[Asterisk-Users] Problems with SIP codec selection

[Kevin P. Fleming]

Jamie Neil wrote:

> Is this a bug, by design or is it a problem specific to Xlite?

Neither. Your analysis is correct. When the SIP INVITE comes in, and the 
user has not been asked to authenticate yet, the codec settings used are 
the "global" ones from the [general] section of sip.conf. This is 
necessary because at this point we do not _have_ a user name, there is 
no way we can look in a user-specific section for preferences/settings.

Once the user has authenticated, then the codec matching uses the 
settings for that user.

However, I think it might be possible to make this work a bit better, 
and not bother checking the codecs for matches if all we're going to do 
is tell the user to authenticate anyway... I'll look into it.