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update to deal with new rtp.c changes 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1146 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
jeremy 2003-07-01 19:11:37 +00:00
parent df8907e2c7
commit 4e9d9c39cb
1 changed files with 12 additions and 2 deletions
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14
channels/h323/chan_h323.c
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@ -684,7 +684,8 @@ static struct oh323_pvt *oh323_alloc(int callid)
/* Keep track of stuff */
memset(p, 0, sizeof(struct oh323_pvt));
p->rtp = ast_rtp_new(NULL, NULL);
p->rtp = ast_rtp_new(sched, io, 1, 0);
if (!p->rtp) {
ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
free(p);
@ -1569,8 +1570,15 @@ static struct ast_rtp *oh323_get_rtp_peer(struct ast_channel *chan)
return NULL;
}
static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp)
static struct ast_rtp *oh323_get_vrtp_peer(struct ast_channel *chan)
{
return NULL;
}
static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp)
{
/* XXX Deal with Video */
struct oh323_pvt *p;
struct sockaddr_in them;
struct sockaddr_in us;
@ -1598,9 +1606,11 @@ static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp)
static struct ast_rtp_protocol oh323_rtp = {
get_rtp_info: oh323_get_rtp_peer,
get_vrtp_info: oh323_get_vrtp_peer,
set_rtp_peer: oh323_set_rtp_peer,
};
int load_module()
{
int res;