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Merged revisions 340971 via svnmerge from

Browse Source 
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
This commit is contained in:
Kinsey Moore
2011-10-14 20:51:19 +00:00
parent e77f1a6ae1
commit 4b9546abdf
2 changed files with 62 additions and 36 deletions
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5
channels/chan_sip.c
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@@ -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)