I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary
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On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change
at some point, mid-call? Or do you mean: Will Asterisk properly
react to such a re-INVITE and change codecs if asked to do so by
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
and
Will Asterisk properly react to such a re-INVITE and change codecs if asked
to do so by the dialog counterparty?
On Tue, May 3, 2011 at 12:56 PM, Alex Balashov
On 05/03/2011 01:16 PM, Gary Graves wrote:
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
I don't know of a way to do that. I suppose it might be possible if a
call were asynchronously transferred to a SIP peer that had different
codec
