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Document Overview

This document is a configuration guide for the Sonus SBC 5XX0 Series (Session Border Controller) when connecting to Skype for Business 2015 (Skype 2015).

This configuration guide supports features provided in Skype for Business 2015, Configuration Guides.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 5XX0 and Skype for Business 2015.

Audience

This technical document is intended for telecommunication engineers for the purpose of configuring the Sonus SBC 5XX0 series aspects of the Virgin Media SIP trunk group together with Skype 2015. There will be steps that require navigating third-party equipment and the Sonus SBC Web browser user interfaces, as well as the Embedded Management Application (EMA). Understanding basic concepts for IP/Routing, SIP/RTP, and TLS are also necessary to complete the configuration and perform any troubleshooting, if necessary.

 

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.

Requirements

The following equipment and software were used for the provided sample configuration:

Table : Requirements

Equipment

Software Version

Sonus Networks

Sonus SBC 5200
BMC
BIOS
ConnexIP OS
SonusDB
EMA
SBX

V05.00.01-R002
V02.11.00
V02.06.00
V03.00.01-R002
V05.00.01-R002
V05.00.01-R002
V05.00.01-R002

Tenor AFP108-09-21

Third-party Equipment

Microsoft Skype for Business 2015

6.0.9319.0

Polycom CX600 Lync Edition4.0.7577.4372

Reference Configuration

The following reference configuration shows connectivity between third-party equipment and the Sonus SBC 5XX0.

 

Figure : Reference Configuration


Support

For any questions regarding this document or the content herein, please contact your maintenance and support provider.

 

Third-Party Product Features

The following features were tested using the Virgin Media test plan:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding
  • FAX
  • DTMF
  • Conference Call
  • Action on eSBC outage (loss of Ethernet , restart of eSBC)
  • Action on Loss of Virgin Media primary SBC

Verify License

No special licensing required.

Skype 2015 Configuration

The following new configurations are included in this section:
  1. PSTN Gateway
  2. Voice Policy
  3. PSTN Usage
  4. Route
  5. Trunk Configuration

1. PSTN Gateway

To configure the PSTN Gateway, select Topology Builder > Shared Components > PSTN Gateways, as shown in the following figures.

 

Figure : New PSTN Gateway

 

 

Figure : Define FQDN

 

 

Figure : Define IP Address Version

 

 

Figure : Define the Root Trunk

2. Voice Policy

To configure Voice Policy, select Control Panel > Voice Routing > Voice Policy, as shown in the following figure.

 

Figure : Voice Policy

3. PSTN Usage

To configure the PSTN Usage, select Control Panel > Voice Routing > PSTN Usage, as shown in the following figure.

 

Figure : PSTN Usage

4. Route

To configure Route, select Control Panel > Voice Routing > Route, as shown in the following figure.

 

Figure : Route

5. Trunk Configuration

To configure the Trunk, select Control Panel > Voice Routing > Trunk Configuration, as shown in the following figure.

 

Figure : Trunk Configuration

 

SBC 5XX0 Configuration

The following commands are provided to configure the the Sonus SBC 5XX0 Series to successfully interoperate with Virgin Media SIP Trunk:


CLI Configuration
#Skype codecs
set profiles media codecEntry SKYPE_G711U codec g711 packetSize 20 law ULaw
set profiles media codecEntry SKYPE_G711A codec g711 packetSize 20 law ALaw
set profiles media codecEntry SKYPE_G711U dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry SKYPE_G711A dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry SKYPE_G711U fax failureHandling continue toneTreatment none
set profiles media codecEntry SKYPE_G711A fax failureHandling continue toneTreatment none
set profiles media codecEntry SKYPE_G711U modem failureHandling continue toneTreatment none
set profiles media codecEntry SKYPE_G711A modem failureHandling continue toneTreatment none
commit


#VIRGIN_MEDIA codecs
set profiles media codecEntry VIRGIN_MEDIA_G729A codec g729a packetSize 20 preferredRtpPayloadType 128
set profiles media codecEntry VIRGIN_MEDIA_G711A codec g711 packetSize 20 law ALaw
set profiles media codecEntry VIRGIN_MEDIA_G711U codec g711 packetSize 20 law ULaw
set profiles media codecEntry VIRGIN_MEDIA_G711A dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry VIRGIN_MEDIA_G729A dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry VIRGIN_MEDIA_G711U dtmf relay rfc2833 removeDigits enable
set profiles media codecEntry VIRGIN_MEDIA_G711U fax failureHandling continue toneTreatment faxRelay
set profiles media codecEntry VIRGIN_MEDIA_G711A fax failureHandling continue toneTreatment faxRelay
set profiles media codecEntry VIRGIN_MEDIA_G729A fax failureHandling continue toneTreatment faxRelay
set profiles media codecEntry VIRGIN_MEDIA_G711U modem failureHandling continue toneTreatment applyFaxTreatment
set profiles media codecEntry VIRGIN_MEDIA_G729A modem failureHandling continue toneTreatment applyFaxTreatment
set profiles media codecEntry VIRGIN_MEDIA_G711A modem failureHandling continue toneTreatment applyFaxTreatment
commit


#Cranback profile
set profiles callRouting crankbackProfile default reason 151 useLocationValue disabled


#Internal Side Configuration
#IP Interface Group
set addressContext default ipInterfaceGroup Private ipInterface pkt0 ceName BARTONCE portName pkt0 ipAddress 10.35.177.239 prefix 26 mode outOfService state disabled
set addressContext default ipInterfaceGroup Private ipInterface pkt0 mode inService state enabled
commit


#IP Static Route
set addressContext default staticRoute 0.0.0.0 0 10.35.177.193 Private pkt0 preference 100
commit


#SBC Configuration for Skype 2015 Trunk
#Packet Service Profile (PSP)
set profiles media packetServiceProfile SKYPE_PSP codec codecEntry1 SKYPE_G711A codecEntry2 SKYPE_G711U
set profiles media packetServiceProfile SKYPE_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711a,g711u,g729 otherLeg g711a,g711u,g729
set profiles media packetServiceProfile SKYPE_PSP packetToPacketControl conditionsInAdditionToNoCommonCodec applyFaxToneTreatment disable differentDtmfRelay enable differentPacketSize enable differentSilenceSuppression enable  honorOfferPreference disable honorAnswerPreference disable different2833PayloadType enable
set profiles media packetServiceProfile SKYPE_PSP packetToPacketControl transcode conditional
commit


#IP Signaling profiles
set profiles signaling ipSignalingProfile SKYPE_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable publishIPInHoldSDP enable routeUsingRecvdFqdn enable sendPtimeInSdp enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags sendRtcpPortInSdp enable storePChargingVector enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes optionTagInRequireHeader suppressReplaceTag enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile SKYPE_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile SKYPE_IPSP ingressIpAttributes flags suppress183WithoutSdp enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes privacy flags includePrivacy enable msLyncPrivacySupport enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes privacy transparency disable privacyInformation pAssertedId
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes transport type1 tcp type2 none type3 none type4 none
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes numberGlobalizationProfile Skype
commit


#Feature Control Profile
set profiles featureControlProfile SKYPE_FCP processDestinationTgrp disable processDestinationTrunkContext disable processEnumdi disable
set profiles featureControlProfile SKYPE_FCP ipProtocolFlags useIpProtocol enable defaultCalledUser enable honorPhoneContextParameter disable
commit


#zone
set addressContext default zone ZONE-INT-VoIP id 50
commit


#SIP signaling port
set addressContext default zone ZONE-INT-VoIP sipSigPort 25 ipInterfaceGroupName Private ipAddressV4 10.35.177.239 portNumber 5060 mode outOfService state disabled siprec disabled transportProtocolsAllowed sip-udp sip-tcp
set addressContext default zone ZONE-INT-VoIP sipSigPort 25 mode inService state enabled
commit


#IP Peer
set addressContext default zone ZONE-INT-VoIP ipPeer SKYPE_PEER policy sip fqdnPort 0
set addressContext default zone ZONE-INT-VoIP ipPeer SKYPE_PEER ipAddress 10.35.180.229 ipPort 5068 defaultForIp false
set addressContext default zone ZONE-INT-VoIP ipPeer SKYPE_PEER authentication incInternalCredentials enabled intChallengeResponse enabled 
commit


#SIP trunk group
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 media mediaIpInterfaceGroupName Private
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 ingressIpPrefix 10.35.180.229 32
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy callRouting elementRoutingPriority SKYPE_ERP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy digitParameterHandling numberingPlan UK_NUM_PLAN
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy media packetServiceProfile SKYPE_PSP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy signaling ipSignalingProfile SKYPE_IPSP signalingProfile DEFAULT_IP_PROFILE
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy digitParameterHandling egressDmPmRule SKYPE_UK_CLD
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 signaling honorMaddrParam enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 signaling authentication intChallengeResponse enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 policy country 44 featureControlProfile SKYPE2015_FCP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-SKYPE2015 state enabled mode inService
commit


#SBC Configuration for Fax Trunk
#Packet Service Profile
set profiles media packetServiceProfile TENORGW_PSP codec codecEntry1 TENORGW_G711u codecEntry2 TENORGW_G729
set profiles media packetServiceProfile TENORGW_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711u,g729,t38 otherLeg g711u,g729,t38
set profiles media packetServiceProfile TENORGW_PSP packetToPacketControl transcode only
set profiles media packetServiceProfile TENORGW_PSP rtcpOptions rtcp enable
commit


#IP signaling profile
set profiles signaling ipSignalingProfile TENORGW_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile TENORGW_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile TENORGW_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable publishIPInHoldSDP enable sendPtimeInSdp enable
set profiles signaling ipSignalingProfile TENORGW_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile TENORGW_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile TENORGW_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable sendSdpInSubsequent18x enable
set profiles signaling ipSignalingProfile TENORGW_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile TENORGW_IPSP egressIpAttributes transport type1 udp type2 none type3 none type4 none
commit



#IP peer
set addressContext default zone ZONE-INT-VoIP ipPeer TENORGW_PEER policy sip fqdnPort 0
set addressContext default zone ZONE-INT-VoIP ipPeer TENORGW_PEER ipAddress 10.35.137.43 ipPort 5084 defaultForIp false
set addressContext default zone ZONE-INT-VoIP ipPeer TENORGW_PEER authentication incInternalCredentials enabled intChallengeResponse enabled
commit


#SIP trunk group
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW media mediaIpInterfaceGroupName Private
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW signaling messageManipulation inputAdapterProfile CenturyLink_HashIn includeAppHdrs disabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW signaling rel100Support enabled acceptHistoryInfo enabled
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW ingressIpPrefix 10.35.137.43 32
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW policy media packetServiceProfile TENORGW_PSP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW policy signaling ipSignalingProfile TENORGW_IPSP signalingProfile TENORGW_SP
set addressContext default zone ZONE-INT-VoIP sipTrunkGroup TWO-WAY-SIP-TENORGW state enabled mode inService
commit


#External Side SBC Configuration
#IP Interface Group
set addressContext default ipInterfaceGroup Public ipInterface pkt2 ceName BARTONCE portName pkt2 ipAddress 216.110.2.228 prefix 28 mode outOfService state disabled
set addressContext default ipInterfaceGroup Public ipInterface pkt2 mode inService state enabled
commit


#IP static route
set addressContext default staticRoute 0.0.0.0 0 216.110.2.225 Public pkt2 preference 100
commit


#SBC Configuration for VIRGIN_MEDIA SIP Trunk
#Packet Service Profile
set profiles media packetServiceProfile VIRGIN_MEDIA_PSP codec codecEntry1 VIRGIN_MEDIA_G711A codecEntry2 VIRGIN_MEDIA_G711U codecEntry3 VIRGIN_MEDIA_G729A
set profiles media packetServiceProfile VIRGIN_MEDIA_PSP packetToPacketControl codecsAllowedForTranscoding thisLeg g711a,g711u,g729,t38 otherLeg g711a,g711u,g729,t38
set profiles media packetServiceProfile VIRGIN_MEDIA_PSP packetToPacketControl conditionsInAdditionToNoCommonCodec differentDtmfRelay enable differentPacketSize enable differentSilenceSuppression enable honorOfferPreference enable 
honorAnswerPreference enable
set profiles media packetServiceProfile VIRGIN_MEDIA_PSP packetToPacketControl transcode conditional
commit


#IP signaling profile
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP ipProtocolType sipOnly
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP commonIpAttributes flags disableMediaLockDown enable includeReasonHeader enable
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP commonIpAttributes flags minimizeRelayingOfMediaChangesFromOtherCallLegAll enable sendPtimeInSdp enable sendRtcpPortInSdp enable storePChargingVector enable
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP egressIpAttributes flags disable2806Compliance enable disableOptionalRegisterParameters enable
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP egressIpAttributes privacy flags includePrivacy enable
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP egressIpAttributes privacy transparency disable privacyInformation pAssertedId
set profiles signaling ipSignalingProfile VIRGIN_MEDIA_IPSP egressIpAttributes sipHeadersAndParameters sipToHeaderMapping calledNumber
commit


#Feature Control Profile
set profiles featureControlProfile VIRGIN_MEDIA_FCP processDestinationTgrp disable processDestinationTrunkContext disable processEnumdi disable
set profiles featureControlProfile VIRGIN_MEDIA_FCP ipProtocolFlags useIpProtocol enable defaultCalledUser enable honorPhoneContextParameter disable
commit

#PatchCheck Profile
set profiles services pathCheckProfile VIRGIN_MEDIA_PCHP protocol sipOptions sendInterval 30 recoveryCount 3

#Zone
set addressContext default zone ZONE-EXT-ACCESS id 20
commit
 
#SIP signaling port
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 ipInterfaceGroupName Public ipAddressV4 216.110.2.227 portNumber 5060 mode outOfService state transportProtocolsAllowed sip-udp
set addressContext default zone ZONE-EXT-ACCESS sipSigPort 10 mode inService state enabled
commit


#IP peer
set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER1 policy sip fqdnPort 0
set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER1 ipAddress 213.106.222.178 ipPort 5060 defaultForIp false
set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER1 pathCheck profile VIRGIN_MEDIA_PCHP state enabled 
commit


set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER2 policy sip fqdnPort 0
set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER2 ipAddress 82.14.171.226 ipPort 5060 defaultForIp false
set addressContext default zone ZONE-EXT-ACCESS ipPeer VIRGIN_MEDIA-PEER2 pathCheck profile VIRGIN_MEDIA_PCHP state enabled 
commit


#SIP trunk group
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA media mediaIpInterfaceGroupName Public sourceAddressFiltering enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA signaling rel100Support enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA signaling authentication authUserPart xxx authPassword yyy intChallengeResponse enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA ingressIpPrefix 213.106.222.178 32
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy callRouting elementRoutingPriority VIRGIN_MEDIA_ERP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy digitParameterHandling numberingPlan UK_NUM_PLAN
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy media packetServiceProfile VIRGIN_MEDIA_PSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy signaling ipSignalingProfile VIRGIN_MEDIA_IPSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy country 44 featureControlProfile VIRGIN_MEDIA_FCP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA policy digitParameterHandling egressDmPmRule VIRGIN_MEDIA_OUTBOUND
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA state enabled mode inService
commit


#SIP trunk group
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 media mediaIpInterfaceGroupName Public sourceAddressFiltering enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 signaling rel100Support enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 signaling authentication authUserPart xxx authPassword yyy intChallengeResponse enabled
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 ingressIpPrefix 82.14.171.226 32
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy callRouting elementRoutingPriority VIRGIN_MEDIA_ERP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy digitParameterHandling numberingPlan UK_NUM_PLAN
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy media packetServiceProfile VIRGIN_MEDIA_PSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy services classOfService DEFAULT_IP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy signaling ipSignalingProfile VIRGIN_MEDIA_IPSP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy country 44 featureControlProfile VIRGIN_MEDIA_FCP
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 policy digitParameterHandling egressDmPmRule VIRGIN_MEDIA_OUTBOUND
set addressContext default zone ZONE-EXT-ACCESS sipTrunkGroup TWO-WAY-VIRGIN_MEDIA2 state enabled mode inService
commit




#Global Call Routing Configuration
#Element Routing Priority
set profiles callRouting elementRoutingPriority VIRGIN_MEDIA_ERP entry nationalType 2 entityType none
set profiles callRouting elementRoutingPriority VIRGIN_MEDIA_ERP entry nationalType 1 entityType trunkGroup
set profiles callRouting elementRoutingPriority VIRGIN_MEDIA_ERP entry internationalType 2 entityType none
set profiles callRouting elementRoutingPriority VIRGIN_MEDIA_ERP entry internationalType 1 entityType trunkGroup
commit


set profiles callRouting elementRoutingPriority SKYPE_ERP entry nationalType 2 entityType none
set profiles callRouting elementRoutingPriority SKYPE_ERP entry nationalType 1 entityType trunkGroup
set profiles callRouting elementRoutingPriority SKYPE_ERP entry internationalType 2 entityType none
set profiles callRouting elementRoutingPriority SKYPE_ERP entry internationalType 1 entityType trunkGroup
commit


#Skype 2015 Routing
set global callRouting routingLabel TO_TWO_WAY_SKYPE routingLabelRoute 0 trunkGroup TWO-WAY-SIP-SKYPE2015 ipPeer SKYPE_PEER proportion 0 cost 1000000 inService inService testing normal
set global callRouting routingLabel TO_TWO_WAY_SKYPE overflowNOA none overflowNPI none routePrioritizationType sequence action routes numRoutesPerCall 10
commit
set global callRouting route trunkGroup TWO-WAY-VIRGIN_MEDIA BARTONCE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_SKYPE2015
commit


#VIRGIN_MEDIA routing
set global callRouting routingLabel TO_TWO_WAY_VIRGIN_MEDIA routingLabelRoute 0 trunkGroup TWO-WAY-VIRGIN_MEDIA ipPeer VIRGIN_MEDIA-PEER1 proportion 0 cost 1000000 inService inService testing normal
set global callRouting routingLabel TO_TWO_WAY_VIRGIN_MEDIA routingLabelRoute 1 trunkGroup TWO-WAY-VIRGIN_MEDIA2 ipPeer VIRGIN_MEDIA-PEER2 proportion 0 cost 1000000 inService inService testing normal
set global callRouting routingLabel TO_TWO_WAY_VIRGIN_MEDIA overflowNOA none overflowNPI none routePrioritizationType sequence action routes numRoutesPerCall 10
commit
set global callRouting route trunkGroup TWO-WAY-SIP-SKYPE2015 BARTONCE standard Sonus_NULL Sonus_NULL all all ALL none Sonus_NULL routingLabel TO_TWO_WAY_VIRGIN_MEDIA
commit


#Digit Parameter Handling
set profiles digitParameterHandling dmPmCriteria VIRGIN_MEDIA_CLD_INT criteriaType digit digitCriteria natureOfAddress value international
set profiles digitParameterHandling dmPmCriteria VIRGIN_MEDIA_CLI_INT criteriaType digit digitCriteria natureOfAddress value international
set profiles digitParameterHandling dmPmCriteria VIRGIN_MEDIA_CLI_NAT criteriaType digit digitCriteria natureOfAddress value national
set profiles digitParameterHandling dmPmCriteria VIRGIN_MEDIA_CLD_NAT criteriaType digit digitCriteria numberLength operation greaterThanOrEquals value 7
set profiles digitParameterHandling dmPmCriteria VIRGIN_MEDIA_CLD_NAT criteriaType digit digitCriteria natureOfAddress value national


set profiles digitParameterHandling dmPmRule VIRGIN_MEDIA_OUTBOUND subRule 0 criteria VIRGIN_MEDIA_CLD_INT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString calledNumber startDigitPosition 0 numberOfDigits 1 value +
set profiles digitParameterHandling dmPmRule VIRGIN_MEDIA_OUTBOUND subRule 1 criteria VIRGIN_MEDIA_CLD_NAT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString calledNumber startDigitPosition 0 numberOfDigits 3 value +44
set profiles digitParameterHandling dmPmRule VIRGIN_MEDIA_OUTBOUND subRule 2 criteria VIRGIN_MEDIA_CLI_INT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString callingNumber startDigitPosition 0 numberOfDigits 1 value +
set profiles digitParameterHandling dmPmRule VIRGIN_MEDIA_OUTBOUND subRule 3 criteria VIRGIN_MEDIA_CLI_NAT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString callingNumber startDigitPosition 0 numberOfDigits 3 value +44

set profiles digitParameterHandling dmPmRule SKYPE_UK_CLD subRule 0 criteria VIRGIN_MEDIA_CLD_NAT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString calledNumber startDigitPosition 0 numberOfDigits 3 value +44
set profiles digitParameterHandling dmPmRule SKYPE_UK_CLD subRule 1 criteria VIRGIN_MEDIA_CLI_NAT ruleType digit digitManipulation digitStringManipulation replacement type constant digitString callingNumber startDigitPosition 0 numberOfDigits 1 value +

 

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Interoperability Test Results

The following table provides results from interoperability compliance testing.

 

Table : Test Results

Interoperability Test NoTest ScenarioTest Setup and Result SummaryTest StatusComments
IOP1vendors eSBC response to SIP OPTIONS messages from SBCNo calls are required for this test. SIP trace to be captured for approx 60 seconds and checked for correct signalling.

For each eSBC, the SBC will periodically send an OPTIONS request to the vendors eSBC to check if its SIP stack is reachable. If a SIP response 200 OK is received from the IP-PBX, the SIP trunk will be placed (or remain) in an In-Service state

e.g. OPTIONS sip:ping@<ip-pbx_IP_Addr>:5060 SIP/2.0
Pass 
IOP2SBC response to SIP OPTIONS messages from vendor eSBCNo calls are required for this test. SIP trace to be captured for approx 60 seconds (depending on agreement) and checked for correct signalling.

Vendors eSBC setup for Solution IP.Addr Mode
eSBC configured to send OPTIONS messages to the SBC on a periodic basis. The SBC responds with SIP response 200OK -
e.g. "OPTIONS sip:ping@192.168.1.10:5060 SIP/2.0"

Check that the eSBC can simultaneously send SIP OPTIONS messages to both the solution SBC addresses.
Pass 
IOP4Basic test call from IP-PBX to PSTN line through SBC-A (using SBC-A IPV4 ip address).IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-A, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
e.g.
Request-Line: INVITE sip:<B-party>@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-A ip.addr TBD>

Check the wireshark trace and confirm that G.711 A law codec with 10 or 20ms packetisation is being used.
Also check to see if INVITE contains Session-Expires header and that it is syntatically correct. Check for Supported Header to see if 'timer' is supported. Ensure response in 200 OK is compatible with INVITEand check for Required Header and if it contains 'timer'. (x-ref IOP9)
Pass 
IOP5Basic test call from IP-PBX to PSTN line through SBC-B (using SBC-B IPV4 ip address)IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-B, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
e.g.
Request-Line: INVITE sip:<B-party>@<SBC-B ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-B ip.addr TBD>

Check the wireshark trace and confirm that G.711 A law codec with 10ms packetisation is being used.
Pass 
IOP7bCalled Number format - vendors eSBC to soft switch number normalisation - Global Dial Plan

Test eSBC capability to send the called number in  one of the following Global number formats (user part of  Request & To URIs)

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX line terminates call.

Configure the eSBC to present the called number in the user part of the Request & To URIs to be sent in one of the following formats

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP8bCalling Number format - vendors eSBC to soft switch number normalisation - Global Dial Plan

Test eSBC capability to send calling number in one of the following Global number formats (user part of FROM & PAI URIs)

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
00yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX terminates call.

Configure the eSBC to present the calling number in the user part of the From & PAI URIs to be sent in the one of the following formats

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
00yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP9bCalled Number format - soft switch to eSBC number normalisation - Global Dial Plan

Test eSBC capability of accepting the called number in one of the following Global number formats (user part of Request & To URIs)

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

Configure the eSBC to accept the called number in the user part of the Request & To URIs in one of the following formats

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Also check to see that the INVITE contains Session-Expires header and that it is syntactically correct. Check for Supported Header and ensure 'timer' is supported. Ensure response in 200 OK is compatible with INVITE and check for Required Header and if it contains 'timer'.
Pass 
IOP10bCalling Number format - soft switch to eSBC number normalisation - Global Dial Plan

Test eSBC capability of accepting the calling number in one of the following Global number formats (user part of FROM & PAI URIs)  

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

Configure the eSBC to accept the calling number in the user part of the Request & To URIs in one of the following formats

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass 
IOP11Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 999Call made from IP-PBX line to the Emergency services using 999. Call answered.
Either party terminates call.
e.g.
Request-Line: INVITE sip:999@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:999@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass 
IOP12Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 112Call made from IP-PBX line to the Emergency services using 112. Call answered,
Either party terminates call.
e.g.
Request-Line: INVITE sip:112@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:112@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass 
IOP13Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 18000 - Text DirectCall made from IP-PBX line using a text direct set to the Emergency services using 18000. Call answered.
Either party terminates call.
e.g.
Request-Line: INVITE sip:18000@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:18000@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>
Pass_With_Caveat 
IOP14IP-PBX Line to PSTN - call answer - Originator disconnectCall made from IP-PBX line to PSTN line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP15IP-PBX Line to PSTN - call answer - Terminator disconnectCall made from IP-PBX line to PSTN line, Answer Call.
PSTN line terminates call
Pass 
IOP16IP-PBX Line to PSTN - Busy subscriberCall made from IP-PBX line to a busy PSTN line (without divert on busy)
Wait for soft switch to return busy response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass 
IOP17IP-PBX Line to PSTN - No answer timeout testCall made from IP-PBX line to a PSTN line (without divert on no answer)
Do not answer call.
Wait for soft switch to return no answer timeout response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass_With_Caveat 
IOP18IP-PBX Line to PSTN - Subscriber not reachableCall made from IP-PBX line to an invalid number.
Wait for soft switch to return response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.
Pass 
IOP19PSTN Line to IP-PBX - call answer - Originator disconnect. Call made from a PSTN line to an IP-PBX line, Answer Call.
Originator disconnects call.
Pass 
IOP20PSTN Line to IP-PBX - call answer - Terminator disconnectCall made from a PSTN line to an IP-PBX line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP21PSTN Line to IP-PBX - busy subscriberCall made from PSTN line to a busy IP-PBX line  (without divert on busy)
Wait for IP-PBX to return busy response.
NoExecSkype server does not support busy line.
IOP22PSTN Line to IP-PBX - No answer timeout test, Invoked by PBXCall made from a PSTN line to an IP-PBX line  (without divert on no answer) Wait for the IP-PBX to return no answer timeout responsePass 
IOP23PSTN Line to IP-PBX - subscriber not reachableCall made from a PSTN line to an invalid number/unprogrammed DDI on the IP-PBX.
Wait for IP-PBX to return response.
Pass 
IOP24Verify CLIP service on IP-PBX line (incoming call from PSTN) Call made from PSTN line to IP-PBX line. PSTN line is set to allow CLI presentation.
Check that CLI is delivered as expected.
Either party terminates call.
Pass 
IOP25Verify CLIR service on IP-PBX line (incoming call from PSTN)Call made from PSTN line to IP-PBX line. PSTN line is set to restrict CLI presentation.
Check that CLI is not delivered as expected.
Either party terminates call.
Pass 
IOP26Verify CLIP service on PSTN line (outgoing call from IP-PBX, From)Ensure number used in From header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends From header containing Calling Line ID (CLI) in the INVITE.

Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates call.
Pass 
IOP27Verify CLIP service on PSTN line (outgoing call from IP-PBX, PAI/PPI)Ensure number used in PAI/PPI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends PAI/PPI header containing Calling Line ID (CLI) in the INVITE.
If PAI header is populated this will be used in preference to the From header.
Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates call.
Pass 
IOP28Verify CLIR service on PSTN line (outgoing call from IP-PBX)Ensure number used in From/PAI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX  line sends From and/or PAI header containing either the Calling Line ID or obscured information in the INVITE.
e.g.
From: "user751000" <sip:+441256751000@192.168.1.10>;tag=12345
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=12345

Ensure that the eSBC restricts presentation of its CLI using privacy-header (Privacy: id or Privacy: user or Privacy: user;id)

Ensure that CLI is NOT presented to the PSTN line.
Either party terminates call.
Pass 
IOP29Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.
Either party terminates call.

Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch?
Pass 
IOP30Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates PSTN)Call made from a PSTN line to an IP-PBX line with call forward to a line in the PSTN, Answer Call.
Either party terminates call.
Pass 
IOP31Verify Call Forward Busy on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with Call Forward Busy (or equivalent) to a line within the IP-PBX, Answer Call.
Either party terminates call.
NoExecSkype server does not support busy line.
IOP32Verify Call Forward No-answer on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)Call made from a PSTN line to an IP-PBX line with Call Forward No-answer (or equivalent) to a line within the IP-PBX, Answer Call.
Either party terminates call.
Pass 
IOP33Verify Call Hold Service on IP-PBX (Incoming call from PSTN)Call made from a PSTN line to an IP-PBX line with Call Hold, Answer call.
IP-PBX line puts the call on hold.
Leave call on hold for 30 seconds and then retrieve call. Ensure speech path is re-established in both directions.
Either party terminates call.
Pass 
IOP34Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party within IP-PBX)Call made from a PSTN line to an IP-PBX line with 3-party conference, Answer call.
IP-PBX line uses the 3-party conference facility to put PSTN line on hold whilst dialling 3rd party. (another IP-PBX line)
Once the 3rd party has answered the call, place the 3 parties in a conference.
Ensure that all parties have a two way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.
Pass 
IOP35Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party PSTN)Call made from a PSTN line to an IP-PBX line with 3-party conference, Answer call.
IP-PBX line uses the 3-party conference facility to put PSTN line on hold whilst dialling 3rd party. (another PSTN line)
Once the 3rd party has answered the call, place the 3 parties in a conference.
Ensure that all parties have a two way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.
Pass 
IOP36Verify do-not-disturb service on IP-PBX line (Incoming call from PSTN)Call made from a PSTN line to an IP-PBX line with do-not-disturb feature active. Ensure IP-PBX line does not ring
PSTN line receives an appropriate announcement or tone

Record the SIP status received from IP-PBX
Pass 
IOP37Verify Call park service on IP-PBX line (Incoming call from PSTN)Call made from a PSTN line to IP-PBX line A with Call Park (or equivalent) feature active, Answer call.
Place the call in the Park condition.
After 10 seconds, retrieve call from IP-PBX line B using the Call Park pick-up code.
Ensure speech path is re-established in both directions.
Either party terminates call.
Pass 
IOP38Verify Call Waiting on an IP-PBX line, involving a PSTN lineCall made from PSTN line A to an IP-PBX line with Call Waiting active, Answer call.
Call made from PSTN line B to the same IP-PBX line which should receive an indication that a second call is waiting.
PSTN line B receives ringback tone.
IP-PBX line answers the call from PSTN line B.
PSTN line A should receive an appropriate indication that they are now on hold.
IP-PBX line toggles the call back to PSTN line A
Ensure speech path is re-established in both directions and that PSTN line B should receive an appropriate indication that they are now on hold.
Either party terminates call.
Pass 
IOP39Verify DTMF transmission from/to IP-PBX - InbandConfigure the IP-PBX/eSBC to send DTMF transmission in-band.

Call made from IP-PBX line to a PSTN line, Answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?
Pass 
IOP40Verify DTMF transmission from/to IP-PBX - RFC 2833 - telephone-event Configure the IP-PBX/eSBC to send DTMF transmission using RFC 2833 - telephone-event.

Call made from IP-PBX line to a PSTN line, Answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?
Pass 
IOP41T.38 Fax transmission mode - PSTN to IP-PBX originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP42T.38 Fax transmission mode - IP-PBX to PSTN originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from IP-PBX line to a PSTN line Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.
FailT.38 fax transmission fails and it fallbacks to G711 fax transmission due to a problem with some pending requests during SIP message exchange after the fax transmssion is detected.
IOP43In-band G.711 Fax transmission mode - PSTN to IP-PBX originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using in-band G.711 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that  in-band G.711 Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP44In-band G.711 Fax transmission mode - IP-PBX to PSTN originationConfigure the ATA/IP-PBX/eSBC such that Fax transmission is sent using  in-band G.711 Fax transmission mode.
Call made from IP-PBX line to a PSTN line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices

Ensure Wireshark trace shows that  in-band G.711  Fax Transmission is used. Check that the fax is transmitted and received as expected.
Pass 
IOP45Test of Call in progress audit function - response to in-call OPTIONS from soft switch to eSBC.Call made from an IP-PBX line to a PSTN line, Answer call.
Leave the  two parties in conversation for 10 minutes.
Ensure both parties have two way speech.
Either party terminates call.

Check wireshark trace to ensure that in-call OPTIONS are sent by the soft switch and that the eSBC responds with status 200OK. Check to see if the eSBC sends any in-call audit SIP messages.
Pass 
IOP46Test of 4 simultaneous calls, 2 inbound, 2 outbound callsConfigure the eSBC such that successive calls route to alternate SBCs (round robin, cyclic etc).
Make 4 simultaneous calls 2 inbound, 2 outbound calls. Answer calls and ensure two way speech path for each call. 
Pass 
IOP47Test of eSBC endpoint restart-recoveryRestart the eSBC and ensure that, after recovery, inbound and outbound calls are successful.Pass 
IOP48Test of eSBC loss of Ethernet link and reconnectionRemove the Ethernet link between the eSBC and CE router. Leave in this condition for at least 3 minutes. Reconnect the Ethernet link and ensure that after approx 2 minutes inbound and outbound calls are successful.Pass 
IOP49Test of Primary SBC loss ** Contact MSL engineer to carry out the following **
On the Primary SBC carry out the ALLSTOP command to disable the SBC.

Call made from IP-PBX line to a PSTN Line.
Call should attempt to route to Primary SBC. On non-response to INVITE, eSBC re-routes the call to the Secondary SBC.
Wait for call answer.
Either party terminates call.

** Contact MSL engineer to carry out the following **
Restart the Primary SBC
Pass 
IOP50Test of eSBC response to UPDATE messages** Contact MSL engineer to carry out the following **
Run UPDATE emulator script ensuring emulator line points to primary SBC

Call made from IP-PBX line to emulator line as provided by MSL engineer.
eSBC should send a packetization time of 10ms
Pass 

 

 

 


Conclusion

This Application Note describes the configuration steps required to enable the Sonus SBC 5XX0 Series to successfully interoperate with Virgin Media SIP Trunk. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in the section Interoperability Test Results.