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authorJoshua Colp <jcolp@digium.com>2015-07-18 13:16:10 -0300
committerJoshua Colp <jcolp@digium.com>2015-07-24 12:43:02 -0300
commit27497217912a82ca243a9d5e9acfbbb597faf323 (patch)
treea461e85b877c4879eadae0f84024fb26735c630b /include/asterisk/res_pjsip_session.h
parent1997b6f677f314f685ece102af9a70ed17fe63b5 (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_session.h')
-rw-r--r--include/asterisk/res_pjsip_session.h2
1 files changed, 2 insertions, 0 deletions
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 5489979ed..2893f66ab 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -79,6 +79,8 @@ struct ast_sip_session_media {
pj_str_t transport;
/*! \brief Scheduler ID for RTP keepalive */
int keepalive_sched_id;
+ /*! \brief Scheduler ID for RTP timeout */
+ int timeout_sched_id;
/*! \brief Stream is on hold */
unsigned int held:1;
/*! \brief Stream type this session media handles */