Can you answer both?<div><br></div><div>Can Asterisk be configured to _initiate_ such a change at some point, mid-call? </div><div><br></div><div>and</div><div><br></div><div>Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty?<br>
<br><div class="gmail_quote">On Tue, May 3, 2011 at 12:56 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">On 05/03/2011 12:43 PM, Gary Graves wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Can you change codecs mid-call upon re-invite?<br>
</blockquote>
<br></div>
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?<div class="im">
<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Can you handle the SDP offer-answer in the 200-ACK instead of the<br>
usual INVITE-200?<br>
</blockquote>
<br></div>
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.<br>
<br>
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</blockquote></div><br></div>