Avaya Solution & Interoperability Test Lab

Application Notes for Skype Connect™ 2.0 Service with Avaya Communication Server 1000 Release 7.5, Avaya Aura® Session Manager 6.1 and Avaya Aura® Session Border Controller 6.0 – Issue 1.0 Abstract These Application Notes describe a solution comprised of Avaya Communication Server 1000E release 7.5, Avaya Aura® Session Manager, Avaya Aura® Session Border Controller and Skype Connect™ 2.0 Service. During the interoperability testing, Avaya Communication Server 1000E was able to interoperate with Skype Connect Service. This test was performed to verify the calls among CS1000E, PSTN and Skype user such as basic call, call forward (all calls, busy, no answer), call transfer (blind and consult), conference, Avaya CallPilot® voice mail,...etc.

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1.

INTRODUCTION ....................................................................................................................................................... 4

2.

EQUIPMENT AND SOFTWARE VALIDATED ............................................................................................................... 6

3.

AVAYA COMMUNICATION SERVER 1000E CONFIGURATION ................................................................................... 7 3.1. LOGIN TO CS1000E ELEMENT MANAGER (EM) .............................................................................................................. 7 3.2. LOGIN TO CS1000E CLI (CORES) ................................................................................................................................. 9 3.3. OBTAIN NODE IP ADDRESS ........................................................................................................................................ 10 3.4. ADMINISTER TERMINAL PROXY SERVER (TPS) ............................................................................................................... 11 3.5. ADMINISTER QUALITY OF SERVICE (QOS)...................................................................................................................... 12 3.6. SYNCHRONIZE THE NEW CONFIGURATION ..................................................................................................................... 13 3.7. ENABLE VOICE CODECS FOR THE NODE ......................................................................................................................... 15 3.8. ENABLE VOICE CODECS ON MEDIA GATEWAYS............................................................................................................... 16 3.9. ZONE AND BANDWIDTH MANAGEMENT ....................................................................................................................... 19 3.9.1 Create a Zone for IP phones............................................................................................................................ 19 3.9.2 Create a Zone for Virtual SIP Trunk................................................................................................................. 21 3.10. ADMINISTER SIP TRUNK GATEWAY.............................................................................................................................. 21 3.10.1 Integrated Services Digital Network (ISDN) ............................................................................................... 21 3.10.2 Administer SIP Trunk Gateway to Session Manager .................................................................................. 23 3.10.3 Administer Virtual D-Channel .................................................................................................................... 25 3.10.4 Administer Virtual Super-Loop ................................................................................................................... 27 3.10.5 Administer Virtual SIP Routes .................................................................................................................... 27 3.10.6 Administer Virtual Trunks .......................................................................................................................... 30 3.11. ADMINISTER DIALING PLANS ...................................................................................................................................... 32 3.11.1 Inbound Call – Configure IDC to Receive Calls............................................................................................ 32

4.

AVAYA AURA® SESSION MANAGER CONFIGURATION ............................................................................................34 4.1. 4.2. 4.3. 4.4. 4.5. 4.6. 4.7. 4.8.

5.

AVAYA AURA® SYSTEM MANAGER LOG ON .................................................................................................................. 34 SETUP DOMAINS ...................................................................................................................................................... 35 LOCATION CONFIGURATION ....................................................................................................................................... 36 ADAPTATIONS.......................................................................................................................................................... 36 SIP ENTITIES ........................................................................................................................................................... 37 ENTITY LINKS ........................................................................................................................................................... 38 ROUTING POLICIES ................................................................................................................................................... 39 DIAL PATTERNS ........................................................................................................................................................ 41

AVAYA AURA® SESSION BORDER CONTROLLER (AA-SBC) CONFIGURATION ...........................................................43 5.1. SERVICE PROVIDER PRE-INSTALLATION WIZARD ............................................................................................................. 43 5.2. AA-SBC INSTALLATION ............................................................................................................................................. 50 5.3. AVAYA AURA® SBC CONFIGURATION........................................................................................................................... 50 5.3.1 Configuring a Route to Skype ......................................................................................................................... 51 5.3.2 Configure Timing Between SIP INVITE ............................................................................................................ 52 5.3.3 Configure Re-INVITE to Skype ........................................................................................................................ 53 5.3.4 Blocking the CS1000E MIME Part in the SIP Message .................................................................................... 53 5.3.5 Configure CS1000E to Reduce the SIP Package Size ....................................................................................... 55

6.

SKYPE CONNECT SERVICE CONFIGURATION ...........................................................................................................57 6.1. 6.2. 6.3. 6.4.

REGISTERING FOR SKYPE MANAGER ............................................................................................................................. 57 CREATING A SIP PROFILE ........................................................................................................................................... 59 BUYING CHANNEL SUBSCRIPTIONS ............................................................................................................................... 61 ALLOCATING SKYPE CREDIT FOR OUTBOUND CALLING ..................................................................................................... 62

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6.5. 6.6. 7.

INBOUND CALLS ....................................................................................................................................................... 63 OUTBOUND CALLS .................................................................................................................................................... 64

VERIFICATION STEPS ..............................................................................................................................................65 7.1. 7.2.

VERIFY TWO-WAY AUDIO WITH CS1000E. ................................................................................................................... 65 VERIFY TWO-WAY AUDIO FROM SKYPE CONNECT TO CS100E. ......................................................................................... 67

8.

CONCLUSION ..........................................................................................................................................................68

9.

ADDITIONAL REFERENCES ......................................................................................................................................68

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1. Introduction This document provides a typical network deployment of Communication Server 1000E (hereafter referred to as CS1000E), Avaya Aura® Session Manager, Avaya Aura® Session Border Controller (AASBC) and Skype Connect™ 2.0. During the interoperability testing, basic CS1000E telephony features were tested such as basic calls, call forward, call transfer, conference, CLID displayed, abandoned call ...etc. Interoperability was also tested with Avaya CallPilot® voice mail. In this configuration, the AA-SBC is configured as a SIP gateway endpoint and registers to Skype’s SBC. This document provides a general guideline. Further information, may be obtained from your Avaya support representative. Configuration of CallPilot to work with CS1000E 7.5 is not covered in this document. It is assumed that CallPilot is already part of the setup and was already functioning. No additional configuration of CallPilot was required to interoperate with Skype Connect. For Security purposes IP addresses and PSTN phone numbers have been masked out or altered in this document.

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Figure 1 illustrates the test configuration used during the interoperability testing between the Communication Server 1000E and Skype Connect Service. This configuration is for a single Communication Server1000E deployment.

Figure 1: Network Diagram Configuration

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2. Equipment and Software Validated The following equipment and software were used during the lab testing: Equipment Avaya CS1000E 7.5 (CPPM)

Software Version Call Server (CPPM): 7.50Q DepList 1: Core Issue: 01 2011-09-13 Signaling Server (CPPM): 7.50.17.00 In system patches: 0

Avaya Aura® System Manager Avaya Aura® Session Manager Avaya Aura® Session Border Controller Avaya IP Soft Phone 2050 Avaya IP Phone 1140E Avaya IP Phone 2007 Avaya IP Phone 1120E Avaya Digital Phone M3904 Avaya CallPilot® 600r Skype Connect™

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Release 6.1, Build 6.1.0.0.7345 Release 6.1, Build 6.1.1.0.611023 Release 6.0 3.04.0003 04.01.13.00 (SIP) 0621C8A (UNIStim) 0624C8O (UNIStim) N/A 5.0 R2.0

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3. Avaya Communication Server 1000E Configuration This document assumes that the Avaya CS1000E was properly installed and configured as per the product documents, for more information about how to install, configure and administer CS1000E please refer to Section 9. This section provides only the steps on how to configure the CS1000E to interoperate with Session Manager, AA-SBC and Skype Connect to place and receive PSTN calls over SIP trunks.

3.1. Login to CS1000E Element Manager (EM) Open an instance of a web browser and connect to the UCM GUI at the following address: http:// or http://. Login using an appropriate Username and Password. The Unified Communications Management screen is then displayed. Click on the element Name of the CS1000E Element Manager as in Figure 2.

Figure 2 – Unified Communications Management page

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The CS1000E Element Manager System Overview page is then displayed as in Figure 3.

Figure 3 – Element Manager System Overview page

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3.2. Login to CS1000E CLI (CoRes) Start by doing an SSH to the IP address of the SSG or Signaling Server with the admin account, then run the command “cslogin” and login with the admin account. Figure 4 is a capture of the CS100E output. login as: admin Avaya Inc. Linux Base 7.50 The software and data stored on this system are the property of, or licensed to, Avaya Inc. and are lawfully available only to authorized users for approved purposes. Unauthorized access to any software or data on this system is strictly prohibited and punishable under appropriate laws. If you are not an authorized user then do not try to login. This system may be monitored for operational purposes at any time. [email protected]'s password: logi admin PASS? The software and data stored on this system are the property of, or licensed to, Avaya Inc. and are lawfully available only to authorized users for approved purposes. Unauthorized access to any software or data on this system is strictly prohibited and punishable under appropriate laws. If you are not an authorized user then logout immediately. This system may be monitored for operational purposes at any time. OVL000

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Figure 4 – CLI Login

3.3. Obtain Node IP Address These Application Notes assume that the basic configuration has already been administered and a Node has already been created. This section describes the steps to obtain the Node ID of the CS1000E IP network to be used with Skype. For further information on Avaya Communications Server 1000E, please consult reference in Section 9. From EM, Select System
IP Network
Nodes: Servers, Media Cards and then click on the Node ID of your CS1000E Element.

Figure 5 – IP Telephony Nodes page The Node Details screen is then displayed with the IP address of the CS1000E node. The Node IP Address is a virtual address which corresponds to the TLAN IP address of the Signaling Server, SIP Signaling Gateway.

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Figure 6 –Node Details page

3.4. Administer Terminal Proxy Server (TPS) On the Node Details page as in Figure 6, scroll down and select Terminal Proxy Server (TPS).

Figure 7 –Node Details page - TPS Check the UNIStim Line Terminal Proxy Server and then click Save as in Figure 8.

Figure 8 – TPS Configuration Details

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3.5. Administer Quality of Service (QoS) From the Node Details page as shown in Figure 9, click on Quality of Service (QoS).

Figure 9 –Node Details page – QoS The default Diffserv values are correct in the below figure. Click Save.

Figure 10 – QoS Configuration Details

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3.6. Synchronize the New Configuration On the Node Details screen as in Figure11, click on Save.

Figure 11 – Synchronize the new configuration – Save The Node Saved screen is then displayed. Click Transfer Now…

Figure 12 – Synchronize the new configuration - Transfer

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The Synchronize Configuration Files screen is now displayed. Select the Signaling Server and click on Start Sync.

Figure 13 – Synchronize the new configuration – Start Sync When the synchronization completes, Select the Signaling Server and click on Restart Applications.

Figure 14 – Synchronize the new configuration – Restart Applications

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3.7. Enable Voice Codecs for the Node From the Left pane of Element Manager, select IP Network
Nodes: Servers, Media Cards (not shown). In the IP Telephony Nodes screen, select the Node ID of this CS1000E system (not shown). On the Node Details screen, click on Voice Gateway (VGW) and Codecs.

Figure 15 – Node Details - Voice Gateway and Codecs In the next screen scroll down the parameters box and check the desired codecs under Voice Codecs. Note that G.729 is checked. Click on Save.

Figure 16 –Voice Gateway and Codec Configuration Details TBH; Reviewed: SPOC 1/16/2012

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Now synchronize the new configuration as in Section 3.6.

3.8. Enable Voice Codecs on Media Gateways In Element Manager select IP Network
Media Gateways from the left pane. Then click MGC from the right pane as shown in Figure 17.

Figure 17 – Media Gateways page

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In the next screen select VGW and IP phone codec profile to expand the section, scroll down to the parameters box and check Codec G729A.

Figure 18 – Media Gateways Configuration Details page

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Then scroll down and click on Save as shown in the next figure.

Figure 19 – Media Gateways Configuration Details page – Save Now reboot the MGC Card for changes to take effect. Select the card and click on Reboot.

Figure 20 – Media Gateways Card Reboot

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3.9. Zone and Bandwidth Management This section describes the steps to create 2 zones: one for IP sets and another one for a SIP Trunk.

3.9.1 Create a Zone for IP phones The following figures show how to configure a zone for IP sets and bandwidth management. If it does not already exist, please click “Add...” button to create a zone for IP sets. The bandwidth strategy can be adjusted to preference. From Element Manager Select IP Network
Zones from the left pane, then click Bandwidth Zones.

Figure 21 –Zones page The Bandwidth Zones screen is then displayed. Select “Add...” to add a new Zone, or select an existing one and click “Edit” to view the current configuration. In this example Zone 1 was used and it was already configured.

Figure 22 –Bandwidth Zones page

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On this screen select “Zone Basic Property and Bandwidth Management”.

Figure 23 –Edit Bandwidth Zone Select MO for Zone Intent (ZBRN) and click Submit. - INTRA_STGY: Codec configuration for local calls. - INTER_STGY: Codec configuration for the calls over trunk. - BQ: G711 is first choice and G729 is second choice. - BB: G729 is first choice and G711 is second choice. - MO: is used for IP phones, VGW ....etc - VTRK: is used for virtual trunk.

Figure 24 –Bandwidth Management configuration details page – IP phone

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3.9.2 Create a Zone for Virtual SIP Trunk Follow Section 3.9.1 to create a zone for virtual trunk. The difference is in Zone Intent (ZBRN) field. Select VTRK for virtual trunk and then click Submit.

Figure 25 –Bandwidth Management configuration details page –virtual trunk

3.10. Administer SIP Trunk Gateway This section describes the steps for establishing a SIP connection between CS1000E and Session Manager with Integrated Services Digital Network (ISDN) configuration.

3.10.1 Integrated Services Digital Network (ISDN) In Element Manager, select Customers in the left pane. The Customers screen is displayed (not shown). Click the link associated with the appropriate customer, in this case 00. The system can support more than one customer with different network settings and options.

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The Customer 00 screen is displayed next. Select Feature Packages.

Figure 26 –Customer - Feature Packages The screen is then updated with a listing of the Feature Packages as below (not all features are shown). Select Integrated Services Digital Network to edit its parameters. The screen is updated with parameters for Integrated Services Digital Network. Check the Integrated Services Digital Network checkbox, and retain the default values for all remaining fields. Scroll down to the bottom of the screen, and click Save (not shown).

Figure 27 –Customer – ISDN configuration page

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3.10.2 Administer SIP Trunk Gateway to Session Manager In Element Manager, select IP Network
Nodes: Servers, Media Cards configuration from the left pane, and in the IP Telephony Nodes screen, select the Node ID of this CS1000E system (not shown). The Node Details screen is then displayed as below. On the Node Details screen, select Gateway (SIPGw).

Figure 28 – Node Details – Gateway configuration Under General tab of the Virtual Trunk Gateway Configuration Details screen, enter the following values for the specified fields, and retain the default values for the remaining fields. - Vtrk Gateway Application: Select SIPGw - SIP Domain Name: provided when user creates a SIP profile on Skype. - Local SIP Port: provided when user creates a SIP profile on Skype. - Gateway endpoint name: Name for this endpoint.

Figure 29 – Virtual Trunk Gateway Configuration Details TBH; Reviewed: SPOC 1/16/2012

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Click on SIP Gateway Settings tab, under Proxy Server Route 1, enter the following values for the specified fields, and retain the default values for the remaining fields. Primary TLAN IP Address: This is the IP address of the Session Manager. Port: 5060 Transport Protocol: UDP

Figure 30 – Virtual Trunk Gateway Configuration Details Next scroll down in the parameters box to the SIP URI Map section. Under the Public E.164 Domain Names section: - Special Number: leave this SIP URI field blank - Unknown: leave this SIP URI field blank Under the Private Domain Names section: - Special Number: leave this SIP URI field blank - Vacant number: leave this SIP URI field blank - Unknown: leave this SIP URI field blank

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The remaining fields can be left at their default values. Click on Save.

Figure 31 – Virtual Trunk Gateway Configuration Details Synchronize the new configuration as in Section 3.6.

3.10.3 Administer Virtual D-Channel In Element Manager select Routes and Trunks
D-Channels from the left pane to display the DChannels screen. In the Choose a D-Channel Number field, select an available D-channel from the drop-down list. Click “to Add”.

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Figure 32 – D-Channels The D-Channels 10 Property Configuration screen is displayed next. Enter the following values for the specified fields, and retain the default values for the remaining fields. D channel Card Type (CTYP): Designator (DES): Interface type for D-channel (IFC):

D-Channel is over IP (DCIP) A descriptive name Meridian Meridian1 (SL1)

Figure 33 – D-Channels configuration details Next click on Basic Options (BSCOPT) and click on the Edit button beside Remote Capabilities.

Figure 34 – D-Channels configuration details

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Then enable ND2 and MWI if CS1000E hosted voice mail will be used.

Figure 35 – D-Channels configuration details Click on Return – Remote Capabilities (not shown), and then click Submit (not shown).

3.10.4 Administer Virtual Super-Loop In Element Manager Select System
Core Equipments
Superloops from the left pane to display the Superloops screen. If a Superloop does not exist, click on the “Add” button to create a new one.

Figure 36 – Administer Virtual Super-Loop page.

3.10.5

Administer Virtual SIP Routes

Select Routes and Trunks
Routes and Trunks from the left pane to display the Routes and Trunks screen. Next to the applicable Customer row, click Add route.

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Figure 37 – Add route Displayed in the next figure is the Edit output for Route 10 that was already configured on this System. It shows the fields that would be added in a new route. When adding a new route, scroll down until the Basic Configuration section is displayed and enter the following values for the specified fields. Retain the default values for the remaining fields. • • • • •

Route Number (ROUT): Select an available route number. Designator field for trunk (DES): Add descriptive text. Trunk Type (TKTP): TIE trunk data block (TIE) Incoming and Outgoing trunk (ICOG): Incoming and Outgoing (IAO) Access Code for the trunk route (ACOD): Select an available access code.

Check the field The route is for a virtual trunk route (VTRK), to enable four additional fields to appear. For the Zone for codec selection and bandwidth management (ZONE) field, enter the zone number that was created in Section 3.9.2. For the Node ID of signaling server of this route (NODE) field, enter the node number that was created or obtained in Section 3.3 Select SIP (SIP) from the drop-down list for the Protocol ID for the route (PCID) field. Check the Integrated Services Digital Network option (ISDN) checkbox to enable additional fields to appear. Enter the following values for the specified fields, and retain the default values for the remaining fields. Scroll down to the bottom of the screen. • • • • •

Mode of operation (MODE): Route uses ISDN Signaling Link (ISLD) D channel number (DCH): D-Channel number created in Section 3.10.3 Network calling name allowed (NCNA): Check the field. Network call redirection (NCRD): Check the field. Insert ESN access code (INAC): Check the field.

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Figure 38 – Route configuration details Then click Submit (not shown).

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3.10.6 Administer Virtual Trunks After clicking on Submit, the Routes and Trunks screen is displayed and updated with the newly added route. Click the Add trunk button next to the newly added route.

Figure 39 – Route and Trunks In this example the Customer 0, Route 10, Trunk type TIE trunk data block screen is displayed. Enter the following values for the specified fields and retain the default values for the remaining fields. Then disable Media Security (sRTP) at the trunk level by editing the Class of Service (CLS) at the bottom basic trunk configuration page shown in Figure 41. The Multiple trunk input number field may be used to add multiple trunks in a single operation, or repeat the operation for each trunk. In the sample configuration, four trunks were created. • • • • • • • • • •

Trunk data block: IP Trunk (IPTI) Terminal Number: Available terminal number Designator field for trunk: A descriptive text Extended Trunk : VTRK Member number: starting member Card Density: Octal Density (8D) Start arrangement Incoming: Immediate (IMM) Start arrangement Outgoing: Immediate (IMM) Trunk Group Access Restriction: Desired trunk group access restriction level Channel ID for this trunk: An available starting channel ID

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Figure 40 – New Trunk configuration details

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For Media Security, select MSNV. Enter the remaining values for the specified fields as shown in the following figure. Scroll down to the bottom of the screen and click Return Class of Service and then click Save (not shown).

Figure 41 – Class of Service configuration details page.

3.11. Administer Dialing Plans In this solution most of the Dialing Plan will be handled by the routing configuration with Session Manager. On the CS1000E there will be configuration to map the online numbers from Skype to extension numbers on the system.

3.11.1 Inbound Call – Configure IDC to Receive Calls To receive the call to the online number 1-202-470-xxx on a phone having DN 58001, one can configure IDC as follows:

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Configure FCR in Customer by ld 15. This section prints FCR configuration details. >ld 21 PT1000 REQ: prt TYPE: fcr TYPE FCR_DATA CUST 0 TYPE FCR_DATA CUST 00 NFCR YES MAXT 100 OCB1 255 OCB2 255 OCB3 255 IDCA YES DCMX 100

Configure IDC with ld 49. This section prints IDC configuration details. >ld 49 DGT000 MEM AVAIL: (U/P): 103093188 USED U P: 481334 77827 DISK SPACE NEEDED: 60 KBYTES REQ prt TYPE idc CUST 0 DCNO

TOT: 103652349

DCNO 0 All Programs > SP Pre-installation Wizard > Run SP Pre-installation Wizard. The SP Pre-installation Wizard will be run in a web browser. Under template, select SBCT from the drop down list, and then click Next Step as shown in the following figure.

Figure 54: SP Pre-installation Wizard - Select a template

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The next step is Network Settings. Configure the internal interface of the AA-SBC to connect to the CS1000E network as shown in the next figure. The IP addresses used here are for example only. - Domain0 IP Address: IP address of System Platform system domain 0, e.g. 10.10.97.211 - CDom IP Address: IP address of System Platform console domain, e.g. 10.10.97.212 - Gateway IP Address: 10.10.97.193 - Network Mask: 255.255.255.192 - SBC: IP address of AA-SBC internal interface, e.g. 10.10.97.213 - Hostname: AASBC - Domain: sip.skype.com

Figure 55: SP Pre-installation Wizard - Network Settings

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The next step is Service logins for SBC (optional). Define passwords for accounts craft, init and dadmin as shown below. Then click on Next Step.

Figure 56: SP Pre-installation Wizard - Services logins for SBC (optional) The next page is for VPN Access configuration. The SIP Trunk connecting to Skype is not behind a VPN, so select No (VPN mode is disabled) and then click Next Step.

Figure 57: SP Pre-installation Wizard - VPN Access

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The next step is to configure Session Border Controller Data. Define the FQDN of Skype’s SBC used for SIP signaling and for RTP as shown in the following figure. The IP addresses used here are for example only. SIP Service Provider Data: - Service Provider: Skype - Port: 5060 - FQDN1: sip.skype.com SBC Network Data: - Public: IP address of AA-SBC to connect to Skype Connect e.g. 10.10.98.107 - Net Mask: 255.255.255.224 - Gateway: 10.10.98.97 Enterprise SIP Server: - SIP Domain: sip.skype.com - IP Address1: the IP address of Session Manager server (please refer to Section 3.10.2) - Transport1: UDP Skype Username: - Username: Enter the username supplied by Skype. - Password: Enter the password supplied by Skype. - Re-type password: Enter the password supplied by Skype. Note: Service Provider parameter set to Skype means that there is a basic pre-configuration which have been set for the AA-SBC to interface with Skype.

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Figure 58: SP Pre-installation Wizard - Session Border Controller Data

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The next page is the Summary. It gives an overview of the configuration as shown in Figure 59. Scroll down then click on the Next Step (not shown)

Figure 59: SP Pre-installation Wizard - Summary

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The next page gives the option to save the configuration as an EPW file (Electronic Pre-installation Worksheet). Click Accept then Save EPW file as shown in figure below.

Figure 60: SP Pre-installation Wizard - Save Save the EPW file for use in the installation of the AA_SBC.

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5.2. AA-SBC Installation To install Avaya Aura® SBC, follow the installation guide provided on http://support.avaya.com. The installation wizard (not shown) is an automation tool. During installation, the EPW file is needed. Please use the EPW file created in Section 5.1 to upload to the wizard. After the installation is complete, continue to configure the AA-SBC as described in Section 5.3.

5.3. Avaya Aura® SBC Configuration This section describes how to configure the additional settings on the AA-SBC to allow the SIP trunk of the CS1000E/SESSION MANAGER to interoperate with Skype service provider. To login to AA-SBC, https://SBCIPAddress/. Enter username as craft and appropriate password to login.

Figure 61: Login to AA-SBC During installation, the information in the EPW file was used to populate the entry “server Telco1”, which is information of Skype’s SBC for SIP Trunking and the entry “server PBX1” which is the information for the Session Manager.

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5.3.1 Configuring a Route to Skype This section shows how to add a route on the outside interface of AA-SBC to Skype. Select Configuration tab
cluster
box:DevSBC3.sip.skype.com
interface eth2
ip ouside
routing. Click on Add Route. Enter the following parameters: - Route-name: ToSkype - Gateway: 10.10.98.97 - Destination: Type: host - Address: 10.9.161.164. This is Skype’s SBC server IP address. Click on Create button to add new route (not shown). Figure 62 bellow shows newly added route, ToSkype.

Figure 62: Route to Skype Configuration

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5.3.2 Configure Timing Between SIP INVITEs This shows how to extend the time between consecutive SIP INVITE messages to send to Skype. Select Configuration tab
vsp. On the vsp page, click on Show Advance button to expand the vsp page options/parameters. Under the general
static stack-settings. Enter the t1 value to be 10000 = 10 seconds as shown in Figure 63.

Figure 63: t1 Time Configuration

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5.3.3 Configure Re-INVITE to Skype This session shows the configuration on the AA-SBC to send additional SIP INVITEs to Skype in case the initial INVITE is lost.. Select Configuration tab
vsp
enterprise
server
sip-gateway Telco
vsp\sessionconfig-pool\entry ToTelco. Click on Show Advance button
sip-setting. Under the other properties, enter value of 5 to the max-retransmissions parameter as shown in Figure 64.

Figure 64: Retransmissions of SIP INVITE To Skype

5.3.4 Blocking the CS1000E MIME Part in the SIP Message This section shows how to configure AA-SBC to remove the CS1000E proprietary MIME body part in the SIP message from CS1000E to Skype. Select Configuration tab
vsp
session-config-pool
entry ToTelco. Under the media session, select bodypart-type. Click on Add allowed-body-part. Enter the following parameters: - bodypart-type: application - application-sub-type: sdp Click on Create button to add the bodypart-type as shown in Figure 65.

Figure 65: Create allowed-bodypart-type

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Click on Add blocked-body-part. Enter the following parameters: - bodypart-type: application - application-sub-type: any Click on Create button to add the bodypart-type as shown in Figure 66.

Figure 66: Create blocked-body-part

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5.3.5 Configure CS1000E to Reduce the SIP Package Size The Skype server limits the incoming package size to its network to 1500 bytes. Therefore, the configuration of AA-SBC needs to be set up to strip off the unnecessary header and body of the SIP message to make it acceptable to the Skype system. Select Configuration tab
vsp
session-config-pool
entry ToTelco. Click on Show Advanced button. Under the header section, header-settings and block-header, click on Edit block-header. Enter the x-nt-corr-id, click on Add button. Then enter x-nt-e164-clid and click OK button as shown in Figure 67.

Figure 67: Add Block-header x-nt

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After adding the blocked-header, the header-settings will be as shown in Figure 68.

Figure 68: SIP header-settings Similar to the header-settings, inbound-header-settings also need to be configured to block the following headers: x-nt-corr-id, x-nt-e164-clid and P-location. Figure 69 will show the blockedheaders.

Figure 69: Inbound header-settings

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6. Skype Connect Service Configuration 6.1. Registering for Skype Manager Before getting started with Skype Connect, it’s necessary to register to use Skype Manager. To do so, visit the Skype for Business website at skype.com/business and follow the instructions below. Additional documentation for Skype Connect and Skype Manager can be found at http://www.skype.com Step 1: Select Business then click on Sign in under Skype Manager (not shown). Step 2: Click on Sign up now.

Figure 70 – Skype Manager Account login and set up

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Step 3: Select No, I don’t have a Skype account and fill out the form to setup an account.

Figure 71 – New Skype Manager Account form Step 4: After the setup of Skype manager completes, login to Skype Manager, and then click on Feature to create a SIP profile.

Figure 72 – Skype Manager home

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6.2. Creating a SIP Profile To use Skype Connect, you need to create at least one SIP Profile in Skype Manager. Creating a SIP Profile is straightforward. Note: regarding new Skype manager account, user needs to download Skype software
Login to Skype client so Skype Manager will be allowed to create Skype profile. A SIP Profile comprises six elements: 1. SIP Credentials: these are the login details needed by AA-SBC to connect to Skype. 2. Skype Credit: to pay for outbound calls, if required. 3. Business accounts: for receiving calls from Skype in CS1000E, if required. 4. Online Numbers: so people can call from landlines and mobiles, which will be directed to CS1000E, if required. 5. A monthly Channel Subscription: which determines the number of concurrent calling channels you want to use with Skype Connect. 6. Your preferred Caller ID: which can be any Online Number you have associated with your SIP Profile or a landline number your company is authorized to use once it has completed Skype's company verification process. A SIP Profile may be configured for outgoing calls: To enable a SIP Profile for outgoing calls, you must allocate Skype Credit to this SIP Profile from your Skype Manager. If the Caller ID option has been set up for this SIP Profile, then it may be used for the outgoing calls going out from a phone without Caller ID configured. A SIP Profile may also be configured for incoming calls: To enable incoming calls, choose one or more business accounts to be associated with the SIP Profile. You can also purchase Online Numbers and associate them to your SIP Profile. Incoming calls to those business accounts or Online Numbers will be directed to your SIP Profile. Do not use the same SIP Profile on more than one CS1000E. Multiple SIP Profiles can be created in Skype Manager. This can be useful if: - Multiple CS1000E and AA-SBCs are configured in the network. This would allow separate SIP Profiles to be created for each System. - Managing outbound calling costs from different parts of your organization is required.

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After being created, there is now a SIP Profile as follows:

Figure 73 – Skype Profile page

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Click on Authentication details to view the SIP User number and Password. This will be used in the setup of the AA-SBC so it can authenticate with Skype Connect.

Figure 74 – Skype Authentication details

6.3. Buying Channel Subscriptions Channel subscriptions are the amount of concurrent calling channels you would like to use with your SIP Profile and are charged on a monthly basis. Skype Connect supports up to 300 simultaneous calling channels, enabling up to 300 concurrent conversations. Please follow Buying Channel Subscriptions, in the Skype user guide, available online.

Figure 75 – Skype Calling Channels

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6.4. Allocating Skype Credit for Outbound Calling In Skype Manager, Skype calls are normally paid for by Skype Credit being allocated to business accounts. However, Skype Connect is different because Skype Credit can be allocated directly to a SIP Profile. Skype Credit allocated to a SIP Profile is used only to pay for outbound calls. Channel Subscriptions and fees for Online Numbers are paid directly from your Skype Manager.

Figure 76 – Skype Outgoing Call Credit

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6.5. Inbound Calls To set up a SIP Profile for inbound calling, you must either: - Associate one or more business accounts with the SIP Profile. - Or, assign one or more Online Numbers to the SIP Profile.

Figure 77 – Skype online numbers

Figure 78 – Skype Business Account

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6.6. Outbound Calls For any SIP Profile, there are three Caller ID options: - Set Caller ID to be a landline number used by your company (provided your company has been verified by Skype). - Set Caller ID to be any Online Number that you have assigned to the SIP Profile. - Choose not to present a Caller ID.

Figure 79 – Skype Business Account

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7. Verification Steps This section includes some steps that can be followed to verify the solution is working.

7.1. Verify Two-way Audio with CS1000E. Verify that IP phones, digital, analog register successfully as shown below: Verify status of IP phone registered: [admin@cppm1 ~]$ isetShow === TPS === Set Information --------------IP Address NAT Model Name --------- ------------ ------------ -----------10.10.98.51 1140E IP Deskphone 10.10.98.29 2007 Phase 2 IP Deskphone 10.10.98.9 1120E IP Deskphone 10.10.98.30 1110 IP Deskphone

Type RegType State -------------------- ---1140 Regular online 2007 Regular online 1120 Regular online 1110 Regular online

Verify status of digital phone registered: >ld 32 NPR000 .stat 4 0 02 00 = UNIT 00 = IDLE 01 = UNIT 01 = IDLE

(3904) (3904)

Verify status of Analog phone registered: >ld 32 .stat 4 0 03 00 = UNIT 00 = IDLE 01 = UNIT 01 = IDLE

(L500) (L500)

Verify the following basic calls in local CS1000E: IP phone------------------------call--------------------IP phone IP phone -----------------------call--------------------SIP Line Client IP Phone -----------------------call--------------------Analog/Fax phone IP Phone -----------------------call--------------------Digital phone SIP Line Client----------------call--------------------Analog phone SIP Line Client----------------call--------------------Digital Phone TBH; Reviewed: SPOC 1/16/2012

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Analog phone------------------call--------------------Digital Phone User can verify the same for calls from opposite direction. Verify that calls are established with two-way voice path and busy status under CS1000E call server as below: Verify status of IP phones which are busy: [admin@cppm1 ~]$ isetShow === TPS === Set Information --------------IP Address NAT Model Name --------- ------------ ------------ -----------10.10.98.51 1140E IP Deskphone 10.10.98.29 2007 Phase 2 IP Deskphone 10.10.98.9 1120E IP Deskphone 10.10.98.30 1110 IP Deskphone

Type RegType State -------------------- ---1140 Regular busy 2007 Regular busy 1120 Regular busy 1110 Regular busy

Verify status of digital phone is busy: >ld 32 NPR000 .stat 4 0 02 00 = UNIT 00 = BUSY 01 = UNIT 01 = BUSY

(3904) (3904)

Verify status analog phone is busy: >ld 32 .stat 4 0 03 00 = UNIT 00 = BUSY 01 = UNIT 01 = BUSY

(L500) (L500)

Verify status of voice gateway if calls are established between IP phone/SIP line Clients to Analog/Digital phones or call to voice message: >ld 32 .stat 4 0 11 00 = UNIT 00 01 = UNIT 01 02 = UNIT 02 03 = UNIT 03

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= = = =

BUSY BUSY BUSY BUSY

(TRK)(IPTN (TRK)(IPTN (TRK)(IPTN (TRK)(IPTN

REG) REG) REG) REG)

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During the call, use pcap tool (ethereal/wireshark) at the TLAN of the media gateway card. Verify that RTP streams are going for calls related to analog, digital or voice message.

7.2. Verify Two-way Audio from Skype Connect to CS100E. Verify basic call between PSTN phones and Avaya phones. Verify Codec, SIP trunk status when call is established under CS1000E call server by tracing DID number. >ld 80 .trac 0 58001 ACTIVE

VTN 096 1 00 01

ORIG VTN 100 1 00 00 VTRK IPTI RMBR 10 1 INCOMING VOIP GW CALL FAR-END SIP SIGNALLING IP: 10.10.97.213 FAR-END MEDIA ENDPOINT IP: 10.10.97.213 PORT: 23258 FAR-END VendorID: SipGW 4 AVAYA-SM-6.1.1.0.611023 TERM VTN 096 1 00 01 KEY 0 SCR MARP CUST 0 DN 58001 TYPE SLUEXT SIGNALLING ENCRYPTION: INSEC UEXT PROXY VTN 100 0 01 31 VTRK IPTI RMBR 11 32 OUTGOING VOIP GW CALL FAR-END SIP SIGNALLING IP: 0.0.0.0 FAR-END MEDIA ENDPOINT IP: 0.0.0.0 PORT: 0 FAR-END VendorID: Avaya IP Phone 1140E (SIP1140e.04.01.13.00) MEDIA PROFILE: CODEC G.729A NO-LAW PAYLOAD 20 ms VAD OFF RFC2833: RXPT N/A TXPT N/A DIAL DN 58001 MAIN_PM ESTD TALKSLOT ORIG 88 TERM 61 EES_DATA: NONE QUEU NONE CALL ID 82 17060 ---- ISDN CALL REF # BEARER CAP HLC = CALL STATE CALLING NO CALLED NO

ISL CALL (ORIG) ---= 385 = VOICE = = =

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10 ACTIVE 1613967xxxx NUM_PLAN:UNKNOWN 1202470xxxx NUM_PLAN:UNKNOWN

TON:UNKNOWN TON:UNKNOWN

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ESN:UNKNOWN ESN:UNKNOWN

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Verify that the SIP Trunk is released when the DID number is released by tracing that DID number under CS1000E call server. >ld 80 .trac 0 58001 IDLE VTN 096 1 00 01

8. Conclusion Telephony features such as basic calls, call forward, call transfer, conference, CLID displayed, abandoned calls, Avaya CallPilot® voice mail, etc test cases have passed. Skype Connect 2.0 is considered to be interoperable with Communication Server 1000E release 7.5, Avaya CallPilot®, Avaya Aura® Session Manager and Avaya Aura® Session Border Controller.

9. Additional References Product documentation for Avaya products may be found at: https://support.avaya.com. [1] Network Routing Service Fundamentals Avaya Communication Server 1000, Release 7.5, Revision 03.02, Nov 2010, Document Number NN43001-130 [2] IP Peer Networking Installation and Commissioning, Avaya Communication Server 1000, Release 7.5, Revision 05.08, April 2011, Document Number NN43001-313, [3] Unified Communications Management Common Services Fundamentals Avaya Communication Server 1000, Release 7.5, Revision 05.08, January 2011, Document Number NN43001-116 [4] SIP Line Fundamentals, Avaya Communication Server 1000Release 7.5, Revision 03.03, November 2010, Document Number NN43001-508 [5] Dialing Plans Reference, Avaya Communication Server 1000, Release 7.5, Revision 05.02, November 2010, Document Number NN43001-283 Skype Connect and Skype Manager Documents are located at http://www.skype.com.

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©2012 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes.

Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at [email protected]

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