Note that the TCP and TLS support is currently considered experimental and

is subject to change while we work out the remaining issues.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -179,8 +179,8 @@ SIP changes
   * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
      were not properly torn down due to network or endpoint failures during an established
      SIP session.
-  * Added TCP and TLS support for SIP.  See doc/siptls.txt and configs/sip.conf.sample for
-     more information on how it is used.
+  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
+     configs/sip.conf.sample for more information on how it is used.
   * Added a new configuration option "authfailureevents" that enables manager events when
     a peer can't authenticate properly. 
   * Added DNS manager support to registrations not referencing a peer entry.

configs/sip.conf.sample

@@ -80,6 +80,12 @@ bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
 				; bindport is the local UDP port that Asterisk will listen on
 bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
 
+;
+; Note that the TCP and TLS support for chan_sip is currently considered
+; experimental.  Since it is new, all of the related configuration options are
+; subject to change in any release.  If they are changed, the changes will
+; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
+;
 tcpenable=yes                   ; Enable server for incoming TCP connections (default is yes)
 tcpbindaddr=0.0.0.0	        ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)