Fundamental of VoIP and Security

    CHAPTER NO.4a

IV.PROTOCOLS: Two major protocols are, H.323 and SIP, can be classified according their role during message transmission. They are involved in call setup, teardown, and modification. RTP (real-time transport protocol) and RTCP (real-time transport control protocol) are media transport protocols and are involved in end-to-end transport of voice and multimedia data. This chapter covers basic overview of two important VoiP protocols are SIP (Session Initiation Protocol) and H.323 Protocol and also discusses regarding SIP, protocol characteristics of SIP and H.323, signaling of SIP and H.323 protocol, Sip method etc, how SIP works with other protocols. Many other important issues that are related with these Protocols are discussed in this chapter.

A. SIP OVERVIEW: SIP (Session Initiation Protocol) is a protocol that is developed support advanced telephony services across the Internet establishing sessions in an IP network. A session could be a straightforward two-way telephone call or a joint multi-media conference session. Internet telephony is growing by its use as a cheap but low quality way to make international phone calls. SIP is part of the IETF standards process and is modeled upon other Internet protocols such as SMTP (Simple Mail Transfer Protocol) and HTTP (Hypertext Transfer Protocol.) SIP is used to establish, change and end calls between the users in an IP-based network. In order to provide telephony services there is a need for a number of different standards and protocols to come together. SIP is described as a control protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. SIP supports session descriptions that allow participants to agree on a set of compatible media types. It also supports user mobility by proxying and redirecting requests to the user's current location. SIP is not tied to any particular conference control protocol.

 1. CHARECTERSTICS OF SIP:  SIP is described as a control protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephony calls and multimedia distribution. Members in a session can communicate through multicast or through a mesh of uni cast relations or through combination of these. Session descriptions are also supported by SIP that permits participants to agree on a set of compatible media types and supports user mobility by proxy and redirect requests to the user's current location too. SIP is not attached to any particular conference control protocol. SIP has to provide the following functions.

a. NAME TRANSLATION AND USER LOCATION: It ensures that the call reaches to the destination, carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported. 

b. FEATURE NEGOTIATION :This allows the group involved in a call, may be multi-party Call, to agree on the features supported and 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.

 c. 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.

 d. CALL FEATURE CHANGES : It allows the user be able to change the call characteristics during the call. For example, a call may have been set up as �voice-only�, but during the call users may want or 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.

 2. SIP SINGLING: The most basic configuration is used to establish a call directly between two end stations without the use of any intermediate proxy servers. In this case the calling party would send an INVITE message to the called party to initiate the session, who would respond with Ringing and OK messages. The calling party would then return an ACK (acknowledgement) message, which would complete the connection, and allow information to be exchanged between the two parties. When the connection is no longer required, one party sends a BYE message, with an OK message returned in response, thus terminating the call. A complex example, which is described in the SIP standard, uses proxy servers in the communication path. A proxy is someone or something that acts on behalf of another person or entity. For example, if you own shares in a public company, and are not able to attend the annual meeting, you send in a proxy ballot that enables another person or group to vote on your behalf. In a similar manner, a SIP proxy server acts as an intermediary for the purpose of making requests on behalf of other SIP clients, and in many cases, functions in a routing mode, forwarding SIP requests to another device that is closer to the ultimate destination (i.e. the called party). As a result, the SIP proxy server has a dual role. It acts as a server when it processes end user request messages, but it can also act as a client when it forwards a message to another device downstream. In large networks may include more than one proxy server in the path from a calling to a called party.

3. SIP METHODS:The Instructions that are used to start call till finish. These are called methods. Let�s discuss them one by one with the description. There are six types of command messages defined.

�  REGISTER is used by a client to register an address with a SIP server

�  INVITE message to the called party to initiate the session

�   ACK (acknowledgement) Complete the connection for both sides to communicate

�  CANCEL Is used to cancel a pending request

�  BYE Terminating a call

�  OPTIONS Is used to query a server about its capabilities

B. SIP-WORKING WITH OTHER PROTOCOL: SIP was designed to solve only a few problems and to work with a broad spectrum of existing and future IP telephony protocols. To this end SIP provides four basic functions. SIP allows for the establishment of user location by translating from a user's name to their current network address. SIP provides for feature negotiation so that all of the participants in a session can agree on the features to be supported among them. SIP is a mechanism for call management - for example adding, dropping, or transferring participants. And finally SIP allows for changing features of a session while it is in progress. All of the other key functions are done with other protocols.

C. SIP-PLAYING & SIGNIFICANT ROLE WITH THE PROTOCOL: SIP is an IETF application layer protocol for establishing, manipulating, and tearing down sessions. SIP's main purpose is to help session originators deliver invitations to potential session participants wherever they may be in a nut shell that is SIP�s role. So SIP is not the universal remedy because it was never built to be that way. Let's review two of the fundamental assumptions behind SIP's design: Reusing Existing Protocols: SIP was designed to specifically reuse as many existing protocols and protocol design concepts. For example, SIP was modeled after HTTP, using URLs for addressing and SDP to convey session information. Maximizing Interoperability: SIP was also designed so that it would be easy to bind SIP functions to existing protocols and applications, such as e-mail and Web browsers. SIP does this by limiting itself to a modular philosophy - just like many other Internet protocols - and focusing on a specific set of functions.

D. INSIDE SIP MESSAGE: The Request line and header field define the nature of the call in terms of services, addresses, and protocol features. The message body is independent of the SIP protocol and can contain anything.

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