[asterisk-commits] russell: trunk r155929 - /trunk/channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Nov 11 10:07:36 CST 2008


Author: russell
Date: Tue Nov 11 10:07:36 2008
New Revision: 155929

URL: http://svn.digium.com/view/asterisk?view=rev&rev=155929
Log:
Remove commentary from the issues list for SIP TCP/TLS

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=155929&r1=155928&r2=155929
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Nov 11 10:07:36 2008
@@ -87,18 +87,9 @@
  * the sip_hangup() function
  */
 
-/*!  \page sip_tcp_tls SIP TCP and TLS support
- * The TCP and TLS support is unfortunately implemented in a way that is not 
- * SIP compliant and tested in a SIP infrastructure. We hope to fix this for 
- * at least release 1.6.2. This code was new in 1.6.0 and won't be fixed for
- * that release, due to the current release policy. Only bugs compared with
- * the working functionality in 1.4 will be fixed. Bugs in new features will
- * be fixed in the next release. As 1.6.1 is already in release
- * candidate mode, there will be a buggy SIP channel in that release too.
- *
- * If you have opinions about this release policy, send mail to the asterisk-dev
- * mailing list.
- *
+/*!  
+ * \page sip_tcp_tls SIP TCP and TLS support
+ * 
  * \par tcpfixes TCP implementation changes needed
  * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
  * \todo Save TCP/TLS sessions in registry




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