One Voice SIP Trunk UK
One Voice SIP
Trunk UK
Issue 3.1b
Page 1
One Voice SIP Trunk UK
Contents
1.
Introduction .................................................................................................................................... 4
2.
Product Description ........................................................................................................................ 4
2.1.
Network Design ........................................................................................................................ 7
2.2.
IP Connect UK Access ............................................................................................................... 8
2.3.
Ethernet Connect UK access ..................................................................................................... 8
2.3.1.
The SIP Trunk .................................................................................................................. 8
2.3.2.
Trunk Hierarchy............................................................................................................... 9
3.
Main Features & Functionality...................................................................................................... 11
3.1.
Public Telephony service ........................................................................................................ 11
3.1.1.
3.2.
Emergency services ....................................................................................................... 12
Centralised PSTN Break-in ...................................................................................................... 12
3.2.1.
Inherited IP Resiliency ................................................................................................... 14
3.2.2.
Call Admission Control .................................................................................................. 14
3.3.
Incoming Call routing options ................................................................................................ 15
3.4.
Differentiated Trunks ............................................................................................................. 15
3.5.
Out of Area connection .......................................................................................................... 15
4.
Supported Endpoint devices ......................................................................................................... 16
4.1.
Supported Codecs .................................................................................................................. 16
1.1.1
Fax Support ................................................................................................................... 17
4.1.1.
Bandwidth calculations ................................................................................................. 18
4.1.2.
Codecs and Voice Quality.............................................................................................. 18
5.
Connection to the Service ............................................................................................................. 20
5.1.
Connection to the Service through IP Connect UK................................................................. 20
5.1.1.
5.2.
Connection to the Service through Ethernet Connect UK ..................................................... 20
5.2.1.
6.
SIP connectivity ............................................................................................................. 20
SIP connectivity ............................................................................................................. 20
Telephone Numbers...................................................................................................................... 22
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One Voice SIP Trunk UK
6.1.
New Number provides ........................................................................................................... 22
6.2.
Number Block Transfer........................................................................................................... 22
6.3.
Number Portability and Number Transfer ............................................................................. 23
6.4.
Mapping Telephone Numbers to Trunks................................................................................ 23
6.5.
Incoming Call Routing ............................................................................................................. 24
6.6.
Other Calling Features ............................................................................................................ 24
6.7.
DDI (Direct Dialling In) ............................................................................................................ 24
6.8.
DNIS (Dialled Number Identification Service) ........................................................................ 24
6.9.
Overflow, call diversion, BT Smartnumbers ........................................................................... 25
7.
Service Wrap ................................................................................................................................. 26
7.1.
Service Options ....................................................................................................................... 26
7.2.
Service Centre......................................................................................................................... 26
7.2.1.
Customer Helpdesk ....................................................................................................... 27
7.2.2.
BT Helpdesk (GS Front Desk)......................................................................................... 27
7.2.3.
Planned maintenance ................................................................................................... 28
7.3.
Proactive Management .......................................................................................................... 28
7.4.
Billing ...................................................................................................................................... 28
7.4.1.
OneBill ........................................................................................................................... 28
7.4.2.
BT Billing Analyst Converge........................................................................................... 28
7.5.
Reporting ................................................................................................................................ 29
7.6.
Service Level Management .................................................................................................... 29
7.6.1.
Service Level Agreements ............................................................................................. 30
7.6.2.
Lead times ..................................................................................................................... 30
7.6.2.1.
Provision............................................................................................................................ 30
7.6.2.2.
Service Availability and Restoration Figures ..................................................................... 31
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One Voice SIP Trunk UK
1.
Introduction
BT One Voice SIP trunk UK is a flexible, resilient, low cost alternative to ISDN30 that uses the data
network to route PSTN traffic. One Voice SIP trunk UK offers in-built resilience at no extra cost –
routing calls to an alternative site if the primary target becomes unavailable.
BT One Voice SIP trunk UK will help to reduce costs by optimising the number of channels needed
compared to ISDN30, offering free on-net calls, and enabling utilization of existing data
infrastructure to build on your current investments. BT One Voice SIP trunk UK works with a wide
range IP PBXs from all the major suppliers and can also connect to TDM PBX’s via the appropriate
gateway devices. BT One Voice SIP trunk UK offers number porting from virtually all geographic
numbers in the UK meaning customers have no need to change from existing telephone numbers.
Included as part of BT One Voice SIP trunk UK are Cloud based Session Border Controllers – this
means the customer does not need to provide their own Session Border Controller, as this
functionality is provided as part of the BT product. There are versions of the BT One Voice SIP trunk
UK product offering inclusive call packages or PAYG PPM rates. The product also gives extensive selfservice MIS reporting with a wide range of statistics available. BT One Voice SIP trunk UK is included
in BT’s compliance with ISO27001 and ISO22301.
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One Voice SIP Trunk UK
2.
Product Description
BT One Voice SIP trunk UK is an alternative to ISDN30 that uses the data network to route PSTN
traffic. It can help improve an organisation's flexibility, offer additional resilience and
operational efficiency whilst helping to reduce costs by:
Minimising the number of channels needed to rent compared to ISDN30
Lower cost channels
Free on-net calls
The ability to flex capacity up and down so you don’t pay all year, just to cope for the peaks
Utilise existing IP Connect infrastructure to build on your current investments
Providing a range of channel types so you only pay for the grade of service you need
In addition One Voice SIP trunk UK offers in-built resilience at no extra cost – routing calls to an
alternative site if the primary target becomes unavailable.
The following functionality is supported by the BT One Voice SIP Trunk Product;

Native SIP protocol support – (This also means that no conversion gateway is required at
the customer premises).

Cloud based Session Border Controllers – this means the customer does not need to
provide their own Session Border Controller, this functionality is provided in the BT
product.

Number Porting – carry your existing (geographic) phone numbers across to One Voice SIP
trunk UK from other BT and OLO products.

PSTN break-in – you can receive calls from the Public Switched Telephony Network
(national, mobile, international).

PSTN break-out – you can make calls to destinations normally reachable over the Public
Switched Telephony Network (national, mobile and international), including short
numbers and premium number services).

Delivery of the Calling Line Identity of the caller onto your IP PBX.

Delivery of Advanced Presentation Number features, supported by Presentation Number
screening.

Trunk level Call Admission Control – specify the maximum number of simultaneous calls a
particular trunk and hence individual customer site can handle.

Trunk Group level Call Admission Control – specify the overall maximum number of
simultaneous calls the platform needs to cater for your complete estate. (multiple
customer sites).

Channel aggregation across sites – buy only the aggregated capacity. This means in
practice that fewer channels need to be bought to obtain the same grade of service
compared to an equivalent ISDN implementation.

Deliver Round Robin Trunk call distribution – ability to load share telephony traffic across
multiple sites evenly.

Priority Based Trunk call distribution – enable you to specify main sites and
backup/overflow sites should the main site fail or become.

Ability for you to purchase Dynamic Sip channels – flex the call throughput capacity in line
with business’s seasonality.
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One Voice SIP Trunk UK

Provision of Call Barring features – to prevent certain calls from taking place.

Provision of Call Diversion features – point incoming calls to different destinations
depending on the condition (available, busy, error) of the Trunk/destination location.

Compatible with BT Smartnumbers – turn on an alternate dial plan for all numbers in case
of inaccessibility of the primary answering location.

Emergency Call Handling – ensure that 999 calls are treated correctly and the appropriate
address information is displayed to the emergency operator.

Directory Listing – add or keep your numbers in the telephone book.
One Voice SIP trunk UK is designed to be an access agnostic public telephony trunking service.
Access agnostic implies that customers will need to have a suitable WAN/data network that
supports SIP Trunks. This includes BTs IPConnect and Ethernet Connect Services
Public Telephony Trunking implies it can be used instead of another telephone trunk line service
(like ISDN30) but also that the service does not connect any end-users directly. The end-user
functionality is expected to be provided either by the IP-PBX that consumes the trunking service
or by the hosted/network platform when that gets integrated to One Voice SIP trunk UK. It also
implies the service is earmarked for conversational telephony and does not support video nor
clear bandwidth 64K data services.
One Voice SIP trunk UK, being access agnostic requires a customer to have – in advance – a
suitable data network. There are currently two accesses that are supported: IP Connect) and
wires only Ethernet Connect
Product Elements
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One Voice SIP Trunk UK
2.1.
Network Design
The Network elements making up One Voice SIP trunk UK include A-SBC’s which are connected
to the customers VPN (Virtual Private Network (IPConnect or Ethernet Connect). Each customer
is connected to multiple A-SBC which are geographically located at 2 different platform POPs
(Points of Presence). Each of the network POPS include core elements of the SIP network
containing a Call Server that manages call controls and features. The POPs are connected using
the 21CN network elements that are then connected to the public network to facilitate
incoming and outgoing PSTN calls.
Customer Site A
PSTN Network
Auto Fail Over
POP 1
Customer Site B
POP 2
Figure 1: Network Design
The physical network equipment is geographically separated as well as locally resilient with 2
nodes at each location.
The various physical components used in the design are:
PE router – The Provider Edge Router of the IP Connect UK core network. The service platform
used by One Voice SIP trunk UK connects in a resilient way with IP Connect UK core, using
double links at each of the two geographically dispersed interconnect locations.
A-SBC or Access SBC – This Session Border Controller acts as a CE (Customer Edge) router to IP
Connect UK and is the demarcation of the interconnect between One Voice SIP trunk UK and the
IP Connect UK VPN network. In addition, the A-SBC does the necessary security functions such
as SIP-telephony aware firewall functionality as well as topology hiding. For Ethernet Connect
UK, the SBC is in High Availability mode. For IP connect UK, the SBC’s are in Island mode.
OSCS Open Session Control Server – This Call Server will ensure authentication, do (static)
registration of SIP trunks, handle routing of calls and be the point where the features of One
Voice SIP trunk UK will be developed.
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One Voice SIP Trunk UK
2.2.
IP Connect UK Access
IP Connect UK is an MPLS (Layer 3) WAN network with Quality of Service tagging.
One Voice SIP trunk UK can be used with IP Connect UK when the customer has a Virtual Private
Network (VPN) in which the various telephony sites are present. Additional connections will be
added to this VPN so that also the SIP Network is reachable.
Once the IP Connect VPN is in place a SIP order can be taken and trunks configured
2.3.
Ethernet Connect UK access
Ethernet Connect is a Layer 2 WAN network with quality of service tagging.
One Voice SIP trunk UK is supported only with wires only Ethernet Connect. Managed is not
supported.
One Voice SIP trunk UK can be used with wires only Ethernet Connect UK when the customer
has established an E-LAN (aka Ethernet Virtual Private Network, ETHN) and put the various
telephony sites using multi-cos onto that E-LAN.
Upon ordering One Voice SIP trunk UK, 8 additional EVC connections (aka Ethernet Dynamic,
ETHD) will be ordered on the customer’s behalf (and charged on the customer’s account). Once
that is in place the SIP order can continue and trunks configured.
2.3.1.
The SIP Trunk
A Trunk in the context of this service is defined as the connection between 1 – privately
reachable - IP address at the customer side and 2 (up to 4) SBC – private cloud reachable – IP
addresses in the SIP network. This connection is capable of handling a set number of
simultaneous calls.
The SIP Platform uses static registration, this implies that the originating IP address and the VPN
used by the IP PBX are hardcoded during the service provision and are used to identify which
Trunk the traffic is coming from.
In normal circumstances a PBX will map to exactly 1 Trunk and all traffic to/from the PBX will be
carried over that single Trunk. In case traffic needs to be separated over multiple trunks, then
the PBX should have the capability to work with more than one IP address.
It is important to note that when a customer wants multiple Trunks, he needs to have at least as
many IP addresses as he wants trunks as the IP address used to send the packets from is used to
establish the identity of the trunk (Static Registration).
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One Voice SIP Trunk UK
2.3.2.
Trunk Hierarchy
To enable flexibility, One Voice SIP trunk UK introduces the concept of Channels, Trunks and
Trunk Groups, which the customer can dimension.
A Channel is the capacity needed to carry a telephone call and is an attribute of both a Trunk
and a Trunk Group. The number of channels determines how big a Trunk or Trunk Group is.
A Trunk being a connection to 1 IP PBX has a designated number of Channels. This capacity is
also referred to as the trunk’s Call Admission Control (CAC).
A Trunk Group in turn also has a designated number of Channels and is a container for Trunks.
So Trunks sit in Trunk Groups and multiple Trunks can sit in the same Trunk Group.
Customers therefore have the opportunity to specify the capacity both at each individual Trunk
as well as for the overarching Trunk Group(s) as well as the mapping between Trunks and Trunk
Groups, providing them with flexibility to effectively manage their aggregated telephony traffic.
In setting the capacity at the Trunk Group lower than the sum of the individual capacities at
each of the underlying Trunks, the customer can effectively manage the aggregation of this
traffic and benefit from it.
A PBX/would be served by at least 1 Trunk (allowing a certain site-level Channel capacity), and
multiple Trunks can be grouped together in a Trunk Group (allowing a certain overall Channel
capacity). This way, it is possible to specify the behaviour of the call delivery to match the
capacity at each site and offer the features described further in this document.
This relationship between the Trunks, Trunk Groups, sites and the channels is further illustrated
below:
P
B
X
Data
VPN
Channels
Site A
Data
VPN
Trunk Group
Site B
Site C
Trunk
Figure 2: Trunk Mapping
As can be seen from the above picture, the Data VPNs ensure transport layer connectivity
between the sites and the SIP trunk platform. Logical connections (Trunks) are then created to
define connection between each site and the platform and specify both the throughput and the
calling capabilities at the access side of the Trunk. This way, BT can ensure that the calls offered
to any particular site will match the technical capacity available/intended for call handling at
that site.
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One Voice SIP Trunk UK
These Trunks are then grouped up to one or more Trunk Groups that have at aggregated level a
number of channels (a throughput defined) as well as specific calling capabilities (features). It is
at this aggregated number of Channels in a Trunk Group that BT charges the customer for the
One Voice SIP trunk UK service.
Using this hierarchy, the customer is in control of his telephony service and can define within his
network the Trunk Group throughput required in total, which can be smaller than the
throughput at each individual Site trunk connection, effectively managing the Grade of Service
(blocking probability) within his access network. Such dimensioning would typically be the result
of Traffic Engineering (Erlang calculations) for a given Grade Of Service (Blocking percentage) or
be the result of past experience the customer has on the peaks expected in his estate and what
he is comfortable with.
The Total amount of SIP trunk Channels defined at the Trunk Group level is the total capacity
available for making / receiving calls to /from the PSTN network.
In all cases, the number of channels available will be limited by the available bandwidth (on EF
class) defined on the access to each site, which needs to be engineered in advance. It is not
possible with IP Connect to dynamically burst outside the allotted EF bandwidth.
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One Voice SIP Trunk UK
3.
Main Features & Functionality
The main features and functionality of One Voice SIP trunk UK are described in this section,
while the managed service options available to customers are described in section 5 of this
document.
3.1.
Public Telephony service
One Voice SIP trunk UK is a genuine Public Telephony service. This means that the service allows
Customers to make/receive calls to/from other public telephony services (the worldwide phone
network). In addition, being a service offered in the UK, the service conforms to the regulatory
requirements dictated by Ofcom for such services. These are:

GC 3 Proper and Effective Functioning of the Network (network integrity & service
reliability) – must take all reasonably practicable steps to maintain, to the greatest extent
possible, network integrity and service reliability but only for the aspects of the network
that they control.

GC 4 Emergency Call – offer access through 999 and 112 short numbers to the emergency
services.

GC 5 Emergency Planning - requires that the communications provider on the request of
certain authorities makes arrangements for the provision or rapid restoration of such
communications services as are practicable and may reasonably be required in disasters.
Although strictly required of PATS services it is unlikely that this GC will have real force in
relation to SIP or VoIP services until such time as they widely displace traditional voice
networks.

In support of GC 8 Operator Assistance, Directories and Directory Enquiry Facilities and GC
19 Provision of Directory Information, the service ensures that the numbers used are
listed in BT’s (and therefore other CP’s) Directories and Operator Assistance and Directory
Enquiry facilities.

GC 10 Transparency and Publication of Information - shall ensure that clear and up to date
information on its applicable prices and tariffs (which for the avoidance of doubt shall not
include bespoke or individual prices and tariffs), and on its standard terms and conditions,
in respect of access to and use by end-users is published, in accordance with the
requirements of this condition. Typically BT GS meets this obligation by publishing
standard prices in the public price list and by referring to its standard terms and
conditions. This GC does not prevent BT GS from agreeing bespoke prices and conditions
and these do not require publication. One Voice SIP trunk UK being a service sold to large
companies, the prices and conditions would tend to be bespoke instead of published
rates.

GC 11 Metering and Billing - must ensure that the amounts stated in their bills represents
and does not exceed the true extent of any such service actually provided to the end-user
in question.

GC 13 Non-Payment of Bills - any measures taken by the communications provider to
effect payment or disconnection shall be proportionate and reasonable and notified in
advance.

GS 17 Allocation, Adoption and Use of Telephone Number and GC 18 Number Portability –
needs to provide number portability as soon as it is reasonably practicable on reasonable
terms, including charges, to any of its end-users who so requests.
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One Voice SIP Trunk UK
The General Conditions can be found in full at:
http://stakeholders.ofcom.org.uk/telecoms/ga-scheme/general-conditions/
3.1.1.
Emergency services
One Voice SIP trunk UK offers access to Emergency services in compliance to Ofcom General
Condition 4.
The access to emergency services consists of two services. Firstly the possibility to dial the
emergency services numbers (999 and 112), secondly the provision of sufficient information to
the emergency services centre so that even in the event of a silent call, the emergency
responders can be dispatched to the right location.
To provide location information a location will be stored to a telephone number entry as part of
every telephone number range that is provided for a customer. This telephone number, when
used as a calling line identity during an emergency call will then allow the emergency service
centre to look up the location registered for that number.
The emergency location information that can be stored is thus static and pre-provisioned. It is
not possible to dynamically provide location information, nor does the SIP protocol at the
moment have an agreed way to convey dynamic (user provided) location information.
As in VOIP the PBX can offer mobility services, it is therefore important to note that it is up to
the PBX to ensure that in the event of a call to an emergency service number the right calling
line number is used that corresponds to the location of the caller.
In the simplest cases, this would be the telephone number normally used by the caller. In more
complex customer scenario’s a set of telephone numbers would be provisioned specifically for
each location and the PBX programmed in such a way that in the event of emergency call, these
dedicated emergency location numbers are used.
3.2.
Centralised PSTN Break-in
On top of pure connectivity to the Public Telephony network (PSTN), the One Voice SIP trunk UK
service offers the feature of centralised PSTN Break-in. This is done by differentiating between
the – transport - capacity needed to connect each site to the platform (supplied by the relevant
data product) and the trunk – service - capacity needed in-globe to communicate to other
customers (commonly coined as break-in/break-out to the PSTN), allowing the customer to
procure trunks at aggregated capacity.
Being able to look at capacity across all sites gives economies of scale (due to non-linear effects
in traffic engineering) and therefore enables a customer to require less trunks than would have
been necessary using classic technologies, This effect please especially when more than 4 sites
of roughly equal size are involved. This aggregation effect plays to the advantage of our
customers and is one of the compelling features for this going to VOIP.
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One Voice SIP Trunk UK
In technical terms, this has to do with non-linearity in the Erlang formula that applies when
calculating the amount of channels needed to serve a particular number of users that randomly
receive telephone calls whilst providing a pre-set Grade of Service (blocking percentage) to
these users : 
Aggregating multiple sites – customer can decide to aggregate a large number of smaller
sites up to one total capacity holding typically 30% less channels whilst still providing the
users at each site with the same or superior grade of service he wants to offer to his end
users as before.
Channels
Trunk
PBX
Trunk Group
Figure 3: Aggregating Multiple Sites
Free resiliency - Dual Parenting – This is where One Voice SIP trunk UK works very differently to
legacy technologies. In legacy technologies twice the capacity was needed (and paid for) both
on access and on trunk level. With One Voice SIP trunk UK a customer buys sufficient access
capacity to each of the sites but buys only the total trunk channels effectively needed across all
the sites. This implements free resiliency option for the full traffic should one of the two sites
loose connectivity. This is more efficient than in legacy technologies where effectively twice the
capacity was needed both on access and on trunk level, whilst with One Voice SIP trunk UK the
trunk level capacity is only procured once.
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One Voice SIP Trunk UK
Figure 4: Dual Parenting
3.2.1.
Inherited IP Resiliency
In contrast to TDM technology that switches calls using the actual Telephone Numbers as
addresses, One Voice SIP trunk UK routes calls using IP addressing to which Telephone numbers
are mapped. This provides an additional level of resiliency, as IP addresses can be reachable
through a number of routes and therefore provides a better connectivity to the service as one
Telephone Number can be mapped to more than one IP address.
It has to be noted that when there are back-up routes designed with a significant lower capacity,
that this can have an impact on call quality when the primary route becomes unavailable, as the
capacity on these routes as well as the fact that they are used would be invisible to the SIP
Trunk service.
3.2.2.
Call Admission Control
The One Voice SIP trunk UK service has Call Admission Control at various levels.
At the trunk level, both Call and Bandwidth based call admission control are applied to ensure
that the offered voice calls do not exceed the amount of channels intended nor the available
bandwidth on that trunk.
At the trunk group level, a Call based Call Admission Control is applied to ensure that the total
offered voice calls do not exceed the amount of channels bought by the customer.
The Call Admission Control feature in One Voice SIP trunk UK is not directional, this means that
there is only one setting and this setting applies across both incoming and outgoing calls. This in
contrast to some ISDN30 implementations where, through configuration of the channel hunting
algorithm, some channels could be reserved for inbound traffic and others for outbound traffic.
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One Voice SIP Trunk UK
3.3.
Incoming Call routing options
Trunks forming part of the One Voice SIP trunk UK service can be configured to route incoming
calls to the customer in two ways, either Priority Based or Round Robin.

Priority Based, also known as Overflow, call distribution means that the subsequent trunk
in a trunk group is only used when the first trunk has reached the maximum call admission
as specified by the customer for that trunk.

Round Robin, or load share, call distribution means that the calls are alternating between
each of the trunks within that trunk group.
In specifying the incoming call routing options for each trunk group, customers can define in
detail how telephone calls should arrive to their estate.
For clarity, trunks that are inactive or out of service will be skipped by the trunk hunting.
3.4.
Differentiated Trunks
In contrast to ISDN30 where it is a one-size-fits-all, with One Voice SIP trunk UK we offer various
types of trunks have been defined, tuned to the use envisaged for them. This allows us to offer
appropriate cost points for each of the types. Levels of Trunks are available:

Normal – this SIP trunk provides the trunk throughput for normal business use, with peaks
& pits of utilisation throughout the day. This is engineered in the backbone of the SIP
trunking platform to offer a match to the present ISDN30 service.

Intense – these are the heavy-duty SIP trunks that are engineered inside the platform to
support a near to constant use, day-in day-out, a service level superior to what is presently
available with ISDN30 trunk telephone lines.

Dynamic – the Dynamic SIP trunk is available to cover short term increased capacity
without the drawback of a longer term contract commitment, and thus allow the business
to flex the calling capacity over time. Because these trunks are designed to cover peaks of
demand, they are technically the same as the Intense trunks, but available on short term
(minimum duration of a month) and on short notice (with a week’s prior notice).
The ability to deliver calls using this service is always limited to the constraints of the data
network (IP Connect UK) Access capacity the customer has installed
All channels in a trunk will be of the same type. A trunk group on the other hand can consist of a
number of trunks, each with their own type.
3.5.
Out of Area connection
Existing geographic telephone numbers, regardless of where in the UK they reside, can be
brought-in to the service, supplying at no extra charge an out of area connection functionality
which allows businesses to appear local whilst calls are being handled in a different location.
Check out the section on number portability as there are constraints as to which numbers can
be ported-in (from other Communication Providers) at present.
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One Voice SIP Trunk UK
4.
Supported Endpoint devices
All devices connected to the service (IP-PBX or SBC) must conform to the SIP Interface
specification that is issued by BT and specifies the concrete working of the SIP Protocol for the
choices that BT has made.

BT continues to test all types of manufactures PBXs supporting a wide variety of types with
associated manufacturer accreditation for interworking. The list of confirmed PBX
continues to grow and includes; Alcatel – OmniPCX Enterprise Rls 10.0.

Audiocodes – Mediant 1000 6.00.X.X.

Avaya – CM6-SM6.1; CM5.2-SM6.1; CS1K 7.5-SM6.1, with or without SBC (ACME, SIPERA,
AASBC).

Cisco Call Manager – CUCM8.6; CUCM8.5 (1); CUCM8.0.3; with or without CUBE8.8 - 8.5CM8.0.3, CCME 7.1 and CVP8.5 (UCCE), CUCM9.1 with/without CUBE.

Microsoft – MS Lync 2010 when connected with an Audiocodes Mediant gateway. MS
Lync 2013 with Audiocodes, ACME or Sonus SBC.

Mitel – Mitel 3300.

Siemens OSV – Openscape v5, v6, v7 with/without Fortigate FW with ACME, SIPERA,
Openscape SBC ACME SBC.

Wicom – SAP BCM Call Model.
There is an ongoing program to onboard other PBX. This program takes input from customer
demand to drive the test schedule.
4.1.
Supported Codecs
One Voice SIP trunk UK is a VOIP service, which implies that the conversation is digitally
encoded for transport over the IP network this encoding is done using Codecs. There are a
currently a large number of codecs available in the market.
For on-net traffic between sites of the customer as well as for traffic between customers on the
network, the SIP platform will allow the end-points to negotiate the appropriate codec for the
call directly and will not restrict the types of codec used, nor does the platform offer
mediation/transcoding of the codec for these calls.
Therefore, for all customers to be able to talk to any other customer on the platform, at the
minimum the PBX of each customer needs to support G.729a. This is the so-called Codec of last
resort.
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One Voice SIP Trunk UK
For the calls handed from/towards the UK PSTN, the customer has the choice between the
following codecs:
G.711 A-Law – This ITU codec dating back to 1972 is the encoding used in the national
PSTN network as all voice conversations are already digitally transmitted. Using this codec
therefore implies that the voice samples do not need any transcoding until they reach the
distant telephone exchange at the far end. This codec has a payload of 64 kbit/s, but
transporting this over IP clear requires 87.2 kbps (20 ms sample, 160 byte payload and 50
pps - packets per second).

G.711 µ-Law – Same as above, but voice samples are transmitted in a different sequence,
used in the U.S.

G.729 annex A – This ITU codec from 1996 uses audio compression and operates at 8
Kbits/s, transporting this over IP Clear requires 23.2 kbps (30 Ms samples, 30 byte payload
and 33 pps). This codec gives a decent voice quality, but automatic voice response units
may struggle with it. Given that the compression is a glossy compression, care needs to be
taken not to have multiple transcoding hops using low bit rate codes in a call. Note that
G.729 annex B is not supported.

iLBC - Internet Low Bitrate Codec is a narrowband speech codec defined in RFC 3951. The
algorithm is a version of block-independent linear predictive coding, with the choice of
data frame lengths of 20 and 30 milliseconds. iLBC handles the case of lost frames through
graceful speech quality degradation. iLBC-encoded speech frames are independent and so
lost frames don’t propagate their impact on voice quality, by using controlled response to
packet loss, delay and jitter. The codec uses a fixed bitrate (15.2 kbit/s for 20 Ms frames,
13.33 kbit/s for 30 Ms frames) and a fixed frame size (304 bits per block for 20 Ms frames,
400 bits per block for 30 Ms frames). The sampling frequency is 8 kHz/16 bit (160 samples
for 20 Ms frames, 240 samples for 30 Ms frames).

G.726 is an ITU-T speech codec standard covering the transmission of voice at rates of 16,
24, 32, and 40 kbit/s. It is primarily used on international trunks in the phone network.
Sampling frequency 8 kHz. This codec generates a bitstream, therefore frame length is
determined by packetisation (typically 80 samples for 10 Ms frame size) Typical
algorithmic delay is 0.125 Ms, with no look-ahead delay. G.726 is a waveform speech
coder which uses Adaptive Differential Pulse Code Modulation (ADPCM).
1.1.1
Fax Support
The One Voice SIP trunk UK service supports the following in dealing with fax (and expects
compatible CPE to also support these)
T.38 Fax Relay, as per ITU T.38 Annex D standard
G.711 pass-through, in which case the platform expects from the CPE support for G.711 pass
through of fax modem signals, with the ability to disable echo cancellation and dynamic jitter
buffers on a per call basis.
G.711 upspeed, in which case the platform expects the CPE to have the ability to upspeed from
G.729 to G.711 if a fax tone is detected, using a SIP re-INVITE mechanism.
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One Voice SIP Trunk UK
4.1.1.
Bandwidth calculations
G.711, G.729 and iLBC codecs are so-called Constant Bit Rate codecs, implying that the
bandwidth consumption is always the same and thus predictable, allowing excellent engineering
of the customer LAN/WAN as well as the platform internally.
For the calculation of the bandwidth required to carry the telephony traffic, the following
formula can be used:

Total frame size = Ethernet header (18 bytes) + IP/UDP/RTP (normal 40, compressed =
2 bytes) + payload

PPS = codec bit rate / voice payload size

Bandwidth = total frame size * PPS
Please note that in ordering EF bandwidth, the Ethernet header does not need to be taken into
account, as EF is a Layer 3 protocol. In dimensioning the – physical – access bearers, the Layer 2
size does play a role.
25 kbps is typically used as dimensioning for G.729a and 100 kbps for G.711.
4.1.2.
Codecs and Voice Quality
Voice Quality is something that is experienced fully end-to-end: from the receiver in one room
attached on the one end to the receiver in another room at another location at the other end.
Many elements therefore influence Voice Quality, most are out of control of this service, such as
packet loss/jitter on the LAN or on the WAN, transmission issues in the PSTN to the distant end.
Voice Quality is expressed by Mean Opinion Score or MOS. MOS tests for voice are specified by
ITU-T recommendation P.800.
The MOS is generated by averaging the results of a set of standard, subjective tests where a
number of listeners rate the heard audio quality of test sentences read aloud by both male and
female speakers over the communications medium being tested. A listener is required to give
each sentence a rating using the following rating scheme:
Mean opinion score (MOS)
MOS
Quality
Impairment
5
Excellent
Imperceptible
4
Good
Perceptible but not annoying
3
Fair
Slightly annoying
2
Poor
Annoying
1
Bad
Very annoying
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The MOS is the arithmetic mean of all the individual scores, and can range from 1 (worst) to 5
(best).
Compressor/decompressor (codec) systems and digital signal processing (DSP) are commonly
used in voice communications, and can be configured to conserve bandwidth, but there is a
trade-off between voice quality and bandwidth conservation. The best codecs provide the most
bandwidth conservation while producing the least degradation of voice quality. Bandwidth can
be measured quantitatively, but voice quality requires human interpretation, although
estimates of voice quality can be made by automatic test systems.
As an example, the following are mean opinion scores for one implementation of different
codecs:
Data rate
[kbit/s]
Codec
Mean opinion score
(MOS)
G.711 (ISDN)
64
4.30
iLBC
15.2
4.14
G.726
32
3.79
G.729a
8
3.7
GSM FR
12.2
3.5
Note actual MOS experienced in the network will be different from the table as it depends on
the implementation of the codec and the material used end to end. However it gives a relative
position of how the codecs compare with each other.
Note that Wideband codecs are not comparable here. At their frequency range of 7 kHz to even
14 kHz, Wideband codecs give a much different user experience, including transfer of music in
close to Compact Disk quality (22 kHz). As opposed to 3.1 kHz for baseband voice that is the
reference in the above table.
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One Voice SIP Trunk UK
5.
Connection to the Service
5.1.
Connection to the Service through IP Connect UK
When the service is accessed through IP Connect UK and requires the Customer’s PBX to have a
static IP address (public or private) reachable through the data VPN as that is used to identify
the originating PBX within the customer’s VPN.
5.1.1.
SIP connectivity
Customer will be offered 2 to 4 RIPE IP addresses of A-SBC (of which the primary & secondary
are geographically diverse) to route SIP requests and media to. It is the responsibility of the
Customer (or his PBX maintainer) to program the IP PBX in such a way that in case the primary
SBC is not reachable outgoing calls are being pointed to the secondary SBC and that incoming
calls from all SBC’s are accepted.
The IP addresses used in the SBC’s are addresses from the public IP address space, so that there
will be no clashes with customer VPN addressing is possible and configuration is
straightforward.
Frequent (30 sec) polling by the A-SBC of the customer PBX acts as a health-check so that the
network knows which connections are still available for traffic. Likewise, the customer PBX can
learn from polling which A-SBC is still running and direct its outgoing calls to a working A-SBC.
This polling is done through sending out SIP OPTIONS messages and should be responded to by
200 OK messages.
In the event of the failure of an A-SBC, ongoing calls would be interrupted and end-users can redial in which case the PBX should use the secondary A-SBC. For incoming calls, the Call Server
inside the network is doing the polling to the A-SBC to ensure call signalling is happening to the
correct A-SBC. It is therefore important that the Customer PBX accepts incoming calls from both
(up to 4) A-SBC’s.
5.2.
Connection to the Service through Ethernet Connect UK
When the service is accessed through Ethernet Connect UK it requires the Customer’s PBX to
have a static IP address (public or private) reachable through the ELAN as that is used to
identify the originating PBX within the customer’s VPN.
5.2.1.
SIP connectivity
Customer’s network will be extended with 8 EVC’s, across 2 sites, 2 equipments per site and
separating signalling from media. The Customer will be offered 2 RIPE IP addresses of the A-SBC
(which are geographically diverse) that are on each site in High Availability configuration to
route SIP requests and media to. It is the responsibility of the Customer (or his PBX maintainer)
to program the IP PBX in such a way that in case the primary SBC is not reachable outgoing calls
are being pointed to the secondary SBC and that incoming calls from all SBC’s are accepted.
The IP addresses used in the SBC’s are addresses from the public IP address space, so that there
will be no clashes with customer VPN addressing is possible and configuration is
straightforward.
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One Voice SIP Trunk UK
Frequent (30 sec) polling by the A-SBC of the customer PBX acts as a health-check so that the
network knows which connections are still available for traffic. Likewise, the customer PBX can
learn from polling which A-SBC is still running and direct its outgoing calls to a working A-SBC.
This polling is done through sending out SIP OPTIONS messages and should be responded to by
200 OK messages.
In the event of the failure of an A-SBC, ongoing calls would be interrupted and end-users can redial in which case the PBX should use the secondary A-SBC. For incoming calls, the Call Server
inside the network is doing the polling to the A-SBC to ensure call signalling is happening to the
correct A-SBC. It is therefore important that the Customer PBX accepts incoming calls from both
(up to 4) A-SBC’s.
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One Voice SIP Trunk UK
6.
Telephone Numbers
A public telephone service goes hand in hand with getting Telephone Numbers on the service.
These could be existing working numbers but also new Telephone Numbers out of BT’s vast pool
of spare geographic telephone numbers. Before starting the ordering process for BT One Voice
SIP trunk UK, one must ensure to make the necessary arrangements for the Telephone Numbers
that finally need to be configured on the service.
There are two ways to get Telephone Numbers on the One Voice SIP trunk UK service:

Allocation of New Numbers, and

Bringing over (porting or transferring) existing Telephone Numbers.
Both these ways are described hereafter. Once the numbers are clear and their bringing across
reserved, one must also design how the telephone numbers map to the SIP trunks. This is
described further below.
6.1.
New Number provides
The One Voice SIP trunk UK service supports the supply of new numbers out of BT’s total pool of
available telephone numbers.
The process by which these can be applied for is documented on the product intranet pages and
consist of

Filling out Number DCF to capture all information, including the required Directory Entry.

Reserving the number at which time the number gets pre-ported to the SIP platform.

Entering the number & its CSS reference onto the SIPCRF.

Submitting the SIPCRF for fulfilment as per BAU process.
For new number provides, the Number gets pointed to the SIP platform in advance of the
Trunks being built. The lead times for New Number provides are therefore much shorter than in
the case of porting over existing numbers.
6.2.
Number Block Transfer
Customers that have entire 10,000 Number blocks can have their number blocks be brought
over in their entirety. This basically re-homes the number block to the VMP platform in BT’s
network and is a good deal for the losing CP as otherwise they pick up the costs of routing the
traffic onwards to BT whilst not getting the related revenue.
Such requests for Number Block Transfer need to be negotiated specifically with the losing CP
and specific lead times apply.
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One Voice SIP Trunk UK
6.3.
Number Portability and Number Transfer
Whilst inside BT we make the distinction between bringing numbers into BT from other
operators (Number Portability) and pointing numbers to a different network/platform inside BT
(Number Transfer). From the point of view of the customer, this is the same experience of being
able to retain an existing number whilst signing up for or going away from SIP Trunks.
Number Porting Request (NPOR)
Number Porting Request (NPOR) is the process by which agreement is given from a CP or BT to
port a number (or range of numbers) to BT at a certain date and time. BT uses the same process
internally for BT to BT transfers as when it is between BT & CP.
NPOR requires a porting request form (the NPOR form) to be filled out.
Experience has learned that NPOR’s often get rejected and not much information is given to be
able to submit a new one. A guide with the best practise and hands-on learning is available in
the library on the product intranet pages.
Below are the theoretical lead times (attempts to submit shorter get automatically rejected, but
once accepted, there is a date change process to attempt to agree a shorter implementation
time)
Installation type - Minimum Order Lead-time
Port Leadtime in
working
days
Single Line
Single Lines with greater than 10 Lines porting at same
installation / time
Multi Line (30 lines / numbers or less)
Multi Line (31 lines / numbers or greater) – Simple DDI
Complex DDI
Sub Port
Lead-time
in working
days
4
7
14
17
7
17
22
10
20
25
Note a Sub Port applies when the number is already ported before, stands for Subsequent
Porting
6.4.
Mapping Telephone Numbers to Trunks
Customers have to map the ranges of Telephone Numbers either to chosen trunks or to chosen
trunk groups. This mapping defines how incoming calls to the Telephone Number will be
handled by the SIP Platform.
When a number is mapped to a Trunk, the incoming traffic to that number will only be
presented to that Trunk.
When on the other hand a number would be mapped to a Trunk Group, then all the incoming
traffic to that number will be applied the trunk hunting algorithm (round robin or priority) to
select dynamically which trunk from that trunk group would be most appropriate to receive the
next incoming call.
This flexibility will allow a customer to specify the call routing behaviour for incoming calls at a
logical level (how should calls to this number behave in my solution) instead of on
physical/technical level.
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One Voice SIP Trunk UK
In addition to mapping the Telephone Numbers to Trunks or Trunk groups, the customer can
specify some features like incoming call divert, incoming CLIR for the trunk
Please note that the Telephone Numbers will still be treated as Ranges, implying that the
features apply to the entire range.
6.5.
Incoming Call Routing
Customer can specify how the calls are to be routed into his estate. He can specify multiple
Trunks and group Trunks together in Trunk Groups. For all trunks in a given trunk group, there
are two types of routing possible: Priority Routing (all calls go to the first the first trunk, then to
the next, etc.) Round Robin (calls alternate between the trunks, first call to 1st trunk, 2nd call to
2nd trunk etc.). Round Robin is ideally suited for load sharing situations, where-as Priority
Routing is more suited for fall-back situations (main site/back-up site) or overflow conditions.
6.6.
Other Calling Features
Being a Trunking service, the One Voice SIP trunk UK service is stripped down from user related
features that can be provided on other platforms. So for features such as Music On Hold, Voice
Mail, etc. The customer has the option to buy a Hosted Voice service or to provide these from
within the customer’s PBX.
6.7.
DDI (Direct Dialling In)
One Voice SIP trunk UK currently provides a DDI functionality in that it allows customer to
specify how many digits of the dialled number should be presented to the customer PBX. When
DDI is activated, up to the last 6 digits can be specified as required by the PBX to do internal
routing to the individual end users.
6.8.
DNIS (Dialled Number Identification Service)
DNIS is supported by the SIP Platform but is not yet supported by UK Inbound Services.
Calls to non-geographic numbers are translated before they reach the platform serving One
Voice SIP trunk UK. As such One Voice SIP trunk UK by default provides the translated number as
the dialled number to the customer PBX/ACD.
There is currently no agreed SIP standard to convey the originally dialled number and carriers
have implemented various ways to circumvent this problem, using the ‘To’ header, or even
vendor bespoke fields. All these implementation have drawbacks in that they don’t work for all
call scenarios or for all PBX/ACD.
BT have implemented the option to use special network routing numbers at the point of
translation so that both a customer chosen reference to the Dialled Number and the location
can be transported over to the SIP platform and that the chosen reference to the Dialled
Number can be presented to the customer PBX/ACD. The numbering for this aligns to Global
Inbound Services DNIS feature that is also using a 7 digit Site Location Code (unique across the
platform) and a 4 digit DNIS (chosen by the customer). However, at the time of this writing the
OSS for the UK Inbound Services still needs upgrading to be able to consume this feature.
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One Voice SIP Trunk UK
Awaiting this permanent solution, an alternative DNIS implementation can be provided by
mapping each dialled number to a dedicated geographic number (in a range) and using the DDI
feature to only present the last few digits from that number.
6.9.
Overflow, call diversion, BT Smartnumbers
One Voice SIP trunk UK supports the following overflow / call diversion mechanisms:

Static call diversion on Busy / Error:
This can be set on a per DDI Range basis. In this case for incoming calls, when the
customer PBX is signalling that the destination is either Busy or is experiencing an Error,
then the SIP Platform will route the call to the alternate destination number (1 number for
the entire DDI range). This destination should be a valid PSTN reachable number.
The One Voice SIP trunk UK service is compatible with BT Smartnumbers, so that
customers, in setting the call divert on the DDI’s to their BT Smartnumber, can benefit
from alternate dial plans for all their office workers, managed directly in the cloud.

Static call diversion Unconditional:
This can be set on a per DDI Range basis. In this case for incoming calls then the SIP
Platform will immediately route the call to the alternate destination number (1 number for
the entire DDI range). This destination should be a valid PSTN reachable number. In this
case the incoming call doesn’t even reach the trunk.
Also in this case, a BT Smartnumber can be used as Diverted-To number, so that
customers, in setting the call divert on the DDI’s to their BT Smartnumber, can benefit
from alternate dial plans for all their office workers, managed directly in the cloud.

Trunk Priority Routing:
As described earlier in this document, DDI Ranges can be associated to the Trunk Group
instead of to the Trunk. In that case, there can also be specific hunting algorithm applied
to the Trunks in the Trunk Group, so that calls first go to the primary trunk and when that
one is maxed out in capacity (or not reachable), then calls overflow to the second trunk in
the trunk group.
Note that in that case only SIP trunks can be part of the hunting. Overflow to non SIP trunk
destinations is not supported.
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One Voice SIP Trunk UK
7.
Service Wrap
The following describes the service elements provided as part of One Voice SIP trunk UK.
7.1.
Service Options
Customers will by default be provided with a Prompt Care level of service. Customers can chose
a higher service level for this service by opting for Total Care for an additional charge. The
services delivered are:

Prompt Care:
Fault Reporting - Customers will be able to report faults Monday to Sunday24/7 including
bank and public holidays. BT will respond within 4 working hours of receipt of a fault and
start remote diagnostics. Response will include the confirmation that the fault is accepted
and the provisions of a fault reference number. BT will advise the customer of the
progress being made to clear the fault. Where applicable, customers can request the
option to divert their line to an alternative number.
Fault Repair - BT will fix faults by the end of the next working day (up to 23:59 hours).

Total Care:
Fault Reporting - Customers will be able to report faults Monday to Sunday 24/7 including
bank and public holidays. BT will respond within 4 working hours of receipt of the fault
and start remote diagnostics. Response will include the confirmation that the fault is
accepted and the provision of a fault reference number. BT will advise the customer of the
progress being made to clear the fault. Where applicable, customers can request the
option to divert their line to an alternative number.
Fault Repair - BT will fix the fault within 24 hours of the fault being reported (unless the
customer has elected for an appointment outside of this time).
7.2.
Service Centre
BT’s Incident Management process will incorporate the following:

Incident Management.

Problem Management.

Known Error tracking.

Request for Change (RFC).

Resolution.
BT will keep the Customer informed of the progress of the fault investigation.
The BT Help Desk will retain overall ownership of all incidents until the incident is cleared and
service restored.
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One Voice SIP Trunk UK
7.2.1.
Customer Helpdesk
The Customer will be responsible for providing its own internal helpdesk (the ‘Customer
Helpdesk’) which will be:

At the discretion of the customer available for the service hours the customer wants to
offer its end-users. (this could for example be 24x7x365 or working hours only).

Familiar with the customer’s telephony and IP Data network estate.

the first point of contact for all its users.

responsible for dealing with day to day management queries relating to users.

Responsible for first diagnostics before dispatching incident to second line (the BT
Helpdesk).
The Customer will provide BT with contact details for a minimum of 2 personnel (and a
maximum of 4) who will be authorised to contact the BT Helpdesk (GS Front Desk) on behalf of
the Customer.
Note, should the customer not provide a 24*7*365 internal helpdesk, BT will not handle the
customer’s out of hours first line calls as part of this service. Specific services from BT may exist
to cover this functionality from BT’s vast portfolio of services.
7.2.2.
BT Helpdesk (GS Front Desk)
BT’s established and operated single fault reporting helpdesk serving the Customer in question
will be dealing with the fault reception, making the already established means of service related
communication and fault reception hours available also to One Voice SIP trunk UK related
incidents.
The Front desk will through Structure Questions try to classify in which domain the fault is and if
the fault is deemed to be with One Voice SIP trunk UK, dispatch the case off to a One Voice SIP
trunk UK specific 2nd line desk.
BT will respond within 4 working hours of receipt of a fault report. If the fault is not cleared
during this period, BT will advise the customer of the progress being made to clear the fault via
the contact. If staff are available BT may, at the customer's request, continue to work on a fault
reported under Prompt Care conditions outside working hours without a break. BT may make
additional charges. The BT Helpdesk will:

Provide support to the Customer’s nominated personnel.

Deal with faults reported by the Customer.

At BT’s discretion, escalate any unresolved issues or faults to its own 2nd line support desk and
3rd line technical support team where necessary. These helpdesks may request support from
equipment suppliers; and

Handle general enquiries from the Customer’s nominated personnel including planned work,
request for information, billing and orders.
The language of the BT Helpdesk will be English.
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One Voice SIP Trunk UK
7.2.3.
Planned maintenance
From time to time, BT may schedule maintenance work planned in advance to be carried out by
BT or on behalf of BT that may cause the Service to be interrupted or suspended (‘Planned
Maintenance’ or ‘Planned Engineering Works (PEW)’). Where possible, Planned Maintenance
will be during low usage periods outside Working Hours.
Before doing so BT will give the Customer as much notice as possible, and whenever practicable
will agree with the Customer when the Service will be suspended. This also applies to any
planned maintenance by 3rd party suppliers.
Emergency maintenance, updates, and other procedures will be scheduled by BT on a case-bycase basis, and notice will be given to the Customer where practicable.
BT will provide a rollback procedure as standard for all planned maintenance.
7.3.
Proactive Management
All servers within the One Voice SIP trunk UK service are connected to a dedicated management
network and will be monitored and managed by BT using the standard set of tools and
techniques. BT will provide 24 hour in-band monitoring of the server hardware, and network
infrastructure and will take necessary actions if a fault is identified, informing customer when
they are impacted by it.
7.4.
Billing
7.4.1.
OneBill
The customer’s first bill will include set up charges for trunk and channel configurations, and the
first month’s charges for the service, in line with those agreed with the customer when the
contract was completed.
Subsequently the customer will be provided with a bill for the service(s) purchased from this
service in accordance to its contracted bill frequency.
If the customer has any existing services from BT, these charges as well as One Voice SIP trunk
UK charges can be added to an existing monthly bill, using OneBill.
Customers of One Voice SIP trunk UK must have an active OneBill; this is for fraud protection
reasons on reverse call fraud.
The bill is itemised, at the product level (e.g. Trunk, DDI, etc.),
7.4.2.
BT Billing Analyst Converge
The OneBill containing One Voice SIP trunk is available through BT Billing Analyst Converge.
Using this tool, customers and BT people can drill down on the spend on the bill to the detail of
individual calls.
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One Voice SIP Trunk UK
7.5.
Reporting
A comprehensive set of 5 reports is available both to customers as well as to internal BT people
dealing with the customer.
These reports can be accessed internally by going to GSEDW and by customers by logging into
GS Portal.
Please note that access to the reports is restricted. In order to obtain access, a request must be
sent to GSEDW team for internal people or to GS Portal team for setting up customer for
reports. More detailed information how-to is available on the product intranet pages.
The reports are:

Weekly Call report

Incoming Call report

Outgoing Call report

Trunk utilisation by Day report

Trunk utilisation Hour by Day report
Some of the reports are showing the Peak Utilisation, this is the true peak as measured at that
time by the Call Server in the core of the network. This is contrast with other implementations
where such peaks would be statistically obtained using some count in a time interval on call
count or call duration, not giving the true peak.
7.6.
Service Level Management
This section explains the BT definitions for Service Level Agreement (SLA) and details how the
One Voice SIP trunk UK Service SLA works.
Ref P1 – P5 definition & SLA T2R/ L2C Excel doc.
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One Voice SIP Trunk UK
7.6.1.
Service Level Agreements
The One Voice SIP trunk UK is capable of supporting the following service levels for platform and
call quality.
Service attribute
Service level Target
Voice Platform Availability (VMP-21C up to
and including the interconnect to IP Clear and
the interconnect to the PSTN/ISDN). This
excludes the IP Clear WAN as well as the
PSTN/ISDN network
99.99%
Voice Service Grade of Service (VMP)
99.00%
Platform capacity shall be engineer so that
call attempts shall be carried successfully to
ensure that no more than
Theoretically one call in 100
(1:100) would have to
queue
PSTN/ISDN network availability is
IP clear itself has an availability of, depending
on the access technology and resiliency
options chosen by the customer for his WAN
7.6.2.
Lead times
7.6.2.1.
Provision
One Voice SIP trunk UK is an application delivered over an IP infrastructure.
The Timelines below are assuming the IP clear network of the customer has sufficient
bandwidth to support the new/changed requirement. Any upgrades on IP clear should have
been completed before the One Voice SIP trunk UK modify order. The above moves and changes
timescales are for One Voice SIP trunk UK platform configuration only and also do not involve the
fitting of LAN CPE or any other customer located Hardware.
The Term "Normal Working Hours" is defined as 08.00-18.00, Monday-Friday inclusive.
The Term "Working Days" is defined as Monday-Friday inclusive.

For adding or deleting SIP Trunks to an existing site, the lead time is 13 working days.

For adding SIP Trunks to a new site, the lead time is 13 working days.

For adding or deleting DDI number ranges, the lead time is 13 working days.

For porting additional numbers in, the lead time is 13 working days.

For changing the level of existing trunks (Sporadic, normal, intense, dynamic), the lead
time is 6 working days.

For adding a hitherto unsupported IP PBX, the lead time will be decided by advance
agreement between the customer, their BT Account Team, and the One Voice SIP trunk UK
Product Line, and will depend on the amount of effort required to test and certify the IP
PBX for supportability.
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One Voice SIP Trunk UK
7.6.2.2.
Service Availability and Restoration Figures
The overall end-to-end One Voice SIP trunk UK service will have annual availability above 99.9%,
and a few Customer sites will have annual availability below 99.9% due to the availability of the
access network. It is important customers understand that, as there will be a distribution of
fault duration, WAN resilience is highly recommended as a way of reducing the impact of any
outage on the customers strategic sites.
BT One Voice SIP trunk UK Service SLA covers BT’s performance in fixing faults. Transients by
their nature are very short lived; other faults will be infrequent and fixed quickly by BT. BT's will
make best endeavours that faults will not be frequent or of long duration and that circuits will
be delivered to the CDD is covered by an SLG scheme which is set out in the following subsections.
The following exclusions apply to the SLA’s

Faults on customer equipment outside of the service boundary.

Faults on network equipment due to customer action.

Faults reported by customer not observed/confirmed by BT.

Disruptions occurring within pre-notified engineering works window.

Failure of access from suspension of service for breach of contract by the customer, or
matters beyond BT’s reasonable control.
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