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Implementation Guide
Publication History
First Release: Version 1.0 April 20, 2012
CHANGE HISTORY
Version
1.0 1.1
Date
4/20/2012 4/23/2012
Change Details
Document Release Document Updates
Changed By
D.Thompson D.Thompson
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Table of Contents
Document Purpose ________________________________________________4 Product Summary _________________________________________________4 Customer Premise Diagram ___________________________________________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 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 IP PBX Configuration for EarthLink SIP Trunking with Adtran SIP Proxy ___________________9 Switchvox IP PBX Configuration ____________________________________________ 9 Product Support and Contact Information __________________________________14 EarthLink SIP Trunking Turn-up Testing Procedure _____________________________15
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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 Switchvox SMB IP-PBX. 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 Engineering, Field Technicians, Remote Maintenance Support, and Marketing and Sales personnel.
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 EarthLinks 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 EarthLinks Hosted Voice and SIP Trunking Product families. In addition to the basic call control, advanced call routing functionality is available with EarthLinks 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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Adtran ETH 0/1 to Customers Ethernet Switch EarthLink T1 from Network to Adtran NET T1 0/1
LINK
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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 Gateways 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 PBXs 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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IP PBX Configuration for EarthLink SIP Trunking with Adtran SIP Proxy
The steps below provide a step-by-step guide for configuration of the Switchvox IP-PBX for the EarthLink SIP Trunking Product. Basic configuration of the Switchvox IP-PBX should be complete and Switchvox MUST be connected to the LAN prior to configuring the system for SIP Trunking.
The account ID and password can be set to anything since the Adtran SIP Proxy will authenticate the PBX based on IP address. Hostname/IP Address must be set to the LAN interface of the Adtran device. Callback Extension is the default extension to call when no incoming call rules are matched on inbound calls. DTMF Mode must be set to RFC2833.
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Switchvox VoIP Provider Configuration Setup -> VOIP Providers -> Peer Settings
Host Type must be set to Peer so that Switchvox doesnt send REGISTER messages to the Adtran SIP Proxy. Apply Incoming Call Rules to Provider is set to yes so that incoming calls are routed through call rules.
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Switchvox VoIP Provider Configuration Setup -> VOIP Providers -> Call Settings
Select ULAW as the codec to use for calls with Earthlink. G729 will show up in the codec list if a G729 license is installed on the Switchvox PBX. Scroll down and click Save SIP Provider.
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Create outgoing call rules to use Earthlink as required. The rules in the example above start with a 9 as the leading digit, make sure to strip the 9 off from the call before sending it to the provider. This is done when modifying or creating an outgoing call rule.
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Create routes for your DIDs to route to extensions on the PBX. DID routes can be for single DIDs or a range of DIDs.
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