How can we improve the voice quality in Asterisk?

Hi,

My SIP call is working using Android SIP client csip.
Voice quality is ok but still not comparable to Skype & Viber.

I know it depends on the network conditions like packet delay, jitter, codecs & other factors as well.
Is there any optimization parameter in Asterisk to improve voice quality ?

Would appreciate your valuable inputs.

Thanks !!
BR///Ankush Makkar

Ultimately compensating for network quality issues are the responsibility of the phones, although you also need to avoid things like running on a VM with inadequate resources.

The main control of quality within Asterisk itself is the choice of codec.

You can request Asterisk to prioritise network traffic, but unless the network is set up for that, it will make no difference. In any case, Skype connections are unlikely to benefit from such optimisations.

just use G711 new algorithm (slower but cleaner)

G.711 is very old! There are two variants, A-law and mu-law (mu- for North America and A- for most of the rest of the world). They are the codecs used on the PSTN, so they will give you telephone quality audio. Asteisk mis-spells mu as u.

You might try G722 , but as David said before the G711 ulaw/alaw are the codecs used on the PSTN, so they will give you telephone quality audio.