Managing VoIP Quality

The TCP/IP suite of protocols was designed with some safeguards against packet latency and alternative routing. When sending a data file, some packets may take a different path than others and the packets may arrive out of order. But, with TCP/IP that is OK because the protocols understand how to sequence the packets and reassemble them in the proper order so that the data arrives intact- even if it is a few milliseconds later.

Well, when you put voice communications on a TCP/IP network the demands are a little different. The PowerPoint presentation being downloaded won’t really matter if it takes a millisecond longer or what order the packets arrive in as long as they arrive. However, voice communications is now. It is real-time. It is imperative that the packets reach the listeners ear in order and with minimal delay in order to facilitate a voice conversation and mirror as closely as possible the voice communications experience that callers are used to with traditional telephone systems. Streaming video for online video calls or video conferencing is even more data intensive and still demands that the data get from point A to point B as quickly as possible, uninterrupted and in order.

This is one of the challenges that organizations face as they attempt to migrate to VoIP and unified communications. The underlying network architecture and available bandwidth are a critical foundation that can make or break the success of VoIP and unified communications in the organization. These needs have opened the door for a whole new generation of networking and quality of service equipment designed specifically around VoIP and bandwidth-intensive streaming media applications.

 Ensuring Voice and Video Quality About More Than Watching Packet Flows.