The telephone system and the Internet are two very different types of networks. The former is based on circuit switching which relies on the principle of reserving a connection between two nodes so that no other traffic can interfere for the duration of the connection. While some may call this process wasteful, there is no doubt that it generates an exceptionally reliable connection. However, it is estimated that anywhere between 30 to 80% of a regular call consists of silence that leads to a waste of bandwidth. Another feature of the telephone network is that everything is standardized and this makes it easy for two phones to communicate with each other.
VoIP on the other hand relies on the Internet which depends on packet switching instead of circuit switching. What this means is that many different types of traffic can be sent over the same link at the same time. This improves the throughput and makes for a far more efficient network. However, the drawback is that it can introduce delays as some packets are lost and are rerouted. Because of this, VoIP calls typically have a little more lag inbuilt into them than regular PSTN calls. There is a tolerance limit below which people don’t mind the latency and don’t even notice it. The aim of a VoIP configuration is to ensure that the latency never goes above this level. Here are few things we can do to reduce the amount of time it takes for a person to hear what the other person is saying.
To start off with, it will help if you select an SIP provider who lives near you. You may have noticed that on international calls using the PSTN system you experience more lag than you otherwise would. This is because the data has to travel over a longer distance. So naturally choosing an SIP provider who is closer to you or the people you talk to will reduce round-trip times and consequently reduce lag as well.
A few other techniques can be deployed such as changing the codecs in order to conserve more bandwidth – but you must keep in mind that higher compression also typically requires more processing power which might take a bit of time. Other parameters such as the ptime value relates to the packetization interval indicating the number of milliseconds encapsulated in a single IP packet. Clearly, the smaller this value, the faster the voice data leaves its source. But be careful when tweaking this parameter. Smaller ptime values dramatically increases the number of requests sent per second.
Contact your Internet Telephony Service Provider to find out the optimal configuration for your SIP VoIP client. Ensuring that all the settings are in line with what your SIP provider expects is one of the best ways to obtain maximum performance for your VoIP system.