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 the dialog counterparty?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200?
Doesn't seem to. Looking at chan_sip.c in 1.6.2.13, there is no call to add_sdp() that is not made either in the context of 1) an initial INVITE request or 2) a re-INVITE or 3) the construction of a response. Nothing in the case of the production of an end-to-end ACK.
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