- Back -

Research carried out

Head-of-the-Line (HOL) blocking is a phenomenon that can occur when TCP is used as a transport protocol in Session Initiation Protocol (SIP) signalling. TCP assumes a single stream of data and ensures that the segments of that stream are delivered in the sequence in which they are sent. In telephony call set-up most of the segments in the stream will not be inter-related. The majority will be independent from each other. The problem occurs when sending independent messages over an order-preserving TCP connection which in turn causes the delivery of messages sent later to be delayed within a receiver's lower layer buffers until an earlier lost/delayed message is retransmitted and arrives. Segments behind the recovering packet are delayed even though they may not be related to it. This can have undesirable effects on the performance of Applications using the protocol.

The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple telephone call between two parties or it could be a collaborative multimedia conference session. The ability to establish these sessions means that a large amount of innovative services become possible, such as voice enriched internet content, make a voice call from a link on a web page, Instant Messaging etc. The Voice over IP (VoIP) community has adopted SIP as its protocol of choice for signalling. At the moment SIP is still evolving and being extended as new technology arises.
When creating a session SIP specifies only what it needs to specify. This saves on the amount of data that needs to be sent, thus making the whole process faster. SIP was developed purely as a mechanism to establish sessions, it is not concerned about the details of a session, it just initiates, terminates and modifies sessions. This simplicity means that SIP is scaleable, extensible, and it sits comfortably in different deployment scenarios and architectures. SIP is a request response protocol that closely resembles two other Internet protocols, HTTP and SMTP. Using SIP, telephony becomes another viable web application and integrates easily into other Internet services. Therefore the protocol is basically a simple toolkit that converged-voice and multimedia services can be created with.

VoIP (voice over IP) is basically voice delivered using the Internet Protocol (IP). In general, this means sending voice information in digital form in individual packets rather than in the circuit-committed protocols of the public switched telephone network. A major advantage of VoIP and Internet telephony is that it is cheaper as it avoids the charges charged by local telephone services. To help ensure that packets get delivered in a timely fashion VoIP uses the real-time protocol (RTP). When using public networks, it is currently quite difficult to guarantee Quality of Service (Qos) in a voice call, for example to guarantee that the call will stay at a certain standard 100% of the time. A more suitable service is possible when provided by private networks specifically for VoIP that can guarantee a certain Quality of Service. One technique used to help ensure faster packet delivery is to ping all possible network Gateways that have access to the public network and choose the optimum path before establishing a TCP connection with the other end.

In order to make any progress in this project, two programming languages were vital. These languages are C++ and Tcl. The C++ code is discussed in greater detail in the chapter concerning the physical implementation of the protocol. The Tcl scripting can be seen in the chapter concerning the simulation of the protocol.