[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

[Dmitriy Serov]

Yes, this means the remote end was not sending any audio stream.
But it shouldn't.
According to [1] 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 
>> v2.06(AAGJ.9)C1
>>
>> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip 
>> trunk).
>> Call using early media (183 Session in progress) and rtp_timeout=10.
>> After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] 
>> res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for 
>> lack of RTP activity in 10 seconds
>>
>> SIP dump is attached.
>>
>> According to [1] 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.
>>
>> [1] https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt
>>
>>
>>
>
>
>

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