update to reflect recent rtp changes

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index b296334..b3d7ea6 100644 (file)
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -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)