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Don't destory rtp until destroy, use rtp_stop instead

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-03-20 22:01:53 +00:00
parent f02b64d258
commit 83016e1bce
3 changed files with 11 additions and 6 deletions
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9
channels/chan_sip.c
View File

@@ -3117,7 +3117,6 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
{
char *to;
char *msg, *c;
struct ast_rtp *rtp;
struct ast_channel *owner;
struct sip_peer *peer;
int pingtime;
@@ -3267,9 +3266,8 @@ retrylock:
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, inet_ntoa(p->sa.sin_addr));
p->alreadygone = 1;
if (p->rtp) {
struct sockaddr_in sin = { AF_INET, };
/* Immediately stop RTP by setting transmit to 0 */
ast_rtp_setpeer(p->rtp, &sin);
/* Immediately stop RTP */
ast_rtp_stop(p->rtp);
}
/* XXX Locking issues?? XXX */
switch(resp) {
@@ -3608,8 +3606,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
p->alreadygone = 1;
if (p->rtp) {
/* Immediately stop RTP */
ast_rtp_destroy(p->rtp);
p->rtp = NULL;
ast_rtp_stop(p->rtp);
}
if (p->owner)
ast_queue_hangup(p->owner, 1);

2
include/asterisk/rtp.h
View File

@@ -95,6 +95,8 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
void ast_rtp_stop(struct ast_rtp *rtp);
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif

6
rtp.c
View File

@@ -586,6 +586,12 @@ void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
memcpy(us, &rtp->us, sizeof(rtp->us));
}
void ast_rtp_stop(struct ast_rtp *rtp)
{
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
}
void ast_rtp_destroy(struct ast_rtp *rtp)
{
if (rtp->smoother)