Code simplification

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index dea5d50..e453e3e 100644 (file)
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -956,7 +956,7 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
        struct jingle_pvt *p, *tmp = client->p;
        struct ast_channel *chan;
        int res;
-       iks *payload_type;
+       iks *codec;
 
        /* Make sure our new call doesn't exist yet */
        while (tmp) {
@@ -974,45 +974,47 @@ static int jingle_newcall(struct jingle *client, ikspak *pak)
                return -1;
        }
        chan = jingle_new(client, p, AST_STATE_DOWN, pak->from->user);
-       if (chan) {
-               ast_mutex_lock(&p->lock);
-               ast_copy_string(p->them, pak->from->full, sizeof(p->them));
-               if (iks_find_attrib(pak->query, JINGLE_SID)) {
-                       ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
-                                                       sizeof(p->sid));
-               }
-
-               payload_type = iks_child(iks_child(iks_child(iks_child(pak->x))));
-               while (payload_type) {
-                       ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")));
-                       ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(payload_type, "id")), "audio", iks_find_attrib(payload_type, "name"), 0);
-                       payload_type = iks_next(payload_type);
-               }
-
-               ast_mutex_unlock(&p->lock);
-               ast_setstate(chan, AST_STATE_RING);
-               res = ast_pbx_start(chan);
-
-               switch (res) {
-               case AST_PBX_FAILED:
-                       ast_log(LOG_WARNING, "Failed to start PBX :(\n");
-                       jingle_response(client, pak, "service-unavailable", NULL);
-                       break;
-               case AST_PBX_CALL_LIMIT:
-                       ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
-                       jingle_response(client, pak, "service-unavailable", NULL);
-                       break;
-               case AST_PBX_SUCCESS:
-                       jingle_response(client, pak, NULL, NULL);
-                       jingle_create_candidates(client, p,
-                                                iks_find_attrib(pak->query, JINGLE_SID),
-                                                iks_find_attrib(pak->x, "from"));
-                       /* nothing to do */
-                       break;
-               }
-       } else {
+       if (!chan) {
                jingle_free_pvt(client, p);
+               return -1;
        }
+       ast_mutex_lock(&p->lock);
+       ast_copy_string(p->them, pak->from->full, sizeof(p->them));
+       if (iks_find_attrib(pak->query, JINGLE_SID)) {
+               ast_copy_string(p->sid, iks_find_attrib(pak->query, JINGLE_SID),
+                               sizeof(p->sid));
+       }
+       
+       /* codec points to the first <payload-type/> tag */     
+       codec = iks_child(iks_child(iks_child(iks_child(pak->x))));
+       while (codec) {
+               ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
+               ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+               codec = iks_next(codec);
+       }
+       
+       ast_mutex_unlock(&p->lock);
+       ast_setstate(chan, AST_STATE_RING);
+       res = ast_pbx_start(chan);
+       
+       switch (res) {
+       case AST_PBX_FAILED:
+               ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+               jingle_response(client, pak, "service-unavailable", NULL);
+               break;
+       case AST_PBX_CALL_LIMIT:
+               ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+               jingle_response(client, pak, "service-unavailable", NULL);
+               break;
+       case AST_PBX_SUCCESS:
+               jingle_response(client, pak, NULL, NULL);
+               jingle_create_candidates(client, p,
+                                        iks_find_attrib(pak->query, JINGLE_SID),
+                                        iks_find_attrib(pak->x, "from"));
+               /* nothing to do */
+               break;
+       }
+
        return 1;
 }