Don't do reinvite if both parties talk diffrent codecs

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1752 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 8 insertions, 0 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index de0712aeb..71cc57240 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6287,10 +6287,17 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
 	return 0;
 }
 
+static int sip_get_codec(struct ast_channel *chan)
+{
+	struct sip_pvt *p = chan->pvt->pvt;
+	return p->capability;	
+}
+
 static struct ast_rtp_protocol sip_rtp = {
 	get_rtp_info: sip_get_rtp_peer,
 	get_vrtp_info: sip_get_vrtp_peer,
 	set_rtp_peer: sip_set_rtp_peer,
+	get_codec: sip_get_codec,
 };
 
 int load_module()
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index cc8d24da7..385b24a6a 100755
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -39,6 +39,7 @@ struct ast_rtp_protocol {
 	struct ast_rtp *(*get_rtp_info)(struct ast_channel *chan);				/* Get RTP struct, or NULL if unwilling to transfer */
 	struct ast_rtp *(*get_vrtp_info)(struct ast_channel *chan);				/* Get RTP struct, or NULL if unwilling to transfer */
 	int (*set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer);	/* Set RTP peer */
+	int (*get_codec)(struct ast_channel *chan);
 	char *type;
 	struct ast_rtp_protocol *next;
 };