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handle on-/off-hold properly with RTP reinvited streams (issue #3974) 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6371 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
kpfleming 2005-08-23 01:50:11 +00:00
parent aa1611ff9d
commit bf469c0a96
1 changed files with 30 additions and 37 deletions
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67
channels/chan_sip.c
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@ -2576,24 +2576,15 @@ static int sip_indicate(struct ast_channel *ast, int condition)
}
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
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_subscriptions(int fd, int argc, char *argv[])
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 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions
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 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions
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 @@ static int sip_show_channel(int fd, int argc, char *argv[])
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 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
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)
return -1;
p = chan->tech_pvt;
if (!p)
return -1;
ast_mutex_lock(&p->lock);
if (rtp)
ast_rtp_get_peer(rtp, &p->redirip);
@ -11498,10 +11484,17 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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);
}
}