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Document Overview

This document provides a configuration guide for Sonus Session Border Controller 5XX0Series (SBC) when connecting to Skype for Business 2015 and Exchange Unified Messaging.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flow between Sonus SBC 5200 and Microsoft Skype for Business, using TCP, TLS, and SRTP.

 

Document History

Table : Document History

DateNameComment
30/Aug/2016Arun MuthusamyInitial Draft
23/May/2017Ankit ShuklaRemoving Diversion SMM as PBX sends diversion

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. Navigating the third-party product as well as the Sonus SBC Command Line Interface (CLI) is required. Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and any required troubleshooting.

Requirements

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

Table : Requirements

Product

Equipment

Software Version

Sonus Networks

Sonus SBC 5200
BMC
BIOS
ConnexIP OS
SonusDB
EMA
SBX

V05.01.00-R000
V02.15.00
V02.06.00
V03.01.00-R000
V05.01.00-R000 
V05.01.00-R000 
V05.01.00-R000

Third-party Equipment

Microsoft

Skype For Business 2015 and Exchange Unified Messaging

Reference Configuration

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


Figure : Microsoft Skype For Business Setup

 

Figure : Microsoft Online and Exchange Setup

 

 

Support

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

 

Third-Party Product Features

Kindly refer to the Microsoft's Skype for Business test plan for complete product features details.

 

Prerequisites

Microsoft's Skype For Business enterprise topology should be deployed with at least two Mediation servers..

 

Verify License

Verify you have necessary licences for making enterprise voice call.

 

Configuration  - SBC Configuration

This section provides a “snapshot” of the Sonus SBC 5200 configuration used during compliance testing. The Sonus SBC 5200 is typically configured for customers by Sonus Networks. The screenshots and partial configuration shown below, supplied by Sonus Networks, are provided for reference only. Other configurations are possible. 

1. Global Configuration

1.1 Codec Entry

Create a Codec Entry with the supported codec on the network.

 

set profiles media codecEntry G711_2833_20 dtmf relay rfc2833
set profiles media codecEntry G711_2833_20 packetSize 20
commit
 
set profiles media codecEntry G711SS_2833_20 codec g711ss sendSid enable dtmf relay rfc2833
set profiles media codecEntry G711SS_2833_20 packetSize 20
commit

1.2 RTCP

Configure RTCP interval.

 

set system media mediaRtcpControl senderReportInterval 5
commit

1.3 SIP Domain

Specify the global SIP Domain name.

 

set global sipDomain vm.testnetwork.com
set global sipDomain access.testnetwork.com
set global sipDomain vm.interopdomain.com
set global sipDomain med01.testnetwork.com
set global sipDomain med02.testnetwork.com
commit

1.4 Feature Control Profile (FCP)

Create a Feature Control Profile (FCP) for the Skype side. The FCP will be specified within the SIP Trunk Group Configuration.

 

set profiles featureControlProfile SKYPE_FCP 
commit

1.5 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.

Subsequent configuration sections (Packet service profiles) do 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

 

1.6 LRBT Profile

Create LRBT profile that will be attached to the Skype side. Enable Dynamic LRBT.

 

set profiles media toneAndAnnouncementProfile LRBT_PROF
set profiles media toneAndAnnouncementProfile LRBT_PROF localRingBackTone signalingTonePackageState enable makeInbandToneAvailable enable
set profiles media toneAndAnnouncementProfile LRBT_PROF localRingBackTone flags useThisLrbtForIngress enable
set profiles media toneAndAnnouncementProfile LRBT_PROF localRingBackTone flags dynamicLRBT enable
commit

 

1.7 Path Check Profile

Create Path Check profile that will be attached to the Skype side.

 

set profiles services pathCheckProfile SKYPE_OPTIONS protocol sipOptions sendInterval 20 replyTimeoutCount 1 recoveryCount 1
commit
set profiles services pathCheckProfile SKYPE_OPTIONS transportPreference preference1 tcp
commit
 
Change the transport preference to TCP-TLS if SKYPE is listening on TLS.

 

2. SKYPE and Exchange On Premise Configuration

 

2.1 Packet Service Profile (PSP)

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

 

set profiles media packetServiceProfile SKYPE_PSP
set profiles media packetServiceProfile SKYPE_PSP codec codecEntry1 G711_2833_20
set profiles media packetServiceProfile SKYPE_PSP rtcpOptions rtcp enable
set profiles media packetServiceProfile SKYPE_PSP preferredRtpPayloadTypeForDtmfRelay 101
set profiles media packetServiceProfile SKYPE_PSP flags ssrcRandomize enable
set profiles media packetServiceProfile SKYPE_PSP silenceInsertionDescriptor g711SidRtpPayloadType 13 heartbeat enable
commit

2.2 IP Signaling Profile (IPSP)

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

 

set profiles signaling ipSignalingProfile SKYPE_IPSP
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags includeReasonHeader enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags sendPtimeInSdp enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes transport type1 tcp
set profiles signaling ipSignalingProfile SKYPE_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes redirect flags forceRequeryForRedirection enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags routeUsingRecvdFqdn enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes sipHeadersAndParameters callForwarding dataMapping none
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes sipHeadersAndParameters callForwarding diversionHeaderTransparency enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes transparencyFlags mwiBody enable
set profiles signaling ipSignalingProfile SKYPE_IPSP commonIpAttributes optionTagInRequireHeader suppressReplaceTag enable
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes numberGlobalizationProfile DEFAULT_IP
commit

 

2.3 IP Interface Group

The configuration below is for a Sonus 52x0 system using a single port for Internal connectivity.

 

set addressContext default ipInterfaceGroup LIF2 ipInterface PKT1_V4 ceName IOTNEXUS portName pkt1 ipAddress 172.16.103.184 prefix 24 altIpAddress fc00::103:f:f:f:118 altPrefix 64
set addressContext default ipInterfaceGroup LIF2 ipInterface PKT1_V4 mode inService state enabled
commit

 

2.4 Zone

This Zone groups the set of objects that are used for the communication to Skype for Business. Configure the domain name.

 

set addressContext default zone SKYPE_ZONE id 4
set addressContext default zone SKYPE_ZONE domainName vm.testnetwork.com
commit

2.5 SIP Signaling Port

A SIP Signaling port is a logical address permanently bound to a specific zone and is used to send and receive SIP call signaling packets.

 

set addressContext default zone SKYPE_ZONE id 4 sipSigPort 4 ipInterfaceGroupName LIF2 ipAddressV4 172.16.103.184 portNumber 5060 ipAddressV6 fc00::103:f:f:f:118 transportProtocolsAllowed sip-tcp,sip-udp,sip-tls-tcp
set addressContext default zone SKYPE_ZONE id 4 sipSigPort 4 state enabled mode inService
commit

 

2.6 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 mgmt server DNS1 ipAddress 172.16.101.165 state enabled
set addressContext default zone SKYPE_ZONE dnsGroup EXT_DNS
commit

2.7 SIP Trunk Group

Create a SIP Trunk Group towards Skype side and assign the Profiles configured above.

 

set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG media mediaIpInterfaceGroupName LIF2
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG signaling honorMaddrParam enabled
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG policy media packetServiceProfile SKYPE_PSP
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG policy signaling ipSignalingProfile SKYPE_IPSP
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG downstreamForkingSupport enabled
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG signaling rel100Support enabled
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG services dnsSupportType a-only
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG media earlyMedia forkingBehaviour firstRtp
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG ingressIpPrefix 172.16.101.0 24
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG ingressIpPrefix :: 0
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG signaling relayNonInviteRequest enabled
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG signaling methods notify allow
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG signaling acceptHistoryInfo enabled
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG policy media toneAndAnnouncementProfile LRBT_PROF
set addressContext default zone SKYPE_ZONE sipTrunkGroup SKYPE_TG mode inService state enabled
commit

2.8 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 172.16.101.0 24 172.16.103.1 LIF2 PKT1_V4 preference 100
set addressContext default staticRoute :: 0 fc00::103:f:f:f:1 LIF2 PKT1_V4 preference 100
commit

 2.9 IP Peer

Create an IP Peer with the Fully-Qualified Domain Name (FQDN) of the end points and assign it to the SP or SKYPE Zone. Assign the path check profile created.

 

set addressContext default zone SKYPE_ZONE ipPeer Exchange_IPP policy sip fqdn exchange.testnetwork.com fqdnPort 5060
set addressContext default zone SKYPE_ZONE ipPeer SKYPE_IPP policy sip fqdn med.testnetwork.com fqdnPort 5068
set addressContext default zone SKYPE_ZONE ipPeer SKYPE_IPP pathCheck profile SKYPE_PATHCHECK 
set addressContext default zone SKYPE_ZONE ipPeer SKYPE_IPP pathCheck profile SKYPE_PATHCHECK hostName med.testnetwork.com hostPort 5068 state enabled
commit

 

3. Service Provider Side Configuration

3.1 Packet Service Profile (PSP)

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

 

set profiles media packetServiceProfile ACCESS_PSP
set profiles media packetServiceProfile ACCESS_PSP codec codecEntry1 G711_2833_20
set profiles media packetServiceProfile ACCESS_PSP rtcpOptions rtcp enable terminationForPassthrough enable
set profiles media packetServiceProfile ACCESS_PSP preferredRtpPayloadTypeForDtmfRelay 101
set profiles media packetServiceProfile ACCESS_PSP silenceInsertionDescriptor g711SidRtpPayloadType 13 heartbeat enable
commit

3.2 IP Signaling Profile (IPSP)

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

 

set profiles signaling ipSignalingProfile ACCESS_IPSP
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeReasonHeader enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags sendPtimeInSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes optionTagInRequireHeader suppressReplaceTag enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags routeUsingRecvdFqdn enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes domainName useZoneLevelDomainNameInContact enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes transport type1 tcp
set profiles signaling ipSignalingProfile ACCESS_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes relayFlags statusCode4xx6xx enable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes transparencyFlags mwiBody enable
set profiles signaling ipSignalingProfile ACCESS_IPSP egressIpAttributes redirect flags forceRequeryForRedirection disable
set profiles signaling ipSignalingProfile ACCESS_IPSP commonIpAttributes flags includeTransportTypeInContactHeader enable
commit

 

3.3 IP Interface Group

The below configuration is for a Sonus 52x0 system using a single port for Internal connectivity.

 

set addressContext default ipInterfaceGroup LIF1 ipInterface PKT0_V4 ceName IOTNEXUS portName pkt0 ipAddress 172.16.102.184 prefix 24
set addressContext default ipInterfaceGroup LIF1 ipInterface PKT0_V4 mode inService state enabled
commit

3.4 Zone

This Zone groups the set of objects that are used for the communication to Skype for Business. Configure the domain name and assign DNS server to the zone.

 

set addressContext default zone ACCESS_ZONE id 2
set addressContext default zone ACCESS_ZONE domainName access.testnetwork.com
commit

3.5 SIP Signaling Port

A SIP Signaling port is a logical address permanently bound to a specific zone which is used to send and receive SIP call signaling packets.

 

set addressContext default zone ACCESS_ZONE id 2 sipSigPort 1 ipInterfaceGroupName LIF1 ipAddressV4 172.16.102.184 portNumber 5060 transportProtocolsAllowed sip-tcp,sip-udp,sip-tls-tcp
set addressContext default zone ACCESS_ZONE id 2 sipSigPort 1 mode inService state enabled
commit

3.6 SIP Trunk Group

Create a SIP Trunk Group towards SP side and assign the Profiles configured above.

 

set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG media mediaIpInterfaceGroupName LIF1
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG policy media packetServiceProfile ACCESS_PSP
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG policy signaling ipSignalingProfile ACCESS_IPSP
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG downstreamForkingSupport enabled
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG signaling rel100Support enabled
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG services dnsSupportType a-only
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG ingressIpPrefix 172.16.100.0 24
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG ingressIpPrefix 172.16.105.0 24
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG ingressIpPrefix 172.16.104.0 24
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG mode inService state enabled
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG signaling honorMaddrParam enabled
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG signaling relayNonInviteRequest enabled
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG signaling methods notify allow
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG policy media toneAndAnnouncementProfile LRBT_PROF
commit

 3.7 IP Peer

Create an IP Peer with the Fully-Qualified Domain Name (FQDN) of the end points and assign it to the SP or SKYPE Zone. Assign the path check profile created.

 

set addressContext default zone ACCESS_ZONE ipPeer PhonerLite_IPP ipAddress 172.16.100.56 ipPort 5060
set addressContext default zone ACCESS_ZONE ipPeer POLYCOM1_IPP ipAddress 172.16.105.99 ipPort 5060
set addressContext default zone ACCESS_ZONE ipPeer POLYCOM2_IPP ipAddress 172.16.105.105 ipPort 5060
set addressContext default zone ACCESS_ZONE ipPeer CUCM_IPP ipAddress 172.16.104.178 ipPort 5060
commit

3.8 IP Static Route

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

 

set addressContext default staticRoute 172.16.100.0 24 172.16.102.1 LIF1 PKT0_V4 preference 100
set addressContext default staticRoute 172.16.104.0 24 172.16.102.1 LIF1 PKT0_V4 preference 100
set addressContext default staticRoute 172.16.105.0 24 172.16.102.1 LIF1 PKT0_V4 preference 100
commit

 

3.9 Routing Label

Create a Routing Label with a single Routing Label Route to bind the SP or SKYPE Trunk Group with the SP or SKYPE IP Peer.

 

set global callRouting routingLabel SKYPE_RL routingLabelRoute 1 trunkGroup SKYPE_TG ipPeer SKYPE_IPP inService inService
set global callRouting routingLabel Exchange_RL routingLabelRoute 1 trunkGroup SKYPE_TG ipPeer Exchange_IPP inService inService
set global callRouting routingLabel PhonerLite_RL routingLabelRoute 1 trunkGroup ACCESS_TG ipPeer PhonerLite_IPP inService inService
set global callRouting routingLabel POLYCOM1_RL routingLabelRoute 1 trunkGroup ACCESS_TG ipPeer POLYCOM1_IPP inService inService
set global callRouting routingLabel POLYCOM2_RL routingLabelRoute 1 trunkGroup ACCESS_TG ipPeer POLYCOM2_IPP inService inService
set global callRouting routingLabel CUCM_RL routingLabelRoute 1 trunkGroup ACCESS_TG ipPeer CUCM_IPP inService inService
commit

3.10 Routing

Routing must be put in place to send calls to the correct destination. For the purpose of this scenario, we have used number based routing, but additional routing options may be used.

The configuration of both standard and username routes are done to ensure that no matter which way the called party is addressed (a number or username) the SBC will route 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 none Sonus_NULL Sonus_NULL standard 7778883000 1 all all ALL none Sonus_NULL routingLabel Exchange_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 77788830 1 all all ALL none Sonus_NULL routingLabel SKYPE_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 9620428030 1 all all ALL none Sonus_NULL routingLabel PhonerLite_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 9620428031 1 all all ALL none Sonus_NULL routingLabel POLYCOM1_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 9620428032 1 all all ALL none Sonus_NULL routingLabel POLYCOM2_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 8030 1 all all ALL none Sonus_NULL routingLabel PhonerLite_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 8031 1 all all ALL none Sonus_NULL routingLabel POLYCOM1_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 8032 1 all all ALL none Sonus_NULL routingLabel POLYCOM2_RL
set global callRouting route none Sonus_NULL Sonus_NULL standard 666 1 all all ALL none Sonus_NULL routingLabel CUCM_RL
set global callRouting route none Sonus_NULL Sonus_NULL username Sonus_NULL Sonus_NULL all all ALL none med01.testnetwork.com routingLabel SKYPE_RL
set global callRouting route none Sonus_NULL Sonus_NULL username Sonus_NULL Sonus_NULL all all ALL none med02.testnetwork.com routingLabel SKYPE_RL
commit

3.11 SIP Message manipulation

Create SIP Adapter profile to remove the transport protocol in the incoming SIP response and attach to SP side TG.

 

set profiles signaling sipAdaptorProfile DELETE_TRANSPORT rule 1 criterion 1 type message message messageTypes all condition exist
set profiles signaling sipAdaptorProfile DELETE_TRANSPORT rule 1 criterion 2 type header header name Contact condition exist
set profiles signaling sipAdaptorProfile DELETE_TRANSPORT rule 1 criterion 3 type parameter parameter condition exist paramType uri name transport
set profiles signaling sipAdaptorProfile DELETE_TRANSPORT rule 1 action 1 type parameter operation delete paramType uri from type parameter value transport
set profiles signaling sipAdaptorProfile DELETE_TRANSPORT rule 1 action 1 type parameter operation delete paramType uri to type parameter value transport
set profiles signaling sipAdaptorProfile DELETE_TRANSPORT state enabled
commit
set addressContext default zone ACCESS_ZONE sipTrunkGroup ACCESS_TG signaling messageManipulation inputAdapterProfile DELETE_TRANSPORT
commit

 

4. Online Exchange (O365) side Configuration

4.1 Packet Service Profile (PSP)

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

 

set profiles media packetServiceProfile OFFICE_PSP
set profiles media packetServiceProfile OFFICE_PSP codec codecEntry1 G711-default
set profiles media packetServiceProfile OFFICE_PSP rtcpOptions rtcp enable
set profiles media packetServiceProfile OFFICE_PSP preferredRtpPayloadTypeForDtmfRelay 101
set profiles media packetServiceProfile OFFICE_PSP silenceInsertionDescriptor g711SidRtpPayloadType 13 heartbeat enable
commit

 

4.2 IP Signaling Profile (IPSP)

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

 

set profiles signaling ipSignalingProfile OFFICE_IPSP
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes flags includeReasonHeader enable
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes flags sendPtimeInSdp enable
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes flags sendRtcpPortInSdp enable
set profiles signaling ipSignalingProfile OFFICE_IPSP egressIpAttributes flags disable2806Compliance enable
set profiles signaling ipSignalingProfile OFFICE_IPSP egressIpAttributes transport type1 tlsOverTcp
set profiles signaling ipSignalingProfile OFFICE_IPSP ingressIpAttributes flags sendSdpIn200OkIf18xReliable enable
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes relayFlags notify enable
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes transparencyFlags mwiBody enable
set profiles signaling ipSignalingProfile OFFICE_IPSP egressIpAttributes redirect flags forceRequeryForRedirection enable
set profiles signaling ipSignalingProfile OFFICE_IPSP commonIpAttributes flags routeUsingRecvdFqdn enable
commit

4.3 IP Interface Group

The below configuration is for a Sonus 52x0 system using a single port for Internal connectivity.

 

set addressContext default ipInterfaceGroup LIF2 ipInterface PKT1_V4 ceName IOTNEXUS portName pkt1 ipAddress 182.74.182.205 prefix 24
set addressContext default ipInterfaceGroup LIF2 ipInterface PKT1_V4 mode inService state enabled
commit

 

4.4 Zone

This Zone groups the set of objects that are used for the communication to Skype for Business. Configure the domain name and assign DNS server to the zone.

 

set addressContext default zone OFFICE_ZONE id 3
set addressContext default zone OFFICE_ZONE domainName vm.interopdomain.com
commit

4.5 SIP Signaling Port

A SIP Signaling port is a logical address permanently bound to a specific zone which is used to send and receive SIP call signaling packets.

 

set addressContext default zone OFFICE_ZONE id 2 sipSigPort 2 ipInterfaceGroupName LIF2 ipAddressV4 182.74.182.205 portNumber 5060 transportProtocolsAllowed sip-tls-tcp
set addressContext default zone OFFICE_ZONE id 2 sipSigPort 2 state enabled mode inService
commit

4.6 SIP Trunk Group

Create a SIP Trunk Group towards SP side and assign the Profiles configured above.

 

set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG media mediaIpInterfaceGroupName LIF2 
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG signaling honorMaddrParam enabled
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG policy media packetServiceProfile OFFICE_PSP
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG policy signaling ipSignalingProfile OFFICE_IPSP
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG downstreamForkingSupport enabled
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG signaling rel100Support enabled
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG services dnsSupportType a-only
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG ingressIpPrefix 0.0.0.0 0
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG mode inService state enabled
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG signaling relayNonInviteRequest enabled
set addressContext default zone OFFICE_ZONE sipTrunkGroup OFFICE_TG signaling methods notify allow
commit

 4.7 IP Peer

Create an IP Peer with the Fully-Qualified Domain Name (FQDN) of the end points and assign it to the SP or SKYPE Zone. Assign the path check profile created.

 

set addressContext default zone OFFICE_ZONE ipPeer OFFICE_IPP policy sip fqdn 8bd26852-6bec-4491-8527-29xx61dxxxx3.um.outlook.com fqdnPort 5060
commit

4.8 IP Static Route

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

 

set addressContext default staticRoute 207.46.58.250 32 182.74.182.193 LIF2 PKT1_V4 preference 100
set addressContext default staticRoute 8.8.8.8 32 182.74.182.193 LIF2 PKT1_V4 preference 100
set addressContext default staticRoute 0.0.0.0 0 182.74.182.193 LIF2 PKT1_V4 preference 100
commit

4.9 DNS Group

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

 

set addressContext default zone OFFICE_ZONE dnsGroup PUBLIC_DNS
set addressContext default dnsGroup PUBLIC_DNS type ip interface LIF2 server PUBLIC_DNS state enabled ipAddress 8.8.8.8
set addressContext default zone OFFICE_ZONE dnsGroup PUBLIC_DNS
commit

4.10 Routing Label

Create a Routing Label with a single Routing Label Route to bind the SP or SKYPE Trunk Group with the SP or SKYPE IP Peer.

 

set global callRouting routingLabel OFFICE_RL routingLabelRoute 1 trunkGroup OFFICE_TG ipPeer OFFICE_IPP inService inService
commit

4.11 Routing

Routing must be put in place to send calls to the correct destination. For the purpose of this scenario, we have used number base routing, but additional routing options may be used.

The configuration of both standard and username routes are done to ensure that no matter which way the called party is addressed (a number or username) the SBC will route 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 none Sonus_NULL Sonus_NULL standard 8888884 1 all all ALL none Sonus_NULL routingLabel OFFICE_RL
commit

 

5. TLS Configuration for Skype and Exchange On-Premise

 

Note: Only difference from previous Section is shown below
Important Note
    
SBX5K does not support MKI. SKYPE_IT tool does not take into account that SBX has not published MKI support in its SDP and still tries to validate SRTP as
SRTP with MKI BIT set. As a workaround, we publish MKI support in SDP and use this new debug xrm command to mark MKI bit in outgoing SRTP/SRTCP  streams and also factor it for incoming SRTP/ SRTCP streams.
 
This command is to be used only for SKYPE certification or qualification in Customer Labs only. We do not recommend enabling this in production enviroment.
 
   
admin@pumal%
unhide debug
Password: ******
#password is sonus1
 
admin@puma%
request sbx xrm debug command "srtpmki enable"
[ok][2014-04-01 16:54:17]
[edit]
MKI Enabled: encLength=1; encValue=0x1; decLength=1
admin@puma%

5.1 Create a configuration object to hold a locally generated RSA key pair

 

 

set system security pki certificate SBC_CERT type local-internal
commit

5.2 Generate Key pair and CSR (certificate signing request) for submission to a Certificate Authority (CA)

 

request system security pki certificate SBC_CERT generateCSR csrSub /C=IN/ST=KA/L=Bangalore/O=Sonus/CN=vm.testnetwork.com keySize keySize2K

5.3 Generate required certificates

 

Note: Follow certification generation procedure given in Appendix A and then copy the SKYPE Server Root Certificate (rootcert.cer) and Microsoft signed SBC Certificate (servercert.pem) into
/opt/sonus/external/ folder of SBC

5.4 Create Crypto Suite Profile

 

set profiles security cryptoSuiteProfile CRYPT_PROF entry 1 cryptoSuite AES-CM-128-HMAC-SHA1-80
commit

5.5 Import SKYPE Root Certificate into database


set system security pki certificate ROOT_CERT type remote fileName Root_CERT.cer state enabled
commit

5.6 Import CA Certified SBC Server Certificate into database


set system security pki certificate SBC_CERT fileName servercert.pem state enabled
commit

5.7 Create TLS Profile


set profiles security tlsProfile TLS_PROF clientCertName SBC_CERT serverCertName SBC_CERT cipherSuite1 rsa-with-3des-ede-cbc-sha cipherSuite2 rsa-with-aes-128-cbc-sha authClient true allowedRoles clientandserver acceptableCertValidationErrors invalidPurpose
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5.8 Configure Packet Service Profile with Crypto Suite


set profiles media packetServiceProfile SKYPE_PSP secureRtpRtcp cryptoSuiteProfile CRYPT_PROF
set profiles media packetServiceProfile SKYPE_PSP secureRtpRtcp flags enableSrtp enable
set profiles media packetServiceProfile SKYPE_PSP secureRtpRtcp flags allowFallback disable
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5.9 Configure IP Signaling Profile


set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes transport type1 tlsOverTcp
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5.10 Attach TLS Profile to SIP Signaling Port


set addressContext default zone SKYPE_ZONE sipSigPort 4 state disabled mode outOfService
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set addressContext default zone SKYPE_ZONE sipSigPort 4 tlsProfileName TLS_PROF
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set addressContext default zone SKYPE_ZONE sipSigPort 4 state enabled mode inService
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6. TLS Configuration for Exchange Online (O365)

 

Note: Only additional config required from previous Section is shown below.
1. New Server certification from public CA needs to be imported.
2. Baltimore certificate in pem formate is found in below site. 
http://certificate.fyicenter.com/319_Root_CA_Baltimore_CyberTrust_Root_CyberTrust_Baltimore_IE.html

 

6.1 Import Baltimore Certificate into database

 

set system security pki certificate MicroSoft_CERT type remote fileName GlobalCert.cer state enable
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6.2 Attach TLS Profile to SIP Signaling Port

 

set addressContext default zone OFFICE_ZONE sipSigPort 2 state disabled mode outOfService
commit
set addressContext default zone OFFICE_ZONE sipSigPort 2 tlsProfileName TLS_PROF
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set addressContext default zone OFFICE_ZONE sipSigPort 2 state enabled mode inService
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Test Results

 

Table : Microsoft SKYPE certification test results

S.NoProcedureObservationResultComment
1408058 [PstnEndPt] calls [SKYPEEndPt] with Caller ID set to 'Anonymous' on [DUT] PASS 
2408059 [PstnEndPt] calls [SKYPEEndPt] with Caller ID set to 'Anonymous' on [DUT]. (Media Bypass OFF) (IPv6) PASS 
3408200 3rd Party Presence headers do not cause [DUT] failure PASS 
4408202 3rd Party Presence headers do not cause [DUT] failure. (Media Bypass OFF) (IPv6) PASS 
5408282 [DUT] accepts [MedSrv] 'pool' certificate for a secure call PASS 
6408064 [SKYPEEndPt] calls IVR number and navigates through the IVR menu before call Connection PASS 
7408106 [SKYPEEndPt] hears Early Media for a call to [PstnEndPt] PASS 
8408112 [DUT] sends PRACK for reliable Early Media for a call from [PstnEndPt] to [SKYPEEndPt] PASS 
9408113 [DUT] sends PRACK for reliable Early Media for call from [PstnEndPt] to [SKYPEEndPt] with SRTP Optional PASS 
10408079 [PstnEndPt]1 calls [SKYPEEndPt] that is set to simultaneous ring to [SKYPEEndPt] and [PstnEndPt]2 answers PASS 
11408109 [DUT] disconnects a forked call if [PstnEndPt] hangs up while phones are ringing PASS 
12408144 [DUT] is able to disconnect a call that isforkedto [SKYPEEndPt]s set to 'Do not disturb'. (Media Bypass OFF) (IPv6) PASS 
13408159 [DUT] disconnects a forked call if [PstnEndPt] hangs up while phones are ringing. (Media Bypass OFF) (IPv6) PASS 
14408062 [SKYPEEndPt] calls [PstnEndPt] with a call duration longer than 32 seconds PASS 
15408063 [DUT] generates [603] response for a call rejected by [PstnEndPt] PASS 
16408065 [DUT] correctly handles non-E.164 number in outbound Request URI PASS 
17408066 [SKYPEEndPt] calls [PstnEndPt] and hangs up before receiving [200] from [DUT] PASS 
18408067 [PstnEndPt] displays [SKYPEEndPt] Caller ID for Outbound Call PASS 
19408068 [PstnEndPt] disconnects established call to [SKYPEEndPt] PASS 
20408069 [PstnEndPt] disconnects established call from [SKYPEEndPt] PASS 
21408071 [DUT] processes phone-context in Request and To URI from [SKYPEEndPt] PASS 
22408072 [SKYPEEndPt] sends INVITE with E.164 number and extension in Request and To URI PASS 
23408073 [DUT] processes call from [SKYPEEndPt] with E.164 number in FROM Header URI PASS 
24408074 [SKYPEEndPt] response to [PstnEndPt] is delayed due to network delay PASS 
25408077 [SKYPEEndPt] calls an IVR number and navigates through the IVR menu after call connection. PASS 
26408078 [DUT] handles call from [MedSrv] with an alias name in the FROM header PASS 
27408081 [MedSrv] renegotiates an existing voice session with a different IP address PASS 
28408085 [DUT] establishes call to [SKYPEEndpt] with configured value of ptime PASS 
29408131 [SKYPEEndPt] calls [PstnEndPt] with a call duration longer than 32 seconds. (Media Bypass OFF) (IPv6) PASS 
30408132 [DUT] generates [603] response for a call rejected by [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
31408133 [SKYPEEndPt] calls IVR number and navigates through the IVR menu before call Connection. (Media Bypass OFF) (IPv6) PASS 
32408134 [DUT] correctly handles non-E.164 number in outboundRequest URI. (Media Bypass OFF) (IPv6) PASS 
33408135 [SKYPEEndPt] calls [PstnEndPt] and hangs up before receiving [200] from [DUT]. (Media Bypass OFF) (IPv6) PASS 
34408136 [PstnEndPt] displays [SKYPEEndPt] Caller ID for Outbound Call. (Media Bypass OFF) (IPv6) PASS 
35408138 [PstnEndPt] disconnects established call from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
36408140 [DUT] processes phone-context in Request and To URI from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
37408141 [SKYPEEndPt] sends INVITE with E.164 number and extension in Request and To URI. (Media Bypass OFF) (IPv6) PASS 
38408142 [DUT] processes call from [SKYPEEndPt] withE.164number in FROM Header URI. (Media Bypass OFF) (IPv6) PASS 
39408143 [SKYPEEndPt] response to [PstnEndPt] is delayed due to network delay. (Media Bypass OFF) (IPv6) PASS 
40408146 [SKYPEEndPt] calls an IVR number and navigates through the IVR menu after call connection. (Media Bypass OFF) (IPv6) PASS 
41408147 [DUT] handlescallfrom [MedSrv] with an alias name in the FROM header. (Media Bypass OFF) (IPv6) PASS 
42408086 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs PASS 
43408090 [PstnEndPt] is able to establish a call with [SKYPEEndPt] using G.711 A-law codec PASS 
44408092 [SKYPEEndPt] is able to establish a call with [PstnEndPt] using G.711 A-law codec PASS 
45408101 [DUT] offers DTMF payload type in the range of 96-127 to [MedSrv] PASS 
46408114 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 U-law codec PASS 
47408119 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 U-law codecs PASS 
48408150 [MedSrv] renegotiates an existing voice session with a different IP address. (Media Bypass OFF) (IPv6) PASS 
49408152 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs. (Media Bypass OFF) (IPv6) PASS 
50408167 [DUT] offers DTMF payload type in the range of 96-127 to [MedSrv]. (Media Bypass OFF) (IPv6) PASS 
51408174 [SKYPEEndPt] is able to establish a call with [PstnEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) PASS 
52408180 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 U-law codec. (Media Bypass OFF) (IPv6) PASS 
53408181 [PstnEndPt] is able to establish a call with [SKYPEEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) PASS 
54408183 [DUT] negotiates Comfort Noise in a call from [SKYPEEndPt] to [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
55408186 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs. (Media Bypass OFF) (IPv6) PASS 
56408187 [SKYPEEndPt] receives a secure call with G.711 U-law codec. (Media Bypass OFF) (IPv6) PASS 
57408093 [DUT] handles multiple RTP streams for a call to [SKYPEEndPt] PASS 
58408123 [DUT] does not change the SSRC of an established outbound SRTP session PASS 
59408126 [DUT] does not change the SSRC of an established inbound SRTP session PASS 
60408190 [DUT] does not change the SSRC of an established inbound SRTP session. (Media Bypass OFF) (IPv6) PASS 
61408193 [DUT] does not change the SSRC of an established outbound SRTP session. (Media Bypass OFF) (IPv6) PASS 
62408091 [PstnEndPt] is able to establish a secure call with [SKYPEEndPt] using G.711 A-law codec PASS 
63408094 [DUT] handles multiple SRTP streams for a secure call to [SKYPEEndPt] PASS 
64408107 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt] PASS 
65408108 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt] when Media Bypass OFF PASS 
66408110 [DUT] disconnects a forked secure call if [PstnEndPt] hangs up while phones are ringing PASS 
67408120 [SKYPEEndPt] receives a secure call with G.711 U-law codec with Media Bypass OFF PASS 
68408122 [SKYPEEndPt] makes a secure call to [PstnEndPt] and [PstnEndPt] later hangs up PASS 
69408129 [SKYPEEndPt] makes a secure call to [PstnEndPt] PASS 
70408130 [SKYPEEndPt] makes a secure call to [PstnEndPt] with call duration more than 32 seconds and SRTP set to Optional PASS 
71408160 [DUT] disconnects a forked secure call if [PstnEndPt] hangs up while phones are ringing. (Media Bypass OFF) (IPv6) PASS 
72408173 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
73408182 [PstnEndPt] is able to establish a secure call with [SKYPEEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) PASS 
74408195 [SKYPEEndPt] makes a secure call to [PstnEndPt] with call duration more than 32 seconds and SRTP set to Optional. (Media Bypass OFF) (IPv6) PASS 
75408196 [SKYPEEndPt] makes a secure call to [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
76408070 [PstnEndPt] calls [SKYPEEndPt] and hangs up while [SKYPEEndPt] is still ringing PASS 
77408080 Inboundcallto [SKYPEEndPt] from [PstnEndPt] with a very long Request-URI in the INVITE PASS 
78408117 [DUT] handles [488] response from the [MedSrv] operating in RTP only mode PASS 
79408118 [DUT] sends its own FQDN in contact header for TLS call from [SKYPEEndPt] to [PstnEndPt] PASS 
80408124 [DUT] sends Crypto attributes in SDP for call from [PstnEndPt] to [SKYPEEndPt] PASS 
81408125 [PstnEndPt] calls [SKYPEEndPt] with security enabled and [SKYPEEndPt] later hangs up PASS 
82408127 [DUT] adds at least one 'crypto' attribute for each media description line in the SDP PASS 
83408137 [PstnEndPt] disconnects established call to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
84408139 [PstnEndPt] calls [SKYPEEndPt] and hangs up while [SKYPEEndPt] is still ringing. (Media Bypass OFF) (IPv6) PASS 
85408145 [DUT] sends single media description line for a call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
86408149 Inboundcallto [SKYPEEndPt] from [PstnEndPt] with a very long Request-URI in the INVITE. (Media Bypass OFF) (IPv6) PASS 
87408151 [DUT] establishescallto [SKYPEEndpt] with configured value ofptime. (Media Bypass OFF) (IPv6) PASS 
88408170 [DUT] with RTP only setting rejectscallfrom [SKYPEEndPt] that requires SRTP. (Media Bypass OFF) (IPv6) PASS 
89408172 [SKYPEEndPt] hears Early Media for a call to [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
90408188 [DUT] sends Crypto attributes in SDP for call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
91408189 [PstnEndPt] calls [SKYPEEndPt] with security enabled and [SKYPEEndPt] later hangs up. (Media Bypass OFF) (IPv6) PASS 
92408191 [DUT] adds at least one 'crypto' attribute for each media description line in the SDP. (Media Bypass OFF) (IPv6) PASS 
93408207 [PstnEndPt]1 calls [SKYPEEndPt] that forwards all calls to [PstnEndPt]2 when Media Bypass OFF PASS 
94408209 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] that forwards the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
95408210 [PstnEndPt]1 calls [SKYPEEndPt] that forwards all calls to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
96408227 [SKYPEEndPt] resumes call to [PstnEndPt] after playing music on hold for 15 minutes PASS 
97408229 [SKYPEEndPt] places a call to [PstnEndPt] on hold and resumes after 12 minutes PASS 
98408234 [SKYPEEndPt] places a call from [PstnEndPt] on hold for 15 minutes and then resumes PASS 
99408242 [DUT] disconnects a call that is on hold when [SKYPEEndPt] hangs up. (Media Bypass OFF) (IPv6) PASS 
100408243 [DUT] disconnects a call that is on hold when [PstnEndPt] hangs up. (Media Bypass OFF) (IPv6) PASS 
101408248 [SKYPEEndPt] places a call from [PstnEndPt] on hold for 15 minutes and then resumes. (Media Bypass OFF) (IPv6) PASS 
102408250 [SKYPEEndPt] places a call to [PstnEndPt] on hold and resumes after 12 minutes. (Media Bypass OFF) (IPv6) PASS 
103408224 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and resumes after 15 minutes PASS 
104408225 [PstnEndPt] places a call with Media Bypass OFF from [SKYPEEndPt] on hold for 15 minutes and then resumes PASS 
105408226 [PstnEndPt] puts [SKYPEEndPt] on hold and resumes after 15 minutes for a secure call PASS 
106408235 [PstnEndPt] places a secure call from [SKYPEEndPt] on hold and then resumes PASS 
107408236 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and then resumes PASS 
108408237 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and resumes after 15 minutes. (Media Bypass OFF) (IPv6) PASS 
109408238 [PstnEndPt] places a call from [SKYPEEndPt] on hold for 15 minutes and then resumes. (Media Bypass OFF) (IPv6) PASS 
110408239 [PstnEndPt] puts [SKYPEEndPt] on hold and resumes after 15 minutes for a secure call. (Media Bypass OFF) (IPv6) PASS 
111408240 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and then resumes. (Media Bypass OFF) (IPv6) PASS 
112408244 [PstnEndPt] places a secure call from [SKYPEEndPt] on hold and then resumes. (Media Bypass OFF) (IPv6) PASS 
113408231 [SKYPEEndPt] plays music when it holds call from [PstnEndPt] to [SKYPEEndPt] PASS 
114408241 [SKYPEEndPt] resumes call to [PstnEndPt] after playing music on hold for 15 minutes. (Media Bypass OFF) (IPv6) PASS 
115408245 [SKYPEEndPt] plays music when it holds call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
116408261 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2 PASS 
117408262 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2 PASS 
118408274 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
119408275 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
120408276 [DUT] does not drop the call when Consultative Transfer by [SKYPEEndPt] to second [PstnEndPt] fails. (Media Bypass OFF) (IPv6) PASS 
121408254 [DUT] includes REFER in ALLOW header in INVITE sent to [MedSrv] PASS 
122408255 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2 PASS 
123408256 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2 PASS 
124408266 [DUT] includes REFER in ALLOW header in INVITE sent to [MedSrv]. (Media Bypass OFF) (IPv6) PASS 
125408267 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
126408268 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) PASS 
127408285 [DUT] uses load balancing to distribute inbound calls among [MedSrv]s in a cluster PASS 
128408286 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TCP PASS 
129408288 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TLS PASS 
130408291 [PstnEndPt] establishes a call with [SKYPEEndPt] when interface of [MedSrv]1 goes down PASS 
131408293 [DUT] fails over incoming call to a second [MedSrv] when the first [MedSrv] does not respond PASS 
132408295 [DUT] fails over incoming call to [MedSrv]2 when [MedSrv]1 sends [503] response. (Media Bypass OFF) (IPv6) PASS 
133408296 [DUT] uses load balancing to distribute inbound calls among [MedSrv]s in a cluster. (Media Bypass OFF) (IPv6) PASS 
134408297 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TCP. (Media Bypass OFF) (IPv6) PASS 
135408299 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TLS. (Media Bypass OFF) (IPv6) PASS 
136408304 [DUT] fails over incoming call to a second [MedSrv] when the first [MedSrv] does not respond (Media Bypass OFF) (IPv6) PASS 
137408309 [DUT] distributes new calls among DNS configured [MedSrv]s PASS 
138408310 [DUT] distributes new calls among DNS configured [MedSrv]s. (Media Bypass OFF) (IPv6) PASS 
139408315 [DUT] sends [414] when unable to handle very long Request URI PASS 
140408316 [DUT] times out after 180 seconds of no response from [SKYPEEndPt] following [100] PASS 
141408317 [DUT] processes [488] response for unsupported codec from [MedSrv] PASS 
142408321 [DUT] disconnects call when [MedSrv] sends [408] for call from [PstnEndPt] PASS 
143408322 [DUT] processes [603] from [SKYPEEndPt] for a secure call PASS 
144408323 [DUT] processes [603] response from [SKYPEEndPt] PASS 
145408324 [DUT] handles call from [SKYPEEndPt] to a user that does not exist in the domain PASS 
146408325 [DUT] generates [486] response from a busy [PstnEndPt] PASS 
147408326 [DUT] processes [486] response from a busy [SKYPEEndPt] PASS 
148408327 [DUT] disconnects call when [MedSrv] sends [501] for call from [PstnEndPt] PASS 
149408328 [DUT] disconnects call when [MedSrv] sends [606] for call from [PstnEndPt] PASS 
150408329 [DUT] responds with [488] when [MedSrv] offers a codec unsupported on the device PASS 
151408330 [DUT] sends [414] when unable to handle very long Request-URI. (Media Bypass OFF) (IPv6) PASS 
152408331 [DUT] times out after 180 seconds of no response from [SKYPEEndPt] following [100]. (Media Bypass OFF) (IPv6) PASS 
153408332 [DUT] processes [488] response for unsupported codec from [MedSrv]. (Media Bypass OFF) (IPv6) PASS 
154408336 [DUT] disconnects call when [MedSrv] sends [408] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
155408337 [DUT] processes [603] from [SKYPEEndPt] for a secure call. (Media Bypass OFF) (IPv6) PASS 
156408338 [DUT] processes [603] response from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
157408339 [DUT] processes [486] response from a busy [SKYPEEndPt]. (Media Bypass OFF) (IPv6) PASS 
158408340 [DUT] disconnects call when [MedSrv] sends [501] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
159408341 [DUT] disconnects call when [MedSrv] sends [606] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
160408342 [DUT] responds with [488] when [MedSrv] offers a codec unsupported on the device. (Media Bypass OFF) (IPv6) PASS 
161408343 [DUT] handlescallfrom [SKYPEEndPt] to a user that does not exist in the domain. (Media Bypass OFF) (IPv6) PASS 
162408344 [DUT] generates [486] response from a busy [PstnEndPt]. (Media Bypass OFF) (IPv6) PASS 
163408347 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs PASS 
164408350 [DUT] fails over incoming call to [MedSrv]2 when [MedSrv]1 sends [503] response PASS 
165408351 [PstnEndPt] places a call from [SKYPEEndPt] on hold for 15 minutes and then resumes PASS 
166408223  Device is able to disconnect a call that is on hold when it receives a BYE from PSTN End Point PASS 
167408076 Device is able to disconnect a call that is on hold when it receives a BYE from PSTN End Point PASS 
168408075 Device can disconnect a forked call when all SKYPE End Points are set to 'Do not Disturb'. PASS 
169408082 Call parking functionality for an inbound call. PASS 
170408222 Device is able to disconnect a call that's on hold when it receives a BYE from Mediation Server. PASS 
171408087 Device is able to negotiate Comfort Noise as part of the SDP negotiation in an outbound call from SKYPE End Point PASS 
172408153 PSTN End Point calls SKYPE End Point1, SKYPE End Point1 parks the call and retrieves it on SKYPE End Point2. (Media Bypass OFF) (IPv6) PASS 
173408162 Device sends PRACK for reliable Early Media for a call from PSTN End Point to SKYPE End Point. (Media Bypass OFF) (IPv6) PASS 
174408116 Device that supports SRTP only rejects call from SKYPE End Point that supports RTP Only PASS 
175408099 Device with RTP only setting rejects call from SKYPE End Point that requires SRTP PASS 
176408163 Device sends PRACK for reliable Early Media forcallfrom PSTN End Point to SKYPE End Point with SRTP Optional. (Media Bypass OFF) (IPv6) PASS 
177  408176 Mediation Server that requires SRTP rejectscallfrom Device that supports RTP only. (Media Bypass OFF) (IPv6) PASS 
178408177 Device that supports SRTP only rejectscallfrom SKYPE End Point that supports RTP Only. (Media Bypass OFF) (IPv6) PASS 
179408178 Device handles 488 Not Acceptable Here response from the Mediation Server operating in RTPonlymode. (Media Bypass OFF) (IPv6) PASS 
180408179 Device sends its own FQDN in contact header for TLS call from SKYPE End Point to PSTN End Point. (Media Bypass OFF) (IPv6) PASS 
181408192 SKYPE End Point makes a secure call to PSTN End Point and PSTN End Point later hangs up. (Media Bypass OFF) (IPv6) PASS 
182408148 PSTN End Point1 calls SKYPE End Point that is set to simultaneous ring to SKYPE End Point and PSTN End Point2 answers. (Media Bypass OFF) (IPv6)  PASS 
183408115 Mediation Server that requires SRTP rejects call from Device that supports RTP only  PASS 
     
 

 

Table : Microsoft O365 certification test results

S.NoProcedureObservationResultComment
1465403 12.1.1 Mailbox login from public phone (On-Premises) PASS 
2465404 12.1.2 Mailbox navigation using VUI (On-Premises) PASS 
3465405 12.1.3 Mailbox navigation using TUI (On-Premises) PASS 
4465406 12.1.4 Leave Voicemail from an internal extension (On-Premises) PASS 
5465407 12.1.5 Leave Voicemail from an external extension (On-Premises) PASS 
6465408 12.1.6 Inbound call handled by Auto Attendant (On-Premises) PASS 
7465417 12.2.1 Voicemail using OWA's Play-On-Phone feature to an external extension (On-Premises) PASS 
8465416 12.2.2 Voicemail using OWA's Play-On-Phone feature to a user's extension (On-Premises) PASS 
9465410 12.3.1 Call transferred to search target (On-Premises) PASS 
10465411 12.3.2 Call transferred to search target busy voicemail (On-Premises) PASS 
11465412 12.3.3 Call transferred to search target no-answer voicemail (On-Premises) PASS 
12465413 12.3.4 Call transferred to search default target (On-Premises) PASS 
13465683 12.4.1 Device supports FAX (On-Premises)   
14465792 12.5.1 MWI Lamp on PBX phone lights up (On-Premises) PASS 
15465793 12.5.2 MWI Lamp on PBX phone turns off (On-Premises) PASS 
16465798 12.6.1 Check Voicemail Button (On-Premises) PASS 
17465799 12.6.2 Call Forward toother UM-Enableduser(On-Premises) PASS 
18465693 13.1.1 Mailbox login from public phone (On-Line) PASS 
19465694 13.1.2 Mailbox navigation using VUI (On-Line) PASS 
20465695 13.1.3 Mailbox navigation using TUI (On-Line) PASS 
21465696 13.1.4 Leave Voicemail from an internal extension (On-Line) PASS 
22465697 13.1.5 Leave Voicemail from an external extension (On-Line) PASS 
23465698 13.1.6 Inbound call handled by Auto Attendant (On-Line) PASS 
24465706 13.2.1 Voicemail using OWA's Play-On-Phone feature to an external extension (On-Line) PASS 
25465705 13.2.2 Voicemail using OWA's Play-On-Phone feature to a user's extension (On-Line) PASS 
26465700 13.3.1 Call transferred to search target (On-Line) PASS 
27465701 13.3.2 Call transferred to search target busy voicemail (On-Line) PASS 
28465702 13.3.3 Call transferred to search target no-answer voicemail (On-Line) PASS 
29465703 13.3.4 Call transferred to search default target (On-Line) PASS 
30465708 13.4.1 Device supports FAX (On-Line)   
31465801 13.6.1 Check Voicemail Button (On-Line) PASS 
32465802 13.6.2 Call Forward toother UM-Enableduser(On-Line) PASS 

Conclusion

These Application Notes describe the configuration steps required for Sonus SBC 5XX0 to successfully interoperate with Skype for Business 2015 and Exchange Unified Messaging. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.