Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment. JR Richardson



Similar documents
Leveraging Asterisk to Deliver Large Scale VoIP Services within a Carrier Environment

Asterisk: A Non-Technical Overview

Gateways and Their Roles

ehealth and VoIP Overview

Internet Telephony Terminology

White Paper. Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market.

NEWT Managed PBX A Secure VoIP Architecture Providing Carrier Grade Service

Cisco CME Features and Functionality

CVOICE Exam Topics Cisco Voice over IP Exam # /14/2005

An XOP Networks White Paper

OAISYS and ShoreTel: Call Recording Solution Configuration. An OAISYS White Paper

Contents. Specialty Answering Service. All rights reserved.

VoIP-PSTN Interoperability by Asterisk and SS7 Signalling

IP- PBX. Functionality Options

Voice over IP Basics for IT Technicians

Crash Course in Asterisk

VoIP from A to Z. NAEO 2009 Conference Cancun, Mexico

Network Overview. Background Traditional PSTN Equipment CHAPTER

CVOICE - Cisco Voice Over IP

Softswitch & Asterisk Billing System

Packetized Telephony Networks

Asterisk & ENUM. Extending the Open Source PBX. Michael Haberler, IPA Otmar Lendl, nic.at

AT&T SIP Trunk Compatibility Testing for Asterisk

Voice over IP (VoIP) Basics for IT Technicians

; Channels 1-8 are incoming voice. ; Channels are for data.

VoIP in the Enterprise

VoIP Survivor s s Guide

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga

Connecting Your Enterprise With Asterisk: IAX to Carriers. Dayton Turner Voxter Communications

Open Source Telephony Projects as an Application Development Platform. Frederic Dickey Director Product Management

Troubleshooting Voice Over IP with WireShark

Overview of Asterisk (*) Jeff Gunther

SIP Trunking with Microsoft Office Communication Server 2007 R2

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Integrate VoIP with your existing network

SIP Trunking. Cisco Press. Christina Hattingh Darryl Sladden ATM Zakaria Swapan. 800 East 96th Street Indianapolis, IN 46240

PETER CUTLER SCOTT PAGE. November 15, 2011

Presented by: John Downing, B.Eng, MBA, P.Eng

GARTNER REPORT: SIP TRUNKING

Commonly Supported Fax/Modem Call Flow Configuration Examples

Software-Powered VoIP

Alcatel-Lucent OXO Configuration Guide. For Use with AT&T s IP Flexible Reach Service. Version 1 / Issue 1 Date July 28, 2009

Three Network Technologies

Fig. Setting up of a VoIP call. Fig. Experimental setup

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX)

Implementing Cisco IOS Unified Communications (IIUC)

BUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS

Integrating VoIP Phones and IP PBX s with VidyoGateway

SIP-Based Solutions in the Contact Center: Using Dialogic Media Gateways with the Genesys Voice Platform

AT&T IP Flexible Reach Service

VitalPBX. Hosted Voice That Works. For You

Allstream Converged IP Telephony

IP PBX using SIP. Voice over Internet Protocol

Copyright and Trademark Statement

B rismark. Open Source IP PBX The Future of Telephony. T: W:

Avaya Call Recording Solution Configuration

VoIP and IP IT Tralee

Selecting the Right SIP Phone for Your IP PBX By Gary Audin May 5, 2014

Converged Telephony Solution. Technical White Paper

IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>>

Advanced LCR (Least Cost Router) With SIP Proxy Server

Using Asterisk with Odin s OTX Boards

Mediatrix 3000 with Asterisk June 22, 2011

A Cable Telephony Case Study: HOT Telecom

Specialty Answering Service. All rights reserved.

Asterisk Calling Card & Billing System

Enterprise open source VoIP with Asterisk

IP Telephony with Asterisk. Sunday A. Folayan

Telco Carrier xsps Solutions.

Enabling Innovation - Unleashing Unified Communications: Best Practices and Case Studies. October 18-19, 2011

Choosing a Dialogic Product Option for Creating a PSTN-HMP Interface

Connect your Control Desk to the SIP world

"Charting the Course... Implementing Cisco IP Telephony & Video, Part 1 v1.0 ( CIPTV1 ) Course Summary

Implementing Cisco IP Telephony & Video, Part 1

Cisco AS5400 Series Universal Gateways How to Order a Cisco AS5400XM Universal Gateway

1. Mobile VoIP solutions and Services:

AudioCodes Gateway in the Lync Environment

Building Voice VPN with Simton IPX

Basics of VoIP Termination

Integrated Communications Platform

Enterprise VoIP. Silvano Gai. ftp://ftpeng.cisco.com/sgai/t2000voip.pdf. Cisco Systems, USA Politecnico di Torino, IT

SIP Trunking DEEP DIVE: The Service Provider

Introducing Cisco Voice and Unified Communications Administration Volume 1

Voice over IP Technologies

SIP Trunking: The New Normal in the Cloud Era

Which VoIP Architecture Makes Sense For Your Contact Center?

Total Recall Max SIP VoIP Call Recording Server

Application Note. Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server

and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG

STUDY PAPER on IP PABX. (IP based PRIVATE AUTOMATIC BRANCH EXCHANGE) Table of Contents. 1. Introduction History & Evolution of PABX...

ReplixFax Fax over IP (FoIP) Technical Overview and Benefits

Implementing Cisco IP Telephony & Video, Part 1 CIPTV1 v1.0; 5 Days; Instructor-led

IP Telephony Deployment Models

Transcription:

Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment JR Richardson

Early VoIP Environment Telecom Act of 1996, mass competition, Telco's needed value add features and capabilities, VoIP held great potential VoIP services very immature and costly Hardware VoIP solutions expensive with few features compared to POTS Software VoIP existed but PC platform processor speeds too low for mass scale

What is Asterisk? Open Source Hybrid TDM and packet voice PBX and IVR platform Runs on general computing platform Analog and Digital telephony interfaces Supports SIP, H323, MGCP, SCCP and IAX2 Dial Plan scripting bound only by imagination of administrator

Hot Debate: Hardware or Software DSP? Digital Signal Processing has been a barrier to low cost VoIP services Hardware solutions are high cost in nature Software DSP technology has been around for years but PC performance low PC performance has increased enough to warrant investigation into large scale soft switch solutions

Hardware vs. Software DSP Exercise Critical planning event for VoIP deployment, calculate PSTN access, TDM to Packet Voice transcoding Large Scale 200K line side extensions Trunk ratio 7:1 28,571 DSP channels required

Hardware DSP Cost Lowest price Dialogic hardware, 2xT1 PCI card, $7,999 Assuming volume discounts @ 50%, $4000/2xT1 card achievable Calculation comes to $83/port ($4000/48) 28,571 trunks / 24 ports per T1 = 1191 T1 s needed / 2 T1 s per card = 596 cards 596 cards x $4000 = $2,384K

Software DSP Cost Digium 4xT1 cards list $1500 Assuming volume discounts @ 50%, $750/4xT1 card achievable (is Mark Spencer watching?) Calculation comes to $7.80/port ($750/96) 28,571 trunks / 24 ports per T1 = 1191 T1 s needed / 4 T1 s per card = 298 cards 298 cards x $750 = $223,500 1/10 th the cost of hardware DSP cards

Processor Loading Sending transcoding function to central CPU has trade offs, processor loading To utilize all 96 channels on 4xT1 card, the server must be robust, driving up cost Dual Zeon 2.8GHz, 1 Gig RAM, SCSI HD With strong vendor relations, can achieve server cost @ $2000 With 1-4xT1 Digium card per server, 298 are servers needed

Processor Loading (cont) 298 servers x $2000 = $596K With hardware DSP cards, same server could easily handle 4 cards With 4 2xT1 cards per server, 149 servers are needed 149 servers x $2000 = $298K Cost for Hardware DSP PSTN GW $2,683K, Software PSTN GW $819K

Dedicated Platform Cost Cisco AS5850 Good port density with 5 CT3 s (190 T1 s), 3360 channels in one fully loaded chassis Lowest cost found $175K Calculation come to $52/port ($175K/3360) 28,572 trunks require 8.5 AS5850 s Cost for this solution $1,513K

Dedicated DSP vs Software DSP $3,000 $2,500 $2,384 $2,682 In M illions $2,000 $1,500 $1,000 $500 $224 $298 $596 $820 $1,513 $0 Interface Card Cost Server Cost Total Cost Hardware Dedicated DSP Chip Software DSP Purpose Built

Rack Cost and Space 8 racks required for software PSTN GW 4 racks required for hardware DSP 3 racks for AS5850 platform Rack kits ~$1000 Space availability more critical than cost of rack hardware

Power Cost Software DSP, 298 servers @ 300 Watts per server = 89.4KW Hardware DSP, 149 servers @ 300 Watts per server = 44.7KW 9 Cisco AS5850 s @ 2.4KW per unit = 21.6KW Average cost per KW = $4,300 Software $390K, Hardware $190K, Cisco AS5850 s $130K

Power Cost In Thousands $450 $400 $350 $300 $250 $200 $150 $100 $50 $0 $390 $195 Software DSP, 298 Servers Dedicated DSP Chip, 149 Servers AC Power Plant $130 Purpose Built, 9 Cisco AS5850's

Codec Cost Codec g.711 $0, high bandwidth 64KB Codec g.729 $10, low bandwidth 8KB Using g.729 codec in this exercise drives up cost of software DSP solution by adding $285,720 Ultimately the network architecture will determine codec usage Also codec usage can change as requirements dictate

Cost Includes Interface Cards, Servers, Racks and Power $3,500 $3,000 $2,881 $2,500 In M illio n s $2,000 $1,500 $1,000 $1,218 $1,503 $1,643 $500 $0 Software DSP Solution Software DSP Solution with g.729 codec Dedicated DSP Soulution Cisco AS5850 Solution VoIP Service for 28,572 PSTN Trunks

Exercise Conclusion Cost exercise intended to show potential cost differential between two methods of converting analog voice to packet voice (Voice quality note) When implemented properly, there is no discernable difference between the two methods

Architecture and Integration Large deployments scale better when hardware platforms are separated by functionality Voice Mail PSTN Gateway Applications Call Conference Customer Aggregation Dial Plan Routing

PSTN Gateway Asterisk can be configured to perform PSTN GW functions to create a virtual VoIP channel per T1 channel In an existing Carrier environment, PSTN traffic through a Class 5 switch Three methods T1 PRI T1 GR303 T1 Dedicated

PSTN Gateway with PRI s Great way to efficiently use trunk capacity Can oversubscribe T1 links with numbers Bundling multiple PRI s gives fault protection Proven technology Provisioning Asterisk can be complex Dynamic nature of PRI s can add undue provisioning strains with data basing functions

PSTN Gateway with GR303 Newer protocol, also takes advantage of over subscribing T1 with numbers Bundling multiple T1 s gives fault protection GR303 Aggregator can be used in between Class 5 switch and Asterisk GW to enhance static soft switch provisioning

VoIP GW Provisioning Model New services require new processes for adds, moves and changes Automation determines level of human and machine involvement with provisioning Adding VoIP services into existing business processes can be complex Setup I/O database picks Correlating VoIP extensions to telephone # s Configuring features per customer Creating dial plan routing

VoIP GW Provisioning Model (cont) Using static VoIP connection channels to PSTN trunks provides minimal impact to daily provisioning tasks When VoIP channels are bonded to Class 5 switch channels, existing provisioning model can facilitate customer turn-up up The VoIP platform can be pre-provisioned provisioned and associated statically to existing databases

VoIP GW Provisioning Model (cont) Conceptually, a static soft switch platform can be achieved throughout all VoIP functions and applications CPE can be pre-assigned to customer aggregation channels Customer Aggregation channels can be pre-routed routed to PSTN GW channels Pre-assigned Voice Mail accounts can be associated with each customer channel

Application Servers Asterisk is modular software like Linux Asterisk can be setup as monolithic application server for specific task This allows server farms to grow as needed prescribed by the customer or function load per application PSTN GW server can handle 96 channels where the same server may handle 500 Voice Mail channels

Customer Aggregation Servers Setup aggregation servers for protocol type, SIP, IAX, MGCP, H323 Firewall each server according to protocol Statically map each CPE channel to a Customer aggregation channel on server Segment Customer IVR or ACD applications on the same server Aggregate multiple Customer groups with different dial plans and call handling routines

Dial Plan Servers Provides the core switching unit for the VoIP platform handling all call routing Uses IAX protocol between PSTN GW, Customer Aggregation and Application servers Drops out of call path once transfer is made Dual server, hardware and software fault redundant, self diagnosis for fail over and load sharing Can reload dial plan without restarting service

Operations and Maintenance 2 maybe 3 VoIP specialist as server quantity increases Create dial plan consistent with best practices of provisioning within host Telco Keep track of software updates that add features or fix bugs Track server usage and resource loading Plan for increasing capacity when needed

Software and Hardware Improvements Future plans, look for increase port density for interface cards, i.e. DS3 Look for added protocols, i.e. SS7 Look for VoIP Hardware Vendors adding IAX protocol Look for integration with presence systems, i.e. Jabber, AIM, MSN

Conclusion State of technology has never been or will ever be static Don t rely on equipment Vendors to add new products to your service portfolio Telco s pronounce themselves as technology companies, but sell services built by Vendors Embrace the ability to build on a stable and robust VoIP platform, Asterisk

Conclusion (cont) Asterisk is a project that delivers into the hands of the novice and experienced alike, a platform for developing telecom applications that bridge the gap between the existing TDM technology and future VoIP technology. Asterisk is well suited for small scale customers and if deployed properly, robust enough to deliver VoIP services to a very, very large customer base.