[asterisk-commits] res pjsip sdp rtp.c: Fix cut-n-paste error (asterisk[master])

[SVN commits to the Asterisk project]

Joshua Colp has submitted this change and it was merged. ( https://gerrit.asterisk.org/5235 )

Change subject: res_pjsip_sdp_rtp.c: Fix cut-n-paste error
......................................................................


res_pjsip_sdp_rtp.c: Fix cut-n-paste error

We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
---
M res/res_pjsip_sdp_rtp.c
1 file changed, 1 insertion(+), 1 deletion(-)

Approvals:
  George Joseph: Looks good to me, approved
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 9f39256..9fc1b8a 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -217,7 +217,7 @@
 	}
 
 	if (!strcmp(session_media->stream_type, STR_AUDIO) &&
-			(session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
+			(session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
 		ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
 				session->endpoint->media.cos_audio, "SIP RTP Audio");
 	} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>