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authorjpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-01 23:01:31 +0000
committerjpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-01 23:01:31 +0000
commitc7ce3d71d5256922347d6d2d7c238c4719a65fa6 (patch)
tree204b6b026496bb5a24cabaa5c2160b3d5080594c /include
parent045e4bf061183e40d4e39eb59f05913a726754c8 (diff)
Merged revisions 289797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289798 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp.h1
1 files changed, 1 insertions, 0 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index aa38ee0fa..acd40d3ec 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -211,6 +211,7 @@ int ast_rtcp_fd(struct ast_rtp *rtp);
int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
+int ast_rtp_senddigit_end_with_duration(struct ast_rtp *rtp, char digit, unsigned int duration);
int ast_rtp_sendcng(struct ast_rtp *rtp, int level);