Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."
authorzuul <zuul@gerrit.asterisk.org>
Fri, 9 Sep 2016 18:56:16 +0000 (13:56 -0500)
committerGerrit Code Review <gerrit2@gerrit.digium.api>
Fri, 9 Sep 2016 18:56:16 +0000 (13:56 -0500)
1  2 
CHANGES
res/res_pjsip/pjsip_configuration.c
res/res_pjsip_session.c

diff --cc CHANGES
+++ b/CHANGES
  --- 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 ----------
  ------------------------------------------------------------------------------
Simple merge
Simple merge