chan_pjsip: Add debug in RTP Engine glue callback

This patch adds some debug statements that aid with determining why a direct
media request may or may not be initiated.
........

Merged revisions 415117 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 4876f02..26dda4a 100644 (file)
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -303,10 +303,12 @@ static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
 
        /* Don't try to do any direct media shenanigans on early bridges */
        if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
+               ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
                return 0;
        }
 
        if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
+               ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
                return 0;
        }
 
@@ -318,6 +320,7 @@ static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
        }
 
        if (direct_media_mitigate_glare(session)) {
+               ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(chan));
                return 0;
        }
 
@@ -329,7 +332,7 @@ static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
        if (changed) {
                ao2_ref(session, +1);
 
-
+               ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(chan));
                if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
                        ao2_cleanup(session);
                }