1 INTECHNOLOGY S SIP TRUNKING SERVICE DEFINITION SD032 v1.8 Issue Date 12 th Sept 2013
2 Revision Control Revision Author Change Date SD032 V1.2 Phil Dyson Updates after review 15 th April 2011 August 2011 SD032 V1.3 Paul Storey Added network diagrams, removed reference to PBX Connect ISDN service SD032 V1.4 Paul Storey Updated Network Architecture Diagram. Added Billing and Bandwidth Calculation Table. 18 th August st October 2011 SD032 V1.5 Paul Storey Added number porting information 15 th November 2011 SD032 V1.6 Paul Storey Added note to Section 4 21 st March 2013 SD032 V1.7 Paul Storey Removed SLA and Service Credit sections 26 th July 2013 SD32 V1.8 Paul Storey Updated Section 5 Charging Policy 12 th Sept 2013 Related Documents Reference Title and Purpose Author Last version  Interoperability Test Plan S. Golding V1.0
3 Scope This document provides a technical overview of the InTechnology s SIP Trunk service PBX Connect SIP. The document will describe the relevant network topology, network elements, supported service features and limitations. The intended audience for this document includes InTechnology Sales, Support, Engineering, Product Management and contracted InTechnology customers.
4 SECTION 1: OVERVIEW SERVICE OVERIEW PBX Connect SIP is a Voice over IP (VoIP) service providing IP based connectivity to customers with existing IP PBXs. The service aims to provide full PSTN equivalence, delivered across InTechnology s managed IP network. The SIP trunk service is delivered from two centrally managed, software based, IP Voice Switches hosted in InTechnology s highly secure and resilient Data Centre environment. The service is sold and billed as a combination of a onetime service activation fee and a per-channel monthly rental. The customer needs to define in advance the number of discrete concurrent voice channels required. TARGET MARKET & BENEFITS PBX Connect SIP is primarily targeted at customers that have PBXs capable of supporting direct IP connectivity and are looking to completely replace their existing ISDN BRI or PRI circuits. InTechnology s SIP Trunking service enables organisations to run both voice and data services via a single converged network connection, maximising investments made in their existing telephony equipment. The PBX Connect SIP should be distinguished from InTechnology s PBX Connect ISDN product(s), whereby customers connect into InTechnology s VoIP network via a Q.931 PRI/BRI gateway. PBX CONNECT BENEFITS The ability to make and receive calls to and from the PSTN (Public Switched Telephone Network) via InTechnology s managed IP network and Tier-1 interconnect partners. Flexibility PBX Connect SIP can be increased delivered in increments of one after initial installation. The only limit is the customer bandwidth. Cost reduction through the reduction or removal of existing ISDN rental charges Free on-net calls between the organization s sites connected using the service Number porting from a number of service providers A Platform to deliver additional UC services. Improved network redundancy InTechnology s SIP trunk service can be configured to deliver multiple levels of failover including (Trunk, Trunk Group, or user), providing customers with a level of resilience typically not implemented in ISDN connected solutions. Competitive tariffs, helping customers reduce ongoing call costs.
5 SECTION 2: SERVICE DESCRIPTION Figure 1 details the typical enterprise SIP Trunk deployment models supported. Depending on the customer type and existing security policies, the customer may/may not choose to deploy their own enterprise Session Border Controller (SBC). PSTN InTechnology Trunk SBCs InTechnology TDM Gateways InTechnology MPLS VPN InTechnology Edge Session Border Controllers InTechnology Voice Platforms CPE CPE CPE IP IP PBX SBC Gateway SBC PRI/ BRI Customer A IP PBX Customer B TDM PBX Partners 1- N IP PBX 2 Figure 1: PBX Connect SIP Service Architecture INTECHNOLOGY VOICE SWITCHES InTechnology s PBX Connect SIP trunk service is delivered from two centrally managed, software based, IP Voice Switches hosted in InTechnology s highly secure and resilient DC (Data Centre) environment. The IP Voice Switch technology is based on tried and tested SIP (Session Initiation Protocol) open standards software providing a highly, scalable, cost effective and future proof platform for the provision of IP voice services. InTechnology s IP Voice platform has been deployed in a highly available configuration across InTechnology Data Centre locations in Harrogate and London. The core IP Voice Switches are maintained and supported by InTechnology 24 / 7 and access to the service is delivered across InTechnology s national IP MPLS Network and via a range of connection technologies including DSL, PPC leased lines and 10Mbps, 100Mbps and 1Gbps Ethernet as required.
6 SESSION BORDER CONTROLLER (SBC) InTechnology have deployed clusters of high availability Edge Session Border Controllers (SBCs) to act as the points of service interconnect between the customer and InTechnology s SIP trunk service. The SBC acts a point of network demarcation, performs NAT/PAT translation and SIP signalling interoperability. NETWORK CONNECTIVITY PBX Connect SIP is delivered over InTechnology s IP MPLS network LANnet. In order to assure call quality and performance across the network, InTechnology have implemented Quality of Service (QoS) systems throughout the LANnet network. A direct connection from the customer site to the nearest LANnet Point of Presence (POP) is required to deliver the service. These connections are available in bandwidths from 1Mbps to 1Gbps. NETWORK FAILOVER SIP trunk failover can only be implemented as standard with InTechnology provided circuits. Due to the diverse nature of client networks any failover of the SIP trunk to another service provider s network will need a level of professional services design work to be carried out, and as such would require commercial coverage to be in place prior a design being produced. PSTN CONNECTIVITY For external calls, the service includes centralized, low-cost PSTN breakout from InTechnology s IP Voice switches. InTechnology s high speed IP and TDM PSTN interconnects enable callers to make and receive PSTN calls. EMERGENCY CALLS Emergency 999 calls may be made using the service where supported by the customer s telephony equipment and dial plan. The customer is responsible for ensuring that InTechnology are provided accurate caller location information.
7 SECTION 3: DEPLOYMENT PROCESS It is necessary to ensure that Customer Premises Equipment (CPE) is compatible with the service. In order to do this, an audit of the PBX equipment may be necessary. Any reconfiguration required for the PBX to work with the service must be undertaken by the customer or their appointed contractor. InTechnology can provide PBX reconfiguration or PBX support services via a partner should the customer require PBX configuration support. PBX INTEROPERABILITY TESTING Due to the proprietary variations of the SIP standard that exist across different vendor platforms, InTechnology operates an interoperability test process for each IP-PBX that requires connecting to the SIP trunk service. This process has been implemented to ensure that a customer s IP-PBX operates in accordance with InTechnology s Interoperability test plan.  TESTING PROCESS 1. The customer must configure their PBX to route some non-business critical, test calls via the provided SIP trunk. The configuration details of the SIP trunk will be provided by InTechnology. (The SIP trunk provided for the purposes of interoperability testing will be with temporary phone numbers only.) 2. The customer may use an InTechnology LANnet connection or a suitable Internet connection of sufficient bandwidth as connectivity for the purposes of the interoperability testing. 3. An InTechnology technical consultant will liaise with an appointed customer contact to carry out the InTechnology Interoperability test plan and subject to successful testing, sign off the test plan with the customer. InTechnology will aim to complete the interoperability testing process within 30 days. During the interoperability testing phase: The customer will not be charged any monthly service charges. The customer will be charged for any chargeable calls made. Whilst every effort will be made, InTechnology offers no assurances that all IP- PBXs will work successfully with the service. In the unlikely event that an IP-PBX cannot successfully operate with the service in accordance with the InTechnology test plan, the customer may cancel the service during the first 30 days by providing InTechnology with written notice. If such notice is not received, the full minimum contract term will become applicable. Note: the customer will be responsible for all configuration works required on the PBX.
8 DECOMMISSIONING PROCESS Upon expiration of the service contract where the customer chooses not to renew with InTechnology, the following steps are followed as part of the decommissioning process: Phase 1: Contractual Expiration of the service contract and a decision by the customer not to renew. May also include early termination by the customer, subject to payment of early termination fees. Phase 2: Service decommission Cessation of the service and removal of user accounts, user access and administrative access from the platform. Phase 3: Number porting Porting of any telephone numbers from InTechnology to the gaining party where this party has a porting agreement with InTechnology.
9 SECTION 4: SERVICE COMPONENTS OFFICE CONNECTIVITY The PBX Connect SIP service is delivered over an InTechnology LANnet connection. If a LANnet connection is already installed, it s usually possible to supply the service over the existing connection without significant modification provided that there is sufficient bandwidth capacity available. PBX CONNECT TECHNICAL DATA The PBX Connect SIP supports both G.729a, and G.711 A-law voice encoding with a sampling period of 20msecs. The encoding algorithm results in a per call bandwidth usage of between kbs. Quality of Service (QoS) is implemented using ingress and egress traffic control at both ends of the LANnet connection. As part of the managed service, InTechnology provides a fully managed multi-service router for each customer site to shape traffic, ensuring that voice traffic is not negatively impacted by other traffic (such as Internet traffic) using the LANnet connection. Each customer is provisioned with a private voice network over LANnet. This ensures security and business grade call quality between sites. LANnet is InTechnology s dedicated 10 GB national MPLS network and is specifically designed to carry high volumes of both voice and data traffic without causing degradation to the voice service. If all available voice channels to a customer site are in use, InTechnology s IP Voice platform will prevent additional calls being setup, therefore, protecting the existing active calls being negatively impacted from call degradation. An incoming call which is unable to complete because the circuit is congested, or the subscribed maximum simultaneous call limit is reached will be treated as if the person they were calling was on the phone. Suitable LANnet connections for PBX Connect SIP service are InTechnology managed ADSL 1000 and managed ADSL 2000, PPC and Ethernet connections. Note: calls made between phones on the same site will not consume a channel because it will route internally through the PBX.
10 CONCURRENT SIP CHANNELS The maximum number of concurrent channels = the available bandwidth/total bandwidth, so if for example a customer has a 1Mb SIP trunk using G.729a with a sample period of 20 ms there will be 1Mb of bandwidth required = 20 concurrent channels available. Table 1 provides an estimate of the bandwidth requirements for VoIP calls using G.711, and G.729a, and the corresponding Mean Opinion Score (MOS). By comparison, GSM has a MOS of 4.1. The scores range from 1 (worst) to 5 (best). Network Codec Sample period Encoded sound bandwidth IP/UDP/RT P overhead Network overhead Total bandwidth* Ethernet G ms 64 kbps 16 kbps 15.2 kbps 95.2 kbps 4.4 G ms 8 kbps 16 kbps 15.2 kbps 39.2 kbps 3.9 DSL G ms 64 kbps 16 kbps 26 kbps 106 kbps 4.4 G ms 8 kbps 16 kbps 18.4 kbps 42.4 kbps 3.9 MOS Table 1: VoIP Bandwidth Calculations Note: InTechnology s SIP trunk customers IP-PBX MUST support both G.711 and G.729a encoding. In the event calls are made into InTechnology s managed ADSL user base, calls will be delivered with G.729a as the preferred codec to ensure quality and performance. PBX CONNECT SIP TRUNK SUPPORTS THE FOLLOWING RFCs RFC SIP: Session Initiation Protocol RFC RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals Note: The IANA Registered Port 5060 should be utilized for all SIP Signaling Messages
11 NUMBERING & NUMBER PORTING In order to facilitate inbound telephone calls, the service may be specified with geographic or non-geographic telephone numbers (DDIs). InTechnology can provide new number /DDI ranges to use with the service, as well as port existing numbers and number ranges from existing service providers. Number porting forms a vital part of InTechnology s the service delivery process, InTechnology have current Geographic & Non-Geographic and porting agreements as listed below in tables 1 &2 that facilitate porting their number ranges onto our service, meaning the customer can retain their existing inbound numbers as part of the Unity or Unity PBX Connect service. Non Geographic Number Portability (NGNP) BT Cable & Wireless (Ex-Energis) Your Communications/Thus Virgin Media (NTL & Telewest Cable & Wireless Magrathea Telephony Services Global Crossing KComms/Affinity Colt Telecom Verizon Table 2: Non-Geographic Porting Agreements Geographic Number Portability (GNP) BT Virgin Media (NTL, Telewest & Eurobell) Thus (incl. Your Comms) Cable & Wireless (Ex-Energis) Cable & Wireless UK (Inner&Outer London, Home Counties, Gloucester, Norwich, Nottingham, Hertford, Guilford, Ipswitch, Northampton, Wolverhampton, Brighton, Shrewsbury & Scotland) Inclarity Telephony Services Viatel (UK) Ltd Magrathea Global Crossing
12 Geographic Number Portability (GNP) Colt Telecom Spitfire KComms (does not include Kingston originating numbers in Hull) Verizon Voxbone Storacall (X-On) Opal Telecom (Subsequent, LCP import only) Lumison (Subsequent LCP import only) Primus (Subsequent LCP import only) Table 3: Geographic Porting Agreements InTechnology pre-sales consultants will gather the necessary information to complete the number port but will need the customer s assistance to identify the current Service Provider and Range Holder of the numbers, the site information including postcode and any associated direct dial inwards (DDI) numbers attached to main billing number. This information will then be included in the management summary of the contract so both parties have full visibility of the porting scope. In the majority of cases the presales consultants are able to obtain this data with just the main billing number and installation postcode from a customer s current invoice. InTechnology will in addition, request the customer provides on letter headed paper a standard letter of authority (LOA). The LOA together with the porting request form allows InTechnology s porting desk to talk directly to the losing operator s helpdesk on any number discrepancies. Once established that a customer s numbers can be ported InTechnology will complete the relevant industry documentation and submit the porting request to the losing operator (and range holder where different) together with the letter of authority. Porting lead-times are a regulated and critical milestone in the overall delivery of the service and it s vital that the data received from the customer and presented to the losing operator and range holder is accurate, as this could result in the porting request being rejected. Once the porting order is accepted by the losing operator, the lead-time commences between the range holder and gaining service provider. To set a realistic expectation this can take up to 22 working days and in very rare cases longer. The InTechnology project manager, assigned to the delivery, will advise on the delivery date once confirmation is received back from the losing operator, the losing operator ultimately controls the date that the port will happen. Until the porting date confirmation is received back from the range holder and service provider any dates quoted will be on an indicative and estimated basis. The installation price a customer is quoted includes any required number porting as long as the port is scheduled to take place during normal working hours (Monday-Friday/9-4), porting request outside of the stated hours will incur additional out of hours costs (bank and public holidays will be treated as an out of hours request). Note: If customer provided numbering information results in a subsequent port rejection, InTechnology may charge the customer for porting resubmission.
13 NETWORK & INDIVIDUAL CALLING FEATURES A range of optional network and calling features can be assigned to the service. Some of these features are chargeable. Some of these features are assigned at group level, meaning they apply to all DDIs within the SIP trunk. Other services are assigned at a DDI level and apply only to that specific DDI. Service Calling Line Identity Presentation (CLIP) Service Description Enables the called party to receive and display the calling party s telephone number before answering the call. The called party will only receive this information if the caller has not restricted the sending of their number (CLI). The called party will require suitable CPE in order to use CLIP. CLIP is assigned at an individual DDI level. Calling Line Identity Restriction (CLIR) Customers can request that their identities (telephone numbers) are not revealed at any time. This service is available free of charge. CLIR is assigned at an individual DDI level. Outgoing Calling Plan Restricts the ability to make outbound calls. Options include blocking all outbound calls and selective blocking based on call type. Call types include: Calls within the group Local calls National calls Freephone calls International calls Directory enquiry calls Premium rate calls including: o 070 o 0871 o 090 Outgoing Calling Plan is assigned at a group level and applies to all DDIs within that group. Incoming Calling Plan Restricts the ability to receive calls. Options include: Allow external inbound calls Partial allow / block external inbound calls only if transferred by another user in the group Block external calls Incoming Calling Plan is assigned at a group level and applies to all DDIs within that group. Call Forwarding Redirects inbound calls to an alternate destination. Call forwarding incorporates three sub-services:
14 Service Service Description Call Forwarding Always redirects all incoming calls to that DDI unconditionally to the specified alternate destination. Call Forwarding Busy redirects incoming calls to the alternate destination if the line associated with the DDI is busy. Call Forwarding No Answer redirects incoming calls to the alternate number if the call remains unanswered within a specified number of rings. Please note: The forwarded leg of the call will be charged to the customer at their pre-agreed tariff. Calls can only be forwarded to telephone numbers permitted by the outgoing calling plan Anonymous Call Rejection (ACR) Call Forwarding is assigned at an individual DDI level. Anonymous Call Rejection enables the rejection of calls from calling parties who have withheld their number. When activated, callers who have withheld their number are informed by an announcement that the person they are calling does not accept calls from anonymous callers. Call Forwarding Advanced Anonymous Call Rejection is assigned at an individual DDI level. Call Forwarding Advanced provides the capability to forward inbound calls intended for a particular DDI to another destination, when the incoming call matches pre-specified criteria. If the incoming call does not match any of the criteria, normal call handling applies. Definable criteria includes: A time schedule e.g. applies all day every day, or outside of business hours Whether the calling party line ID is private or unavailable A list of up to 12 phone numbers The criteria can be combined within predicates (for example, incoming call from this number and within business hours and during work week). Multiple predicates can be defined and the call is forwarded when at least one of the predicates is met. The user can associate a different destination with each predicate, or use the same destination for all predicates. Call Forwarding Advanced is assigned at an individual DDI level.
15 SIP TRUNK USER MOBILITY PACK The SIP trunk user mobility pack is designed to allow a user to take advantage of the mobility features available on InTechnology s voice platform whilst still utilising their existing PBX. Each DDI may be specified with this optional service pack that provides the user of that DDI with additional functionality delivered directly from InTechnology s voice platform. This pack provides the user with the Unity PC Assistant Toolbar. This software application sits on the user s Windows PC desktop and integrates into Microsoft Outlook and Internet Explorer. The toolbar provides quick and easy control of the features contained within the Mobility pack including: Call Forwarding Always Call Forwarding Busy Call Forwarding No Answer CommPilot Express Outlook Integration Remote Office Simultaneous Ring Personal Sequential Ring The PC Assistant Toolbar has a minimum supported level of hardware, software and network requirements. These are defined in the Technical Data section of this service definition. WEB ADMINISTRATION PORTAL The PBX Connect SIP service includes access to a secure web portal which may be used to administer certain features of the service. Access to this portal should only be provided to appointed personnel who are authorized to administer the organization s telephony services. Examples of what the portal may be used for include configuring the network and calling services.
16 PRODUCT ORDERING INFORMATION Part code N-PBX-111 N-PBX-115 Description PBX Connect SIP (per channel) PBX Connect SIP Service Establishment (Per Customer) INDIVIDUAL CALLING & NETWORK FEATURES Part code N-PBX-120 N-PBX-121 N-PBX-122 N-PBX-123 N-PBX-124 N-PBX-125 N-PBX-126 Description Calling Line Identity Presentation (CLIP) Calling Line Identity Restriction (CLIR) Outgoing Calling Plan Incoming Calling Plan Call Forwarding Call Forwarding Advanced Anonymous Call Rejection SIP TRUNK USER MOBILITY PACK Part code N-PBX-130 Description SIP Trunk User Mobility Pack
17 SECTION 5: CHARGING POLICY CHARGING STRUCTURE SIP Trunk service charges are structured as follows: One time service setup charge Fixed recurring monthly charge calculated based on the number of contracted SIP channels Per minute charges for voice calls Fixed monthly recurring charges for optional services Details of the actual prices for each element are detailed in the SIP Trunk service master price list. INVOICING SIP Trunk services will be invoiced on the 6 th of every month. BILLING Billing is post-paid. Reconciliation is done daily, using a process to verify that all CDRs have been collected. A customer can access the InTechnology InForm portal to review all unbilled data. By default all calls will be billed to individual users. Alternatively, if required the service can be configured to bill to a Charge To number, which is equivalent to an ISDN Main Bill Number (MBN). The final billing setup and requirements will be agreed and tested during the setup of the service. CALL DETAIL RECORDS Call Detail Records (CDRS) will be produced for all calls, rated daily and made available for the customer to review and download via InTechnology s online Inform portal. REPORTING The following billing reports are available for customers to view and download via the Inform web portal. Itemised Monthly Service Charges Itemised Unbilled Calls Usage By Site/Sub-site/User Top Ten Spend Analysis
18 SECTION 6: SERVICE CAPABILITIES & LIMITATIONS CAPACITIES PBX Connect SIP service can be deployed with a minimum of 20 channels. Subsequent additional channels can be added in increments of 1 SERVICE LIMITATIONS The maximum number of simultaneous PSTN calls that can be made using the service is limited to the amount of available bandwidth on the customer WAN connection The service can only be delivered via managed InTechnology network circuits of adequate bandwidth The service cannot be delivered over ADSL Max, ADSL 512kbps, SDSL or any form of bonded xdsl circuit The service is delivered subject to successful completion of interoperability testing FEATURE LIMITATIONS The following ISDN type Network & Calling Features may not be supported. If any of the listed features are required, it must be tested and validated during the initial Interoperability phase. Connected Line Identity Presentation (COLP) Call Deflection Caller Redirect Customer controlled call forwarding of calls (using the PBX) Malicious call indication Data services are currently not supported via the service Signalling between PBXs is not supported using the service Fax is not supported; this includes analogue or ISDN fax machines connected via the PBX or directly to the gateway Analogue devices such as modems, bank machines and franking machines are not supported LIMITATIONS WITH TRANSFERING & FORWARDING CALLS Call Quality on Transferred and Forwarded Calls PBX Connect SIP The conversion from G.711 (PCM) to G.729a (also called transcoding) compresses the call, reducing the bandwidth required across the network to ~50kbps per call. In certain call scenarios, calls can end up being transcoded from G.711 (PCM) to G.729a more than once. In this event, call quality will become degraded. InTechnology cannot provide support for call quality issues on calls that have been transcoded more than once. Please see the Supported and Unsupported Call Scenarios section of this document. Note: It is mandatory that G.729a codec must as a minimum be supported, if not the preferred codec. Number Presentation on Forwarded Calls
19 The customers PBX must supply the number that diverted the call to the network, if the PBX fails to do this it must send the pilot number for all diverted calls. This is not applicable to transferred calls. Supported and Unsupported Call Scenarios This section applies to: PBX Connect SIP service where transcoding between G.729a and another codec occurs on the customer PBX (i.e. the handsets are not running G.729a) G.729 G.729 G.711 PSTN InTechnology ADSL User G.729 G.729 CPE CPE Customer A Customer B IP-PBX Ethernet LAN IP-PBX Ethernet LAN G.711 G.711 IP PHONES IP PHONES Figure 2: Network Call Scenario InTechnology will not provide support for call quality issues for call scenarios that are not stated as supported in this section of the service definition. SUPPORTED CALL SCENARIOS The PBX may make calls to the PSTN, to Unity IP Voice* users and to another PBX connected to the service The PBX may receive calls from the PSTN, from Unity IP Voice users and from another PBX connected to the service The PBX may receive calls from the PSTN that have been transferred or forwarded by a Unity IP Voice user Calls made by the PBX to Unity IP Voice users may be transferred or forwarded by the Unity IP Voice user to the PSTN, to other Unity IP Voice users or to another PBX connected to the service *Unity IP Voice is InTechnology s hosted IP Telephony service
20 EXCEPTIONS Inbound calls may not be transferred or forwarded by the PBX, except to other internal telephone extensions connected to the same PBX Where more than one PBX is connected to the service, calls made between the PBXs may not then be transferred or forwarded OVERVIEW OF SUPPORTED & UNSUPPORTED CALL SCENARIOS Call Scenario Description PBX to PSTN A call is made from the PBX to the PSTN Yes PSTN to PBX A call is made from the PSTN to the PBX Yes PBX to Unity A call is made from the PBX to a Unity IP Voice user Yes Unity to PBX A call is made by a Unity IP Voice user to the PBX Yes PSTN to PBX to PSTN PSTN to PBX to Unity PSTN to Unity to PBX PBX to Unity to PSTN Unity to PBX to PSTN Unity to PBX to Unity PBX to PBX PSTN to PBX to PBX PBX to PBX to PSTN PBX to PBX to Unity Unity to PBX to PBX PBX to Unity to PBX PSTN to PBX to PBX to PSTN A call from the PSTN is transferred or forwarded by the PBX to another party on the PSTN A call from the PSTN is transferred or forwarded by the PBX to a Unity IP Voice user A call from the PSTN is transferred or forwarded by a Unity IP Voice user to the PBX A call from the PBX is transferred or forwarded by a Unity IP Voice user to another party on the PSTN A call from a Unity IP Voice user is transferred or forwarded by the PBX to another party on the PSTN A call from a Unity IP Voice user is transferred or forwarded by the PBX to another Unity IP Voice user A call is made from a PBX connected via the service to another PBX connected via the service A call from the PSTN is transferred or forwarded by the PBX to another PBX also connected via the service A call made from a PBX connected via the service, to another PBX connected via the service, is transferred by the 2 nd PBX to another party on the PSTN A call made from a PBX connected via the service, to another PBX connected via the service, is transferred by the 2 nd PBX to a Unity IP Voice user A call made from a Unity IP Voice user to a PBX connected via the service is transferred or forwarded to another PBX connected via the service A call made from a PBX connected via the service is transferred or forwarded by a Unity IP Voice user to another PBX connected via the service A call made from the PSTN is transferred or forwarded by a PBX connected via the service to another PBX connected to the services. This call is then transferred or forwarded by the 2 nd PBX to another party on the PSTN Call Quality Supported No No Yes Yes No No Yes No No No No Yes No
INTECHNOLOGY S SIP TRUNKING SERVICE DEFINITION SD032 v1.7 Issue Date 26 TH July 2013 Revision Control Revision Author Change Date SD032 V1.2 Phil Dyson Updates after review 15 th April 2011 August 2011
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EarthLink Business SIP Trunking Switchvox SMB 5.5 & Adtran SIP Proxy Implementation Guide Publication History First Release: Version 1.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed
nexvortex SIP Trunking Implementation & Planning Guide V1.5 510 S PRING S TREET H ERNDON VA 20170 +1 855.639.8888 Introduction Welcome to nexvortex! This document is intended for nexvortex Customers and
PSTN Calling & Network Features This document details the range of optional Calling and Network features which can be added to Pink Connect Analogue and ISDN lines to provide additional functionality and
Title Series Managing IP Centrex & Hosted PBX Services Date July 2004 VoIP Performance Management Contents Introduction... 1 Quality Management & IP Centrex Service... 2 The New VoIP Performance Management
1. What is VoIP/ IPT?...1 2. Should I use a Hosted or DIY IP Telephony service??...2 3. Why should you now seriously consider using VoIP?...2 4. Why should I use BT?...3 5. Why should I use BT for Hosted
Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya
5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,
Cisco Unified Communications 500 Series IP PBX Provisioning Guide Version 1.0 Last Update: 02/14/2011 Page 1 DISCLAIMER The attached document is provided as a basic guideline for setup and configuration
Integrated Voice Service Guide Save money and maximize bandwidth efficiency Version 201009 TABLE OF CONTENTS TABLE OF CONTENTS...2 PRODUCT OVERVIEW...3 WHAT IS INTEGRATED VOICE?...3 HOW IT WORKS...4 BASE
Conditions for ICT Partner Solutions Service Schedule for BT Cloud 1. Provision of Service The Service will be provided by BT to the Customer using BT s Supplier. For the avoidance of doubt no contractual
SIP Trunking, ITSP Checklist Note: This is not an official document of any kind so please do your own homework and research and also note that Vocale Ltd (owner of the SIP School) does not accept any responsibility
Contents 1 Introduction... 2 2 PBX... 3 3 IP PBX... 4 3.1 How It Works... 4 3.2 Functions of IP PBX... 5 3.3 Benefits of IP PBX... 5 4 Evolution of IP PBX... 6 4.1 Fuelling Factors... 6 4.1.1 Demands from
Service Guide Save money and maximize bandwidth efficiency Learn More: Call us at 877.634.2728. www.megapath.com Table of Contents Product overview... 3 What is integrated voice?... 3 How it works... 4
MITEL SIP CoE Technical Configuration Notes Configure MCD 4.1 for use with SKYPE SIP Trunking SIP CoE 10-4940-00120 NOTICE The information contained in this document is believed to be accurate in all respects
RENTAL OF UNLIT TELECOMMUNICATIONS FIBRE Core optic fibre pair rental: use of unlit optic fibre pair between any switching facilities Cross Connection fibre pair rental: use of an unlit optic fibre pair
SIP Trunking Christina Hattingh Darryl Sladden ATM Zakaria Swapan Cisco Press 800 East 96th Street Indianapolis, IN 46240 SIP Trunking Contents Introduction xix Part I: From TDM Trunking to SIP Trunking
Jargon Buster Access Server An Access Server or Network Access Server connects devices to a Local Area Network (LAN) or Wide Area Network (WAN). Internet Service Providers are able to provide customers
Creating the Unified Multi-Service Demarcation Point Powered by AudioCodes Multi-Service Business Gateways (MSBG) The Challenge Enterprise organizations building their ICT infrastructure face many challenges.
Receptionist-Small Business Administrator guide What is it? Receptionist-Small Business works with your desk phone, soft phone, or mobile device so you can control calls, monitor the lines of employees,
ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.
= UNITY MEET ME CONFERENCING SERVICE DEFINITION SD051 v1.0 Issue Date 29th June 2012 Unity Meet Me Conferencing Service Definition Service Summary Unity Meet Me Conferencing is InTechnology s hosted audio
Convergence: The Foundation for Unified Communications Authored by: Anthony Cimorelli, Senior Product Marketing Manager Onofrio Norm Schillaci, Principal Sales Engineer Michelle Soltesz, Director, Marketing
MITEL SIP CoE Technical Configuration Note Configure MiVB for use with Netcall Telecom Liberty SIP Trunking SIP CoE 14-4940-00338 NOTICE The information contained in this document is believed to be accurate
SIP Matters at 360 Solutions Flexible, resilient and cost effective call routing for your voice traffic 360 Solutions What is SIP Trunking? Our SIP Trunking solution will connect your site directly into
TXI Telephony Product Book 2012 2012 PRICING, HOW TO This section details the common pricing scenarios for the following: Receptionist Call Centre SIP Trunking Fax to Email Messaging Receptionist: Receptionist
Configuring Interactive Intelligence ININ IP PBX For tw telecom SIP Trunking service USER GUIDE Version 1.0 August 23, 2012 Copyright 2012 by tw telecom inc. All Rights Reserved. This document is the property
SIP Trunking Guide: Get More For Your Money 07/17/2014 WHITE PAPER Overview SIP trunking is the most affordable and flexible way to connect an IP PBX to the Public Switched Telephone Network (PSTN). SIP
Telappliant Hosted Edition is the next generation call management solution providing ultimate flexibility and features, all via a high speed Internet connection. Hosted Edition provides all of the benefits
N3 Programme N3 Voice Services There are now in excess of 30,000 NHS connections on N3, which provide a highquality, fully-managed IP infrastructure. The N3 network supports data applications, voice and
BT Hosted IPT (VoIP) What is Convergence? Currently, we see applications on distinct technologies Voice PBX / ISDN / PSTN Data LAN / WAN Video Dedicated VC link What is Convergence? Currently, we see applications
ehealth and VoIP Overview Voice over IP (VoIP) configurations can be very complex. Your network could contain a variety of devices, applications, and configuration capabilities to support voice traffic.
Whitepaper - Guide to migrating to SIP There are both cost savings and productivity benefits to be had by moving from the old circuit switched telephony world to SIP. This short guide describes the steps
Setup Reference Guide for KX-NS1000 to SBC SIP Trunking Method of connection by "WAN Global IP address directly" (i.e. SBC is the Perimeter Router device.) Panasonic IP-PBX (KX-NS1000 Version2 series),
AT&T Connect Video Conferencing Functional and Architectural Overview v9.5 October 2012 Video Conferencing Functional and Architectural Overview Published by: AT&T Intellectual Property Product: AT&T Connect
Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital
The Evolved Office APPLICATION PLATFORM REQUIREMENTS TM 989 Old Eagle School Road Suite 315 Wayne, PA 19087 USA 610.964.8000 www.evolveip.net Release: 16.0 Table of Contents The Evolved Office Assistant
MITEL SIP CoE Technical Configuration Notes Configure MCD 6.X for use with babytel SIP trunks SIP CoE 13-4940-00266 NOTICE The information contained in this document is believed to be accurate in all respects
G Cloud Service Description Premier SIP Trunk Software as a Service October 2015 Premier Choice Telecom has been a trusted provider of communications solutions for businesses for over 15 years. Our clients
MITEL SIP CoE Technical Configuration Notes Configure the Mitel 3300 MCD 10.1 with the MCD 4.1 SP1 for use with Integra Telecom NOTICE The information contained in this document is believed to be accurate
Frequently Asked Questions about Integrated Access Phone Service How are local, long distance, and international calls defined? Local access transport areas (LATAs) are geographical boundaries set by the