EarthLink Business SIP Trunking/media/9c7c9065fb284439a5faba8f42f4724… · Cisco Call Manager and...

EarthLink Business

SIP Trunking

Cisco Call Manager and Cisco CUBE

Customer Configuration Guide

Implementation Guide

2

Proprietary and Confidential – EarthLink Business

Publication History

First Release: Version 2.0 – April 20, 2012

CHANGE HISTORY

Version Date Change Details Changed By

1.0 4/5/2012 Original Document Draft Thomas Maurin

2.0 4/20/12 Document Updates Dantley Thompson

AUTHOR: Dantley Thompson EarthLink Engineering Thomas Maurin World Wide Technologies


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Table of Contents

Document Purpose ________________________________________________ 4

Product Summary _________________________________________________ 4

Network Architecture and Design ________________________________________ 5

Media Attributes and Codec Negotiation ____________________________________ 6

Codec Support ______________________________________________________ 6

G.711u ___________________________________________________________ 6

G.729a ___________________________________________________________ 6

Packetization Time ____________________________________________________ 6

DTMF Support ______________________________________________________ 6

Fax and Modem Support Requirements ____________________________________ 7

North American Numbering Plan Format ____________________________________ 7

Quality of Service Policy _____________________________________________ 7

EarthLink SIP Trunking to IP PBX Interoperability _______________________________ 8

Adtran Software Version Tested ___________________________________________ 8

IP PBX Software Version Tested ____________________________________________ 8

EarthLink Open Issues & Non-Supported Features ________________________________ 8

Cisco CUCM & CUBE Open Issues & Non-Supported Features _________________________ 8

IP PBX Configuration for EarthLink SIP Trunking with Adtran SIP Proxy ___________________ 9

CUCM Configuration ___________________________________________________ 9

CUBE Configuration __________________________________________________ 13

Product Support and Contact Information __________________________________ 17

EarthLink SIP Trunking Turn-up Testing Procedure _____________________________ 18


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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 Cisco Call Manager with Cisco CUBE.

The configuration outlines the Cisco Call Manager and Cisco CUBE terminating to the Adtran route

passing via the Adtran SIP Stateful 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.

Figure 1-EarthLink SIP Trunking-Network Topology

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

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

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G1

G2

G3

G4

CONSOLE

G4G2

Power over Ethernet

2

EarthLink T1 from Network to Adtran NET T1 0/1

Adtran ETH 0/1 to Customers Ethernet Switch

IP PBX to Customers Ethernet Switch

Adtran 900e/Rear-View

EarthLink

Network

Customer’s Layer 2 Ethernet Switch

Cisco Unified Communications Manager Cisco Unified Border

Element

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. 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 Cisco Unified Communication Manager 7.1(3)

Cisco Unified Border Element (CUBE) 15.1-4.M4

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 be billed correctly.

Cisco CUCM & CUBE Open Issues & Non-Supported Features SIP Refer message enhancements were not added to CUCM until 8.6

Cisco best practices & EarthLink recommend the use of a CUBE (Cisco Unified Border Element) to

connect SIP trunks to a service provider. Earthlink has tested connecting to the Earthlink SIP Trunk

both with and without using a CUBE router. Earthlink supports both methods but recommends the

use of a CUBE router. With the use of a CUBE router, the SIP trunk from the CUCM will be built to

the CUBE and the SIP trunk will be built from the CUBE to the Adtran SIP Proxy.

When CUCM is not deployed using CUBE, MTP resources are required. MTP resources are licensed

and MUST be purchased and prior to implementation of SIP Trunking to EarthLink.

If MTP resources are provided via the CUCM application, G.711u only is supported.

G.729 to G.711 upspeed for faxing is ONLY supported via the CUCM with CUBE.

A multi-server array, such as CUCM Publisher and Subscriber or CUCM Clustering is ONLY supported

when CUCM is implemented with CUBE.


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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 CUCM for the EarthLink SIP

Trunking Product. Basic configuration of the CUCM and CUBE should be complete and be connected to

the LAN prior to configuring the system for SIP Trunking.

CUCM Configuration The screen-shots below are CUCM version 7.1(3). These steps outline the configuration of the CUCM to

work with EarthLink’s SIP Trunking product with the Adtran SIP Proxy. For more detailed information the

Cisco Knowledgebase can be used.

When a SIP Trunk is built to a CUBE, the

MRGL does not need to contain MTP

resources.

Media Termination

Point is not required

when building the SIP

trunk to a CUBE.


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The SIP Trunk Configuration on

CUCM should have a Media

Resource Group List that

contains MTP resources only if

the SIP Trunk is being built

straight to the Earthlink Adtran.

If the SIP Trunk remains built to

a CUBE, then MTP resources are

not required.

If the SIP trunk is being built from the

CUCM straight to the Earthlink

Adtran, the Media Termination Point

Required box needs to be checked.


11


Asserted-identity needs to be checked

711ulaw is the recommended

codec.

RFC 2833 is the supported signaling

method.

The Destination Address should be the

address of the CUBE or alternatively it can

be the Earthlink Adtran.


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Redirecting Diversion Header Delivery –

Outbound needs to be checked for

transfers and call-forwards to work

correctly

The SIP Realm can be found on the System Menu. This is only

needed if a CUBE will not be used. Earthlink will supply the

information needed for each install.


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CUBE Configuration The following configuration is from a CUBE running version 15.1-4.M4. These steps outline the

configuration of the CUBE to work with EarthLink’s SIP Trunking product with the Adtran SIP Proxy. For

more detailed information the Cisco Knowledgebase can be referenced.

The basic CUBE configuration as tested by Earthlink is shown below.

voice service voip

ip address trusted list1

ipv4 172.31.1.0 255.255.255.0

mode border-element2

allow-connections sip to sip3

modem passthrough protocol codec g711ulaw4

sip

asserted-id pai5

early-offer forced6

g729 annexb-all7

!

voice class codec 18

codec preference 1 g711ulaw

codec preference 2 g729r8

!

dial-peer voice 100 voip9

1 This command explicitly enables those source IP addresses from which you would like to add to the trusted list for legitimate VoIP

calls. See more about this Toll-Fraud Prevention Feature at http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080b3e123.shtml 2 This command is used to enable other commands used in the CUBE configuration

3 This command allows SIP-to-SIP calls

4 This command enables modem passthrough globally

5 This command enables the forwarding of the PAI which Earthlink requires in order to properly process transferred and forwarded

calls.

6 This command ensures that the CPE sends the initial SDP for coed negotiation.

7 This command enables otherwise incompatible versions of g729 to connect calls. This command is needed to support DTMF of

g729. Earthlink recommends using g711 as the preferred codec.

8 This section of commands defines the preferred codecs list that will be applied to the dial-peers. Earthlink recommends using g711

as the preferred codec.

http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080b3e123.shtml


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destination-pattern .T

session protocol sipv2

session target sip-server10

voice-class codec 111

dtmf-relay rtp-nte12

ip qos dscp ef signaling13

no vad

!

dial-peer voice 103 voip14

destination-pattern 555….

session protocol sipv2

session target ipv4:172.31.1.35

voice-class codec 1

dtmf-relay rtp-nte

ip qos dscp ef signaling

no vad

!

sip-ua 15

authentication username 2562419305 password 7 110C1817031A07050A21 realm

static.voiplab.deltacom.net

sip-server ipv4:172.31.1.1

9 This is the SIP dial-peer that will send calls to Earthlink.

10 This command defines that the destination of the dial-peer will be the sip server that is defined in the sip-ua section of the config

11 This command applies the previously defined codec list to the dial-peer.

12 This command configures the dial-peer to use RFC2833 for DTMF signaling

13 This command configures the dial-peer to use a DSCP value of 46 for SIP signaling traffic

14 This is the dial-peer that is use to route inbound calls to the CUCM. The CUCM IP address is defined as the target of the dial-peer.

15 This section defines the Earthlink SIP proxy information. The authentication username, password and realm will be provided by

Earthlink. Earthlink will also provide the SIP proxy IP address to be used in this section.


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Earthlink recommends that g711 be used as the codec for all RTP streams, however, Earthlink does

support the use of g729. In the event that g729 and g711 codecs need to be mixed together, the CUBE

will need to make use of DSP transcoders. This functionality requires the use of PVDM modules. The

configuration below shows how to configure the DSP modules to be used as transcoders and how to

register them to the telephony service on the CUBE router.

sccp local GigabitEthernet0/216

sccp ccm 172.31.1.38 identifier 1 version 7.0 17

sccp18

!

!

sccp ccm group 119

bind interface GigabitEthernet0/2

associate ccm 1 priority 1

associate profile 1 register CCME_XCODE

!

dspfarm profile 1 transcode 20

codec g729br8

codec g729r8

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 4

associate application SCCP

!

16

This command defines the interface that will be used for the IP traffic for the DSP resources.

17 This command defines the IP address of the telephony server that will be used to register the DSP resources to. In this example

the CUBE is the telephony server and therefore the IP address used here is also the address of the same CUBE.

18 This command enables skinny for the DSP resources.

19 This section defines the registration information for the DSP resources.

20 This section configures the parameters used for the transcoding sessions.


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telephony-service21

sdspfarm units 1

sdspfarm transcode sessions 8

sdspfarm tag 1 CCME_XCODE

max-ephones 3

max-dn 6

ip source-address 172.31.1.38 port 2000

21

The telephony service section configures the router to run the telephony service so that the DSPs can register to the CUBE router.


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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:

http://www.earthlinkbusiness.com/

(800)239-3000

24x7 Support Availability

Cisco Support (TAC):

http://www.cisco.com/en/US/support/tsd_cisco_worldwide_contacts.html

(800) 553-2447

24x7 Support Availability

http://www.earthlinkbusiness.com/

http://www.cisco.com/en/US/support/tsd_cisco_worldwide_contacts.html


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EarthLink SIP Trunking Turn-up Testing Procedure To ensure proper call negotiation can be established between EarthLink and the Cisco VoIP system, 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:

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