Monday, 19 March 2012


SIP (Session Initiation Protocol)  
Session Initiation Protocol (SIP) is an application layer protocol used to establish, modify and terminate multimedia sessions such as VoIP Calls. SIP also can invite new sessions to existing sessions such as multicast conferences. Basically it’s referred as signaling protocol in VoIP environment that can handle call establishment, call control and call termination and generating CDR (Call Detail Record) for billing purposes.

The Session Initiation Protocol (SIP) is an IETF (Internet Engineering Task force )-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams.

Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol.         

The following features of SIP play a major role in the enablement of IP telephony and VoIP:
  • Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
  • Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported � recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
  • Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
  • Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
  • Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.
Voice over IP (VoIP)
VoIP is a technology sending voice over packet networks. Earlier people were using PSTN networks and Mobile Networks to communicate each other. The rapid growth of internet and network technology introduced voice over data networks with carrier grade quality. In simple terms, VoIP means making or receiving phone calls over Internet or Internal Network.

Voice-over-IP (VoIP) implementations enable users to carry voice traffic (for example, telephone calls and faxes) over an IP network.

There are 3 main causes for the evolution of the Voice over IP market:

    • Low cost phone calls
    • Add-on services and unified messaging
    • Merging of data/voice infrastructures
The following protocols are used in VOIP:  

MegacoH.248-           Gateway Control Protocol

MGCP-                       Media Gateway Control Protocol

MIME-                         Multipurpose Internet Mail Extensions

RVP over IP-              Remote Voice Protocol over IP Specification

SAPv2-                      Session Announcement Protocol

SDP -                         Session Description Protocol

SGCP-                       Simple Gateway Control Protocol

SIP-                            Session Initiation Protocol Skinny

Skinny-                       Client Control Protocol (SCCP)

A VoIP system consists of a number of different components: Gateway/Media Gateway, Gatekeeper, Call agent, Media Gateway Controller, Signaling Gateway and a Call manager

The Gateway converts media provided in one type of network to the format required for another type of network. For example, a Gateway could terminate bearer channels from a switched circuit network (i.e., DS0s) and media streams from a packet network (e.g., RTP streams in an IP network). This gateway may be capable of processing audio, video and T.120 alone or in any combination, and is capable of full duplex media translations. The Gateway may also play

audio/video messages and performs other IVR functions, or may perform media conferencing.

In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a network: H.323 (ITU), MGCP (level 3, Bellcore, Cisco, Nortel), MEGACO/H.GCP (IETF), SIP (IETF), T.38 (ITU), SIGTRAN (IETF), Skinny (Cisco) etc.

Coders are used for efficient bandwidth utilization. Different coding techniques for telephony and voice packet are standardized by the ITU-T in its G-series recommendations: G.723.1, G.729, G.729A etc.

The coder-decoder compression schemes (CODECs) are enabled for both ends of the connection and the conversation proceeds using Real-Time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the protocol stack.

Quality of Service
A number of advanced methods are used to overcome the hostile environment of the IP net and to provide an acceptable Quality of Service. Examples of these methods are: delay, jitter, echo, congestion, packet loss, and missordered packets arrival. As VoIP is a delay-sensitive application, a well-engineered, end-to-end network is necessary to use VoIP successfully. The Mean Opinion Score is one of the most important parameters that determine the QoS.

The following are examples of services provided by a Voice over IP network according to market requirements:

Phone to phone, PC to phone, phone to PC, fax to e-mail, e-mail to fax, fax to fax, voice to e-mail, IP Phone, transparent CCS (TCCS), toll free number (1-800), class services, call center applications, VPN, Unified Messaging, Wireless Connectivity, IN Applications using SS7, IP PABX and soft switch implementations.

Difference between VoIP and SIP

VoIP and SIP are related terms in context of Voice over IP. VoIP is Voice over Internet Protocol and SIP is Session Initiation Protocol. SIP is one of the signaling protocols used in voice over IP. H323 is another Signaling protocol does similar function as SIP. Basically comparing VoIP and SIP is like comparing Apple and Orange but since most people use VoIP and SIP to a same context of Voice over IP technology, devices and applications we have differentiated VoIP and SIP below.

 (1)VoIP is a technology used in modern telecommunication networks whereas SIP is a signaling       protocol (control protocol) used in VoIP

(2)General Term VoIP includes Signaling and Media whereas SIP only refers Signaling plane.

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