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Fix call timeouts with rtp bridge etc (bug #5252)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-10-13 18:27:54 +00:00
parent 6fea13619e
commit dd11afca42
7 changed files with 56 additions and 77 deletions
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6
rtp.c
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@@ -1499,7 +1499,7 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
of Asterisk. */
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
struct ast_frame *f;
struct ast_channel *who, *cs[3];
@@ -1513,7 +1513,6 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
char iabuf[INET_ADDRSTRLEN];
void *pvt0, *pvt1;
int to;
int codec0,codec1, oldcodec0, oldcodec1;
memset(&vt0, 0, sizeof(vt0));
@@ -1635,7 +1634,6 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
}
return AST_BRIDGE_RETRY;
}
to = -1;
/* Now check if they have changed address */
ast_rtp_get_peer(p1, &t1);
ast_rtp_get_peer(p0, &t0);
@@ -1677,7 +1675,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
memcpy(&vac0, &vt0, sizeof(vac0));
oldcodec0 = codec0;
}
who = ast_waitfor_n(cs, 2, &to);
who = ast_waitfor_n(cs, 2, &timeoutms);
if (!who) {
if (option_debug)
ast_log(LOG_DEBUG, "Ooh, empty read...\n");