[Asterisk-cvs] asterisk/channels chan_sip.c,1.811,1.812
kpfleming
kpfleming
Mon Aug 22 21:48:07 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv11223/channels
Modified Files:
chan_sip.c
Log Message:
handle on-/off-hold properly with RTP reinvited streams (issue #3974)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.811
retrieving revision 1.812
diff -u -d -r1.811 -r1.812
--- chan_sip.c 23 Aug 2005 01:44:28 -0000 1.811
+++ chan_sip.c 23 Aug 2005 01:50:11 -0000 1.812
@@ -2576,24 +2576,15 @@
}
res = -1;
break;
- case AST_CONTROL_HOLD: /* We are put on hold */
- /* The PBX is providing us with onhold music, but
- should we clear the RTP stream with the other
- end? Guess we could do that if there's no
- musiconhold class defined for this channel
- */
+ case AST_CONTROL_HOLD: /* The other part of the bridge are put on hold */
if (sipdebug)
- ast_log(LOG_DEBUG, "SIP dialog on hold: %s\n", p->callid);
+ ast_log(LOG_DEBUG, "Bridged channel now on hold%s\n", p->callid);
res = -1;
- ast_set_flag(p, SIP_CALL_ONHOLD);
break;
- case AST_CONTROL_UNHOLD: /* We are back from hold */
- /* Open RTP stream if we decide to close it
- */
+ case AST_CONTROL_UNHOLD: /* The other part of the bridge are back from hold */
if (sipdebug)
- ast_log(LOG_DEBUG, "SIP dialog off hold: %s\n", p->callid);
+ ast_log(LOG_DEBUG, "Bridged channel is back from hold, let's talk! : %s\n", p->callid);
res = -1;
- ast_clear_flag(p, SIP_CALL_ONHOLD);
break;
case -1:
res = -1;
@@ -3447,6 +3438,7 @@
if ((bridgepeer=ast_bridged_channel(p->owner))) {
/* We have a bridge */
/* Turn on/off music on hold if we are holding/unholding */
+ struct ast_frame af = { AST_FRAME_NULL, };
if (sin.sin_addr.s_addr && !sendonly) {
ast_moh_stop(bridgepeer);
/* Indicate UNHOLD status to the other channel */
@@ -3460,19 +3452,8 @@
p->owner->uniqueid);
}
ast_clear_flag(p, SIP_CALL_ONHOLD);
- /* Somehow, we need to check if we need to re-invite here */
- /* If this call had a external native bridge, it's broken
- now and we need to start all over again.
- The bridged peer, if SIP, now listens
- to RTP from Asterisk instead of from
- the peer
-
- So IF we had a native bridge before
- the HOLD, we need to somehow re-invite
- into a NATIVE bridge afterwards...
-
- */
-
+ /* Activate a re-invite */
+ ast_queue_frame(p->owner, &af);
} else {
/* No address for RTP, we're on hold */
append_history(p, "Hold", req->data);
@@ -3489,6 +3470,8 @@
ast_moh_start(bridgepeer, NULL);
if (sendonly)
ast_rtp_stop(p->rtp);
+ /* Activate a re-invite */
+ ast_queue_frame(p->owner, &af);
}
}
return 0;
@@ -4319,6 +4302,8 @@
reqprep(&req, p, SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
+ if (sipdebug)
+ add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
ast_rtp_offered_from_local(p->rtp, 1);
add_sdp(&req, p);
/* Use this as the basis */
@@ -7527,8 +7512,8 @@
static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions)
{
#define FORMAT3 "%-15.15s %-10.10s %-21.21s %-15.15s\n"
-#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %s %s\n"
-#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-6.6s%s %s\n"
+#define FORMAT2 "%-15.15s %-10.10s %-11.11s %-11.11s %-4.4s %-7.7s %s \n"
+#define FORMAT "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-4.4s %-7.7s%s %s\n"
struct sip_pvt *cur;
char iabuf[INET_ADDRSTRLEN];
int numchans = 0;
@@ -7537,7 +7522,7 @@
ast_mutex_lock(&iflock);
cur = iflist;
if (!subscriptions)
- ast_cli(fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Seq (Tx/Rx)", "Format", "Last Msg");
+ ast_cli(fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Seq (Tx/Rx)", "Format", "Hold", "Last Msg");
else
ast_cli(fd, FORMAT3, "Peer", "User", "Call ID", "URI");
while (cur) {
@@ -7547,6 +7532,7 @@
cur->callid,
cur->ocseq, cur->icseq,
ast_getformatname(cur->owner ? cur->owner->nativeformats : 0),
+ ast_test_flag(cur, SIP_CALL_ONHOLD) ? "Yes" : "No",
ast_test_flag(cur, SIP_NEEDDESTROY) ? "(d)" : "",
cur->lastmsg );
numchans++;
@@ -7739,6 +7725,7 @@
ast_cli(fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), ntohs(cur->sa.sin_port));
ast_cli(fd, " Received Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->recv.sin_addr), ntohs(cur->recv.sin_port));
ast_cli(fd, " NAT Support: %s\n", nat2str(ast_test_flag(cur, SIP_NAT)));
+ ast_cli(fd, " Audio IP: %s %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" );
ast_cli(fd, " Our Tag: %08d\n", cur->tag);
ast_cli(fd, " Their Tag: %s\n", cur->theirtag);
ast_cli(fd, " SIP User agent: %s\n", cur->useragent);
@@ -11478,15 +11465,14 @@
return rtp;
}
-/*--- sip_set_rtp_peer: Set the data needed to RE-INVITE this call
- so that the peers media go between them, outside of Asterisk. ---*/
+/*--- sip_set_rtp_peer: Set the RTP peer for this call ---*/
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
{
struct sip_pvt *p;
+
p = chan->tech_pvt;
- if (!p)
+ if (!p)
return -1;
-
ast_mutex_lock(&p->lock);
if (rtp)
ast_rtp_get_peer(rtp, &p->redirip);
@@ -11498,10 +11484,17 @@
memset(&p->vredirip, 0, sizeof(p->vredirip));
p->redircodecs = codecs;
if (!ast_test_flag(p, SIP_GOTREFER)) {
- if (!p->pendinginvite)
+ if (!p->pendinginvite) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
+ }
transmit_reinvite_with_sdp(p);
- else if (!ast_test_flag(p, SIP_PENDINGBYE)) {
- ast_log(LOG_DEBUG, "Deferring reinvite on '%s'\n", p->callid);
+ } else if (!ast_test_flag(p, SIP_PENDINGBYE)) {
+ if (option_debug > 2) {
+ char iabuf[INET_ADDRSTRLEN];
+ ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
+ }
ast_set_flag(p, SIP_NEEDREINVITE);
}
}
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