Codecs are an important element in quality Voice over IP (VoIP) communications and video conferencing. Codec comes from the combination of COder/DECoder and defines the way analog voice (audio) is compressed and converted into digital IP packets as they flow over IP network infrastructure and are then decompressed into analog audio for the receiver on the other side of the conversation.Codecs play a role in both the quality of voice sound as well as the bandwidth consumed by the session.
There are several different standards-based and proprietary Codecs commonly used today.
- G.711 – This ITU standard Codec is uncompressed and thus consumes higher bandwidth than other Codecs so is recommended for areas with higher available bandwidth. This will also mean the quality will be better.Two versions of this are available – U-law that is compatible with the T1 standards used in North America and Japan and A-Law which conforms to E1 requirements used broadly around the world. This is a royalty free codec.
- G.722 – This ITU standard, with the highest consumption of bandwidth among the most common Codecs, has exceptional quality. As all patents have expired, this is a royalty free codec.
- G.729 – One of the most commonly used Codec in VoIP, it offers good voice quality while having low bandwidth requirements. Use of this Codec will require a license.
Issues related to Codecs can impact the quality that users experience. For instance if remote offices are bandwidth constrained, configuring G.722 could exacerbate the congestion issue due to the high bandwidth demands G.722 require. A failed international call could be tied to not enabling G.729 as a negotiated codec. Determining the reason for call quality issues in this part of a VoIP implementation requires tools that closely monitor performance of codec-related metrics.