[asterisk-commits] mmichelson: branch group/pimp_my_sip r379827 - /team/group/pimp_my_sip/res/

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Tue Jan 22 09:13:09 2013
New Revision: 379827

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379827
Log:
Change SDP answer not to have random origin id and version.

Some phones, and Asterisk's old chan_sip, do not like if the SDP
version in an answer is lower than the version in the offer and will
immediately hang the call up after the INVITE transaction completes.

Now we base our SDP answer id and version off the incoming SDP offer.
This makes my Polycom phone actually manage to keep a call up rather
than immediately hanging up.


Modified:
    team/group/pimp_my_sip/res/res_sip_session.c

Modified: team/group/pimp_my_sip/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pimp_my_sip/res/res_sip_session.c?view=diff&rev=379827&r1=379826&r2=379827
==============================================================================
--- team/group/pimp_my_sip/res/res_sip_session.c (original)
+++ team/group/pimp_my_sip/res/res_sip_session.c Tue Jan 22 09:13:09 2013
@@ -711,7 +711,8 @@
 		return;
 	}
 	answer = PJ_POOL_ZALLOC_T(inv->pool, pjmedia_sdp_session);
-	answer->origin.version = answer->origin.id = (pj_uint32_t)(ast_random());
+	answer->origin.version offer->origin.version + 1;
+	answer->origin.id = offer->origin.id;
 	pj_strdup2(inv->dlg->pool, &answer->origin.user, "Asterisk");
 	/* XXX Hardcoded origin net stuff for now. Replace with transport
 	 * stuff later