# Media Inactivity detection **URL:** \ **Category:** Asterisk SIP\ **Created:** [December 28, 2018, 12:35pm UTC](https://community.asterisk.org/t/media-inactivity-detection/77811 "2018-12-28T12:35:54Z")\ **Posts on this page:** 3\ **Page:** 1
**Author:** ![krishnapriya](https://avatars.discourse-cdn.com/v4/letter/k/51bf81/32.png) [@krishnapriya](https://community.asterisk.org/u/krishnapriya)\ **Post date:** [December 28, 2018, 12:35pm UTC](https://community.asterisk.org/t/media-inactivity-detection/77811/1 "2018-12-28T12:35:55Z")
If a call is established with registered user in Asterisk, and there is no RTP/RTCP received by Asterisk from far end user, how Asterisk is going to handle the call? The observation I had is, Asterisk disconnects the call with Asterisk hangup code 44 exactly at 30th second. Could you guys please confirm above observation. ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [December 28, 2018, 12:59pm UTC](https://community.asterisk.org/t/media-inactivity-detection/77811/2 "2018-12-28T12:59:14Z")
Obviously not true, otherwise IAX and circuit switched calls would all fail at 30 seconds! (Neither of these technologies uses RTP, so can never receive RTP or RTCP.) For chan\_sip it is configurable with rtptimeout, and rtpholdtimeout, in the channel technology configuration file. The names are different for chan\_pjsip. Both these technologies seem not to apply a timeout by default. The cause code may vary with technology. ---
**Author:** ![krishnapriya](https://avatars.discourse-cdn.com/v4/letter/k/51bf81/32.png) [@krishnapriya](https://community.asterisk.org/u/krishnapriya)\ **Post date:** [December 28, 2018, 2:00pm UTC](https://community.asterisk.org/t/media-inactivity-detection/77811/3 "2018-12-28T14:00:39Z")
Thank you David… as you have suggested, it can be configured using **rtptimeout** value in chan\_sip channel.