Introduction
UK Market Overview – Hosted VoIP and SIP Trunking are now established and credible options for customers. In December 09 there were 420,000 hosted extensions and 100,000 SIP Trunks in the UK with these figures growing 11% quarter on quarter [source: Illume Consulting].
Who Are The Competition? In general the main competition at the point of sale is from hardware based PABX solutions. It can be beneficial to have another hosted proposition on the table to help reinforce the concept to the customer, especially as you will not get outgunned on features. Customer hot buttons change as the size increases. While VanillaIP is a credible option at the low end, we are stronger at 15 extensions and above.
Customer Profile Hot Buttons Smaller
Customers
• Anywhere from 2-10 users – single site
• Normally the decision maker is the owner/operator
• Functionality wise they fall into two extremes, those that want a feature rich platform [where we are strong] or those that want a cheap and cheerful solution [Asterix].
• No home workers
SME and Multisite
• 20 users and upwards – often multi-site or with remote workers • Decision maker often the IT Manager, Ops Manager or FD • More interested in being able to admin the system themselves • Interested in centralised billing
• Often requires performance metrics on inbound/outbound calls • Typically have proper IT infrastructure in place and may have
What is VoIP? – In simple terms, VoIP is normal telephone conversations that have been packetised for delivery over IP networks. The value of VoIP is not as a delivery mechanism, but rather the applications; how they work and what they can do to drive the business forward. If VoIP is a means to this end, then fine. Too often however VoIP has been presented as the ends in itself.
VanillaIP is “pure” end-to-end VoIP rather than the largely hybrid solutions on the market currently [such as Panasonic and IP Office]. Mitel and Cisco are both pure VoIP CPE [customer premise equipment] solutions.
VanillaIP is a phone system from the average user’s point of view. This is not necessarily a technology led sale, unless the customer wants it to be.
Skype – Skype has done us favours in raising awareness but it has serious limitations as a business solution; it’s a peer-to-peer system, no central intelligence, no business functions [ie one number, music on hold, hunt groups, reporting, etc]
What is Hosted? Introduce the concept that everything is hosted – there is no intelligence residing locally on the customer site. Most customers now have a degree of hosting in their business, either off-site storage/back-up, hosted Exchange, hosted CRM [like salesforce.com] or Head Office hosting servers and applications for remote sites. So, hosted voice is often an extension of an existing business practice.
What is QoS? – Quality of Service [QoS] is an umbrella term that refers to a set of techniques designed to ensure good call quality. Simply put, these prioritise VoIP calls across the network ahead of non time-sensitive traffic like email and web browsing. The biggest hurdle in ensuring QoS is on the customers local LAN. Our best practice for installation at a customer site recommends either a separate physical LAN, or VLAN, for the IP phones.
What is SIP? Session Initiation Protocol [SIP] is an industry standard for setting up and tearing down communications sessions. Our BroadSoft platform is SIP based. The key advantage for customers of a SIP based system is that it protects their investment in handsets, in that they can be used on another SIP system. Although BroadSoft is SIP based it is not “open standard”. The signalling for feature activation and from the switch to the applications is proprietary [then same as for Mitel, Cisco etc].
What is Unified Communications? UC has replaced “Convergence” as the industry buzz word of choice. Everyone has a view of what UC means depending on the relative strengths of their product. In general, UC means combining all elements of the customers communications mix. This includes unified voicemail, presence, instant messaging, directory look-up, conferencing, CRM integration, mobile integration, calendar integration etc.
An important part of UC is the ability to access and manage all services in one place. The Uboss provisioning portal provides a single point of entry and real-time management of all users and services on the VanillaIP platform.
Key Customer Benefits
User Benefits
Full PBX Feature Set – Hunt groups, reception consoles, conference calling, hot desking, computer telephony, music on hold and all other standard PABX features are all available.
Advanced User Options - Advanced user functionality like fixed mobile convergence [FMC], simultaneous ring, presence and instant messaging can be allocated based on the individual user requirement, without investing in the whole business.
Advanced Group Options – Call recording, call centre, international numbers and intelligent network queuing and reporting can be added, again, without the need for capital investment.
Business Benefits
Simplicity of Deployment – Having no hardware onsite [except phones and switches] reduces complexity for customers and Resellers. VanillaIP IP handsets are self aware and will auto configure as soon as they are connected to the network. This greatly simplifies the initial deployment and the ability for the customer to change or add phones going forward. 50% of our customers self install. There are no ongoing Maintenance charges [though the reseller may build in some kind of maintenance if they wish]
Centralised Billing – The customer’s total telecoms spend, including ADSL, users, services and calls can be combined in a single bill, broken down by site or as necessary. If VanillaIP billing is being used, the invoice history, with detailed breakdown, is available online within Uboss.
Cost Control – As an extension of centralised billing, VanillaIP allows customers to get an exact picture and control of the telecoms spend. Often customers have SIDN and analogue circuits that they don’t want to disconnect in case something stops working.
Scalability – Customers can flex their services as necessary. For seasonal customers, or those that are strongly project based [such as construction companies] this positive and negative scalability ensures they only pay for services they are actually consuming.
Uboss Web Based System Admin - Users, analogue lines and ADSL can all be provisioned through Uboss, either by the Reseller or the customer. The value of Uboss, particularly to larger or multi-site customers cannot be overstated.
Economic Model - Opex vs Capex – A recurring revenue model with reduced upfront charges can help customers access operational budgets
Home Worker Support – Home workers can be deployed without any VPN or special connection. All they need is a business ADSL connection. They can have the same numbering plan as the office and be visible to Reception Console and Unity users.
Multi-Site Networking – Remote sites benefit from centralised admin, centralised reception, and desk-to-desk dialling. Members of group services such as hunt group and call centre queues can sit across multiple sites.
Number Portability – All existing UK numbers can be ported – there are no restrictions on taking numbers away from the local exchange region. Number porting takes 25 working days.
Credit Limits - The VanillaIP Uboss portal has pre-set call credit limits for both individual users and customer businesses. When these limits are breached the phone or phones are toll barred automatically. In the event that a rogue employee makes unauthorised calls the maximum customer exposure is £20 + VAT, the default call credit limit.
Network Overview
Customer installations will vary depending on the type of access [leased line etc] and internal data network that the customer has. In general, the IP phones will be installed on a separate physical LAN [or VLAN] with a separate, voice only, ADSL or leased line.
SIP Endpoint – A SIP endpoint is a device that the user talks into. This includes IP hardphones [Polycom and Cisco] and softphones [Bria].
Network Cabling – Being network devices, the phones need to be connected to the central switch via Cat5/6 network cabling. The phones are shipped with an RJ45 patch lead to connect to the wall. The IP321 does not have a spare witch port in the back, unlike the IP331 and IP550/650. Where the second switch port is being used for a PC the Reseller or customer needs to create a VLAN for the PC’s.
Voice Switch/LAN – Where a separate switch is being used a simple, unmanaged L2 switch is sufficient.
Softphones – Softphones sit on the same LAN as the PCs, as opposed to IP hard phones on a separate LAN. For this reason we do not have as much control over the call quality, unless the customer or Reseller can prioritise VoIP traffic. We recommend using softphones as a remote worker option rather than being widely deployed in an office environment.
Phased Rollout/Trial – By leaving an existing PBX/ISDN in place, diverting existing numbers to VanillaIP phones and replacing PBX phones on users desks, Resellers can offer customers a level of assurance that they can roll back should they have any problems. The reseller will incur licence costs in doing this though.
Connection To Apps – The client apps [Unity, Call Centre Supervisor and Reception] communicate to the BroadWorks switch over the PC network DSL. There is no physical connection between the phone and the PC
G.729/G.711 Codecs – A codec is an algorithm that is used to chop up the voice for transport over the IP network. VanillaIP offers two voice codecs, G.729 and G.711, that con be specified on a per user basis. Both codes are industry standards. The difference between the two is that G.711 is generally accepted to have better voice quality, though for most users they will not notice the difference. 90% of the users on the VanillaIP platform are connected on G.729. G.729 calls require 40k of bandwidth while G.711 calls require 100k.
MoS Scoring - Mean Opinion Score [MoS] is an industry measure used when comparing call quality. It is generally accepted that an ISDN call has a MoS score of 4.1. When using G.729 codec the MoS score is 3.9. When using G.711 the MoS score is 4.1. We undertake that 90% of calls will be at or above MoS 3.9.
Forecasting Bandwidth Requirement – The upstream speed on the DSL is bottleneck. For new office sites you will need approximately 1 x “channel” for every 3 extensions. For existing customers use the current number of ISDN channels as a guide. Factor on getting 60% of the upstream on ADSL. We suggest that an 860k upstream business ADSL can carry up to 12 concurrent calls. Where a customer needs more than 12 concurrent calls, it is possible to use 2 x ADSL, with different phones registering to different default gateways for the router on each circuit.
Home Workers – When home workers are installing Polycom IP321 phones, remember that these do not come with a local power supply. Residential ADSL services such as Talk Talk are not suitable for VoIP. If the home worker has an odd router that does not support, it may be easier to provide a replacement router. It is worth factoring on 20% of home routers being incompatible.
Partitioning Existing Leased Lines – If the customer has an existing leased line with sufficient spare capacity, the WAN provider can partition off part of the circuit for voice. Depending on the router, this can then provide a separate Ethernet connection for the voice network.
IP Addressing – The phones support DHCP and will be assigned a private IP address [192.168.X.X] and default gateway from the local router. There is no need for static IP’s. The Session Border Controller [SBC] in the core maps all private IPs. Individual IP phones send “keep alive” messages to the VanillaIP network every minute.
Phone Set-Up – Once the phone is connected it will pull down the users config file and service profile, based on the unique MAC address of the phone, from the secure VanillaIP FTP server.
Power Supply – The phones need Power over Ethernet [PoE] switches and phone power bricks.
VanillaIP Network Core – Fully geographically redundant based in 2 x data centres [Telehouse and Global Switch] with automatic failover. High level of redundancy within each site with redundant servers and redundant disk arrays. Refer to Network Security doc for more information
Throttling Simultaneous Calls – VoIP handsets will attempt to make calls regardless of how much bandwidth is available at the local customer site. In extreme situations, this can compress all calls to fit across the customer WAN resulting in poor quality. We have an optional service that can limit the total calls carried over a WAN circuit ensuring optimal bandwidth. Any additional calls that are attempted will get unobtainable tone. This is essentially the same as ISDN lines on a PBX system. If you only have 8 ISDN channels then you can only carry 8 concurrent calls.
No Tromboning of Internal LAN Calls – Once a call has been set-up, the actual voice call is an RTP stream. For internal calls, the RTP stream connects across the LAN, it doesn’t go outside on the DSL and “trombone” back in. All PSTN break-in/break-out is via BT, there is no IP trunking for international calls.
Connecting Analogue Devices – Fax machines can be connected through ATA or 4/8/24 port Vegastream POTS gateway. PDQ’s and modems need to remain on separate analogue lines.
Customer Site Redundancy
We have customers connected on WAN circuits from ADSL up to 100MB leased lines. The VanillaIP solution is not dependant on this access, provided it is of sufficient quality and has enough capacity to support the required number of customer calls. Redundancy options will vary depending on the type of access the customer has.
Redundant Voice WAN - Local redundancy options provide back-up in case the primary voice WAN fails. For customers using ADSL for VoIP, having a stand-by circuit is a cost-effective option. In this case VanillaIP would install ADSL from separate unbundled ISP’s [provided this was possible in the local exchange]. This ensures that an exchange problem is less likely to disable both circuits.
Router Auto Fail-Over – The above scenario can be extended by installing a router that terminates both ADSL circuits. This does not bond them to share combined bandwidth but gives automatic fail-over if the router senses that the primary ADSL is offline.
Local LAN Device Security - VanillaIP security does not extend to managing threats on the customers local LAN. Customers can optionally use 802.1X to block devices that are not permitted on the LAN. This requires individual devices to be authenticated with username/password on a server before they can physically connect to a switch port.
Reroute Calls – In the event that the local site is offline, calls are still being delivered to the customer numbers, voicemail, auto-attendants etc as the VanillaIP core is still up. Customers can have calls temporarily rerouted to mobiles, home workers or other remote sites.
Call Forward Not Reachable – An optional service for Premium users is Call Forward Not Reachable. This allows for configuring a location [for example a mobile] where a call should be redirected when the main device is unreachable [eg. If ADSL is down].
Disaster Recovery – If the office has suffered a catastrophic and prolonged service failure [such as flooding or major power outage] then VanillaIP offers unparalleled DR options. All IP handsets can be reconnected in another location [such as user homes or DR facility] where they will reconnect to the core and operate as normal.
Services and Licensing
There are three elements to VanillaIP licensing; User packages [Standard, Premier and Power, User Options and Group Services. 90% of the VanillaIP users are Standard Users. Refer to the current pricing for which user services are in which user package and the separate feature explanations for what these services do.
User Options
Simultaneous Ring – Another number can be set to ring simultaneously with the cust DDI. Best used in conjunction with Unity as this provides easy access to turn Sim Ring on/off. Sim Ring is only available on the Premier User licence
Remote Office – Allows the user to temporarily specify another number to act as their extension/DDI. Best used in conjunction with Unity. Inbound calls will be delivered to the temp Remote Office number. Outbound calls made from Unity will also ring the temp Remote Office number and will also ring the external party. The outbound CLI presented to the called party will be the user’s extension/DDI number. Remote Office is part of the Premier User licence package.
Shared Call Appearance – Up to three SIP endpoints [IP hard phones, WiFi IP phones or IP Softphone] can be assigned to the same extension/DDI. All phones will ring simultaneously.
Unified Voicemail – Voicemail provides Busy and No Answer greetings with the option to press 0 for a caller to escape. The user can specify where the 0 escape points [such as mobile, Reception etc] with Unity or from Uboss. Messages can be forwarded or copied to an email account. Again, the user can determine these options within Unity or Uboss. If messages are Forwarded, they are not stored centrally and so cannot be accessed remotely via the phone. A third option allows for notification of messages to be sent via email. This option would suits users that were concerned about storage etc with messages clogging email systems. When forwarded/copied to email the messages are in .wav format. Approximately 1MB for every 1 min of message. Whichever option is taken, there is no synchronisation between the email and voicemail box. Ie If a user deleted their copied voice message from their email, the message still sits in their voicemail on the system and their message wait lamp will continue to flash. To access voicemail from the phone, users must dial 5555 SEND. The first time they do this they are prompted through recording their message etc. On some phones, voicemail can be accessed through the context sensitive menu.
Fax Messaging [Fax-to-email] – Users can be assigned a separate DDI as a personal or group fax. This provides routing of inbound fax messages to the designated recipients email box. Fax messaging does not provide for outbound fax. Fax messaging can only be assigned to a user with voicemail.
Hot Desking – BroadSoft has the concept of Hot Desk Host and Hot Desk Guest. A Guest must log on to a Host to be able to make and receive calls under their extension. Both the Hot Desk Host and Premier User package can act as Hosts. Both Standard and Premier users can logon to a Host. Log on/off is done through the voice portal or through Unity
Directed Call Pick-Up with Barge-In – Power users can silently barge in to another users live conversation using barge-in. This is activated by a feature access code. To prevent unauthorised use of this service, users whose calls not allowed to be monitored need the Premier user package which includes Barge-In Exempt.
Group Services
Automated Attendant –DISA [dial the extension you want], 0 for Operator and dial by name are supported. A licence is required for each level of Auto Attendant. If the receiving caller needs to know what option the caller pressed when exiting the AA, then the calls need to be routed to a hunt group or call centre queue.
Hunt Groups – Any number of hunt groups can be set-up, between sites if required. Users can be a member of more than one hunt group simultaneously. When a hunt group call is delivered, the group name will be alpha tagged on the screen of display phones. Circular, group, Rotary and ACD call delivery options are available. There is no comfort message played to callers queued to a hunt group. Hunt group calls will not follow personal diverts/sim ring etc that a user has on their own extension/DDI.
Music On Hold – MoH is a group wide service that will support all users in the customer group. The music is uploaded to the system as a wav file.
Call Forward Selective – This service can be assigned to a hunt group, call centre or user and changes the routing of the call based on time and day profiles. Typically CFS is used to route after hours main number calls to a separate voicemail.
Paging – The Instant Group Call service allows multiple paging groups to be created. Premier users can dial the extension number of the Paging Group and send a message through the hands-free speakerphone of any user not currently on a call.
Call Pick-Up – A call pick-up instance allows users to pick-up one another’s calls. Specific extension or group pick-up can be used. Users not in the same group cannot pick-up calls. Calls can be picked-up using Unity or feature access codes.