Table of Contents
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.
- For additional information on Microsoft, visit http://microsoft.com.
- For additional information on Sonus SBC, visit https://ribboncommunications.com
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
Document History
| Date | Name | Comment |
|---|---|---|
| 30/Aug/2016 | Arun Muthusamy | Initial Draft |
| 23/May/2017 | Ankit Shukla | Removing 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:
Requirements
| Product | Equipment | Software Version |
|---|---|---|
| Sonus Networks | Sonus SBC 5200 | 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.
Microsoft Skype For Business Setup
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 commit
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 commit
5.9 Configure IP Signaling Profile
set profiles signaling ipSignalingProfile SKYPE_IPSP egressIpAttributes transport type1 tlsOverTcp commit
5.10 Attach TLS Profile to SIP Signaling Port
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
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 commit
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 commit set addressContext default zone OFFICE_ZONE sipSigPort 2 state enabled mode inService commit
Test Results
Microsoft SKYPE certification test results
| S.No | Procedure | Observation | Result | Comment |
|---|---|---|---|---|
| 1 | 408058 [PstnEndPt] calls [SKYPEEndPt] with Caller ID set to 'Anonymous' on [DUT] | PASS | ||
| 2 | 408059 [PstnEndPt] calls [SKYPEEndPt] with Caller ID set to 'Anonymous' on [DUT]. (Media Bypass OFF) (IPv6) | PASS | ||
| 3 | 408200 3rd Party Presence headers do not cause [DUT] failure | PASS | ||
| 4 | 408202 3rd Party Presence headers do not cause [DUT] failure. (Media Bypass OFF) (IPv6) | PASS | ||
| 5 | 408282 [DUT] accepts [MedSrv] 'pool' certificate for a secure call | PASS | ||
| 6 | 408064 [SKYPEEndPt] calls IVR number and navigates through the IVR menu before call Connection | PASS | ||
| 7 | 408106 [SKYPEEndPt] hears Early Media for a call to [PstnEndPt] | PASS | ||
| 8 | 408112 [DUT] sends PRACK for reliable Early Media for a call from [PstnEndPt] to [SKYPEEndPt] | PASS | ||
| 9 | 408113 [DUT] sends PRACK for reliable Early Media for call from [PstnEndPt] to [SKYPEEndPt] with SRTP Optional | PASS | ||
| 10 | 408079 [PstnEndPt]1 calls [SKYPEEndPt] that is set to simultaneous ring to [SKYPEEndPt] and [PstnEndPt]2 answers | PASS | ||
| 11 | 408109 [DUT] disconnects a forked call if [PstnEndPt] hangs up while phones are ringing | PASS | ||
| 12 | 408144 [DUT] is able to disconnect a call that isforkedto [SKYPEEndPt]s set to 'Do not disturb'. (Media Bypass OFF) (IPv6) | PASS | ||
| 13 | 408159 [DUT] disconnects a forked call if [PstnEndPt] hangs up while phones are ringing. (Media Bypass OFF) (IPv6) | PASS | ||
| 14 | 408062 [SKYPEEndPt] calls [PstnEndPt] with a call duration longer than 32 seconds | PASS | ||
| 15 | 408063 [DUT] generates [603] response for a call rejected by [PstnEndPt] | PASS | ||
| 16 | 408065 [DUT] correctly handles non-E.164 number in outbound Request URI | PASS | ||
| 17 | 408066 [SKYPEEndPt] calls [PstnEndPt] and hangs up before receiving [200] from [DUT] | PASS | ||
| 18 | 408067 [PstnEndPt] displays [SKYPEEndPt] Caller ID for Outbound Call | PASS | ||
| 19 | 408068 [PstnEndPt] disconnects established call to [SKYPEEndPt] | PASS | ||
| 20 | 408069 [PstnEndPt] disconnects established call from [SKYPEEndPt] | PASS | ||
| 21 | 408071 [DUT] processes phone-context in Request and To URI from [SKYPEEndPt] | PASS | ||
| 22 | 408072 [SKYPEEndPt] sends INVITE with E.164 number and extension in Request and To URI | PASS | ||
| 23 | 408073 [DUT] processes call from [SKYPEEndPt] with E.164 number in FROM Header URI | PASS | ||
| 24 | 408074 [SKYPEEndPt] response to [PstnEndPt] is delayed due to network delay | PASS | ||
| 25 | 408077 [SKYPEEndPt] calls an IVR number and navigates through the IVR menu after call connection. | PASS | ||
| 26 | 408078 [DUT] handles call from [MedSrv] with an alias name in the FROM header | PASS | ||
| 27 | 408081 [MedSrv] renegotiates an existing voice session with a different IP address | PASS | ||
| 28 | 408085 [DUT] establishes call to [SKYPEEndpt] with configured value of ptime | PASS | ||
| 29 | 408131 [SKYPEEndPt] calls [PstnEndPt] with a call duration longer than 32 seconds. (Media Bypass OFF) (IPv6) | PASS | ||
| 30 | 408132 [DUT] generates [603] response for a call rejected by [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 31 | 408133 [SKYPEEndPt] calls IVR number and navigates through the IVR menu before call Connection. (Media Bypass OFF) (IPv6) | PASS | ||
| 32 | 408134 [DUT] correctly handles non-E.164 number in outboundRequest URI. (Media Bypass OFF) (IPv6) | PASS | ||
| 33 | 408135 [SKYPEEndPt] calls [PstnEndPt] and hangs up before receiving [200] from [DUT]. (Media Bypass OFF) (IPv6) | PASS | ||
| 34 | 408136 [PstnEndPt] displays [SKYPEEndPt] Caller ID for Outbound Call. (Media Bypass OFF) (IPv6) | PASS | ||
| 35 | 408138 [PstnEndPt] disconnects established call from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 36 | 408140 [DUT] processes phone-context in Request and To URI from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 37 | 408141 [SKYPEEndPt] sends INVITE with E.164 number and extension in Request and To URI. (Media Bypass OFF) (IPv6) | PASS | ||
| 38 | 408142 [DUT] processes call from [SKYPEEndPt] withE.164number in FROM Header URI. (Media Bypass OFF) (IPv6) | PASS | ||
| 39 | 408143 [SKYPEEndPt] response to [PstnEndPt] is delayed due to network delay. (Media Bypass OFF) (IPv6) | PASS | ||
| 40 | 408146 [SKYPEEndPt] calls an IVR number and navigates through the IVR menu after call connection. (Media Bypass OFF) (IPv6) | PASS | ||
| 41 | 408147 [DUT] handlescallfrom [MedSrv] with an alias name in the FROM header. (Media Bypass OFF) (IPv6) | PASS | ||
| 42 | 408086 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs | PASS | ||
| 43 | 408090 [PstnEndPt] is able to establish a call with [SKYPEEndPt] using G.711 A-law codec | PASS | ||
| 44 | 408092 [SKYPEEndPt] is able to establish a call with [PstnEndPt] using G.711 A-law codec | PASS | ||
| 45 | 408101 [DUT] offers DTMF payload type in the range of 96-127 to [MedSrv] | PASS | ||
| 46 | 408114 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 U-law codec | PASS | ||
| 47 | 408119 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 U-law codecs | PASS | ||
| 48 | 408150 [MedSrv] renegotiates an existing voice session with a different IP address. (Media Bypass OFF) (IPv6) | PASS | ||
| 49 | 408152 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs. (Media Bypass OFF) (IPv6) | PASS | ||
| 50 | 408167 [DUT] offers DTMF payload type in the range of 96-127 to [MedSrv]. (Media Bypass OFF) (IPv6) | PASS | ||
| 51 | 408174 [SKYPEEndPt] is able to establish a call with [PstnEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) | PASS | ||
| 52 | 408180 [SKYPEEndPt] makes a call to [PstnEndPt] with G.711 U-law codec. (Media Bypass OFF) (IPv6) | PASS | ||
| 53 | 408181 [PstnEndPt] is able to establish a call with [SKYPEEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) | PASS | ||
| 54 | 408183 [DUT] negotiates Comfort Noise in a call from [SKYPEEndPt] to [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 55 | 408186 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs. (Media Bypass OFF) (IPv6) | PASS | ||
| 56 | 408187 [SKYPEEndPt] receives a secure call with G.711 U-law codec. (Media Bypass OFF) (IPv6) | PASS | ||
| 57 | 408093 [DUT] handles multiple RTP streams for a call to [SKYPEEndPt] | PASS | ||
| 58 | 408123 [DUT] does not change the SSRC of an established outbound SRTP session | PASS | ||
| 59 | 408126 [DUT] does not change the SSRC of an established inbound SRTP session | PASS | ||
| 60 | 408190 [DUT] does not change the SSRC of an established inbound SRTP session. (Media Bypass OFF) (IPv6) | PASS | ||
| 61 | 408193 [DUT] does not change the SSRC of an established outbound SRTP session. (Media Bypass OFF) (IPv6) | PASS | ||
| 62 | 408091 [PstnEndPt] is able to establish a secure call with [SKYPEEndPt] using G.711 A-law codec | PASS | ||
| 63 | 408094 [DUT] handles multiple SRTP streams for a secure call to [SKYPEEndPt] | PASS | ||
| 64 | 408107 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt] | PASS | ||
| 65 | 408108 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt] when Media Bypass OFF | PASS | ||
| 66 | 408110 [DUT] disconnects a forked secure call if [PstnEndPt] hangs up while phones are ringing | PASS | ||
| 67 | 408120 [SKYPEEndPt] receives a secure call with G.711 U-law codec with Media Bypass OFF | PASS | ||
| 68 | 408122 [SKYPEEndPt] makes a secure call to [PstnEndPt] and [PstnEndPt] later hangs up | PASS | ||
| 69 | 408129 [SKYPEEndPt] makes a secure call to [PstnEndPt] | PASS | ||
| 70 | 408130 [SKYPEEndPt] makes a secure call to [PstnEndPt] with call duration more than 32 seconds and SRTP set to Optional | PASS | ||
| 71 | 408160 [DUT] disconnects a forked secure call if [PstnEndPt] hangs up while phones are ringing. (Media Bypass OFF) (IPv6) | PASS | ||
| 72 | 408173 [SKYPEEndPt] hears Early Media for a secure call to [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 73 | 408182 [PstnEndPt] is able to establish a secure call with [SKYPEEndPt] using G.711 A-law codec. (Media Bypass OFF) (IPv6) | PASS | ||
| 74 | 408195 [SKYPEEndPt] makes a secure call to [PstnEndPt] with call duration more than 32 seconds and SRTP set to Optional. (Media Bypass OFF) (IPv6) | PASS | ||
| 75 | 408196 [SKYPEEndPt] makes a secure call to [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 76 | 408070 [PstnEndPt] calls [SKYPEEndPt] and hangs up while [SKYPEEndPt] is still ringing | PASS | ||
| 77 | 408080 Inboundcallto [SKYPEEndPt] from [PstnEndPt] with a very long Request-URI in the INVITE | PASS | ||
| 78 | 408117 [DUT] handles [488] response from the [MedSrv] operating in RTP only mode | PASS | ||
| 79 | 408118 [DUT] sends its own FQDN in contact header for TLS call from [SKYPEEndPt] to [PstnEndPt] | PASS | ||
| 80 | 408124 [DUT] sends Crypto attributes in SDP for call from [PstnEndPt] to [SKYPEEndPt] | PASS | ||
| 81 | 408125 [PstnEndPt] calls [SKYPEEndPt] with security enabled and [SKYPEEndPt] later hangs up | PASS | ||
| 82 | 408127 [DUT] adds at least one 'crypto' attribute for each media description line in the SDP | PASS | ||
| 83 | 408137 [PstnEndPt] disconnects established call to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 84 | 408139 [PstnEndPt] calls [SKYPEEndPt] and hangs up while [SKYPEEndPt] is still ringing. (Media Bypass OFF) (IPv6) | PASS | ||
| 85 | 408145 [DUT] sends single media description line for a call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 86 | 408149 Inboundcallto [SKYPEEndPt] from [PstnEndPt] with a very long Request-URI in the INVITE. (Media Bypass OFF) (IPv6) | PASS | ||
| 87 | 408151 [DUT] establishescallto [SKYPEEndpt] with configured value ofptime. (Media Bypass OFF) (IPv6) | PASS | ||
| 88 | 408170 [DUT] with RTP only setting rejectscallfrom [SKYPEEndPt] that requires SRTP. (Media Bypass OFF) (IPv6) | PASS | ||
| 89 | 408172 [SKYPEEndPt] hears Early Media for a call to [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 90 | 408188 [DUT] sends Crypto attributes in SDP for call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 91 | 408189 [PstnEndPt] calls [SKYPEEndPt] with security enabled and [SKYPEEndPt] later hangs up. (Media Bypass OFF) (IPv6) | PASS | ||
| 92 | 408191 [DUT] adds at least one 'crypto' attribute for each media description line in the SDP. (Media Bypass OFF) (IPv6) | PASS | ||
| 93 | 408207 [PstnEndPt]1 calls [SKYPEEndPt] that forwards all calls to [PstnEndPt]2 when Media Bypass OFF | PASS | ||
| 94 | 408209 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] that forwards the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 95 | 408210 [PstnEndPt]1 calls [SKYPEEndPt] that forwards all calls to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 96 | 408227 [SKYPEEndPt] resumes call to [PstnEndPt] after playing music on hold for 15 minutes | PASS | ||
| 97 | 408229 [SKYPEEndPt] places a call to [PstnEndPt] on hold and resumes after 12 minutes | PASS | ||
| 98 | 408234 [SKYPEEndPt] places a call from [PstnEndPt] on hold for 15 minutes and then resumes | PASS | ||
| 99 | 408242 [DUT] disconnects a call that is on hold when [SKYPEEndPt] hangs up. (Media Bypass OFF) (IPv6) | PASS | ||
| 100 | 408243 [DUT] disconnects a call that is on hold when [PstnEndPt] hangs up. (Media Bypass OFF) (IPv6) | PASS | ||
| 101 | 408248 [SKYPEEndPt] places a call from [PstnEndPt] on hold for 15 minutes and then resumes. (Media Bypass OFF) (IPv6) | PASS | ||
| 102 | 408250 [SKYPEEndPt] places a call to [PstnEndPt] on hold and resumes after 12 minutes. (Media Bypass OFF) (IPv6) | PASS | ||
| 103 | 408224 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and resumes after 15 minutes | PASS | ||
| 104 | 408225 [PstnEndPt] places a call with Media Bypass OFF from [SKYPEEndPt] on hold for 15 minutes and then resumes | PASS | ||
| 105 | 408226 [PstnEndPt] puts [SKYPEEndPt] on hold and resumes after 15 minutes for a secure call | PASS | ||
| 106 | 408235 [PstnEndPt] places a secure call from [SKYPEEndPt] on hold and then resumes | PASS | ||
| 107 | 408236 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and then resumes | PASS | ||
| 108 | 408237 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and resumes after 15 minutes. (Media Bypass OFF) (IPv6) | PASS | ||
| 109 | 408238 [PstnEndPt] places a call from [SKYPEEndPt] on hold for 15 minutes and then resumes. (Media Bypass OFF) (IPv6) | PASS | ||
| 110 | 408239 [PstnEndPt] puts [SKYPEEndPt] on hold and resumes after 15 minutes for a secure call. (Media Bypass OFF) (IPv6) | PASS | ||
| 111 | 408240 [PstnEndPt] places a secure call to [SKYPEEndPt] on hold and then resumes. (Media Bypass OFF) (IPv6) | PASS | ||
| 112 | 408244 [PstnEndPt] places a secure call from [SKYPEEndPt] on hold and then resumes. (Media Bypass OFF) (IPv6) | PASS | ||
| 113 | 408231 [SKYPEEndPt] plays music when it holds call from [PstnEndPt] to [SKYPEEndPt] | PASS | ||
| 114 | 408241 [SKYPEEndPt] resumes call to [PstnEndPt] after playing music on hold for 15 minutes. (Media Bypass OFF) (IPv6) | PASS | ||
| 115 | 408245 [SKYPEEndPt] plays music when it holds call from [PstnEndPt] to [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 116 | 408261 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2 | PASS | ||
| 117 | 408262 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2 | PASS | ||
| 118 | 408274 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 119 | 408275 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Consultative Transfers to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 120 | 408276 [DUT] does not drop the call when Consultative Transfer by [SKYPEEndPt] to second [PstnEndPt] fails. (Media Bypass OFF) (IPv6) | PASS | ||
| 121 | 408254 [DUT] includes REFER in ALLOW header in INVITE sent to [MedSrv] | PASS | ||
| 122 | 408255 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2 | PASS | ||
| 123 | 408256 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2 | PASS | ||
| 124 | 408266 [DUT] includes REFER in ALLOW header in INVITE sent to [MedSrv]. (Media Bypass OFF) (IPv6) | PASS | ||
| 125 | 408267 [PstnEndPt]1 calls [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 126 | 408268 [PstnEndPt]1 makes a secure call to [SKYPEEndPt] and [SKYPEEndPt] Blinds Transfers the call to [PstnEndPt]2. (Media Bypass OFF) (IPv6) | PASS | ||
| 127 | 408285 [DUT] uses load balancing to distribute inbound calls among [MedSrv]s in a cluster | PASS | ||
| 128 | 408286 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TCP | PASS | ||
| 129 | 408288 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TLS | PASS | ||
| 130 | 408291 [PstnEndPt] establishes a call with [SKYPEEndPt] when interface of [MedSrv]1 goes down | PASS | ||
| 131 | 408293 [DUT] fails over incoming call to a second [MedSrv] when the first [MedSrv] does not respond | PASS | ||
| 132 | 408295 [DUT] fails over incoming call to [MedSrv]2 when [MedSrv]1 sends [503] response. (Media Bypass OFF) (IPv6) | PASS | ||
| 133 | 408296 [DUT] uses load balancing to distribute inbound calls among [MedSrv]s in a cluster. (Media Bypass OFF) (IPv6) | PASS | ||
| 134 | 408297 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TCP. (Media Bypass OFF) (IPv6) | PASS | ||
| 135 | 408299 [DUT] responds to OPTIONS as keep alive to [MedSrv] over TLS. (Media Bypass OFF) (IPv6) | PASS | ||
| 136 | 408304 [DUT] fails over incoming call to a second [MedSrv] when the first [MedSrv] does not respond (Media Bypass OFF) (IPv6) | PASS | ||
| 137 | 408309 [DUT] distributes new calls among DNS configured [MedSrv]s | PASS | ||
| 138 | 408310 [DUT] distributes new calls among DNS configured [MedSrv]s. (Media Bypass OFF) (IPv6) | PASS | ||
| 139 | 408315 [DUT] sends [414] when unable to handle very long Request URI | PASS | ||
| 140 | 408316 [DUT] times out after 180 seconds of no response from [SKYPEEndPt] following [100] | PASS | ||
| 141 | 408317 [DUT] processes [488] response for unsupported codec from [MedSrv] | PASS | ||
| 142 | 408321 [DUT] disconnects call when [MedSrv] sends [408] for call from [PstnEndPt] | PASS | ||
| 143 | 408322 [DUT] processes [603] from [SKYPEEndPt] for a secure call | PASS | ||
| 144 | 408323 [DUT] processes [603] response from [SKYPEEndPt] | PASS | ||
| 145 | 408324 [DUT] handles call from [SKYPEEndPt] to a user that does not exist in the domain | PASS | ||
| 146 | 408325 [DUT] generates [486] response from a busy [PstnEndPt] | PASS | ||
| 147 | 408326 [DUT] processes [486] response from a busy [SKYPEEndPt] | PASS | ||
| 148 | 408327 [DUT] disconnects call when [MedSrv] sends [501] for call from [PstnEndPt] | PASS | ||
| 149 | 408328 [DUT] disconnects call when [MedSrv] sends [606] for call from [PstnEndPt] | PASS | ||
| 150 | 408329 [DUT] responds with [488] when [MedSrv] offers a codec unsupported on the device | PASS | ||
| 151 | 408330 [DUT] sends [414] when unable to handle very long Request-URI. (Media Bypass OFF) (IPv6) | PASS | ||
| 152 | 408331 [DUT] times out after 180 seconds of no response from [SKYPEEndPt] following [100]. (Media Bypass OFF) (IPv6) | PASS | ||
| 153 | 408332 [DUT] processes [488] response for unsupported codec from [MedSrv]. (Media Bypass OFF) (IPv6) | PASS | ||
| 154 | 408336 [DUT] disconnects call when [MedSrv] sends [408] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 155 | 408337 [DUT] processes [603] from [SKYPEEndPt] for a secure call. (Media Bypass OFF) (IPv6) | PASS | ||
| 156 | 408338 [DUT] processes [603] response from [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 157 | 408339 [DUT] processes [486] response from a busy [SKYPEEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 158 | 408340 [DUT] disconnects call when [MedSrv] sends [501] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 159 | 408341 [DUT] disconnects call when [MedSrv] sends [606] for call from [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 160 | 408342 [DUT] responds with [488] when [MedSrv] offers a codec unsupported on the device. (Media Bypass OFF) (IPv6) | PASS | ||
| 161 | 408343 [DUT] handlescallfrom [SKYPEEndPt] to a user that does not exist in the domain. (Media Bypass OFF) (IPv6) | PASS | ||
| 162 | 408344 [DUT] generates [486] response from a busy [PstnEndPt]. (Media Bypass OFF) (IPv6) | PASS | ||
| 163 | 408347 [SKYPEEndPt] receives a call from [PstnEndPt] with G.711 A-law and/or G.711 U-law codecs | PASS | ||
| 164 | 408350 [DUT] fails over incoming call to [MedSrv]2 when [MedSrv]1 sends [503] response | PASS | ||
| 165 | 408351 [PstnEndPt] places a call from [SKYPEEndPt] on hold for 15 minutes and then resumes | PASS | ||
| 166 | 408223 Device is able to disconnect a call that is on hold when it receives a BYE from PSTN End Point | PASS | ||
| 167 | 408076 Device is able to disconnect a call that is on hold when it receives a BYE from PSTN End Point | PASS | ||
| 168 | 408075 Device can disconnect a forked call when all SKYPE End Points are set to 'Do not Disturb'. | PASS | ||
| 169 | 408082 Call parking functionality for an inbound call. | PASS | ||
| 170 | 408222 Device is able to disconnect a call that's on hold when it receives a BYE from Mediation Server. | PASS | ||
| 171 | 408087 Device is able to negotiate Comfort Noise as part of the SDP negotiation in an outbound call from SKYPE End Point | PASS | ||
| 172 | 408153 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 | ||
| 173 | 408162 Device sends PRACK for reliable Early Media for a call from PSTN End Point to SKYPE End Point. (Media Bypass OFF) (IPv6) | PASS | ||
| 174 | 408116 Device that supports SRTP only rejects call from SKYPE End Point that supports RTP Only | PASS | ||
| 175 | 408099 Device with RTP only setting rejects call from SKYPE End Point that requires SRTP | PASS | ||
| 176 | 408163 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 | ||
| 178 | 408177 Device that supports SRTP only rejectscallfrom SKYPE End Point that supports RTP Only. (Media Bypass OFF) (IPv6) | PASS | ||
| 179 | 408178 Device handles 488 Not Acceptable Here response from the Mediation Server operating in RTPonlymode. (Media Bypass OFF) (IPv6) | PASS | ||
| 180 | 408179 Device sends its own FQDN in contact header for TLS call from SKYPE End Point to PSTN End Point. (Media Bypass OFF) (IPv6) | PASS | ||
| 181 | 408192 SKYPE End Point makes a secure call to PSTN End Point and PSTN End Point later hangs up. (Media Bypass OFF) (IPv6) | PASS | ||
| 182 | 408148 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 | ||
| 183 | 408115 Mediation Server that requires SRTP rejects call from Device that supports RTP only | PASS | ||
Microsoft O365 certification test results
| S.No | Procedure | Observation | Result | 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) | |||
| 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) | |||
| 31 | 465801 13.6.1 Check Voicemail Button (On-Line) | PASS | ||
| 32 | 465802 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.
Overview
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