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.
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.
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Thank you David… as you have suggested, it can be configured using rtptimeout value in chan_sip channel.