Here is an example:
Call comes in via PSTN... ulaw is the native format of the channel.
On the sip side you have g729,ulaw as the codec order. That call
will end up being ulaw because we send the native format as our first
choice above all because we don't want to transcode.
/b
On Aug 15, 2005, at 1:10 PM, Tony Hoyle wrote:
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently
(use sip show peer <peer> and look for "codec order" entry), so,
if you have prefered codec g729 for your sip phone/peer, asterisk
will use them (regardles of codec setting for other party - if
codecs does not match, asterisk will try to transcode between)
imho ;-)
It does seem to be a weakness of asterisk.. it's creating load on
the server when it doesn't need to.
Really it should look at the capabilities of both ends and see if
there's a common set, and only start transcoding if there's no
overlap.
Tony
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