Visão Geral

Curso Voice Over IP FundamentalsDescubra como e por que a Voz sobre IP funciona e entenda o que é SIP.  Neste Curso Voice Over IP Fundamentals somente com palestras, você aprenderá conceitos básicos de como o Protocolo de Internet (IP) carrega um pacote de Voz sobre IP (VoIP). Você aprenderá os fundamentos da arquitetura do Protocolo de Iniciação de Sessão (SIP), serviços IP relacionados ao SIP, as vantagens e desvantagens do Tronco SIP, bem como o Protocolo Relacionado à Qualidade de Serviço (QoS).

Objetivo

Após realizar este Curso Voice Over IP Fundamentals você será capaz de:

• Conceitos básicos de como o Protocolo de Internet (IP) transporta um pacote VoIP
• Vantagens e desvantagens do SIP Trunking
• Entenda como DHCP e DNS oferecem suporte à telefonia IP
• Protocolo de Transporte em Tempo Real (RTP)
• Protocolo de Iniciação de Sessão (SIP) – Configuração de chamada, Mensagens Instantâneas, Presença
• Protocolo de Descrição de Sessão (SDP)
• Proxy SIP, Controlador de Borda de Sessão (SBC) e softswitch SIP
• Análise do Protocolo de Controle de Gateway de Mídia (MGCP)
• Arquitetura MGCP
• Como implementar QoS para garantir a mais alta qualidade de voz em suas redes IP
• O impacto do jitter, latência e perda de pacotes em redes VoIP
• Como o Wireshark pode decodificar e solucionar problemas de fluxos de chamadas RTP, SIP e MGCP
• Discuta o trixbox Softswitch e o proxy SIP
• Discuta gateways SIP e softphones

Publico Alvo

Esta Curso Voice Over IP Fundamentals é para pessoas que precisam entender a tecnologia VoIP. Gerentes de TI, pessoal técnico de vendas/marketing, consultores, designers e engenheiros de rede, engenheiros de design de produtos desenvolvendo produtos de serviços integrados, técnicos e gerentes de telecomunicações integrando serviços de PBX em redes de dados e administradores de sistemas que gerenciarão uma rede convergente se beneficiariam deste curso.

Pre-Requisitos

• TCP/IP Networking

Materiais

Conteúdo Programatico

Packetizing Voice

1. Telephony Architecture
2. Introduction to the VoIP Standards
3. Connecting VoIP to PSTN
4. Traffic Engineering
5. PSTN to VoIP Using Magic
6. Voice Digitization
7. Companding Mu-Law vs. A-Law
8. Time Division Circuit Switching
9. Voice Packet
10. The 20-Millisecond Voice Packet
11. The 60-Millisecond Voice Packet
12. The Voice Packet Header
13. Other Voice Packet Sample Sizes
14. Voice Packet Analysis
15. Voice Packet Analysis: Other Voice Packet Sample Sizes
16. QoS Overview
17. Latency
18. Packet Loss
19. Jitter
20. Controlling Delay
21. Sources of Delay
22. The First Voice Packet
23. The Second Voice Packet
24. The Third Voice Packet
25. Jitter Buffer Under Perfect Conditions
26. An Adaptive Jitter Buffer

SIP Trunking

1. The Legacy Circuit Switch
2. VoIP Phases
3. VoIP Phase 1: LAN Connect the Line Side
4. VoIP Phase 2: Decompose the Switch Cabinet
5. VoIP Phase 3: Shrink the MGs and Add Survivability
6. VoIP Phase 4: Add SIP Trunking
7. VoIP Phase 5: Eliminate the Old MGs
8. VoIP Phase 6: Add EMUN
9. VoIP Phase 7: Mass Acceptance of SIP Trunking with ENUM?
10. SIP Trunking Costs
11. Other Means of Connection
12. The “Old PBX” can do SIP Trunking if the Vendor Offers the Software
13. SIP Trunking Protocols
14. Peer-to-Peer RTP
15. Hairpin RTP
16. Disadvantages and Advantages of SIP Trunking
17. Disadvantages
18. Advantages
19. ITSPs
20. SIP Trunking Examples
21. SIP Trunk Outbound Call
22. Public VoIP

VoIP in the LAN

1. IP and Ethernet
2. A Sample Ethernet Switched Network
3. MAC Addresses
4. IP MAC Address Learning
5. known Destination MAC Addresses
6. Flood the Broadcast
7. Response to Flooded Packet
8. Learning Port Information
9. Switching
10. MAC Table Aging
11. Ethernet Communications Limits
12. Virtual LANs
13. VLAN Trunk
14. VLAN Tags
15. Untagged Frames
16. Port-Based VLANs
17. Broadcast Frame in VLAN 10
18. VLAN Trunking for VoIP Phones
19. IEEE 802.3af Device Detection
20. IEEE 802.3af Power Classifications
21. QoS at Layer 2
22. VLAN Tagging Process
23. IEEE 802.1q Frame Tagging

IP Networking

1. One-Way vs. Both-Way Routing
2. Static Routing
3. Subnet Masks and Routing
4. Routing and Switching
5. Routing Protocols
6. Distance Vector Routing
7. Link-State Routing
8. TCP/IP Review
9. Transmission Control Protocol (TCP) vs. User Datagram Protocol (UDP)
10. Connection-Oriented Protocol (TCP)
11. TCP/IP Packet Format and Operation
12. Connectionless Protocols (UDP)
13. UDP Packet
14. DNS
15. Basic Method of DNS
16. Dial Plan Essentials
17. Dial Plan Example
18. Digit Map
19. Enbloc vs. Overlap
20. Common Modifications to REGEX
21. Symbols
22. Regular Expressions
23. Metacharacters
24. Matching
25. Normalization Examples

SIP-Related IP Services

1. DHCP Option for SIP
2. DHCP Discover
3. DHCP Offer
4. Root-Level Domain Registration
5. Basic Method of DNS
6. Why Start with ENUM?
7. ENUM: NAPTR Query
8. ENUM: NAPTR Response
9. Locating SIP Servers: An Example
10. NAPTR Response
11. SRV Query
12. SRV Response
13. A Record Query
14. Regular Expressions
15. The Metacharacters

Voice Compression

1. Voice Compression Hardware
2. ASICs
3. DSPs
4. Mean Opinion Scores
5. Codecs
6. G.711, G.723.1, G.726
7. G.728 and G.729
8. Voice Compression
9. Formants
10. The Predictor
11. PCM Sampling
12. Voice Compression Algorithms
13. ADPCM Compression
14. Vocoder
15. G.729 Example
16. Codec Comparison Exercise
17. Zero Packet Loss
18. Ten Percent Packet Loss
19. Twenty Percent Packet Loss
20. T.38 Fax Spoofing
21. Call Setup
22. Discovering the Fax Tone
23. T.30 Negotiation
24. Shifting to 9.6 Kbps
25. T.38 Phase

Real-Time Transport Protocol (RTP)

1. RTP Architecture
2. RTP and RTP Control Protocol
3. Encapsulating the Voice Packet
4. RTP Ports
5. RTP Profile
6. Payload Types
7. Mapping Payload Type to Codec Type
8. How H.323 Identifies the Payload Type
9. NTP vs. RTP Timestamp
10. RTP Timestamps
11. RTP Timestamps and Silence Suppression
12. RTP Timestamps and Jitter Calculation
13. Controlling Jitter
14. Jitter Buffer Delay
15. Mixers
16. Synchronization Source
17. Conference Bridge Adds CSRC
18. RTP Header
19. UDP Packet with RTP Header and Voice
20. Required Fields
21. Version
22. Padding Bit
23. Extension Bit
24. CSRC
25. Market Bit
26. Payload Type
27. Sequence Number
28. Timestamp
29. SSRC
30. The Format-Specific Parameter (fmtp) Attribute
31. RFC 2833 Example: A Dialing Event
32. Transmitter Processing
33. Receiver Processing
34. Controlling Serialization Delay
35. Perfect Candidate for LFI and RTP Header Compression
36. RTP Header Compression Process (RFC 2508)
37. RTP Header Compression Format
38. RTCP
39. RTCP QoS: Round-Trip Delay Calculation
40. Sender Reports
41. Receiver Reports
42. Source Descriptions
43. Source Description Items
44. Other RTCP Packets

SIP Architecture

1. SIP User Agents
2. SIP Requests (Methods)
3. SIP Response Codes
4. SIP Proxy
5. SIP Back-to-Back UA
6. Session Border Controller
7. Forking Proxy
8. SIP Redirect Proxy
9. Global SIP Architecture
10. Overview of Operation
11. Classic SIP Trapezoid
12. INVITE Request
13. Session Description Protocol
14. Proxy Function
15. 180 Response
16. 200 Final Response
17. BYE
18. INVITE and ACK
19. SIP Functional Stack
20. SIP Core Documents and Extensions

SIP Call Flow Examples

1. SIP Call Analysis
2. SIP Registration with Authentication
3. SIP Call without INVITE Authentication
4. The 100rel Process
5. Busy Number
6. Abandoned Call (Cancel)
7. SIP Redirect (Call Forward)
8. Call Transfer
9. E&M Tie Trunk
10. See a Problem
11. Solution: SIP 183 Response

Session Description Protocol

1. Session Description Protocol
2. v= Header
3. o= Header
4. s= Header
5. c= Header
6. t= Header
7. m= Header
8. a= Header
9. Offer/Answer Model
10. Offer/Answer: Example 1
11. Offer/Answer: Example 2
12. SDP Offer/Answer Rules
13. UPDATE Method
14. RTP SEND and RECV Defined
15. Media Direction and RTCP
16. How RTCP Works
17. Placing a Call on HOLD

SIP NAT Traversal

1. SIP NAT Traversal
2. One-Way Voice Results
3. Full Cone NAT
4. IP Address Restricted NAT
5. Port Restricted NAT
6. Symmetric NAT
7. Simple Traversal of UDP through NATs
8. Traversal Using Relay NAT
9. NAT with Embedded SIP Proxy
10. Public VoIP Example

Media Gateway Control Protocol (MGCP)

1. Protocol Comparison
2. MGCP Call Model
3. Hairpin Call Example
4. Defined Endpoints
5. MGCP Commands
6. MGCP Syntax Example
7. Return Codes
8. Return Code Table
9. Parameter Lines
10. DTMF Package
11. Line Package
12. Digit Maps
13. MGCP Trace Procedure
14. MGCP Trace (Steps 1-8)
15. MGCP Trace (Steps 9-14)
16. MGCP Trace (Steps 15-22)
17. MGCP Trace (Steps 23-28)
18. MGCP Established Call
19. MGCP Trace (Steps 29-36)
20. MGCP Trace (Steps 37-40)

Queuing

1. CoS vs. QoS
2. Leaky Bucket
3. First In, First Out
4. Type Classification
5. Session ID Classification (Fair Queuing)
6. Dequeuing
7. 16. QoS-Related Protocol
8. Sources of Delay
9. Packetization Delay
10. Algorithmic Delay (Look Ahead)
11. Coder Processing Delay (Think Time)
12. Queuing Delay
13. Serialization Delay
14. Low-Speed Link
15. How 56-Kbps Links Cause Jitter
16. Upgrade to T1/E1 and Prioritize Voice
17. QoS Technology Solutions: Differentiated Services (DiffServ)
18. Supporting a VoIP Call with DiffServ
19. ToS Field
20. DiffServ Process at the Edge Router
21. DiffServ Process in the Core
22. DiffServ Highlights
23. Traffic Engineering: An Art Form
24. Measuring Engineering
25. Grade of Service
26. Appendix A: Glossary
27. Appendix B: H.323