[asterisk-commits] dvossel: trunk r290649 - in /trunk: ./ channels/chan_gtalk.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Wed Oct  6 16:09:14 2010
New Revision: 290649

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=290649
Log:
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.
........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_gtalk.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/channels/chan_gtalk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=290649&r1=290648&r2=290649
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Wed Oct  6 16:09:14 2010
@@ -173,7 +173,7 @@
 
 /* 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 @@
 	.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 @@
 
 	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 @@
 	  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_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 @@
 	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 @@
 	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");