EarthLink Business SIP Trunking. Avaya IPO IP PBX Customer Configuration Guide

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1 EarthLink Business SIP Trunking Avaya IPO IP PBX Customer Configuration Guide

2 Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011 Original Document Draft Dantley Thompson 1.0 8/30/2011 Distributed draft for Peer Review Dantley Thompon AUTHOR: Dantley Thompson EarthLink Engineering 2

3 Table of Contents Document Purpose 4 Product Summary 4 Network Architecture and Design 5 Media Attributes and Codec Negotiation 7 Codec Support 7 G.711u 7 G.729a 7 Packetization Time 7 DTMF Support 7 Fax and Modem Support Requirements 8 North American Numbering Plan Format 8 Quality of Service Policy 8 EarthLink SIP Trunking to IP PBX Interoperability 9 Adtran Software Version Tested 9 IP PBX Software Version Tested 9 EarthLink Open Issues & Non-Supported Features 9 Avaya IPO Open Issues & Non-Supported Features 9 IP PBX Configuration for EarthLink SIP Trunking with Adtran SIP Proxy 10 Avaya IPO IP PBX Configuration 10 Product Support and Contact Information 26 EarthLink SIP Trunking Turn-up Testing Procedure 27 3

4 Document Purpose The purpose of this document is to provide a detailed technical description and best practices for successful implementation of the EarthLink SIP Trunking Product for the Avaya IPO with the Adtran SIP Proxy. This document provides information relative to the overall network topology as well as definition and configuration standards for each device associated with the product. Also described within this document are product guidelines and product limitations. This document is to serve as product reference and guide to EarthLink Customers. Product Summary The EarthLink Business SIP Trunking product is a complete VoIP (Voice over IP) solution based on the SIP (Session Initiation Protocol) signaling protocol. The SIP Protocol is responsible for set-up and tear-down of voice calls and overall feature and functionality. The SIP Trunking product can be offered as an overlay to several of EarthLink s existing products such as Internet and MPLS based products. EarthLink Business SIP Trunking solution will be served off a MetaSphere Call Feature Server (CFS) fronted by an ACME packet SBC (Session Border Controller). The CFS acts as the centerpiece for call control and feature interaction. The EarthLink Business SIP Trunking Product will primarily use Adtran CPE (Customer Premise Equipment) configured as a SIP Proxy. The MetaSphere CFS Platform is a geo-redundant, high availability solution and serves as the primary element in EarthLink s Hosted Voice and SIP Trunking Product families. In addition to the basic call control, advanced call routing functionality is available with EarthLink s SIP Trunking product with MetaSphere Enhanced Application Server (EAS) Platform which consists of multiple applications and servers integrated into high availability solution. The Acme Packet SBC masks private to public IP Address space to provide a safe and secure means of communication between the SIP Server and IP PBX. All SIP traffic destined to, or originating from the MetaSphere CFS, traverses through the ACME Packet SBC. The same policy relates to the CPE device installed at the customer premise. The Acme Packet SBC and Adtran CPE, utilizing SIP Proxy, both resolve NAT (Network Address Translation) related issues exposed when SIP traffic passes through a firewall. 4

5 SPEAKER HEADSET PHONE/EXIT PAGE LEFT ABC DEF 0SW IP MUTE * GHI JKL 4 5 PQRS TUV SPEAKER HEADSET # WXYZ 9 PHONE/EXIT PAGE LEFT MNO 6 ABC DEF 0SW IP MUTE GHI JKL MNO PQRS TUV 7 8 * 0 # WXYZ 9 REDIAL REDIAL PAGE RIGHT DROP PAGE RIGHT DROP OPTIONS TRANSFER CONFERENCE OPTIONS TRANSFER CONFERENCE OK ME SS AGE P HONE CON TAC TS MENU C AL L L AB C D EF GHI J KL MNO FORWARD P QRS T UV WXYZ HEADSET VOLU ME 0 # * SPEAKER MUTE HOLD HOLD [ T1/Ethernet Implementation Guide Network Architecture and Design The EarthLink Business SIP Trunking solution consists of several key network elements that are connected to the existing core routing infrastructure. The MetaSwitch Call Feature Server, IP/TDM Gateways, and Acme Packet SBC s are geographically diverse with reach-ability at both layer two and layer three to provide failover capability and redundancy. Split-Horizon DNS servers are used to resolve the SIP domain to the appropriate regional SBC. Adtran CPE will be connected to the EarthLink network via the traditional means such as Ethernet, PPP (Point to Point Protocol), or MLPPP (Multilink Point-to Point Protocol). T1, or bonded T1 services MUST be provisioned to either the Adtran TA5000 or directly to the Cisco 7609 (Edge Router) to allow for proper QoS (Quality of Service) behavior. As mentioned earlier in this document, EarthLink s SIP Trunking product can be offered as an overlay to other Earthlink Products and Services. The first diagram below provides a high level look at the primary components that complete the SIP Trunking product. The second diagram provides a detailed layout for the connections between the Adtran CPE and Customers IP PBX. EarthLink SIP Trunking Test Bed Information and Configuration Metaswitch Application Server: Metaswitch CFS ver.7.3 ACME Packet: Net-Net 4250 ver.sc6.1.0 MR-7 Patch 4 (Build 855) Adtran CPE Type: Adtran TA916e A4.08 (SIP Stateful Proxy) PBY Type: Avaya IPOffice ver.6.0(8) Metaswitch and ACME configuration DID Number Range PBX Phone Assignment Sip Binding = Avaya IPOffice (AA_VM) PBX IP = (Digital) Acme IP (OUTSIDE/PUBLIC) = (ANTNLAB ACME PUBLIC) (Digital) ACME IP (INSIDE/PRIVATE) = (Analog Fax) ACME SESSION AGENT = lab-avaya-ipo (IP) Media Gateway model = Avaya IPO Use DN for Identification = TRUE Authentication required = FALSE Sip user name = Sip password = deltacom Sip domain name = static.voiplab.deltacom.net Trusted = TRUE PBX = DNIS = 10 digit Charge Number required = TRUE Charge Number = Calling Pty # = NA DID's = SIP Figure 1-EarthLink SIP Trunking-Network Topology 5

6 Figure 2-EarthLink SIP Trunking-Connections from Adtran CPE to IP PBX 6

7 Media Attributes and Codec Negotiation Codec Support A voice codec (coder/decoder) is a hardware/software module/algorithm that takes an analog or digital voice stream and encodes it into an IP packet. For the EarthLink Business SIP Trunking Product, we currently support two (2) of the most common codec s utilized in the continental United States, G.711u and G.729a. The preferred codec offered by EarthLink in the default configuration model is G.711u, then G.729a. Basically this means that the call will negotiate using the G.711u codec first, as long as the terminating end sends G.711u as the first or primary offered codec. The paragraphs below provide more detailed information related to the codec s and other requirements associated with proper negotiation of the media/rtp. G.711u G.711u is the most common uncompressed audio codec deployed in the US. Because it is uncompressed, it supports the highest level of quality for the call. Typically the G.711u consumes 90Kbps-100Kbps per call. The standard sampling rate of 8kHz is used for the G.711u codec. G.729a G.729a is the most common codec utilized to support compressed audio utilized in the US. Because it is compressed, it is perceived to have a lower voice quality than that of G.711u, however most people would never be able to tell the difference. Typically the G.729 consumes 30Kbps-40Kbps per call. The standard sampling rate of 8kHz is used for the G.729a codec. Packetization Time Packetization Time determines how often the audio stream is sampled and how often an IP packet is created. The standard packetization times are 10ms, 20ms, 30ms, and 40ms. EarthLink Media Gateway s have been statically configured to use a 20ms packetization time. IP Phones and/or Voice Applications will need to configure their equipment for a 20ms packetization time before audio traffic can be reliably passed across the EarthLink IP Voice network. DTMF Support EarthLink supports the transmission of Dual-Tone Multi-frequency (DTMF) digits through the implementation of RFC2833. This RFC covers the basis of including DTMF digits within the media/rtp path of the call. EarthLink recommends for Customers to configure their IP PBX s and/or Voice Applications to use RFC2833 to allow for DTMF to be passed properly and detected across the EarthLink IP Voice network. 7

8 Fax and Modem Support Requirements Currently, analog devices such as faxes and modems MUST be provisioned using the G.711u codec only. SIP to analog lines are supported as SIP Lines off the Adtran FXS Ports or a Cisco 2102 ATA (Analog Terminal Adapter). The customer may also configure the IP PBX to use analog extensions for faxes and modems. This method can be supported utilizing the G.711u codec only. T.38 is currently not supported. North American Numbering Plan Format Currently, the EarthLink Business Hosted Voice product only supports the North American Numbering Plan Format. A Global Numbering Plan Format, such as E.164, is currently not supported. Quality of Service Policy To ensure the best possible voice quality, EarthLink will mark and match all VoIP traffic related to SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol). EarthLink VoIP and/or Real-Time based appliances and applications are configured to use DSCP (Differentiated Services Code Point) 46 for all signaling traffic (SIP) and DSCP 46 for all Real-Time traffic (RTP) for Layer three priority. The Customers IP PBX MUST also be configured to use DSCP 46 to provide prioritization for SIP and RTP. Marking the DSCP field in the IP packet header will allow for packet classification to be matched and provide priority across EarthLink s network. This also ensures QoS specifications outlined in SLA (Service Level Agreements) can be sufficiently met between EarthLink and the customer. 8

9 EarthLink SIP Trunking to IP PBX Interoperability SIP Trunking interoperability testing was performed between EarthLink and the IP PBX. All phases of the test plan were executed against the actual configuration used in a customer deployment. The information below provides the Adtran and IP PBX software versions tested as well as an issue summary and non-supported elements discovered during compliance testing in the EarthLink Lab. Adtran Software Version Tested Adtran TA908e version A4.08 IP PBX Software Version Tested Avaya IPO version 6.0(8) EarthLink Open Issues & Non-Supported Features Registration is currently not supported for the EarthLink SIP Trunking Product. When the originating calling number in present in the FROM Header, the main billing telephone number or DID belonging to the trunk group must be provided via the PAI (P- Asserted Identity) Header or via the Diversion Header on Call Transfer and Call Forward calls for the call to pass through the Metaswitch and billed correctly. Correct calling name not being sent correctly when on-net calls are made on the same Metaswitch from Business Group to IP PBX. Avaya IPO Open Issues & Non-Supported Features Authorization/Challenge on INVITE without Registration is not supported by the Avaya IPO. Authorization must be turned off/ Unchecked in the Metaswitch. The Avaya IPO is unable to send originating calling number of Call Forwarded Calls to the PSTN over the SIP Trunk. 9

10 IP PBX Configuration for EarthLink SIP Trunking with Adtran SIP Proxy The steps below provide a step by step guide for configuration of the Avaya IPO (IP Office) IP PBX for the EarthLink SIP Trunking Product. Basic configuration of the Avaya IPO should be complete and the Avaya IPO MUST be connected to the LAN prior to configuring the system for SIP Trunking. Avaya IPO IP PBX Configuration The following screen-shots below are using Avaya s IP Office Manager Application version 9.0(5). These steps outline the configuration of the Avaya IPO version 6.0(8) to work with EarthLink s SIP Trunking product with the Adtran SIP Proxy. For more detailed information the Avaya IP Office Knowledgebase application can be used. Avaya IPO SIP Trunk License Verification Confirm the Avaya IPO is licensed for SIP Trunk Channels. The number of Instances will determine how many concurrent calls can be handled simultaneously. From the Navigation Pane, select License Select SIP Trunk Channels to verify a License if Valid and the number of Instances for number of concurrent calls 10

11 Avaya IPO System Configuration-LAN 1/LAN Settings Tab Click on the LAN 1 and then LAN Settings Tab Note the IP Address of the Avaya IPO Note the Subnet Mask of the Avaya IPO Enable NAT MUST NOT be checked Select the System from the Navigation Pane. From the Navigation Pane, select System and then the correct Avaya IPO System name Click on LAN 1 and then LAN Settings to view the IP information for the IPO (These settings should all be configured prior to the SIP Trunking configuration) Verify the settings above, making note of the IP Address of the Avaya IPO 11

12 Avaya IPO System Configuration-LAN 1/VoIP Tab Click on the LAN 1 and then VoIP Tab Check SIP Trunks Enable Select the System from the Navigation Pane. Set all DiffServ Settings to B8 (Hex) & DSCP 46 From the Navigation Pane, select System and then the correct Avaya IPO System name Click on LAN 1 and then VoIP to configure the VoIP Settings Check SIP Trunks Enable Set all DiffServ Settings to B8 DSCP (Hex) and DSCP 46 12

13 Avaya IPO System Configuration-LAN 1/Network Topology Tab Click on the LAN 1 and then Network Topology Tab Select the System from the Navigation Pane. From the Navigation Pane, select System and then the correct Avaya IPO System name Click on LAN 1 and then Network Topology Tab to configure the Network Topology Settings All fields should look as shown above. This should be the default setting for this page. If configuration does exist in the Network Topology page, set all fields as shown above 13

14 Avaya IPO IP Route Configuration Click on the IP Route Tab Enter the default gateway address Select LAN 1 Select IP Route from the Navigation Pane. From the Navigation Pane, select IP Route Click on the (Default Route) The IP Address field should be The IP Mask field should be The Gateway IP Address should be the default Gateway for the Avaya IPO. Typically, this is the IP Address assigned to the Adtran Ethernet port designated as the default gateway for the LAN. For Destination, Select LAN 1 from the drop down menu All fields should look as shown above. This should be the default setting for this page. If configuration does exist in the Network Topology page, set all fields as shown above 14

15 Avaya IPO Line Configuration-SIP Line Tab Click on the SIP Line Tab Enter the Line (Trunk Group) Number Select In Service Select Check OOS Set to Diversion Header if using Twinning Feature Enter the IP of the Adtran Ethernet port in the ITSP Field. Set to Request URI for the Routing Method Set to LAN 1 Set to UDP Set to 5060 From the Navigation Pane, select Line Select the SIP Line Tab For the Line Number, Enter the desired Trunk Group Number For the ITSP IP Address Field, Enter the IP Address of the Adtran Ethernet Port assigned as the Default Gateway for the LAN 1. Check the box for In Service Check the box for Check OOS For the Call Routing Method, Set to Request URI For the Layer 4 Protocol, Set to UDP For the Use Network Topology, Set to LAN 1 For the Send Port, Set to 5060 For the Send Caller ID, Set to Diversion Header if using the Twinning Feature (This allows the originating Caller ID to be passed to the Twinned endpoint) 15

16 Avaya IPO Line Configuration-SIP URI Tab Click on the SIP URI Tab Click OK, Once Complete with the Edit Channel Highlight the SIP Line and Click Edit For Local URI, Contact, and Display Name, Select Use Internal Data For Registration, Set to 0: <None> For Incoming and Outgoing Group, Enter the SIP Line Number Click OK, Once Complete with the SIP URI tab For Max Calls per Channel, Set to Max Number of calls Under the same Line Settings from the Navigation Pane Select the SIP URI Tab Highlight the Line Group Entry, Click Edit (An Edit Channel Window opens for the Line) For the Local URI, Set to Use Internal Data For the Contact, Set to Use Internal Data For the Display Name, Set to Use Internal Data For the Call Routing Method, Set to Request URI For the Registration, Set to 0: <None> For the Incoming Group, Enter the correct SIP Line Number For the Outgoing Group, Enter the correct SIP Line Number For the Max Calls per Channel, Set to the Max Number of calls for the SIP Line Click OK in the Edit Channel Window Click OK for the SIP URI Window 16

17 Avaya IPO Line Configuration-VoIP Tab Click on the VoIP Tab Set Call Initiation Timeout to 4 For the Compression Mode, Select G.711u 64k or select the desired Codec from the drop down list. Click Advanced to select Multiple Codecs if desired. Set DTMF Support to RFC2833 Uncheck All 3 boxes above Under the same Line Setting from the Navigation Pane Select the VoIP Tab Highlight the Line Group Entry, Click Edit (An Edit Channel Window opens for the Line) For the Local URI, Set to Use Internal Data For the Contact, Set to Use Internal Data For the Display Name, Set to Use Internal Data For the Call Routing Method, Set to Request URI For the Registration, Set to 0: <None> For the Incoming Group, Enter the correct SIP Line Number For the Outgoing Group, Enter the correct SIP Line Number For the Max Calls per Channel, Set to the Max Number of calls for the SIP Line Click OK in the Edit Channel Window Click OK for the SIP URI Window 17

18 Avaya IPO ARS Configuration ARS Route ID Check Secondary Dial Tone Select System Tone Select Out of Service Route Select System Default (4) Check In Service Click OK, once complete From the Navigation Pane, select ARS Right Click on the ARS Menu and Select New An ARS Route ID will be generated automatically (This will be the route ID referenced in other sections of the configuration, so make a note of it) For the Route Name, Enter a unique name for the route For the Dial Delay Time, Set to System Default (4) Check the In Service box For the Out of Service Route, select the appropriate route if using Fail-Over. Note: The Out of Service Route can be to anther SIP Trunk group or Analog Copper Lines, PRI, etc. (Not required if no back-up route exists) Check the Secondary Dial Tone box Select System Tone from the drop down menu under Secondary Dial Tone 18

19 Avaya IPO ARS Configuration ARS Route ID Click Add In the same ARS window, click Add The New Short Code Window opens as shown below 19

20 Complete the Fields as shown below for each ARS Short Code Select the correct Line Group ID (This MUST match the SIP Line specified in the SIP Line Section above) For the Short Codes below with address, Enter the IP of the Adtran Ethernet port assigned as the default gateway for the LAN. with Each Short Code 20

21 21

22 Avaya IPO Short Code Configuration Select Dial Enter N (for any number) Select the correct Line Group ID Select the 9N Short Code Select United States for Locale From the Navigation Pane, select Short Code Scroll Down in the Short Code List and Select 9N (This is for dialling 9+number for Trunk Group Access for the Internal Extensions) For the 9N Short Code, Set the fields as shown in the above window, specifying the correct Line Group ID that was designated created in the ARS table in the steps above For Code, Enter 9N (Should exist already by default) For the Feature, Select Dial For Telephone Number, Enter N For the Line Group ID, select the correct SIP Line Group specified in the ARS table For Locale, Select United States (Unless you are not in the United States) DO NOT check Force Account Code (If account codes are required, this can be changed after testing has been completed) 22

23 Avaya IPO Incoming Call Route Configuration-Standard Tab Select Any Voice Select the correct SIP Line Group Enter the 10 digit DID Select the desired Music on Hold Filename From the Navigation Pane, select Incoming Call Route Right Click on Incoming Call Route, select New For Bearer Capability, Select Any Voice For Line Group ID, Select the correct SIP Line Group specified in the SIP Line page For Incoming Number, Specify the full DID coming from the EarthLink Network (Typically 10 digits) For Hold Music Source, Select the desired.wav Filename or leave at default All other fields can be left at default Note: This process will need to be completed for each Incoming DID 23

24 Avaya IPO Incoming Call Route Configuration-Destination Tab Click on the Destination Tab Select an available user for the DID to terminate Select a Fallback Extension if desired Under the same Incoming Call Route from the Navigation Pane Click on the Destination Tab For Destination, Select a user from the drop down menu that the DID will terminate to For Fallback Extension, Select a second user from the drop down menu if desired (This is not required) Note: This process will need to be completed for each Incoming DID 24

25 Avaya IPO User Configuration-SIP Tab Click on the SIP Tab Enter the 10 Digit DID Enter a SIP Display Name Enter the 10 Digit DID Select the correct User/Extension from the Navigation Pane under User From the Navigation Pane, select User Click on the SIP Tab (Use the scroll bars to scroll all the way to the right) For SIP Name, Enter the 10 Digit DID Number (This number will be in the FROM Header in the in the outgoing INVITE) For SIP Display Name (Alias), Enter the 10 Digit DID Number (This will in the SIP Display Field in the FROM Header on the outgoing INVITE) For Contact, Enter the 10 Digit DID Number (This number will be in the Contact Header) Note: This process will need to be completed for each Incoming DID 25

26 Product Support and Contact Information The information below provides contact information for assistance in configuration and troubleshooting EarthLink s SIP Trunking service. EarthLink Support: (800) or 24x7 Support Availability Avaya Support (NTAC):(877) or support.avaya.com/css/appmanager/public/support 24x7 Support Availability 26

27 EarthLink SIP Trunking Turn-up Testing Procedure To ensure proper call negotiation can be established between EarthLink and the IP PBX, the test steps below MUST be executed during the initial turn-up process. SIP Trunking Test Steps: 1. Test an outbound call to a Local Number. Check for Ring-back, 2-way Audio, and Call Quality. 2. Test an outbound call to a Long Distance Number. Check for Ring-back, 2-way Audio, and Call Quality. 3. Test an outbound call to an International Number. Check for Ring-back, 2-way Audio, and Call Quality. 4. Test an outbound call to a Toll-Free Number. Check for Ring-back, 2-way Audio, and Call Quality. 5. Test an inbound call that lasts greater than 10 minutes 6. Test an outbound call that lasts greater than 10 minutes 7. Test simultaneous inbound and outbound calls to PSTN 8. Test an outbound Call to Operator 0 9. Test an outbound Call to Directory Assistance Test a 911 Call (IDENTIFY TO THE 911 OPERATOR THAT THIS IS A TEST). Ask them to provide phone number, address and secondary or alternate number if available. 11. Test an inbound call to an internal DID. Check for Ring-back, 2-way Audio, and Call Quality. 12. Test an inbound call to Auto-Attendant. Check DTMF and Call Quality 13. Test an outbound call to an Auto-Attendant/IVR and verify DTMF 14. Test Call Transfer off-site 15. Test Call Forward off-site Notes: 27

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