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Table of Contents

 

Document Overview

This document is intended for Ribbon's C20, AS, and SBC technical staff and any individual tasked with the integration of Ribbon's C20-AS-SBC with the SIP Trunking solution with the Avaya IP Office on the enterprise side. This document includes the configuration guide for the Ribbon series SBC Core, Enterprise SBC, C20, and AS to provide the SIP Trunk solution with Avaya IP Office PBX.

 

Scope

This is a general reference document that requires user input during the configuration of the Ribbon SBC - SWe Core, Enterprise SBC Edge 1000, C20, AS, and Avaya IP Office. 

Non-Goals

All test results and observations for Ribbon's C20-AS-SBCs and Avaya IPO interoperability have been mentioned in a separate Test Report document and are not part of this Configuration Guide. 

Introduction

Configuration Guides provide the system integrator with the necessary information to construct an integration similar to what has been built for certification or Interoperability Test (IOT) purposes. The IOT reference architecture is shown below in the Network Diagram section.

This configuration guide is divided into the following main sections:

  • Avaya IP Office

  • Ribbon SBC - SWe Core

  • Ribbon Enterprise 1K - Edge SBC

  • Ribbon AS 

  • Ribbon C20

Solution Release 

Solution Release

DateSolution ReleaseComment
16/Jan/20191.3NA

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon and third-party product. Navigating third-party product as well as the Ribbon SBC command line interface (CLI) is required. Understanding about the Ribbon SBC Edge 1000, SBC SWe Core, SESM AS, C20, and Avaya IPO is also necessary to complete the configurations and for the basic troubleshooting.

Requirements

The sample configuration uses the following equipment and software:

Requirements

Product

Equipment

Software Version

Ribbon Products

 

CSBC (SBC SWe Core):

OS:

SonusDB:

EMA:

 

E-SBC (1K SBC)

AS

C20

CIM

MAS

 

V07.00.00-R000

V06.00.00-R000

V07.00.00-R000

V07.00.00-R000

 

8.0

AS12.1 (MCP_19.0.20.2_2017-11-28-1107)

R19

V 10.0.0, REV 4532

V 16.0.0.865

Third-party Equipment

Avaya IPO 500 V2 - PBX

10.1.0.2 Build-2

Avaya 9611G SIP - IPO Phone 17.1.1
Avaya 9611G H323 - IPO Phone 26.6
Polycom Sound point IP450 - PSTN 14.1.1
Polycom Sound point IP450 - PSTN 24.1.1
Polycom Sound point IP450 - PSTN 34.1.1
Panasonic Fax KX - FP701CX Analog MachineNA
IADNA

Administration and Debugging Tools


Wireshark

Lx Tool Log Analyzer and Syslog server

2.6.4

2.1.0.4

System Overview

The main IOT Report for this activity captures the results where the registration model is a static PBX setup from the Avaya IP Office perspective to the AS.

Network Diagram

Avaya IP Office PBX Configuration

This section provides the procedure for configuring the Avaya IP Office to support connectivity to the Ribbon C20-AS SIP Trunking solution through the SBC. This section requires you to have proper knowledge of the Avaya IP Office usage, configuration, and support in general, and experience with the product platform. Assuming that the basic configuration was already setup, the following screen captures show the SIP trunk configuration on the Avaya IP Office during the test execution. Avaya IP Office is configured using the Avaya IP Office Manager PC application with administrator login credentials.

Avaya IP Office Manager

The Avaya IP Office Manager was loaded onto the tester’s PC and allowed user login and access to the Avaya IP Office PBX. With Avaya IP Office Manager loaded on your local PC, select Program Files (x86) > Avaya > IP Office > Manager. Select the “Manager” application.

 

 




Enter the Service User Password for the Administrator user.

 

 

System Settings

To access the System settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the System tab.

 

 

LAN1

Access to the IP Office was gained through the LAN side of the PBX (LAN1). The SIP PBX phones also registered through the LAN side of the PBX.
To access the LAN1 settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the LAN1 tab.

LAN Settings

Enter the IP Address and IP Mask for the LAN side of the IP Office PBX.

VOIP

Check the SIP Registrar Enable box to allow the SIP phones to register to the IP Office PBX.

The UDP and TCP Protocols were set to 5060.

Network Topology

The UDP and TCP Public Ports were set to 5060.

LAN2

The SIP Trunk to the E-SBC and C20 SIP Trunking Solution uses the WAN connection of the IP Office. To access the LAN2 settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the LAN2 tab.

Note

During Interoperability testing, LAN2 was not used because LAN1 was connected to the Ribbon Enterprise SBC Edge 1000 and the Enterprise SBC had public internet connectivity over WAN.

 

LAN Settings

 

 

 

Voicemail

To access the Voicemail settings, click the name of the IP Office system. Select Sonus IP Office > System > system name “Sonus IP Office” and then click the Voicemail tab.

Note

Voicemail pro was installed on Enterprise network and was integrated with the IPO PBX. The voicemail server was hosted with IPO.

 

 

  1. For the voicemail type, select Voicemail Pro.
  2. Enter the Voicemail pro IP address in the Voicemail IP Address field.

VOIP

SIP Line - Line 17 

IPO SIP Line is a PBX SIP Trunk on IPO towards the Enterprise SBC Edge 1000. 

 

Transport
Note

Mention Transport with the next hop or Proxy IP address. During interoperability, the SBC Edge 1000 packet interface IP address was configured on the IPO SIP Trunk and SIP Line with transport protocol UDP and port 5060.

 

 

 

SIP URI

 

VOIP

 

T.38 Fax

 

SIP Credentials

 

Analog Extension

 

 

Analog 

 

SIP Extension

 

 

VOIP

 

T.38 Fax

H.323 Extension

 

 

 

VOIP

 

SIP User - 250

 

Voicemail

 

Short codes

 

SIP

 

 

Analog user - 211

 

 

 

SIP

H.323 User - 212 

 

 

Short codes

 

SIP

Voicemail

 

Incoming call Route 

 

 

Destinations

 

 

Short Code 

 

Ribbon SBC SWe (Core) Configuration

Prerequisites

The SBC SWe is deployed with the correct build and running with the necessary license.

 

SBC Configuration

This section provides a sample of the Ribbon SBC SWe configuration used during the interoperability testing. The following commands and configurations are only for reference, other configurations are also possible based on the customer's requirement.

Global Configuration

Codec Entry

Create Codec Entry with the supported codec in the network.

 

set profiles media codecEntry G729A-IOT-TEST dtmf relay rfc2833
set profiles media codecEntry G729A-IOT-TEST packetSize 20
commit
 
set profiles media codecEntry G711_ALAW_PTIME_20 dtmf relay rfc2833
set profiles media codecEntry G711_ALAW_PTIME_20 packetSize 20
commit

 

RTCP

Configuring the RTCP interval.

 

set system media mediaRtcpControl senderReportInterval 5
commit

DSP Resource Allocation

This configuration only applies if the SBC has been deployed with (hardware) DSP resources. If it has not, executing this configuration step has no negative impact.

 

Note

The subsequent configuration section (Packet Service Profiles) does not attempt transcoding, so the lack of compression resources will not impact the overall SBC configuration in this document.

 

 

set system mediaProfile compression 75 tone 25
commit

 

 

SBC Configuration for C20/AS Side 

Packet Service Profile (PSP)

Create a Packet Service Profile (PSP) for the C20/AS. The PSP will be specified within the SIP trunk group configuration.

 

set profiles media packetServiceProfile CORE_PSP codec codecEntry1 G729A-IOT-TEST
set profiles media packetServiceProfile CORE_PSP codec codecEntry2 G711_ALAW_PTIME_20
set profiles media packetServiceProfile CORE_PSP packetToPacketControl transcode transcoderFreeTransparency
set profiles media packetServiceProfile CORE_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg ""
set profiles media packetServiceProfile CORE_PSP packetToPacketControl codecsAllowedForTranscoding otherLeg ""
set profiles media packetServiceProfile CORE_PSP rtcpOptions rtcp enable
set profiles media packetServiceProfile CORE_PSP preferredRtpPayloadTypeForDtmfRelay 101
commit
 

 

IP Signaling Profile (IPSP)

Create an IP signaling profile for the C20/AS side. The IPSP will be specified within the SIP trunk group configuration.

 
set profiles signaling ipSignalingProfile CORE_IPSP 
set profiles signaling ipSignalingProfile CORE_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes flags noPortNumber5060 enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes relayFlags updateWithoutSdp enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags mwiBody enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags referredByHeader enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags sipfragBody enable
set profiles signaling ipSignalingProfile CORE_IPSP commonIpAttributes transparencyFlags unknownBody enable
set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes privacy transparency enable
set profiles signaling ipSignalingProfile CORE_IPSP ingressIpAttributes flags sendUpdatedSDPin200Ok enable
set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes transport type1 udp
set profiles signaling ipSignalingProfile CORE_IPSP egressIpAttributes transport type2 tcp
commit

 

IP Interface Group

Create an IP interface group and assign its interface and IP address.

set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 ceName IOTSBC1 portName pkt0
set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 ipAddress 172.28.xxx.xx
set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 prefix xx
set addressContext default ipInterfaceGroup CORE_LIF ipInterface IPIF2 mode inService state enabled
commit

 

Zone

This Zone groups the set of objects that communicate to the SESM AS.

set addressContext default zone CORE id 3
set addressContext default zone CORE remoteDeviceType appServer
commit


SIP Signaling Port

A SIP signaling port is a logical address that sends and receives SIP call signaling packets and is permanently bound to a specific zone.

 

set addressContext default zone CORE sipSigPort 3 ipInterfaceGroupName CORE_LIF ipAddressV4 172.28.xxx.xx portNumber 5060 transportProtocolsAllowed sip-udp,sip-tcp
set addressContext default zone CORE sipSigPort 3 state enabled mode inService
commit

 

DNS Group

DNS Groups set DNS objects that may be used for DNS resolution within a particular Zone.

 

set addressContext default dnsGroup EXT_DNS
set addressContext default dnsGroup EXT_DNS type ip interface IPIF2 server DNS2 ipAddress x.x.x.x state enabled
set addressContext default zone CORE dnsGroup EXT_DNS
commit


SIP Trunk Group

Create a SIP Trunk Group for the SESM-AS side and assign the IPSP and PSP configured above. For ingressIpPrefix, replace x.x.x.x with the IP address prefix that you want to allow. Multiple SESM IP addresses can be configured.

 

set addressContext default zone CORE sipTrunkGroup CORE_STG media mediaIpInterfaceGroupName CORE_LIF
set addressContext default zone CORE sipTrunkGroup CORE_STG signaling honorMaddrParam enabled
set addressContext default zone CORE sipTrunkGroup CORE_STG policy media packetServiceProfile CORE_PSP
set addressContext default zone CORE sipTrunkGroup CORE_STG policy signaling ipSignalingProfile CORE_IPSP
set addressContext default zone CORE sipTrunkGroup CORE_STG services dnsSupportType a-srv-naptr
set addressContext default zone CORE sipTrunkGroup CORE_STG ingressIpPrefix x.x.x.x x
set addressContext default zone CORE sipTrunkGroup CORE_STG signaling relayNonInviteRequest enabled
set addressContext default zone CORE sipTrunkGroup CORE_STG media sdpAttributesSelectiveRelay enabled
set addressContext default zone CORE sipTrunkGroup CORE_STG mode inService state enabled
commit


IP Static Route

Create a default route to the subnet's next hop IP for the interface and IP Interface Group.

 

set addressContext default staticRoute X.X.X.X X x.X.X.X  CORE_LIF IPIF2 preference 100
commit


IP Peer

Create an IP Peer with the AS IP address and assign it to the CORE zone.

 

set addressContext default zone CORE ipPeer CORE_PEER ipAddress x.x.x.x ipPort 5060       
commit

 

SBC configuration for the Enterprise SBC Edge 1000 side 

Packet Service Profile (PSP)

Create a Packet Service Profile (PSP) for the SBC Edge side. The PSP will be specified within the SIP Trunk Group Configuration.

 

set profiles media packetServiceProfile ACCESS_PSP codec codecEntry1 G729A-IOT-TEST
set profiles media packetServiceProfile ACCESS_PSP codec codecEntry2 G711_ALAW_PTIME_20
set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl transcode transcoderFreeTransparency
set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg ""
set profiles media packetServiceProfile ACCESS_PSP packetToPacketControl codecsAllowedForTranscoding otherLeg ""
set profiles media packetServiceProfile ACCESS_PSP rtcpOptions rtcp enable
set profiles media packetServiceProfile ACCESS_PSP preferredRtpPayloadTypeForDtmfRelay 101
commit

 

IP Signaling Profile (IPSP)

Create an IP Signaling Profile (IPSP) for the SBC Edge side. The IPSP will be specified within the SIP Trunk Group Configuration.

 

set profiles signaling ipSignalingProfile ACCESS_IPSP 
set profiles signaling ipSignalingProfile ACCESS_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags relayDataPathModeChangeFromOtherCallLeg enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags noPortNumber5060 enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags dialogEventPackage enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags info enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags refer enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags updateWithoutSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags authcodeHeaders enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags mwiBody enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags referredByHeader enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags sipfragBody enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags unknownBody enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes privacy transparency enable
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags sendUpdatedSDPin200Ok enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes transport type1 udp
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes transport type2 tcp
commit

 

IP Interface Group

Create IP Interface Group and assign its interface and IP address for connectivity towards the SBC Edge.

 

set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 ceName IOTSBC1 portName pkt1
set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 ipAddress xxx.xx.xxx.xx
set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 prefix xx
set addressContext default ipInterfaceGroup ACCESS_LIF ipInterface IPIF0 mode inService state enabled
commit

Zone

This Zone groups the set of objects that communicate with the SBC Edge 1000.

 

set addressContext default zone ACCESS id 2
commit

SIP Signaling Port

A SIP Signaling port is a logical address permanently bound to a specific zone that sends and receives SIP call signaling packets.

 

set addressContext default zone ACCESS id 2 sipSigPort 2 ipInterfaceGroupName ACCESS_LIF ipAddressV4 XXX.XXX.XXX.XXX portNumber 5060 transportProtocolsAllowed sip-tcp,sip-udp
set addressContext default zone ACCESS id 2 sipSigPort 2 mode inService state enabled
commit

SIP Trunk Group

Create a SIP Trunk Group towards the Enterprise SBC Edge 1000 side and assign the PSP and IPSP configured above.

set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG media mediaIpInterfaceGroupName ACCESS_LIF
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG signaling honorMaddrParam enabled
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG policy media packetServiceProfile ACCESS_PSP
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG policy signaling ipSignalingProfile ACCESS_IPSP
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG services dnsSupportType a-srv-naptr
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG ingressIpPrefix x.x.x.x x
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG signaling relayNonInviteRequest enabled
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG media sdpAttributesSelectiveRelay enabled
set addressContext default zone ACCESS sipTrunkGroup ACCESS_STG mode inService state enabled
commit
 

IP Peer

Create an IP Peer with the Fully-Qualified Domain Name (FQDN) or IP address of the endpoint and assign it to the PSTN Side.

 

set addressContext default zone ACCESS ipPeer ACCESS_PEER ipAddress x.x.x.x ipPort 5060       
commit

IP Static Route

Create a default route to the subnet’s next hop IP for the interface and IP Interface Group.

 

set addressContext default staticRoute X.X.X.X X X.X.X.1 ACCESS_LIF IPIF0 preference 100
commit

 

Routing Label

Create a Routing Label with a single Routing Label Route to bind the egress Trunk Group for the C20-AS and the Enterprise SBC Edge 1000 IP Peer.

 

set global callRouting routingLabel ACCESS_RL routingLabelRoute 1 trunkGroup ACCESS_STG ipPeer ACCESS_PEER inService inService
set global callRouting routingLabel CORE_RL routingLabelRoute 1 trunkGroup CORE_STG ipPeer CORE_PEER inService inService
commit

Routing

Routing must be put in place to send calls to the correct destination. Additional routing options may be used based on the requirements.

The configuration of both standard and username routes are done to ensure that no matter how the called party is addressed (a number or username) the SBC routes the message to the Core. Create Route entries for standard Trunk Group routing with Matching Criteria and a Routing Label destination.

set global callRouting route trunkGroup ACCESS_STG IOTSBC1 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel CORE_RL
set global callRouting route trunkGroup ACCESS_STG IOTSBC1 username Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel CORE_RL
set global callRouting route trunkGroup CORE_STG IOTSBC1 standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel ACCESS_RL
commit

 

Configure the Ribbon E-SBC (SBC Edge 1000)

Configuring the SBC 1000 is done through the SBC’s integrated web server. This guide assumes that the operator has already done the initial configuration that positions the SBC Edge 1000 on the IP network.

To start the configuration process, use a standard web browser to connect to the IP or FQDN address of the SBC. Supply the username and password to complete the login process.

 

 

Use the Initial Task configuration procedure to position the SBC Edge 1000 between the Ribbon SBC Core and Avaya IPO PBX. This task will create SIP components and call routing basics.

Avaya IPO PBX Side Configuration

Create profiles with a specific set of characteristics that correspond to the Avaya IPO PBX. This profile creation includes configuring the following entities on the SBC Edge 1000.

  • Media Profile
  • Media List
  • SIP Profile
  • SIP Server Table
  • Signaling Group
  • Call Routing Table

 

Media Profile

Create a Media Profile towards the Interactive Intelligence side with G711a law as the first codec, G711u law as the second codec, and T.38 Fax as the third codec.

To create or modify a Media Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media Profile.

Media List

Media List allows you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which you must first define in Media List. These lists allow you to accommodate specific transmission requirements and SIP devices that only implement a subset of the available voice codecs.

To create or modify a Media List:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media List.

SIP Profile

To create or modify an existing SIP Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Profiles.

 

SIP Server Table

SIP Server Tables contain information about the SIP devices connected to the SBC Edge 1000. The entries in the tables provide information about the IP Addresses, ports, and transport protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.

To create or modify an existing SIP Server Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Server Tables.

 

Signaling Group

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, they specify protocol settings and link to server, media, and mapping tables.

To create or modify an existing Signaling Group:

  • In the WebUI, click the Settings tab.
  • In the left navigation pane, click Signaling Groups.
  • From the Create Signaling Group drop down box, click SIP Signaling Group.

 


IP Address/FQDN in Federated IP/FQDN: Specifies the IP Address or Fully Qualified Domain Name of a server from which the SBC Edge 1000 will accept SIP messages. It should have the IP of Avaya IPO PBX.

Call Routing Table

Call Routing allows calls to be carried between signaling groups, which allows calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

To create or modify an existing Call Routing Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Call Routing Table.

 

 

The following are definitions of fields in the Call Routing Table:

  • Destination Signaling Group: Specifies the Signaling Groups used as the destination of calls.

  • Media Mode: Media Mode should be RTP DSP.

  • Media Transcoding: It should be enabled in order for the SBC Edge 1000 to transcode.

  • Media List: Specifies the Media List used for this call route.

SBC Core Side Configuration

Create profiles with a specific set of characteristics that correspond to the SBC Core. It includes configuring the following entities on the SBC Edge 1000:

  • Media Profile
  • Media List
  • SIP Profile
  • SIP Server Table
  • Signaling Group
  • Call Routing Table

Media Profile

Create Media Profile towards Interactive Intelligence side with G711a law as the first codec, G711u law as the second codec, and T.38 Fax as the third.

To create or modify a Media Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media Profile.

Media List

Media List allows you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which must first be defined in Media List. These lists allow you to accommodate specific transmission requirements and SIP devices that only implement a subset of the available voice codecs.

To create or modify a Media List:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media List.

SIP Profile

To create or modify an existing SIP Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Profiles.

 

SIP Server Table

SIP Server Tables contain information about the SIP devices connected to the SBC Edge 1000. The entries in the tables provide information about the IP Addresses, ports, and transport protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.

To create or modify an existing SIP Server Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Server Tables.

Signaling Group

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, they specify protocol settings and link to server, media, and mapping tables.

To create or modify an existing Signaling Group:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Signaling Groups.
  3. From the Create Signaling Group drop down box, select SIP Signaling Group.

 

 

 

 

 

 

 


IP Address/FQDN in Federated IP/FQDN: Specifies the IP Address or Fully Qualified Domain Name of a server from which the SBC Edge 1000 will accept SIP messages. It should have the IP of Avaya IPO PBX.

Call Routing Table

Call Routing allows calls to be carried between signaling groups, which allows calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation TablesMessage TranslationsCause Code Reroute TablesMedia Lists and the three types of Signaling Groups (ISDNSIP and CAS).

To create or modify an existing Call Routing Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Call Routing Table.

 

 

 


The following are descriptions of some fields in the Call Routing Table:

  • Destination Signaling Group: Specifies the Signaling Groups used as the destination of calls.

  • Media Mode: Media Mode should be RTP DSP.

  • Media Transcoding: It should be enabled in order for SBC Edge 1000 to transcode.

  • Media List: Specifies the Media List used for this call route.

Configure Ribbon AS

System Management Console Configuration

To configure the Static SIP PBX you will need access to the Ribbon AS System Management Console.

Documentation

The following documentation was used to configure the AS: 630-01981-01_02.01_1.2_C20_ Enterprise SIP Trunking Solution C20-EXPERiUS SIP Trunking Solution Guide.

Static SIP PBX Addresses

The Static SIP PBX does not send SIP REGISTER messages to notify the network of their signaling address, therefore, the address of the SIP PBX must be provisioned in the network. Configure the address of the private side of the SBC that represents the static SIP PBX.

Select Network Data and Mtc > Addresses. Add an address as shown in the following example.

 

Static SIP PBX External Nodes

Configure an external node by mapping the Static SIP PBX address to the node name. Select Network Data and Mtc > External Nodes. Add an External Node as shown in the example below.

C20 SIP PBX

From Network Data and Mtc, select C20 Converged Softswitch Integration, SIP PBX, and then C20 SIP PBX. Add a C20 SIP PBX as referenced below using the C20 CIP PBX “tkv4st.”

 

AYT Profiles

The AYT Profile is mandatory if the AYT Audit was selected in the C20 SIP PBX setup as shown in the preceding screen capture. Here you will select SIP OPTIONS or INFO messages will be used and valid responses to the AYT audit. The AYT audits will be sent by the C20/AS to monitor the connectivity to the SIP PBX. From Network Data and Mtc select AYT Profiles.

 

SIPTrunkingdefaultOptions was used during testing.

SIP/PRI Header Mapping

A new Header Map was created for this test. This maps specific SIP messages to specific PRI messages. The default Header Map was modified (see the following screen capture). From Network Data and Mtc select the C20 Converged Softswitch Integration folder and then SIP/PRI Header Mapping.

 

 

Note

The PRI header Original Called Party Number was changed from the History-Info header to the Diversion header. This instructs the AS to use the Diversion header containing the Original Called party Number. When the Call is sent to the CIM Voice Mail server, the Diversion header is in the INVITE message.

 

SIP Profiles

A custom SIP Profile may be required to support different types of SIP PBX. In this case Ribbon recommends you copy the SipPBX SIP Profile and only change the settings required to support the particular SIP PBX. From Network Data and Mtc select the SIP Profiles folder and then SIP Profiles.

Then give the profile a name and description.


 

Select Requests

The following default settings are viewed when clicking the Select Requests link.

 

Select Headers

The following default settings are viewed when clicking the Select Headers link.

 

Select Allow Methods

The following default settings are viewed when clicking the Select Allow Methods link.

SIP Authorized Methods

Configure the AS to require authentication for both SIP INVITE and REGISTER transactions. From Network Elements select Session Managers, SessionManagerX, and then Authorized Methods.

 

SESM Configuration Parameters

SESM Configuration Parameters and associated values are set globally within the AS and are not unique to the SIP PBX. From Network Elements select Session Managers, SESMx, and then Configuration Parameters.

 

 

Authentication With Integrity

For each AS SESM, set the AuthenticationWithIntegrity parameter to false and Parm Group = Authentication

 

Long Call Duration Audits

This parameter determines the length of time, in minutes, between endpoint audits. Duration is used to detect abandoned calls. A value of zero deactivates the duration parameter. 

Note

Make sure Parm Group = LongCall.

 

 

SIP PBX Configuration

Defines a name for the SIP PBX to be used in other associated provisioning entities.

From Network Elements select Session Managers > SessionManager1 > C20 Converged Softswitch Integration > SIP PBX Integration > SIP PBX Configuration. The SIP PBX Configuration used for our setup was Tkv4StaticSipPbx.

 

 

 

SIP PBX Route Configuration

Establishes links between the AS SIP PBX entity and the C20 by associating the SIP PBX name, defined in the AS SIP PBX Configuration, with an ISDN Logical Terminal ID (LTID) and Virtual Media Gateway defined in the C20.

 

SIP PBX Link Maintenance

After the SIP PBX is configured in the Provisioning Manager shown in the following screen capture, you will be able to bring up the link.

 

 

 

AS Provisioning Manager

The AS Provisioning Manager is the AS web-based interface accessible at the following link: http://<prov blade IP address>:8443/prov.

Service Node for SIP PBX

Defines the SIP PBX Entity, established through the AS System Manager, as an AS Service Node. In the Translations tab select Service Node.

The following are definitions of fields in the Service Node screen:

  • Node type: Select a node type based on the SIP Profile that has been established for this SIP PBX.
  • Select Address Name: Select the short name of the SIP PBX.
  • Location: Choose Other unless another location is required for the SIP.
  • PBX. Location choices are limited to those available within the SIP PBX’s assigned domain “Is trusted”: When selected the AS will send the P-Asserted-Identity header to the SIP PBX.
  • For the Node name assign the Domain to the Node by clicking the Blue Domain link.

 

SIP PBX

From the Solution tab select SIP PBX then create a SIP PBX. Specify a username and password for the SIPPBX that will be used to authenticate associated transactions specified in the Authorized Methods.

The following is a SIP PBX example.

 

 

 

 

 

 

Configure Ribbon C20

C20 Management Tools

Gateway

Add a Gateway of type VOIP_VPN associated with the SIP PBX Entity (in this case tekVizion4SIP-PBX). The IP address must be the service address of the AS SESM on which the SIP PBX is configured.

Carriers

Define Carrier(s) that will be used to assign the ISDN PRI trunks in C20 associated with the SIP PBX Entity. In this example, carrier EPG_002 for (SIP PBX) gateway tekVizion4SIP-PBX has 23 channels assigned starting at 2. These will be used as bearer channels. Carrier EPG_001 for tekVizion4SIP-PBX has a single channel (1) that will serve as the ISDN PRI “D channel” for signaling.

Note

The node number is (92).

C20 Call Agent

Table CLLI

Define the Trunk Group name and size.

Table TRKGRP

Define an ISDN PRI trunk group supporting the SIP PBX.

GRPKEY is derived from table CLLI.

GRPTYP = PRA is required for ISDN PRI trunk group.

LTID is derived from table LTDEF and cannot be datafilled manually. Therefore, enter $ for this field when initially adding the SIP PBX trunk group.

 

Table TRKSGRP

Define a trunk subgroup entry for the SIP PBX ISDN PRI trunk group SGRPKEY value TEKVIZION is derived from table CLLI; 0 means it is the 0th trunk subgroup for TEKVIZION.

SGRPVAR ISDN is required to indicate ISDN PRI.

PMTYPE, GWCNO, GWCNODENO, and GWCTRMNO are all values based on the SIP PBX gateway defined by the Gateway Controller provisioning.

Note

The value for GWCTRMNO (92), defined in Gateway Controller Carrier provisioning, is the D channel for the SIP PBX ISDN PRI trunk group.

 

 

Table TRKMEM

Define trunk members for the SIP PBX trunk group using the bearer path carrier channels (total 23) defined at the Gateway Controller.

 

Table LTDEF

Define the Logical Terminal ID ISDN 15 for use with the SIP PBX trunk group.

Table LTMAP

Map the Logical Terminal ID, defined in table LTDEF, to the SIP PBX trunk group
(TEKVIZION_4).

  • LTKEY is the LTID defined in table LTDEF
  • CLLI is the CLLI of the SIP PBX trunk group
  • All remaining values are DEFAULT

Table LTCALLS

Define translation paths for LTID ISDN 15.

Public routes use XLARTE (translation route selector) XLALEC.

 

Table LTDATA

Define service data associated with LTID ISDN 15.

OPTION PRI_IP_PROT is used to define the IP protocol for ISDN 14 as SIP (required for SIP PBX trunks).

 

Table MSGRTE

If using centralized C20 voice mail, table MSGRTE is used to route message waiting indications back to the mailbox subscriber.

In this example, any MWI for DNs in this range are sent to the PRA trunk group TEKVIZION_4 (LTID ISDN 15).

 

Routing

PSTN to Avaya IP Office call.

An example of the translation flow to route calls from PSTN line to SIP PBX trunk members is shown in the TRAVER outputs below. 

Avaya IP Office to PSTN Call

An example of the translation flow to route calls from SIP PBX trunk members to PSTN line is shown in the TRAVER outputs below. 

 

Dynamic Registration SIP PBX Setup

Dynamic PBX Registration for Avaya IPO is supported by C20-AS. Currently the Ribbon SBC Edge does not support the Dynamic Registration feature. So dynamic PBX registration related configuration is out of scope in this configuration guide.

Conclusion

This detailed document describes the configuration steps required for the Ribbon SESM AS, C20, SBC Core, and SBC Edge 1000 to successfully interoperate with Avaya IPO over SIP Trunking solution.

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