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update to reflect recent rtp changes 

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41272 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
russell 2006-08-29 13:55:54 +00:00
parent 1a0e185890
commit fb567961c7
1 changed files with 11 additions and 7 deletions
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18
channels/chan_jingle.c
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@ -185,7 +185,7 @@ static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from,
/*----- RTP interface functions */
static int jingle_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
struct ast_rtp *vrtp, int codecs, int nat_active);
static struct ast_rtp *jingle_get_rtp_peer(struct ast_channel *chan);
static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static int jingle_get_codec(struct ast_channel *chan);
/*! \brief PBX interface structure for channel registration */
@ -405,18 +405,22 @@ static int jingle_answer(struct ast_channel *ast)
return res;
}
static struct ast_rtp *jingle_get_rtp_peer(struct ast_channel *chan)
static enum ast_rtp_get_result jingle_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
{
struct jingle_pvt *p = chan->tech_pvt;
struct ast_rtp *rtp = NULL;
enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
if (!p)
return NULL;
return res;
ast_mutex_lock(&p->lock);
if (p->rtp)
rtp = p->rtp;
if (p->rtp) {
*rtp = p->rtp;
res = AST_RTP_TRY_NATIVE;
}
ast_mutex_unlock(&p->lock);
return rtp;
return res;
}
static int jingle_get_codec(struct ast_channel *chan)