]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
res_pjsip_sdp_rtp.c: Fix cut-n-paste error
authorRichard Mudgett <rmudgett@digium.com>
Thu, 16 Mar 2017 20:45:57 +0000 (15:45 -0500)
committerRichard Mudgett <rmudgett@digium.com>
Thu, 16 Mar 2017 20:49:13 +0000 (14:49 -0600)
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

res/res_pjsip_sdp_rtp.c

index 8c1c9383d3b301c4741499d54b404ad7c30bf149..171e68bdd7da4a95e2d22ddd9bbb6a6b9fc7b8ee 100644 (file)
@@ -218,7 +218,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
        }
 
        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) &&