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EarthLink Business SIP Trunking ININ IC3 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking · 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)

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Page 1: EarthLink Business SIP Trunking · 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)

EarthLink Business

SIP Trunking

ININ IC3 IP PBX

Customer Configuration Guide

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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 Thompon

1.1 10/17/2013 Modified for ININ IC3 Mike Machnik

AUTHOR: Dantley Thompson EarthLink Engineering

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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

ININ IC3 Open Issues & Non-Supported Features _________________________________ 9

IP PBX Configuration for EarthLink SIP Trunking with Adtran CPE _____________________10

ININ IC3 IP PBX Configuration ____________________________________________ 10

Line Configuration ________________________________________________ 10

Line Menu 10

Identity (Out) Menu.................................................................................................................................................. 11

Audio Menu .............................................................................................................................................................. 12

Transport Menu ........................................................................................................................................................ 14

Session Menu ........................................................................................................................................................... 15

Authentication Menu ............................................................................................................................................... 16

Proxy Menu .............................................................................................................................................................. 17

Access Menu ............................................................................................................................................................. 18

Region Menu ............................................................................................................................................................ 19

SIP Proxy Support ____________________________________________20

Fax Caveats _________________________________________________20

E911 Support ________________________________________________20

Product Support and Contact Information __________________________________21

EarthLink SIP Trunking Turn-up Testing Procedure _____________________________22

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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 ININ IC3 with the Adtran CPE.

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

Oracle/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) and will not be configured as a SIP Proxy. The ININ IC3 will handle the natting and

the 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 Oracle/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 Oracle/Acme Packet SBC. The same policy relates to

the CPE device installed at the customer premise. The Oracle/Acme Packet SBC will 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 Oracle/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. 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.

MetaSwitch Application Server

Acme Packet SBC

T1/Ethernet

Title: EarthLink Business SIP Trunking Test Bed Network Topology

Rev.01 Date: 8/30/2011 Drawing by: Dantley Thompson

Eth 0/1

Eth 0/2

EarthLink Business Product CertificationSIP Trunking Network Topology

SIP SIP SIP

SIPEarthLink

VoIPNetwork

Split-HorizonDNS Server

SIP

PSTN

Cisco P.E.

IP Station

IP Station

Analog FaxAdtran FXS 0/1

Adtran CPE

OK

PHONEMESSAGE

CONTACTS MENU CALL LOG

VOLUME

ABC DEF

JKL MNOGHI

1 2 3

4 5 6

TUV WXYZPQRS

7 8 9

*

[

0 #

.,@

MUTESPEAKER

HEADSET

FORWARD

OK

PHONEMESSAGE

CONTACTS MENU CALL LOG

VOLUME

ABC DEF

JKL MNOGHI

1 2 3

4 5 6

TUV WXYZPQRS

7 8 9

*

[

0 #

.,@

MUTESPEAKER

HEADSET

FORWARD

CustomerEthernet Switch

IC Server

Figure 1-EarthLink SIP Trunking-Network Topology

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2

EarthLink T1 from Network to Adtran NET

T1 0/1

Adtran ETH 0/1 to Customers Ethernet Switch

Adtran 900e/Rear-View

EarthLink

Network

IC Server

1 3 5 7 9 1 1 1 3 1 5 1 7 1 9 2 1 23 G3G1

LINK / A C T

S T A T PoE

1

2

3

4

5

6

7

8

9

10

11

12

13

14

15

16

17

18

19

20

21

2 4 6 8 1 0 1 2 1 4 1 6 1 8 2 0 2 2 24

22

23

24

G1

G2

G3

G4

CONSOLE

G4G2

Power over Ethernet

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 A4.09

IP PBX Software Version Tested ININ IC3 version 4.0 SU3

Phones Polycom IP550 version 4.0.1.13681

EarthLink Open Issues & Non-Supported Features Registration is currently not supported for the EarthLink SIP Trunking Product.

T38 faxing is not currently supported.

ININ IC3 Open Issues & Non-Supported Features Call transfer with REFER is not supported. Call transfer is performed with Re-INVITE and media is

anchored on the ININ IC3 server.

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IP PBX Configuration for EarthLink SIP Trunking with Adtran CPE The steps below provide a step by step guide for configuration of the ININ IC3 for the EarthLink SIP

Trunking Product. Basic configuration of the ININ IC3 should be complete and the ININ IC3 must be

connected to the LAN prior to configuring the system for SIP Trunking.

ININ IC3 IP PBX Configuration SIP trunk configuration for ININ IC3 provided by Andrew Marshall of Interactive Intelligence.

Line Configuration

The line page has a vast majority of the configuration options required for SIP Carrier setup. This

is the section that configures the connection to the carrier’s servers, any authentication or

registration information, and basic configuration needs.

Any reference to a menu, while talking about the line configuration, will refer to the options on the

left side of the line configuration page, and tabs will refer to the standard tab interface across the

top of the line configuration page.

Line Menu

Figure 3: Line Menu Line Configuration Page

Active

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The active box should be checked. This activates the line. If this box is not checked, the line will

not be available for any function. This can also be affected by right clicking on the line in

Interaction Administrator, dropping to the Set Active menu option, and selecting Yes.

Domain Name

This box should contain the Fully Qualified Domain Name (FQDN) of the IC server. It will

automatically be appended to all REGISTER requests sent by Interaction Center and used in the

FROM header on outgoing SIP messages.

Enable T.38 Faxing

EarthLink Business’s SIP Carrier service does not support the T.38 faxing protocol by default.

Leave this box unchecked if you do not wish to use T.38 Faxing.

Remainder of Line Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

Identity (Out) Menu

Figure 4: Identity (Out) Menu Line Configuration Page

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Calling Address

Figure 3: Configure Line Value Dialog

Click the “…” button next to the Line Value 1 box to bring up the Configure Line Value dialog shown

in Figure 3. In the Address box, enter the Main DID to be used as the Default User portion on

outgoing calls.

Remainder of Identity (Out) Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

Audio Menu

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Figure 4: Audio Menu Line Configuration Page

Audio Path

This is, for the most part, the choice of the client with respect to the business being done on the

server. However, there are several important caveats.

Dynamic audio for SIP carriers has significantly less delay as compared to Always-In

audio (~100ms).

The audio will be brought into the Media Server when set to Dynamic Audio for any call

that is recorded, conferenced, placed on hold, or is navigating the IVR.

DTMF Type

This is up to the discretion of the user. EarthLink Business supports both In-Band and Out-of-Band

(RFC2833) DTMF Types.

Remainder of Audio Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

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Transport Menu

Figure 5: Transport Menu Line Configuration Page

Transport Protocol

This option should be set to UDP.

Receive Port

This option should be set to 5060 (the standard SIP port), unless an agreement for an alternative

port has been agreed upon with EarthLink Business.

Remainder of Transport Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

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Session Menu

Figure 6: Session Menu Line Configuration Page

Media Timing/Media reINVITE Timing

This dropdown pair controls Delayed Media support. Setting both to Normal is the recommend

method by Interactive Intelligence for all SIP Carriers.

Remainder of Session Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

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Authentication Menu

Figure 7: Authentication Menu Line Configuration Page

This box must be checked to enable authentication to the SIP Carrier. The User Name and

Password fields should be filled out with the appropriate information provided by the SIP Carrier.

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Proxy Menu

Figure 8: Proxy Menu Line Configuration Page

Prioritized list of Proxy IP addresses

This box is somewhat of a misnomer in the case of some SIP Carriers. In the case of EarthLink

Business there may not be a single IP that is needed. In this case, they provide a Fully Qualified

Domain Name (FQDN) to a machine or cluster that handles the requests. To enable the service to

work properly, this FQDN is entered into the Address box and the port number (generally 5060

unless otherwise directed) is entered into Port Number box. Additionally, the IP Addresses that the

FQDN resolves to can be entered in place of the FQDN to enable proper service functionality.

Remainder of Proxy Menu Options

These have no major direct impact on the SIP carrier configuration, and should be addressed

according to business needs.

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Access Menu

Figure 9: Access Menu Line Configuration Page

All computers will be

This should be set to Denied Access to limit the remote endpoints that this line will accept calls

from.

Exceptions

Each IP Address provided by EarthLink Business should be added to the list to grant them access to

this Line. As of 10/3/2013, this list cannot be configured with FQDN or DNS SRV records.

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Remainder of Access Menu Options

These have no major direct impact on the SIP Carrier configuration, and should be addressed

according to business needs.

Region Menu

Figure 10: Region Menu Line Configuration Page

Location should be set according to business needs. However, the user should take care to assure

the location supports the proper codecs supported by the SIP Carrier.

In the case of EarthLink Business, only G.711 mu-law and G.729 are supported, so selecting a

location that does not have either of these as an option would cause the line not to function

properly. EarthLink Business does not have a particular business model preference for either

codec, so this is up to the discretion and needs of the user.

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SIP Proxy Support For EarthLink Business the Interaction SIP proxy is not supported. This information is included for

completeness and in the case that it may be supported in the future. SIP Proxy is not typical in a

standard deployment and is used in a failover scenario in the unlikely event of both IC servers

failing.

Fax Caveats

EarthLink Business does not support T.38 faxing. However, if the customer would like to use an

analog fax machine connected to the network, the way to do this is with an analog to SIP FXS

device connecting an analog fax machine to the IP network. The FXS device will pass the SIP

information on allowing for G.711 pass-through (which is the carrying of the fax signal through the

voice packets on the network).

E911 Support

EarthLink Business currently supports E911.

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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

24x7 Support Availability

http://www.earthlinkbusiness.com/support/support.xea

Interactive Intelligence:

24x7 Support Availability

[email protected]

https://my.inin.com/_layouts/inin/pages/login.aspx?ReturnUrl=%2f

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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: