Which VoIP codec is the best?

A VoIP codec is principally an algorithm which is set as a server program. This is utilized within a hardware device to convert voice data into digital signals. The signals are then transmitted through the internet or other system as you make a call.

In a conventional phone, analog voice data are transferred electronically starting from a caller’s phone to the recipient’s line. This takes place in a close circuit that stays open once the call is connected. Voice signals and other sound patterns are changed into an electromagnetic state through analog modulation within the circuit system of the telephone set. After this, the analog data are changed back into audible signals by the phone device of the receiver’s end.

How VoIP works?

Unlike the traditional telephone system, analog signals are converted to digital format in a VoIP system. This is basically switched to ones and zeros that denote details within a computer system. These are used to transfer sound signals between lengthy distances. This makes advances for computer and telephones using a packet-switched systems and circuit-switched set-up.

The most frequently used codecs in VoIP

After knowing how VoIP works, it can also be helpful to understand the VoIP codec used today. Codecs come as programs that perform the same data transmission to digital form. There are various codecs for text messaging, fax transmissions, videos, and audios.

  • G.711: This codec is otherwise known as PCM or pulse code modulation. This has 64kbit/s and known as the language of advanced digital phones among typical platforms. This codec renders precise speech indicator and only needs low processor strength. It is used in fax transfer in VoIP system, but needs a 128 bandwidth to make effective voice calls.
  • G.722: This VoIP codec is employed with different compressions to save on bandwidth and adjusts to varying system congestion. This offers high quality audio with its 16kbps. This is twice that of the typical telephone system and is valuable for VoIP system along LANs that are bandwidth-ready.
  • G.723.1: This codec is mostly utilized among VoIP networks as it only requires a low bandwidth. It provides a high quality audio and high compression. This can also be carried out in a dial-up system, although it entails lots of processing power.
  • G.726: It is an enhanced version of the G.723 and G.721 codec. This can be modified to several bandwidths and is utilized in international phone system and in wireless telephones.
  • G.729: As an error-resistance codec, this provides an excellent bandwidth usage at 8kbit/s. However, it requires license when used in devices and is also expensive when it comes to processing.
  • GSM: This is a free VoIP codec that is similar to the one used in GSM mobile phones. It is readily obtainable in several software and hardware platforms, and also has a great compression ratio and less bandwidth constraint.
  • iLBC: This is a speech codec which is considered as royalty-free that has 15kbit/s bandwidth. It can offer a forceful voice communication quality.
  • Speex: Known as an open-source acoustic codec, this is made chiefly for speech functions. Having a remarkable bandwidth ranging between 2.5 and 44 kbit/s, this is can minimize bandwidth usage though entails a CPU processor.

When learning how VoIP works, it follows to know the minute details that functions within the system including the VoIP codec.

Post Navigation