[Asterisk-code-review] chan_sip: SIP route header is missing on UPDATE (asterisk[19])

[Kevin Harwell]

Kevin Harwell has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/18453 )

Change subject: chan_sip: SIP route header is missing on UPDATE
......................................................................

chan_sip: SIP route header is missing on UPDATE

if Asterisk need to send an UPDATE before answer
on a channel that uses Record-Route:
it will not include a Route header

ASTERISK-29955

Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef
---
M channels/chan_sip.c
1 file changed, 7 insertions(+), 3 deletions(-)

Approvals:
  Kevin Harwell: Looks good to me, approved; Approved for Submit



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b6aff80..77685b2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12443,9 +12443,8 @@
 	 * Send UPDATE to the same destination as CANCEL, if call is not in final state.
 	 */
 	if (!sip_route_empty(&p->route) &&
-			!(sipmethod == SIP_CANCEL ||
-				(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)) ||
-				(sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)))) {
+		!(sipmethod == SIP_CANCEL ||
+			(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)))) {
 		if (p->socket.type != AST_TRANSPORT_UDP && p->socket.tcptls_session) {
 			/* For TCP/TLS sockets that are connected we won't need
 			 * to do any hostname/IP lookups */
@@ -12453,6 +12452,11 @@
 			/* For NATed traffic, we ignore the contact/route and
 			 * simply send to the received-from address. No need
 			 * for lookups. */
+		} else if (sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)) {
+			/* Calling set_destination for an UPDATE in early dialog
+			 * will result in mangling of the target for a subsequent
+			 * CANCEL according to ASTERISK-24628 so do not do it.
+			 */
 		} else {
 			set_destination(p, sip_route_first_uri(&p->route));
 		}

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: 19
Gerrit-Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef
Gerrit-Change-Number: 18453
Gerrit-PatchSet: 3
Gerrit-Owner: Mark Petersen <asterisk.org at zombie.dk>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-CC: Mark Petersen <bugs.digium.com at zombie.dk>
Gerrit-MessageType: merged
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