VoIP
Duration: 2 days

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Synopsis This course introduces the myriad protocols used to convey voice and video over the Internet Protocol. It explains both the legacy H.323 protocol stack and the SIP standard, as well as the RTP protocol common to both. This course also discusses SCCP ("Skinny"), a proprietary protocol used in legacy Cisco IP telephony. Students obtain hands-on experience using custom H.323 and SIP capable software.
Target Audience VoIP system developers and implementors. Network Administrators
Prerequisites
Objectives
  • Explain the basic concepts of VoIP
  • Explain how VoIP differs from traditional (SS7) phone systems
  • Explain the inner workings of the H.323 stack
  • Debug H.323 connectivity issues
  • Explain the SCCP (Skinny) protocol
  • Explain the SIP protocol, both version 1 and version 2
  • Debug SIP connectivity issues
Exercises Sample exercises include setting up two fully functional VoIP systems - one based on H.323 (with Microsoft Netmeeting), and the other on SIP (using softphones)
Modules
Day 1
1. Introduction to VoIP
1 hours
This module presents the basic concepts of Voice over IP (VoIP) and Video over IP (VVoIP) in a clear and thorough way. These concepts are then solidified and built upon during the rest of the course.
  • What is VoIP? What is VVoIP?
    • Circuit switching vs. packet switching
      • Digital vs. analog
        • Constraints - bandwidth, latency, and more
          • Codecs
            2. H.323
            1 hours
            This module presents the H.323 protocol suite, or "Stack". We cover the various devices and myriad protocols using for signaling, and demonstrate use cases and call establishment.
            • H.323 Terminology - Terminals, Gateways, Gatekeepers and MCUs
              • H.323 standards: v2 through v6
                • H.323 Vendors and products
                  • The H.323 Protocol Stack
                    • H.323 call flow
                      3. H.225.0 - RAS
                      1 hours
                      We focus on the Registration/Admission/Status protocol, or RAS - a key component of H.323 and the protocol behind the Gatekeepers.
                      • RAS explained
                        • Registration functions
                          • Endpoint lookup
                            • Admission functions
                              • Status functions
                                4. H.225.0 - Q.931
                                1 hours
                                Next, we discuss Q.931 - the signaling protocol used in call virtual circuit establishment:
                                • What is Q.931?
                                  • Message Sequence: Setup/Call Proceeding/Progress/Alerting/Connect
                                    • Gatekeeper assisted vs. Gatekeeper routed
                                      5. H.245
                                      1 hours
                                      The final component is H.245, which is used to establish the actual call channels.
                                      • What is H.245?
                                        • Message Sequence: MasterSlave/TerminalCapability/OpenLogicalChannel
                                          • Fast Start
                                            • Security
                                              • Firewall compatibility and H.323v6
                                                6. SCCP (Skinny)
                                                1-2 hours
                                                The Selsius Call Control Protocol (affectionally known as "Skinny") is a proprietary protocol that drives legacy Cisco phones. Poorly document, it was inherited by Cisco upon the acquisition of Selsius, and is still used in Call Manager. This module describes it in as much detail as possible, showing actual captures and packet disassembly.
                                                • SCCP? Skinny?
                                                  • Message Sequence
                                                    • Sample call flows
                                                      • Analyzing SCCP
                                                        Day 2
                                                        7. RTP/RTCP
                                                        1-2 hours
                                                        H.323, Skinny, or SIP - the actual call data, be it voice, video or data, is carried in RTP. This so-called "Real Time Protocol" and its annex "Real Time Control Protocol" (RTCP) are used to convey data with minimal delay.
                                                        • RTP Messages
                                                          • Header Compression
                                                            • Sample flows
                                                              • RTCP
                                                                • Determining QoS parameters and call quality with RTCP
                                                                  8. SIP
                                                                  3-4 hours
                                                                  The Session Initiation Protocol (SIP) in both v1 (RFC2543) and v2 (RFC3261) is the force behind modern VoIP and messaging systems. We cover this HTTP-like protocol in depth.
                                                                  • SIP standards
                                                                    • SIP extensions
                                                                      • Sample SIP Flow
                                                                        • SIP Requests
                                                                          • SIP Responses
                                                                            • Headers
                                                                              • Error/Response codes
                                                                                • SDP
                                                                                  • SIP Firewalling issues