1. Packetizing Voice
- 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/IP Review
- Transmission Control
- Protocol (TCP)
- VoIP protocols that use TCP
- User Datagram Protocol (UDP)
- VoIP protocols that use UDP
5. SIP-Related 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 define the RTP session
- Session Description Protocol
- H.245 terminal capabilities
- 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. SIP Syntax
- Request Message
- Response Message
- The Start Line
- Via Header
- SIP Dialog
- From Header
- To Header
- Call-ID Header
- Dialog State
- CSeq Header
- Max-Forwards Header
- Proxy-Authenticate Header
- Contact Header
- Expires Header
- User-Agent Header
- Content-Length Header
- Allow Header
- Supported Header
- Content-Type Header
12. Session Description Protocol
- v= Header
- o= Header
- s= Header
- c= Header
- t= Header
- m= Header
- a= Header
- Offer/Answer Model
- SDP Offer/Answer Rules
- UPDATE Method
- RTP SEND and RECV Defined
- Media Direction and RTCP
- How RTCP Works
- Placing a Call on HOLD
13. 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 Exampl
14. Media Gateway Control Protocol (MGCP)
- Architecture
- Verbs: CRCX, MDCX
- Responses
- Packages (DTMF, Line, Trunk, Generic, etc.)
- Parameter lines\
- Sample call flow protocol analysis
15. H.323
- ASN.1 primer
- H.323 architecture
- Gatekeeper
- Gateway
- MCU
- Terminal
- H.323 versions
- H.323 gatekeeper-controlled call flow example
16. 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
17. QoS-Related Networking Protocols
- Differentiated Services (DiffServ)
- Call Admission Control
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 w