[asterisk-users] Codec negotiation

[Ryan Wagoner]

I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my voip provider and transcoding these to ulaw. In
sip.conf for the provider I have deny=all and allow=ulaw. This can
cause potential audio degrading and wastes cpu cycles. If Asterisk
knows the trunk only supports ulaw why would it offer g722 to the
phone.

Ryan