[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
serov.d.p at gmail.com
Mon Dec 19 04:36:56 CST 2016
Yes, this means the remote end was not sending any audio stream.
But it shouldn't.
According to  before remote end send OK or ACK there is one way SDP,
no any audio stream.
PJSIP channel (option rtp_timeout) does not take this one.
Isn't it a mistake? What could be workarounds?
19.12.2016 11:33, Jean Aunis пишет:
> This means the remote end was not sending any audio stream, or the
> audio stream was not received by Asterisk. The problem may have many
> different reasons, but often it is a network-related issue.
> Le 16/12/2016 à 21:19, Dmitriy Serov a écrit :
>> Today I faced a problem. Please help to solve this problem.
>> Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware
>> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip
>> Call using early media (183 Session in progress) and rtp_timeout=10.
>> After 10 seconds: [2016-12-16 13:53:15] NOTICE
>> res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for
>> lack of RTP activity in 10 seconds
>> SIP dump is attached.
>> According to  before called user agent send OK or ACK there is one
>> way SDP.
>> In sip dump (attached) i didn't find such SIP packets. Whether that
>> call was canceled due to RTP inactivity?
>> Any help is welcome.
>>  https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt
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