[Asterisk-code-review] chan sip: Trigger reinvite if the SDP answer is included in ... (asterisk[13])

Jenkins2 asteriskteam at digium.com
Thu Apr 27 16:01:55 CDT 2017


Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5536 )

Change subject: chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
......................................................................


chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK

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
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 1 deletion(-)

Approvals:
  George Joseph: Looks good to me, approved
  Jenkins2: Approved for Submit
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2ba52ab..ea77aff 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -28900,7 +28900,7 @@
 					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);

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Jean Aunis - Prescom <jean.aunis at prescom.fr>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>



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