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authorJoshua Colp <jcolp@digium.com>2015-07-18 13:16:10 -0300
committerJoshua Colp <jcolp@digium.com>2015-07-24 12:43:43 -0300
commit309dd2a4090ccdd1ea31d8d5415a645daddd3883 (patch)
tree625855c3191c70217c6087593767baf3e0a0cb6d /include/asterisk/res_pjsip.h
parenta105461f9eadbacb684fe3751cddae2a7f400dea (diff)
pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Diffstat (limited to 'include/asterisk/res_pjsip.h')
-rw-r--r--include/asterisk/res_pjsip.h4
1 files changed, 4 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 432a1689e..decb4177a 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -504,6 +504,10 @@ struct ast_sip_media_rtp_configuration {
unsigned int encryption_optimistic;
/*! Number of seconds between RTP keepalive packets */
unsigned int keepalive;
+ /*! Number of seconds before terminating channel due to lack of RTP (when not on hold) */
+ unsigned int timeout;
+ /*! Number of seconds before terminating channel due to lack of RTP (when on hold) */
+ unsigned int timeout_hold;
};
/*!