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

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index c8ad051d3671b3bd2262328888541415757d4c3f..b34486a59d137549c4a49dce70ffe73f71189c24 100644 (file)
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -217,7 +217,7 @@ 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) &&