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When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.

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(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2008-03-04 18:05:28 +00:00
parent 40425a24bf
commit 36bb1f9d46
3 changed files with 22 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files
channels/chan_sip.c
+6 -1
View File
@@ -3661,8 +3661,10 @@ static int sip_answer(struct ast_channel *ast)
if (option_debug > 1)
ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name);
res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
} else
} else {
ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
}
}
ast_mutex_unlock(&p->lock);
return res;
@@ -3695,6 +3697,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_rtp_new_source(p->rtp);
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
@@ -3920,9 +3923,11 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
res = -1;
break;
case AST_CONTROL_HOLD:
ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
include/asterisk/rtp.h
+2
View File
@@ -162,6 +162,8 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Setting RTP payload types from lines in a SDP description: */
void ast_rtp_pt_clear(struct ast_rtp* rtp);
/*! \brief Set payload types to defaults */
main/rtp.c
+14
View File
@@ -172,6 +172,7 @@ struct ast_rtp {
struct ast_rtcp *rtcp;
struct ast_codec_pref pref;
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
int set_marker_bit:1; /*!< Whether to set the marker bit or not */
};
/* Forward declarations */
@@ -1994,6 +1995,12 @@ int ast_rtp_settos(struct ast_rtp *rtp, int tos)
return res;
}
void ast_rtp_new_source(struct ast_rtp *rtp)
{
rtp->set_marker_bit = 1;
return;
}
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
rtp->them.sin_port = them->sin_port;
@@ -2641,6 +2648,13 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
}
}
}
/* If we have been explicitly told to set the marker bit do so */
if (rtp->set_marker_bit) {
mark = 1;
rtp->set_marker_bit = 0;
}
/* If the timestamp for non-digit packets has moved beyond the timestamp
for digits, update the digit timestamp.
*/