This document provides a configuration guide for Sonus Session Border Controller 5XX0Series (SBC) when connecting to Skype for Business 2015 and Exchange Unified Messaging.
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
Document History
Date | Name | Comment |
---|---|---|
23/June/2017 | Ankit Shukla | Initial Draft |
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.
The following equipment and software were used for the sample configuration provided:
Requirements
Product | Equipment | Software Version |
---|---|---|
Sonus Networks | Sonus SBC 5200 | V06.00.00-R000 |
Third-party Equipment | Microsoft | Skype For Business 2015 and Exchange Unified Messaging |
The following reference configuration shows connectivity between the third-party and the Sonus SBC 5XX0.
Microsoft Skype For Business Setup
Microsoft Online and Exchange Setup
For any questions regarding this document or the content herein, contact your maintenance and support provider.
Kindly refer to the Microsoft's Skype for Business test plan for complete product features details.
Microsoft's Skype For Business enterprise topology should be deployed with at least two Mediation servers.
Verify you have necessary licences for making enterprise voice call.
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.
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
Configure RTCP interval.
set system media mediaRtcpControl senderReportInterval 5 commit
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
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
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
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
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 server is listening on TLS.
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
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
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 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
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
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 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
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
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
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
Create an IP Peer with the Fully-Qualified Domain Name (FQDN) of the end points and assign it to the Service Provider (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
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
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
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
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
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
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
Create an IP Peer with the Fully-Qualified Domain Name (FQDN) of the end points and assign it to the SP or SKYPE Zone and 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
Create a default route to the subnet’s next hop IP 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
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
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 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
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
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
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
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
This Zone groups the set of objects that are used for the communication to Skype for Business. Configure the domain name. 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
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 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
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
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. FQDN for OFFICE_IPP will be provided by Microsoft O365.
set addressContext default zone OFFICE_ZONE ipPeer OFFICE_IPP policy sip fqdn 8bd26852-6bec-xxxx-8527-29ee61ddxxxx.um.outlook.com fqdnPort 5060 commit
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
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
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
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
Note: Only difference from previous Section is shown below
Important Note SBX5K does not support MKI. LYNC_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 LYNC 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%
set system security pki certificate SBC_CERT type local-internal commit
request system security pki certificate SBC_CERT generateCSR csrSub /C=IN/ST=KA/L=Bangalore/O=Sonus/CN=vm.testnetwork.com keySize keySize2K
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
set profiles security cryptoSuiteProfile CRYPT_PROF entry 1 cryptoSuite AES-CM-128-HMAC-SHA1-80 commit
set system security pki certificate ROOT_CERT type remote fileName Root_CERT.cer state enabled commit
set system security pki certificate SBC_CERT fileName servercert.pem state enabled commit
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 commit
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 commit
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes transport type1 tlsOverTcp commit
set addressContext default zone SKYPE_ZONE sipSigPort 4 state disabled mode outOfService commit set addressContext default zone SKYPE_ZONE sipSigPort 4 tlsProfileName TLS_PROF commit set addressContext default zone SKYPE_ZONE sipSigPort 4 state enabled mode inService commit
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 format is found in below site. It need to be converted into .cer by using openssl. http://certificate.fyicenter.com/319_Root_CA_Baltimore_CyberTrust_Root_CyberTrust_Baltimore_IE.html openssl x509 -outform der -in cert.pem -out cert.cer
set system security pki certificate MicroSoft_CERT type remote fileName GlobalCert.cer state enable commit
set addressContext default zone OFFICE_ZONE sipSigPort 2 state disabled mode outOfService commit set addressContext default zone OFFICE_ZONE sipSigPort 2 tlsProfileName TLS_PROF commit set addressContext default zone OFFICE_ZONE sipSigPort 2 state enabled mode inService commit
Microsoft Skype For Business certification test results
S .No | Test Case Id | Test Case Description | Result | Observation | Comment |
1 | 408347 | 3.2.1 Skype for Business Client receives a call from PSTN End Point with G.711 A-law and/or G.711 U-law codecs | Pass | ||
2 | 408351 | 3.2.2 PSTN End Point places a call from Skype for Business Client on hold for 15 minutes and then resumes | Pass | ||
3 | 408348 | 3.2.3 PSTN End Point1 calls Skype for Business Client that forwards the call to PSTN End Point2 | Pass | ||
4 | 408352 | 3.2.4 PSTN End Point calls Skype for Business Client1 that performs Blind Transfer to Skype for Business Client2 with REFER | Pass | ||
5 | 408349 | 3.2.5 PSTN End Point1 calls Skype for Business Client that escalates the call to a conference by inviting PSTN End Point2 | Pass | ||
6 | 408350 | 3.2.6 Device fails over incoming call to Mediation Server2 when Mediation Server1 sends 503 Service Unavailable response | Pass | ||
7 | 408083 | 4.2.6 Skype for Business Client1 calls PSTN End Point, Skype for Business Client1 parks the call and retrieves it on Skype for Business Client2 | Pass | ||
8 | 408070 | 4.2.8 PSTN End Point calls Skype for Business Client and hangs up while Skype for Business Client is still ringing | Pass | ||
9 | 408084 | 4.2.9 PSTN End Point calls Skype for Business Client that later parks the call but does not retrieve it | Pass | ||
10 | 408111 | 4.2.14 PSTN End Point1 calls Skype for Business Client that is set to simultaneous ring to IVR number on a PSTN endpoint | Pass | ||
11 | 408080 | 4.2.15 Inbound call to Skype for Business Client from PSTN End Point with a very long Request-URI in the INVITE | Pass | ||
12 | 408065 | 4.3.1 Device correctly handles non-E.164 number in outboundRequest URI | Pass | ||
13 | 408085 | 4.3.2 Device establishes call to Skype for Business Client with configured value of ptime | Pass | ||
14 | 408063 | 4.3.3 Device generates 603 Decline response for a call rejected by PSTN End Point | Pass | ||
15 | 408078 | 4.3.4 Device handles call from Mediation Server with an alias name in the FROM header | Pass | ||
16 | 408075 | 4.3.5 Device is able to disconnect a call that isforkedto Skype for Business Clients set to 'Do not disturb' | Pass | ||
17 | 408087 | 4.3.6 Device negotiates Comfort Noise in a call from Skype for Business Client to PSTN End Point | Pass | ||
18 | 408073 | 4.3.7 Device processes call from Skype for Business Client with E.164 number in FROM Header URI | Pass | ||
19 | 408071 | 4.3.8 Device processes phone-context in Request and To URI from Skype for Business Client | Pass | ||
20 | 408088 | 4.3.9 Device sends Comfort Noise packets to Skype for Business Client when a call is muted | Pass | ||
21 | 408076 | 4.3.11 Device sends single media description line for a call from PSTN End Point to Skype for Business Client | Pass | ||
22 | 408066 | 4.3.14 Skype for Business Client calls PSTN End Point and hangs up before receiving 200 OK from Device | Pass | ||
23 | 408062 | 4.3.15 Skype for Business Client calls PSTN End Point with a call duration longer than 32 seconds | Pass | ||
24 | 408077 | 4.3.16 Skype for Business Client calls an IVR number and navigates through the IVR menu after call connection | Pass | ||
25 | 408074 | 4.3.19 Skype for Business Client response to PSTN End Point is delayed due to network delay | Pass | ||
26 | 408072 | 4.3.20 Skype for Business Client sends INVITE with E.164 number and extension in Request and To URI | Pass | ||
27 | 408081 | 4.3.21 Mediation Server renegotiates an existing voice session with a different IP address | Pass | ||
28 | 408082 | 4.3.22 PSTN End Point calls Skype for Business Client1, Skype for Business Client1 parks the call and retrieves it on Skype for Business Client2 | Pass | ||
29 | 408069 | 4.3.23 PSTN End Point disconnects established call from Skype for Business Client | Pass | ||
30 | 408068 | 4.3.24 PSTN End Point disconnects established call to Skype for Business Client | Pass | ||
31 | 408067 | 4.3.25 PSTN End Point displays Skype for Business Client Caller ID for Outbound Call | Pass | ||
32 | 408101 | 4.4.1 Device offers DTMF payload type in the range of 96-127 to Mediation Server | Pass | ||
33 | 408092 | 4.4.2 Skype for Business Client is able to establish a call with PSTN End Point using G.711 A-law codec | Pass | ||
34 | 408086 | 4.4.3 Skype for Business Client makes a call to PSTN End Point with G.711 A-law and/or G.711 U-law codecs | Pass | ||
35 | 408114 | 4.4.4 Skype for Business Client makes a call to PSTN End Point with G.711 U-law codec | Pass | ||
36 | 408119 | 4.4.5 Skype for Business Client receives a call from PSTN End Point with G.711 U-law codecs | Pass | ||
37 | 408090 | 4.4.6 PSTN End Point is able to establish a call with Skype for Business Client using G.711 A-law codec | Pass | ||
38 | 408112 | 4.5.1 Device sends PRACK for reliable Early Media for a call from PSTN End Point to Skype for Business Client | Pass | ||
39 | 408064 | 4.5.3 Skype for Business Client calls IVR number and navigates through the IVR menu before call Connection | Pass | ||
40 | 408106 | 4.5.4 Skype for Business Client hears Early Media for a call to PSTN End Point | Pass | ||
41 | 408104 | 4.6.1 Device does not change the SSRC of an established inbound RTP session | Pass | ||
42 | 408100 | 4.6.3 Device does not change the SSRC of an established outbound RTP session | Pass | ||
43 | 408093 | 4.6.5 Device handles multiple RTP streams for a call to Skype for Business Client | Pass | ||
44 | 408103 | 4.6.6 Device may send RTCP sender and receiver reports | Pass | ||
45 | 408097 | 4.6.8 Device sends RTCP packets when Skype for Business Client places call on hold | Pass | ||
46 | 408128 | 4.6.9 Device sends RTCP packets while playing music on hold | Pass | ||
47 | 408109 | 4.7.1 Device disconnects a forked call if PSTN End Point hangs up while phones are ringing | Pass | ||
48 | 408079 | 4.7.3 PSTN End Point1 calls Skype for Business Client that is set to simultaneous ring to Skype for Business Client and PSTN End Point2 answers | Pass | ||
49 | 466206 | 4.9.1 Inbound Call QoS Remediation | Pass | ||
50 | 466207 | 4.9.2 Outbound Call QoS Remediation | Pass | ||
51 | 408225 | 5.1.1 PSTN End Point places a call to Skype for Business Client on hold for 15 minutes and then resumes (Media Bypass OFF) | Pass | ||
52 | 408231 | 5.2.1 Skype for Business Client plays music when it holds call from PSTN End Point to Skype for Business Client | Pass | ||
53 | 408234 | 5.3.1 Skype for Business Client places a call from PSTN End Point on hold for 15 minutes and then resumes | Pass | ||
54 | 408229 | 5.3.2 Skype for Business Client places a call to PSTN End Point on hold and resumes after 12 minutes | Pass | ||
55 | 408227 | 5.3.5 Skype for Business Client resumes call to PSTN End Point after playing music on hold for 15 minutes | Pass | ||
56 | 408207 | 6.1.1 PSTN End Point1 calls Skype for Business Client that forwards all calls to PSTN End Point2 when Media Bypass OFF | Pass | ||
57 | 408258 | 7.1.1 Device generates INVITE with Replaces and Referred-By headers when it receives a REFER request | Pass | ||
58 | 408254 | 7.1.2 Device includes REFER in ALLOW header in INVITE sent to Mediation Server | Pass | ||
59 | 408259 | 7.1.3 Device maintains the original session when rejecting a call transfer with REFER | Pass | ||
60 | 408257 | 7.1.4 Device supports Hairpin Elimination for Blind Transfer with REFER | Pass | ||
61 | 408255 | 7.1.7 PSTN End Point1 calls Skype for Business Client and Skype for Business Client Blinds Transfers the call to PSTN End Point2 | Pass | ||
62 | 408263 | 7.2.1 Device does not drop the call when Consultative Transfer by Skype for Business Client to second PSTN End Point fails | Pass | ||
63 | 408264 | 7.2.2 Device supports Hairpin Elimination for Consultative Transfer with REFER | Pass | ||
64 | 408261 | 7.2.4 PSTN End Point1 calls Skype for Business Client and Skype for Business Client Consultative Transfers to PSTN End Point2 | Pass | ||
65 | 408213 | 8.1.1 Skype for Business Client1 calls Skype for Business Client2 and escalates the call to a conference, inviting PSTN End Point and later removing it | Pass | ||
66 | 408214 | 8.1.3 PSTN End Point establishes a call with the Conference Auto Attendant | Pass | ||
67 | 408309 | 9.1.1 Device distributes new calls among DNS configured Mediation Servers | Pass | ||
68 | 408311 | 9.1.2 Device honors TTL when distributing new calls among DNS configured Mediation Servers | Pass | ||
69 | 408286 | 9.2.1 Device responds to OPTIONS as keep alive to Mediation Server over TCP | Pass | ||
70 | 408289 | 9.2.3 Device resumes sending calls to Mediation Server when it starts receiving OPTIONS response from that Mediation Server | Pass | ||
71 | 408287 | 9.2.4 Device sends periodic OPTIONS message as keep alive to Mediation Server | Pass | ||
72 | 408291 | 9.2.6 PSTN End Point establishes a call with Skype for Business Client when interface of Mediation Server1 goes down | Pass | ||
73 | 408293 | 9.2.7 Device fails over incoming call to a second Mediation Server when the first Mediation Server does not respond | Pass | ||
74 | 408306 | 9.2.8 Device utilizes failover and does not offer new calls to a failed Mediation Server | Pass | ||
75 | 408058 | 10.1.1 PSTN End Point calls Skype for Business Client with Caller ID set to 'Anonymous' on Device | Pass | ||
76 | 408321 | 11.1.1 Device disconnects call when Mediation Server sends 408 Request Timeout for call from PSTN End Point | Pass | ||
77 | 408327 | 11.1.2 Device disconnects call when Mediation Server sends 501 Not Implemented for call from PSTN End Point | Pass | ||
78 | 408328 | 11.1.3 Device disconnects call when Mediation Server sends 606 Not Acceptable for call from PSTN End Point | Pass | ||
79 | 408325 | 11.1.4 Device generates 486 Busy Here response from a busy PSTN End Point | Pass | ||
80 | 408324 | 11.1.5 Device handles call from Skype for Business Client to a user that does not exist in the domain | Pass | ||
81 | 408326 | 11.1.6 Device processes 486 Busy Here response from a busy Skype for Business Client | Pass | ||
82 | 408317 | 11.1.7 Device processes 488 Not Acceptable Here response for unsupported codec from Mediation Server | Pass | ||
83 | 408323 | 11.1.9 Device processes 603 Decline response from Skype for Business Client | Pass | ||
84 | 408329 | 11.1.10 Device responds with 488 Not Acceptable Here when Mediation Server offers a codec unsupported on the device | Pass | ||
85 | 408315 | 11.1.11 Device sends 414 Request-URI Too Long when unable to handle very longRequest URI | Pass | ||
86 | 438961 | 11.1.12 Reason headers are included for all 4xx, 5xx and 6xx responses | Pass | ||
87 | 438951 | 11.1.13 Reason headers are included for all BYE requests | Pass | ||
88 | 438956 | 11.1.14 Reason headers are included for all CANCEL requests | Pass | ||
89 | 408162 | 4.2.5 Device sends PRACK for reliable Early Media for a call from PSTN End Point to Skype for Business Client. (Media Bypass OFF) (IPv6) | Pass | ||
90 | 408155 | 4.2.10 PSTN End Point calls Skype for Business Client that later parks the call but does not retrieve it. (Media Bypass OFF) (IPv6) | Pass | ||
91 | 408153 | 4.2.12 PSTN End Point calls Skype for Business Client1, Skype for Business Client1 parks the call and retrieves it on Skype for Business Client2. (Media Bypass OFF) (IPv6) | Pass | ||
92 | 408148 | 4.2.13 PSTN End Point1 calls Skype for Business Client that is set to simultaneous ring to Skype for Business Client and PSTN End Point2 answers. (Media Bypass OFF) (IPv6) | Pass | ||
93 | 408164 | 4.6.7 Device may send RTCP sender and receiver reports. (Media Bypass OFF) (IPv6) | Pass | ||
94 | 408159 | 4.7.2 Device disconnects a forked call if PSTN End Point hangs up while phones are ringing. (Media Bypass OFF) (IPv6) | Pass | ||
95 | 408273 | 7.1.6 PSTN End Point calls Skype for Business Client1 that performs Blind Transfer to Skype for Business Client2 with REFER. (Media Bypass OFF) (IPv6) | Pass | ||
96 | 408216 | 8.1.2 Skype for Business Client1 calls Skype for Business Client2 and escalates the call to a conference, inviting PSTN End Point and later removing it. (Media Bypass OFF) (IPv6) | Pass | ||
97 | 408282 | 4.1.1 Device accepts Mediation Server 'pool' certificate for a secure call | Pass | ||
98 | 428361 | 4.1.2 Device offers Device pool certificate for a secure call | Pass | ||
99 | 408127 | 4.2.1 Device adds at least one "crypto" attribute for each media description line in the SDP | Pass | ||
100 | 408117 | 4.2.2 Device handles 488 Not Acceptable Here response from the Mediation Server operating in RTP only mode | Pass | ||
101 | 408124 | 4.2.3 Device sends Crypto attributes in SDP for call from PSTN End Point to Skype for Business Client | Pass | ||
102 | 408118 | 4.2.4 Device sends its own FQDN in contact header for TLS call from Skype for Business Client to PSTN End Point | Pass | ||
103 | 408115 | 4.2.7 Mediation Server that requires SRTP rejects call from Device that supports RTP only | Pass | ||
104 | 408125 | 4.2.11 PSTN End Point calls Skype for Business Client with security enabled and Skype for Business Client later hangs up | Pass | ||
105 | 408089 | 4.3.10 Device sends Comfort Noise packets to Skype for Business Client when secure call is muted | Pass | ||
106 | 408116 | 4.3.12 Device that supports SRTP only rejects call from Skype for Business Client that supports RTP Only | Pass | ||
107 | 408099 | 4.3.13 Device with RTP only setting rejects call from Skype for Business Client that requires SRTP | Pass | ||
108 | 433140 | 4.3.17 Skype for Business Client makes a secure call to an IVR andpastesa string of conference ID digits which are recognized by the Device and IVR | Pass | ||
109 | 408098 | 4.3.18 Skype for Business Client makes a secure call to an IVR number and navigates through the IVR menu after receiving 200 OK from Device | Pass | ||
110 | 408113 | 4.5.2 Device sends PRACK for reliable Early Media for call from PSTN End Point to Skype for Business Client with SRTP Optional | Pass | ||
111 | 408126 | 4.6.2 Device does not change the SSRC of an established inbound SRTP session | Pass | ||
112 | 408123 | 4.6.4 Device does not change the SSRC of an established outbound SRTP session | Pass | ||
113 | 408105 | 4.6.10 Device sends SRTCP sender and receiver reports for a secure call | Pass | ||
114 | 408110 | 4.8.1 Device disconnects a forked secure call if PSTN End Point hangs up while phones are ringing | Pass | ||
115 | 408094 | 4.8.3 Device handles multiple SRTP streams for a secure call to Skype for Business Client | Pass | ||
116 | 408107 | 4.8.4 Skype for Business Client hears Early Media for a secure call to PSTN End Point | Pass | ||
117 | 408108 | 4.8.5 Skype for Business Client hears Early Media for a secure call to PSTN End Point when Media Bypass OFF | Pass | ||
118 | 408129 | 4.8.6 Skype for Business Client makes a secure call to PSTN End Point | Pass | ||
119 | 408122 | 4.8.7 Skype for Business Client makes a secure call to PSTN End Point and PSTN End Point later hangs up | Pass | ||
120 | 408130 | 4.8.8 Skype for Business Client makes a secure call to PSTN End Point with call duration more than 32 seconds and SRTP set to Optional | Pass | ||
121 | 428537 | 4.8.9 Skype for Business Client places a secure call to PSTN End Point and call is up for more than 30 minutes with session timer enabled on Device | Pass | ||
122 | 439173 | 4.8.10 Skype for Business Client places a secure call to PSTN End Point and call is up for more than 30 minutes with session timer enabled on Device (Media Bypass OFF) | Pass | ||
123 | 408120 | 4.8.11 Skype for Business Client receives a secure call with G.711 U-law codec with Media Bypass OFF | Pass | ||
124 | 408091 | 4.8.12 PSTN End Point is able to establish a secure call with Skype for Business Client using G.711 A-law codec | Pass | ||
125 | 408235 | 5.1.2 PSTN End Point places a secure call from Skype for Business Client on hold and then resumes | Pass | ||
126 | 408224 | 5.1.3 PSTN End Point places a secure call to Skype for Business Client on hold and resumes after 15 minutes | Pass | ||
127 | 428506 | 5.3.3 Skype for Business Client places secure call to PSTN End Point on hold after 30 minutes and then resumes | Pass | ||
128 | 439169 | 5.3.4 Skype for Business Client places secure call to PSTN End Point on hold after 30 minutes and then resumes (Media Bypass OFF) | Pass | ||
129 | 408206 | 6.1.2 PSTN End Point1 makes a secure call to Skype for Business Client that forwards the call to PSTN End Point2 with Media Bypass OFF | Pass | ||
130 | 408205 | 6.1.3 PSTN End Point1 makes a secure call to Skype for Business Client that has call forwarded to PSTN End Point2 | Pass | ||
131 | 408260 | 7.1.5 Device supports Hairpin Elimination for secure Blind Transfer with REFER | Pass | ||
132 | 408256 | 7.1.8 PSTN End Point1 makes a secure call to Skype for Business Client and Skype for Business Client Blinds Transfers the call to PSTN End Point2 | Pass | ||
133 | 408265 | 7.2.3 Device supports Hairpin Elimination for secure Consultative Transfer with REFER | Pass | ||
134 | 408262 | 7.2.5 PSTN End Point1 makes a secure call to Skype for Business Client and Skype for Business Client Consultative Transfers to PSTN End Point2 | Pass | ||
135 | 408288 | 9.2.2 Device responds to OPTIONS as keep alive to Mediation Server over TLS | Pass | ||
136 | 408285 | 9.2.5 Device uses load balancing to distribute secure inbound calls among Mediation Servers in a cluster | Pass | ||
137 | 408322 | 11.1.8 Device processes 603 Decline from Skype for Business Client for a secure call | Pass | ||
138 | 408166 | 4.6.11 Device sends SRTCP sender and receiver reports for a secure call. (Media Bypass OFF) (IPv6) | Pass | ||
139 | 408160 | 4.8.2 Device disconnects a forked secure call if PSTN End Point hangs up while phones are ringing. (Media Bypass OFF) (IPv6) | Pass | ||
140 | 408268 | 7.1.9 PSTN End Point1 makes a secure call to Skype for Business Client and Skype for Business Client Blinds Transfers the call to PSTN End Point2. (Media Bypass OFF) (IPv6) | Pass | ||
141 | 408275 | 7.2.6 PSTN End Point1 makes a secure call to Skype for Business Client and Skype for Business Client Consultative Transfers to PSTN End Point2. (Media Bypass OFF) (IPv6) | Pass |
Microsoft O365 certification test results
S.No | Test Case Id | Test Case Description | Result | Observation | Comment |
1 | 465403 | 12.1.1 Mailbox login from public phone (On-Premises) | Pass | ||
2 | 465404 | 12.1.2 Mailbox navigation using VUI (On-Premises) | Pass | ||
3 | 465405 | 12.1.3 Mailbox navigation using TUI (On-Premises) | Pass | ||
4 | 465406 | 12.1.4 Leave Voicemail from an internal extension (On-Premises) | Pass | ||
5 | 465407 | 12.1.5 Leave Voicemail from an external extension (On-Premises) | Pass | ||
6 | 465408 | 12.1.6 Inbound call handled by Auto Attendant (On-Premises) | Pass | ||
7 | 465417 | 12.2.1 Voicemail using OWA’s Play-On-Phone feature to an external extension (On-Premises) | Pass | ||
8 | 465416 | 12.2.2 Voicemail using OWA’s Play-On-Phone feature to a user’s extension (On-Premises) | Pass | ||
9 | 465410 | 12.3.1 Call transferred to search target (On-Premises) | Pass | ||
10 | 465411 | 12.3.2 Call transferred to search target busy voicemail (On-Premises) | Pass | ||
11 | 465412 | 12.3.3 Call transferred to search target no-answer voicemail (On-Premises) | Pass | ||
12 | 465413 | 12.3.4 Call transferred to search default target (On-Premises) | Pass | ||
13 | 465683 | 12.4.1 Device supports FAX (On-Premises) | Not Tested | Not tested due unavailability of FAX server. | |
14 | 465792 | 12.5.1 MWI Lamp on PBX phone lights up (On-Premises) | Pass | ||
15 | 465793 | 12.5.2 MWI Lamp on PBX phone turns off (On-Premises) | Pass | ||
16 | 465798 | 12.6.1 Check Voicemail Button (On-Premises) | Pass | ||
17 | 465799 | 12.6.2 Call Forward toother UM-Enableduser(On-Premises) | Pass | ||
18 | 465693 | 13.1.1 Mailbox login from public phone (On-Line) | Pass | ||
19 | 465694 | 13.1.2 Mailbox navigation using VUI (On-Line) | Pass | ||
20 | 465695 | 13.1.3 Mailbox navigation using TUI (On-Line) | Pass | ||
21 | 465696 | 13.1.4 Leave Voicemail from an internal extension (On-Line) | Pass | ||
22 | 465697 | 13.1.5 Leave Voicemail from an external extension (On-Line) | Pass | ||
23 | 465698 | 13.1.6 Inbound call handled by Auto Attendant (On-Line) | Pass | ||
24 | 465706 | 13.2.1 Voicemail using OWA’s Play-On-Phone feature to an external extension (On-Line) | Pass | ||
25 | 465705 | 13.2.2 Voicemail using OWA’s Play-On-Phone feature to a user’s extension (On-Line) | Pass | ||
26 | 465700 | 13.3.1 Call transferred to search target (On-Line) | Pass | ||
27 | 465701 | 13.3.2 Call transferred to search target busy voicemail (On-Line) | Pass | ||
28 | 465702 | 13.3.3 Call transferred to search target no-answer voicemail (On-Line) | Pass | ||
29 | 465703 | 13.3.4 Call transferred to search default target (On-Line) | Pass | ||
30 | 465708 | 13.4.1 Device supports FAX (On-Line) | Not Tested | Not tested due unavailability of FAX server. | |
31 | 465801 | 13.6.1 Check Voicemail Button (On-Line) | Pass | ||
32 | 465802 | 13.6.2 Call Forward toother UM-Enableduser(On-Line) | Pass |
These Application Notes describe the configuration steps required for the 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.