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authorWalter Doekes <walter+github@wjd.nu>2016-09-12 10:28:17 +0200
committerWalter Doekes <walter+asterisk@wjd.nu>2016-09-12 03:40:54 -0500
commitda8ba990d13c0dd396b625cc98a9e00ca85d986a (patch)
tree21c24f7b0b95cc379f2a54830bd5eec8c8bf1bdb /channels
parent7580a736bb30577d7557aac9165894dc9f9583e6 (diff)
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
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c5
1 files changed, 4 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b03a3e34d..9c9a407ad 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -26057,12 +26057,15 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
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)) {