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 <[email protected]>wrote: > 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. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
