server (OTES):

- Acts as relay towards the OT server where the web application is hosted
- Relays application data flows of the conferencing clients (document and screen sharing, IM etc)

Internet
Corporate
Internet
firewall

OTES

HTTPS

Guest participant
office network

RTP
SIP over TCP/UDP

OpenTouch SBC (OT SBC):

OTC-Web

SRTP

- ICE-lite support (for IP connectivity)


- SIP signaling and media traffic between
remote WebRTC clients and OT server
- SRTP to RTP

Guest participant
over Internet

- Terminates the SIP signaling in a Javascript module


- Manages the media handling thru the WebRTC API
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OT-SBC

SIP over WebSocket


secured

HTTPS

OTC Web:

OTBE/OTMS

DMZ
SRTP

OTC-Web
OTC-Web

HTTPS

Corporate network

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OpenTouch server
(OTBE/OTMS):

- Web application server. Provides SIP


configuration to the client (thru a data
channel flowing thru OTES)
- Hosts Data/Audio/Video conferences

No usage of WebRTC within


OTC Web inside the
corporate network with
current product release !

Use case 2: Remote SIP users


WebRTC in OTC Web

OT SBC is the mandatory WebRTC/SIP Gateway:


-

SIP over Secure Web Socket as a signaling protocol managed by the OT SBC

OT SBC supports the ICE protocol for IP connectivity of WebRTC clients


NB: OT SBC implements an ICE-lite and HNAT (Hosted NAT mechanism) to handle IP addressing for remote
OTC Web with WebRTC. OT SBC is able to reply to STUN requests from the browser with WebRTC even if
it does not implement a complete STUN server. The advantage is that the OT SBC can have a NATted WAN
interface meaning a private IP@ behind a Firewall (instead of public IP@)

Encryption mandatory between WebRTC clients and the OT SBC:


-

Support of DTLS/SRTP for encryption key exchange managed by the OT SBC

OT SBC supports WebRTC feature as of product release 2.1.1 (a specific software


patch is delivered with this release refer to Technical Release Note document for more details -)

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Use case 3: Encryption 8088/8082 for CV users


Encrypted peer-to-peer audio calls between 8088/8082 sets
OT SBC is installed in front of the OTBE/OTMS server and is used to handle the TLS sessions termination for
SIP signaling between the endpoints and the SBC
Secure end to end peer-to-peer calls with Direct SRTP for voice between two 8088/8082 desk phone sets
OTBE/OTMS

Trusted environment

OT SBC

Enterprise LAN

SIP TLS
8088/8082 phone set

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8088/8082 phone set

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SIP
Secure Media
SRTP

Use case 3: Encryption 8088/8082 for CV users


Encrypted peer-to-peer audio calls between 8088/8082 sets

Feature details:
- The OT SBC is used to proxy all the SIP signaling control sessions established between the different
8088/8082 phone sets and the OTBE/OTMS server. These sessions are secured with standard TLS
protocol (SIP TLS) and audio media is encrypted with standard SRTP
- Encryption can be enabled or disabled for each 8088/8082 phone set individually. A 8088/8082
phone set that is connected directly to the OTBE/OTMS server (i.e. not connected through the OT
SBC) will not provide encryption of the communications
- Visual indication (padlock icon) on the 8088/8082 phone set screen showing to the user that the
set is configured to encrypt communications up to the SBC. The padlock icon is not contextual (i.e.
not depending of the called party encryption capability) but is permanent
- Direct SRTP between two encrypted phone sets (Media Release feature)

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Use case 3: Encryption 8088/8082 for CV users


Encrypted peer-to-peer audio calls between 8088/8082 sets

Encryption perimeter:
- End-to-end encryption is provided for the use case of a peer-to-peer call between two Conversation
users equipped with a 8088/8082 phone set
- Partial encryption when the Conversation user with a secure 8088/8082 phone is calling another
Conversation user equipped with a non-secure device/client (ex. 8012 or 8068 DeskPhone, OTC PC
etc): SRTP between the secure 8088/8082 phone and the OT SBC, and RTP (no encryption) between
the OT SBC and the other device
- Partial encryption when the Conversation user with a secure 8088/8082 phone set is calling a
Connection user equipped with any device/client (even if this user has an IP Premium DeskPhone on
the OXE Call Server secured with SSM module): SRTP between the secure 8088/8082 phone and the
OT SBC, and RTP (no encryption) between the OT SBC and the other device

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Use case 3: Encryption 8088/8082 for CV users


Encrypted peer-to-peer audio calls between 8088/8082 sets

Encryption perimeter:
- Partial encryption when the Conversation user with a secure 8088/8082 phone set is calling an
OpenTouch application such as audio conferencing, messaging service (voicemail
deposit/consultation, online recording): SRTP between the secure 8088/8082 phone and the OT SBC,
and RTP (no encryption) between the OT SBC and the OpenTouch server

- Video capability must be disabled on the secure 8088/8082 phone set to be able to support
encryption of the audio stream with OT SBC in current product release

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Use case 3: Encryption 8088/8082 for CV users


Mixing OXE IP Touch Security feature and OT SBC
NOT RECOMMENDED CONSIDERING
INCONSISTENCY OF ENCRYPTION
BETWEEN ALL USERS !

OTMS

OXE CS

OT SBC

SSM-RM
IPSec

NOE
phone
set

SIP
TLS

SIP

A
SRTP

NOE
phone
set

RTP

8088/8082 phone set


CV user

SRTP

8088/8082 phone set


CV user

SIP TLS

IPSec

SIP
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SRTP

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Secure Media SRTP


RTP

Use Cases Summary


Public SIP Trunk:
PCS (Premium Customer Support) for OXE/OTBE feature Public SIP Trunk
Interworking tests with Carrier/SP to be carried out jointly with ALE
Contact ALE local representatives for more information

Remote SIP users:


OTC clients supporting VoIP/Video stack (based on standard SIP protocol) or OTC Web with
WebRTC connected remotely through the public Internet
Encryption based on SIP TLS or Secure WebSocket (for WebRTC) and SRTP over the public Internet
Refer to design guidance in this presentation

8082 and 8088 encryption (CV user only):


For phone sets deployed in the LAN only / independent from the IP Touch Security feature of OXE
(no end-to-end encryption)
Refer to design guidance in this presentation

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AGENDA

1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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Features Overview
Customer needs

Interoperability
Connectivity

QOS

Security

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SBC capabilities
SIP signaling gateway (B2BUA) and WebRTC support
Transport mediation (SIP UDP/TCP/TLS)
Programmable SIP call-flow manipulation (headers,)

NAT/PNAT traversal (level 5 SIP session)


Hosted NAT Traversal (remote SIP end-points)
SIP sessions routing (multi-Carriers, multi-PBX)
IPv4/IPv6

Call Admission Control management


CDR generation (for VoIP performance)
802.1Q and TOS/Diffserv marking
High Availability
compatible with OXE spatial redundancy
OT SBC redundancy mechanism (active/standby)

VoIP aware firewall


ALG SIP
Dynamic Pinholing for voice media
DoS (Denial of Service) prevention
Hidden internal IP PBX topology
TLS encryption for SIP and SRTP for audio+video media
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Features evolution with release 2.1.x

Support of Multi-Tenant (required for Cloud environment)

Serviceability/Management:

Simplification of object model (mono SRD)

HA (High Availability) enhancement: automatic upgrade of redundant server based on software release of the
active server

New SNMP MIBs for SIP Interfaces monitoring

Enhancements on Syslog feature (log tags, device identifier, activity logs management)

Support of WebRTC for remote user with OTC Web over the public Internet

Support of Transcoding for:

DTMF RFC2833 to in band audio (G711)

G711 encoding A law <- -> Mu law

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Packaging
Software delivery for OT SBC

OpenTouch SBC is a pure software solution. The software delivery contains:


- the Operating System CentOS 6.0
OpenTouch
- the SBC software packages
SBC R2.1.x

The OT SBC software must be installed on a compatible virtualized infrastructure in a


dedicated Virtual Machine
As of the MLE Offer H1-2016, ALE does not deliver anymore the OT SBC software pre-packaged
on an appliance server for new system

OT SBC software is compatible with the virtualized infrastructure listed below:


- VMware vSphere ESXi 5.x (5.5 recommended)
- MS Hyper-V
- KVM

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Packaging
Software delivery for OT SBC
SW solution
OT SBC
No Dongle

OS Linux CentOS

VM
Hypervis
or

Not delivered by ALE:


* Licenses for Vmware/MS HyperV/KVM
* Hardware appliance server

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Product Limits
Definitions:
Remote SIP users registered : the SIP users are registered on the communication server through the
OT SBC. The registered SIP users can use UDP/TCP or TLS (for secured a SIP connection)
Simultaneous SIP sessions (also named SIP Calls) used to establish a conversation between two peers
using audio only or audio+video, with or without encryption (TLS for SIP and SRTP for audio/video
media). NB: a SIP session with audio+video consumes two media resources instead of one in case of
audio only (potential impact on the provisioning level)
WebRTC sessions : it is a SIP session with support of WebRTC for a remote user connected through
public Internet and using the OTC Web client on a compatible web browser to join a conference
managed by the OpenTouch server. A WebRTC session is always encrypted
Transcoding sessions : it is a SIP session with support of transcoding feature (ex. DTMF RFC2833 to in
band audio, G711 Alaw Mulaw)
SRTP to RTP sessions : a SRTP to RTP session is a SIP session with encryption of the audio/video
media stream on the untrusted network side (OTC clients connected from the public Internet). This
option does not apply to WebRTC sessions that are always encrypted

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Product Limits (cont)

OT SBC R2.1.1

SIP calls
SIP calls Remote
Transcodi
SRTP/RTP WebRTC
(audio (audio+vide SIP
ng
Sessions Sessions
only)
o)
users
Sessions

OTSBC virtualized
sizing:
x1 vCPU*
RAM 4GB
1000
500
1000
1000
750
300**
OTSBC virtualized
sizing:
x4 vCPU*
RAM 8GB
4000
2000
6000
4000
750
N/A
* 64-bit Intel CPU of at least 2.5 GHz, with support for hardware virtualization (Intel VT-x) enabled
with Advanced Vector Extensions (AVX) and AES-NI support (Sandy-Bridge architecture or newer)
** With x3 additionnal dedicated vCPU (4GB add. RAM/8GB in total)
NB: a SIP session with audio+video consumes two media resources instead of one in case of audio only

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High Availability

OpenTouch SBC supports 1+1 redundancy in Active/Standby mode to provides high


availability (HA)
Redundancy is based on a second identical physical server on which the same software
release is installed (it is a second OT SBC instance)
In redundancy configuration one of the NIC available on the physical server is used for
Primary server to Backup server synchronization information (keep alive packets, sessions
state, configuration synchronization). This allows to keep all active sessions (signaling and
media) upon switchover
In virtualized environment a second OT SBC server can also be configured (with a second
identical Virtual Machine). In such configuration an additional VNIC should be provisioned for
each OT SBC server (primary and backup)
Redundancy principle is based on sharing a Virtual IP address floating between the primary
and the backup SBC. Primary and backup OT SBC servers must be in the same IP subnet
The configuration operations performed on the active SBC via the web interface are
automatically replicated on the standby SBC

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High Availability (cont)


OT SBC
Primary

OT SBC
Backup
Maintenance interface

WAN

(used for HA logical interface between the two devices)

WAN Virtual @IP

LAN

LAN Virtual @IP

The switchover triggers are:


- Manual switchover (command via the management interface)
- Loss of Ethernet connectivity (at Maintenance interface level)
- Keep alive timeout (by default 500ms)

Network constraint: Primary and Backup OT SBC servers must be in the same subnet (same broadcast domain)

- For more technical details refer to the OT SBC User Manual (see HA Configuration section) Presales Presentation | May 2016
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High Availability (cont)


OTMS (w/ or w/o redundancy)

OXE CS Main

OXE CS Standby

OXE Spatial redundancy


@IP:192.168.100.1

@IP:10.0.50.10

Subnet 1

Subnet 2
OT SBC (w/ or w/o redundancy)

OT SBC is compatible with OXE Call Server local and spatial redundancy
OT SBC is compatible with OTMS HA feature (available as of product release 2.1.1)

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High Availability (cont)


Primary Data Center

Backup Data Center

OTMS (w/ or w/o redundancy)

OXE CS Main

OXE CS Standby

OXE Spatial redundancy

OT SBC#1

Main
SIP path

SIP Provider
network

Backup
SIP
path

OT SBC#2

WAN

In case of geographical redundancy (e.g. two Data Centers Primary and Backup) with no
possibility of broadcast domain then one OT SBC system must be configured for each
location. Each OT SBC system can be configured with or without local redundancy

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High Availability (cont)


Important Notes:
OpenTouch Multimedia Services (OTMS) solution supports local server redundancy as of
product release 2.1.1. But geographical redundancy of OTMS is not yet supported. It means
that in a configuration with OXE in spatial redundancy (when the OXE Main server and the OXE
Standby server are installed in two different distant locations), then the OTMS servers
(Primary and Secondary) must be installed in one of the locations (usually in the main site). In
case of loss of connectivity with this location then the software OTC clients managed by the
OTMS will be out of service until connectivity with the location where OTMS servers are
installed is back

OTC clients (OTC PC/MAC, OTC iPhone, OTC iPad, OTC Android smartphone, OTC Android
Tablet) do not support natively the configuration of two SBC systems (by FQDN or @IP).
Consequently the configuration of two different OT SBC systems, one in Primary Data
Center and one in Backup Data Center (with or without internal HA feature configured on
each), cannot be proposed

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Management

OpenTouch SBC is managed and configured with its own management interface (it is not
integrated into the OmniVista 8770 solution)

OpenTouch SBC supports SNMP for supervision and alarms (SNMP TRAP)

A Web based interface provides a user-friendly, secured (HTTPS), graphical user interface
(GUI) to configure the OpenTouch SBC. This web interface is accessible from any web
browser (ex. MSFT Internet Explorer)

The GUI allows to manage:


- configuration
- operation (software upgrade / configuration upload / system reset)
- fault management (real time alarms with severity)
- performance monitoring (voice calls statistics based on CDR)
- security (password policy / RBAC / access control based on @IP)

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Management (cont)
Login screen

Areas of the Web GUI

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Management (cont)
SBC Wizard for configuration:
Off line tool to simplify configuration of the OT SBC
How does it work?
Choose the required use case for example:
- OXE Public SIP Trunk interoperability with a certified Service Provider
- Connectivity with remote user with OTC iPad etc
Enter specific topology/networking parameters of the project
SBC Wizard generates a file the SIP configuration part of OT SBC
The tool is updated based on successful interoperablity tests with a Service Provider (between PBX and
Provider network and SIP service)

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Licensing control

OpenTouch SBC licensing is based on a dedicated license file installed and controlled locally
on the server (physical or virtualized)
There is no interaction with the other communication solutions license files (ex. OXE or
OpenTouch BE/MS)
License file for an OpenTouch SBC system is generated from the ALE ordering tools. See the
details below about how to proceed to get the license file corresponding to your OT SBC
configuration

How to get the license file corresponding to the configuration of OT SBC ?


BP needs to open an e-SR, and include the following information:
- That the e-SR is a request for the OT SBC license (no specific title/subject is mandated, so long as it is clear that the request is
for an OT SBC license)
- The Alcatel-Lucent order number corresponding to the purchase of the OT SBC
- The Host ID (product key+serial number) of the physical or virtual server on which the OT SBC software is installed (as shown on
the system details page of the web-based OT SBC administration console)
Note: if the OT SBC redundancy option has been purchased, to activate it a second Host ID (corresponding to the backup appliance
server or virtual machine in case of virtualization) must be supplied

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AGENDA

1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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How to Quote Principles


Licensing based on:
Number of Remote SIP users registered : it is the number of remote SIP clients effectively connected
to the communication server through the OT SBC
Number of simultaneous SIP Calls : a call is a SIP signaling session crossing and managed by the SBC.
It is used to establish a conversation based on audio only or audio&video. Depending on the caller
and the called parties, one or two SIP calls can be consumed for a conversation. This is illustrated
with the two examples below
A) a local user calling the public network
through a SIP Trunk

B) Two remote users with SIP Softphone

PBX
PBX

IMS

OT SBC

IP Phone
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OTC client
(PC/Smartphone/Tablet)

x1 SIP call

Internet

OT SBC

x2 SIP calls
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How to Quote Principles (cont)


Options
Number of SRTP to RTP sessions (calls with encryption for audio/video media steam). It is an option
applying to a SIP call (meaning that there can be calls with encryption and calls without encryption)
except for WebRTC sessions that are always encrypted
Number of WebRTC sessions. It is an option applying to a SIP call
Number of Transcoding sessions. It is an option applying to a SIP call
Redundancy (second OT SBC instance for High Availability)

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OT SBC Licensing Model (1/2)

N x SRTP (SDES)
channels

N x WebRTC
channels

encryption

SIP calls
options
N x transcoding channel

transcoding

mini = M/10

N x SIP calls

1 x OTSBC
engine

M x SIP users

OTC clients (including OTC Web/WebRTC)

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OT SBC Licensing Model (2/2)

SIP call

SIP user

SIP SRTP
(SDES)

SIP WebRTC
(DTLS)

Transcoding

N' (N'<=N)

N'' (N''<N)

Remote SIP users

M (mini M'/10)

M'

M'' (M''<=M)

8088/8082 encryption

P (mini P'/10)

P'

Licenses

Use cases
SIP Trunk

WebRTC

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OT SBC in ACTIS
OpenTouch Suite for MLE configuration:

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OT SBC in ACTIS (cont)


OpenTouch Suite for MLE configuration:
Indicate the number of Simultaneous SIP calls
that are managed by the OT SBC. It should be at
minimum 10% of the number of remote users

OT SBC product
release

Video Channels: the number of video (+ audio) channels


corresponds to part of the Simultaneous SIP calls". The
remaining number Simultaneous SIP calls" corresponds
to audio sessions (without video). For provisioning level
control only (no associated license)
Number of simultaneous calls where SRTP to RTP is required
for audio/video media.
NB: The value of Simultaneous SIP calls must be equal or
higher than the value of SRTP to RTP

Indicate the number of WebRTC sessions that are


managed simultaneously through the OT SBC

Redundancy option providing


a second OT SBC instance
with Active/Standby principle
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OT SBC Design&HTQ_ed03

Indicate the number of Remote SIP users


that are registered simultaneously on the
system through the OT SBC
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Indicate the number of


Transcoding sessions

OT SBC in ACTIS (cont)


Configuration example for WebRTC feature:

WebRTC
As an example, if the need is to allow x10 OTC Web
users to access simultaneously an OT audio
conference through WebRTC compatible browser

Enter: 10 SIP calls

Enter: 10 WebRTC sessions


to enable WebRTC with OTC Web
client

Enter: 10 remote SIP users


to enable remote users with
OTC Web/WebRTC
to register through the OT SBC

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OT SBC in ACTIS (cont)


OTBE configuration:

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Quotation use case Public SIP Trunk


Example of OT SBC configuration used for Public SIP Trunk only:

PSTN

Corporate network

SIP Provider
network

OXE/OTMS

S1

RTP

OT SBC

RTP
SIP Call1 (S1)

External call to/from PSTN


(A calls an external number)

IP Phone
User A
RTP
SIP

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Quotation use case Public SIP Trunk


OT Suite for MLE or OTBE:

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Public SIP Trunks

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Quotation use case Public SIP Trunk


OT Suite for MLE or OTBE:

Public SIP Trunks

Additional x50 SIP Trunk


licenses (for the total of
x200 SIP Trunk channels
configured)

Bundle pack license for


[SIP Trunk+SBC Call] licenses for
up to 150
Additional x50 SIP Call licenses (for the total of
x200 SIP Calls configured)

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Quotation use case Remote SIP Users


Example of OT SBC configuration used for Remote SIP Users only:

Corporate
Data Center
network

OXE/OTMS

PSTN

SIP
S2

S1

S3

OT SBC

Internet

SRTP

SIP Provider
network

In this scenario OT SBC is used only to


interconnect remote users. It is not
used for Public SIP Trunk with the
SIP Provider.

RTP

SIP Call1 (S1)

SRTP

Internal call with another remote user (A calls B)

SIP Call2 (S2)

External call to/from PSTN (A calls an external

SIP Call3 (S3)

number)

Softphone

User B
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Softphone

User A

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SRTP

RTP

SIP TLS

SIP

Quotation use case Remote SIP Users


Example of OT SBC configuration used for Remote SIP Users only (cont):
- Objective is to size the OT SBC that is at the border of the public Internet and that is used for calls
-

between remote users with softphone (for peer-to-peer calls) and also calls from a softphone to the PSTN
for incoming and outgoing calls
We consider the internal traffic that are the calls from one remote user to another through the OT SBC. As
explained in previous slides, such internal call will consume two SIP Calls at the level of the OT SBC. We
take as assumption that this internal traffic represents 40% of the total traffic
We consider the external traffic that is the incoming and outgoing calls of a remote user from/to the
PSTN. As explained in previous slides, such external call will consume one SIP Call at the level of the OT
SBC. We take as assumption that this internal traffic represents 60% of the total traffic
Erlang-to-Session/Call calculation can be applied to calculate the total number of SIP Calls that would be
required in the OT SBC based on the total number of Remote SIP Users

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Quotation use case Remote SIP Users


Example of OT SBC configuration used for Remote SIP Users only (cont):
60% traffic with external PSTN
one external call = 1 SIP Call OT SBC
40% traffic internal (between remote users)
one internal call = 2 SIP Calls OT SBC (one for remote user caller / one for remote user called)
External traffic:
(X users*0.16E)*0.6= Nerlangs
ErlangTable(Nerlangs,blocking 0.01%) = Nchannels = Nexternal_SIPCalls
Internal traffic:
(X users*0.16E)*0.4= Perlangs
ErlangTable(Perlangs,blocking 0.01%) = Pchannels
Pchannels*2 = Pinternal_SIPCalls

Total_SIPCalls = Nexternal_SIPCalls + Pinternal_SIPCalls

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Quotation use case Remote SIP Users


Example of OT SBC configuration used for Remote SIP Users only (cont):
Dimensioning based on the number of Remote SIP Users configured:

200 users => 38 (Nexternal_SIPCalls) + 56 (Pinternal_SIPCalls) = 94 Total_SIPCalls


600 users => 86 (Nexternal_SIPCalls) + 126 (Pinternal_SIPCalls) = 212 Total_SIPCalls
1000 users => 132 (Nexternal_SIPCalls) + 188 (Pinternal_SIPCalls) = 320 Total_SIPCalls

Example with x1000 remote SIP users with encryption:

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Quotation use case Remote SIP Users + Public SIP Trunk


In this configuration we consider that the same OT SBC system is used for
two usages:
1) Remote SIP Users connectivity (OTC software clients)
2) Public SIP Trunk connectivity with SIP Provider network
The configuration of the OT SBC system takes into account these two usages
in terms of sizing for Remote SIP Users (each user of the solution is
considered potentially as a remote user) and sizing for SIP calls (internal
traffic between remote SIP users and external traffic to the Public SIP
Trunks)

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Quotation use case Remote SIP Users + Public SIP Trunk

SIP calls usage in OT SBC depending on call scenario:


OXE/OTMS

Corporate
Data Center
network

SIP Provider
network

S2

S1

PSTN

RTP

S3 S4

OT SBC

SIP Call1 (S1)

Internet

SIP Call2 (S2)

SRTP
SRTP

Internal call with another remote user (A calls B)

SIP Call3 (S3)

External call to/from PSTN (A calls an external

SIP Call4 (S4)

number)

Softphone

User B
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Softphone

User A
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SRTP

RTP

SIP TLS

SIP

Quotation use case Remote SIP Users + Public SIP Trunk

OT SBC sizing considerations:


Sizing OT SBC (remote SIP users+Public SIP Trunk):
60% traffic with external (through Public SIP Trunks)
one external call = x2 SIP Calls OT SBC (one for remote user / one for SIP Trunk)
40% traffic internal (between remote users)
one internal call = x2 SIP Calls OT SBC (one for remote user caller / one for remote user called)
External traffic:
(X users*0.16)*0.6= Nerlangs
ErlangTable(Nerlangs,blocking 0.1%) = Nchannels * 2 = Nexternal_SIPCalls
Internal traffic:
(X users*0.16)*0.4= Perlangs
ErlangTable(Perlangs,blocking 0.1%) = Pchannels * 2 = Pinternal_SIPCalls

Total_SIPCalls = Nexternal_SIPCalls + Pinternal_SIPCalls

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Nbr users

Total Erlang

Total channels

Total SIP calls

250

40

60

120

500

80

106

212

1000

160

194

388

2000

320

366

732

3000

480

534

1068

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Quotation use case 8088/8082 Encryption for CV users


Example of OT SBC configuration for 8088/8082 encryption:
OXE/OTMS
PSTN

OT SBC

LAN
8088
OT Conversation

8088
OT Conversation

Direct SRTP
(audio only)

8088
OT Conversation

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SIP Call through OT SBC


Peer-to-peer encrypted calls (between
8088/8082 sets)
Direct SRTP
External traffic to PSTN
SRTP to RTP
Access to media server OT (voicemail,
conference etc)
SRTP to RTP

8088
OT Conversation

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Quotation use case 8088/8082 Encryption for CV users


Example of OT SBC configuration for 8088/8082 encryption:

Example with

x50 8088 encrypted sets

Set the number of Simultaneous SIP


calls to the number of 8088 sets
configured with encryption to handle
the possibility for all the 8088 sets
be in a conversation simultaneously

Set the number of SRTP/RTP sessions


for simultaneous calls from a 8088 set
to non encrypted destination (ex.
in/out PSTN, voice mail or conference
etc). Put the same number as 8088
sets configured with encryption to
handle the possibility for all the 8088
sets be in a conversation
simultaneously
Set the number of Remote SIP users to the number of 8088
sets configured with encryption
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FAQ
How to purchase multiple OT SBC systems for the same OXE or OTBE system ?
In some configuration it could be of interest to configure a second or more OT SBC systems (with
independent configuration) for the same OXE or OTBE system.
The first OT SBC system can be configure in ACTIS, the second one must be purchase directly
through eBUY with the commercial items corresponding to OT SBC product. Refer to the
operational guidelines OTS MLE Stand alone Applications available in the eBP website for
details on the process. See screenshot below for illustration:

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How to Quote Services and Lab&Demo


Support Services
OT SBC support services are included in the SPS contract

Demo lab special discount


Composed of one OT SBC system for up to 25 remote SIP users and up to 25 SIP Calls (including SRTP
to RTP , WebRTC and Transcoding capabilities)
Specific SDR to apply on this demonstration configuration (refer to Lab&Demo terms and conditions
official document)

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AGENDA

1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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Training
Technical training (installation and configuration) is provided through Audiocodes partner
training programs (named as Alcatel-Lucent OpenTouch Session Border Controller).
Sales/Presales training : no dedicated program for OpenTouch SBC. The existing ALE training
programs for our IP Communications solutions present the information related to the
OpenTouch SBC product.
- Refer to ALE Educational Services catalog for more information -

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AGENDA

1. General Information
2. Use Cases Details
3. Technical Details
4. Quotation Guidance
5. Product Training Information
6. Reference Documentation

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Reference documentation
Available in the eBusiness Portal :
Product > Communication Platforms > OT Session Border Controller

Feature List & Provisioning Level information (integrated in the FL/PL OpenTouch excel file)

Standard Offer (Security section for OT SBC Product Description)


OpenTouch SBC datasheet

Resources Center > Presales Documentations


Presales Presentation - OpenTouch SBC (this presentation)
Presales Presentation Solution for Remote User Access
PCS (Premium Customer Support) form for Public SIP Trunk feature/configuration

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Reference documentation
Available in the eBusiness Portal :
Customer Support > Technical Support
Technical Knowledge Base (installation and administration manual for OT SBC is integrated into the
OpenTouch technical documentation)
TechCom document Certified SIP Provider/Carrier for OXE reference list document (TC2005)

Business Operation

OTS MLE Standalone Applications Order Entry Guide

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ANNEX

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FAQ
1) How to get the software delivery for OT SBC ?
The software delivery can be downloaded from eBP (Technical Support/Software Download). However it is
not visible in the eBP by default: the BP must request TechSupport to get visibility of this specific software
package in the eBP. To do so, the BP can get the "Request For Security Patches" form in the eBP, fill this form
and tick the "OpenTouch SBC" option in the form. The BP will then get the access to the software.

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Hosted NAT Traversal principal

Call Server

Translation of private IP addresses of


equipments in the corporate LAN at the
level of the SBC (Hosted NAT Traversal)

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enterprise.alcatel-lucent.com

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