earthlink business sip trunking · the earthlink business sip trunking product is a complete voip...

EarthLink Business

SIP Trunking

Avaya IPO IP PBX

Customer Configuration Guide

Implementation Guide

2

Proprietary and Confidential – EarthLink Business

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


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


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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.


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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.

T1/Ethernet

SIP

OK

P HONEME SS AGE

CONTACTS MENU CAL L L OG

VOLUME

AB C DEF

J KL MNOGHI

1 2 3

4 5 6

T UV WXYZP QRS

7 8 9

*

[

0 #

. ,@

MUTESPEAKER

HEADSET

FORWARD

PHONE /E XI TP AGELE FT

P AGER IGHT

OP TIONS

HOLD

T RANS FE R

CONFE RENCE

DROP

RED IAL

MUTE

HEADS ET

S P EAKER

0SW IP 1 2 3

64 5

97 8

#*

0

ABC DEF

JKL MNOGH I

TU VQ RSP XYZW

PHONE /E XI T P AGELE FT

P AGER IGHT

OP TIONS

HOLD

T RANS FE R

CONFE RENCE

DROP

RED IAL

MUTE

HEADS ET

S P EAKER

0SW IP 1 2 3

64 5

97 8

#*

0

ABC DEF

JKL MNOGH I

TU VQ RSP XYZW

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 2562419310 3000 (AA_VM)

PBX IP = 10.198.254.20 2562419311 3001 (Digital)

Acme IP (OUTSIDE/PUBLIC) = 97.66.255.136 (ANTNLAB ACME PUBLIC) 2562419312 3002 (Digital)

ACME IP (INSIDE/PRIVATE) = 10.188.6.19 2562419313 3030 (Analog Fax)

ACME SESSION AGENT = lab-avaya-ipo 2562419314 3040 (IP)

Media Gateway model = Avaya IPO

Use DN for Identification = TRUE

Authentication required = FALSE

Sip user name = 2562419310

Sip password = deltacom

Sip domain name = static.voiplab.deltacom.net

Trusted = TRUE

PBX = 256249310

DNIS = 10 digit

Charge Number required = TRUE

Charge Number = 2562419310

Call ing Pty # = NA

DID's = 2562419310 - 2562419314

EarthLink SIP Trunking Test Bed Information and Configuration

Figure 1-EarthLink SIP Trunking-Network Topology


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


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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.


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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.


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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.


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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

� 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

Confirm the Avaya IPO is licensed for SIP Trunk

Channels. The number of Instances will

determine how many concurrent calls can be

handled simultaneously.


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Avaya IPO System Configuration-LAN 1/LAN Settings Tab

� 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

Select the System from the

Navigation Pane.

Note the IP Address of the Avaya IPO

Click on the LAN 1 and

then LAN Settings Tab

Enable NAT MUST

NOT be checked

Note the Subnet Mask of the Avaya IPO


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Avaya IPO System Configuration-LAN 1/VoIP Tab


� 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

� Click OK once complete


Navigation Pane.

Check SIP Trunks Enable

Click on the LAN 1 and

then VoIP Tab

Set all DiffServ Settings to

B8 (Hex) & DSCP 46

Click OK once complete


13


Avaya IPO System Configuration-LAN 1/Network Topology Tab


� 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



Navigation Pane.

Click on the LAN 1 and then

Network Topology Tab



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Avaya IPO IP Route Configuration

� From the Navigation Pane, select IP Route

� Click on the 0.0.0.0 (Default Route)

� The IP Address field should be 0.0.0.0

� The IP Mask field should be 0.0.0.0

� 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


Select IP Route from the

Navigation Pane.

Click on the IP Route Tab


Enter the default gateway address

Select LAN 1


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Avaya IPO Line Configuration-SIP Line Tab

� 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)


Set to UDP

Enter the IP of the Adtran

Ethernet port in the ITSP Field.

Enter the Line (Trunk Group) Number

Click on the SIP Line Tab

Set to 5060 Set to LAN 1

Set to Request URI for

the Routing Method

Select In Service

Select Check OOS

Set to Diversion Header if

using Twinning Feature



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Avaya IPO Line Configuration-SIP URI Tab

� 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 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

For Local URI, Contact, and

Display Name, Select Use

Internal Data

Click on the SIP URI Tab

Highlight the SIP Line and Click Edit

For Registration, Set to 0: <None>

For Incoming and Outgoing Group, Enter the SIP Line Number

For Max Calls per Channel,

Set to Max Number of calls

Click OK, Once Complete with the Edit Channel

Click OK, Once Complete with the SIP URI tab


17


Avaya IPO Line Configuration-VoIP Tab

� 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 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

Click on the VoIP Tab

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 Call Initiation Timeout to 4

Set DTMF Support to RFC2833

Uncheck All 3 boxes above


18


Avaya IPO ARS Configuration

� 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


Select System Default (4)

Check Secondary Dial Tone

ARS Route ID

Check In Service

Select System Tone

Select Out of Service Route

Click OK, once complete


19


Avaya IPO ARS Configuration

� In the same ARS window, click Add

� The New Short Code Window opens as shown below

ARS Route ID

Click Add


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 the @ip address, Enter the IP of the Adtran Ethernet

port assigned as the default gateway for the LAN.

� Click OK once complete with Each Short Code


21



22


Avaya IPO Short Code Configuration

� 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)


Select the 9N Short Code

Select Dial

Enter N (for any number)

Select the correct Line Group ID

Select United States for Locale



23


Avaya IPO Incoming Call Route Configuration-Standard Tab

� 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

Select the correct SIP Line Group


Select Any Voice

Enter the 10 digit DID

Select the desired Music on Hold Filename


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Avaya IPO Incoming Call Route Configuration-Destination Tab

� 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)



Select an available user for the DID to terminate


Click on the Destination Tab

Select a Fallback Extension if desired


25


Avaya IPO User Configuration-SIP Tab

� 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)



Enter the 10 Digit DID


Click on the SIP Tab

Select the correct User/Extension from the Navigation Pane under User

Enter the 10 Digit DID

Enter a SIP Display Name


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)239-3000 or http://www.earthlinkbusiness.com/

� 24x7 Support Availability

Avaya Support (NTAC): (877) 295-0099 or

http:// support.avaya.com/css/appmanager/public/support

� 24x7 Support Availability


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 “411”

10. 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:

earthlink business sip trunking · the earthlink business sip trunking product is a complete voip...

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