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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-04 18:08:42 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2008-03-04 18:08:42 +0000
commit151876ff4c5529acf6dcdc24e2273f3c02e1f299 (patch)
tree3c694cacd2589a20ba8082115a507e1c811319b5
parent56cdf42bca5f2e7dd685090e90adbb73fd4db637 (diff)
Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105675 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c7
-rw-r--r--include/asterisk/rtp.h2
-rw-r--r--main/rtp.c15
3 files changed, 23 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4ed3e3d75..f5cbda2b5 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4824,8 +4824,10 @@ static int sip_answer(struct ast_channel *ast)
if (p->t38.state == T38_PEER_DIRECT) {
change_t38_state(p, T38_ENABLED);
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, FALSE);
+ }
}
sip_pvt_unlock(p);
return res;
@@ -4858,6 +4860,7 @@ 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, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
@@ -5095,9 +5098,11 @@ 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 */
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 8d853903b..5a857f016 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -170,6 +170,8 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
+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 */
diff --git a/main/rtp.c b/main/rtp.c
index f5e22a953..5eed32207 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -175,6 +175,8 @@ struct ast_rtp {
enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
+
+ int set_marker_bit:1; /*!< Whether to set the marker bit or not */
};
/* Forward declarations */
@@ -2324,6 +2326,12 @@ int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc)
return ast_netsock_set_qos(rtp->s, tos, cos, desc);
}
+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;
@@ -3036,6 +3044,13 @@ 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.
*/