Merged revisions 48168 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines

Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48169 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 16 insertions, 6 deletions

diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index f64bc125b..e5ec59de6 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -163,7 +163,6 @@ struct gtalk_container {
 };
 
 static const char desc[] = "Gtalk Channel";
-static const char type[] = "Gtalk";
 
 static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
 
@@ -192,7 +191,7 @@ static int gtalk_get_codec(struct ast_channel *chan);
 
 /*! \brief PBX interface structure for channel registration */
 static const struct ast_channel_tech gtalk_tech = {
-	.type = type,
+	.type = "Gtalk",
 	.description = "Gtalk Channel Driver",
 	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
 	.requester = gtalk_request,
@@ -220,7 +219,7 @@ static struct in_addr __ourip;
 
 /*! \brief RTP driver interface */
 static struct ast_rtp_protocol gtalk_rtp = {
-	type: "gtalk",
+	type: "Gtalk",
 	get_rtp_info: gtalk_get_rtp_peer,
 	set_rtp_peer: gtalk_set_rtp_peer,
 	get_codec: gtalk_get_codec,
@@ -921,10 +920,12 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
 	fmt = ast_best_codec(tmp->nativeformats);
 
 	if (i->rtp) {
+		ast_rtp_setstun(i->rtp, 1);
 		tmp->fds[0] = ast_rtp_fd(i->rtp);
 		tmp->fds[1] = ast_rtcp_fd(i->rtp);
 	}
 	if (i->vrtp) {
+		ast_rtp_setstun(i->rtp, 1);
 		tmp->fds[2] = ast_rtp_fd(i->vrtp);
 		tmp->fds[3] = ast_rtcp_fd(i->vrtp);
 	}
@@ -1790,7 +1791,7 @@ static int load_module(void)
 
 	/* Make sure we can register our channel type */
 	if (ast_channel_register(&gtalk_tech)) {
-		ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
+		ast_log(LOG_ERROR, "Unable to register channel class %s\n", gtalk_tech.type);
 		return -1;
 	}
 	return 0;
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 03f204453..aab598a31 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -186,6 +186,9 @@ void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
 /*! \brief Compensate for devices that send RFC2833 packets all at once */
 void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
 
+/*! \brief Enable STUN capability */
+void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
+
 int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
 
 int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
diff --git a/main/rtp.c b/main/rtp.c
index 838105290..3d394a043 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -183,6 +183,7 @@ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader,
 #define FLAG_P2P_NEED_DTMF              (1 << 5)
 #define FLAG_CALLBACK_MODE              (1 << 6)
 #define FLAG_DTMF_COMPENSATE            (1 << 7)
+#define FLAG_HAS_STUN                   (1 << 8)
 
 /*!
  * \brief Structure defining an RTCP session.
@@ -545,6 +546,11 @@ void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate)
 	ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
 }
 
+void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable)
+{
+	ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
+}
+
 static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type)
 {
 	if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) ||
@@ -2913,8 +2919,8 @@ static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
 /*! \brief Helper function to switch a channel and RTP stream into callback mode */
 static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod)
 {
-	/* If we need DTMF or we have no IO structure, then we can't do direct callback */
-	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || !rtp->io)
+	/* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */
+	if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io)
 		return 0;
 
 	/* If the RTP structure is already in callback mode, remove it temporarily */