VoIP systems: Challenges And Solutions Part 2 VoIP Quality of Service Issues
May 4, 2015
VoIP Quality of Service Issues
Jitter: Jitter refers to non-uniform packet delays. It is often caused by low bandwidth situations in VoIP and can be exceptionally detrimental to the overall QoS. Variations in delays can be more detrimental to QoS than the actual delays themselves. Jitter can cause packets to arrive and be processed out of sequence.
Latency: Latency in VoIP refers to the time it takes for a voice transmission to go from its source to its destination. Ideally, we would like to keep latency as low as possible but there are practical lower bounds on the delay of VoIP.
Packet loss: Packet loss is another major QoS issue for VoIP systems. VoIP is exceptionally intolerant of packet loss. Packet loss can result from excess latency, where a group of packets arrives late and must be discarded in favor of newer ones. It can also be the result of jitter, that is, when a packet arrives after its surrounding packets have been flushed from the buffer, making the received packet useless.
Bandwidth: In computer networks, bandwidth is often used as a synonym for data transfer rate – the amount of data that can be carried from one point to another in a given time period (usually a second). So it is obvious that the more bandwidth we have better the call quality.One of the great attractions of VoIP, data and voice sharing the same wires, is also a potential headache for implementers who must allocate the necessary bandwidth for both networks in a system normally designed for one. Congestion of the network causes packets to be queued, which in turn contributes to the latency of the VoIP system. Low bandwidth can also contribute to non-uniform delays (jitter), since packets will be delivered in spurts when a window of opportunity opens up in the traffic.
Session Initiation Protocol (SIP): The Session Initiation Protocol (SIP) is a signaling protocol, widely used in VoIP systems it is extremely popular. The SIP protocol is simple and text based like the HTTP protocol. The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP requires a SIP server and a SIP client to work properly.
Real time transport protocol (RTP): The real time transport protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP is UDP based and due to this is not highly reliable but due to the nature of VoIP traffic hundred percent reliability is not essential. RTP is designed for end-to-end, real-time, transfer of stream data. The protocol provides facilities for jitter compensation and detection of out of sequence arrival in data, which are common during transmissions on an IP network. RTP allows data transfer to multiple destinations through IP multicast. Stay tune for our 3rd part to this series next week VoIP security issues.
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