10. IP Telephony

Technological innovation is changing the way we communicate. This time it is coming in the form of changing the way telephone calls are transmitted. It brings with it several new capabilities that change the meaning of the phrase telephone call through the use of Voice over Internet Protocol (VoIP). Basically, VoIP means "voice transmitted over a packet data network." VoIP is often referred to as IP Telephony because it uses the IP protocols to make possible enhanced voice communications throughout the world, wherever IP connections have been delivered. IP Telephony unites a company’s many locations- including mobile workers- into a single converged communications network. Telephony calls using VoIP go above and beyond what’s been possible in the past. When it comes to placing telephone calls, VoIP provides a range of support services and features unequalled in the world of telephony, but above all delivers them at low cost.

How Does VoIP Work?

Voice over Internet Protocol means basically what the acronym states: Voice travels over an Internet Protocol.

Internet Protocol refers to the type of rules that the network uses to send and receive signals. IP Telephony works by converting voice communications into data packets. Conveniently, it runs on the popular Ethernet

LAN (local area network) technology, which currently supports over 96 percent of the worlds companies’ LANs.

Circuit-switched or Time-Division Multiplexed Telephony

Before digital networking with the Internet took off, everyone had to use the "Plain Old Telephone

Services" (POTS). These run over a network called the Public Switched Telephone Network (PSTN). The PSTN has been around since the telephone was invented in either analog or digital form using circuit switched technology where the telephone call gets exclusive bi-directional use of a wire – or circuit – while the call is in progress. Because the circuit is exclusive to each conversation, PSTN and private branch exchanges (PBXs) must be sized to cope with peak demand and have enough circuits available for all expected conversations.

This is not a flexible approach and results in a lot of infrastructure investment that the telephone companies need to recoup, via the cost of access charges and calls. The Internet has changed this – where data services have driven down access charges and allowed voice to "travel for free" over a multipurpose data network.

Packet-Switched Telephony

Unlike circuit-switched connections, which always require use of dedicated bi-directional circuit for the duration of a call, VoIP technology has enabled telephony and other new and novel features and services to run over fixed and wireless networks including private local area networks. These newer network types use packetswitched protocols. Packet-switched VoIP puts voice signals into packets. Along with the voice signals, VoIP packets include both the sender’s and receiver’s network addresses. VoIP packets can traverse any VoIPcompatible network. Along the way, they can choose alternate, shared paths because the destination address is included in the packet. The routing of the packets is not dependent on any particular network route which means the network provides can provide a reliable service at a fraction of the cost of circuit switched providers.

What Advantage Does IP Office Have?

IP Office can provide support of PSTN, SIP, POTs, digital time division multiplexed telephones AND digital IP telephones all on the same system. This means you don’t have to abandon the past to embrace the future, IP

Office allows all the technologies to co-exist. IP Office connects to the PSTN and to IP trunks (the VoIP equivalent) so providing a "Hybrid" PBX function – where both legacy and future technologies can be used together to minimize operating costs and optimize business communications through both voice and data.

IP Office has digital telephones built on both TDM and IP technology that provide the same user interface offering a flexible choice of solution that can mix, for example TDM telephones in the office and IP telephones at a remote site of at home. With the choice of IP telephones including real and virtual (software) telephones,

IP Office can take communications to a new level.

Buying IP Office allows you choice – you can use the pure POTs or the pure VoIP capabilities of IP Office, or use both at the same time to allow seamless technology transition of your business without the disruption of having to choose between them now.

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IP Office and IP Telephony

In order to make use of VoIP, IP Office uses signaling protocols called H.323 , and Session Initiation Protocol

(SIP) which allow IP Office to establish end-to-end connections for the voice path through the IP network. It ensures each end of the connection is able to transmit and receive voice and provides the network addressing for end to end packet transmission. IP Office also allows for connecting between the different technologies by translating the signals they use, for example an analog telephone may wish to connect to a VoIP destination.

This requires both the signaling and voice transmission to be translated – IP Office does this easily as it contains technology elements called gateways and gatekeepers that enable translations to happen.

With a conventional telephone system you plug your analog or digital TDM telephone into an extension socket connected to your PBX or Key System. With IP Telephony you connect your IP telephone to your IP PBX via the LAN. There are two basic types of IP telephones:

· A physical telephone, which looks very similar to a standard telephone (IP Hard Phone)

· A software application (IP Office Video Softphone) which runs on the user's PC, allowing them to use either a headset/microphone to make/receive calls anywhere they have IP connection

IP telephony has the advantage of allowing extensions to be deployed both locally and remotely through the use of IP routing and IP VPN services.

When making use of IP telephony, there are a number of data centric considerations such as which data types have priority on the IP network when there is contention. This is set with IP/TCP "quality of service" and should not be ignored. In situations where LAN Bandwidth is limited, a quality of service capable LAN switch should be used to ensure voice packets are transmitted with the required priority on the network. If not, the conversation carried over IP appears as broken up (due to delays) or has unacceptable delays introduced in the conversation (latency and jitter). With IP hardphones there is need for Power over Ethernet (PoE), or local phone power supplies to be provided to the telephones as the IP telephones are no longer powered by IP

Office – a list of Avaya approved PoE options is available at the end of this section.

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Gateways, Gatekeepers and H.323 - Technology Overview

IP Office uses the H.323 signaling protocol which has the following architectural components

· Telephones are H.323 service endpoint devices that can support Audio calls. Other types of H.323

devices can support video as part of H.323

· Gateways provide media translation to allow calls to be made to non-H.323 devices, for instance an analog telephone or the public network to connect with a H.323 device

· Gatekeepers control the call processing and security for H.323 devices

· Multipoint Connection Units (MCU) for conferences by adding together media streams

These elements are grouped together in what is known as an H.323 zone (a zone is analogous to a PABX).

Each zone has a single Gatekeeper that can be considered as the brains of the system dealing with call distribution, call control and the management of resources. On power-up, IP telephones, Gateways and MCU make registration requests to a Gatekeeper which then authenticates (accepts or rejects) their request to become a member of the zone. Once accepted, a telephone wishing to make a call sends a call set-up message to the Gatekeeper which then determines how to route the call and will then send an alert to the called telephone, or if the call is to a non-H.323 telephone establish the call via a Gateway within the zone.

The design of IP Telephony systems has been driven by open standards. Digital IP telephones, Gateways and

Gatekeepers all support the H.323 standard and it is this that allows devices from different manufacturers to work together. IP Office has an optional integral Gateway (Voice Compression Modules) and Gatekeeper functionality required to provide a fully functional IP Telephony solution.

SIP - Session Initiation Protocol

IP Office supports SIP, which is a signaling protocol used for creating session oriented connections between two or more locations in an IP network. SIP trunking is a service offered by and ITSP (Internet Telephony

Service Provider) that permits businesses with an IP Office installed to use Voice-over-IP (VoIP) outside the enterprise network by using the same connection as in Internet connection.

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IP Telephony Features

· Gatekeeper

The IP Office gatekeeper allows the registration of up to 384 IP extensions on the IP500, less the number of analog and digital TDM telephones already configured on the system.

· Gateway

The Voice Compression Module provides the H.323 gateway function that allows IP extensions to make calls to other non-IP devices. The maximum number of simultaneous calls is limited by the number of channels available on the Voice Compression Module. IP Office must be fitted with an optional Voice Compression Module to enable IP telephony.

· Silence Suppression

Silence suppression is a technique used to make the best use of available bandwidth, such as the connection over which the caller is listening, not speaking. Silence suppression works by sending descriptions of the background noise, rather than the actual noise itself, during gaps in conversation, thereby reducing the number and frequency of voice packets sent on the network. Background noise is very important during a telephone call. Without noise the call will feel very unnatural and give a perception of poor quality. Ensure that silence suppression configuration matches at both ends of the

SCN trunk for correct operation and also for improved sound quality.

· Compression

IP Office supports a wide range of voice compression standards including G.711, G.729a and G.722.

The method of compression can be either automatically established on a call-by-call basis or configured on an individual extension basis.

G.722 is supported on SIP Trunks, the 9600 Series sets, 96x1 Series Sets, B179 Conference Phones,

IP Office Video Softphone, 1100/1200 IP Phones and 3 rd party SIP phones. G.722 is not supported for

5600 and 1600 Series sets.

· Fast Start

When fast start is supported by an IP extension, this facility reduces the protocol overhead allowing an audio path to be established more quickly.

· Out of Band DTMF

When out of Band DTMF is configured on an IP extension, the extension will signal to the other end of the connection which digits need to be regenerated by a local DTMF generator on behalf of the sending IP extension. This is useful when navigating external voicemail systems and Auto-Attendants.

· Direct Media Path

Direct Media Path allows the speech path between two IP extensions (after call setup) to be routed directly to each other. This allows the IP Office system to free up voice compression resources after establishing the end to end connection, allowing the resources to be used in the most efficient way.

· Auto-Create Extensions

IP Office can automatically create an extension entry for new IP telephones added onto the local area network. In cases where the local area network is not secure this facility can be disabled, but simplifies installation of IP telephone systems

· Fax Transport (Avaya Proprietary)

Fax Transport allows fax calls to be routed over VoIP trunks between IP Office systems on an IP network using a proprietary IP Office transport protocol.

· Fax Transport T.38

IP Office supports the standardized protocol "T.38" for transporting FAX calls between IP Office and

SIP trunks or SIP endpoints. T.38 is only supported on IP500/IP500 V2 systems with IP500 VCM,

IP500 VCM V2 and or IP500 Combo cards. T.38 is not supported on Server Edition systems.

T.38 allows the reliable transmission of Fax messages over a IP channel independent of Codec used for the Voice communication.

Platforms Supported: IP500, IP500 V2 only, must be fitted with VCM32 or VCM64 module.

Trunk Types Supported:

Extensions Supported:

SIP

SIP

T.38 Transport Layers Supported: UDPTL (with optional redundancy error correction).

T.38 Versions Supported: 0-3

Call Types Supported: Voice calls which transition to fax relay on detection of fax tones.

Calls which are negotiated as fax only.

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· T.38 Fax Fallback

SIP Trunks and SIP Extensions can now configure the "Fax Transport Support" to "T38 Fallback" so that outgoing fax calls use T38 fax, but when the called destination doesn't support T.38 and rejects the call, a re-invite is sent for fax transport over G.711. Incoming audio calls that detect fax tones also initiate fax transport using T.38 Fallback. This allows IP Office to support additional deployments where T.38 fax may not be universally available.

· Display Directory Name on In-Bound SIP Trunk Calls

IP Office Manager will now include a configuration option that will allow the selection of either the

CLID or directory name as the default display for inbound calls. In previous IP Office releases only

CLID display was supported over SIP trunks for incoming calls. This enhances IP Office user experience by providing the flexibility to select between personalizing the display with the personal or system directory name, and seeing the default CLID.

· Enhanced CLIR

Enable SIP Trunk PAI and Privacy headers by default, and format "From, PAI and Privacy." This feature allows caller and calling party anonymity while still providing the necessary billing and traceability information (E911 services) to the network. This enhancement aligns the functionality with implementation guidelines outlined in the SIPconnect 1.1 Technical Recommendation.

· Early Media and Support for PRACK

SIP trunks can be configured to support early media by adding "100rel to Supported" header in the

INVITE: This allows support for in-band announcements such as: branding from discount or bulk long distance providers; caller comfort in terms of progress indications when extraordinary wait times are inherent in the call scenario, for instance when trying to locate a cell phone; as well as, the ability to provide country specific ring back and other progress tones. This feature also facilitates the ability to set up conferences in the IP domain before call answer, for instance, in certain call recording scenarios or for automatic dialers’ conferencing in agents.

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SIP Trunks

SIP Trunk enhancements have been included that allow service providers to continue to broaden their SIP trunk offerings.

SME customers are increasingly evaluating and adopting SIP trunks. IP Office responds to these market trends, with several new SIP trunk capabilities, designed to simplify and address a wider set of implementation scenarios.

The IP Office architecture brings the benefits of Session Initiated Protocol (SIP) to the enterprise customer through SIP Service Providers. To help enable the SIP Service Provider, Avaya offers a SIP Compliance

Testing Program (GSSCP) that will validate the operation of the IP Office solution with the Service Provider’s

SIP trunk offering.

For more information about Avaya’s SIP Compliance Testing Program (GSSCP), please refer to the following link: https://enterpriseportal.avaya.com/ptlWeb/bp/so/CS201172610238111040

SIP Trunk Capabilities

SIP From in Clear when privacy is requested

Many SIP Service Providers require that the From field contain account information for outbound calls. SIP

Lines will now support the Send From in Clear field. IP Office will send the caller ID in the clear in the SIP

From header for outbound anonymous calls. The default of Send From in Clear is disabled. This new capability is configurable on a per line basis.

SIP User-Agent header included to identify IPO in SIP trunk calls

IP Office identifier will now be included in the SIP header. This addition will enhance troubleshooting capabilities when additional identification is required. The identifier will be configurable as the system administrator is permitted to enter identifying (User-Agent) information into the SIP messages. User-Agent headers are generated for IP Office initiative messages such as INVITE, OPTIONS, CANCEL, etc. This feature will also support enhanced SIP Server header information in all 18x response messages such as Trying,

Ringing, etc.

IP Office can send UPDATE without SDP for session refresh over SIP trunks

The addition of this new audit mechanism will validate that the connection is in place. If not, the IP Office will free the trunk, thereby improving network resource utilization.

IP Office can now negotiate RFC 4028 for session refresh over SIP Trunks. It can either be the refresher or will expect UPDATE messages from remote party, depending on negotiation. The session will be terminated if IP

Office doesn’t receive an UPDATE/200 OK message for session refresh. In earlier IP Office releases, RE-INVITE messages were used as the refresher. IP Office is now RFC 4028-compliant. By default, UPDATE Supported is not enabled and IP Office will send RE-INVITE messages for session refresh. If UPDATE Supported is Allow or

Auto and the other party supports UPDATE (specified in the "Allow" header), IP Office will send UPDATE messages for session refresh.

From header shall be used by IPO as calling identity whatever presence of P-Asserted-

Identity (PAI)

Although service providers typically use PAI for Caller ID delivery, additional flexibility is sometimes required.

IP Office answers this requirement by providing an option to use the From header. The administrator will have the option of selecting which Caller ID information to use for displaying on the user equipment. By default, IP

Office will display PAI caller ID present in the SIP/Tel URI consistent with the behaviour of previous releases. If the checkbox is set, the caller ID present in the From SIP URI will be used instead.

Turn off OPTIONS if registration can do keep-alive

Some service provider implementations utilize registration to maintain the status of lines instead of OPTIONS.

OPTIONS requests are sent to regularly check if the SIP Line is in service. It is possible to send frequent

REGISTER requests to keep the line in service rather than using OPTIONS.

Fax G711 pass through mode including SG3

Super G3 fax machines utilize higher speeds than normal G3 or earlier fax machines. In order to broaden IP

Office fax support, IP Office now supports Super G3 fax transmission.

This includes Fax Transport support for sending and receiving of faxes over G.711. IP Office will detect fax tones in G.711 calls between Super G3 fax machines.

IP Office should support nonce-caching for proactive inserting SIP credentials in outgoing SIP requests

Traditionally, IP Office sends the Authorization/Proxy-Authorization only if the network specifically requests this for one IP Office request. Now, IP Office will cache the nonce received with the challenge or with the

Authentication-Info header. IP Office will use the information for proactively inserting the SIP credentials in all the outgoing SIP messages.

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The support of nonce-caching will speed authentication, reduce network signaling traffic, and improve the efficiency of message handling by IP Office.

All outgoing SIP requests may be authenticated by SIP network with the exception of

Cancel method

IP Office will now support additional levels of authentication by broadening the scope of message types to which the IP Office will respond.

All methods shall support presence of (proxy) authorization headers except Cancel (RFC compliance). At present, Invite, Re-Invite, Update, and BYE shall support the presence of such headers. Before the introduction of these changes, IP Office did not support challenges for Update message.

Template support for new features included in SIP Trunk

Each of the new Manager configuration settings have been added to SIP Line templates.

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SIP Endpoint Support

Session Initiation Protocol (SIP) is an open signaling protocol for establishing any kind of real-time communication session. The communication session can involve voice, video, or instant messaging, and can take place on one of many devices that people use for communicating: laptop computer, PDA, cell phone, IM client, IP phone, and so on. SIP has been developed in the Internet Engineering Task Force (IETF) by common participation from various vendors, including Avaya.

Avaya IP Office supports SIP for telephony functions to enable the usage of standard based SIP endpoints for

Voice and Fax communication. In pure SIP systems, IP Office expands the feature set beyond the SIP standard, offering a wealth of IP Office features also on SIP endpoints delivering a feature rich system that a pure-SIP server based on the SIP standard only can’t deliver. With that, IP Office delivers the best of both worlds, supporting standard based IP telephones while delivering a wealth of features consistently between

SIP, digital and Avaya IP endpoints.

IP Office SIP endpoint implementation is built on two major SIP components: SIP User agents, and SIP Server components.

SIP Components

SIP Endpoints (User Agents)

User agents (UAs) are applications in SIP endpoints (such as a SIP phone, cell phone, PDA, or workstation) that interface between the user and the SIP network.

SIP Servers

IP Office has implemented the required functionality of the SIP servers mentioned below not only to provide

SIP endpoint support but also to allow full interoperability between SIP endpoints, other IP telephones based on H.323, Digital and Analog telephones as well as IP Office trunks (Analog, digital or SIP based)

SIP servers provide centralized information and enablement services in a SIP ecosystem. The core SIP servers and their functions are summarized here. IP Office provides the required the features of the following two servers for Voice and FAX communication.

· Registrar Server

When SIP IP telephones come online, they need to make sure that others are aware that they’re available to take and make calls. The Registrar authenticates and registers the IP phone (often directly related to a specific user) when it comes online, and then stores information on the telephones logical identities.

· Proxy Server

A proxy server takes SIP requests, processes them, and passes them downstream while sending responses upstream to other SIP servers or devices. A proxy server may act as both a server and a client, and can modify a SIP request before passing it along. A proxy is involved only in the setup and teardown of a communication session. After user agents establish a session, communications occur directly between the parties.

Functionality of the following two SIP servers are generally available by IP Office using existing IP Office functionality. Therefore, while functionality is provided, e.g. allowing hot desking (also for users using a SIPendpoint) in a small community network, a consistent methodology between SIP and non SIP endpoints is used to deliver those features

· Location Service

As users roam, the network needs to be continually aware of their locations. The location service is a database that keeps track of users and their locations. The location service gets its input from the registrar server and provides key information to the proxy and redirect servers. IP Office provides hot desking support, delivering a similar functionality but working consistently between SIP and non SIP endpoints.

· Redirect Server

If users are not in their home domains, sessions bound for them needs to be redirected to them.

The redirect server maps a SIP request destined for a user to the device "closest" to the user. In IP

Office, call forwarding and Follow me functionality is used to provide again consistent functionality between all type of endpoints.

Supported functionality for SIP endpoints in IP Office

SIP endpoints are supported on IP Office for Voice (Audio) and Fax (T.38) communication.

This allows the usage of standard compliant IP telephones using the open SIP standard, giving customers a choice of endpoints of different manufacturers including special purpose devices like conference telephones, hotel telephones or terminal adapters.

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In order to use a non-Avaya SIP endpoint with IP Office, a "Third party IP endpoint license" is needed. This license will continue to support endpoints based on the H.323 standard but will also be required for generic SIP endpoints on IP Office. Avaya IP Office SIP telephones use the IP End Point license.

SIP Endpoint support is fully integrated into IP Office core. No other components are needed. SIP endpoints will need VCM module capacity in IP Office like any other IP phone.

Next to SIP telephones, SIP terminal adapters are supported to connect analog telephones and fax machines.

This offers a flexibility to support Fax machines and Audio/T.38

SIP extensions function like any other IP Office extension: This means they

· Can make and receive calls to any other extension, independent of type of extension

· Delivers end to end Media just like any other IP telephone on IP Office. For calls between two SIP extensions of a SIP extension and an Avaya IP telephone, the audio is transmitted end to end for basic telephone calls. (Conferences etc. However require a VCM resource). See chapter "VCM modules" for details

· Can use short codes and authorization codes like any other telephones

· Transmit In band call progress tones are delivered from IP Office

· A SIP telephones needs to register with IP Office like any other IP telephone, Authentication with

Username and password is possible

· SIP extensions support "auto create" in IP Office to make installation fast and efficient. Successful registration of an endpoint will consume one third party license

· On one IP address, several extensions can register with IP Office, each consuming a license. This enables the connection of SIP terminal adapters with more than one analogue port, giving a different extension number to each of the ports.

Advanced features:

SIP endpoints support a number of extended features according to the "SIP service samples-draft", also referred to as "Sipping-19". This includes:

· Calling line identification

· Hold/Consultation Hold

· Attended/Unattended Transfer

· Message Waiting

· Do not disturb

· Conference Add

Some telephones support several call appearances making it easy to switch between calls. Please not that this does not include "bridged appearances" or "(outside)-line appearances."

A large number of additional features are supported on IP Office using Feature activation keys. These features include but not limited to:

· Call forward: Unconditional/Busy/no Answer

· Follow me

· Park/Unpark

· Music on Hold

· Meet me conferencing

· Conference join

· Ring back when free

SIP endpoints also support Computer Telephony Integration “CTI” and therefore applications like Avaya one-

X™ Portal for IP Office: the following features are supported with Avaya one-X™ Portal for IP Office and via the TAPI interface:

· Outgoing call (without remote activation of speakerphone/headset)

· Hang up

· Hold

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· Attended/Unattended transfer

· Conference (IP Office based)

· Voicemail collect

· Set forwarding/DND (IP Office based)

· Park/Ride (IP Office based)

The features work similar like "CTI" features in combination with an analog telephone, e.g. an outgoing call will first ring the SIP phone and after connect the outgoing call will be placed. SoftConsole is currently not supported in combination with SIP-endpoints.

Session Initiation Protocol

· Rec. E.164 [2] - ITU-T Recommendation E.164: The international public telecommunication numbering plan

· RFC 2833 [7] - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

· RFC 3261 [8] - SIP: Session Initiation Protocol

· RFC 3263 [10] - Session Initiation Protocol (SIP): Locating SIP Servers

· RFC 3264 [11] - An Offer/Answer Model with Session Description Protocol (SDP)

· RFC 3323 [14] - A Privacy Mechanism for the Session Initiation Protocol (SIP)

· RFC 3489 [18] - STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address

Translators (NATs)

· RFC 3824 [24] - Using E.164 numbers with the Session Initiation Protocol (SIP)

· RFC 1889 – RTP

· RFC 1890 - RTP Audio

· RFC 4566 – SDP

· RFC 3265 - Event Notification

· RFC 3515 - SIP Refer

· RFC 3842 - Message Waiting

· RFC 3310 – Authentification

· RFC 2976 – INFO

· RFC 3323 - Privacy for SIP (PAI) and draft-ietf-sip-privacy-04 (RPID)

For codecs supported, refer to chapter to VoIP Standards Supported .

While great care has been taken to be compliant with SIP standards, no guarantee can be given that all devices claiming support of SIP will work flawlessly. The SIP standard is constantly evolving with new features and methods introduced. Also, while being standard compliant, not all devices implement all options of the standard, making it hard to almost impossible to predict if a device will work- Avaya will only confirm functionality of devices that we have tested and will publish a list of devices that have been tested including – if required – implementation details and software version used on that device.

At the time of writing, the following devices have been tested successfully for Audio and/or Fax transmission.

SIP Telephones: · Polycom Soundpoint IP 320, IP 601

· Avaya 1100 Series/1200 Series

· Avaya B179 conference phone

SIP clients on mobile cell telephones:

· Nokia S60 v3 SIP client (e.g. Nokia E61i mobile cell phone)

MAC and Windows versions

SIP Analog Terminal Adapters · Quick Edition A10 ATA

SIP PC-based softphones:

· Patton single line M-ATA

· Innovaphone IP22, IP24, IP28

· CounterPath eyebeam/xlite

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Softphone

Video enabled devices

· IP Office Video Softphone (Using advanced SIP features): Does not require a “SIP-station” license but works as part of the user package.

· Avaya 1010 Video

· Avaya 1040 Video with 4-party MCU

· Polycom VVX 1500

· Grandstream GXV3140

A list will be provided in the IP Office knowledge base and on http://support.avaya.com

Supported functionality for SIP communication in IP Office

IP Office supports both Video and Audio communication for SIP devices.

It is therefore possible to connect a Video-endpoint to IP Office for rich multimedia communication:

Video is supported

· In an IP Office system locally

· In a IP Office small Community network

· To a Video-ready SIP trunk, e.g. to Avaya Aura

A few typical Call scenarios:

· End to end Video call in an IP Office system: E.g. a call from a Video Softphone to an Avaya 1010 system in a senior manager’s office.

· Video in a Small Community Network: Calls e.g. between two Avaya 1010 telephones in the headquarter and a remote Office on the SCN

· Video Conference leveraging the Avaya 1040 Video Conferencing system. One user can use the 1040 while three other users are e.g. in remote locations on a SCN, home office etc.

· Multi-party video call using the softclient-based video-conferencing capability. In this case, all video conference participants use the IP Office Video Softclient.

Video details:

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· All video communication is end to end, meaning that IP Office does not natively manage or perform video conferencing.

· The softphone handles all video conferencing initiated by users logged into the softphone.

· Video endpoints can make calls to all telephones and trunk lines on IP Office (analog, digital IP) but will work as audio devices only when calling these telephones. They can also receive audio calls from these telephones/lines.

· Video needs a high bandwidth in the network, depending on codec/Video quality it can be up to 1

Mbit/sec. Through planning of the capacity it has to be assured that there is sufficient bandwidth available for both audio and video. A network assessment is mandatory and has to take the bandwidth requirements into account. Please refer to the details of the product(s) to be used for exact

Video requirements.

· All Video connections are “end to end”. IP Office does not negotiate e.g. Video Codes; this will be done by the Video endpoints (via IP Office) only. The Video-capability is therefore depending on the used devices. It was confirmed that IP Office works with both H.263 as well as H.264 Codecs

· Video calls have limited functionality. While with most devices hold and transfer works, this is of course depended on the tested device. IP Office feature access codes are available for Video telephones and e.g. call forwarding will function with video devices. Forward to voice mail will work but will only record audio streams, no video calls.

· IP Office supports a single video call per session only. No dual Video channels for e.g. Video and sharing of applications are possible.

· A single endpoint can accept several video calls in parallel. That makes it possible to leverage

“MCU” (Multi conference Unit) functionality in an endpoint like inside the Avaya 1040 system. It makes Video conferences between a device with MCU and several other endpoints possible.

· Typical bandwidth requirements for HD video:

Video bandwidth needed per call (transmit and receive, Audio bandwidth not included) for e.g. the

Avaya 1010/1040 Video Conferencing systems.

o 1010: 1 Mbps for 720p/30fps o 1040:

· 768 Kbps for 720p 30fps

· 1.1 Mbps for 720p 60fps

· 1.7 Mbps for 1080p 30fps

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IP Office Interoperability

A number of interoperability scenarios with other Avaya products have been tested and are supported. The scenarios include:

· IP Office 9.0 and Avaya Aura Communication Manager 6.3

· IP Office 9.0 and Avaya Aura Session Manager 6.3.4

· IP Office 9.0 and BCM 6.0

· IP Office 9.0, BCM 6.0 and CS1000 7.6 with Session Manager

IP Office and Communication Manager Interoperability

The H323 protocol is used for trunks between the IP Office nodes and the CM. The IP Offices are a network of

SCNs and can use advanced SCN feature set between each other.

· Protocol Support: o IP Office to CM: H.323

o IP Office to IP Office: H323 - SCN Networking

· Platform Requirements: o IP Office:

· Release: IP Office 9.0

· Edition: Essential, Preferred, Advanced, and Server Edition o Communication Manager:

· Release: CM 6.3

· Centralized Voicemail: Centralized Voicemail for all IP Office systems in the SCN network

· Phones: All phones supported on the individual call servers and releases including Avaya IP Phone(H323),

Avaya Digital phones, BCM 7000 Series Digital phones, BCM1100/1200 Series IP Phones (SIP), DECT R4 and Analog phones.

The advanced feature set of SCN Networking is available between the IP Office nodes.

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IP Office and Session Manager Interoperability

The SIP protocol is used for trunks between the IP Office nodes and the SM. The IP Offices are a network of

SCNs and can use advanced SCN feature set between each other.

· Protocol Support: o IP Office to SM: SIP o IP Office to IP Office: H323 - SCN Networking

· Platform Requirements: o IP Office:

· Release: IP Office 9.0

· Edition: Essential, Preferred, Advanced, and Server Edition o Session Manager:

· Release: SM 6.3.4

· Centralized Voicemail: Centralized Voicemail for all IP Office systems in the SCN network

· Phones: All phones supported on the individual call servers and releases including Avaya IP Phone(H323),

Avaya Digital phones, BCM 7000 Series Digital phones, BCM1100/1200 Series IP Phones (SIP), DECT R4 and Analog phones.

The advanced feature set of SCN Networking is available between the IP Office nodes.

IP Office and BCM Interoperability

The SIP interoperability support between IP Office and BCM50/450 allows customers to migrate a network of

BCM, step by step, to an IP Office SCN or Server Edition network.

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· Protocol Support: o IP Office to BCM: SIP o BCM to BCM: SIP or SIP/MCDN o IP Office to IP Office : H323 – SCN or SIP o It is not possible to network IP Office and BCM via the MCDN or SCN protocol directly

· Platform Requirements: o IP Office:

· Release: IP Office 9.0

· Edition: Essential, Preferred, Advanced, and Server Edition o BCM:

· Release: BCM50 and BCM450 6.0

· Centralized Voicemail: o The IP Office system may use a centralized Voicemail PRO system in the SCN network o The BCM systems may use a centralized voicemail within the BCM MCDN network. o Centralized Voicemail across IP Office and BCM systems is not supported.

· Phones: All phones supported on the individual call servers and releases including Avaya IP Phone(H323),

Avaya Digital phones, BCM 7000 Series Digital phones, BCM1100/1200 Series IP Phones (SIP), DECT R4 and Analog phones.

The advanced feature set of IP Office SCN Networking is available between the IP Office and BCM. MCDN

Networking features are also available between the IP Office and BCM. The standard SIP features described below are supported.

IP Office, BCM, and CS1000 Interoperability

This configuration allows migrating a network of BCM and CS1000 to the IP Office step by step by adding IP

Office systems to it.

· Protocol Support: o IP Office to BCM: SIP o IP Office to CS1000: SIP o IP Office to IP Office: SIP or H323 - SCN Networking o BCM to BCM: SIP or SIP/MCDN

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o BCM to CS1000: SIP or SIP/MCDN o Only UDP will be supported and not TCP

Note:IP Office to CS1000 using H.323 or PRI trunks is not supported, and IP Office to CS1000 using

NRS is no longer supported.

· Platform Requirements: o IP Office:

· Release: IP Office 9.0

· Edition: Essential, Preferred, Advanced and Server Edition o BCM:

· Release: BCM50 and BCM450 6.0

o CS1000:

· Release: CS1000 7.6 with Session Manager

· Centralized Voicemail: Centralized Voicemail for all IP Office systems in the SCN network. BCM systems may use centralized voicemail of the CS1000. IP Office systems cannot use centralized voicemail of

CS1000.

· Phones: All phones supported on the individual call servers and releases including Avaya IP Phone

(H.323), Avaya Digital phones, BCM 7000 Series digital phones, BCM1100/1200 Series IP phones (SIP),

DECT R4 phones, and Analog phones.

SIP Standard Call Feature Support

The following call features are supported for the standard SIP Networking connections described above:

· Standard Call Features: o Basic Call Completion o Handling of busy called party o DTMF and ring-back tone o Hold and Retrieve o Call Waiting presentation o Called Number display o Calling number and name display o Abandoned call

· Call Redirection: o Call Forward (Forward ALL, Forward Busy, Forward No Answer) o Call Transfer (Attended and Unattended) o Call Redirection (To PSTN, Abandoned, Busy)

· Conferencing: o AdHoc and MeetMe conferencing

· PSTN Toll Bypass: o Toll Bypass allows each system to leverage the trunk connections of the other system in the network to avoid international and long distance charges.

Video Collaboration Solution

The Radvision Scopia solution based on the XT5000/4200 platform and the Elite 6000 platform using Radvision

Scopia 7.7 firmware is currently supported in IP Office release 8.1.

For Scopia 8.0, the video solution has been broken into small and medium deployment scenarios. The small solution using Scopia 8.0 firmware will be ready for 8.1 FP1. The medium solution has been tested and works with release 8.1 FP1 with some exceptions for Flare to support Pin codes when connected to IPO. IPO 9.0 will provide the needed support for this to remove this exception.

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The new XT5000 Video Server for IP Office is now available and works with IP Office release 8.12 FP1 – SP5 and 9.0. This variant of the XT5000 platform is integrated to IP Office and must register to the IP Office as an

Avaya SIP endpoint for the Video Server component to function (MCU). Release 9.0 includes the fixes from 8.1

SP5 and some additional features to support flare functionality when used with the Radvision Elite MCUs.

IP Office supports:

· Video Collaboration Solution for IP Office

· XT5000 Video Server for IP Office with 3.2 Firmware (must register with IPO for MCU to function)

· XT5000 Endpoint w/embedded video server for IP Office with 3.2 Firmware (must register with IPO for

MCU to function)

· Scopia XT5000 Endpoint with embedded MCU with 3.2 firmware

· Scopia XT4200 Endpoint with 3.2 firmware

· Scopia XT240 Endpoint with 3.2 firmware (requires a 3rd party SIP license)

· Scopia Elite XT6000 with 7.7 firmware

Remote Phone Support

Remote Working - NAT Traversal

The new Remote Worker feature allows the connection of remote 9600 Series IP Telephones with the H.323

FW which resides behind a NAT Router to IP Office. The configuration does not require any VPN concentrator equipment with the IP Office system.

With the Remote Worker feature enabled, remote 9600 H.323 IP Phones can connect to the IP Office system even if the IP Office is located behind a NAT router. The sets are authenticated on the IP Office in the same way as sets in the private network. The IP Office will determine that a set is located outside the private network based and relay the VOIP RTP traffic to ensure it transverses the NAT router.

It is important to note that the H.323 signaling and the media traffic is not encrypted, but the proprietary binary format adds a basic level of encryption.

To reach the IP Office from the remote private network remote H.323 IP Phone needs to be configured to the public IP address of the NAT router hosting the IP Office. Configurable ports need to be forwarded to the IP

Office. The IP Office requires a valid public IP address be configured for the feature to function. The public IP address can be statically configured or dynamically discovered via a STUN server.

The Remote Working feature needs to be enabled in the IP Office Manager. The Remote Worker feature requires the Essential Edition license which provides four (4) remote worker seats with the IP Office system.

Additional remote worker capability is available with the Teleworker User package and a Preferred Edition license.

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Product Description

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VPN Phone Functionality

VPN Phone is a full-featured IP Telephony solution that provides secure communication over public ISP networks to an IP Office system at the company headquarters.

It is a software-only product that runs on the standard 5610/5620/5621 or 4610/21 IP telephones. In combination with one of these telephones and the most popular VPN gateway products, the software extends enterprise telephony to remote locations.

VPN functionality is supported on the new 9600 IP telephones, not requiring a separate software load.

VPN telephones offer the full IP Office telephony features that are available on IP Office IP telephones at the user's desktop in a remote location like a home-office:

There is no restriction on VPN phone usage.

VPN Phone is ideal for IP Office customers supporting "work-at-home" users:

· Virtual Office workers

· Remote workers

· Remote call center

· Business continuity support

· Very small locations that require a single phone only

· Temporary installations such as conferences, off-site meetings, and trade shows

VPN Phone has been tested with a number of VPN-gateways from major vendors like Cisco or Juniper as well as with smaller VPN-access devices from companies like Adtran, Kentrox, Netgear, and SonicWall. Refer to the support pages (support.avaya.com) for a list of available application notes on VPN-gateways tested with each line of telephones.

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Avaya Session Border Controller for Enterprise

The Avaya Session Border Controller for Enterprise is a flexible, fit-for-purpose solution that includes a set of standard security features. This Session Border Control (SBC) provides all of the functionality required for

Small-Medium enterprises to terminate SIP trunks without the complexity and higher price associated with typical SBCs. The Avaya SBCE scales up to 150 simultaneous sessions per Portwell server (2,000 with other server options).

Enterprise-Class functionality at SME Pricing

· Blocks toll fraud

· Blocks DoS, DDos and Call Walking attacks

· GUI driven

· Scalable to largest IP Office environments

· A fraction of the overall solution cost

Easily upgradable to the Avaya Session Border Controller for Advanced (for future releases)

· VPN-less SIP remote worker protection

· Enhanced Toll Fraud and DoS protection

The Avaya Session Border Controller for Enterprise will be included in the GSSCP reference architecture for IP

Office, as a recommended option to complement the base reference architecture, where appropriate to the specific customer requirements.

For more information about Avaya’s Session Border Controller for Enterprise, please navigate to the following

URLs:

· https://enterpriseportal.avaya.com/ptlWeb/gs/products/P0996 (Avaya SBCAE)

· https://enterpriseportal.avaya.com/ptlWeb/gs/products/P0997 (Avaya SBCE)

VoIP FAQ

Network Requirements

Quality of Service (QoS) is a measure of the performance of a network that reflects the availability of network service and the quality of network transmissions. The term itself refers to a number of networking technologies and techniques and does not necessarily restrict itself to any single protocol or standard.

There are a number of measures that can be taken on the LAN and WAN to make them 'good enough' to carry voice traffic. Some of these are the implementation of standards based QoS protocols while are simply a matter of network architecture and good network management practices.

The term 'good enough' is intentional. Every customer will have different expectations and different budgets to work to. Some will be willing to upgrade their networks to use the best possible equipment and practices. To others the additional expense may be viewed as unnecessary.

Examples of standards based Quality of Service protocols include DiffServ and 802.1p/q.

What are Voice Compression Modules (VCM's) for?

VCM's are required to support the following scenarios:

· Internal phone calls between an IP device and a non-IP device.

·

·

Analog/digital telephones to IP trunks (SIP/H.323) including managed Frame Relay and managed IP

VPN (provides echo cancellation).

IP telephones to ISDN or PSTN trunks (convert IP to TDM and vice-versa).

IP device to IP device

Call progress tones (for example dial tone, secondary dial tone, etc) do not require voice compression channels with the following exceptions:

· Short code confirmation, ARS camp on and account code entry tones require a voice compression channel.

· Devices using G.711, G.722, or G.723 require a voice compression channel for all tones except call waiting.

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When a call is connected:

· If the IP devices use the same audio codec no voice compression channel is used.

· If the devices use differing audio codecs, a voice compression channel is required for each.

"Direct Media" is a VoIP concept that circumvents resources (TDM bus, Gateway) within the system and improves the voice quality. If two IP devices are connected on the same system, a direct LAN connection between them will be established once the call has been set up (as long as they use the same Codecs). Direct media is supported between Avaya IP telephones and SIP telephones.

It is possible for an IP device to be physically located at one site while being registered at a different site. In this case, even for VoIP across the WAN the VCM would not be used, as long as the two IP devices involved in a phone call are registered on the same system.

Bandwidth Required For Each Voice Call?

The bandwidth used varies depending on the compression method chosen. IP Office supports standards listed below. These will occupy approximately 10K and 13K of bandwidth respectively.

Use the following chart to choose the most appropriate compression algorithm for your available bandwidth.

Audio Codec

G.723.1 (6.3K)

G.729a

G.711 (64K)

G.722 (64K)

RTP Voice

Data Payload

24 Bytes

20 Bytes

160 Bytes

160 Bytes

Packets per

Second

33.33

50

50

50

LAN

(bps)

20,800

29,600

85,600

85,600

% Overhead

LAN

225%

270%

34%

34%

WAN

(bps)

9,867

13,200

69,200

69,200

% Overhead

WAN

Algorithmi c Delay

(milliseconds)

54%

65%

8%

8%

80

40

20

20

Typical Bandwidth Required For Video Calls

This is for Video transmit only. To the Video part, the audio channel needs to be added.

· Video bandwidth needed per call (transmit and receive, Audio bandwidth not included) as sample for the

Avaya 1010-1040 for HD-Video:

· 1010/1020: 1 Mbps for 720p/30fps

Typical Bandwidth for standard Video quality, for example with IP Office Video Softphone.

Resolution Pixels Typical Net Video

Bandwidth Requirement

H.263

64 kbps

H.264

45 kbps Low

Standard

176 x 144 (QCIF)

320 x 200 (QVGA)

High 640 x 480 (VGA)

162 kbps

776 kbps

114 kbps

545 kps

Acceptable Delay?

End-to-end delay should be 150 milliseconds or below.

How Many Simultaneous Calls Can I Get Down My Link?

The following chart illustrates the theoretical maximum number of simultaneous voice calls that can be delivered over a WAN for a given link speed. This does not take into account any bandwidth that may be required for data traffic between sites or the physical limit of VoIP calls for the specific version of IP Office in use.

The number of simultaneous voice calls can be in excess of the capabilities of the individual platform, where the calls transit the switch as data traffic. In this situation compression resources are not used but obviously must be catered for in the overall bandwidth provision.

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Compression

Algorithmic Delay

(seconds)

Number of Calls

- 64Kbps Link

- 128Kbps Link

- 256Kbps Link

- 512Kbps Link

- 1Mbps Link

- 2Mbps Link

G.723.1 (6.3K)

0.08

6

12

25

51

103

207

G.729a (8K)

0.04

4

9

19

38

77

155

G.711 (64K)

0.02

0

1

3

7

14

29

What is the Maximum Number of Simultaneous VoIP Calls

Each IP Office can be fitted with optional Voice Compression Modules (VCM) to support VoIP connections.

The IP500 V2 is capable of supporting two VCM 32/64 modules allowing up to 128 simultaneous calls. With combination cards, another 10 VCMs are provided per card. Up to two combination cards are supported with IP

Office.

Network Assessment

With IP Office, optimum network configurations can support VoIP with a perceived voice quality equivalent to that of the Public Switched Telephone Network (PSTN). However, not every network is able to take advantage of VoIP transmissions. It is important to distinguish between basic compliance with the minimal VoIP standards and validated support for QoS which is needed to run VoIP applications over a data network.

With the exception of standalone configurations where IP telephones connect directly connected to the ports on IP Office, Avaya now requires that all customers formally audit their networks for IP telephony readiness before attempting to install any VoIP application.

A network assessment should normally include:

· Physical inventory of all equipment inclusive of the current version of code, and configurations as needed.

· An accurate and complete network topology for all networked sites, inclusive of IP addressing and physical/logical connections.

· An evaluation of the network's topology to check that the design is both sound and reasonable.

· Measurement of packet loss, jitter and delay over the course of multiple days and measured on a per minute basis. A graphical representation of the data is the preferred output method.

· Examination of QoS/Class of Service (CoS) parameters in place in the network.

· Summary of findings and possible actions to correct problems.

The assessment should leave you confident that the implemented network will have the capacity for the foreseen data and voice traffic, and can support H.323, DHCP, TFTP, and jitter buffers in H.323 applications.

With this in mind, if you require support during or after an IP Office VoIP installation, a copy of your network assessment documentation will be requested by your support channel.

For more details about available tools, resources and services to enable you to audit your network for VoIP readiness, please contact your local Avaya representative.

IP Packet Flow Control

While a high-performance switch forwards data packets at full wire speed to and from its ports simultaneously, there may be times when a switch port may not be able to accept packets at the rate it is receiving them.

For example, the switch port may be receiving packets from multiple ports at the same time, or the switch port may be receiving packets from a port operating at a faster speed. For instance, the sending port might be operating at 100 Mbps, while the receiving port operates at 10 Mbps; or the sending port might operate at

1000 Mbps, while the receiving port operates at 100 or 10 Mbps. If data packets arrive for a port that is saturated with other packets, the packets may overflow the port’s buffer, resulting in dropped packets and lost data.

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Flow control is a congestion-control mechanism that prevents data loss at congested ports. Flow control prevents packet loss by controlling the flow of data from the transmitting device to ensure that the receiving device can handle all of the incoming data.

IEEE 802.3 flow control is used on Avaya IP telephones operating in full-duplex mode. If the receiving device becomes congested, it sends a pause frame to the transmitting device. The pause frame instructs the transmitting device to stop sending packets for a specific period of time. The transmitting device waits the requested time before sending more data.

VoIP Standards Supported

IP Office supports the following protocols and standards:

· H.323 V2 (1998), Packet-based multimedia communications systems.

· Q.931, ISDN user-network interface layer 3 specification for basic call control.

· H.225.0 (1998), Call signaling protocols and media stream packetization for packet-based multimedia communication systems.

· H.245 (1998), Control protocol for multimedia communication.

· Session Initiation Protocol.

· Audio CODECs: o

G.711 A-Law/Mu-Law (64K).

o G.723.1 MP-MLQ (6.3K).

o

G.722.

· Silence Suppression.

· Fax Relay (IP Office to IP Office Fax Transport over IP).

· T.38 Fax support (SIP trunks and SIP endpoints).

· Local End Echo Cancellation 25ms.

· Out of band DTMF.

· Jitter buffer, 5 frames of jitter buffer.

· Internet Standards/Specification (in addition to TCP/UDP/IP).

o

RFC 1889 – RTP/RTCP, Real Time and Real Time Control Protocol.

o RFC 2507, 2508, 2509 – Header Compression.

o

RFC 2474 – DiffServ, Type of Service field configurable.

o RFC 1990 - PPP Fragmentation.

o RFC 1490 - Encapsulation for Frame Relay.

o

RFC 2686 - Multiclass Extensions to Multilink PPP.

o RFC 3261 - Session Initiation Protocol (SIP).

o

RFC 3489 - STUN.

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Chapter 11.

Public and Private Voice

Networks

Product Description

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