]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
chan_sip: Set Quality of Service for video rtp instance
authorJonathan Rose <jrose@digium.com>
Tue, 25 Sep 2012 16:24:34 +0000 (16:24 +0000)
committerJonathan Rose <jrose@digium.com>
Tue, 25 Sep 2012 16:24:34 +0000 (16:24 +0000)
(closes issue ASTERISK-20201)
Reported by: ddkprog
Patches:
    chan_sip.c.diff uploaded by ddkprog (license 6008)
........

Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 381ce447bd99f7dc8c67daf7545d1bfb6b8b3e03..315f0fb358f3840eb7ea8b64f8025664302779c8 100644 (file)
@@ -5310,6 +5310,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
                ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
 
                ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+               ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
        }
 
        if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {