]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
authorJean Aunis <jean.aunis@prescom.fr>
Thu, 20 Apr 2017 07:13:13 +0000 (09:13 +0200)
committerJean Aunis - Prescom <jean.aunis@prescom.fr>
Wed, 26 Apr 2017 14:51:30 +0000 (09:51 -0500)
Some equipments may send a re-INVITE containing an SDP in the final ACK
request. If this happens in the context of direct media, the remote end
should be updated with a re-INVITE.
This patch queues an "update RTP peer" frame to trigger the re-INVITE,
instead of the "source change" frame wich was used previously.

ASTERISK-26951

Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6

channels/chan_sip.c

index 2ba52ab7d31ba7fc3ebbe92c4af70ffe7739c577..ea77aff14c5a310dff94bf0550df2aeb46e22f94 100644 (file)
@@ -28900,7 +28900,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
                                        return -1;
                                }
                                if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
-                                       ast_queue_control(p->owner, AST_CONTROL_SRCCHANGE);
+                                       ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
                                }
                        }
                        sched_check_pendings(p);