Voice over IP Foundations Training

Who Should Attend?

• Core concepts of how Internet Protocol (IP) carries a VoIP packet
• Advantages and disadvantages of SIP Trunking
• Understand how DHCP and DNS supports IP telephony
• Real-Time Transport Protocol (RTP)
• Session Initiation Protocol (SIP) – Call set up, Instant Messaging, Presence
• Session Description Protocol (SDP)
• SIP proxy, Session Border Controller (SBC), and SIP softswitch
• Media Gateway Control Protocol (MGCP) analysis
• MGCP architecture
• How to implement QoS to ensure the highest voice quality over your IP networks
• The impact of jitter, latency, and packet loss on VoIP networks
• How Wireshark can decode and troubleshoot RTP, SIP, and MGCP call flows
• Discuss trixbox Softswitch and SIP proxy
• Discuss SIP gateways and softphones

Module 1 Packetizing Voice

• Telephony Architecture
• Introduction to the VoIP Standards
• Connecting VoIP to PSTN
• Traffic Engineering
• PSTN to VoIP Using Magic
• Voice Digitization
• Companding Mu-Law vs. A-Law
• Time Division Circuit Switching
• Voice Packet
• The 20-Millisecond Voice Packet
• The 60-Millisecond Voice Packet
• The Voice Packet Header
• Other Voice Packet Sample Sizes
• Voice Packet Analysis
• Voice Packet Analysis: Other Voice Packet Sample Sizes
• QoS Overview
• Latency
• Packet Loss
• Jitter
• Controlling Delay
• Sources of Delay
• The First Voice Packet
• The Second Voice Packet
• The Third Voice Packet
• Jitter Buffer Under Perfect Conditions
• An Adaptive Jitter Buffer

Module 2 SIP Trunking

• The Legacy Circuit Switch
• VoIP Phases
• VoIP Phase 1: LAN Connect the Line Side
• VoIP Phase 2: Decompose the Switch Cabinet
• VoIP Phase 3: Shrink the MGs and Add Survivability
• VoIP Phase 4: Add SIP Trunking
• VoIP Phase 5: Eliminate the Old MGs
• VoIP Phase 6: Add EMUN
• VoIP Phase 7: Mass Acceptance of SIP Trunking with ENUM?
• SIP Trunking Costs
• Other Means of Connection
• The “Old PBX” can do SIP Trunking if the Vendor Offers the Software
• SIP Trunking Protocols
• Peer-to-Peer RTP
• Hairpin RTP
• Disadvantages and Advantages of SIP Trunking
• Disadvantages
• Advantages
• ITSPs
• SIP Trunking Examples
• SIP Trunk Outbound Call
• Public VoIP

Module 3 VoIP in the LAN

• IP and Ethernet
• A Sample Ethernet Switched Network
• MAC Addresses
• IP MAC Address Learning
• known Destination MAC Addresses
• Flood the Broadcast
• Response to Flooded Packet
• Learning Port Information
• Switching
• MAC Table Aging
• Ethernet Communications Limits
• Virtual LANs
• VLAN Trunk
• VLAN Tags
• Untagged Frames
• Port-Based VLANs
• Broadcast Frame in VLAN 10
• VLAN Trunking for VoIP Phones
• IEEE 802.3af Device Detection
• IEEE 802.3af Power Classifications
• QoS at Layer 2
• VLAN Tagging Process
• IEEE 802.1q Frame Tagging

Module 4 IP Networking

• One-Way vs. Both-Way Routing
• Static Routing
• Subnet Masks and Routing
• Routing and Switching
• Routing Protocols
• Distance Vector Routing
• Link-State Routing
• TCP/IP Review
• Transmission Control Protocol (TCP) vs. User Datagram Protocol (UDP)
• Connection-Oriented Protocol (TCP)
• TCP/IP Packet Format and Operation
• Connectionless Protocols (UDP)
• UDP Packet
• DNS
• Basic Method of DNS
• Dial Plan Essentials
• Dial Plan Example
• Digit Map
• Enbloc vs. Overlap
• Common Modifications to REGEX
• Symbols
• Regular Expressions
• Metacharacters
• Matching
• Normalization Examples

Module 4 SIP-Related IP Services

• DHCP Option for SIP
• DHCP Discover
• DHCP Offer
• Root-Level Domain Registration
• Basic Method of DNS
• Why Start with ENUM?
• ENUM: NAPTR Query
• ENUM: NAPTR Response
• Locating SIP Servers: An Example
• NAPTR Response
• SRV Query
• SRV Response
• A Record Query
• Regular Expressions
• The Metacharacters

Module 5 Voice Compression

• Voice Compression Hardware
• ASICs
• DSPs
• Mean Opinion Scores
• Codecs
• G.711, G.723.1, G.726
• G.728 and G.729
• Voice Compression
• Formants
• The Predictor
• PCM Sampling
• Voice Compression Algorithms
• ADPCM Compression
• Vocoder
• G.729 Example
• Codec Comparison Exercise
• Zero Packet Loss
• Ten Percent Packet Loss
• Twenty Percent Packet Loss
• T.38 Fax Spoofing
• Call Setup
• Discovering the Fax Tone
• T.30 Negotiation
• Shifting to 9.6 Kbps
• T.38 Phase

Module 6 Real-Time Transport Protocol (RTP)

• RTP Architecture
• RTP and RTP Control Protocol
• Encapsulating the Voice Packet
• RTP Ports
• RTP Profile
• Payload Types
• Mapping Payload Type to Codec Type
• How H.323 Identifies the Payload Type
• NTP vs. RTP Timestamp
• RTP Timestamps
• RTP Timestamps and Silence Suppression
• RTP Timestamps and Jitter Calculation
• Controlling Jitter
• Jitter Buffer Delay
• Mixers
• Synchronization Source
• Conference Bridge Adds CSRC
• RTP Header
• UDP Packet with RTP Header and Voice
• Required Fields
• Version
• Padding Bit
• Extension Bit
• CSRC
• Market Bit
• Payload Type
• Sequence Number
• Timestamp
• SSRC
• The Format-Specific Parameter (fmtp) Attribute
• RFC 2833 Example: A Dialing Event
• Transmitter Processing
• Receiver Processing
• Controlling Serialization Delay
• Perfect Candidate for LFI and RTP Header Compression
• RTP Header Compression Process (RFC 2508)
• RTP Header Compression Format
• RTCP
• RTCP QoS: Round-Trip Delay Calculation
• Sender Reports
• Receiver Reports
• Source Descriptions
• Source Description Items
• Other RTCP Packets

Module 7 SIP Architecture

• SIP User Agents
• SIP Requests (Methods)
• SIP Response Codes
• SIP Proxy
• SIP Back-to-Back UA
• Session Border Controller
• Forking Proxy
• SIP Redirect Proxy
• Global SIP Architecture
• Overview of Operation
• Classic SIP Trapezoid
• INVITE Request
• Session Description Protocol
• Proxy Function
• 180 Response
• 200 Final Response
• BYE
• INVITE and ACK
• SIP Functional Stack
• SIP Core Documents and Extensions

Module 8 SIP Call Flow Examples

• SIP Call Analysis
• SIP Registration with Authentication
• SIP Call without INVITE Authentication
• The 100rel Process
• Busy Number
• Abandoned Call (Cancel)
• SIP Redirect (Call Forward)
• Call Transfer
• E&M Tie Trunk
• See a Problem
• Solution: SIP 183 Response

Module 9 Session Description Protocol

• Session Description Protocol
• v= Header
• o= Header
• s= Header
• c= Header
• t= Header
• m= Header
• a= Header
• Offer/Answer Model
• Offer/Answer: Example 1
• Offer/Answer: Example 2
• SDP Offer/Answer Rules
• UPDATE Method
• RTP SEND and RECV Defined
• Media Direction and RTCP
• How RTCP Works
• Placing a Call on HOLD

Module 10 SIP NAT Traversal

• SIP NAT Traversal
• One-Way Voice Results
• Full Cone NAT
• IP Address Restricted NAT
• Port Restricted NAT
• Symmetric NAT
• Simple Traversal of UDP through NATs
• Traversal Using Relay NAT
• NAT with Embedded SIP Proxy
• Public VoIP Example

Module 11 Media Gateway Control Protocol (MGCP)

• Protocol Comparison
• MGCP Call Model
• Hairpin Call Example
• Defined Endpoints
• MGCP Commands
• MGCP Syntax Example
• Return Codes
• Return Code Table
• Parameter Lines
• DTMF Package
• Line Package
• Digit Maps
• MGCP Trace Procedure
• MGCP Trace (Steps 1-8)
• MGCP Trace (Steps 9-14)
• MGCP Trace (Steps 15-22)
• MGCP Trace (Steps 23-28)
• MGCP Established Call
• MGCP Trace (Steps 29-36)
• MGCP Trace (Steps 37-40)

Module 12 Queuing

• CoS vs. QoS
• Leaky Bucket
• First In, First Out
• Type Classification
• Session ID Classification (Fair Queuing)
• Dequeuing
• 16. QoS-Related Protocol
• Sources of Delay
• Packetization Delay
• Algorithmic Delay (Look Ahead)
• Coder Processing Delay (Think Time)
• Queuing Delay
• Serialization Delay
• Low-Speed Link
• How 56-Kbps Links Cause Jitter
• Upgrade to T1/E1 and Prioritize Voice
• QoS Technology Solutions: Differentiated Services (DiffServ)
• Supporting a VoIP Call with DiffServ
• ToS Field
• DiffServ Process at the Edge Router
• DiffServ Process in the Core
• DiffServ Highlights
• Traffic Engineering: An Art Form
• Measuring Engineering
• Grade of Service
• Appendix A: Glossary
• Appendix B: H.323