Real-Time Streaming Protocol or RTSP is a protocol associated with the client-server application control protocol. The protocol is mainly used for repositioning with regards to streaming media retrieved over a communication channel. Among the uses of this protocol include functions such as rewinding, fast-forwarding, pausing, and resumption of media access. Real-time Protocol or RTP is a protocol associated with the transport layer in the OSI architecture. It operates over UDP and is concerned with payload type identification, packet sequence numbering, and time stamping. This protocol provides data about the data and doesn’t provide any additional data handling service. Real-Time Control Protocol or RTCP works in conjunction with RTP. The main purpose of RTCP is the synchronization of different media streams within an RTP packet. An RTP packet could contain a video and audio stream, which through RTCP will be executed in a synchronized fashion.
Session Initiation Protocol or SIP is used to set up a call across the internet. Among the assumptions behind the SIP is the assumption that all calls joint sessions such as video conferences take place over the internet. As such, it is important to be able to reach the desired individual regardless of their location. Through this protocol, the caller can alert the receiver and map a path to link the two together. H323 is a protocol that is similar to SIP in that it is used as a means for signaling real-time communication. Whereas SIP is a single protocol that works with RTP, H323 is a suite of integrated protocols that has its roots in ITU or telephony. SIP borrows largely from HTTP and is thus web suited, while H323 is more telephony-suited owing to its ITU roots. SIP is relatively simple in comparison to H323.
Jitter refers to the variation in packet delays within the same packet stream. For best results with streaming media, a loss tolerant link is best suited as the minimal number of delays will cause fewer glitches in presentation. Delay in the case of streaming media refers to the duration between the receipts of streaming data packets. The fewer the delays encountered between packets, the less the jitter will be and hence the better the presentation. A jitter buffer may be necessary because though there is a long delay, there is no loss of data. For improved presentation, the buffer will assist in ensuring smooth play.