[Asterisk-code-review] chan sip: Allow target refresh (Contact update) on re-INVITE. (asterisk[master])

Anonymous Coward asteriskteam at digium.com
Tue Sep 13 10:26:51 CDT 2016


Anonymous Coward #1000019 has submitted this change and it was merged.

Change subject: chan_sip: Allow target refresh (Contact update) on re-INVITE.
......................................................................


chan_sip: Allow target refresh (Contact update) on re-INVITE.

Previously, the Contact was stored only on initial INVITE and on any
18X and 200. That meant that after re-INVITEs from *us* the Contact
could get updated, but after re-INVITEs from the *peer*, it did not.

This changeset fixes this inconsistency, properly allowing target
refreshes through re-INVITES (RFC3261, 12.2).

If your strictrtp setting allows it, this change allows you to switch
the source IP of a connected/calling device mid-call with a simple
re-INVITE from the new IP.

ASTERISK-26358 #close

Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435
---
M channels/chan_sip.c
1 file changed, 4 insertions(+), 1 deletion(-)

Approvals:
  George Joseph: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 28ba05f..8d84d55 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -26113,12 +26113,15 @@
 		copy_request(&p->initreq, req);		/* Save this INVITE as the transaction basis */
 		if (sipdebug)
 			ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
+
+		/* Parse new contact both for existing (re-invite) and new calls. */
+		parse_ok_contact(p, req);
+
 		if (!p->owner) {	/* Not a re-invite */
 			if (req->debug)
 				ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
 			if (newcall)
 				append_history(p, "Invite", "New call: %s", p->callid);
-			parse_ok_contact(p, req);
 		} else {	/* Re-invite on existing call */
 			ast_clear_flag(&p->flags[0], SIP_OUTGOING);	/* This is now an inbound dialog */
 			if (get_rpid(p, req)) {

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Walter Doekes <walter+asterisk at wjd.nu>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>



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