[asterisk-commits] mmichelson: branch mmichelson/correct_sdp_answer r206696 - /team/mmichelson/c...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jul 15 11:40:39 CDT 2009
Author: mmichelson
Date: Wed Jul 15 11:40:35 2009
New Revision: 206696
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=206696
Log:
Looking at the example in RFC 3264, it appears that we should NOT clear
the offered_media when sending a reinvite. This seems really strange, but
hey, whatever.
Modified:
team/mmichelson/correct_sdp_answer/channels/chan_sip.c
Modified: team/mmichelson/correct_sdp_answer/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mmichelson/correct_sdp_answer/channels/chan_sip.c?view=diff&rev=206696&r1=206695&r2=206696
==============================================================================
--- team/mmichelson/correct_sdp_answer/channels/chan_sip.c (original)
+++ team/mmichelson/correct_sdp_answer/channels/chan_sip.c Wed Jul 15 11:40:35 2009
@@ -7158,7 +7158,6 @@
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
- memset(p->offered_media, 0, sizeof(p->offered_media));
add_sdp(&req, p, 1, 0);
/* Use this as the basis */
initialize_initreq(p, &req);
@@ -7181,7 +7180,6 @@
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug)
add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
- memset(p->offered_media, 0, sizeof(p->offered_media));
add_sdp(&req, p, 0, 1);
/* Use this as the basis */
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