In our previous tutorials, we have considered the two major architectures that have been developed to support Voice over IP (VoIP) networks: H.323, developed by the International Telecommunications Union Telecommunications Standard Sector (ITU-T), and the Session Initiation Protocol (SIP), developed by the Internet Engineering Task Force (IETF). In our first tutorial on H.323, we considered the history and architecture of that standard, and looked at the four key components of an H.323 system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs). This tutorial extends the discussion of H.323 by focusing on the architecture of the H.323 terminal, and the protocols that are required to support the terminal functions.
The actual title of H.323, Packet-Based Multimedia Communications Systems yields a few clues regarding the protocols that will be required to support a H.323-compatible terminal. The Packet-Based part indicates that we are considering a packet switched, not circuit switched network environment, such as the Internet and its foundation protocol, the Internet Protocol (IP). The Multimedia part indicates that support for voice, data and/or video systems may be required, depending upon the end user objectives and the channel deployed in support of that end user communication. The Systems part indicates that connectivity between H.323 environments will be required, and with that comes issues of multivendor interoperability, that raises the immediate attention of many network managers.
To support these functions requires an intelligent terminal that takes audio, video and/or data inputs, begins a dialogue with another compatible system, and reliably transports that information across a packet network to the other system. TheH.323 terminal therefore requires several key components:
- Audio coder/decoder (codec): takes the analog audio signal from a microphone and converts it into a digital format that can be transmitted across the packet network. At the receiver, the opposite function (decoding) is performed, thus reconstructing the analog signal for human consumption. H.323 references other ITU-T standards for audio codecs that have been previously developed. These include: G.711, G.722, G.723, G.728 and G.729, all of which have specific encoding algorithms, data rates, and related technical specifications. At least one audio codec (G.711, operating at 64 Kbps) is required within the H.323 system, and other codecs may be optionally included.
- Video codec: takes the video information from a camera, and converts it into a transmittable form. At the receiver, an inverse function is performed (again called decoding), so that it can be displayed for human consumption. Two ITU-T video codec standards are referenced in H.323: H.261 and H.263. Video is an optional media type for H.323, therefore the video codec may or may not be included in the system. (This brings about the first interoperability challenge. If my H.323 terminal supports both audio and video, but yours only supports audio, we can therefore talk, but not share video information. Both of our terminals are compliant with H.323, however, but not completely compatible, since the video support is an optional feature.)
- User Data Channel: provides a data channel within the H.323 system to support applications such as still images, file transfers, audiographics conferences, and database access. To support the audiographics conferencing function, the ITU-T has developed the T.120 standard, which complements H.323.
- Registration, Admission, and Status (RAS): Gatekeepers are optional devices that provide for network management functions, such as bandwidth management. When a Gatekeeper is active on the network, endpoints register with that Gatekeeper using a process called RAS, which stands for Registration, Admission, and Status. When no Gatekeeper exists on the network, RAS is not used. The RAS signaling function is defined in the complementary H.225 standard.
- Call Signaling: the process of establishing or taking down a connection between two communicating entities is called signaling, which finds its historical roots in the on-hook, off-hook, dial tone, busy tone, and other signals that have been incorporated into the telephone network for decades. H.323 endpoints must also establish a connection between themselves prior to further communication, and this call signaling process for H.323 endpoints is defined in the H.225 standard.
- End-to-End Control Signaling: the H.323 systems must have a means to govern the operation of the two communicating endpoints. Another complementary standard, designated H.245, provides functions such as opening and closing logical channels, exchanging station capabilities, requesting a particular mode of operation, controlling the end-to-end flow of information, determining round trip delay, and so on. Note that the End-to-End Control Signaling is separate from the H.225 Call Signaling, and that the Call Signaling channel operates first (to establish the connection) prior to the Control Signaling functions (which control the communication once the channel has been established).
- Media Stream Packetization: the voice, data, and/or video information (which, in the case of voice and video, will have passed through a codec) must be placed inside a packet for delivery to the remote H.323 system. The H.225 standard specifies that the IETF Real-time Transport Protocol/Real-time Transport Control Protocol (RTP/RTCP) be used for these functions.
- Network Interface: the packet-based network interface is implementation-specific, and therefore outside the scope of H.323. From a product standpoint, however, many H.323-compliant devices support the Ethernet/IEEE 802.3 standard at the network interface, making for simpler integration of the H.323 device with most local area networks.
As demonstrated above, the H.323 standard is really an “umbrella” under which other standards, such as H.225, H.245 and RTP/RTCP fit. In our next tutorial, we will examine how all of the standards work together to provide end-to-end signaling and endpoint communication.
Copyright Acknowledgement: © 2005 DigiNet ® Corporation, All Rights Reserved
Mark A. Miller, P.E. is President of DigiNet ® Corporation, a Denver-based consulting engineering firm. He is the author of many books on networking technologies, including Voice over IP Technologies, and Internet Technologies Handbook, both published by John Wiley & Sons.