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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
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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.
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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.
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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
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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: