[Asterisk-code-review] chan_sip: SIP route header is missing on UPDATE (asterisk[18])
Kevin Harwell
asteriskteam at digium.com
Tue Apr 26 16:47:23 CDT 2022
Kevin Harwell has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/18452 )
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 beb512b..4960207 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12441,9 +12441,8 @@
* 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 */
@@ -12451,6 +12450,11 @@
/* 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: 18
Gerrit-Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef
Gerrit-Change-Number: 18452
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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