Background Definitions Public switched telephone network PSTN Private

Background Definitions • Public switched telephone network (PSTN) • Private Branch e. Xchange (PBX) • Voice over Internet Protocol (Vo. IP) • Session Initiation Protocol (SIP) • Internet Telephony Service Provider (ITSP) • Uniform Resource Identifier (SIP URI) • Multiple Points of Presence (MPOP)

UCOIP • Skype for Business Certification Program • Testing and qualification of third party solutions for interoperability with Microsoft UC • Independent testing by third party labs based on standards based open documentation • SIP trunking providers supported with Lync Server 2013 will be supported with Skype for Business Qualified Gateway Qualified PBX Supported PBX

Typical Legacy Enterprise PBX PSTN Numbering Plan 31 -20 -500 1000 to +31 -20 -500 1999 Class of Service Inbound/Outbound Local, National Class of Service Outbound only Local, National, and International Dialing Habits 4 digit internal extensions 9 for an outside line 3 digits + extension for other locations

Decision 1: Legacy PBX integration Connect Skype for Business directly to the PSTN Connect Skype for Business to the Legacy PBX PSTN

Decision 2: POTS/TDM or SIP Trunking Connecting through a Gateway PSTN Connecting through SIP Trunk PSTN SIP TDM

Direct Connection Through a Gateway • A gateway is a physical device that connects two incompatible networks • The gateway translates signaling and media between Skype for Business (SIP) and the PSTN • Use supported gateways (UCOIP) Skype for Business Mediation Server Pool Qualified PSTN Gateway PSTN SIP TDM

Direct Connection Through SIP Trunking • IP connection that establishes a SIP communications link between your organization and an Internet telephony service provider (ITSP) beyond your firewall • Use supported SIP Trunking Provider (UCOIP) Session Border Controller (SBC) Skype for Business Mediation Skype for Business Server Pool Qualified IP-PSTN Gateway PSTN Enterprise Network SIP VPN ITSP Network TDM

User Configuration

Connecting Through PBX by Using SIP PSTN Skype for Business Mediation Skype for Business Server Pool Qualified or supported IP-PBX SIP TDM IP endpoint

Connecting Through PBX by Using a Gateway Skype for Business Pool Skype for Business Mediation Server Qualified IP-PSTN Gateway PSTN TDM or unsupported PBX SIP TDM IP endpoint

PSTN Sizing 1. In replacement scenarios, existing call volume is known 2. Account for new behaviors and features: Simultaneous ringing • PSTN conferencing • Dial-in audio conferencing • Mobile users • • Use Erlang B calculations when appropriate

Inter-Trunk Routing - Overview • Skype for Business Server 2015 supports call routing from an incoming trunk to an outgoing trunk to provide routing functionalities to other telephony systems • A possible alternative for PBX Integration scenario’s • By enabling inter-trunk routing, the following routing paths (among others) are enabled: Incoming PSTN calls to an IP-PBX system via Lync • Outgoing IP-PBX calls to a PSTN network via Lync • Outgoing IP-PBX calls to another IP-PBX system via Lync •

Inter-Trunk Routing – Description • Skype for Business Server 2015 allows to the associate a set of PSTN usages on an incoming trunk to determine a call route to an outgoing trunk • These PSTN usages are used to determine destination for incoming call on a trunk, if the call can’t be terminated locally No local client or other entity is found (essentially, the RNL fails) • No match to Call. Park range or Unassigned Numbers range • • Inter-trunk routing call authorization scope is at the trunk level • The same call authorization applies to all incoming calls arriving via the trunk, that can’t be terminated locally on a client • Media bypass in inter-trunk routing calls is supported

Inter-Trunk Routing IP-PBX to IP-PBX Peer to Peer Routing without Skype for Business Server 2015 Inter-Trunk Routing

Inter-Trunk Routing – Signaling and Media Flow • • • •

Configuring Inter-Trunk Routing • Use the Skype for Business Management Shell • Configure a voice route New-Cs. Voice. Route -Identity Redmond. Route -Pstn. Usages @{add=“Redmond"} -Pstn. Gateway. List @{add="Pstn. Gateway: redmondgw 1. contoso. com"} • Add a PSTN usage to a trunk configuration: • New -PSTNUsages property has been added to CSTrunk. Configuration • Or use the Skype for Business Control Panel

Mediation Server • Collocation vs. Standalone Collocation can offer significant server count reduction • Standalone may be preferable for network zone placement or workload isolation • • Media Bypass and Scalability Scale based on hardware and transcoding mix • For planning, do not count calls with media bypass • • Pool vs. Single Server • Can gateway or SIP trunk support DNS load balancing?

Media Bypass

Location Based Routing Hyderabad Skype for Business Pool Bangalore Gateway Hyderabad Gateway Skype for Business Mediation Server PSTN

Interworking Routing-History Lync Server 2010: Multiple PSTN gateways can be associated with the same Mediation Server pool (1: N); a single PSTN gateway is associated with a single Mediation Server pool; a single SIP listening port on the Mediation Server and on the gateway is used in the association. Lync Server 2013: Introduces M: N Interworking routing. A particular PSTN gateway can be associated with multiple Mediation Server pools or the same Mediation Server pool with multiple unique associations. Skype for Business Server 2015: Introduces M: N Interworking routing. A particular PSTN gateway can be associated with multiple Mediation Server pools or the same Mediation Server pool with multiple unique associations.

Trunk and IP-PBX Interworking • Multiple trunks between a Mediation Server and PSTN gateway can be defined to represent IP-PBX SIP termination • Each trunk will be associated with the appropriate route for outbound calls from Mediation Server to IP-PBX • For inbound calls, per-trunk policy will be applied • Trunk configuration will be scoped globally or per trunk; similarly, dial plan can be scoped per trunk • Representative Media IP is a pertrunk parameter, allowing for Media Bypass Mediation Server IP-PBX Port A Trunk 1 Port A 1 Port B Trunk 2 Port B 1 Port n Trunk n Port n 1

Trunk and IP-PBX Interworking-Real Life Trunk 1: MS 10 to PBX 01 Mediation Server (MS 10) PBX 01 port: 5060 Signaling IP: PBX-1 Media IP: MTP-1 IP-PBX/Gateway (PBX 01) Trunk 2: MS 10 to PBX 01 port: 5061 Signaling IP: PBX-1 Media IP: MTP-2

Configuration Details • Topology Builder: • Define the PSTN Gateway and Trunks Define the MTP as the Alternate Media IP address • Use different gateway listening ports for each trunk • • Publish the topology • Windows Power. Shell: Identify the trunk IDs • Use Windows Power. Shell to configure media IP addresses for the remaining trunks • Verify the media IP address for the trunks •

Trunks and Resiliency Mediation Server MS 1 Trunk 1 Port B Port A Gateway GW 1 Trunk 2 Gateway GW 2 Trunk 3 Port E Port C Mediation Server MS 2

Multiple Sites to the Same Service Provider Lync Server 2010: • Virtual gateways must be defined to allow connectivity from multiple Mediation Server pools to the same Session Border Controller (SBC) FQDN • Virtual gateway FQDNs all resolve to the same IP SBC sbc 1. provider. com PSTN address • TLS cannot be used because the SBC certificate does not contain the virtual gateway’s name • Gateway-specific inbound policies cannot be applied when virtual gateways are used (RNL of the IP-address does not resolve to virtual gateway) Lync Server 2013 & Skype for Business: • Separates PSTN gateways and trunks • Enable you to connect multiple trunks to one gateway • Enables the use of TLS • Allows for gateway-specific inbound policies Trunk 1 Site 01 Mediation Pool Lync Pool Trunk 2 MPLS Site 02 Mediation Pool

M: N Interworking-Trunk Definition

Auxiliary Calling Information Skype for Business Server 2015 Incoming Call to +1 (989) 555 0200 PSTN Phone +1 (999) 555 2001 User Bob +1 (989) 555 0200 Simultaneous Ring: +1 (999) 555 1000 INVITE sip: +19995551000@192. 168. 1. 41; user=phone SIP/2. 0 PSTN Phone FROM: sip: +19995552001@contoso. com; user=phone +1 (999) 555 1000 TO: sip: +19995551000@192. 168. 1. 41; user=phone HISTORY-INFO: sip: +19895550200@se 01. contoso. local; user=phone ms-retarget-reason=forwarding, sip: +19995551000@se 01. contoso. local; user=phone P-ASSERTED-IDENTITY: <tel: +19995552001> SIP Header sent to 19995551000

Fast Failover and Options Polling • Gateway Log 1 d: 0 h: 12 m: 15 s OPTIONS sip: 192. 168. 1. 41 SIP/2. 0 FROM: <sip: se 01. tailspin. local: 5068; transport=Tcp; ms-opaque=6 b 773 cd 98097 b 3 f 8>; epid=BE 80 B 79150; tag=cdee 90 d 70 TO: <sip: 192. 168. 1. 41> CSEQ: 3 OPTIONS CALL-ID: 598 db 21985 cb 4 d 38 a 5 e 89 a 410987464 a MAX-FORWARDS: 70 VIA: SIP/2. 0/TCP 192. 168. 1. 52: 59546; branch=z 9 h. G 4 b. K 3 b 462 b 11 CONTACT: <sip: se 01. tailspin. local: 5068; transport=Tcp; maddr=192. 168. 1. 52> CONTENT-LENGTH: 0 USER-AGENT: RTCC/5. 0. 0. 0 Mediation. Server • Mediation Server Log 1 d: 0 h: 12 m: 15 s SIP/2. 0 200 OK Via: SIP/2. 0/TCP 192. 168. 1. 52: 59546; branch=z 9 h. G 4 b. K 3 b 462 b 11 From: <sip: se 01. tailspin. local: 5068; transport=Tcp; msopaque=6 b 773 cd 98097 b 3 f 8>; epid=BE 80 B 79150; tag=cdee 90 d 70 To: <sip: 192. 168. 1. 41>; tag=1 c 1952373857 Call-ID: 598 db 21985 cb 4 d 38 a 5 e 89 a 410987464 a CSeq: 3 OPTIONS Contact: <sip: 192. 168. 1. 41: 5060; transport=tcp> Supported: 100 rel Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE Server: Audiocodes-Sip-Gateway-/v. 5. 80 A. 053. 005

Call-Routing Reliability-Lost Connection Skype for Business Mediation Server Pool Route Policy: For the example session only, Gateway (GW-01 and GW-03 in that order) can be used Qualified Gateways options GW-01 options 503 response MS-01 options GW-02 options MS-02 Front-End Server SIP Configured Trunk GW-03 Control messages

Call-Routing Reliability—Gateway Down Skype for Business Mediation Server Pool Route Policy: For the example session only Gateway GW-01 and GW-03 in that order can be used Qualified Gateways options GW-01 options 504 response MS-01 GW-02 options MS-02 Front-End Server SIP Configured Trunk GW-03 Control messages

Call-Routing Reliability and Retries Skype for Business Server 2015 (FE) 10 -sec timer-1: starts Timer-1: continues Timer-1: expires 10 -sec timer-1: starts Timer-1: continues Timer-1: stops (MS) (GW 1) Invite (trunk 1) 183 response Failed Connection Cancel (trunk 1) Invite (trunk 2) 183 response Invite 18 x response (GW 2)

Call-Routing Reliability—Next-Hop Proxy • The Mediation Server tracks its next-hop proxy and backup next-hop proxy by sending out periodic options polls: • Backup next-hop proxy is defined by pool pairing • If the primary next-hop proxy is found to be down (failure to answer to five options polls in a row), new invites from gateways are sent to the backup next-hop proxy • Additionally, a 10 -second timer is used for incoming calls, so if the primary next-hop proxy is used for a call and no SIP response is received within this time, the call is rerouted to the backup next-hop proxy

Voice Routing Coexistence Home Server Mediation Server Supported Skype for Business 2015 Yes Skype for Business Server 2015 and Lync Server 2013 2015 2013 Yes 2013 2015 Yes Skype for Business Server 2015 and Lync Server 2010 2015 2010 Yes 2010 2015 No Outbound Calls Inbound Calls Mediation Next-hop Server Home Server Skype for Business Server 2015 Yes Skype for Business Server 2015 and Lync Server 2013 2015 2013 Yes 2013 2015 Yes Skype for Business Server 2015 and Lync Server 2010 2015 2010 Yes 2010 2015 Yes

Lesson 5: Call via Work - Expanding Voice interoperability to the PBX phone Skype Voice for PBX Users End-users can make voice calls using any PSTN phone, including existing PBX endpoints Leverages existing Direct SIP connectivity between PBX systems and Skype for Business User Experience Server dials out to PSTN or Deskphone number to connect user, then connects with far-end destination Features Presence update & call control from rich client Mid-call control capabilities preserved on PBX phone

Call via Work - Components 1. User instantiates call from Skype rich client 2. Skype for Business Server places call to Destination 6 PSTN 3. PBX routes call and local user answers. 5 Skype Server Pool 4. When Server sees this call answered, places 4 far-end call. Here the server will use PBX user’s DID as ANI PBX 2 1 user’s PBX station set (or to any other PSTN phone number) Local call 3 (or to any other local PBX endpoint) Far-end call Skype for Business PBX Station 5. PBX routes call out to PSTN with user’s DID 6. Far-end call answers & call is established with client acting as control channel

Establishing a call

Mid call controls

Adding Modalities to a Skype for Business call

Adding Modalities (IM)

Multiple Calls • User warned on accepting/placing 2 nd call • Lose control of the 1 st call from client when second call is started. • Remote participant activity Remote participant may accept or place another call from/to someone • This will make the call on PBX Phone go on hold for the local user, • Conversation Window will not update to show the accurate status of the call. •

Ending a call • Placing the receiver of the PBX phone on the handset • Clicking the hang-up button • Close out (“x”) on the Conversation Window

Conversation History • Works as expected • The initial inbound calls are not shown in Conversation History view. • Inbound missed calls PBX or Gateway should support Reason header “Call completed elsewhere ” in the CANCEL message • If PBX does not send this Reason header, Server will treat incoming call as missed. •

Meetings • Client • Dialog will prompt for meeting join preference auto-populated • Focus dials out to user’s Cv. W configured number Click to Join • Meet Now & Ad-hoc Group Call • Ad-hoc incoming group calls •

Inbound Calls • Call via Work is Outbound Only • Inbound experience to both client & phone achieved when Skype is first in line & forwarded with Call FW settings • When PBX is first in line, inbound call will land only on desktop phone.

Presence Scenario Behavior Outbound Cv. W Call Presence will change to “In a Call” Outbound Meet Now / Group Call Presence will change to “In a Conference Call” Inbound Cv. W Call – Answered on PBX No change to presence Inbound Cv. W Call – Answered on Skype Presence will change to “In a Call” Inbound Meet Now / Group Call Presence will change to “In a Conference Call”

Policy and User configuration

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