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chan_pjsip: Add debug in RTP Engine glue callback

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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
This commit is contained in:
Matthew Jordan
2014-06-04 14:13:07 +00:00
parent 4f603c5da3
commit 6b423e311b
1 changed files with 4 additions and 1 deletions
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5
channels/chan_pjsip.c
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@@ -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);
}