1 One Voice SIP trunking Technical Outline for Customers Issue 1.2, 11 March 2014 BT One. Communications that unify
2 Overview BT One What One Voice SIP trunking is all about The Proposition The Call Service The Solution The Coverage BT One. Communications that unify 2
3 The Proposition One Voice SIP trunking Connects your Locations in several Countries via the Global BT Network Provides a Native SIP Trunk Access to connect your PBX/UC Systems, with optional TDM/H.323 support Protects your communications Through dedicated MPLS VPN connectivity to your sites, disconnected from the Internet and protected against any foreign access through 3rd party Through multiple options to support redundant accesses and redundant systems on your side. Integrates both Public PSTN and Private VPN Telephony on the same Accesses With Direct Dial-In access from any national and international origination to your locations; you can keep your existing phone numbers for your locations or get new ones from BT With Full Access to regular national and international fixed, mobile and satellite voice network services, domestic inbound call services as well as domestic short code services With Private Telephony among your locations within your own private One Voice global VPN dial plan Complies to the Legal & Regulatory Requirements and fulfils e. g. your Rights for For Privacy: Place anonymous calls yourself and control incoming anonymous calls For Call Control: Control calls of your users to specific PSTN number types, like Premium numbers For Emergency Calls: Call any emergency number in your country and always get safely connected to the right local emergency service responsible for your location
4 The Call Service One Integrated Telephony Service One Voice SIP trunking provides Voice and Fax/Data Call Service for Public and Private Telephony The Public Telephony Service is suitable to replace your existing PSTN service. It comprises: Inbound Calling to the public phone numbers for your locations (Direct Dial-In) from any national or international public network origin. Your phone numbers are number ported to BT, or you get new numbers from BT if desired. National Outbound Calling to the Public Telephony Network of the country where your respective location is situated. These are calls to destinations in the Fixed and Mobile Networks and to Inbound Service Numbers, as well as to domestic Short Code Services incl. Emergency Services International Outbound Calling to Fixed and Mobile Networks as well as to Inbound Services of the desired country (as far as callable from abroad), satellite networks and generic international services. The Private Telephony Service makes your internal telephony more effective. It comprises: On-Net Calling between your national and international locations within your own private BT Global VPN dial plan. The VPN access via SIP trunking complements the TDM-based access variants. Your SIP trunking locations can be integrated seamlessly into your existing VPN service. Public and Private Calls can be placed equally via the same lines with very user-friendly Dialling Rules.
5 Other BT Networks The Solution Central SIP Trunking Host Platform Your IP PBXs BT Global Managed Voice Network Local Voice POP (SBC) BT MPLS IP Access Network National PSTN Networks International PSTN Networks Your IP PBXs are connected to the BT Global Managed Voice network which provides the World-wide Voice Services of BT and connects to the National and International PSTN Networks as well as to other BT networks. Your IP PBXs access our network at Local Voice POPs which contain in the first place a Session Boarder Controller (SBC = firewall). The IP connectivity is provided by the BT MPLS network. Internally, the accesses terminate at a Central Host Platform that maps your phone numbers to your access trunks. It organises the call service in both call directions. This system talks SIP to your PBXs.
6 The Coverage Current Coverage of One Voice SIP trunking Europe Netherlands UK (launch 2014) Belgium France Germany Turkey Switzerland America Africa United States South Africa Asia Pacific Australia (launch 2014) Hong Kong (2014) Singapore (launch 2014 with restrictions at day 1) The rollout to further countries is currently being prepared.
7 Details BT One How One Voice SIP trunking works Building the Access Defining the SIP Trunks Connecting Your Systems Placing PSTN and VPN Calls Bringing it Live BT One. Communications that unify 7
8 Details BT One How One Voice SIP trunking works Building the Access BT One. Communications that unify 8
9 Standard Access via BT IP Connect (BT MPLS) Local Voice POP BT MPLS with dedicated VPN for each customer Your Site A Handover Points Your Site B The BT MPLS network is used to provide the connectivity between your sites and the local BT Voice POP that serves the country where your sites are situated. BT IP Connect is the related BT product. It provides MPLS networking as a managed service. An MPLS VPN is created within this MPLS network which contains your sites and the BT Voice POP that needs to be connected. No other customer or any foreign party can access this VPN. Per country, one VPN is created. The access to the MPLS network is realised at the nearest BT MPLS POP where a provider edge or PE router is available. At your site, an customer edge or CE router will be installed that connects to the PE router at the BT MPLS site. This CE router is the handover point where you connect your systems (at according Ethernet ports).
10 Resilience Option Redundant IP Accesses Local Voice POP Your Site All platforms and components of the BT Voice Network and the BT MPLS IP access network are built consequently in a redundant way, preferably using High Availability redundancy with hot-standby failover. In this way we ensure a very high availability of the network platforms. In order to improve the resiliency of the solution further, also the access connectivity to the BT MPLS network can be made redundant. The access for an individual site will then be built resilient with redundant PE/CE routers and redundant cabling between them.
11 Resilience Option Geographically Redundant IP Accesses Local Voice POP Your Site Optionally, the CE routers on your site can be connected to PE routers in different locations as well. Should one IP access POP fail become entirely unavailable for any reason damage to the site like fire, loss of power supply, loss of transmission links then the operation continues via the access to the secondary IP POP.
12 Resilience Option Redundant IP Accesses Local Voice POP Own WAN Your Site A Your Site B In case of several sites, each site can of course have redundant accesses as well This may be independent redundant accesses for each site Or you may connect both sites via an own WAN and use the access of the other site as redundancy Or you combine both options just as you like!
13 Resilience Option PBX Redundancy Local Voice POP Within your own infrastructure, you may also want to apply redundancy. An option is that you let the PBX of one site act as a backup PBX for the other site, and vice versa In this case, we configure additional SIP access trunks for each site to the respective backup PBX. For each site, the primary and backup trunks form a trunk group that jointly serve their related site. For the call service provided to each site it makes no difference whether the calls are exchanged with the primary or backup PBX.
14 Resilience Option Dual Homing Local Voice POP Your Site A Secondary Voice POP Your Site B For selected countries, BT operates several Voice POPs. In this case, your systems can be connected to two independent, geographically separated Voice POPs. This option is usually called Dual Homing. Both your PBX/UC system and the BT SIP trunking host know the paths via both POPs. If one POP fails, simply choose the path via the secondary POP. For the call service it makes no difference whether the calls travel via the primary or secondary POP. Dual Homing is currently available in: Germany, Netherlands, USA It is currently being extended to: Belgium, France, Switzerland
15 A developing Alternative: Access via BT Ethernet Connect Local Voice POP BT Ethernet Network Your Site As an alternative to the MPLS-based IP access, BT is developing the Ethernet-based network services and their use as access technology for the BT One Voice services. The related IP product is BT Ethernet Connect. In the same way as in case of the MPLS access, BT will provide the connectivity plus an access device to your site. This access device (NTE) will be the handover point. Similar to the MPLS access, the Ethernet access provides a protected IP connection between your site and the Voice POP that is not shared with or accessible for any other customer or foreign party. BT Ethernet access for One Voice SIP trunking is possible in: Germany, Netherlands, Belgium, United Kingdom
16 Details BT One How One Voice SIP trunking works Defining the SIP Trunks BT One. Communications that unify 16
17 SIP Trunks Central SIP Trunking Host Platform Local Voice POP (SBC) Location 1 Location 2 SIP Trunks are distinguished logical connections between your PBX system(s) and the BT Voice POP(s). Each SIP trunk is actually a unique combination of peer addresses on either side (PBX and SBC) which talk to each other. (Peer address = IP address + UDP/TCP port number used for SIP) The SIP trunks correspond to the physical access lines (E1, S 2M, n S 0 ) in the TDM world, but without being bound to a specific copper or fiber line. The physical bandwidth is provided by the IP (MPLS/Ethernet) cloud. Within the BT network, it is actually the Central SIP Trunking Host where the SIP trunks are hosted; the POP SBC acts as an entry gate at the network border. It prolongs your SIP trunks internally to the Trunking Host. The Trunking Host is the entity that really talks SIP to your PBXs, and that will organise your calls! The Central SIP Trunking Host needs at least 1 SIP trunk to each of your PBXs.
18 Profile of a SIP Trunk On the Central SIP Trunking Host, each SIP trunk is associated with a Profile that defines how the trunking host needs to work with your PBX system and what the call service needs to look like. Profile parameters are, for example: Typical Profile Parameters of a One Voice SIP trunk and the Call Service that it provides Remote End Point Traffic Capacity Location Number Format, Dial Plan Feature Settings What is the peer address of your system to which we send the SIP messages? Via which Voice POP can it be reached, and which alternative POP can eventually be used (Dual Homing)? What are the traffic limitations of this trunk specified in in/out/bothway direction, measured in numbers of parallel call sessions or amount of IP bandwidth by the call sessions? Which location is associated with this SIP trunk? What are PSTN and VPN numbers of the location, the user extensions, the default extension? In which Emergency Area is your location situated that is: which Police, Ambulance or Fire Service station needs to get your calls? Or can we alternatively safely identify the location and all its specific data by the calling phone number (CLI) that your system sends? Does your system exchange public PSTN or private VPN number formats? To which country or numbering plan relate all the PSTN phone numbers (calling, called etc.) that your system sends and expects? (Is it UK, France, North America, Hong Kong, other?) Shall calls to certain number types (like Premium Rate) be blocked? Shall CLI Restriction be applied permanently on outgoing/incoming calls? Shall incoming anonymous calls be rejected?
19 Groups of SIP Trunks On a higher level, it is also possible to group several SIP Trunks. This is intended for priority / backup constructions and for load distribution purposes. Profile of a Trunk Group Trunks in the Group Traffic Capacity Routing Mode Load Distribution Priority Routing Which individual SIP trunks are combined in this trunk group? What are the traffic limitations for the whole of the trunk group? Shall the traffic be distributed over the individual trunks (Load Distribution), or shall the trunks be filled in a specific order (Priority Routing)? Which share of the traffic shall each trunk receive? In which order shall the trunks be used? The next trunk will be chosen when the current one is either full or currently not usable (trunk down, error from PBX etc.). Note that these trunk groups are independent of Dual Homing constructions. In case of Dual Homing, the same individual SIP trunk is replicated via a second Voice POP. These doubled trunks can also be members of Trunk Groups. These Trunk Groups can be applied very flexibly, for example for distributing the calls to a cluster of PBX/UC servers on your side, or for overflowing traffic from a primary server to a secondary/backup server that can serve the respective locations alternatively.
20 Details BT One How One Voice SIP trunking works Connecting Your Systems BT One. Communications that unify 20
21 PBX/UC Systems working with One Voice SIP trunking Features of the One Voice SIP trunking Service Alcatel Avaya Cisco Microsoft Mitel OmniPCX Call Management System (CMS) Cisco Unified Call Manager Microsoft Lync Open Communications Server (OCS) 3300MX Siemens OpenScape, HiPath 8000 Sonus NBS5200 The table above shows a number of examples of PBX types that One Voice SIP trunking supports today. The list of supported manufacturers and systems will steadily grow further. Major types also undergo a certification test in the BT test lab. The following slides briefly discuss typical connection scenarios using the examples of Cisco Call Manager and Microsoft Lync, and show how also H.323- and TDM-based PBX systems can be connected.
22 Connecting Cisco Unified CallManager (CUCM) Access Router CUCM Local Voice POP Your Cisco Environment Cisco Unified CallManager is a certified IP-PBX/UC system that can be connected via One Voice SIP trunking. You can connect your own CallManager platform via One Voice SIP trunking, or use our BT One Enterprise Cisco product which equally applies One Voice SIP trunking for the access to public and private telephony services. CallManager is capable to support SIP-based network connections natively and directly without the need of additional gateways. The compact Call Manager version Call Manager Express is likewise supported. Older releases of Cisco CallManager using H.323 trunks can be connected on a non-standard basis. For this, the local BT Voice POP will need to be equipped with a H.323/SIP interworking function see below.
23 CallManager for Multiple Locations Voice POP Country A Access Router CUCM Voice POP Country B Shared CUCM CallManager is a good example of a PBX system that can operated multiple locations as a central PBX. It can even be used sensibly for serving locations in neighbour countries (when distances and capacities allow this). Such systems usually group the user phones by the locations where they are situated, being aware how much IP bandwidth is available towards each location, knowing which location-specific dial plans need to be applied, which SIP trunks can be used to hand over the PSTN calls to the public network, and finally how the identity of the calling user needs to be turned into a public CLI for the location where the caller or phone is situated. One Voice SIP trunking supports such central PBX constructions. The access will be installed to the site where the central PBX is situated, and one or several SIP trunks will be configured between the BT Voice POPs of the involved countries and CallManager. In a simple approach we configure 1 SIP trunk for each location as if each had its own PBX. One Voice SIP trunks can also be shared by several locations. The preconditions are that the locations are in the same country and that the location of the caller can be identified uniquely by the CLI that CallManager transmits. This is mandatory, and it must be ensured also in case of CLI Restriction and Call Forwarding.
24 Connecting Microsoft Lync Access Router Media Server Lync Local Voice POP Your Lync Environment Microsoft Lync is a certified IP-PBX/UC system that can be connected via One Voice SIP trunking. You can connect your own Lync platform via One Voice SIP trunking, or use our BT One Enterprise Lync product which equally applies One Voice SIP trunking for the access to public and private telephony services. Microsoft Lync is capable to support SIP-based network connections natively. However, please be aware that Microsoft applies their own, proprietary audio codec RTaudio that is not supported by regular telephony networks and services. Therefore, the audio streams need to be transcoded from RTaudio to a standard codec like G.711 or G.729. For this purpose, Microsoft requires the use of a Media Server or alternatively another IP- PBX as a gateway towards the telephony network. This is key in Microsofts architecture for Lync. A concept discussion can be found here: Suitable alternative media servers and IP-PBXs can be found here: see: Qualified IP-PBXs and Gateways. Microsofts reference provider for Media Servers is AudioCodes.
25 Voice POP Country A Distributed Lync Environments Access Router Media Server Central Lync Servers Voice POP Country B Access Router Media Server Call Signalling via Central Servers Call Media directly to Media Servers Microsoft Lync is designed to provide enterprise UC services in distributed environments, that is, for companies with many sites and locations in the same country or even worldwide. In such an environment, many server roles may be centralised, even in one location that serves many countries. Also the Mediation Servers can be shared, but they can be centralised only to a limited extend. These servers carry the voice media streams that must be exchanged between the user phones and the BT Voice POPs serving the respective country where these phones are located. It will usually be very ineffective to send the media streams over a long distance from the phones to a central Media Server in another country or even on another continent, only to send them straight back towards the BT Voice POP in the originating country! Therefore, the mediation servers are typically decentralised, keeping the path between the phones, the Media Servers and the local BT POP short. Typically, each country should have at least one own Mediation Server. For this, BT will deliver the access lines, routers and SIP trunks to the locations of the Mediation Servers.
26 Support of Enterprise SBCs Local Voice POP Access Router Enterprise SBC PBX/UC Systems In the same way as BT on the network side, you may want to operate an Enterprise SBC that decouples your PBX/UC infrastructure from any external components and accesses outside your organisation and implements a well-defined and well-controlled entry gate for the same. This can be required due to security considerations, but it may also have very practical advantages to hide your infrastructure behind an SBC. One Voice SIP trunking fully supports this approach. In fact, it makes not much difference if the BT Voice POPs connect directly to a PBX/UC system or if that happens via an intermediate SBC or other SIP gateway. In order to achieve a cost-efficient solution, your Enterprise SBCs will typically be installed in central locations. BT will then build the accesses to these sites where your SBCs are situated, and hand over all SIP trunks there. A redundancy of the SBCs can easily be integrated into an overall access resilience concept as shown in the previous chapter. The SIP trunks will then be replicated accordingly for terminating at the set of your SBCs. The traffic over the redundant SBCs and SIP trunks can be distributed in load sharing or priority mode. Well-known Enterprise SBCs that we have operated successfully with One Voice SIP trunking are, for example: ACME Packet Net-Net SD, Genband S3 (now Quantix), Cisco Cube, Avaya SIPERA
27 Connecting H.323 and TDM/ISDN Systems H.323 IW H.323 Local Voice POP SIP Access Router H.323 PBX TDM IW Access Router TDM PBX One Voice SIP trunking is a native SIP access product, meaning that it applies the SIP protocol for the control of VoIP calls. However, there can be situations where you may want to use One Voice SIP trunking with an existing H.323 PBX like earlier versions of Cisco Call Manager or with an existing TDM PBXs. A reason can be to introduce the new access while planning to replace a legacy PBX system at a later date. For such cases, One Voice SIP trunking can provide the necessary interworking to SIP: VoIP calls using the H.323 protocol are interworked to SIP by the Session Border Controller (SBC) of our Local Voice POP. TDM (ISDN/analogue) calls will be interworked by the Access Router at your site. For this, a different router model will be selected that provides the necessary ISDN or analogue interfaces and a SIP/TDM Interworking function.
28 Details BT One How One Voice SIP trunking works Placing PSTN and VPN Calls BT One. Communications that unify 28
29 Choice of the Number Format between PBX and Network For each SIP trunk that connects to your PBX/UC systems, we configure which Number Format the exchanged phone numbers shall have. In the first place, one basic decision is necessary: Shall the phone numbers exchanged between your PBX and the network be: Public PSTN phone numbers, or Private VPN phone numbers? What is the difference The standard number format will usually be the PSTN format. The official national dial plan of the country where your location is situated will be applied, which allows you to dial national and international PSTN destinations and domestic short codes including emergency just like from a normal subscriber line. Additionally, private VPN phone numbers can be dialled after a special VPN trunk prefix. One Voice SIP trunking defines a unified trunk prefix for this in most countries this prefix is 88. The alternative number format is the Global VPN format. This format is useful when existing TDM-based Global VPN accesses shall be migrated to SIP trunking. Existing PBX dialling rules and dial plan configurations can simply be taken over. The agreed number format must be configured both on the PBX/UC system and in the network on the Central SIP trunking host. It is then applied to all exchanged phone numbers ie to called, calling and redirecting numbers.
30 Dialling Rules EMEA (NL, BE, FR, CH, UK, DE, TR, SA) Standard Dialling Rules based on domestic PSTN formats International PSTN call: 00 + country code + area/subscriber no. (00) 44 National PSTN call: 0 + area code + subscriber number (0) 20 Domestic short code call: Short code without any prefix etc. 112 Private Global VPN call 88 + private site ID + user extension (88) Alternative Dialling Rules based on Global VPN formats Private Global VPN call Private site ID + user extension National PSTN call 00 + own country code + area/subscriber no. (00) International PSTN call 00 + other country code + area/subscriber no. (00) 44 20
31 Dialling Rules North America Standard Dialling Rules based on domestic PSTN formats International PSTN call: country code + area/subscriber no. (011) 44 National PSTN call: 1 + area code + subscriber number (1) 212 (New York) Domestic N11 short code call: Short code without any prefix etc. 911 Private Global VPN call 88 + private site ID + user extension (88) Alternative Dialling Rules based on Global VPN formats Private Global VPN call Private site ID + user extension National PSTN call own country code + area/subscriber no. (011) International PSTN call other country code + area/subscriber no. (011) 44 20
32 Dialling Rules Other Countries The Dialling Rules for other countries follow the same basic rules: Generally, we apply the default international and national trunk prefixes as defined in that country. Local dialling is not supported see below for comments on this subject. Short codes and other domestic number types are dialled without any trunk prefix Private VPN numbers are dialled with a special VPN trunk prefix. This is usually the 88, but especially in countries where no national trunk prefix is used we need to choose a different code to avoid code conflicts. Example: Dialling rules for Australia other international prefix, inbound services in domestic range 1 International PSTN call: country code + area/subscriber no. (0011) 44 Domestic standard PSTN call: 0 + area code + subscriber number (0) 29 (Sydney) Domestic inbound call: 1 + service number (tollfree) Emergency call: Short code without any prefix etc. 000 Private Global VPN call 88 + private site ID + user extension (88) Example: Dialling rules for Singapore no national trunk prefix, other VPN prefix International PSTN call: country code + area/subscriber no. (000) 44 Domestic standard PSTN call: Subscriber/service number starting with 3,6,8 or 9 6 Emergency call: Short code 999 Private Global VPN call 44 + private site ID + user extension (44)
33 Local Dialling Keep it Safe and Simple One Voice SIP trunking supports dialling on the national level, but no local dialling. Where users shall have the option to dial locally for calls within the geographic code area, omitting the area code and eventual trunk prefix, this needs to be supported by the PBX with according translation rules respectively site dial plans. This is in fact common practice. Many countries allow local dialling, but by far not all countries. The tendency is that countries go away from local dialling. Where this is allowed, the rules where local dialling is allowed and where not are often inconsistent. The most prominent example is North America, where individual dialling rules are defined for each geographic area. Remark for North America: All national numbers are dialled always with the trunk prefix 1 and area code, including local calls and 555 calls to local directory enquiries. We do not differentiate dial patterns for local and long-distance calls. Alternative Trunk Prefixes, Regional Dial Codes for Cross-border Calls One Voice SIP trunking allows calling all supported national and international services with the standard national and international trunk prefixes. However, it does not support the use of alternative trunk prefixes for placing calls via alternative phone services respectively via other carriers. Examples: France, Australia, Singapore One Voice SIP trunking allows calling all destinations of the national dialling plan. This applies also to overseas areas that have been mapped into the numbering plan of the main land. However, regional cross-border dialling with special dial codes for destinations in a neighbour country is not supported. These need to be called with their international dial codes. Example: Singapore. Global Address Formats, aka E.164 format Some PBX/UC manufacturers prefer the Global Address format ( + followed by the E.164 country code), or even declare this format a standard number format for SIP. However, this format does not allow you to formulate domestic codes (like the emergency short codes 112, 999 or 911, or inbound service codes eg in Australia) without additional context attributes to the number, which are not commonly supported by the manufacturers, yet. In fact, many PBX systems simply do not even support the + character. Therefore, One Voice SIP trunking focuses on the official national number formats. This ensures that all services can be called.
34 Details BT One How One Voice SIP trunking works Bringing it Live BT One. Communications that unify 34
35 Organisational Issues Plan & Build 1. The BTRegional Order Manager will plan with you which access configuration shall be built to each of your sites (based on BT MPLS and/or Ethernet), and which voice configuration shall be implemented on top of this. This comprises, which individual PBX systems shall be connected for your locations, which SIP trunks are needed, and what all the related phone numbers, feature settings etc. shall be. 2. The BT Access & IP Engineering team will build the access lines, access routers to your site(s). 3. The BT Voice Engineering team will then configure the SIP trunk and the voice service and test it together with you. BT provides a test & turn-up document that describes the necessary preparations and the test cases. The test results will be documented in a test protocol; on completion of the tests it will be signed off and handed over to you. 4. When the voice service to your PBX is successfully built, it will be fully activated. When you want use your One Voice SIP trunking access(es) with your existing PBX phone number(s), and these are till this moment operated by another operator, then we will now initiate the number porting process so that your phone numbers are switched over from your previous operator to BT. When this has happened, you receive all incoming calls to your phone numbers via BT and the new One Voice SIP trunk accesses. 5. After that, your new One Voice SIP trunking access is live.
36 Access Configuration Preparation Issues The BT Regional Order Manager will clarify the individual needs with you for: Suitable space to install the access equipment; this should be collocated with your PBX/UC equipment Power supply for the access router equipment. Cabling that needs to be installed for the access lines. SIP Trunks The BT Regional Order Manager will clarify with you: The details of your PBX/UC system: which elements for PBX, Media Servers, SBCs etc. need to be connected, which models (manufacturer, product, software version) do you use. The details for each SIP trunk: IP addresses, UDP/TCP ports for SIP, trunk group and backup/dual Homing configurations, phone numbers associated with each trunk, trunk sizes, voice and fax media codecs, number formats, feature settings eg for CLI Restrictions, specific SIP parameters to be used, etc. Test & Turn-up When the connectivity between your systems and the BT network is established, the SIP trunks will be configured. The BT Voice Engineer will have prepared all this by an agreed day. Likewise, the configurations of your systems and their cable connection to the BT access router should be completed by then. For the turn-up and test, your systems will need to be equipped with test phones (also fax, when applicable). These devices must be able to place and receive calls via the new SIP trunk(s), so that call tests can be carried out with the BT engineer. The turn-up and tests need to be done together with the administrators of your PBX/UC system. For this they need to be available at an agreed day to support the procedure, both for placing and accepting test calls and for eventually analysing the test calls and debugging the configuration of your systems.
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EarthLink Business SIP Trunking Toshiba IPedge Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011
Fax over IP Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary About this document This document describes how Fax over IP works in general
A Light Reading Webinar Session Border Controllers in Enterprise Thursday, October 7, 2010 Hosted by Jim Hodges Senior Analyst Heavy Reading Sponsored by: Speakers Natasha Tamaskar VP Product Marketing
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
61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Abstract These Application Notes describe the procedures
Mediatech 2015 Implementation of Voice over IP and Audio over IP in the Studio environment Wilfried Hecht, Managing Director AVT Audio Video Technologies GmbH Nordostpark 12 90411 Nuernberg Germany Email:
SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG
Dialogic Diva SIPcontrol Software converts Dialogic Diva Media Boards (Universal and V-Series) into SIP-enabled PSTN-IP gateways. The boards support a variety of TDM protocols and interfaces, ranging from
SBC 1000/2000 Configuration Guide with Lync 2013 for Windstream/ LPAETEC SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: April 10, 2015 Revision Date Revised By Comments 0.1 12/03/2015 Roman
Configuration Guide SBC 1000 / SBC 2000 Series Configuration Guide (For Microsoft Lync Server 2013) For use with AT&T s IP Flexible Reach Enhanced Features Service on MIS, MPLS PNT or AT&T VPN Disclaimers
1 SIP Carriers 1.1 Telstra 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can be found
CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,
Skype Connect Product Datasheet Copyright Skype Limited 2011 Discover what Skype Connect can do for your business Provide connectivity between your business and the Skype community. By adding Skype Connect
EarthLink Business SIP Trunking ININ IC3 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011
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
Communications Transformations 2: Steps to Integrate SIP Trunk into the Enterprise The Changing Landscape IP-based unified communications is widely deployed in enterprise networks, both for internal calling
VoIP Logic: Disaster Recovery and Resiliency VoIP Logic: Options for Disaster Recovery and Resiliency The ability to keep telephone systems operating in the event of a service impairment or a catastrophic
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
Software-Powered VoIP Ali Rohani Anthony Murphy Scott Stubberfield Unified Communications Architecture Core Scenarios UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN
07 2015 2 Efficient communication anynode is a Session Border Controller that is entirely a software based solution. It works as an interface for any number of SIP UAs for example, SIP phones and SIP PBXs,
Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user
BlackBerry Mobile Voice System for SIP Gateways and the Avaya Aura Session Manager Version: 5.3 Feature and Technical Overview Published: 2013-06-19 SWD-20130619135120555 Contents 1 Overview...4 2 Features...5
Packetized Telephony Networks Benefits of Packet Telephony Networks Traditionally, the potential savings on long-distance costs was the driving force behind the migration to converged voice and data networks.
EarthLink Business SIP Trunking Cisco Call Manager and Cisco CUBE Customer Configuration Guide Publication History First Release: Version 2.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed
BT One for Communications IP Voice Reseller Deliver the VoIP services your customers demand without costly infrastructure investment or increased management complexity. We manage the technology to enable
An Allstream White Paper Best Practices for deploying unified communications together with SIP trunking connectivity An Allstream White Paper 1 Table of contents Correct identification and maximization
OVERVIEW OPTUS OUTBOUND VOICE SOLUTIONS ANSWER THE CALL TO KEEP COSTS DOWN CONTENTS YOUR VOICE SERVICES NEED TO KEEP UP WITH YOUR BUSINESS 01 EVOLVING AT A PACE THAT SUITS YOU 02 OUTBOUND VOICE SOLUTIONS
AN-SBC-100 Sangoma Session Border Controllers Using DNS SRV to Provide High Availability Scenarios Contents 1. Sangoma Session Border Controllers - High Availability Solution...1 2. What is DNS SRV?...1
Configuring the Synapse SB67070 SIP Gateway for Broadvox GO! SIP Trunking 2012 Advanced American Telephones. All Rights Reserved. AT&T and the AT&T logo are trademarks of AT&T Intellectual Property licensed
VitalPBX Hosted Voice That Works For You Vital Voice & Data s VitalPBX VVD Hosted PBX solutions provide you with the stability of a traditional telephone PBX system and the flexibility that only a next
Siemens OpenScape Voice V7 SIP Connectivity with OpenScape SBC V7 to Integra SIP Service February 2013 Page 1 of 47 Document History Date Version / Issue Modified by Section(s) Affected High-Level Description
Whitepaper: Microsoft Office Communications Server 2007 R2 and Cisco Unified Communications Manager Integration Options Document Summary This document provides information on several integration scenarios
Configuring the Sonus SBC 5200 with Level(3) SIP Trunks and Lync 2013 Application Notes Last Updated: July 23, 2014 Sonus Equipment Type Version SBC 5200 SBC 5000 04.00.01R0 3rd Party Equipment Type Version
White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary
Course 20337B: Enterprise Voice and Online Services with Microsoft Lync Server 2013 Course Details Course Outline Module 1: Voice Architecture This module introduce Enterprise Voice features of Lync Server
Service Schedule for BT MeetMe with Dolby Voice 1. INTERPRETATION The following definitions shall apply, in addition to those in the Conditions for BT Conferencing Services. Account means the identifier
BroadSoft Partner Configuration Guide Microsoft Lync 2010 SIP Trunking August 2012 Document Version 1.6 9737 Washingtonian Blvd Suite 350 Gaithersburg, MD USA 20878 Tel +1 301.977.9440 WWW.BROADSOFT.COM
Colt VoIP Access 2010 Colt Technology Services Group Limited. All rights reserved. Business requirements Are you looking for ways to simplify management of national or even international voice services
MITEL SIPCoE Technical Configuration Notes Configure Inn-Phone SIP Phone for use with MCD SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not
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