SIP.edu Project TERENA VoIP Workshop
|
|
|
- Adam Carr
- 8 years ago
- Views:
Transcription
1 SIP.edu Project TERENA VoIP Workshop Dennis Baron June 5, 2005 Page 1
2 SIP.edu Goals Grow SIP connectivity on the Internet Increase value proposition for end-user SIP adoption Promote converged identity - -style addressing Provide a useful service; while supporting enhanced campus services and experiments Means Publishing cookbook with several alternative recipes Obtaining corporate sponsorship and promotional pricing Cisco, Avaya, Pulver.com so far Build community of SIP practitioners Page 2
3 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:dbaron@mit.edu It s time to put E.164 phone numbers behind us! A.G. Bell did not say: , come here. I need you! Page 3
4 SIP.edu Architecture (Phase 1) SIP User Agent DNS SRV query sip.udp.bigu.edu INVITE DNS SIP Proxy INVITE SIP- PRI / CAS bigu.edu telephonenumber where mail= bob Directory Bob's Phone Page 4
5 SIP.edu 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. INVITE bigu.edu location DB SIP Registrar REGISTER (Contact: ) Bob's SIP Phone Page 5
6 SIP.edu 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 Connects to existing or Centrex Could also connect to proprietary VoIP system Page 6
7 SIP.edu 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 extension eg. SIP proxy checks dial plan Routes call to gateway rings phone Page 7
8 SIP.edu Configuration SIP Server Internet2 Network SIP user wants to call DNS Server SIP/PRI LDAP Server PSTN Page 8
9 DNS SRV Lookup DNS SRV Internet2 SIP Server Network DNS lookup for MIT.EDU DNS Server SIP/PRI LDAP Server PSTN Page 9
10 SIP INVITE Internet2 SIP SIP Server Network SIP INVITE to DNS Server SIP/PRI LDAP Server PSTN Page 10
11 LDAP Lookup Internet2 SIP Server LDAP Network LDAP lookup for dbaron returns x21232 DNS Server SIP/PRI LDAP Server PSTN Page 11
12 Call Sent to SIP Server Internet2 Network SIP INVITE to x21232 via DNS Server SIP SIP/PRI LDAP Server PSTN Page 12
13 Media Stream via to Internet2 RTP SIP Server Network SIP user talks to at x21232 DNS Server SIP/PRI LDAP Server PSTN Page 13
14 Sip to SIP Calling Internet2 RTP SIP Server Network SIP user talks to at his SIP phone DNS Server SIP/PRI LDAP Server PSTN Page 14
15 SIP to and PSTN Calling SIP Server Internet2 Network SIP user calls or DNS Server SIP/PRI LDAP Server PSTN Page 15
16 and PSTN to SIP Calling SIP Server Internet2 DNS Server LDAP Server Network SIP/PRI PSTN user calls or user calls PSTN Page 16
17 SIP.edu Reachable Users Page 17
18 SIP.edu 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 18
19 SIP.edu Just do it! Questions? Page 19
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
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
Implementing SIP.edu
Implementing SIP.edu Internet2 Spring Member Meeting Dennis Baron April 26, 2006 Page 0 Agenda SIP.edu Basics ISN Why SIP.edu? MIT Case Study Page 1 Basics Page 2 Goals SIP Connectivity Build a large base
Internet2 Member Meeting SIP.edu Initiative
Member Meeting SIP.edu Initiative Dennis Baron October 14, 2003 Page 1 MIT Implementation Page 2 SIP@mit.edu Implementation Pingtel SIPxchange SIP proxy server SIP registrar server SIP media server (voice
Internet2 and the Enterprise
Internet2 and the Enterprise Spring VON, Santa Clara, CA March 31 st, 2004 Agenda Overview of Internet2 Projects Ben Teitelbaum, Internet2 Henning Schulzrinne, Columbia University University Case Studies
Internet2's Voice and Integrated Communications Initiatives
Internet2's Voice and Integrated Communications Initiatives Ben Teitelbaum http://people.internet2.edu/~ben/ SI-501 Mini-Talk Ann Arbor, MI February 9 th, 2005 Voice Not just a crusty
SIP-based VoIP Deployment in Taiwan
SIP-based VoIP Deployment in Taiwan Aaron Solomon (a.k.a. Dr. Quincy Wu in Taiwan) TWAREN solomon@ipv6.club.tw 2004.01.29 1 Outline Introduction to TWAREN NTP SIP-based VoIP Platform Plans of VoIP Working
Connective Middleware for Voice and Integrated Communications
Connective Middleware for Voice and Integrated Communications Submission to TERENA Networking Conference 2004 Ben Teitelbaum < ben@internet2.edu > 3025 Boardwalk, Suite 200 Ann Arbor, MI 48108, USA Keywords
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
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
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...
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
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 len_evenchik@harvard.edu sip:len.evenchik@harvard.edu AT&T Dimension PBX, 1980 Lecture Agenda Welcome
High Availability Configuration Guide
New Rock Technologies, Inc. MX Voice-Fax Gateway Series HX4E MX8A MX60 MX120 Website: http://www.newrocktech.com Email: gs@newrocktech.com Document Version: 201509 Contents 1 Overview... 1-1 1.1 Function
SIP Basics. CSG VoIP Workshop. Dennis Baron January 5, 2005. Dennis Baron, January 5, 2005 Page 1. np119
SIP Basics CSG VoIP Workshop Dennis Baron January 5, 2005 Page 1 Outline What is SIP SIP system components SIP messages and responses SIP call flows SDP basics/codecs SIP standards Questions and answers
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.
Ram Dantu. VOIP: Are We Secured?
Ram Dantu Professor, Computer Science and Engineering Director, Center for Information and Computer Security University of North Texas rdantu@unt.edu www.cse.unt.edu/~rdantu VOIP: Are We Secured? 04/09/2012
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. (saverio.niccolini@netlab.nec.de) VoIP and Videoconferencing 1 st
High Availability Configuration Guide
New Rock Technologies, Inc. MX Voice-Fax Gateway Series High Availability Configuration Guide HX4 MX8 MX60 MX120 http://www.newrocktech.com Tel: +86 21-61202700 Fax: +86 21-61202704 Document Version: TB0-E002-P
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
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 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
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
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
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
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
Application Notes for configuring Avaya IP Office IP500 R7.0 with 2Ring NetFAX R3.0 Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for configuring Avaya IP Office IP500 R7.0 with 2Ring NetFAX R3.0 Issue 1.0 Abstract These Application Notes describe the configuration steps
(Refer Slide Time: 6:17)
Digital Video and Picture Communication Prof. S. Sengupta Department of Electronics and Communication Engineering Indian Institute of Technology, Kharagpur Lecture - 39 Video Conferencing: SIP Protocol
VoIP Application Note:
VoIP Application Note: Configure NEC UX5000 w/ BroadVox SIP Trunking Service P/N 0913226 Date: 8/12/09 Table of Contents: GOAL... 3 PREREQUISITES... 3 SIP TRUNKING INFORMATION PROVIDED BY BROADVOX:...
How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions
How to Configure the Avaya IP Office 6.1 for use with Integra Telecom SIP Solutions Overview This document provides a reference for configuration of the Avaya IP Office to connect to Integra Telecom SIP
IP Telephony Technology. IEEE ComSoc Meeting. Corey Coffin, SE Cisco Systems
IP Telephony Technology IEEE ComSoc Meeting Corey Coffin, SE Cisco Systems 1 Agenda IP Convergence Call Manager Deployment Models Case Studies Summary 2 Toll Bypass and IP Telephony PBX PBX Router/GW IP
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.
Session Initiation Protocol (SIP) The Emerging System in IP Telephony
Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia
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
Connecting with Free IP Call
Connecting with Free IP Call Free IP Call (http://www.freeipcall.com/) offers telephone service using the VoIP standard SIP. The service allow users making/receiving VoIP calls to/from VoIP telephone numbers
Release Notes for NeoGate TE100 16.18.0.X
Release Notes for NeoGate TE100 16.18.0.X ===Firmware Version: V16.18.0.2==== Applicable Model: NeoGate TE100 Release Date: October 25th, 2014 http://www.yeastar.com 1/6 1. New Features 1. Added support
An Introduction to VoIP Protocols
An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this
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
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
Towards Junking the PBX: Deploying IP Telephony. What is a PBX?
Towards Junking the : Deploying IP Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh Columbia University {wenyu,lennox,hgs,kns10}@cs.columbia.edu We describe our departmental IP telephony
Citrix EasyCall Gateway Pre-Installation Checklist
Citrix EasyCall Gateway Pre-Installation Checklist This checklist is for EasyCall Gateway administrators. Information for telephony system integrators is provided in the EasyCall Gateway Telephony System
NAT TCP SIP ALG Support
The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the
Application Note. SIP Domain Management
Application Note SIP Domain Management 28 March 2008 Table of Contents 1 WHAT IS A SIP DOMAIN?... 1 2 LOCAL SIP DOMAIN... 2 3 OTHER SIP DOMAIN... 3 4 DNS CONSIDERATIONS... 4 5 USING A PUBLIC DNS... 5 6
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
Japan Registry Service. ENUM Trial in Japan. NGI2 & IPv6 DNS Operation Workshop 5 Dec 2003 Yoshiro YONEYA <yone@jprs.co.jp> 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.
Mediatrix 3000 with Asterisk June 22, 2011
Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration
Configuration Notes 290
Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...
Lab Hours. We need to allocate 3 hours in this week for hands-on lab hours ( Nov 13 th 14:10-17:00).
Lab Hours We need to allocate 3 hours in this week for hands-on lab hours ( Nov 13 th 14:10-17:00). 17:00). The instructor will set up the SIP server. Every student will bring a labtop or desktop PC and
Best Practices for Securing IP Telephony
Best Practices for Securing IP Telephony Irwin Lazar, CISSP Senior Analyst Burton Group Agenda VoIP overview VoIP risks Mitigation strategies Recommendations VoIP Overview Hosted by VoIP Functional Diagram
Ray McElheney Phone Systems. Let us show you how to save money With our VoIP Phone Systems
Ray McElheney Phone Systems Let us show you how to save money With our VoIP Phone Systems Why Switch to VoIP? The savings - Our customers save 40% - 70% off traditional phone service. VoIP gives you more
SIP and PSTN Connectivity. Jiri Kuthan, iptel.org sip:jiri@iptel.org September 2003
SIP and PSTN Connectivity Jiri Kuthan, iptel.org sip:jiri@iptel.org September 2003 Outline PSTN Gateways. PSTN2IP Demo Integration challenges: CLID Interdomain Trust Gateway Location Outlook: Reuse of
CIPT1_8- Implementing Cisco Unified Communications Manager Part 1
CIPT1_8- Implementing Cisco Unified Communications Manager Part 1 Description Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8.0 prepares you for implementing a Cisco Unified Communications
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
Introduction to DE-CIX NGN. Andreas Sturm andreas.sturm@de-cix.net
Introduction to DE-CIX NGN Andreas Sturm andreas.sturm@de-cix.net Today s Situation: How does it look like? Current situation Carrier s Data and Voice business operations are controlled by different departments
How to Build a Simple Virtual Office PBX System Using TekSIP and TekIVR
How to Build a Simple Virtual Office PBX System Using TekSIP and TekIVR This document explains how to build a simple virtual office PBX system using TekSIP and TekIVR. In this example following components
SIP Trunking Quick Reference Document
SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG
Internet Standards - Emergency Services
Internet Standards - Emergency Services Hannes Tschofenig (Chair IETF ECRIT Working Group) Mail comments to Hannes.Tschofenig@nsn.com and/or ecrit@ietf.org. Architectural Considerations Layer 7 VoIP, Inc.
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
TALKSWITCH VOIP NETWORK TROUBLESHOOTING GUIDE
TALKSWITCH DOCUMENTATION TALKSWITCH VOIP NETWORK TROUBLESHOOTING GUIDE RELEASE 3.24 CT.TS005.008001 ANSWERS WITH INTELLIGENCE COPYRIGHT INFORMATION TalkSwitch. Copyright 2006. All Rights Reserved. Reproduction,
159.334 Computer Networks. Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT)
Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT) Presentation Outline Basic IP phone set up The SIP protocol Computer Networks - 1/2 Learning Objectives
ACCELERATOR 6.3 TDM PBX INTEGRATION GUIDE
ACCELERATOR 6.3 TDM PBX INTEGRATION GUIDE April 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 2004-2014
SIP Tutorial. VoIP Workshop Terena 2005 Poznan Poland. By Stephen Kingham mailto:stephen.kingham@aarnet.edu.au sip:stephen.kingham@aarnet.edu.
SIP Tutorial VoIP Workshop Terena 2005 Poznan Poland By Stephen Kingham mailto:stephen.kingham@aarnet.edu.au sip:stephen.kingham@aarnet.edu.au Stephen Kingham Copyright Stephen Kingham 2004 This work is
Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Microsoft Office Communications Server 2007 R2 and Avaya IP Office PSTN Call Routing - Issue 1.0 Abstract These Application
Implementing Cisco Unified Communications Manager Part 1 Course IPT1 v9.0; 5 Days, Instructor-led
Implementing Cisco Unified Communications Manager Part 1 Course IPT1 v9.0; 5 Days, Instructor-led Course Description Implementing Cisco Unified Communications Manager, Part 1 (IPT1) v9.0 prepares you for
VoIP Server Reference
IceWarp Server VoIP Server Reference Version 10 Printed on 12 August, 2009 i Contents VoIP Service 1 Introduction... 1 V10 New Features... 3 SIP REFER... 3 SIP Call Transfer Agent Settings... 3 NAT traversal
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
Configuring Positron s V114 as a VoIP gateway for a 3cx system
Assumptions: Configuring Positron s V114 as a VoIP gateway for a 3cx system The IP address of the V114 is 192.168.1.2 The IP address of the 3CX PBX System is 192.168.1.110 3CX already has some IP phones
Application Note: Cisco Integration with Onsight Connect
Application Note: Cisco Integration with Onsight Connect Table of Contents Application Note:... 1 Cisco Integration with Onsight Connect... 3 Direct Onsight Device to Cisco Endpoint Calls... 3 Cisco Unified
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
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
Implementing Cisco Unified Communications Manager Part 1, Volume 1
Implementing Cisco Unified Communications Manager Part 1, Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your Training
How to make free phone calls and influence people by the grugq
VoIPhreaking How to make free phone calls and influence people by the grugq Agenda Introduction VoIP Overview Security Conclusion Voice over IP (VoIP) Good News Other News Cheap phone calls Explosive growth
Cisco Unified Communications IP Telephony Part 1 (CIPT1) v8.0. Course Objectives. Associated Certification. Required Exam(s) Price.
Cisco Unified Communications IP Telephony Part 1 (CIPT1) v8.0 Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8.0 prepares you for implementing a Cisco Unified Communications Manager
Connecting with Vonage
Connecting with Vonage Vonage (http://www.vonage.com/) offers telephone service using the VoIP (Voice over Internet Protocol) standard SIP (Session Initiation Protocol). The service allow users making
4602 SIP Telephone Release 1.1 Quick Setup Guide
4602 SIP Telephone Release 1.1 Quick Setup Guide This Quick Setup guide is intended for technicians installing Avaya 4602 SIP telephones. Before performing the procedures in this guide, make sure the following
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.,
Expert Reference Series of White Papers. The Role of Session Manager in Applying Features to Calls via Sequenced Applications, Part 1
Expert Reference Series of White Papers The Role of Session Manager in Applying Features to Calls via Sequenced Applications, Part 1 1-800-COURSES www.globalknowledge.com The Role of Session Manager in
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 hvu@swin.edu.au 8/March/2006 Page 2 Swinburne University
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
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
Cisco EXAM - 300-075. Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product. http://www.examskey.com/300-075.
Cisco EXAM - 300-075 Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Buy Full Product http://www.examskey.com/300-075.html Examskey Cisco 300-075 exam demo product is here for you to test the
Connecting with sipgate
Connecting with sipgate sipgate (http://www.sipgate.co.uk/) offers telephone service using the VoIP standard SIP. sipgate covers every area code within the United Kingdom and provides a local Direct Inward
SIP: Ringing Timer Support for INVITE Client Transaction
SIP: Ringing Timer Support for INVITE Client Transaction Poojan Tanna (poojan@motorola.com) Motorola India Private Limited Outer Ring Road, Bangalore, India 560 037 Abstract-The time for which the Phone
NTP VoIP Platform: A SIP VoIP Platform and Its Services
NTP VoIP Platform: A SIP VoIP Platform and Its Services Speaker: Dr. Chai-Hien Gan National Chiao Tung University, Taiwan Email: chgan@csie.nctu.edu.tw Date: 2006/05/02 1 Outline Introduction NTP VoIP
SIP Proxy. SIP Proxy. Bicom SYSTEMS. SIP Proxy... Advanced Simplicity
Session border controller solution offering high performance, security and SIP compliance connectivity. 1 DIAGRAM : Registration SIP Client registration for all users (, Business, Hosted PBXware and Wholesale)
Mediatrix 4404 Step by Step Configuration Guide June 22, 2011
Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...
A Comparative Study of Signalling Protocols Used In VoIP
A Comparative Study of Signalling Protocols Used In VoIP Suman Lasrado *1, Noel Gonsalves *2 Asst. Prof, Dept. of MCA, AIMIT, St. Aloysius College (Autonomous), Mangalore, Karnataka, India Student, Dept.
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
Mobility User mobility Service mobility 3GPP
Advanced Networking Voice over IP & Other Multimedia Protocols Some SIP Advanced Topics Renato Lo Cigno Renato.LoCigno@disi.unitn.it Mobility User mobility Service mobility 3GPP ENUM Advanced Networking
ConneXon s response to Ofcom Consultation Document
ConneXon s response to Ofcom Consultation Document Changes to General Conditions and Universal Service Conditions Detailed Response to Question 4: Do you agree with our proposals for emergency call numbers
Configuration Notes for Aastra MX-ONE in Ascom IP-DECT System
Configuration Notes for in Ascom IP-DECT System Contents 1 Introduction... 1 1.1 Abbreviations and Glossary... 1 2 General... 2 3 IP-DECT Configuration... 2 3.1 DECT... 2 3.2 Users... 3 4 MX-ONE Configuration...
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
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
ScopTEL TM IP PBX Software. Back to Back SIP Trunking Configuration
Back to Back SIP Trunking Configuration Usage Cases Usage Cases Implementing DNIS: SIP TIE trunks: A private network is created to dial extensions between systems using Access Codes Tandem Dialing: PSTN
Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670
Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Businesses Save Money with Toshiba s New SIP Trunking Feature Unlike gateway based solutions, Toshiba s MIPU/ GIPU8 card
VoIP Services. Maurice Duault mduault@cisco.com. 2001, Cisco Systems, Inc. All rights reserved.
VoIP Services Maurice Duault mduault@cisco.com 2001, Cisco Systems, Inc. All rights reserved. 1 Agenda Distributed Softswitch Residential Business VoIP access Multiservice over VPN Managed IP Telephony
nexvortex SIP Trunking Implementation & Planning Guide V1.5
nexvortex SIP Trunking Implementation & Planning Guide V1.5 510 S PRING S TREET H ERNDON VA 20170 +1 855.639.8888 Introduction Welcome to nexvortex! This document is intended for nexvortex Customers and
Application Notes Rev. 1.0 Last Updated: February 3, 2015
SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v8.6 for Level 3 Voice Complete SM Deployments Application Notes Rev. 1.0 Last Updated: February 3, 2015 Contents 1 Document Overview...
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