m EarthLink Business SIP Trunking Cisco CUCM 9.1 with CUBE Customer Configuration Guide
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 9.1 Mike Machnik AUTHOR: Dantley Thompson EarthLink Engineering 2
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 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 21 Product Support and Contact Information 27 EarthLink SIP Trunking Turn-up Testing Procedure 28 3
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. 4
1 2 ABC 4 5 GHI JKL 7 8 PQRS * TUV 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7970 SERIES -? + 1 2 ABC 4 5 GHI JKL 7 8 PQRS * TUV 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7970 SERIES -? + 1 2 ABC 4 5 GHI JKL 7 8 PQRS * TUV 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7970 SERIES -? + Implementation Guide Network Architecture and Design The EarthLink Business SIP Trunking solution consists of several key network elements that are connected to the existing core routing infrastructure. The MetaSwitch Call Feature Server, IP/TDM Gateways, and 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. EarthLink Business Product Certification SIP Trunking Network Topology PSTN Cisco UCM Adtran 1234 Ethernet Switch Adtran CPE Cisco P.E. IP Station Eth 0/1 T1/Ethernet SIP EarthLink VoIP Network SIP SIP SIP Acme Packet SBC MetaSwitch Application Server Cisco Router/CUBE Voice Gateway Split-Horizon DNS Server IP Station Title: EarthLink Business SIP Trunking Test Bed Network Topology Rev.01 Date: 8/30/2011 Drawing by: Dantley Thompson Figure 1-EarthLink SIP Trunking-Network Topology 5
S T A T LINK PoE / A C T 1 2 3 4 5 6 7 8 1 9 10 12 13 14 15 16 17 18 19 20 21 22 23 24 G1 G2 G3 G4 CONSOLE CF 1 CF 0 ISM 1 3 5 7 9 1 1 1 3 1 5 1 7 1 9 2 1 23 G1 G3 2 4 6 8 1 0 1 2 1 4 1 6 1 8 2 0 2 2 24 PVDM1 PVDM0 Power over Ethernet EN EN AUX CONSOLE GE 0/1 GE 0/2 L L S 0 1 0 G2 L G4 Implementation Guide Adtran 900e/Rear-View EarthLink Network 2 EarthLink T1 from Network to Adtran NET T1 0/1 Adtran ETH 0/1 to Customers Ethernet Switch Cisco 2900 CUBE EHWIC 3 EHWIC 2 EHWIC 1 EHWIC 0 G E / 0 2911 Cisco UCM SM SLOT 1 USB Figure 2-EarthLink SIP Trunking-Connections from Adtran CPE to IP PBX 6
Media Attributes and Codec Negotiation Codec Support A voice codec (coder/decoder) is a hardware/software module/algorithm that takes an analog or digital voice stream and encodes it into an IP packet. For the EarthLink Business SIP Trunking Product, we currently support two (2) of the most common codec s utilized in the continental United States, G.711u and G.729a. The preferred codec offered by EarthLink in the default configuration model is G.711u, then G.729a. Basically this means that the call will negotiate using the G.711u codec first, as long as the terminating end sends G.711u as the first or primary offered codec. The paragraphs below provide more detailed information related to the codec s and other requirements associated with proper negotiation of the media/rtp. G.711u G.711u is the most common uncompressed audio codec deployed in the US. Because it is uncompressed, it supports the highest level of quality for the call. Typically the G.711u consumes 90Kbps-100Kbps per call. The standard sampling rate of 8kHz is used for the G.711u codec. G.729a G.729a is the most common codec utilized to support compressed audio utilized in the US. Because it is compressed, it is perceived to have a lower voice quality than that of G.711u, however most people would never be able to tell the difference. Typically the G.729 consumes 30Kbps-40Kbps per call. The standard sampling rate of 8kHz is used for the G.729a codec. Packetization Time Packetization Time determines how often the audio stream is sampled and how often an IP packet is created. The standard packetization times are 10ms, 20ms, 30ms, and 40ms. EarthLink Media Gateway s have been statically configured to use a 20ms packetization time. IP Phones and/or Voice Applications will need to configure their equipment for a 20ms packetization time before audio traffic can be reliably passed across the EarthLink IP Voice network. DTMF Support EarthLink supports the transmission of Dual-Tone Multi-frequency (DTMF) digits through the implementation of RFC2833. This RFC covers the basis of including DTMF digits within the media/rtp path of the call. EarthLink recommends for Customers to configure their IP PBX s and/or Voice Applications to use RFC2833 to allow for DTMF to be passed properly and detected across the EarthLink IP Voice network. 7
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. 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 be configured to use DSCP 46 to provide prioritization for SIP and RTP. Marking the DSCP field in the IP packet header will allow for packet classification to be matched and provide priority across EarthLink s network. This also ensures QoS specifications outlined in SLA (Service Level Agreements) can be sufficiently met between EarthLink and the customer. 8
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 CUCM 9.1 CUBE c2900-universalk9-mz.spa.152-4.m4.bin Phones Cisco 7960 7940 SCCP and SIP 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 9
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: 10
Trunk Configuration to CUBE: 11
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Trunk to Cisco Unity Connection: 15
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Transcoder config. CUBE will be the MTP: 20
Cisco CUBE Configuration CUBE Configuration: Current configuration : 6080 bytes Last configuration change at 15:32:43 UTC Fri Oct 25 2013 by version 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 boot-end-marker 21
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 cisco network 192.168.1.0 255.255.255.0 default-router 192.168.1.1 dns-server 8.8.8.8 8.8.4.4 option 150 ip 192.168.1.2 ip domain name voiplab.elnk.us 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 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 22
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 voice class codec 2 codec preference 1 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 interface Embedded-Service-Engine0/0 no ip address shutdown 23
interface GigabitEthernet0/0 ip address 207.X.X.X 255.255.255.254 ip access-group 101 in no ip unreachables ip nat outside ip virtual-reassembly in duplex full speed 100 service-policy output test interface GigabitEthernet0/1 ip address 192.168.1.1 255.255.255.0 ip nat inside ip virtual-reassembly in duplex full speed 100 ip forward-protocol nd no ip http server ip http secure-server ip nat inside source list 1 interface GigabitEthernet0/0 overload ip nat inside source static 192.168.1.2 207.X.X.X ip nat inside source static 192.168.1.3 207.Y.Y.Y ip route 0.0.0.0 0.0.0.0 207.X.X.X access-list 1 deny 192.168.1.3 access-list 1 deny 192.168.1.2 access-list 1 permit 192.168.1.0 0.0.0.255 access-list 101 deny ip host 213.249.66.114 any access-list 101 permit ip any any 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 24
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 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 1 voip description CUCM destination-pattern 774237... 25
session protocol sipv2 session target ipv4:192.168.1.2 voice-class codec 2 dtmf-relay rtp-nte ip qos dscp ef signaling no vad sip-ua 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 192.168.1.1 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# 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: 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 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: 28