Express Talk

[Oj234]

Hello,

 

I'm using asterisk for VoIP calls with SIP extensions.

Actually, the software I'm using for the SIP extensions is X-lite but the sound quality is horrible.

I would like to know , in express talk, wich codecs are used/supported and if the sound quality is good.

[pythonpoole]

Express Talk supports G.711 (alaw/ulaw), GSM, and I think it also supports G.726. It's the same for all of NCH's other VoIP software. The reason NCH doesn't support G.729 or G.723 is because of their proprietary nature which would require the purchase of a license for each person using the software (making it impossible to provide for free, and also increasing the price of the business edition).

 

I have no problem with the quality of Express Talk calls, but here are some tips to help you out:

 

G.711 provides crystal clear audio quality, but will break up the voice if there isn't enough bandwidth available (be warned, it uses a lot of bandwidth!)

G.726 provides good quality clear audio, using much less bandwidth (with less or no voice break-ups)

GSM (same codec used on GSM cellphones) provides satisfactory audio quality (not as clear) using very little bandwidth (virtually no voice break-ups)

 

The quality of audio can also be affected by many other factors. To improve audio quality and reduce voice break-ups:

- Be sure to forward the necessary ports on your router/firewall

- Be sure to have a stable Internet connection that is relatively fast (consider upgrading if you have lower-end hispeed)

- If your router supports it, be sure to enable and set-up the Quality of Service feature. This will help put priority on VoIP traffic over regular Internet traffic

 

Other factors that may affect the quality (things you can't really change):

- The distance between you and the VoIP server (the larger the distance, the larger the delay, and also the greater chance of audio quality degrading)

- How your ISP decides to manage your Internet traffic. If your ISP doesn't put priority on VoIP traffic, it could result in broken audio

- The bandwidth available on the server end (of your VoIP provider) could be causing voice quality

- The codecs allowed by your VoIP service could be limiting your choices to help save bandwidth or improve audio clarity