Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."

diff --cc CHANGES
index 2851b2639ed8271922bad185da9360a2f007c936,6032dd8a053a6471f9e436ce819a6687b4be5ab2..bf7bc75de92bcaf6978b8089bfb2b544f327f402
--- 1/CHANGES
--- 2/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 ----------
  ------------------------------------------------------------------------------

diff --cc res/res_pjsip/pjsip_configuration.c
Simple merge

diff --cc res/res_pjsip_session.c
Simple merge