earthlink business sip trunking/media/3a4223f3f76e4dd0... · the earthlink business sip trunking...

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EarthLink Business

SIP Trunking

Cisco CUCM 10.5 with CUBE

Customer Configuration Guide

Implementation 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 11/06/2013 Modified for Cisco CUCM 10.5 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 ________________ Error! Bookmark not defined.

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

Cisco CUCM Open Issues & Non-Supported Features ______________________________ 9

IP PBX Configuration for EarthLink SIP Trunking with Adtran CPE _____________________10

Cisco CUCM IP PBX Configuration _________________________________________ 10

Cisco CUBE Configuration ______________________________________________ 17

Product Support and Contact Information __________________________________23

EarthLink SIP Trunking Turn-up Testing Procedure _____________________________24


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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 Cisco CUCM with CUBE behind

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 Cisco CUBE will handle the NAT 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

Oracle SBC

Title: EarthLink Business SIP Trunking Test Bed Network Topology

Rev.02 Date: 3/23/2016 Drawing by: Mike Machnik

SIP SIP SIP

EarthLinkVoIP

Network

PSTN

Cisco P.E.

Cisco Router/CUBEVoice Gateway172.27.199.1

Adtran 1234Ethernet Switch

Cisco UCM BE6000Prov 172.27.199.10CUCM 172.27.199.11Unity 172.27.199.12CUP 172.27.199.13

IP Station

IP Station

IP Station

Split-HorizonDNS Server

Adtran CPE

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

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

LINK /A C T

S TA T PoE

1

2

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4

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8

9

10

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12

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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

G4G2Power over Ether net

Cisco 2900 CUBE

Cisco UCM

BE6000

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 122 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.

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 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 R10.3.2

IP PBX Software Version Tested CUCM 10.5

VM ESXi 5.5

CUBE c2900-universalk9-mz.SPA.152-4.M4.bin

Phones Cisco CP-8841 SIP88xx.10-2-2-16

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

T38 faxing is not currently supported.

Cisco CUCM Open Issues & Non-Supported Features No current open issues.


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IP PBX Configuration for EarthLink SIP Trunking with Adtran CPE The steps below provide a basic guide for the configuration of the CUCM with CUBE for the EarthLink SIP

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

CUBE must be connected to the LAN prior to configuring the system for SIP Trunking. The Cisco CUBE

will anchor the media and will be the MTP resource.

Cisco CUCM IP PBX Configuration

Cisco CUCM configuration:


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Trunk Configuration to CUBE:


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Media Termination Point Configuration.


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Deice Pool Configuration.


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Server Configuration.

Cisco CUBE Configuration

CUBE Configuration:

Current configuration : 6080 bytes ! ! Last configuration change at 19:47:34 UTC Mon Feb 22 2016 by cube ! NVRAM config last updated at 19:47:37 UTC Mon Feb 22 2016 by cube ! NVRAM config last updated at 19:47:37 UTC Mon Feb 22 2016 by cubeversion 15.2 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname cube ! boot-start-marker boot system flash:c2900-universalk9-mz.SPA.152-4.M4.bin


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boot-end-marker ! ! card type command needed for slot/vwic-slot 0/2 ! card type command needed for slot/vwic-slot 0/3 enable secret enable password 7 ! no aaa new-model ! ip cef ! ip dhcp excluded-address 192.168.1.1 192.168.1.10 ip dhcp excluded-address 192.168.1.254 ! ip dhcp pool CUCM_10 network 172.27.199.0 255.255.255.192 default-router 172.27.199.1 dns-server 8.8.8.8 8.8.4.4 option 150 ip 172.27.199.11 domain-name ciscolocal.com ! ip domain name voiplab.elnk.us ip host pcp.ciscolocal.com 172.27.199.10 ip host ucm-pub.ciscolocal.com 172.27.199.11 ip host ucn1.ciscolocal.com 172.27.199.12 ip host imp1.ciscolocal.com 172.27.199.13 ip name-server 207.230.65.90 ip name-server 207.230.65.74 no ipv6 cef multilink bundle-name authenticated ! crypto pki trustpoint TP-self-signed-2170615189 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-2170615189 revocation-check none rsakeypair TP-self-signed-2170615189 ! crypto pki certificate chain TP-self-signed-2170615189 certificate self-signed 01 3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 32313730 36313531 3839301E 170D3133 30383031 31393134 33325A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 31373036 31353138 3930819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100990C 20491A39 45530863 8FB0D84F 0A5BAE76 318847EB BFE85A72 BFDF413F BAB3E2F4 2CDF65AF 877C59E5 C51A4D72 62B70FA4 B7636E42 763B778F 05C4A938


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971F5E7F 56FA3458 FEB3A3A4 B16D57EC 9B15AAE7 18023184 95D26E9F 0975C7BD A275C10B 4CC4135A A07C5C20 ADA3E41A CD51AA5E 972499FA D5E7D22C 2CFE8D60 01190203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603 551D2304 18301680 14C24A83 3338E672 D25548C4 F6D89A5D 4C9642D3 E2301D06 03551D0E 04160414 C24A8333 38E672D2 5548C4F6 D89A5D4C 9642D3E2 300D0609 2A864886 F70D0101 05050003 81810053 2D0AF887 78460733 4F9E25F7 8EEBC1FF A16E222F 16BBF223 82E82C15 EB4FDBA1 45419053 7B5BF1BE 91195158 47E62FFE E734E84C 2654941D C1C570EA CB4995DF C4823B29 C22862A7 795A3AC4 116F0BB5 333B998E 040B90B5 2D3D336B CA49B7E9 2D82EC15 B607145B 3C344854 A37EDC20 D6DFBA7F 77B3C796 613E8B42 E82401 quit voice-card 0 dspfarm dsp services dspfarm ! voice service voip ip address trusted list ipv4 207.X.X.0 255.255.255.0 ipv4 10.0.0.0 255.0.0.0 mode border-element allow-connections sip to sip modem passthrough protocol codec g711ulaw sip asserted-id pai early-offer forced g729 annexb-all ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! license udi pid 73-11834-09 sn FTX163884VC hw-module pvdm 0/0 ! username cube password 7 0822594C0C4B554641 ! redundancy ! class-map match-all testaf41 match ip dscp af41 class-map match-all test match ip dscp ef ! policy-map test class test class testaf41 !


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interface Embedded-Service-Engine0/0 no ip address shutdown ! interface GigabitEthernet0/0 ip address 207.X.X.X 255.255.255.254 ip access-group 101 in no ip unreachables ip virtual-reassembly in duplex full speed 100 service-policy output test ! interface GigabitEthernet0/1.199 encapsulation dot1Q 199 ip address 172.27.199.1 255.255.255.192 ip virtual-reassembly in end! ip forward-protocol nd ! no ip http server ip http secure-server ! ip route 0.0.0.0 0.0.0.0 207.X.X.X ! control-plane ! voice-port 0/0/0 ! voice-port 0/0/1 ! voice-port 0/0/2 ! voice-port 0/0/3 ! mgcp profile default ! sccp local GigabitEthernet0/1 sccp ccm 192.168.1.2 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/1 associate ccm 1 priority 1 associate profile 2 register CUBE_MTP associate profile 1 register MTPFC99479E4F81 associate profile 3 register CUBE_CONF


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! dspfarm profile 1 transcode codec g729abr8 codec g729ar8 codec g711alaw codec g729r8 codec g729br8 codec g711ulaw maximum sessions 4 associate application SCCP ! dspfarm profile 3 conference codec g729br8 codec g729r8 codec g729abr8 codec g729ar8 codec g711alaw codec g711ulaw maximum sessions 10 associate application SCCP ! dspfarm profile 2 mtp codec g711ulaw maximum sessions hardware 50 associate application SCCP ! dial-peer voice 100 voip description Catch-all dialing destination-pattern .T session protocol sipv2 session target sip-server voice-class codec 1 dtmf-relay rtp-nte ip qos dscp ef signaling no vad ! dial-peer voice 2 voip description CUCM-10 destination-pattern 774237182[0-4] session protocol sipv2 session target ipv4:172.27.199.11 voice-class codec 1 dtmf-relay rtp-nte ip qos dscp ef signaling no vad ! sip-ua


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sip-server dns:stat-msblt.voiplab.elnk.us ! gatekeeper shutdown ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 8 sdspfarm tag 1 MTPFC99479E4F81 max-ephones 1 max-dn 1 ip source-address 172.27.199.1 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! line con 0 exec-timeout 0 0 password 7 060506324F41 line aux 0 line 2 no activation-character no exec transport preferred none transport output pad telnet rlogin lapb-ta mop udptn v120 ssh stopbits 1 line vty 0 4 login local transport input all ! scheduler allocate 20000 1000 ! end cube#


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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/support/support.xea

(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/support/support.xea

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 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:

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