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TCP or UDP – Which protocol does VoIP use?

When it comes to telephone systems and the future, landlines are already out of the competition. VoIP is an advanced telephony that allows its users to make calls over the internet without the need of traditional landline networks. To do so, VoIP converts analogue data in to packets and then sends it over the internet to its destination. This service is made possible thanks to a set of listening ports, of which the primary ones are TCP and UDP ports.

What is TCP and UDP ports?

TCP stands for Transmission Control Protocol, in which data packets connect directly from source to destination and stay connected for the duration of the transfer. This method is usually very accurate and guarantees the delivery of data packets in its original order. That’s why it’s used where accuracy is more important than efficiency such as websites and emails.

The TCP process progresses in the following way:

  • Endpoint A sends data 1 to endpoint B
  • Endpoint B receives data 1 and send confirmation to endpoint A
  • Endpoint A received the confirmation and proceed to send data 2.
  • If Endpoint A didn’t receive the confirmation within a certain period, the transmission will be initiated again before sending any more data.

TCP accuracy means audio traffic through this protocol will be less efficient on resources. This is due to TCP’s insistence on guaranteed packet delivery and missing packets will break the entire call.

UDP on the other hand stands for User Datagram Protocol, which is very different to TCP as it’s a connectionless protocol that has no control over errors. This means that data packets will be sent as quickly as possible without taking actions to ensure that all packets are received or that they have been received in the correct order. These errors are usually translated as jitters in a VoIP call. UDP is ideal for real-time applications and services where it’s important to keep the stream of data and information going.

 

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Why UDP is used with VoIP?

UDP provides a better experience for VoIP users as it allows them to enjoy a real-time and uninterrupted call without any delay. Errors such as packets losses have minor impact on the output audio and usually goes unnoticed. TCP errors on the other hand translated with streaming services manifest themselves differently. For example, a YouTube video that keeps freezing every now and then for few seconds while trying to deliver a high-quality image. VoIP users won’t really appreciate such experience when they are making that all important phone call.

How to improve the quality of a VoIP call?

The first thing you need to look at when trying to improve the quality of your VoIP call is your broadband. VoIP relies on the internet and thus, a slow internet connection will negatively impact the quality of your call. For example, if you run a call centre that usually handle many concurrent calls at the same time, you may want to upgrade your bandwidth to minimise jitters and improve your client’s experience.

Another thing you should consider is your router configurations. You need to make sure that your router has been setup to prioritize VoIP traffic on your network so that your VoIP call won’t be affected by other users using the same network to stream videos or download large files. Learn more about minimising VoIP jitters in our article “VoIP jitters – can you defeat them?

You might also be interested to read “Is VoIP legal around the world?

If you have more questions, please don’t hesitate to call us on: 03300881182, or send us an email to: sales@vipvoip.co.uk

 

 

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