Merged revisions 290648 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines

  Fixes gtalk outbound DTMF to work properly.

  Outbound DTMF with gtalk needs to be done within the RTP stream.  I discovered
  this after investigating a packet capture from the gmail client.  Instead of
  performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive
  on the RTP stream using RFC2833 way of doing things.  Chan_gtalk also had an issue
  with negotiating RTP payload type 106 for the telephony-event and then sending
  DTMF as payload 101.  This has been resolved by always negotiating 101 as the payload
  type like we do everywhere else.  With this patch, incoming google voice calls forwarded
  to Asterisk via gtalk work.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index 0647d3e6c49c371306f6f17d04b8a7edaa08f0b4..00f873ab4ade724579b829cbaf3b46e5508beeb1 100644 (file)
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -173,7 +173,7 @@ AST_MUTEX_DEFINE_STATIC(gtalklock); /*!< Protect the interface list (of gtalk_pv
 
 /* Forward declarations */
 static struct ast_channel *gtalk_request(const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause);
-static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);
+/*static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);*/
 static int gtalk_sendtext(struct ast_channel *ast, const char *text);
 static int gtalk_digit_begin(struct ast_channel *ast, char digit);
 static int gtalk_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
@@ -201,7 +201,10 @@ static const struct ast_channel_tech gtalk_tech = {
        .send_text = gtalk_sendtext,
        .send_digit_begin = gtalk_digit_begin,
        .send_digit_end = gtalk_digit_end,
-       .bridge = ast_rtp_instance_bridge,
+       /* XXX TODO native bridging is causing odd problems with DTMF pass-through with
+        * the gtalk servers. Enable native bridging once the source of this problem has
+        * been identified.
+       .bridge = ast_rtp_instance_bridge, */
        .call = gtalk_call,
        .hangup = gtalk_hangup,
        .answer = gtalk_answer,
@@ -412,7 +415,7 @@ static int gtalk_invite(struct gtalk_pvt *p, char *to, char *from, char *sid, in
 
        if (codecs_num) {
                /* only propose DTMF within an audio session */
-               iks_insert_attrib(payload_telephone, "id", "106");
+               iks_insert_attrib(payload_telephone, "id", "101");
                iks_insert_attrib(payload_telephone, "name", "telephone-event");
                iks_insert_attrib(payload_telephone, "clockrate", "8000");
        }
@@ -997,6 +1000,8 @@ static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const
          return NULL;
        }
        ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
+       ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_DTMF, 1);
+       ast_rtp_instance_dtmf_mode_set(tmp->rtp, AST_RTP_DTMF_MODE_RFC2833);
        ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
 
        /* add user configured codec capabilites */
@@ -1590,14 +1595,39 @@ static int gtalk_sendtext(struct ast_channel *chan, const char *text)
 
 static int gtalk_digit_begin(struct ast_channel *chan, char digit)
 {
-       return gtalk_digit(chan, digit, 0);
+       struct gtalk_pvt *p = chan->tech_pvt;
+       int res = 0;
+
+       ast_mutex_lock(&p->lock);
+       if (p->rtp) {
+               ast_rtp_instance_dtmf_begin(p->rtp, digit);
+       } else {
+               res = -1;
+       }
+       ast_mutex_unlock(&p->lock);
+
+       return res;
 }
 
 static int gtalk_digit_end(struct ast_channel *chan, char digit, unsigned int duration)
 {
-       return gtalk_digit(chan, digit, duration);
+       struct gtalk_pvt *p = chan->tech_pvt;
+       int res = 0;
+
+       ast_mutex_lock(&p->lock);
+       if (p->rtp) {
+               ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration);
+       } else {
+               res = -1;
+       }
+       ast_mutex_unlock(&p->lock);
+
+       return res;
 }
 
+/* This function is of not in use at the moment, but I am choosing to leave this
+ * within the code base as a reference to how DTMF is possible through
+ * jingle signaling.  However, google currently does DTMF through the RTP. 
 static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration)
 {
        struct gtalk_pvt *p = ast->tech_pvt;
@@ -1623,8 +1653,8 @@ static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duratio
        ast_aji_increment_mid(client->connection->mid);
        iks_insert_attrib(gtalk, "xmlns", "http://jabber.org/protocol/gtalk");
        iks_insert_attrib(gtalk, "action", "session-info");
-       /* put the initiator attribute to lower case if we receive the call
-        * otherwise GoogleTalk won't establish the session */
+       // put the initiator attribute to lower case if we receive the call
+       // otherwise GoogleTalk won't establish the session
        if (!p->initiator) {
                char c;
                char *t = lowerthem = ast_strdupa(p->them);
@@ -1651,7 +1681,7 @@ static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duratio
        ast_mutex_unlock(&p->lock);
        return 0;
 }
-
+*/
 static int gtalk_sendhtml(struct ast_channel *ast, int subclass, const char *data, int datalen)
 {
        ast_log(LOG_NOTICE, "XXX Implement gtalk sendhtml XXX\n");