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authorMartin Pycko <martinp@digium.com>2003-11-15 00:52:49 +0000
committerMartin Pycko <martinp@digium.com>2003-11-15 00:52:49 +0000
commite3b0a66805b6f127468421b2992d105586342f60 (patch)
tree716e601171e9673cc9ce68df4d2c14576fcdf21c /include/asterisk/rtp.h
parent6f9da41a6ce99e94e6bf0ed3d1e2314b1ec1d0fa (diff)
Don't do reinvite if both parties talk diffrent codecs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk/rtp.h')
-rwxr-xr-xinclude/asterisk/rtp.h1
1 files changed, 1 insertions, 0 deletions
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;
};