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VoIP Protocols: SIP and H.323

Chapter Description

In this sample chapter from CCNP Collaboration Call Control and Mobility CLACCM 300-815 Official Cert Guide, you will review the function components of Session Initial Protocol (SIP), exam Session Description Protocol (SDP) fundamentals, and examine the H.323 communication protocol.

This chapter includes the following main topics:

  • Overview of SIP: This section provides a brief history of SIP and describes its functional components, the different methods commonly used in SIP, and the process by which a call is set up and torn down.

  • Introduction to SDP: This section examines Session Description Protocol (SDP) fundamentals and describes the offer/answer framework.

  • Overview of H.323: This section covers H.323 basics, including the various components typically included in H.323 networks and the flow for a typical H.323 call.

This chapter covers the following CLACCM 300-815 exam topics:

  • ■ 1.1 Troubleshoot these elements of a SIP conversation

    • ■ 1.1.a Early media

    • ■ 1.1.c Mid-call signaling (hold/resume, call transfer, conferencing)

    • ■ 1.1.e UPDATE

  • ■ 1.2 Troubleshoot these H.323 protocol elements

    • ■ 1.2.b Call set up and tear down

  • ■ 1.3 Troubleshoot media establishment

The field of communications has come a very long way since the introduction of the telephone in the 1800s by Alexander Graham Bell. Voice over IP (VoIP) traces its roots back to as early as the 1920s, when the first advancement in reproducing speech electronically and transmitting it over long distances was made. Decades later, in 1974, a significant milestone was achieved when the first voice datagram was transmitted over ARPANET, the precursor to the Internet. The year 1974 also saw another significant milestone in the history of the Internet: the introduction of Transmission Control Protocol (TCP), which would revolutionize the way information was transmitted over the Internet. Experiments carried out in subsequent years adequately demonstrated the need to develop a more flexible protocol for the transmission of real-time traffic classes. This led to the introduction of User Datagram Protocol (UDP), which has gone on to become the default transport layer protocol for real-time applications.

The next big leap in the world of real-time communications occurred in 1995, when an Israeli company by the name of VocalTec pioneered the first widely available Internet phone. At that time, it was possible to make calls between two such phones over the Internet, but speech quality, reliability of connection establishment, and the overall user experience were huge hindrances in preventing VoIP technology from becoming the next big wave in telecommunications.

However, transmitting real-time traffic, like voice and video, over the Internet at a fraction of the cost incurred in circuit-switched networks was such an exciting prospect that equipment manufacturers could not abandon it. The introduction of broadband Internet, with its always-on capability, greatly improved connection reliability, voice quality, and the user experience. This seemed to be the inflection point at which VoIP went mainstream, as corporations realized the immense cost benefits associated with this technology. Consequently, equipment manufacturers invested significant amounts of money and time in developing product lines with an abundance of features and customization options.

Over the next couple years, standards organizations such as the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) and the Internet Engineering Task Force (IETF) took up the task of developing and publishing standards related to VoIP. These standards have become the backbone protocols and enablers on which modern, real-time communications infrastructures operate today.

The two most significant signaling protocols used for real-time communications are Session Initiation Protocol (SIP) and H.323. The ITU-T first standardized the H.323 suite of protocols, which is used to define how multimedia sessions are established. Just a few years later, the IETF began standardizing a competing signaling protocol for VoIP: SIP. It’s worth noting that one of the most powerful design elements of SIP was its maximum reuse of existing Internet standards, such as Domain Name System (DNS), Hypertext Transfer Protocol (HTTP), Session Description Protocol (SDP), and Transport Layer Security (TLS). This made it easier for SIP to be deployed and integrated into existing environments while also allowing vendors to reuse these other protocols in their SIP applications.

While H.323 admittedly has a few advantages over SIP, it was the easily consumable HTTP-like text-based approach of SIP and its flexibility, extensibility, and ease of implementation that won out. Thus, SIP has become the de facto signaling protocol for real-time multimedia communications today. For this reason, we have a very brief introduction to H.323 in this chapter but spend more time on the SIP signaling protocol and its use of SDP to describe and negotiate multimedia sessions.

“Do I Know This Already?” Quiz

The “Do I Know This Already?” quiz allows you to assess whether you should read the entire chapter. If you miss no more than one of these self-assessment questions, you might want to move ahead to the “Exam Preparation Tasks” section of the chapter. Table 2-1 lists the major headings in this chapter and the “Do I Know This Already?” quiz questions related to the material in each of those sections to help you assess your knowledge of these specific areas. The answers to the “Do I Know This Already?” quiz appear in Appendix A, “Answers to the ‘Do I Know This Already?’ Quiz Questions.”

Table 2-1 “Do I Know This Already?” Foundation Topics Section-to-Question Mapping

Foundation Topics Section


Overview of SIP


Introduction to SDP


Overview of H.323


  • 1. Which of the following devices involves colocated UAC and UAS functionality for the forwarding and processing of SIP requests?

    1. a. SIP proxy

    2. b. Registrar server

    3. c. Redirect server

    4. d. B2BUA

    5. e. Location server

  • 2. Which of the following messages are server error final responses? (Choose two.)

    1. a. 404 Not Found

    2. b. 503 Service Unavailable

    3. c. 488 Unacceptable Media

    4. d. 301 Moved Temporarily

    5. e. 500 Internal Server Error

  • 3. Which headers are required for a SIP INVITE request? (Choose two.)

    1. a. Call-ID

    2. b. Expires

    3. c. Remote-Party-ID

    4. d. Session-ID

    5. e. Contact

  • 4. In an early offer call, which two SIP messages carry the SDP message body? (Choose two.)

    1. a. INVITE

    2. b. 100 Trying

    3. c. 200 OK

    4. d. ACK

    5. e. BYE

  • 5. Which media codecs require the a=rtpmap SDP attribute due to the use of dynamic payload numbers? (Choose two.)

    1. a. G711ulaw

    2. b. G711alaw

    3. c. OPUS

    4. d. H.264

    5. e. G.729

  • 6. Which SIP request can be used to update an existing media session between two user agents? (Choose two.)

    1. a. INVITE

    2. b. PRACK

    3. c. UPDATE

    4. d. REGISTER

    5. e. OPTIONS

  • 7. Which H.323 protocol performs the media negotiation during session establishment?

    1. a. H.225

    2. b. H.245

    3. c. H.450

    4. d. H.264

  • 8. What H.225 TCP port is used to establish non-secure signaling?

    1. a. 1719

    2. b. 1720

    3. c. 1721

    4. d. 5060

    5. e. 5061

  • 9. With which of the following is H.245 negotiated before the call connects?

    1. a. Slow start

    2. b. Fast start

    3. c. Early offer

    4. d. Delayed offer

Foundation Topics

2. Overview of SIP | Next Section

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