[Asterisk-Users] codec negotiation

[Ronald Wiplinger]

We have four settings for the codec.

How will it be negotiated?
How should it be negotiated in relation to the available bandwidth?
Is there an influence by using canreinvite=yes ?

Phone A has a setting for the priority of codec
Sip.conf has (maybe even different) settings for the priority of codec 
of this phone A
Sip.conf has codec settings for the destination phone B
Phone B has (maybe even different) settings for the priority of codec

Which codec will be taken?
a. if the call goes via * ?
b. if the call will be completed with canreinvite=yes ?

Which codec should be enforce depending on the bandwidth?

Thanks for thinking with me ;-)


bye

Ronald Wiplinger