S I P T R U N K I N G S E R V I C E D E S C R I P T I O N

S I P T R U N K I N G S E R V I C E D E S C R I P T I O N

Contents 1. 2. 3. 4. 5. 6. What is SIP Trunking... 4 The Product Overview... 4 Channels & Trunks... 5 Channels... 5 Differentiated Channel Types... 6 Dynamic Capacity... 6 SIP Channel Configuration... 7 Channel Aggregation... 7 Priority Call Routing... 7 Round Robin Call Routing... 8 Call Admission Control (CAC)... 9 Call Attempts per Second (CAPS)... 9 SIP Trunking Features... 9 Call Diverts... 9 Anonymous Call Reject... 9 Incoming Call Barring... 10 Outbound Call Barring... 10 Numbering and number features... 10 Number Management... 10 New Numbers... 10 Ported Numbers... 10 Emergency Services Database Updates... 11 Calling Line Identity (CLI)... 11 Calling Line Identity Restriction (CLIR)... 12 CLI Format Options... 12 Screening Lists... 12 Class 1-5 CLI Support... 12 Override Number Range... 13 Long Duration Calls... 14 Equipment... 14 Service Overview... 15 Virgin Media Business Bandwidth... 15 Bandwidth Calculations & Considerations... 16 Technical configurations... 16 Platform Components... 16 External use Page 2 of 26

Supported Codecs... 17 End User Firewall Security... 18 CPE Protocols and Ports Overview... 18 Transcoding... 19 Hosted NAT Traversal... 19 Fax, PDQ & Alarm Support... 19 7. 8. 9. 10. 11. Customer Responsibilities... 20 Ahead of sale... 20 Ahead of deployment... 21 Day of deployment... 22 In-life service... 22 Order Journey... 23 Order Pack... 23 Provisioning... 24 Data Services... 24 Number Porting Process - Import... 25 Number Porting - Export... 25 Customer Service... 25 Fault Management Centre (FMC)... 25 Billing... 26 MACC's... 26 External use Page 3 of 26

1. What is SIP Trunking SIP Trunking is an IP based voice solution utilising the SIP standard. SIP Trunking enables customers to make secure IP based calls, usually originating from an IP phone or desk-based client (soft phone), to the PSTN (Public Switched Telephone Network) and vice versa. Over the past few years the technology has started to replace traditional voice services such as ISDN, as it enables customers to consolidate both their voice and data telecommunications over a single network connection e.g. a WAN (Wide Area Network) rather than having to deploy dedicated TDM (Time Division Multi) voice circuits. This converged approach typically leads to cost savings, which makes SIP Trunking a very attractive product if you are looking to consolidate or reduce your communications spend. SIP ( Session Initiation Protocol ) is an open standard which is being universally adopted by communications hardware and software vendors because it enables them to standardise their technologies and interoperate with other products and services more easily. 2. The Product Overview SIP Trunking is a cost effective and highly resilient business call and line service that offers you a logical migration path from traditional TDM based voice services such as ISDN. Through our SIP Trunking offering, Virgin Media Business can offer a dynamic, feature rich and flexible service for our customers. The following functionality is available as part of this SIP Trunking service: PSTN Break-in and Break-out make and receive calls from the Public Switched Telephone Network (national, mobile, international). Number Porting transfer your existing phone numbers to numbers from other TDM or SIP carriers. New Numbers order new numbers for any exchange area in the UK. Geographic Numbering Freedom configure numbers on a Trunk from any UK area code without the geographic restrictions of ISDN. Class 1-5 CLI use presentation numbers for outbound calls that are not hosted on the Trunk (using Number Screening Lists). Call Admission Control CAC is configured for both available bandwidth and Channels purchased. Channel Aggregation buy only the aggregated capacity needed across multiple Trunks. This means in practice that fewer channels need to be bought to obtain the same grade of service compared to an equivalent ISDN implementation. Round Robin Call Distribution load-share calls across Trunks within a Trunk Group. Priority Based Call Distribution specify main and back-up Trunks within a Trunk Group. 3 Core Channel Types giving you the flexibility in your service offering. Dynamic SIP Channels flex Channel capacity in line with seasonal demand. Trunk Call Barring features prevent certain call types from taking place. Call Diversion divert incoming calls to different destinations based on availability of the Trunk or permanently. Emergency Call Handling ensure 999 calls are treated correctly and that appropriate address information is displayed to the emergency operator for each number. External use Page 4 of 26

3. Channels & Trunks SIP Trunking offers the concept of Channels, Trunks and Trunk Groups. Channels A Channel allows one call to be made at any one time. The number of Channels required for a deployment is based on the maximum number of simultaneous calls that will be generated by the PBX. A Trunk provides a logical connection, over whichever access medium is deployed, from the End Customer PBX to the SIP platform. Each Trunk must be identified with a unique IP address. One or more channels are configured within a Trunk. A Trunk Group contains one or more Trunks. Trunk Groups have a number of features including resilience by offering round robin or priority based distribution of inbound calls between Trunks. Channels are also configured against the Trunk Group allowing aggregation of channel capacity for multiple Trunks within a Trunk Group, so you only pay for the throughput needed. We only charge for Channel capacity at the Trunk Group level and the resilience options are included in that Channel price. The relationship between Channels, Trunks and Trunk Groups is shown in the diagram below: External use Page 5 of 26

Differentiated Channel Types Where traditional ISDN voice services offer a single channel type for a one size fits all service type, SIP Trunking has more flexibility and offers 3 channel types. Each has been defined for a specific traffic use and offers differentiated pricing to suit. Basic this type of SIP Channel is suited for occasional use, typically smaller satellite offices or home office locations that don t produce a constant flow of telephony traffic. This is engineered to match traditional ISDN2 services and has a similar 12:1 contention through the platform. Standard this type of SIP Channel provides the throughput for normal business use, with peaks & troughs of utilisation throughout the day. This is engineered in the backbone platform to align with traditional ISDN30 services and has a similar 4:1 contention through the platform. Premium these are the heavy-duty SIP Channels that are engineered inside the platform to support a near to constant use, day-in day-out, a service level superior to that currently available with ISDN30 lines. A 1:1 contention is applied through the platform. Please note that monthly rental charges can vary on the type of channels taken. The ability to deliver calls using this service is always limited to the constraints of the data network access capacity provisioned as detailed further section Scalability and Capability. The following can be used as a guide only. Actual traffic profiles should be used to assess Channel type and number of Channels required. ISDN2 services with 2, 4, or 6 channels can typically be replaced using Basic Channels. ISDN30 services with e.g. 8, 15, 20 or 30 channels can typically be replaced using Standard Channels (quantity dependent on Call Attempts per Second required). ISDN30 used in contact centres with diallers can be replaced with Premium Channels (quantity dependent on Call Attempts per Second required). Where multiple sites with individual PBXs each with individual ISDN2 or ISDN30 lines are being migrated to a centralised PBX with centralised SIP Trunking, Premium Channels may also be needed. Dynamic Capacity In addition to the three channel types detailed above, SIP also offers a Dynamic SIP Channel option. It is designed to deal with short term demand for additional capacity allowing customers to flex capacity when needed, to suit a seasonal calling campaign for example. Dynamic channels can be provided for up to 50% of the SIP channel capacity. For example, if a customer has 100 premium channels they will be able to take up to 50 dynamic channels. A lower monthly rental is charged for dynamic channels whilst they are deactivated. As and when required the dynamic channels can be activated at the quantity you wish, it doesn t have to be all dynamic channels available. For the period of time the channels are active the rental charge will increase to that of the SIP channel rental for the primary rental change, basic, standard or premium. External use Page 6 of 26

SIP Channel Configuration Channel Aggregation As advised earlier, Trunks are configured within a single Trunk Group and channels are configured against both the Trunk and Trunk Group. This allows for aggregation of channel capacity for multiple Trunks into the Trunk Group, so you only pay for the throughput needed, as channel capacity is only charged for at the Trunk Group level. In the diagram above, you can see three PBXs each with a SIP Trunk configured with 20, 30 and 15 channels, 65 channels in total. After looking at the usage across those sites, a maximum of 50 channels may actually be in use at any one time. Determined by your preferred call routing plan, configured by Virgin Media Business, 50 Channels can therefore be configured at the Trunk Group level and only 50 would be billed offering cost savings if migrating from ISDN. Priority Call Routing Also known as Overflow, with priority call routing, Trunks are assigned a priority within the Trunk Group to receive inbound calls. The primary Trunk will receive all inbound calls under normal conditions. If the primary Trunk Call Admission Control (CAC) threshold is reached (i.e. the max configured channels or bandwidth limit is hit) any additional calls are sent to the next Trunk in a Trunk Group up until the Trunk Group CAC is reached. Likewise, if the primary Trunk becomes unavailable then calls are routed to the next Trunk in the Trunk Group. External use Page 7 of 26

Outbound calls can be sent from any of the Trunks in the Trunk Group when using priority call distribution. Round Robin Call Routing Also known as Load Sharing, Round Robin call routing distributes incoming calls across all trunks configured within the Trunk Group. This is done using a platform algorithm that delivers calls approximately evenly across each of trunks. Outbound calls can be sent from any of the Trunks in the Trunk Group when using Round Robin call distribution. External use Page 8 of 26

Call Admission Control (CAC) Call Admission Control applies at both the Trunk and Trunk Group level to ensure service is provided with the quality expected. At the Trunk level, both maximum calls (channels) and the bandwidth available for voice calls is configured for Call Admission Control. This ensures that the voice calls offered by the platform do not exceed the number of channels or the available bandwidth on that trunk. At the Trunk Group level, a maximum channels based Call Admission Control is applied to ensure that the total offered voice calls do not exceed the number of channels paid for. Note that the Call Admission Control feature in SIP Trunking from Virgin Media Business is not directional i.e. there is only one setting that applies across both incoming and outgoing calls. This in contrast to some ISDN30 implementations where inbound and outbound channel limits can be configured separately. This may however be achieved by configuring multiple Trunks. Call Attempts per Second (CAPS) Call Attempts per Second (CAPS) or Calls per Second (CPS) refers to the number of calls a system can setup in a single second. Note that this is different to the number of simultaneous calls. When Trunks are provisioned, the platform uses signalling rate controls to set the maximum achievable CAPS based on the quantity and type of channels ordered. There is a CAPS Calculator built within the order pack which shows the CAPS rate against the volume of channels ordered. The calculation shows the CAPS that the 3 channel types (Basic, Standard or Premium) will be able to provide. No Channels Basic Standard Premium 500 0.425 1.276 5.102 An example of the CAPS calculator 4. SIP Trunking Features Call Diverts Call diverts are configured on number ranges. Options available are as follows: Call divert unconditional All incoming calls will be immediately delivered to an alternate destination number (one number for the entire DDI range). This destination should be a valid UK reachable number. Call divert busy Incoming calls to a Trunk that has reached its CAC limits will be delivered to an alternate destination number (one number for the entire DDI range). This destination should be a valid UK reachable number. Call divert error Where there is an error reaching the PBX terminating a Trunk, calls will be delivered to an alternate destination number (one number for the entire DDI range). This destination should be a valid UK reachable number. SIP error codes 408, 500 and 503 from the PBX will trigger the call divert on error, as will two failed SIP Option messages. Anonymous Call Reject Incoming calls that have a presentation number marked as Anonymous or withheld (using 141 prefix or using privacy headers) can be rejected at the platform level if required. This is a feature configured against specific number ranges. External use Page 9 of 26

Incoming Call Barring Inbound call barring can be configured on individual number ranges if required. When activated, all inbound calls to the number range will be barred. Outbound Call Barring Outbound call barring is configured on Trunks. Options available are as follows: Permanent OCB, excluding emergency requesting this will not allow any outgoing calls except those to emergency numbers. Premium OCB requesting this will not allow outgoing calls to premium rate (09) numbers. International OCB requesting this will not allow outgoing calls to international numbers. Operator controlled OCB requesting this will not allow outgoing operator controlled calls (100, 155, 198, 118 and 195). Numbering and number features Number Management To create a fully working SIP Trunking service, numbers need to be assigned to either Trunks or Trunk Groups in order to receive inbound calls. You are able to purchase new numbers or arrange to port existing telephone numbers. There isn t a restriction to local exchange area codes like ISDN, and new or ported numbers can be assigned to a Trunk for any exchange area in the UK. For example, and 0203 number can be allocated to a Trunk based in Aberdeen. When a number range is mapped to a Trunk, the incoming traffic to that range will only be presented to that Trunk. Where a number range is mapped to a Trunk Group, then the Trunk hunting algorithm (round robin or priority) is used to select which Trunk will receive the next incoming call. New Numbers New geographic numbers can be requested as part of an order for Trunk Groups and Trunks. Contiguous blocks of up to 100 numbers can be requested and a single order may contain multiple blocks. New numbers are ordered directly against a Trunk or Trunk Group and cannot be moved so please only order the quantity required. Please note 0203 has now replaced 0207 and 0208 for new London number requests. Ported Numbers Geographic number(s) can also be ported to the service. Both Single Line and Multi-Line porting is supported. Once a number port request has been placed and accepted by the Losing Communications Provider (LCP), these numbers can be added to a Trunk or Trunk Group. External use Page 10 of 26

Emergency Services Database Updates This product offers access to emergency services in compliance to Ofcom General Condition 4, which consists of two elements; the ability to dial the emergency services numbers (999 and 112) and the provision of sufficient information to the emergency services operator so that even in the event of a silent call, the emergency response is dispatched to the right location. Virgin Media Business will gather required information to support Emergency Service Database Updates but it s the responsibility of the end user to provide caller location data for addition to the Emergency Services Database (ESDB). When an emergency call is made from a SIP endpoint, information is presented to the operator to flag it as a VoIP call. This flag ensures the operator asks the caller to confirm their location as the CLI presented may not represent the actual location. In the event of a silent call the operator utilises location information in the ESDB to assist in locating the caller. Where the operator needs further support they can call a 24x7x365 helpdesk to assist with tracking down the caller location. When new numbers are added to Trunks or Trunk Groups, location information is taken from the Trunk or Trunk Group contact details supplied and submitted to the ESDB. A number cannot be activated on the service until location details are provided. The ESDB location information is therefore static and pre-provisioned. It is not possible to provide location information dynamically, nor does the SIP protocol, at the moment, have an agreed way to convey dynamic (user provided) location information. It is the end customer s responsibility to ensure this information is correct and request Virgin Media Business to make any changes to keep up-to-date. It is recommended that a PSTN fixed line or mobile telephony device is available in the event that any of your IP voice or data services are unavailable. If ADSL is deployed then a PSTN line is available and a fixed line device should be plugged into this. Calling Line Identity (CLI) The product supports both Network CLI and Presentation CLI which can be set independently for outbound calls. The Network CLI is used within carrier networks and is the number used by the Emergency Services if a call is made. It should therefore always be a UK geographic number. A presentation number is a number nominated or provided by the caller that is presented to the receiving party when a call is made. It can identify that caller and can be used to make a return or subsequent call. The options for each are as follows: Default a separate default value can be set for the Network and Presentation CLI. For example, a receptionist s number may be used for all calls generated from the Trunk. From Header the From Header in the SIP messaging can be used to set network and presentation CLI on a per call basis. PAID Header the P-Asserted-Identity header in the SIP messaging can be used to set network and presentation CLI on a per call basis. Where the From or PAID headers are being used by the PBX to send the network or presentation number on a per call basis, if the PBX sends invalid information, the default value will be sent instead. Invalid information may be a number not assigned to the Trunk or Trunk Group and not included in the screening list, or information incorrectly formatted. If however the PBX indicates that the CLI should be withheld (see CLIR section below) then no number will be sent. Also note that some other communications providers may incorrectly display the network CLI instead of the presentation number. This is contrary to the Ofcom CLI Code of Practice and a feature of the called party s service and not unique to this product. External use Page 11 of 26

Calling Line Identity Restriction (CLIR) Restricting or withholding the presentation CLI is typically carried out on the PBX but SIP can restrict the presentation CLI for all calls on a Trunk if required. If the PBX supports the function, it can be applied on a per call basis. SIP will accept a 141 prefix or Anonymous as well as the privacy ID to identify calls that need the presentation CLI withheld. Calls will appear as a Withheld or Unknown number to the called device, depending on the receiving networks ability to treat the call. Some local mobile and international networks for example are unable to treat the call properly and present Unknown instead of Withheld. CLI Format Options For incoming calls SIP Trunking calls will send the full dialable number to the End User PBX. The format options are dialable and global (example below) which can be set differently for the Incoming Called Party and Incoming Called Party numbers presented. This can be configured on a per Trunk basis. Dialable = 02031234567 Global = +442031234567 Some PBXs are unable to handle both formats so it is essential the settings on the portal match the actual PBX configuration. Screening Lists All numbers built directly onto Trunks or Trunk Groups, whether a new number or ported in, can be used as the presentation CLI as standard. Where other numbers need to be used that are hosted on another platform, SIP has a screening list facility that can be configured on a per Trunk basis. Additional single numbers or number ranges can be added to the screening list. Up to 50 entries in the list are permitted (as ranges can be added that allows far more than 50 numbers if required). A letter of authority will be required from the owner of the number allowing them use of that number. Class 1-5 CLI Support SIP enables all five CLI classifications to be used. See brief description of each below along with the way in which SIP can support it. Type 1 CLI A presentation number that is applied at the network level for all calls generated from a Trunk. This is handled by SIP by configuring: Presentation Number Source = Default Presentation Number Default = The required presentation number Any Presentation numbers sent from the PBX are replaced with the Default by WSIPT Type 2 CLI Presentation numbers that can identify the extension number of the caller. Although the number will be generated by the PBX, the network provider is able to check authenticity. This is handled by SIP by configuring: Presentation Number Source = From or PAID (PBX configured to match) PBX only sends Presentation numbers that are configured on the Trunk Presentation Number Default = Number to be used if PBX sends an invalid number External use Page 12 of 26

Type 3 CLI Presentation numbers for callers that may be in a different geographical location to the PBX. With ISDN that may mean that the caller s number is not hosted on the Trunk generating the outbound call. The SIP service essentially treats Type 3 Presentation Numbers in the same way as Type 2 as SIP does not limit numbers on a Trunk to a local exchange area: Presentation Number Source = From or PAID (PBX configured to match) Screening List = all valid numbers or range(s) added PBX only sends Presentation numbers that are configured on the Trunk or in screening list Presentation Number Default = Number to be used if PBX sends an invalid number SIP cannot validate that the CLI is correct for the caller only that it is a valid CLI for the specific Trunk. Consequently you have a contractual responsibility to ensure that numbers sent from the PBX are valid for the originating party (particularly for emergency calls). Type 4 CLI Where an inbound call is passed back out again retaining the original inbound caller s CLI. On the outbound leg the number is generated by the PBX although should be the same number as the number received on the inbound leg. If you wish to use type 4 services you will need to sign a contractual commitment that they will only submit CLIs that have been received from the public network. Unlike other types of presentation numbers, type 4 numbers may not always be dialable; this will depend on the nature of the number received on the inbound leg. The SIP service does not technically support type 4 Presentation Numbers but can be achieved as follows: Presentation Number Source = From or PAID (PBX configured to match) Screening List = ranges added to cover all inbound caller Presentation number scenarios Presentation Number Default = Number to be used if PBX sends an invalid number Type 5 CLI Where presentation CLIs need to be used that are not configured on a Trunk or Trunk Group but the user has permission to use it. Type 5 presentation numbers are generated by the PBX. A typical scenario is a call centre making calls on behalf of more than one client. The SIP service essentially treats Type 5 presentation numbers in the same way as Type 2. Presentation Number Source = From or PAID (PBX configured to match) Screening List = all valid numbers or range(s) added PBX only sends Presentation numbers that are configured on the Trunk or in screening list Presentation Number Default = Number to be used if PBX sends an invalid number There is a contractual responsibility on you as the customer to ensure that numbers sent from the PBX are valid for the originating party. Override Number Range Although number ranges remain associated with the original Trunk or Trunk group they were assigned to, SIP offers a facility to move numbers to another SIP service using a feature called Override Number Range. Whole ranges or a subset of a range can be moved to another Root Service, Trunk Group or Trunk. This can be useful when you need to move numbers around. External use Page 13 of 26

Long Duration Calls Calls made on the platform will remain up for a maximum duration of 24 hours at which point the call will be dropped. Equipment In order to provide a secure, stable and high quality service, all equipment connecting to the SIP platform must be on the SIP Authorised Equipment list located here www.virginmediabusiness.co.uk/voice-over-ip-support-guides All devices on the list have undergone detailed testing to ensure complete interoperability with the SIP platform, and are backed up by collateral to allow your organisation to configure equipment to work with SIP. These guides are usually available from the CPE vendor with copies also held by Virgin Media Business www.virginmediabusiness.co.uk/voice-over-ip-support-guides The main categories of equipment that can be used on the SIP platform are discussed in the sections below. IP PBX or SBC An IP PBX as the name suggests is IP enabled and can be directly connected to the SIP platform when it has successfully been through interop testing. An SBC (session boarder controller) may also be used with the PBX if included during interop testing, in which case the SBC would terminate the SIP Trunk. Where a PBX does not proxy the media (where individual IP phones behind it need IP address resolution), it can only be connected when fronted with a device which can resolve each IP address of the phones sitting behind it (e.g. an SBC). TDM PBX via Media Gateway TDM PBXs cannot connect directly to the SIP platform as the protocols used are not directly compliant with SIP. However, use of a Media Gateway (ISDN to SIP converter) which sits between the PBX and SIP allows you to connect to any TDM PBX directly to the service. This gateway acts as a bridge to translate TDM voice into SIP and carries calls to and from the End Customer PBX to SIP. Unsupported Equipment If equipment connecting directly to the SIP service isn t detailed within the SIP Authorised Equipment it cannot be used and service won t be provided. Equipment Configuration All SIP equipment connecting to SIP should be configured as per the vendor configuration guide using the details supplied during provision. No usernames or passwords are required as SIP uses IP authentication (rather than SIP registration) to authorise all connections to the platform. The IP address used by the PBX for the SIP Trunk therefore needs to be hardcoded during the service provision and is used to identify which Trunk traffic is coming from. In most deployments, a PBX will map to exactly one Trunk and all traffic to and from the PBX will be carried over that single Trunk. Where traffic needs to be separated over multiple trunks, the PBX should have the capability to support more than one IP address. External use Page 14 of 26

5. Service Overview The SIP Trunking Service from Virgin Media Business can be optimised through combining with connectivity and professional services, including consultancy, from Virgin Media Business. Scalability and Capability The SIP Trunking service is highly flexible and is suitable for all MLE organisations and can be scaled to suit your needs, including capacity that can be provided from 15 to 5,000 channels with 3 channel types. With the appropriate volume of channels and channel type, call rates can be up to 50 call attempts per second to support the highest traffic requirements. Dynamic channels can be utilised to meet customers high demand periods. The above channel capacity and call volumes are available as standard. If you require a service with great than 5,000 channels or that supports a higher call rate than 50 calls attempts per second then we can discuss you requirements. Please speak to us about this. SIP Trunking from Virgin Media Business is available across IPVPN and MIA Access types. We have geographically resilient connections into the SIP Trunking Voice platform. It s recommended that resilience is considered into your sites because, if you lose connectivity, you ll also lose access to the SIP service for that site. Virgin Media Business Bandwidth As SIP is delivered over IP services, the product will consume bandwidth on the customer s access type. The amount of bandwidth consumed is dependent on: Number of simultaneous calls required during busy hours Traffic Type Inbound and Outbound calling patterns Calls per second Choice of Codec Type, G711 or G729 Use of access for both voice and data and whether QoS is applied As Virgin Media Business will be providing the access service alongside SIP we want to ensure that neither product is to the detriment of the other as part of the solution. To guide you with the amount of bandwidth required, there will be a recommended bandwidth calculator as part of the order pack, please see below. This will indicate as to whether your existing or new access service will be able to support the volume of usage required for SIP to work effectively. The calculation is based against upstream as this will be the primary impacted bandwidth based on calls made. Upstream bandwidth needs to be double the expected voice bandwidth. The recommendation is made in a green and red system based on: Green Required SIP bandwidth is <70% of purchased bandwidth o The current bandwidth can cope with the SIP service that will be delivered over it unless the current data use is high and already reaching capacity on purchased bandwidth. Red Required Cloud Voice bandwidth is 91% of, or exceeds, the purchased bandwidth o We recommend an upgrade of bandwidth to avoid any deterioration of the SIP Trunking product. External use Page 15 of 26

Please note, the Order Pack can only provide a guideline and is not able to take into consideration existing usage of bandwidth based against non-voice usage. For more specific calculations, the below details can be used. Bandwidth Calculations & Considerations To determine the number of simultaneous/concurrent voice calls that can be achieved it is important to note that the available bandwidth is determined by the UPSTREAM. We have pulled together the below table to detail the bandwidth volumes and maximum user channels availability within the individual access technologies: Access circuits D/S Mbps U/S Mbps Max choice B/W Max no. of G.711 & G.722 calls Ethernet 10Mb 10 10 5 36 83 Ethernet 20Mb 20 20 10 71 150 Ethernet 30Mb 30 30 15 107 150 Ethernet 40Mb 40 40 20 143 150 Ethernet 50Mb 50 50 25 150 200 Ethernet 60Mb 60 60 30 150 300 Ethernet 70Mb 70 70 35 150 300 Ethernet 80Mb 80 80 40 150 300 Ethernet 90Mb 90 90 45 150 300 Ethernet 100Mb 100 100 50 357 750 Ethernet 200Mb 200 200 100 714 750 Ethernet 500Mb 500 500 250 750 750 Ethernet 1000Mb 1000 1000 500 1000 1000 Max no. of G.729 calls We apply and adhere to the following network recommendations: 6. Technical configurations Platform Components The SIP Trunking services are made up of a number of hardware, network and software service components that work together to provide the user s access to an IP based telephony services. Even though the platform components are not owned or managed at a platform or trunk access level by Virgin Media Business, for completeness a description of these main components from a solution perspective are shown below: Application Servers - These operate at the core and handles users, groups & subscription services. Media Servers These enable a broad array of media features including, Auto Attendant, Music on Hold, service announcements etc. External use Page 16 of 26

Network Servers These enable scalability and geographic redundancy, dial plans and Voice VPNs. Profile Servers These store the customer and user data. XSPs These provide additional services such as toolbars, soft clients, device management and APIs. Collaborate Servers (UMS/USS) These provide instant messaging and collaboration services. Session Border Controllers (SBCs) These control the entry points into the platform, The SBC s take care of things like Security, NAT traversal and call routing processes. Business Zone This web-based application server enables configuration of users, the Business Zone enables service set-up. Supported Codecs SIP is a VoIP service, which means that a conversation is digitally encoded for transport over the IP network and this encoding is done using a codec. There are a number of codecs available for use. G.711 is a codec used for voice compression and is comparable to PSTN quality calls. It uses approximately 85-100kbps of bandwidth to carry one simultaneous call. Typically a G711 call will provide a MOS score of 4.0 and above. G.729 is a codec used for voice compression and is comparable to ISDN quality calls. It uses approximately 24-35kbps of bandwidth to carry one simultaneous call. Typically a G729 call will provide a MOS score of 3.7 and above. External use Page 17 of 26

End User Firewall Security When deploying SIP, you may need to amend local firewall polices to allow SIP to function correctly. See table below with the TCP/UDP ports used: Protocol SBC IP Address Customer SBC/PBX IP Address Source Ports Destination Ports Customer Source & Destination Ports SIP (Signalling) See Trunk Order KCI Configured on portal UDP 5060 or TCP 1024-65535 UDP 5060 or TCP 5060 Configured on portal RTP (Media) See Trunk Order KCI Any as negotiated in SDP UDP 32768-65535 UDP 32768-65535 Any as negotiated in SDP The level of configuration will depend on the make and model of the firewall used. CPE Protocols and Ports Overview Device Protocol Destination Destination Port IP Phone & ATA Signalling SIP _SIP_udp.ipcommsbtwbslnws09.bt.com 147.152.35.102/29 147.152.35.110/29 UDP/TCP 5060 to 5075 IP Phone & ATA Media RTP 147.152.35.100/29 147.152.35.108/29 UDP16384 to 32766 IP Phone & ATA NTP europe.pool.ntp.org UDP/TCP 123 IP Phone & ATA DNS Supplied locally UDP/TCP 53 Cisco Linksys Download & Configuration HTTPS dm-linksys.yourwhc.co.uk 193.113.10.34 193.113.11.36 TCP 443 Cisco Small Business Download & Configuration HTTPS dm-csb-yourwhc.co.uk 193.113.10.33 193.113.11.35 TCP 443 Panasonic Download & Configuration HTTPS dm.yourwhc.co.uk 193.113.10.10 193.113.11.10 TCP 443 Polycom Download & Configuration HTTPS dm.yourwhc.co.uk 193.113.10.10 193.113.11.10 TCP 443 Yealink Download & Configuration HTTPS dm.yourwhc.co.uk 193.113.10.10 193.113.11.10 TCP 443 External use Page 18 of 26

Transcoding When calls are made over the SIP platform, end-points are able to negotiate the appropriate codec directly. The calling party typically includes a list of the codecs supported in priority order in the invite. The called party then compares that list to its own supported codecs and selects a common codec from the list to be used for the call. Where no common codec is found, the call will fail unless a device in the middle (e.g. SIP SBC) can act as a translator and perform Transcoding. Hosted NAT Traversal Network Address Translation (NAT) is often used to translate private IP addresses used for devices on a LAN, into a public IP address that can be routed over a WAN. This translation can cause problems with SIP if not handled correctly as SIP messages do not match the IP packet used to send it. As IP authentication is used the IP address used by the end customer SIP equipment has to be known. Some SIP equipment when assigned a private IP address behind a NAT connection can be configured to use the external public address in the SIP messaging. If the SIP equipment cannot be configured in this way, it can also be managed by configuring the Hosted NAT Traversal feature on the Trunk. When using the Hosted NAT Traversal feature, the internal and external IP addresses and ports used will need to be known. Fax, PDQ & Alarm Support Fax Support The SIP platform supports Group 3 (G3) fax transmission using T.38 and G.711 up-speed when used with compatible CPE. Due to the wide variety of proprietary extensions to fax standards, we cannot guarantee interworking between all fax terminals. T.38 as standard does not support all extensions used by G3 fax machines such as the v.34bit transmission speeds used by Super G3 fax terminals. SIP supports the following when dealing with fax and compatible CPE and should be able to support the following: T.38 Fax Relay as per ITU T.38 Annex D standard. G.711 pass-through CPE is expected to pass-through fax modem signals, with the ability to disable echo cancellation and dynamic jitter buffers on a per call basis. G.711 up-speed CPE is expected to up speed from G.729 to G.711 if a fax tone is detected, using a SIP re-invite mechanism. PDQ Machines Support Credit Card (Process Data Quickly) machines used over SIP connections unfortunately have problems and may not work with any consistency. This is due to the wide variety of machines and different implementations and the codecs used by ATAs have been designed to compress voice, not the analogue signals sent and received by modems. We use G.711 pass-through for these calls, where the data is carried in a VoIP call encoded as audio as these calls are very sensitive to network packet loss, jitter and clock synchronization. Where high compression encoding techniques (e.g. G.729) are used some tonal signals may not be transported correctly across IP network. SIP will not guarantee any PDQ machines will work on the network. You can however try to use them but it is advised that they are tested first before deploying to End Customers. External use Page 19 of 26

Alarm systems Many alarm companies are now developing services to work with VoIP but many do not work correctly. Consequently SIP will not guarantee inter working with these deployments. 7. Customer Responsibilities Across the product portfolio, there are consistencies on what remains your liability throughout your Virgin Media Business SIP Trunking journey: Connectivity Support Ensuring the necessary equipment is available and able to support the site connectivity requirements. Either the SBC, Gateway or IP PBX connecting to the service must be on the approval list. Ensuring adequate rack space, desktop space and electrical power for all CPE. This will be part of the access circuit provision. Providing access necessary for Virgin Media Business and assigned personnel to fulfil deployment services to all sites to be part of the access circuit provision. Ensuring all requirements are fed through the account manager and detailed in the SIP Trunking order pack. Delegating a single administrative point of contact for Virgin Media Business service issues. Serving as the life-cycle maintenance contact. The Customer Administrator will be responsible for conducting the on-going activities required to maintain and administer the account. Activities include, but are not limited to: o Submitting feature account changes, user profile/parameter updates such as name and office location changes. It s imperative that any changes regarding 999 emergency databases are provided to Virgin Media Business. Contacting Virgin Media Business concerning all service issues. Assigning the Default Calling Number (DCN) to a person/agent who will be available to answer emergency calls from public safety personnel at each location. Further guidance can be found in the below FAQs: Ahead of sale Do you have an access circuit that suitable for SIP Trunking to be provided over? o The circuit needs to be Ethernet based and can be either a Virgin Media Business IPVPN circuit or Managed Internet circuit. o If you don t have a suitable circuit then please speak to us about ordering an appropriate service. Do you have an available port on the CE router of the access circuit? o If you are or intend to use the access circuit for shared services then you ll need to ensure that there is an available port to use for SIP trunking. o If you don t have an available port on the router then speak to us and we can discuss what is required to make your service SIP ready. Do you require an access resilient service? o If you require a resilient service to ensure you can continue to receive or make calls on failure of access circuit or equipment then you will require additional circuits. o If you don t have an available secondary circuit to use then please speak to us about available service and configurations to ensure the correct resilient solution to meet your needs. External use Page 20 of 26

How many channels and what kind of SIP channel do you need? o The volume of channels can be from 15 up to 5000 channels. o The type of channel will determine the volume of call attempts per second (CAPS Rate) the service can manage. o The volume of channels will determine the volume of simultaneous calls the service will be able to manage. Do you require Dynamic SIP channels? o Do you have seasonal or campaign driven increases in call traffic? o Do you need to expand and decrease the volume of calls you can make and receive at short notice? o If yes then dynamic channels can help you manage this demand. Do you have the available bandwidth to support the volume of channels you require? o We have set a recommended maximum number of channels against the available upload bandwidth of the access circuit: o 10Mbps upload 28 channels o 20Mbps upload 69 channels o 30Mbps upload 104 channels o 40Mbps upload 138 channels o 50Mbps upload 173 channels o 100Mbps upload 345 channels o 200Mbps upload 694 channels o 500Mbps upload 1,736 channels o 1Gbps upload 3,400 channels Do you wish to keep your telephone numbers and therefore need numbers to be ported? o If yes, you ll need to complete a Letter of Authority to authorise us to port the numbers away from your existing telecoms provider. The LOA form is within the order pack which can be provided to you. o Even if you have existing numbers or voice services with Virgin Media Business, the numbers need to be ported. The porting enables the services to move from traditional voice telephony to new Voice over IP services. o If you don t have numbers to port then we can provide you with new numbers from all geographic regions to support your requirements. What connectivity limitations would impact the SIP Trunking service? o Delay Latency of more than 150ms. o Jitter more than 30ms. o Packet loss more than 1%. Do you have the correct equipment to support SIP Trunking? o SIP Trunking requires a Session Border Controller (SBC), Media Gateway or IP PBX to connect to. o If you do have any of the above equipment please refer to the authorised equipment list to ensure your equipment is supported. o If you don t have any of the above equipment or don t have equipment on the authorised list then please speak to us as we can provide the appropriate tech to support your requirements. Ahead of deployment Do you need Quality of Service (QoS) on your LAN? o Though QoS is not mandatory for SIP Trunking service, any potential quality degradation can be reduced drastically by enabling end-to-end QoS. o If you re concerned about your LAN infrastructure, a LAN audit Professional Service can be purchased from us. External use Page 21 of 26

Do you need Quality of Service (QoS) outside of your LAN? o Though QoS is not mandatory for the SIP Trunking service, any potential quality degradation can be reduced drastically by enabling end-to-end QoS. o QoS is only available over Ethernet based IPVPN circuits within a WAN. Introductions/amendments of QoS on the WAN must be completed via the usual Virgin Media Business IPVPN alteration methods, completing a CR3 form. o QoS is not available over Managed Internet Circuits. Have you requested service configuration to meet your requirements? o SIP Trunking is a very tailorable service and can be configured to support a wide variety of requirements. o The service can be configured to aggregate channels across trunk groups and trunks and it can support bandwidth and channel controls to protect your priority traffic. o If you have any queries around available options and configurations then please refer to SIP trunking support documentation or speak to us. Do you have support to configure your equipment to work with the SIP Trunking Services? o SIP equipment needs to be configured correctly to work with specific SIP Trunking. If you have already been using SIP Trunking it doesn t mean the same configuration will work with our SIP Trunking. o We can provide you with the appropriate supporting configuration guide for your equipment to allow you to correctly set-up you service. Do you have an internal firewall? o When deploying SIP, you may need to amend local firewall polices to allow SIP to function correctly. See table below with the TCP/UDP ports used: Protocol SBC IP Address Customer SBC/PBX IP Address Source Ports Destination Ports Customer Source & Destination Ports SIP (Signalling) See Trunk Order KCI Configured on portal UDP 5060 or TCP 1024-65535 UDP 5060 or TCP 5060 Configured on portal RTP (Media) See Trunk Order KCI Any as negotiated in SDP UDP 32768-65535 UDP 32768-65535 Any as negotiated in SDP Day of deployment Do you have your equipment connected and configured? o We won t be able to deploy the service unless you have your equipment connected to the access circuit and correctly configured. o We will test and prove the service but we won t be able to support configuration issues for equipment that we have not provided. Do you have the appropriate support available? o We would suggest your equipment maintainer is available on the day of deployment to support with any issues that may arise with configuration. o If you have ordered equipment from us then we will ensure the service is provided and configured to support the deployment of SIP trunking. In-life service What happens if SIP Trunking doesn t work? External use Page 22 of 26

o Please follow the Virgin Media Business fault support process as soon as a fault is found. Can I increase the volume of channels or DDI numbers allocated? o Please contact your sales representative who will arrange this for you. o You must make sure you have sufficient upload bandwidth to accommodate any additional channels. Can I change the configuration of the SIP Trunking? o Yes, you can change the configuration of the service, please contact your sales representative to discuss this. Can I reduce the service I have? o It s possible to reduce the service you have. If the service is still in contract term, early termination charges may be applied based on the remaining term left for the service. Can I end the service if it s no longer wanted? o Please inform your sales representative or visit www.virginmediabusiness.co.uk and request to cease the service. If you re still in contract you may face early termination charges. 8. Order Journey Order Pack Once your requirements have been established for SIP Trunking, it is possible to produce a quote. An Order Pack has been created to ensure all required information is captured, quoted and signed for by you, from quote through to order entry handover. Key information captured within the order pack: Access circuit type (If not installed at time of quote the expected access method can be used) Access resilience required Volume of channels required Type of channel required (Basic/Standard/Premium) Dynamic channels required New numbers required Ported numbers required If desired, you can then send over a pdf of the quote from Virgin Media Business. Once satisfied with the quote, the order pack will continue with the information provided within the quote to drive an order form which can be signed for by your organisation. If any of the quote details have changed, the quote must be revised before the order pack can be completed and an order form provided. You will also be signing up to the Virgin Media Business standard terms, SIP Trunking special terms and any voice policies applied. Once the service is signed for, your organisation, site and user details will need to be captured. The information can either be captured by the sales representative with your guidance, or you can complete the required information in your own time. External use Page 23 of 26

Provisioning Note, if the SIP Trunking is delivered over a new access service, this will be prioritised to be installed ahead of the SIP service. This is to ensure that correct provisioning of the service and actual use of the service can be obtained when live. On submission of the SIP order, Virgin Media Business will send a welcome email to the nominated recipient at your organisation. Throughout the order delivery, the SIP Order Manager will advise of your order progress. Data Services If the customer has ordered professional services to be performed by Virgin Media Business, the team will engage with you to arrange a time, date and location for the professional services to be performed. External use Page 24 of 26