diff options
| author | Kinsey Moore <kmoore@digium.com> | 2011-10-14 20:51:19 +0000 |
|---|---|---|
| committer | Kinsey Moore <kmoore@digium.com> | 2011-10-14 20:51:19 +0000 |
| commit | 4b9546abdfb732758371e64976d65debe81fc067 (patch) | |
| tree | e31fff60e060655098b472b0da47c3e520ba7766 /channels | |
| parent | e77f1a6ae19d5a51b839d8419ae6b1203770f6cb (diff) | |
Merged revisions 340971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines
Merged revisions 340970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
(closes issue ASTERISK-18400)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 2d340f735..ff9689895 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -9127,6 +9127,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp); + /* Ensure RTCP is enabled since it may be inactive + if we're coming back from a T.38 session */ + ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1); if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) { ast_clear_flag(&p->flags[0], SIP_DTMF); @@ -9143,6 +9146,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action } else if (udptlportno > 0) { if (debug) ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n"); + /* Silence RTCP while audio RTP is inactive */ + ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0); } else { ast_rtp_instance_stop(p->rtp); if (debug) |