This hands-on VoIP training course provides essential VoIP and data networking knowledge including - how VoIP works, why VoIP works, and how to use it. On the first day of training, participants will configure an IP network using Cisco routers and switches, learning IP fundamentals in order to make VoIP easier to understand. The remaining four days of training will focus on VoIP and IP telephony. The training is 60% practical and 40% lecture. The lecture portion of the training uses technically detailed slides that illustrate the subject matter. Text-only slides are kept to a minimum. The practical portion consists of 30 hands-on, skills-building labs where attendees will gain proficiency with some of the most popular VoIP software such as Wireshark, trixbox (formerly Asterisk@Home), Linksys Ethernet phone, SIP-based ATA, SIP-based Server and PBX products from Brekeke Software, Inc. The training course includes 1-year access to a 50-book Online Networking Reference Library with titles specially selected to reinforce course concepts. You'll Learn...
Who Would Benefit Agenda 1. Packetizing Voice
2. VoIP in the LAN
3. IP Networking
4. TCP / IP and UDP
5. SIP-Related IP Services
6. Voice Encoding and Compression
7. Real-Time Transport Protocol (RTP)
8. SIP Architecture
9. SIP Uniform Resource Indicators (URIs)
10. SIP Call Flow Examples Review how SIP calls are set up for applications like PSTN, instant messaging, VoIP, and more in this technical, in-depth analysis of the protocol.
11. SIP Syntax
12. Session Description Protocol
13. SIP NAT Traversal
14. Media Gateway Control Protocol (MGCP)
15. H.323
16. Queuing
17. QoS Related Networking Protocols
Labs (30) Lab 1: Install the network hardware. Lab 2: Configure Cisco IOS Command Line Interface via Telnet and console port access. Lab 3: Configure VLAN for secure voice and data separation. Lab 4: Configure an IP network using static routing. Lab 5: Configure a DNS zone, NAPTR, SRV, and A records as needed to support VoIP services. Lab 6: Configure DHCP services on your LAN to support VoIP gateways and phones. Lab 7: Call without a SIP proxy. Lab 8: Register a UA with a proxy. Lab 9: Configure a SIP Ethernet phone. Lab 10: Configure the SIP Server. Lab 11: Network SIP Proxies. Lab 12: Configure a SIP softphone. Lab 13: Configure a Wi-Fi radio. Lab 14: Configure a Wi-Fi SIP phone. Lab 15: Use Ethereal and Port Spanning to capture and analyze RTP. Lab 16: Configure various CODECs and make test calls to compare voice quality (G.711, G.729, and G.723.1). Lab 17: Reduce bandwidth consumption by 50% or more by increasing packet intervals and witness the QoS tradeoff. Lab 18: CODEC bandwidth testing. Test the amount of bandwidth actually consumed by different types of voice compression. Lab 19: Silence suppression and witness any QoS tradeoff. Activate and test silence suppression. Lab 20: Configure automatic CODEC negotiation and observe how SIP negotiates CODECS (OFFER/ANSWER). Lab 21: DTMF RFC 2833 and SIP INFO. Configure two different techniques that support accurate and reliable DTMF transmission. Lab 22: Use Ethereal to capture and analyze RTCP (QoS) reports. Lab 23: SIP REGISTER authentication. Configure a SIP phone to authenticate prior to joining a SIP network. Lab 24: SIP INVITE authentication. Configure a SIP proxy to confirm the calling party prior to processing the call. Lab 25: SIP Call Flow analysis. Using Wireshark, analyze typical call processing such as a normal call, busy call, abandoned call, and call transfer. Learn how to use Wireshark to troubleshoot problems with call processing. Lab 26: Configure trixbox. Configure the system that promises " a PBX is 30 minutes". Configure SIP extensions, voice mail, and meetme conference. Lab 27: SIP Trunking. Configure SIP trunking between two SIP PBXs, and learn the process of connecting to the PSTN using ITSP rather than buying your own PSTN gateways and connecting using conventional TDM or analog methods. Lab 28: Install SolarWinds Engineer's Edition and use WAN Killer and SolarWinds SNMP tools to test QoS performance. Lab 29: Configure diff-serv on your VoIP gateway. Lab 30: Configure various queuing strategies, apply service policies on your router, and witness the results. Perform file transfer and voice services on the same network and witness the results of proper and poor QoS configuration.
Registration
Fees Class begins at 8:30 AM each day and concludes at 4:30 PM unless otherwise directed. Please arrive early on the first day to sign-in and meet fellow attendees. If you register less than a week in advance of a class, please bring your confirmation letter. Business casual attire is appropriate. To register, click on the "Book Now" button or please call (708) 246-0320 Seminar Schedule
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Payment is due prior to the Seminar. Cancellation Policy. Registrants may cancel up to fourteen days in advance of the seminar start date for a full refund, less administrative fees of $500. Or, you may transfer your registration to another member of your company at no additional charge. Registrants canceling within fourteen days of the seminar will receive training credit, less administrative fees of $500 toward any other Resource Center seminar. In the unlikely event that a seminar must be cancelled, you will be notified at least two weeks prior to the seminar date. Seminar provider is not responsible for losses due to cancellation including losses on advanced purchase airfares. As seminars are cancelled for under-enrollment from time to time, we strongly recommend that registrants traveling by air purchase only refundable tickets. |
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The Resource Center for Customer Service
Professionals LLC
PO Box 401, Western Springs, IL 60558
Tel: (708) 246-0320 Fax: (708)
246-0251
Copyright © 2005-2008 Resource
Center for Customer Service Professionals, LLC. All rights reserved.
Last modified May 12, 2008