kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-13 00:04:53 +00:00
Code Issues Packages Projects Releases Wiki Activity

Merged revisions 340971 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/branches/10

................
  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)
  ........
................


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
Download Patch File Download Diff File Expand all files Collapse all files

5
channels/chan_sip.c
View File

@@ -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)

21
res/res_rtp_asterisk.c
View File

@@ -2382,10 +2382,12 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (property == AST_RTP_PROPERTY_RTCP) {
if (value) {
if (rtp->rtcp) {
ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
return;
}
/* Setup RTCP to be activated on the next RTP write */
if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
return;
}
@@ -2420,6 +2422,25 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
rtp->rtcp->schedid = -1;
return;
} else {
if (rtp->rtcp) {
if (rtp->rtcp->schedid > 0) {
if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
/* Successfully cancelled scheduler entry. */
ao2_ref(instance, -1);
} else {
/* Unable to cancel scheduler entry */
ast_debug(1, "Failed to tear down RTCP on RTP instance '%p'\n", instance);
return;
}
rtp->rtcp->schedid = -1;
}
close(rtp->rtcp->s);
ast_free(rtp->rtcp);
rtp->rtcp = NULL;
}
return;
}
}
return;