[Asterisk-Users] Drops due to codecs?

[Daniel Flickinger]

Hello,

   It is my understanding that on the softphone side, asterisk is only 
responsible for establishing the session between two phones.  If this is the 
case, does it matter what type of audio codecs the two phones are using?  And 
if it does matter, are there any codecs that cause problems with asterisk 
bridging two SIP connections?  Thanks for your helpful input,

Daniel