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SYNOPSIS INTEROPERABILITY IN UNIFIED SERVICES USING SIP

1. Introduction
Purpose
Todays communication environment is characterized by the usage of several services. Each of them needs to be accessed in a specific way. For example, a user has to employ two different devices or services if he wants to make a phone-call and to send a fax. The more services a user wants to employ, the more devices or applications he has to operate on. To overcome this restriction, a new category of telecommunication service solutions has gained momentum UNIFIED COMMUNICATION , these solutions have the aim to fulfil the vision of integrating different communication services. Unified communication is very useful for knowledge workers, information workers, and service workers. With an increasingly mobile workforce, businesses are rarely centralized in one location, in such a scenario unified communication provides potential to speed up the business processes .

Unified communications
Unified communication is the integration of real time communications like chat, video conferencing, and speech recognition with non-real time communications like email , voicemail, sms and ,fax allowing each and every individual to manage their several different means of communications using a single service instead of many different services Unified communication aims to minimize the communication response time or delay. The term presence is an important concept in unified communications which represents the availability and willingness of a person to communicate

In May 2010.the Unified Communications Interoperability Forum was formed which was an alliance between companies that create and tests interoperability between UC products and existing communications.

Unified communication and SIP


Session Initiation Protocol (SIP) is the industry standard protocol used by most unified communications solution vendors. Each mode of communication IM, presence, conferencing, video, and voice, all uses SIP for its transport. Communication systems based on the SIP standard have come a long way over the past several years. SIP provides a simplified and efficient approach for implementing unified communication and is thus widely used protocol for offering this integrated service to users

About SIP
SIP (Session Initiation Protocol) is a signaling protocol for multimedia session control which was published by the Internet Engineering Task Force (IETF) in 1999. SIP is actually an application layer protocol for establishing, manipulating, and tearing down sessions. SIP serves four major purposes: Establishment of user location

Provides feature negotiation so that all of the participants in a session can agree on a feature to be supported among them. Mechanism for call management functions. Allows for changing features of a session when it is in progress. SIP follows the client/server model. SIP is not a vertically integrated communications system. SIP is rather a component that can be used with other IETF protocols to build a complete multimedia architecture. Typically, these architectures will include protocols such as the Realtime Transport Protocol (RTP) (RFC 1889 [28]), Real-Time streaming protocol (RTSP) (RFC2326 [29]), the Media Gateway Control Protocol (MEGACO) (RFC 3015 [30]) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) (RFC 2327 [1]). 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. SIP does not provide services. Rather, SIP provides primitives that can be used to implement different services SIP Protocol Structure

The application layer provides application programs with an interface to communicate and transfer data across the network. Following are the SIP protocol layers:

Transaction user creates a client transaction and sends the request, the destination IP address, port number, and transport service to which the request must be sent. The transaction layer handles application layer retransmissions and timeouts, and matches responses to requests.. All SIP components contain a transport layer to send requests and responses over network transports. For connection-oriented transports, the transport layer determines the type of connection to use for a request or response. The network transport can be Transmission Control Protocol (TCP), User Datagram Protocol (UDP), Transport Layer Security (TLS), or Stream Control Transmission Protocol (SCTP). The transport layer also determines how a client can send requests and responses, and how a server can receive requests and responses over the network.

The network transport layer is the transport layer in the IP stack. It enables transfer of data between end points by using the services of the network layer. This layer has two primary protocols, TCP and UDP. The TCP supports reliable and sequential packet delivery through error recovery and flow control mechanisms. The UDP is a simple message-based connectionless protocol compared to TCP. The SCTP is yet another transport layer protocol that application developers can use to transmit data between end points. The network layer routes data packets from the sender to the receiver in the network. The most common network layer protocol is IP

2.

Interoperability

of

SIP

in

heterogeneous

environments

Due to the limitations of the current version of the Internet protocol (IPv4) extensive research and standardization work has been done in developing a new protocol, namely IPv6. The most obvious reason for using IPv6 with SIP is naturally the huge amount of available addresses. This is especially important when considering 3G architectures with millions of SIP based mobile phones all requiring their own IP addresses. But phones are not the only IP capable devices from the SIP point of view. Internet-capable gaming stations or even appliances are also thought to be triggered by SIP As an integration mechanism for enabling the communication between an IPv4 only capable device and an IPv6 only capable device we chose the protocol translator approach. This allows for simple end devices and networks that only need to support one of the IP versions. This protocol translator called the SIP Protocol Gateway (PGW) is intended to be located on the borderline between pure IPv6 and pure IPv4 clients. It runs on a dual-stack machine to be able to speak and listen to both protocol-families. It can be considered as a proxy, which modifies the SIP messages sent by an IPv4/IPv6 host to be understood by an IPv6/IPv4 host. The SIP-PGW consists of three components:1. MSP- The Mini SIP Proxy (MSP) receives SIP messages, modifies them, installs UDP mappings for RTP communication and forwards the SIP messages to another proxy. The

MSP depends on two outbound proxies: one for IPv4 and one for IPv6 targets, as it does not route requests itself. If a SIP request message is received by an IPv4 interface, it is sent out to the IPv6 proxy and vice versa. SIP response messages are routed by their second via header. 2. UDP-forwarding daemon (UFWDD) - This entity manages the IPv4 and IPv6 address spaces of the gateway and acts as a network address translator. Packets received on the IPv4/IPv6 side are sent to an IPv6/IPv4 host using a sending address and port number allocated from the IPv6/IPv4 address and port space of the gateway. 3. Control protocol - For requesting the allocation of addresses and mapping between the protocol families,both the MSP and FWDD communicate via UDP messages. This also allows both components to reside on different machines.

3. Implementation Detail
To converge all communication services at one place we have to build up a setup which consists of several servers and services .The each and every components shown in the picture are described below and this will also provides the insight of the implementation of our project. The various components shown in figure are described as follows : SIP proxy The SIP proxy is the central component of our solution. It is responsible for registering the users and keeping the location database (maps IP to SIP addresses). The entire SIP routing and signaling is handled by the SIP proxy, and it is also responsible for end-user services such as call forwarding, white/blacklist, speed dialing, and others. User administration and provisioning portal One important component is the user administration and provisioning portal. In the portal, the user may subscribe to the service and should also be capable of buying credits, changing passwords, and verifying his or her account. On the other hand, administrators should be able to remove users, change user credits, and grant and remove privileges. Provisioning is the process used to make it easier for administrators to provide automatic installation of user agents such as IP phones, analog telephony adapters, and soft-phones.

PSTN gateway To communicate with the public switched telephone network, a PSTN(Public Switched Telephone Network) gateway is required. Usually, this gateway will interface the PSTN using E1 or T1 trunks. Basically it converts TDM signal to SIP signal. Media server The SIP proxy never handles the media. Services such as IVRs, voicemail, conference, or anything related to media should be implemented in a media server. Media Proxy or RTP Proxy for NAT traversal Any SIP provider will have to handle NAT traversal for their customers. The Media Proxy is an RTP bridge that helps the users behind symmetric firewalls to access the SIP provider. Without them, it won't be possible to service a large share of the user base. Accounting and CDR generation A server is used for Authentication and Authorization. In CDR generation the duration of the calls is calculated and it also deals with issue related to call forwarding. Interoperability Engine It works for handling interoperability issue between IPv4 and IPv6. Mail Exchange Server Mail Exchange Server comprises of the 4 parts IMAP SERVER fetches the mail. SMTP SERVER sends the mail. IPFAX server which sends and receive FAX . Filter Engine decides what service is used when. Local Database It stores mail , faxes ,voice mail ,user data ,enterprise Directory and lots of other information used by different services for implementing unified communications. User Agent The SIP terminal (IP phones, ATAs, softphones, and so on) devices which are actually communicating using the SIP.

Ldap Server It is the server dedicated to access Enterprise Directory and to get other information related to user when signaling a call.

4. References
1.http://www.ipv6.com/articles/voip/Session-Initiation-Protocol.htm 2.RFC3261(http://www.ietf.org/rfc/rfc3261.txt) 3.www.ieeexplore.ieee.org 4.Building Telephony System By Flavio E. Golcanves 5.http://www.cs.columbia.edu/~coms6181/slides/11/sip_long.pdf 6. http://en.wikipedia.org/wiki/Unified_communications 7. Unified communications system by Diego Besprosvan