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The following page gives a rough overview about several protocols. Information about the OSI model and its protocol stack as the basis for most protocols can be found here:

H.245 Terminal Control Protocol
H.323 Terminal Control Protocol

HDLC High-level Data Link Control
HTTP Hypertext Transfer Protocol
IP Internet Protocol
  TCP Transmission Control Protocol
  UDP User Datagram Protocol
  RTP Real-time Transport Protocol
  RTCP RTP Control Protocol
  IPv6 Internet Protocol Version 6

Q.931 Signalling Protocol

SIP Session Initiation Protocol
SMTP Simple Mail Transfer Protocol



H.245 Terminal Control Protocol
H.245 requires the numbered simple retransmission protocol (NSRP) and control channel segmentation and reassembly layer (CCSRL) sublayer support to ensure reliable operation. H.245 requires mobile terminals to support NSRP and SRP modes. In addition H.245 control protocol provides following functionalities and services:

  • Master-slave determination is provided to determine which terminal is the master at the beginning of the session. Due to the fact that H.245 is symmetric control protocol, it is necessary to decide the master terminal, which has the right to decide the conditions in case of the conflict.
  • Capability exchange is provided to exchange the capabilities both terminal supports, such as optional modes of multiplexing, type of audio/video codecs, data sharing mode and its related parameters, and/or other additional optional features.
  • Logical channel signaling is provided to open/close the logical channels for media transmission. This procedure also includes parameter exchange for the use of this logical channel. Multiplex table initialization/modification is provided to add/delete the multiplex table entries.
  • Mode request is provided to request the mode of operation from the receiver side to the transmitter side. In H.245, the choice of codecs and its parameters are decided at the transmitter side considering decoder's capability, so if the receiver side has a preference within its capability, this procedure is used. Round-trip delay measurement is provided to enable accurate quality characteristic measurement.
  • Loopback testing is provided for use during development or in the field to assure proper operation.
  • Miscellaneous call control commands and indications are provided to request the modes of communication, flow control such as conference commands, jitter indication and skew, or to indicate the conditions of the terminal, to the other side.

H.245 uses the abstract syntax notation 1 (ASN.1) to define each message parameters that provides readability and extensibility effectively. After the multiplexing level synchronization between communicating parties is completed the first logical channel opened (channel 0) is H.245 call control with the CCRL and NSRP to assure that the H.245 channel will be highly reliable and can use large packets during operation.

H.323 Terminal Control Protocol
H.323 is an ITU recommendation and defines the interworking of network elements and protocols to allow multimedia transmission of voice, video, chat, file sharing, whiteboarding etc. through an unreliable packet-based network. H.323 ties together a number of existing recommendations and is often considered to be just the call signalling element of a multimedia call, responsible for setting up and clearing down calls.
Incorporated are a number of mandatory and optional H323 entities. The most important elements are listed below:

  • H.323 Gatekeeper (optional)
    One of its purposes is to provide basic Admission Control onto a network by authorising (or refusing) communications between other H.323 entities. It also provides an address translation service.
  • H.323 Gateway
    A gateway provides a protocol conversion service between H.323 terminals and other terminals that do not support H.323.
  • H.323 Multipoint Control Unit
    An MCU provides services that allow three or more endpoints to take part in a conference call. An MCU comprises a Multipoint Controller for handling call control and optional Multipoint Processors for handling the media exchange (voice, video etc.) in a conference.
  • H.323 Terminal
    An H.323 Terminal is an endpoint on a network which provides the two-way communications with another H.323 terminal, Gateway or MCU. A terminal may provide speech only, speech and data, speech and video, or speech, data and video.

H.323 is not the only recommendation for multimedia transmission. A newer alternative protocol is being developed by the Internet Engineering Task Force called SIP xxxxxxx. It is yet to been seen which protocol (if any) will become dominant.

HDLC High-level Data Link Control
HDLC is a popular ITU defined protocol used in data networking applications such as cellular base station switch controllers, frame relay switches, high bandwidth WAN links, xDSL and modem error correction. This protocol is responsible for transmitting data between network points. It organizes data into units, following the bit oriented packet transmission mode, and sends it across a network to a destination that verifies its successful arrival. The data stream and transmission rate is controlled from the network node (PCM highway clock) with a back pressure mechanism. This eliminates additional synchronization and buffering of the data at the network interface. Different variations of the protocol are used in different networks. For example, ISDN's D-channel uses a slightly modified version of HDLC.

HTTP Hypertext Transfer Protocol
HTTP has been in use by the World-Wide Web global information initiative since 1990. It's the network protocol used to deliver virtually all files and other data (called resources) on the World Wide Web, whether they're HTML files, image files, query results, or anything else. HTTP is an application-level protocol with the lightness and speed necessary for distributed, collaborative, hyper media information systems. It is a generic, stateless, object-oriented protocol which can be used for many tasks, such as name servers and distributed object management systems, through extension of its request methods. A feature of HTTP is the typing and negotiation of data representation, allowing systems to be built independently of the data being transferred. Usually.

IP Internet Protocol
The connectionless Internet Protocol is a low level protocol and is responsible for the delivery of packets (or datagrams) between host computers. It does not establish a virtual connection through a network prior to commencing transmission. This is the task for higher level protocols. In its most basic form, the IP header comprises 20 octets.
IP makes no guarantees concerning reliability, flow control, error detection or error correction. The result is that datagrams could arrive at the destination computer out of sequence, with errors or not even arrive at all.
Thus IP is not well suited for example to voice transmission. Real time applications such as voice and video require guaranteed connection with consistent delay characteristics. Higher layer protocols need to be installed to address these issues up to a certain extent.
Nevertheless, IP succeeds in making the network transparent to the upper layers involved in voice transmission through an IP based network. Generally, there are two protocols - TCP and UDP - available at the transport layer when transmitting information through an IP network. TCP and UDP protocols enable the transmission of information between the applications on host computers. These processes are associated with unique port numbers (HTTP application is usually associated with port 80).

TCP  Transmission Control Protocol
TCP is a connection oriented protocol and establishes a communications path prior to transmitting data. It handles sequencing and error detection, ensuring that a reliable stream of data is received by the destination application.

UDP  User Datagram Protocol
In common with IP, UDP is a connectionless protocol and routes data to it's correct destination port, but does not attempt to perform any sequencing, or to ensure data reliability.

RTP  Real-time Transport Protocol
RTP is the protocol used to provide timing and synchronisation for digitised voice and video being transmitted through a packet network. When transmitted through an IP network, RTP relies on the lower layer UDP protocol to transport it through a network between computer applications. Real time applications require mechanisms to be in place to ensure that a stream of data can be reconstructed accurately. Datagrams must be reconstructed in the correct order, and a means of detecting network delays must be in place.
In order to reduce the effects of jitter, data must be buffered at the receiving end of the link so that it can be played out at a constant rate. To support this requirement, two protocols have been developed.  These are RTP (Real-time Transport Protocol) and RTCP (RTP Control Protocol).

RTCP RTP Control Protocol
RTCP provides feedback on the quality of the transmission link. RTP transports the digitised samples of real time information. RTP and RTCP do not reduce the overall delay of the real time information and they do not make any guarantees concerning quality of service.

Wireless technologies that promise to integrate voice and web data in an IP-based mobile communications system known as the Fourth Generation (4G) wireless network. The future broadband network and its evolution will be characterized by six transitions:

  • Transition from a dial-up, circuit-switched network to a data-oriented network
  • Transition from mere connectivity to service creation platforms
  • Transition from a copper-based network towards an all-optical network
  • Convergence of mobile and fixed networks
  • Operator profitability through converged mobile and broadband services
  • Transition to IPv6 networks

IPv6 Internet Protocol Version 6
IPv6 or "IP Next Generation” (IPng) is designed by the IETF to replace the current version Internet Protocol Version 4 ("IPv4") which is now nearly twenty years old and - remarkably resilient - is beginning to have problems. Most importantly, there is a growing shortage of IPv4 addresses, which are needed by all new machines added to the Internet. IPv6 fixes a number of problems in IPv4. It also adds many improvements to IPv4 in areas such as routing and network autoconfiguration. IPv6 is expected to gradually replace IPv4, with the two coexisting for a number of years during a transition period.

  • Expanded Routing and Addressing Capabilities
    IPv6 increases the IP address size from 32 bits to 128 bits, to support more levels of addressing hierarchy and a much greater number of addressable nodes and simpler auto-configuration of addresses.
  • A new type of address called a "anycast address"
    Identifies sets of nodes where a packet sent to an anycast address is delivered to one of the nodes. The use of anycast addresses in the IPv6 source route allows nodes to control the path which their traffic flows.
  • Header Format Simplification
    Some IPv4 header fields have been dropped or made optional, to reduce the common-case processing cost of packet handling and to keep the bandwidth cost of the IPv6 header as low as possible. The IPv6 header is only twice the size of the IPv4 header.
  • Improved Support for Options
    Changes in the way IP header options are encoded allows for more efficient forwarding, less stringent limits on the length of options, and greater flexibility for introducing new options in the future.

Authentication and Privacy Capabilities
IPv6 includes the definition of extensions which provide support for authentication, data integrity and confidentiality. This is included as a basic element of IPv6 and will be included in all implementations.

Q.931 Signalling Protocol
Q931 is a call signalling protocol used in ISDN networks for setting up and clearing down calls. Q.931 is also used for establishing H323 calls. H.225 call control messages are embedded within the user-to-user elements of Q.931 messages to provide additional information not available in Q.931 such as IP address information.

SIP  Session Initiation Protocol
All 3G wireless standards groups have agreed to use the IP based signaling protocol SIP, Session Initiation Protocol, for voice and multimedia services. SIP messages are text based and similar in format to existing Internet protocols such as HTTP. It has been argued that SIP is more suited to Internet applications and is easier to implement in software.

SMTP Simple Mail Transfer Protocol
SMTP is used for sending e-mails. The recipient does not have to be online when the message is sent. The e-mail application uploads the message to a server running SMTP, which stores the message until the recipient’s SMTP server is available, then forwards the message. The recipient’s SMTP server runs another service that places the message into the recipient’s mailbox, where it waits for the recipient to use yet a different protocol, such as Post Office Protocol version 3 (POP3) or Internet Message Access Protocol (IMAP), to download the message.

top  (last update: October 2005)

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