NGN
Fundamental NGN Protocols

SIP protocol

Session Initiation Protocol (SIP) is an application-layer control protocol that handles the setup, modification, and tear-down of multimedia sessions. Media can be added to (and removed from) an existing session. SIP is used in combination with other protocols to describe the session characteristics to potential session participants. SIP is based on a request and response transaction model similar to HTTP. Each transaction consists of a request that invokes a particular method or a function on the server and at least one response.

SIP supports five facets of establishing and terminating multimedia communications:

SIP is a text-based protocol suggested and standardized in RFC 3261. SIP has been proposed as a part of a unit based of following protocols.

SIP protocol stack

Consequential protocols

A lot of SIP functions depend from other protocols. SIP defines establishment, termination and call modification and SIP use other protocols as Real-time Transport Protocol (RTP) for transporting real-time data and providing QoS feedback, the Real-Time Streaming Protocol (RTSP) for controlling delivery of streaming media, the Media Gateway Control Protocol (MEGACO) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) for describing multimedia sessions. Therefore, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.

SDP

SDP (Session Description Protocol) [4] is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation.

When initiating multimedia teleconferences, voice-over-IP calls, streaming of video, or other sessions, there is a requirement to convey media details, transport addresses, and other session description meta data to the participants. SDP provides a standard representation for such information, irrespective of how that information is transported. SDP is intended to be general purpose so that it can be used in a wide range of network environments and applications. However, it is not intended to support negotiation of session content or media encodings: this is viewed as outside the scope of session description.

An SDP session description includes the following:

As resources necessary to participate in a session may be limited, some additional information may also be desirable:

RTP

The goal of this part is to present RTP (Real Time Transport Protocol) [14] in the way that it will be easily understandable also by a beginner. It contains deeper description of the main RTP protocol, but also protocols that stand next to RTP, cooperate with it and fill its gaps. The structure of RTP will be shown, to introduce the protocol body to the reader with a close view of its parts and theirs functions.

RTP provides end-to-end delivery services suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. Real-time means that not only correct results are required, but also a sufficient time in which the result is delivered. That is why the delivery of the audio or video data is typically delay sensitive. According to this RTP use timestamp and control mechanisms for synchronizing different streams with timing properties.

RTCP

RTCP (Real-time Transport Control Protocol) [4] is an application layer protocol designed to control of data delivery in real-time and to measure the QoS. It is defined in RFC 3550 published in July 2003. RTP protocol uses the RTCP protocol, which transports the following additional information for the management of the session. RTCP is based on the periodic transmission of control packets to all participants in the session. The underlying protocol must provide multiplexing of the data and control packets, like UDP protocol that allows the multiplexing of RTP data packets and RTCP control packets. RTCP protocol requires the sending of information periodically by the participants of the session. RTP packets only transport user’s data, whereas RTCP packets only transport in real time the supervision.

Protocol RTCP performs these principal functions:

Control the media flow and adapt it to all the participants of the RTP session. By having each participant send its control packets to all the others, each can independently observe the number of participants. This information is used to calculate the rate at which the packets are sent.

DIAMETER

DIAMETER [4] is a member of “AAA” protocols collection, derived from its predecessor RADIUS protocol. It is a peer to peer protocol , used for handling service requests such as user validation, network resource control, connection and session management, wireless or roaming charging, billing applications etc.

Diameter sessions consist of exchange of commands and AVPs between servers and clients and unlike Radius, uses peer to peer architecture rather than more classic client/server scheme. Each node may initiate a message (request) at any time, as example, server may abort a service to specific user. Diameter is defined in terms of base protocol and a set of applications. This design allows protocol to be extended for new access technologies. The base protocol provides basic mechanism for reliable transport, delivery and error handling.

MEGACO/H.248

This protocol has been established to cover the need of IP networks and services to interoperate with traditional networks (e.g. PSTN) and provide the same services over both types of networks (IP, Traditional). This enables separation of call control from media conversion. Megaco/H.248 is defined as master / slave architecture based protocol which is used for communication between MGC (Media Gateway Controller, sometimes called a call agent or softswitch, which dictates the service logic of that traffic) and one or more decomposed MGs (Media Gateways), which converts circuit-switched voice to packet-based traffic.

Megaco/H.248 instructs an MG to connect streams coming from outside a packet or cell data network onto a packet or cell stream such as RTP. Megaco/H.248 is similar to MGCP from an architectural standpoint and the controller-to-gateway relationship, but Megaco/H.248 supports a broader range of networks, such as ATM.

SIGTRAN

In view of functionality and performance the user make high demands on modern telecommunication networks. Using IP (Internet Protocol) signaling messages will be transmitted over TCP (Transmission Control Protocol) or UDP (User Datagram Protocol). These transport protocols are not designed to meet the requirements given by a signaling system used in a circuit switched network like PSTN/ISDN (Public Switched Telephone Network/Integrated Services Digital Network). So the working group SIGTRAN was founded by the IETF (Internet Engineering Task Force) to develop a new protocol, based on IP, in consideration of given requirements by the existing switched telephone network.

This protocol, named SCTP (Stream Control Transmission Protocol) has some advantages in comparison to TCP. The SCTP offers a fundament to initiate and run secured transport connections using IP networks to transmit signaling information. Based on SCTP, several adaptation layers enable the transmission of upper layer protocols, i.e. ISUP (ISDN User Part), SCCP (Signaling Connection Control Part) and DSS1 (Digital Subscriber System No. 1).