From: zuul Date: Fri, 9 Sep 2016 18:56:16 +0000 (-0500) Subject: Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint." X-Git-Tag: 15.0.0-beta1~654 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=9d54dd04bbdad6849aee77536ab12c5fa6620680;p=thirdparty%2Fasterisk.git Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint." --- 9d54dd04bbdad6849aee77536ab12c5fa6620680 diff --cc CHANGES index 2851b2639e,6032dd8a05..bf7bc75de9 --- a/CHANGES +++ b/CHANGES @@@ -12,14 -12,14 +12,21 @@@ --- Functionality changes from Asterisk 14 to Asterisk 15 -------------------- ------------------------------------------------------------------------------ +chan_sip +------------------ + * If an offer is received with optional SRTP (a media stream with RTP/AVP but + which contains a crypto line) chan_sip will now accept it and enable SRTP. + If you would like to do optional SRTP on outbound you will need to create + a dialplan that dials with it enabled initially and if it fails fall back to + without. + res_pjsip + ------------------ + * Added endpoint configuration parameter "preferred_codec_only". + This allow asterisk response to a SIP invite with the single most + preferred codec rather than advertising all joint codec capabilities. + This limits the other side's codec choice to exactly what we prefer. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ---------- ------------------------------------------------------------------------------