Dmitriy Serov
2016-Dec-19 10:36 UTC
[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
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 ?????:> > 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 ? 21:19, Dmitriy Serov a ?crit : >> 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 >> >> >> > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161219/0ccd2720/attachment.html>
Joshua Colp
2017-Jan-03 17:58 UTC
[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote:> 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?Looking at the code we don't take that scenario into account it seems. Please file an issue[1] and we'll see if we can do something about it. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
Dmitriy Serov
2017-Jan-03 18:24 UTC
[asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity
Joshua, issue has been filed. Thank you! https://issues.asterisk.org/jira/browse/ASTERISK-26689 03.01.2017 20:58, Joshua Colp ?????:> On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: >> 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? > Looking at the code we don't take that scenario into account it seems. > Please file an issue[1] and we'll see if we can do something about it. > > [1] https://issues.asterisk.org/jira >