Voice over IP Foundations Training Class

1. VoIP Fundamentals

• 
  Key architectural VoIP components


• 
  End-to-end voice transmission


• 
  Packetizing voice (encapsulation)


• 
  Transmission time allocation


• 
  QoS and capacity considerations


• 
  Sources of delay
  
  
  
  • 
    Coder processing delay (Think Time)
  
  
  • 
    Algorithmic delay (Look Ahead)
  
  
  • 
    Packetization delay
  
  
  • 
    Serialization delay
  
  
  • 
    Queuing delay
  
  
  
  


• 
  Jitter buffer function


• 
  VoIP QoS requirements: Packet Loss, Latency, Jitter

2. VoIP in the LAN

• 
  MAC address


• 
  IP address and ARP


• 
  Ethernet switching


• 
  Logical and physical segmentation


• 
  VLAN - 802.1Q/P


• 
  802.3af - Power over Ethernet (POE)

3. IP Networking

• 
  IP addressing


• 
  Static routing


• 
  OSPF


• 
  EIGRP

4. TCP and UDP

• 
  Transmission Control Protocol (TCP)


• 
  VoIP protocols that use TCP


• 
  User Datagram Protocol (UDP)


• 
  VoIP protocols that use UDP

5. IP Services

• 
  DNS


• 
  How SIP uses DNS


• 
  DHCP


• 
  How IP telephony uses DHCP

6. Voice Encoding and Compression

• 
  G.711 u-law and A-law


• 
  G.729


• 
  G.723.1

7. Real-Time Transport Protocol (RTP)

• 
  Dealing with Packet Loss, Latency, Jitter


• 
  How various protocols defines the RTP session
  
  
  
  • 
    Session Description Protocol
  
  
  • 
    H.245 Terminal Capabilities
  
  
  • 
    The RTP profile
  
  
  • 
    The RTP payload type field
  
  
  • 
    RTP telephony events (RFC 2833)
  
  
  
  


• 
  How RTP removes jitter


• 
  How RTP handles packet loss


• 
  How RTP identifies the talking party


• 
  How RTP handles silence suppression


• 
  How RTP is used to mix voice (conference calls)


• 
  The RTP header


• 
  RTP Control Protocol (RTCP)
  
  
  
  • 
    SDES
  
  
  • 
    Sender/receiver reports
  
  
  • 
    Bye reports
  
  
  
  

8. SIP Architecture

• 
  SIP architecture
  
  
  
  • 
    Proxy: stateful, stateless, call stateful, Session Border Controller
  
  
  
  


• 
  SIP methods: INVITE, ACK, BYE, CANCEL, REGISTER, INFO, PRACK, etc.


• 
  SIP response codes: 1xx, 2xx, 3xx, 4xx, 5xx, 6xx


• 
  SIP headers (To:, From:, Call-ID:, Allows: Required, Via)


• 
  Session Description Protocol (SDP)


• 
  SIP Addressing, Session Control, and Call Setup

9. SIP Uniform Resource Indicators (URIs)

• 
  Understand the format of SIP URIs and how URIs interoperate with PSTN dialing
  plans, email systems, and web pages


• 
  Generic URI information (RFC 2396)


• 
  Direct or Proxy


• 
  PSTN number (RFC 2808)


• 
  Instant messaging


• 
  Presence


• 
  In registrations

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.

• 
  Call attempt - unsuccessful


• 
  Presence subscription


• 
  Registration


• 
  Presence notification


• 
  Instant Message Exchange


• 
  Call setup - successful


• 
  Cancel


• 
  Vacant number


• 
  100rel


• 
  www authenticate

11. Media Gateway Control Protocol (MGCP)

• 
  Architecture


• 
  Verbs: CRCX, MDCX


• 
  Responses


• 
  Packages (DTMF, Line, Trunk, Generic, etc.)


• 
  Parameter lines


• 
  Sample call flow protocol analysis

12. H.323

• 
  ASN.1 primer


• 
  H.323 architecture
  
  
  
  • 
    Gatekeeper
  
  
  • 
    Gateway
  
  
  • 
    MCU
  
  
  • 
    Terminal
  
  
  
  


• 
  H.323 versions


• 
  H.323 gatekeeper controlled call flow example

13. Queuing

• 
  Priority queuing


• 
  Weighted fair queuing


• 
  Weighted precedence


• 
  Traffic policing and traffic shaping


• 
  Low latency queuing


• 
  The effects of data traffic and fair queuing on VoIP


• 
  Mixing voice and data traffic effectively


• 
  Determining bandwidth needs for voice traffic


• 
  Assessing the impact of voice on data networks
  
  
  
  • 
    Low speed links
  
  
  • 
    High speed links
  
  
  
  

14. QoS Related Networking Protocols

• 
  Differentiated Services (DiffServ)


• 
  Call Admission Control

This class is for people who need to understand VoIP technology. IT managers, technical sales/mar