document G.722.1/.1C support

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
17 years ago · 3854faf2d7
parent 2a53f2ec98
commit 3854faf2d7
1 changed files with 3 additions and 0 deletions
--- a/3
+++ b/3
@ -48,6 +48,7 @@ SIP Changes
   first INVITE is generated - SIPRemoveHeader()
 * Channel variables set with setvar= in a device configuration is now 
   set both for inbound and outbound calls.
+ * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.

 Skinny Changes
 --------------
@ -135,6 +136,8 @@ Miscellaneous
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
   give you the correct output for an asterisk box behind nat. It will give you the
   externhost and localnet settings.
+ * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
+   can connect calls in passthrough mode, as well as record and play back files.

 Asterisk Manager Interface
 --------------------------