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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-12-26 04:31:58 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-12-26 04:31:58 +0000
commitcb366eccb630d34261b3e2d455470c30d24fa9c6 (patch)
tree0740b6f5add270b039bb3ab1f862ff5f8b5cf7e6 /include
parent9b1d7a3c1c3bbe9e8b2369121c644ee713507faf (diff)
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48964 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h2
1 files changed, 2 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index c0b3d4cfb..f6f26e942 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -209,6 +209,8 @@ char *ast_rtp_get_quality(struct ast_rtp *rtp);
/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
int ast_rtcp_send_h261fur(void *data);
+void ast_rtp_new_init(struct ast_rtp *rtp);
+
void ast_rtp_init(void);
int ast_rtp_reload(void);