Implementing SIP.edu
|
|
|
- Ashlee James
- 10 years ago
- Views:
Transcription
1 Implementing SIP.edu Internet2 Spring Member Meeting Dennis Baron April 26, 2006 Page 0
2 Agenda SIP.edu Basics ISN Why SIP.edu? MIT Case Study Page 1
3 Basics Page 2
4 Goals SIP Connectivity Build a large base of SIP-reachable Internet2 users by making existing campus PBX, Centrex, and VoIP systems reachable via SIP SIP Addressing Facilitate the convergence of communications identities by promoting the use of addresses for voice and multimedia communications Page 3
5 Means Publishing cookbook with several alternative recipes Obtaining corporate sponsorship and promotional pricing Cisco, Avaya, etc. Build community of SIP practitioners Page 4
6 Architecture (Phase 1) SIP User Agent DNS SRV query sip.udp.bigu.edu INVITE DNS SIP Proxy INVITE SIP-PBX Gateway PRI / CAS PBX bigu.edu telephonenumber where mail= bob Campus Directory Bob's Phone Page 5
7 Architecture (Phase 2) SIP User Agent DNS DNS SRV query sip.udp.bigu.edu SIP Proxy INVITE If Bob has registered, ring his SIP phone; Else, call his extension through the PBX. INVITE bigu.edu location DB SIP Registrar REGISTER (Contact: ) Bob's SIP Phone Page 6
8 Why Phone NUMBERS? Users should not be burdened with device addresses, when it s people they really care about Addresses should be mnemonic and empower enterprises to manage the identities of their users sip:[email protected] It s time to put E.164 phone numbers behind us! A.G. Bell did not say: , come here. I need you! Page 7
9 Components DNS Server Add SIP SRV records to existing servers SIP Proxy Server Also acts as SIP registrar Can support aliases for legacy phone numbers Mimics campus dial plan LDAP Server (or other source of directory data) Has mapping of to phone number SIP Gateway Connects to existing PBX or Centrex Could also connect to proprietary VoIP system Page 8
10 Call Flow Example SIP DNS lookup for MIT.EDU points to SIP proxy Sends INVITE to to proxy SIP proxy checks MIT directory Maps call to PBX extension eg. SIP proxy checks dial plan Routes call to PBX gateway PBX rings phone Page 9
11 Examples All-in-One Asterisk as both proxy and gateway Soekris 4801 server with Sangoma T1/E1 card Approximately $675USD Astlinux (Asterisk + micro Linux) Directory lookup via file or LDAP Still under development and testing Page 10
12 Examples Vendor Solution Avaya SIP Converged Communications Server (CCS) Communications Manager Media Server and Gateway Handle Based Dialing service (LDAP plugin) SIP N' Go Starter Kit Page 11
13 gaps SIP is more than voice Video and IM are important too Presence services change the user experience Chickens without eggs only gets you half way We ve made everybody SIP reachable, now who s going to call them? The 12-digit keypad problem will be with us for awhile What do we do until the devices have a 21 st century user interface? Page 12
14 ISN Page 13
15 Old World / New World Radically new devices / services Deep bureaucratic hierarchy Telco provider control [email protected] The world is flat (almost) Be your own provider Page 14
16 How to SIP from a 12-key phone? Old World* IP Desk Phones Legacy Desk Phones Cell Phones Emerging New World PSTN * Transitional period during which we have to support these devices will last a long time! Solution: numeric aliases Page 15
17 ITAD Subscriber Numbers (ISN) 4257*260 locally assigned IP Telephony Administrative Domain (ITAD) ITADs Defined by Telephony Routing over IP (TRIP) [RFC3219] Globally unique Lots of them ( ) IANA is already set up to allocate ISN resolution works just like ENUM Page 16
18 Assigned ITADs (as of 3/15/06) Academic Internet2 Hofstra University UCLA MIT Stanford University of Alaska Fairbanks University of California, Berkeley Florida State University University of Manitoba University of Oregon Royal Institute of Technology NE Worcestershire College Trent University University of North Carolina Corporate Enterprises Sterling National Bank Apple Computer VoIP Service Providers Free World Dialup Government State of Oregon University of Texas, Austin Other Columbia University BizFu (web hosting) UCSD Manitoba New Democratic Party Taiwan Academic Network Packet Clearing House +36 others Stealth Communications SIPcall.com RCN Corporation VoIPteq SIP Broker VoIP Solution Providers Tello Iotum Digium Page 17
19 ISN Status Trial just starting up Supported by Internet2, Packet Clearing House, MIT, Tello ISN Cookbook Published Recipes for SER and Asterisk 103 ITADs assigned so far Page 18
20 ISN in Four Easy Steps 1. Request an ITAD from IANA Simple piece of Approximate two week turnaround 2. Publish your ITAD/ISN information in DNS Option1: Put full NAPTR in root zone *.xxx.freenum.org IN NAPTR "u" "E2U+sip Option2: Have root zone delegate to your own nameservers 3. Enable inbound ISN calling 4. Enable outbound ISN calling Option1: Native ISN lookup Option2: Using Tello SIP redirector Option3: Using Tello private ENUM Page 19
21 E.164 vs. GDS vs. ISN vs. SIP E.164 GDS ISN SIP AOR Example *260 [email protected] Familiarity Phone numbers H.323 video users Huh? addresses Delegating Authority ITU, national government, ViDeNet, national gatekeepers IANA ICANN, TLD registrars Address Structure Hierarchical / geographical Hierarchical / geographical/ organization local*domain local@domain Non-numeric Characters Ignored No * Only Yes Portability Varies by country??? With domain owner s cooperation With domain owner s cooperation With domain owner s cooperation Fragmentation Public ENUM + multiple private ENUMs One space One space One space Page 20
22 Why? Page 21
23 Motivations Provides a useful service Easy to get started Lots of options Facilitates inter-campus communications Opens the way for innovation Build I/T staff skills Help break down organization/cultural barriers Encourage early technology adopters Set PBX migration path Page 22
24 Quotes This project was initiated by the need to provide reliable, IP based phones for the Toolik Lake research station located north of the Brooks Range. University of Alaska Fairbanks sipeth: Internet Telefonie for the ETH Zurich: This project has been inspired by the Internet2 SIP.edu initiative. During the exploration process many new ideas have led to a new vision for our project. ETH Zurich Our SIP.edu infrastructure has allowed us to utilizing our Internet2 connections to reestablish the telephone tie lines connecting out two institutions. MIT and WHOI Page 23
25 Deployments Page 24
26 at MIT Page 25
27 MIT Integrated Comm. Project SIP.edu deployments On Pingtel SIPxchange in March 2003 Moved to OpenSER in 2005 ICP Goals Develop a next generation digital integrated communications services strategy Conduct experiments applying selected technologies in education, research colaboration, and community Page 26
28 ICP Experiments Presence service to support Plasma Fusion research Dormitory collaboration spaces Collaboration services for MIT Singapore Alliance Faculty virtual office hours Virtual communities for MIT Cambridge student exchange Page 27
29 ICP Outputs MIT WHOI tie lines Shuttletrack IVR tel: Media Lab Fluid Voice project ispots Voice mail to pilots VoIP pilot Page 28
30 MIT SIP Architecture Screening Incoming AuthN Outgoing Signing External Proxy External Proxy Internet Gateways Routing/Dial Plan AuthZ Accounting Internal Proxy Internal Proxy Gateways Services Services Vmail to Conferencing etc. Registration AuthN Personal Options Personal Proxy Personal Proxy Personal Proxy Services Personalization Page 29
31 MIT SIP Usage Calls per day Total MIT SIP calls over service lifetime /7/04 9/21/04 10/5/04 10/19/04 11/2/04 11/16/04 11/30/04 12/14/04 12/28/04 1/11/05 1/25/05 2/8/05 2/22/05 3/8/05 3/22/05 4/5/05 4/19/05 5/3/05 5/17/05 5/31/05 6/14/05 6/28/05 7/12/05 7/26/05 8/9/05 8/23/05 9/6/05 9/20/05 10/4/05 10/18/05 11/1/05 11/15/05 11/29/05 12/13/05 12/27/05 1/10/06 1/24/06 2/7/06 2/21/06 3/7/06 3/21/06 Date Page 30
32 Outstanding Tasks Generate call billing records Improved web interface Support for additional devices Improved voice mail integration PBX Message Waiting Indication IMAP integration Location management for 911, etc. Presence service Page 31
33 Conclusion Page 32
34 Questions? Page 33
35 More Information? SIP.edu Web Page Mailing list (see web page) Thursday conference calls (2:00 Eastern) SIP.edu Cookbook ISN Cookbook Page 34
36 More Information? Contact: Dennis Baron, MIT or, if you must! isn:21232*270 tel: Page 35
Implementing VoIP at an institution using the SIP.edu cookbook
Implementing VoIP at an institution using the SIP.edu cookbook SEEREN2 Winter School, Kopaonik, Serbia, VoIP workshop Dennis Baron Milivoje Mirovic, AMRES March 12 th, 2007. Page 0 Motivations Provides
SIP.edu Project TERENA VoIP Workshop
SIP.edu Project TERENA VoIP Workshop Dennis Baron June 5, 2005 Page 1 SIP.edu Goals Grow SIP connectivity on the Internet Increase value proposition for end-user SIP adoption Promote converged identity
SIP.edu Project. CSG VoIP Workshop. Dennis Baron January 5, 2005. Dennis Baron, January 5,2005 Page 1. np120
SIP.edu Project CSG VoIP Workshop Dennis Baron January 5, 2005 Page 1 SIP.edu Goals Grow SIP connectivity in Internet2 Increase value proposition for end-user SIP adoption Promote convergence of voice
Internet2 Member Meeting SIP.edu Initiative
Member Meeting SIP.edu Initiative Dennis Baron October 14, 2003 Page 1 MIT Implementation Page 2 [email protected] Implementation Pingtel SIPxchange SIP proxy server SIP registrar server SIP media server (voice
VoIP at MIT. Merit VoIP Seminar. Dennis Baron April 3, 2008. Dennis Baron, April 3, 2008 Page 1. np163
VoIP at MIT Merit VoIP Seminar Dennis Baron April 3, 2008 Page 1 Outline MIT VoIP Project Goals VoIP Service Models VoIP Architecture Rollout Plan Outstanding Issues Quick Questions Page 2 Voice over IP
Preparatory Meeting for Phase 2 of Philippine National ENUM Trial
Preparatory Meeting for Phase 2 of Philippine National Trial IP Telephony Group Advanced Science and Technology Institute Department of Science and Technology December 12, 2005 NCC-CICT Dialing Scheme
VoIP Project Objectives and Design
VoIP Review IT Partners Wednesday, July 23, 2007 1 Agenda Project objectives Financial overview Current status New VoIP architecture Enterprise services demo Personal sip accounts Timeline Strategic Issues
SIP-based VoIP Deployment in Taiwan
SIP-based VoIP Deployment in Taiwan Aaron Solomon (a.k.a. Dr. Quincy Wu in Taiwan) TWAREN [email protected] 2004.01.29 1 Outline Introduction to TWAREN NTP SIP-based VoIP Platform Plans of VoIP Working
IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week
Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.
MIT s Current SIP Infrastructure. Mark Silis MIT Information Services and Technology February 2, 2006
MIT s Current SIP Infrastructure Mark Silis MIT Information Services and Technology February 2, 2006 Current SIP Implementation Utilizes the IETF standards based SIP protocol Comprised of several different
ENUM: an Enabler for VoIP and Next Generation Services
ITU Workshop on Origin Identification and Alternative Calling Procedures (Geneva, Switzerland, 19-20(AM) 2012) ENUM: an Enabler for VoIP and Next Generation Services Steven D. Lind Senior Member of the
2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)
Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication
VoIP and Videoconferencing: are they the same?
VoIP and Videoconferencing: are they the same? Dr. Saverio Niccolini Research Staff Member now @ Network Laboratories, NEC Europe Ltd. ([email protected]) VoIP and Videoconferencing 1 st
Internet Voice, Video and Telepresence Harvard University, CSCI E-139. Lecture #5
Internet Voice, Video and Telepresence Harvard University, CSCI E-139 Lecture #5 Instructor: Len Evenchik [email protected] sip:[email protected] AT&T Dimension PBX, 1980 Lecture Agenda Welcome
The SIP School- 'Mitel Style'
The SIP School- 'Mitel Style' Course Objectives This course will take delegates through the basics of SIP into some very technical areas and is suited to people who will be installing and supporting SIP
AARNet VoIP update and peering VoIP
AARNet VoIP update and peering VoIP VoIP Summit at APAN Conference and Internet 2 Joint Tech s Conference January 2004 www.aarnet.edu.au H.323 Architecture AARNet Reliability QoS Monitor Billing System
Two Standards: H.323 / SIP Bridging both worlds. João Pereira - FCCN - Portugal
Two Standards: H.323 / SIP Bridging both worlds João Pereira - FCCN - Portugal 1 Presentation FCCN - Portuguese NREN (RCTS) administrator and a research unit; We ve been using H.323 videoconferencing since
SIP Essentials Training
SIP Essentials Training 5 Day Course Lecture & Labs COURSE DESCRIPTION Learn Session Initiation Protocol and important protocols related to SIP implementations. Thoroughly study the SIP protocol through
Multimedia Service Platform
Multimedia Service Platform Turnkey solution for SIP based communications Adrian Georgescu AG Projects http://ag-projects.com Current telecommunications landscape From the old PSTN only the E.164 numbering
Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2)
Overview Voice-over over-ip (VoIP) ENUM VoIP Introduction Basic PSTN Concepts and SS7 Old Private Telephony Solutions Internet Telephony and Services VoIP-PSTN Interoperability IP PBX Network Convergence
Juha Heinänen [email protected]
From Voice over IP to Voice over Internet Juha Heinänen [email protected] From VoIP to VoINET VoIP replaced wires in PBX and PSTN backbones with IP preserves the traditional, centralized telephony service
Japan Registry Service. ENUM Trial in Japan. NGI2 & IPv6 DNS Operation Workshop 5 Dec 2003 Yoshiro YONEYA <[email protected]> Copyright 2003 JPRS
ENUM Trial in Japan NGI2 & IPv6 DNS Operation Workshop 5 Dec 2003 Yoshiro YONEYA Background Typical ENUM world PSTN SIP Server MGW Location Servre Mail, ifax, SMS, etc. PSTN ENUM Infrastructure
Avaya IP Office 8.1 Configuration Guide
Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.
Internet Technology Voice over IP
Internet Technology Voice over IP Peter Gradwell BT Advert from 1980s Page 2 http://www.youtube.com/v/o0h65_pag04 Welcome to Gradwell Gradwell provides technology for every line on your business card Every
Future of VoIP. Patrik Fältström, [email protected]
Future of VoIP Patrik Fältström, [email protected] 1 Evolution of VoIP Phase Enterprise Data Applications VoIP Transport Connection Oriented Packet Based (IP) TDM SS7 over IP Emulation Terminals Terminal Emulation
Configuration Notes 0215
Mediatrix Digital and Analog VoIP Gateways DNS SRV Configuration for a Redundant Server Solution (SIP) Introduction... 2 Deployment Scenario... 2 DNS SRV (RFC 2782)... 3 Microsoft Server Configuration...
SIP Trunking DEEP DIVE: The Service Provider
SIP Trunking DEEP DIVE: The Service Provider Larry Keefer, AT&T Consulting UC Practice Director August 12, 2014 2014 AT&T Intellectual Property. All rights reserved. AT&T, the AT&T logo and all other AT&T
Deploying SIP Phones into Unified Communications Solutions with the Dialogic Media Gateway Series
Communication is an important part of business enterprises, and many are adopting Unified Communications (UC) solutions to improve business processes and enhance communications. These UC solutions typically
MODELLING OF INTELLIGENCE IN INTERNET TELEPHONE SYSTEM
MODELLING OF INTELLIGENCE IN INTERNET TELEPHONE SYSTEM Evelina Nicolova Pencheva, Vessela Liubomirova Georgieva Department of telecommunications, Technical University of Sofia, 7 Kliment Ohridski St.,
Ram Dantu. VOIP: Are We Secured?
Ram Dantu Professor, Computer Science and Engineering Director, Center for Information and Computer Security University of North Texas [email protected] www.cse.unt.edu/~rdantu VOIP: Are We Secured? 04/09/2012
How To Implement A Cisco Vip From Scratch
Overview of Cisco VoIP Infrastructure Solution for SIP The Cisco VoIP Infrastructure Solution for SIP implements a voice-over-packet network design using SIP to provide telephony services. It lays the
Building a Scalable Numbering Plan
Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user
White Paper. Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market.
White Paper Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market Produced For: Produced By: July 2006 The Evolving Role of Hardware as a
SIP Trunking using the EdgeMarc Network Services Gateway and the Mitel 3300 ICP IP-PBX
June 26th, 2014 SIP Trunking using the EdgeMarc Network Services Gateway and the Mitel 3300 ICP IP-PBX Page 1 of 30 Table of Contents 1 Overview... 3 2 Prerequisites... 3 3 Network Topology... 4 4 Description
PSTN IXC PSTN LEC PSTN LEC STP STP. Class 4. Class 4 SCP SCP STP. Switch. Switch STP. Signaling Media. Class 5. Class 5. Switch.
As we enter the 21st century, we are experiencing a telecommunications revolution. From a technological perspective, the distinction between voice information and other kinds of data is blurring as circuit-switched
Using Asterisk with Odin s OTX Boards
Using Asterisk with Odin s OTX Boards Table of Contents: Abstract...1 Overview...1 Features...2 Conclusion...5 About Odin TeleSystems Inc...5 HeadQuarters:...6 Abstract Odin TeleSystems supports corporate
SIP and PSTN Connectivity. Jiri Kuthan, iptel.org sip:[email protected] September 2003
SIP and PSTN Connectivity Jiri Kuthan, iptel.org sip:[email protected] September 2003 Outline PSTN Gateways. PSTN2IP Demo Integration challenges: CLID Interdomain Trust Gateway Location Outlook: Reuse of
Supporting Multiple PBXs in Hybrid Deployment Models
NetVanta Unified Communications Technical Note Supporting Multiple PBXs in Hybrid Deployment Models Application Note for Hybrid Deployments The goal of this technical note is to ensure that you can leverage
Development of SIP-H.323 Gateway Project
Development of SIP-H.323 Gateway Project Ruston Hutchens QUESTnet 2005 Thursday 7 th July v2 SIP-H.323 Gateway project Motivation Large deployment base of H.323 terminals (over 2.9 million calls placed
Taiwan SIP/ENUM Trial Project. Vincent W.S. Chen Executive Director, TWNIC Email: [email protected]
Taiwan SIP/ENUM Trial Project Vincent W.S. Chen Executive Director, TWNIC Email: [email protected] October 1, 2003 Agenda Project Structure Phase I Trial Objectives SIP/ENUM Trial Architecture Application
ENUM and VoIP. Numbering and Dialing Plans. RIPE 46 VoIP and ENUM Tutorial 1. September 2003. Richard STASTNY
ENUM and VoIP Numbering and Dialing Plans RIPE 46 VoIP and ENUM Tutorial 1. September 23 Richard STASTNY ÖFEG/TELEKOM AUSTRIA, Postbox 147, 113-Vienna enum:+43 664 42 41 E-Mail: [email protected]
BUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS
BUILDING LARGE CAMPUS ASTERISK-BASED PABX SYSTEMS.0 Background The abundance of Local Area Networks such as those found on most African Universities, and recent advances in Computer and Telephony integration
Avaya Aura SIP Trunking Training
Avaya Aura SIP Trunking Training 5 Day Course Lecture & Demo WHO NEEDS TO ATTEND This class is suited to those who are new to administering Avaya systems, need to know how to send calls through an Avaya
VoIP Survivor s s Guide
VoIP Survivor s s Guide Can you really save $, improve operations, AND achieve greater security and availability? Presented by Peggy Gritt, Founder and CEO of the VoIP A non-biased organization for the
Digium Switchvox AA65 PBX Configuration
Digium Switchvox SIP Trunking using Optimum Business SIP Trunk Adaptor and the Digium Switchvox AA65 IP-PBX v23695 Goal The purpose of this configuration guide is to describe the steps needed to configure
Introducing Cisco Voice and Unified Communications Administration Volume 1
Introducing Cisco Voice and Unified Communications Administration Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your
IP PBX SH-500N WWW.HIPERPBX.COM
IP PBX SH-500N COMPANIES THAT WANT TO EXPAND AND IMPROVE THEIR TELEPHONE SYSTEM IP PBX SH-500N The IP PBX SH-500N is designed for companies that want to expand and improve their telephone system, and/or
Application Notes for the Ingate SIParator with Avaya Converged Communication Server (CCS) - Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for the Ingate SIParator with Avaya Converged Communication Server (CCS) - Issue 1.0 Abstract These Application Notes describe the configuration
Impact of enum and IP telephony
Impact of enum and IP telephony Scott Bradner Harvard University IPtel - 1 The Internet as Global Telephone Co. u key Internet concept: end-to-end (e2e) hosts connected to Internet can communicate without
Software-Powered VoIP
Software-Powered VoIP Ali Rohani Anthony Murphy Scott Stubberfield Unified Communications Architecture Core Scenarios UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN
Delivering UC Solutions UC Summit
Siemens Enterprise Communications Delivering UC Solutions UC Summit David Leach Unified Communications Consultant Siemens Enterprise Communications 1 2012 Siemens Enterprise Communications, Inc. Business
Introducing Personeta
Introducing Personeta Solution for Next Generation Converged Services Agenda Business trends The Solution Introducing Personeta Value Proposition Architecture TappS Services Key selling points 2 Communication
Asterisk: A Non-Technical Overview
Asterisk: A Non-Technical Overview Nasser K. Manesh [email protected] Millenigence, Inc. 5000 Birch St., Suite 8100 Newport Beach, CA 92660 June 2004, Revised December 2004 Executive Summary Asterisk
SIP A Technology Deep Dive
SIP A Technology Deep Dive Anshu Prasad Product Line Manager, Mitel June 2010 Laith Zalzalah Director, Mitel NetSolutions What is SIP? Session Initiation Protocol (SIP) is a signaling protocol for establishing
SIP and ENUM. Overview. 2005-03-01 ENUM-Tag @ DENIC. Introduction to SIP. Addresses and Address Resolution in SIP ENUM & SIP
and ENUM 2005-03-01 ENUM-Tag @ DENIC Jörg Ott 2005 Jörg Ott 1 Overview Introduction to Addresses and Address Resolution in ENUM & Peer-to-Peer for Telephony Conclusion 2005 Jörg Ott
mobile unified communications client and docking station
FREQUENTLY ASKED QUESTIONS mobile unified communications client and docking station What are the target customer characteristics of a Mobile UC subscriber? + Verizon Wireless as mobile carrier. Mobile
Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment. JR Richardson
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,
Kerio Operator. Getting Started Guide
Kerio Operator Getting Started Guide 2011 Kerio Technologies. All rights reserved. 1 About Kerio Operator Kerio Operator is a PBX software for small and medium business customers. Kerio Operator is based
Asterisk PBX Features
Asterisk PBX Features Automated Attendant Allows callers to be automatically transferred to a user s extension without the intervention of a receptionist. Some businesses use the PBX telephone system to
Cisco TelePresence Video Communication Server Basic Configuration (Control with Expressway)
Cisco TelePresence Video Communication Server Basic Configuration (Control with Expressway) Deployment Guide Cisco VCS X8.1 D14651.08 August 2014 Contents Introduction 4 Example network deployment 5 Network
How Telecom Italia Empowers Customer Service from the IMS Cloud
How Telecom Italia Empowers Customer Service from the IMS Cloud Giacomo De Filippis, IPCC Senior Program Manager, Telecom Italia Guillaume Calot, Strategic Business Director EMEA, Alcatel-Lucent Enterprise
Truly Unified Communications. This could be your corporate network:
The Comdasys MC Solution brings the features of your IP-PBX to your mobile phone in the palm of your hand. Its two components, the MC Controller and the MC Client, will enhance your business mobility.
Telephony Telephony more than just a phone system.
Telephony Telephony more than just a phone system. Telephony En Pointe ECS provides an end-to-end telephony solution based around the latest bleeding edge technology from Cisco Systems the Cisco Hosted
Kyle Haefner Communications Programmer Telecommunications Colorado State University. Open Source Telecommunications. Applications to Education
Kyle Haefner Communications Programmer Telecommunications Colorado State University Open Source Telecommunications Applications to Education The Ongoing Dilemma Current telephony equipment is becoming
Avaya Aura SIP Trunking Training
Avaya Aura SIP Trunking Training 5 Day Course Lecture & Demo WHO NEEDS TO ATTEND This class is suited to those who are new to administering Avaya systems and would like to know more about the SIP protocol.
VoIP Services. Maurice Duault [email protected]. 2001, Cisco Systems, Inc. All rights reserved.
VoIP Services Maurice Duault [email protected] 2001, Cisco Systems, Inc. All rights reserved. 1 Agenda Distributed Softswitch Residential Business VoIP access Multiservice over VPN Managed IP Telephony
Voice over IP Basics for IT Technicians
Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements
OfficeServ 7100 IP-PBX. SIP Trunking using the Optimum Business Sip Trunk Adaptor and the Samsung
SIP Trunking using the Optimum Business Sip Trunk Adaptor and the Samsung OfficeServ 7100 IP-PBX 1 Table of Contents 1. Overview 3 2. Prerequisites 3 3. OfficeServ 7100 PBX Configuration 3 3.1 Network
November 2013. The Business Value of SIP Trunking
November 2013 S P E C I A L R E P O R T The Business Value of SIP Trunking Table of Contents Introduction... 3 What Is SIP Trunking?... 3 What Is the Demand for SIP Trunking?... 5 How Does SIP Trunking
THINKTEL COMMUNICATIONS DIGIUM G100/G200 PRI OVER IP SIP TRUNKING
THINKTEL COMMUNICATIONS DIGIUM G100/G200 PRI OVER IP SIP TRUNKING TA B L E O F C O N T E N T S 1.1 NETWORK DIAGRAM... 3 1.2 COLLABORATION OF MONARQUE TELECOM... 3 1.3 CONNECTING TO THE DIGIUM G100... 4
Implementation notes on Integration of Avaya Aura Application Enablement Services with Microsoft Lync 2010 Server.
Implementation notes on Integration of Avaya Aura Application Enablement Services with Microsoft Lync 2010 Server. Introduction The Avaya Aura Application Enablement Services Integration for Microsoft
Cisco Unified Communications 500 Series
Cisco Unified Communications 500 Series IP PBX Provisioning Guide Version 1.0 Last Update: 02/14/2011 Page 1 DISCLAIMER The attached document is provided as a basic guideline for setup and configuration
Internet Telephony Terminology
Internet Telephony Terminology Understanding the business phone system world can be a daunting task to a lay person who just wants a system that serves his or her business needs. The purpose of this paper
ACCELERATOR 6.3 ASTERISK 1.4 INTEGRATION GUIDE
ACCELERATOR 6.3 ASTERISK 1.4 INTEGRATION GUIDE October 2014 Tango Networks, Inc. phone: +1 469-229-6000 3801 Parkwood Blvd, Suite 500 fax: +1 469-467-9840 Frisco, Texas 75034 USA www.tango-networks.com
SIP Trunk Configuration Guide. using
SIP Trunk Configuration Guide using www.cbeyond.net 1-877-441-9783 The information contained in this document is specific to setting up SIP connections between Vertical SBX IP 320 and Cbeyond. If you require
Chapter 2 PSTN and VoIP Services Context
Chapter 2 PSTN and VoIP Services Context 2.1 SS7 and PSTN Services Context 2.1.1 PSTN Architecture During the 1990s, the telecommunication industries provided various PSTN services to the subscribers using
Device SIP Trunking Administrator Manual
Table of Contents Device SIP Trunking Administrator Manual Version 20090401 Table of Contents... 1 Your SIP Trunking Service... 2 Terminology and Definitions... 2 PBX, IP-PBX or Key System... 2 Multi-port
VoIP & Internet Telephony
VoIP & Internet Telephony Peter Gradwell www.gradwell.com 01225 800 800 3-3a Gloucester St, Bath, BA1 2SE www.gradwell.com 1 Gradwell dot com Ltd Founded 1998 whilst at University 7 Staff Bath, Bristol,
DNS SRV Usage June 22, 2011
DNS SRV Usage June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Scope of this Document... 3 DNS SRV (RFC 2782)... 3 Deployment Scenario... 4 Mediatrix Unit Configuration... 5 Scenario
Network Overview. Background Traditional PSTN Equipment CHAPTER
CHAPTER 1 Background Traditional PSTN Equipment Traditional telephone services are engineered and offered over the public switched telephone network (PSTN) via plain old telephone service (POTS) equipment
Summary - ENUM functions that maps telephone numbers to Internet based addresses - A description and the possible introduction to Sweden
DATE REFERENCE NO. 30 March 2001 01-9734 Summary - ENUM functions that maps telephone numbers to Internet based addresses - A description and the possible introduction to Sweden AUTHOR Joakim Strålmark
Detecting Spam in VoIP Networks. Ram Dantu Prakash Kolan
Detecting Spam in VoIP Networks Ram Dantu Prakash Kolan More Multimedia Features Cost Why use VOIP? support for video-conferencing and video-phones Easier integration of voice with applications and databases
EXPLOITING SIMILARITIES BETWEEN SIP AND RAS: THE ROLE OF THE RAS PROVIDER IN INTERNET TELEPHONY. Nick Marly, Dominique Chantrain, Jurgen Hofkens
Nick Marly, Dominique Chantrain, Jurgen Hofkens Alcatel Francis Wellesplein 1 B-2018 Antwerp Belgium Key Theme T3 Tel : (+32) 3 240 7767 Fax : (+32) 3 240 8485 E-mail : [email protected] Tel : (+32)
Table of Contents. Confidential and Proprietary
Table of Contents About Toshiba Strata CIX and Broadvox SIP Trunking... 1 Requirements... 2 Purpose, Scope and Audience... 3 What is SIP Trunking?... 4 Business Advantages of SIP Trunking... 4 Technical
Using DNS SRV to Provide High Availability Scenarios
AN-SBC-100 Sangoma Session Border Controllers Using DNS SRV to Provide High Availability Scenarios Contents 1. Sangoma Session Border Controllers - High Availability Solution...1 2. What is DNS SRV?...1
Peer to Peer Settlement for Next Generation IP Networks Using the ETSI OSP Protocol (ETSI TS 101 321) for Cascading Peering Settlements
Peer to Peer Settlement for Next Generation IP s Using the ETSI OSP Protocol (ETSI TS 101 321) for Cascading Peering Settlements Table of Contents 1 Introduction... 1 2 Requirements... 2 3 The ETSI Open
VoIP and IP-IC Regulatory aspects
VoIP and IP-IC Regulatory aspects Giovanni Santella [email protected] What is VoIP? VoIP call scenarios (1) PC-to-PC PC-to-Phone VoIP call scenarios (2) Phone-to-Phone The value chain for the provision
ACCELERATOR 6.3 ASTERISK LINES INTEGRATION GUIDE
ACCELERATOR 6.3 ASTERISK LINES INTEGRATION GUIDE January 2014 Tango Networks, Inc. phone: +1 469-229-6000 3801 Parkwood Blvd, Suite 500 fax: +1 469-467-9840 Frisco, Texas 75034 USA www.tango-networks.com
Asterisk & ENUM. Extending the Open Source PBX. Michael Haberler, IPA Otmar Lendl, nic.at
Asterisk & ENUM Extending the Open Source PBX Michael Haberler, IPA Otmar Lendl, nic.at Why a ENUM-enable a PBX? your PBX doubles as an IP/PSTN gateway for your existing numbers becomes a dual contact
FOR COMPANIES THAT WANT TO EXPAND AND IMPROVE THEIR TELEPHONE SYSTEM
IP PBX VH-500 FOR COMPANIES THAT WANT TO EXPAND AND IMPROVE THEIR TELEPHONE SYSTEM IP PBX VH-500 The Virtual IP PBX VH-500 is an unified communication system hosted in the cloud, and it's an excellent
Outline. VoIP Research Workshop February, Canberra. VoIP Workshop. VoIP in AARNet (+) Group discussion. Summary, what s next?
VoIP Research Workshop February, Canberra HAI VU Outline VoIP Workshop VoIP in AARNet (+) Group discussion Summary, what s next? http://caia.swin.edu.au [email protected] 8/March/2006 Page 2 Swinburne University
