I have two box in different site connected via WAN link, now i am experiencing very poor voice quality.
What could be the problem?, i have tried to use ulaw,alaw and gsm codec but still no improvements.
also when i ping the end pbx i get delay of Max 160ms.
what could be the problem 
This is a support question not a discussion.
You have an overloaded network and/or your RTP traffic does not have enough priority. You may or may not have control over the part that is overloaded, other than by changing ISP. It is unlikely that your ISP will honour attempts to prioritise traffic, so if the bottleneck is not within your LAN or the outbound hop to your ISP’s router, you will probably not be able to fix the priority.
(This could also happen if you were running Asterisk on VMs without sufficient dedicated CPU or RAM.)
The ping times for UDP in RTP port range and with the same QoS markings as the RTP should be of the order of 20ms or less.
Thank you david55
that means has nothing to do with codecs…
The choice of codec has some impact on bandwidth used. However your ping time is too high for good VoIP, and the packet overhead tends to dominate the codec bit rate.
ulaw and alaw are telephone quality, as they are the codecs used for the PSTN (depends on location as to which).
gsm is standard mobile phone quality, so has more obvious artefacts and is not good for speech.