2018-04-10

Rainbow
WebRTC Gateway
Technical Business Engine
April 2018
1 Patrick LEMONNIER
History
Edition 01: Creation of the document for “Rainbow – Gateway”
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.
Rainbow – WebRTC Gateway - ed01 2

Pre-Sales COPYRIGHT © 2018 ALE INTERNATIONAL. ALL RIGHTS RESERVED.
Agenda
1. Introduction
2. Overview
3. Features
4. Misc
5. Deployment
6. Licenses & Quotation
7. Topology
8. Use cases
9. Multi-device
10. Annexes

Introduction

Rainbow WebRTC Gateway
Introduction
 The objective of this presentation is to bring technical information on the WebRTC gateway
permitting a media communication between Rainbow and OXE PBX or OXO PBX.
 If there are specifics points to a PBX type, OXE or OXO naming are mentioned.
 Topologies, flows, use cases and sizing are treated in this document.
 This presentation is destined to PreSales, BP and Customers.
 The content is aligned with GA of the WebRTC gateway available in June/July 2018.
 Should be with Rainbow 1.43

Overview

What is it?
 Rainbow WebRTC gateway is a software component relaying media communications

between the WebRTC Rainbow applications, and standard RTP-based PBX ecosystem
components:
 Voice between rainbow clients, phones and off-net calls,
 Unified multi-device experience (one-number): softphone and deskphone.
 This gateway is located in customer premises.
 Complements the ALE Hybrid solution based on Rainbow and PBX (OXE or OXO).
Media flows between a Rainbow client and a deskphone
WebRTC SIP/RTP
WebRTC Gateway
Rainbow WebRTC client Any deskphone/sofphone

Service levels
 What can be done from a Rainbow client?
PBX not connected PBX connected PBX connected to Rainbow and
to Rainbow to Rainbow WebRTC gateway deployed
Rainbow Cloud Rainbow Cloud Rainbow Cloud
• Establish calls between • Establish calls between • Establish calls between Rainbow
Rainbow clients only Rainbow clients only clients
• No RCC on deskphones • RCC between deskphones HTTPS • RCC between deskphones HTTPS HTTPS
• Establish calls between a Rainbow
client and a deskphone
Rainbow Rainbow
Agent Agent
SIP WebRTC Gateway
PBX PBX PBX
Company premises Company premises Company premises
Encrypted media flow

Non encrypted media flow

Flows (1/2)
 The role of the WebRTC gateway is twofold :

 It’s a signaling translator, permitting exchange between XMPP/Jingle used by Rainbow clients and
SIP used by terminals/clients on PBX side. (J2S: Jingle-to-SIP gateway)
 It’s also a media proxy between encrypted DTLS-SRTP media (WebRTC) and “regular” RTP media
used on PBX side.
Media flows from and to the WebRTC Gateway
Rainbow Cloud Extension to JID

resolver service phone/client
Rainbow user/client identification is done
identification is done with an extension
with a JID
HTTPS/REST
PBX Extension
PBX
HTTPS/XMPP/Jingle Jingle2SIP SIP PBX Signaling
or public extension
Gateway
Public
DTLS-SRTP media (WebRTC) WebRTC RTP media
Carriers
Media Proxy
Any deskphone/sofphone
Rainbow WebRTC client WebRTC Gateway

Rainbow WebRTC Gateway Rainbow Cloud XMPP Servers
Flows (2/2) Extension to JID

PCG resolver service
TURN
 The WebRTC Gateway is located on the LAN,
HTTPS/XMPP/Jingle
near the PBX. HTTPS/
REST
 Does not require to deploy it in the DMZ. Internet
WebRTC indirect
media (DTLS-SRTP)
HTPPS/
HTTPS/XMPP/Jingle Rainbow WebRTC
 As Rainbow clients, it is always at the origin
XMPP/
Jingle client on WAN
of IP traffic towards the Rainbow cloud, and

maintains connections.
 So, it is always reachable from cloud components. WebRTC direct
WebRTC Gateway
media (DTLS-SRTP)
Rainbow WebRTC
client on LAN
SIP Trunk
 Pay attention that no SBC is required. PBX ecosystem PBX ecosystem

media (RTP) Media (RTP)
 The WebRTC gateway talks directly with the PBX device

PBX device
on LAN
PBX signalization
Rainbow server in the Cloud. on LAN PBX signalization
PBX
Company premises
 Some IS/IT may want to deploy it in the DMZ.
Dashed line means the flow is encrypted.

Features

OXE - Telephony features (1/3)
 Telephony
 Make call, take call, release call.
 Hold call/retrieve call, divert call, second call, alternate call.
 3 party conference.
 Multi-device management
 Incoming calls make all user’s OXE devices and Rainbow client ring.
 Shift a VoIP call from 4G to wifi and/or wifi to 4G

 This is managed automatically by WebRTC stack.
 Not possible to shift from a VoIP 4G call to cellular 2G call in case of voice quality issue.
 Shift a call seamlessly between a Rainbow client and a deskphone is targeted in H2/2018

OXE - Telephony features (2/3)
 Conferences
 Thanks to DTMF support, it’s possible to join from the Rainbow VoIP client (Pc/Smartphone) any audio
conference (dial-in):
• OXE meet-me conference,
• OTMS conference,
• Rainbow audio conference (Pgi).
 A Rainbow client can be included in a Hunting Group.

 Thanks to the integration of the Rainbow client through a Remote Extension (REX).
 However, if the Rainbow user is busy in a pure Rainbow conversation, he is seen available from the PBX
perspective.
 IVR can route calls to the VoIP Rainbow client.

 In fact the rooting is done to the Remote Extension associated to the Rainbow client.
 There is no notion of explicit routing to a Rainbow client at the PBX level.
Telephony features (3/3)
 Voice mail
 From his Rainbow VoIP client, a user:
• Can consult his voice messages through TUI commands (DTMF).
• Can deliver a voice message if the recipient is forwarded on his
voice mail.
 Pay attention, currently access to the voicemail from the

Rainbow client is possible from PC client only.
• Smartphone client will provide this possibility at the GA
(June/July 2018).

User interaction
 From the desktop user interface, the user can:

 Define the set from which he will make calls (office phone, computer, home phone, mobile phone).
 Define if incoming calls will ring on his main set (office phone) and on his secondary set (ex : mobile)
or on his main set only.
 Forward calls to his voice mail or to another number.
 From his smartphone user interface, the user can:

 Define on incoming calls if his smartphone will ring or not (dual ring or not).
 Forward calls to his voice mail or to another number.

Graphical user Interface
 Menu to launch calls is unchanged compare to a simple PBX

connection.
 Menu accessible from the icon “Connected PBX” is enriched

when the WebRTC gateway is available.
 Selection of the device from which the call will be done (dial from).

Misc

Codecs
 The WebRTC Gateway supports: Opus, G722 and G711 codecs.

 It doesn’t support transcoding.
 Reminder:
 On OXE side G729, G722, G711 codecs are supported.
• G722 is not supported on hardware media gateways and OMS supports it for OXE Conferences (all types) only.
• G722 is supported by NOE IP sets (3G & EE).
 On OXO side G711, G722 and G729 codec are supported.

• G722 is not supported by VoIP DSP for transcoding.
 The codec G711 and G722 can then be used between PBX and the WebRTC Gateway.
 OXE case: If transcoding is needed in case of WAN topology, it is done by OXE gateway.

Recording
Rainbow Cloud
 At the client level
HTTPS
 Voice recording can be done by the user with the

Rainbow client.
 The recording is save locally or on the Rainbow user
storage space. HTTPS
WebRTC
 At the OXE level Duplicated voice flow
 Voice recording done thanks to OmniPCX Record OmniPCX Record 3 party

WebRTC Gateway
Server
solution using a DRLink recording. IP flow embedding
- Voice of user A
conference
- Voice of user B OmniPCX Record
 Any Rainbow user associated to a Remote Extension

- And all voice to record
Packetizer
TDM
(physical link) RTP
(REX) and doing a call through the WebRTC gateway. SIP Trunk
PCM Board
• Not possible to record voice between two Rainbow users CSTA (DR Link)
using their Rainbow VoIP client.
 Additional compressor resources must be provisioned OXE
for 3-party conferences, depending on the number of Monitored REX associated
simultaneous recordings. IP Phone to a Rainbow user

DTMF
 DTMF sending from Rainbow VoIP client in the Band following the RFC 4733/2833 mechanisms.
 RTP payload format conveys DTMF and tones.
 Besides decryption of the DTS-SRTP flow, the WebRTC Gateway doesn’t modify these payloads.
 Dedicated dial pad is provided in Rainbow client.
 For OXE:
 Due to REX usage, a pool of Q23 DTMF detector on OXE side is currently needed.
 Pay attention: OXE Hardware gateway is then needed (OMS doesn’t provide such resources).
 For OXO, nothing specific.

Deployment

Installation
 In an OXE context:
 Require OXE release 12.1 MD2, 12.2 or later.
 The WebRTC gateway is installed over a dedicated VMWare VM in the customer premises:
• On a dedicated physical server (ESXi),
• Or on the same physical server (ESXi) as for OXE (one VM for WebRTC gateway and one VM for OXE).
 The VM software (OVF) is delivered by ALE and requires a VMWare release 6.X (6.0, 6.5).
 The Operating System is Debian release 9, provided by ALE with the OVF.
 Needed to connect previously OXE to Rainbow:
• Check the presence of the PBX,
• Identify users on the OXE network.
 In an OXO context:
 Require OXO Connect 3.0 or OXO Connect Evolution 3.0.
 For the first step the installation is done on a separated physical server identical to OXE.
• A deployment directly embedded in OXO Connect Evolution is planned.
Capacity & VM Sizing
 The WebRTC gateway load depends on the number of simultaneous calls between PBX and
Rainbow clients.
 The VM has been sized for a typical configuration 100 simultaneous sessions and require:
 2 vCPU
 4800 MHz for CPU reservation
 2 GB vRAM
 20 GB vDISK
 1 GB vNIC
 Pay attention, that the VM sizing provided here is not yet completely finalized and may not
perfect for XL project.
 It will be adjusted in the coming months after additional tests.
 The maximal limit in terms of simultaneous calls is then not yet defined.
 If more than 100 simultaneous calls is required, contact the Technical Business Engine team.

Capacity – UCaaS cases
 UCaaS (Hybrid softphony users)
 Rainbow Hybrid Softphony services require making PBX users multi-device by allocating one VT-
Anyphone(OXO)/REX(OXE) virtual resources per user, necessary for using Rainbow applications as VoIP
termination.
 For OXE:
• The number of Rainbow Hybrid Softphony users is limited to 5000 (corresponding to the number of REX on OXE).
- Pay attention that, this limit is common with the TDM, Analog and DECT phones.
 For OXO:
• The number of Rainbow Hybrid Softphony users is limited to 50.

Capacity – CPaaS cases
 CPaaS cases allow pure Rainbow user to call PBX extensions:
 Don’t require that the PBX user’s device be associated to Remote extension (REX/VT-Anyphone).
 It’s then not an hybrid softphony use case.
 All PBX users can therefore receive calls from Rainbow applications.
 The limit is the limit of the number of simultaneous calls through the WebRTC gateway.

SIP Trunk configuration
 SIP Trunk establishment and monitoring

 The SIP trunk between the PBX and the WebRTC gateway is always established at the initiative of the PBX.
 The PBX is also responsible to maintain the trunk.
• If the link failed then the PBX retried periodically to reestablish it.
 SIP Trunk characteristics

 The level of telephony services on the SIP trunk between is limited to basic SIP calls.
 The PBX trunk must therefore be configured with a limited services category:
• “public” ISDN trunk on OXE,
• or “heterogeneous” level of service on OXO.

High Availability - OXE (1/2)
 High Availability based on VMWare HA:
 It’s a cold active/standby model.
 Compatible with local and spatial OXE HA.
 Only one instance of the WebRTC gateway deployed with a pair of OXE CS (Active/Standby).
 Primary and secondary see only one gateway through the trunk.
 VMware cluster with a minimum of two hosts and VMware HA enabled.
 Hosts of the cluster located in a single IP subnetwork.
• Layer 2 only (incl. extended LAN with stretched cluster).
 If the hosts are spread over an extended LAN: VMware stretched cluster pre-requisites.
 License
 No specific license on WebRTC gateway side.
 VMware vSphere (Standard, Enterprise & Enterprise Plus) or vSphere Essentials Plus kit.
• VMware HA service is not available with ESXi free hypervisor or vSphere Essentials kit.

High Availability – OXE (2/2)
 Local or spatial redundancy
One site Two sites Two sites

Local HA Local HA for WebRTC gateway Spatial HA for
Spatial HA for OXE WebRTC gateway and for OXE
Rainbow Cloud Rainbow Cloud Rainbow Cloud
VMware stretched cluster
WebRTC Gateway WebRTC Gateway
WebRTC Gateway VMWare HA WebRTC Gateway VMWare HA WebRTC Gateway WebRTC Gateway
VMWare HA
SIP Trunk SIP Trunk SIP Trunk

SIP Trunk SIP Trunk
SIP Trunk
OXE HA Link OXE HA Link OXE HA Link

OXE 1 OXE 2 OXE 1 OXE 2 OXE 1 OXE 2
Site 1 Active Standby Active Standby Active Standby
Site 1 Site 2 Site 1 Site 2

License & Quotation

Licenses & quotation (OXE)
 Rainbow side:
 WebRTC Gateway services are accessible to users with a Business or Enterprise subscription.
 OXE side:
 SIP trunk is free of charge for WebRTC gateway connection (bypassed).
 One REX (Remote Extension) license is required per Rainbow user with or without a deskphone.
• Needed for multi-device configuration on the PBX.
• And for the One Number notion.
 For DTMF, a pool of Q23 resources corresponding to the number of simultaneous calls through the
WebRTC gateway.
 Additional compressors might be required (not calculated by Actis) depending on.
• Topology and codecs used.
- For example: when a call must be established between a Rainbow client and a phone set through a WAN implicating G729 codec.
• Number of simultaneous communications going through the WebRTC gateway.
 Quotation rules are those usually apply for REX and simultaneous calls. So there is no specific
fields in Actis related to the Rainbow WebRTC gateway.
Licenses & quotation (OXO)
 Rainbow side:
 WebRTC Gateway services are accessible to users with Business or Enterprise subscription
 OXO side:
 SIP trunk licenses (one per channel)
• Recommendation use: SIP Trunk in mode not reserved
 One Anyphone license (UTL licenses) is required per Rainbow user with a deskphone

Hybrid services & subscription
 OXE services depending on the subscription
Subscription
Integration Hybrid Services with PBX or WebRTC gateway Essentiel Business Enterprise
Business phone control w/ basic call control (call, answer, release) l l l
Phone presence l l l
Screen popup on Rainbow clients when phone rings l l l
BPX Connection
Call logging (phone call history) l l l

Caller identification, contact search l l
Call control up to 3 participants, call forwarding l l
Voicemail (notifications, call) l l
Telephony calls l l
Gateway
WebRTC
PSTN calls l l

Topology

UCaaS - OXE Topology (1/3) Rainbow Cloud
 Single node
 One WebRTC Gateway per OXE node.
 On a single site or on multi-site.
 OXE High Availability is supported.
Domain 1
Domain 1 OXE CS
OXE CS WebRTC GW
WAN
WAN
Domain 1
WebRTC GW
OXE CS WebRTC GW
Domain 2 Domain 3 Domain 1 Domain 2

Single Site Multi Site Multi Site
WebRTC Gateway with OXE CS WebRTC Gateway on a remote site
UCaaS - OXE Topology (2/3)
 OXE in Network and users with Enterprise subscription spread on all nodes
 One WebRTC gateway per OXE node
ABC/F
Node 3
ABC/F
OXE CS WebRTC GW
Node 2
OXE CS WebRTC GW
Node 1
ABC/F
WebRTC GW OXE CS WAN
Domain 1 Domain 2

UCaaS - OXE Topology (3/3) OXE’s users with a Rainbow
Essential subscription only
 OXE in Network and users with Business or Enterprise subscription

not spread on all nodes
 Others users have no subscription or a ABC/F
Essential subscription only
ABC/F Node 3
 One WebRTC gateway per OXE node having OXE CS
users with a Rainbow Business or Enterprise
subscription
Node 2
OXE CS WebRTC GW
Node 1
ABC/F
WebRTC GW OXE CS WAN
OXE’s users with a Rainbow
Business or Enterprise subscription
OXE’s users with a Rainbow

Business or Enterprise subscription
Domain 1 Domain 2

CPaaS - OXE Topology
 OXE Network – Centralized WebRTC Gateway

 One WebRTC Gateway for several OXE.
 Eligible for specific cases only: ABC/F
• Rainbow CPaaS mainly Node 3
• Low traffic ABC/F
OXE CS WebRTC GW
Node 2
OXE CS
Node 1 WAN
ABC/F
OXE CS
Domain 1 Domain 2

Use cases

Use cases introduction
 The WebRTC gateway, which brings voice interoperability between Rainbow Client and
PBXs ecosystem, addresses two use case families:
 UCaaS use cases:

• A Rainbow user makes and receives its PBX calls on Rainbow’s VoIP client for Mobile, PC and Browser, in
addition to his existing PBX device.
• Pay attention, this UCaaS use case requires that the PBX user is provisioned with REX/VT.
 CPaaS use cases:

• Brings softphone capabilities to dedicated integration into a customer ecosystem
• Remote extension is not requested.
• Currently, only calls from the Rainbow clients to PBX extension are supported.

CPaaS Use case
 Enables audio media relationship between Rainbow applications and devices of a connected
PBX.
Rainbow Cloud
 Especially targeted for Verticals and integration HTTPS
of Rainbow capabilities in business processes. WebRTC
Customer
HTTPS HTTPS
 Typical example: User “Customer” browsing a Company’s website on WebRTC
the Internet clicks in the web page to enter in a voice/video call with
an “Employee” of the company on a PBX’s device and obtains any
support. Rainbow
Agent
 “Customer” has an account on Rainbow. It can be a Guest account created SIP

WebRTC Gateway
for the duration of the session or a permanent Guest account assigned to the OXE RTP
Customer. This Customer account is part of the Company domain of Rainbow.

Employee
 “Employee” can be a real employee or a Queue of a Contact Center, an ACD, 21855..
Company premises
a Hunting Group.
UCaaS Use case (1/2)
 Calls from a Rainbow client
 Any telephonic extension can be reached, Rainbow Cloud
 inside or outside the company. HTTPS
WebRTC
HTTPS
WebRTC
WebRTC Gateway RTP PSTN

RTP
SIP Trunk
21855.. OXE +33155..
Inside Outside

UCaaS Use case (2/2)
 Calls to a Rainbow client
 From any telephonic extension, Rainbow Cloud
 Distributed to PBX devices associated with the HTTPS
Rainbow user.
 Incoming calls rings devices according to WebRTC
user’s settings. Any phone HTTPS
BOB DECT phone
Office phone
 With the WebRTC Gateway
WebRTC Gateway PSTN
 On “Rainbow” routing choice, the smartphone is SIP Trunk
fully managed by Rainbow and no more thru OXE
(no GSM overflow). Incoming calls are VoIP. OXE +33155..
 On “GSM” routing choice, the smartphone is

ALICE calls BOB
managed by the OXE and incoming calls are 21855.. Internal or external
cellular.
Multi-device

Multi-devices aspect (1/2)
 Deskphone + Softphone + Smartphone
 The user can either configure is remote extension to his Incoming PSTN/PBX calls don’t rings the Rainbow clients.
Rainbow client or to “other” phone. But user can take call from the Rainbow client in RCC
The remote extension

Rainbow Cloud is configured with the
“other” number
HTTPS
REX/VT
REX/VT
Remote extension is configured
with the number (prefix) of the WebRTC Gateway
Main Secondary Rainbow application
Main Secondary
SIP Trunk
Routing on UCaaS clients PBX Routing on other number

Multi-devices aspect (2/2)
 Softphone + Smartphone
 The user can either configure is remote extension Incoming PSTN/PBX calls don’t rings the Rainbow clients.
to his Rainbow client or to “other” phone. But user can take call from the Rainbow client in RCC
Rainbow Cloud
HTTPS
REX/VT
REX/VT
Remote extension is configured The remote extension
with the number (prefix) of the WebRTC Gateway is configured with the
Rainbow application “other” number
SIP Trunk
Routing on UCaaS clients PBX Routing on other number

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Rainbow

Rainbow – WebRTC Gateway - ed01 2

Rainbow – WebRTC Gateway - ed01 3

Rainbow – WebRTC Gateway - ed01 4

 This presentation is destined to PreSales, BP and Customers.

Rainbow – WebRTC Gateway - ed01 5

Rainbow – WebRTC Gateway - ed01 6

 Rainbow WebRTC gateway is a software component relaying media communications

 This gateway is located in customer premises.

Media flows between a Rainbow client and a deskphone

Rainbow WebRTC client Any deskphone/sofphone

Rainbow – WebRTC Gateway - ed01 7

PBX PBX PBX

Company premises Company premises Company premises

Encrypted media flow

Rainbow – WebRTC Gateway - ed01 8

 The role of the WebRTC gateway is twofold :

Media flows from and to the WebRTC Gateway

Rainbow Cloud Extension to JID

Rainbow – WebRTC Gateway - ed01 9

Flows (2/2) Extension to JID

 Does not require to deploy it in the DMZ. Internet

of IP traffic towards the Rainbow cloud, and

 Pay attention that no SBC is required. PBX ecosystem PBX ecosystem

 The WebRTC gateway talks directly with the PBX device

Rainbow – WebRTC Gateway - ed01 10

Rainbow – WebRTC Gateway - ed01 11

 Shift a VoIP call from 4G to wifi and/or wifi to 4G

Rainbow – WebRTC Gateway - ed01 12

 A Rainbow client can be included in a Hunting Group.

 IVR can route calls to the VoIP Rainbow client.

 Pay attention, currently access to the voicemail from the

Rainbow – WebRTC Gateway - ed01 14

 From the desktop user interface, the user can:

 From his smartphone user interface, the user can:

Rainbow – WebRTC Gateway - ed01 15

 Menu to launch calls is unchanged compare to a simple PBX

 Menu accessible from the icon “Connected PBX” is enriched

Rainbow – WebRTC Gateway - ed01 16

Rainbow – WebRTC Gateway - ed01 17

 The WebRTC Gateway supports: Opus, G722 and G711 codecs.

 On OXO side G711, G722 and G729 codec are supported.

Rainbow – WebRTC Gateway - ed01 18

 Voice recording can be done by the user with the

 At the OXE level Duplicated voice flow

 Voice recording done thanks to OmniPCX Record OmniPCX Record 3 party

- Voice of user B OmniPCX Record

 Any Rainbow user associated to a Remote Extension

Rainbow – WebRTC Gateway - ed01 19

 For OXO, nothing specific.

Rainbow – WebRTC Gateway - ed01 20

Rainbow – WebRTC Gateway - ed01 21

Rainbow – WebRTC Gateway - ed01 23

Rainbow – WebRTC Gateway - ed01 24

Rainbow – WebRTC Gateway - ed01 25

 SIP Trunk establishment and monitoring

 SIP Trunk characteristics

Rainbow – WebRTC Gateway - ed01 26

Rainbow – WebRTC Gateway - ed01 27

 Local or spatial redundancy

One site Two sites Two sites

Rainbow Cloud Rainbow Cloud Rainbow Cloud

VMware stretched cluster

WebRTC Gateway WebRTC Gateway