[asterisk-commits] file: branch file/issue8855 r203708 - in /team/file/issue8855: channels/ conf...

[SVN commits to the Asterisk project]

Author: file
Date: Fri Jun 26 14:42:03 2009
New Revision: 203708

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=203708
Log:
Update documentation and just make a comedia option to nat that performs symmetric RTP without forcing rport.

Modified:
    team/file/issue8855/channels/chan_sip.c
    team/file/issue8855/configs/sip.conf.sample

Modified: team/file/issue8855/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/file/issue8855/channels/chan_sip.c?view=diff&rev=203708&r1=203707&r2=203708
==============================================================================
--- team/file/issue8855/channels/chan_sip.c (original)
+++ team/file/issue8855/channels/chan_sip.c Fri Jun 26 14:42:03 2009
@@ -22831,6 +22831,10 @@
 			ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
 			ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
 			ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
+		} else if (!strcasecmp(v->value, "comedia")) {
+			ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
+			ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
+			ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
 		}
 	} else if (!strcasecmp(v->name, "canreinvite")) {
 		ast_set_flag(&mask[0], SIP_REINVITE);
@@ -22911,9 +22915,6 @@
 	} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
-	} else if (!strcasecmp(v->name, "symmetricrtp")) {
-		ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
-		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_SYMMETRICRTP);
 	} else
 		res = 0;
 

Modified: team/file/issue8855/configs/sip.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/team/file/issue8855/configs/sip.conf.sample?view=diff&rev=203708&r1=203707&r2=203708
==============================================================================
--- team/file/issue8855/configs/sip.conf.sample (original)
+++ team/file/issue8855/configs/sip.conf.sample Fri Jun 26 14:42:03 2009
@@ -625,20 +625,10 @@
 ; However, this is only useful if the external traffic can reach us.
 ; The following settings are allowed (both globally and in individual sections):
 ;
-;        nat = no                ; default. Use NAT mode only according to RFC3581 (;rport)
-;        nat = yes               ; Always ignore info and assume NAT
-;        nat = never             ; Never attempt NAT mode or RFC3581 support
-;        nat = route             ; route = Assume NAT, don't send rport 
-;                                ; (work around more UNIDEN bugs)
-;
-; It is additionally possible to enable support for symmetric RTP without enabling NAT
-; support for signaling. This is accomplished using the symmetric_rtp configuration
-; option.
-;
-;         symmetric_rtp = no    ; default. Do not explicitly enable symmetric RTP support.
-;                               ; Some nat options will enable it regardless.
-;         symmetric_rtp = yes   ; Explicitly enable symmetric RTP support.
-;
+;        nat = no                ; Default. Use rport if the remote side says to use it.
+;        nat = force_rport       ; Force rport to always be on.
+;        nat = yes               ; Force rport to always be on and perform symmetric RTP.
+;        nat = comedia           ; Use rport if the remote side says to use it and perform symmetric RTP.
 
 ;----------------------------------- MEDIA HANDLING --------------------------------
 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's