[Asterisk-code-review] res pjsip sdp rtp.c: Fix cut-n-paste error (asterisk[14])

[Richard Mudgett]

Richard Mudgett has uploaded a new change for review. ( https://gerrit.asterisk.org/5234 )

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(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/34/5234/1

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 8c1c938..171e68b 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -218,7 +218,7 @@
 	}
 
 	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) &&

-- 
To view, visit https://gerrit.asterisk.org/5234
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>