PROTOCOLS USED IN VoIP

Table of Contents: 1. Introduction 2. H.323 protocol Standard 3. Session Initiation protocol 4. Conclusion 5. References

Introduction:
The Voice over Internet Protocol (VoIP) is internet telephony in which the user can perform a phone call over broadband internet instead of the old analog telephone lines. The concept of VoIP is very flexible such that it allows the user to make and receive calls to and from traditional landline numbers, generally for free [1]. The voice signals are first digitized, then compressed and converted to Internet Protocol (IP) packets and hence transmitted over the IP network. The working of VoIP depends on certain parameters such as end-user equipment, network components, call processors, gateways and some protocols [1] [2]. The components used in VoIP makes use of the traditional circuit switched networks and thus saving a lot of bandwidth and expenditure. The protocols used in VoIP are mainly signaling protocols which are used to make or tear down calls, convey the information required to the end users and also negotiate capabilities [2]. As mentioned above, the methodology in VoIP connection usually consists of a pattern of signaling transactions between the gateways. Some of the basic protocol standards that are used in VoIP are as follows: A) H.323 B) Session Initiation Protocol (SIP) C) Inter-Asterix Exchange protocol (IAX) Supporting Protocols: D) Media Gateway Control Protocol (MGCP) E) Real-time transport Protocol and Real-Time Control Protocol (RTP-RTCP) F) Real-Time Streaming Protocol (RTSP) G) Resource Reservation Protocol (RSVP) H) Session Description Protocol (SDP) I) Session Announcement protocol (SAP) [3].

Proprietary Protocols: J) Skinny Client Control Protocol (SCCP) K) UNISTIM In this paper we will discuss about the two standard and prominent protocols SIP and H.323. H.323 Protocol Standard: This protocol is specified by the International Telecommunications Union (ITU) which makes the vendors follow the rules for an IP based real-time communications and videoconferencing including audio, video and data [3]. H.323 provides the mechanisms for video communications and data collaboration, in combination with the ITU-T T.120 series of standards. This standard refers to many other family standards such as H.245, H.225, and H.450 etc [4]. H.323 governs the requirements for visual telephony which are mostly technical such as the transmission of audio and video in packet based network. There are basically four components in the standard which are: A) Terminals: These are the local area network client endpoints which facilitate two way communications and should also facilitate the supporting protocols. B) Gateways: These are the endpoints on the network that facilitate the two way communication between the terminals of an IP network and other terminals in the network, just like a translator [4]. C) Gatekeepers: A very important component of the H.323 system which performs the duties of a manager. Provides services to all the registered endpoints. Some basic services are address translation; admissions control, call signaling, call authorization, bandwidth management and call management [3]. D) Multipoint Control Units (MCUs): As the name suggests, the MCU are endpoints that provide the option for three or more terminals or even gateways to participate in a multipoint conference.

As discussed above, H.323 is a combination of several different protocols which are used in different stages of a VoIP call [4]. The procedures used in communication between the endpoints and the gatekeepers are: Gatekeeper discovery: In this process the endpoint uses to recognize the gatekeeper with which the registration should be done. When the endpoint broadcasts a gatekeeper request message seeking the required gatekeeper, the gatekeepers in the network respond with a gatekeeper confirmation message [2] [3]. Any gatekeeper which does not wish to register will send a rejection message and if no message comes in the specified time interval, then this will lead to a timeout. Endpoint registration: This is the process in which the endpoint enters a zone and then informs the gatekeeper of its purpose with its details with a request message for registration. The gatekeeper will respond with a confirmation or a rejection [4]. Admissions, bandwidth change, and status and disengage: These messages are exchanged between a gatekeeper and an endpoint and are used to facilitate admissions control and bandwidth management functions. Call setup process in H.323: Call Establishment: The initiation of the call is done firstly by the registration process that is done when the RAS request message is sent by the caller. This request is then received by the gatekeeper who confirms the request by sending back an acknowledgement. The destination will be registered in the same way by the gatekeeper. Now the caller again sends a call signaling setup message this time to the destination through the gatekeeper [4]. Now both the caller and destination will send and receive the acknowledgements one after another by the use of the H.245 terminal capability set message and open logical channel message.

[4] Call Termination: The destination endpoint will initiate the call release by sending an H.245 end session command message to the caller which then confirms this by sending an end session command message to the destination [4]. Now a disconnection request is sent to the gatekeeper from both the caller and destination to the gatekeeper who in turn sends back an confirmation message about the status. In brief, the call process can be explained in the following steps: 1) discovering gatekeeper 2) Endpoint registration with gatekeeper 3) Endpoint sets up call and has capability to exchange places with the gatekeeper. 4) Call is established 5) When the endpoint is done, the call can be terminated by the endpoint or the gatekeeper can also initiate the termination [3] [4].

Session Initiation Protocol (SIP): A new generation of IP based service which is now used as an alternative or replacement to H.323 is the session Initiation Protocol. This is the pioneer in enabling multi-user sessions irrespective of the content. SIP based services include the local and long distance telephony, instant and voice messaging, video conferencing etc [3]. SIP is emerging from the shadows of the more established protocols such as hyper text transfer protocol (HTTP) and simple mail transfer protocol (SMTP). One of the advantages of SIP is that it works with the preexisting protocols taking care of the authentication, location and QoS etc [5]. SIP functions independently of the network transport protocol and the participants end devices can establish, modify or even terminate a connection irrespective of the content either it is voice, video, data. SIP is very flexible in the way that it allows the construction of messages, applications etc. using programming languages such as Java [5]. It generally functions over User datagram protocol (UDP) or transmission control protocol (TCP), but as explained above it can be run on different protocols such as the IP, ATM or XX.25 [3]. The basic requirements in the communications in SIP are user location services, session establishment, session participant management and limited feature establishment. The flexibility of SIP allows it to be used for many applications such as gaming, music and video on demand with voice, video conferencing also [3]. SIP makes use of two major components in a session which are the SIP user agent (UA) and the SIP network server. The SIP user agent (UA) has two branches, the user agent client (UAC) which initiates the call and the user agent server (UAS) that responds to the call [5]. The SIP network server controls the signaling related to the name resolution and user location. This is classified into three major categories such the SIP register server, SIP proxy server and SIP redirect server. The SIP register server takes care of the registration messages which it receives from the endpoints. These messages contain the information of the current user location and this then directs the SIP addresses with the physical location in the domain. The SIP proxy

server sends these messages to different proxy servers creating a structure similar to a tree so that the messages reach the destination [6]. The two different operating modes for these servers are stateless in which the server does not store any information once the request is sent and the second one is stateful in which the server saves the earlier routing information and then use it for improvement in the delivery [5]. The SIP redirect enables the endpoints to locate the required address by redirecting them to different server. Call Setup in SIP: The call setup process in SIP is relatively very simple when compared to other protocols such as H.323 and MGCP. When a caller s UA is sent to the redirect server using the INVITE message, then the redirect server uses this and enquires the location server to establish a path to the called party [6]. After this the redirect server transfers that particular information back again to the caller who in turn confirms this by sending an acknowledgement. Now, the caller will send a request again to the device or endpoint determined by the redirection server. Then the destination location will get this information and then send an acknowledgement to set up the call. This whole process is done through different redirect proxy servers.

[6]

A call is initiated in SIP using some basic messages which are relayed between the endpoints, servers or the proxy servers and gateways. INVITE: As the name suggests, it is a message for the session to initiate. The message will contain the information about the type of session. ACK: Gives an acknowledgement about the session establishment and this corresponds only to the INVITE. BYE: This message terminates or ends the session. CANCEL: If an INVITE is sent, hen this message is used to cancel a request that is pending. OPTIONS: Using this message, the user can have the capability to inquire REGISTER: With this message, the address is given to a current location and this will be a source for user data [5] [6]. Conclusion: In this paper, we have discussed the two prominent protocols used in VoIP and IP telephony such as H.323 and SIP. This paper also covers the components, characteristics and call flow process that is used in the protocols. It can thus be concluded that even though there are different protocols that are being used in IP telephony, but for now SIP and H.323 are the most used protocols and will take some considerable time to replace these. References: [1]Voice Over IP: Protocols and standards by Rakesh Arora [2] Understanding Voice over IP protocols, Cisco System Service provider solutions Engineering, February 2002 [3] Protocols for VoIP: http://examples.oreilly.com/9780596009625/openbook/ch08.pdf [4] SIP: Protocol Overview by RADVISION, the VoIP experts [5] Understanding SIP by Ubiquity, a white paper [6] Session Initiation Protocol Technology by IXIA