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<div class="moz-cite-prefix">Yes, this means the remote end was not
sending any audio stream.<br>
But it shouldn't.<br>
According to [1] before remote end send OK or ACK there is one way
SDP, no any audio stream.<br>
PJSIP channel (option rtp_timeout) does not take this one.<br>
<br>
Isn't it a mistake? What could be workarounds?<br>
<br>
19.12.2016 11:33, Jean Aunis пишет:<br>
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cite="mid:c6fdf25d-6c69-ac41-4628-53b011c56244@prescom.fr"
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<p>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.<br>
</p>
<br>
<div class="moz-cite-prefix">Le 16/12/2016 à 21:19, Dmitriy Serov
a écrit :<br>
</div>
<blockquote
cite="mid:78f84ea9-f0af-4d8f-23ac-78846f54aabf@gmail.com"
type="cite">Today I faced a problem. Please help to solve this
problem. <br>
<br>
Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT,
firmware v2.06(AAGJ.9)C1 <br>
<br>
Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN
(pjsip trunk). <br>
Call using early media (183 Session in progress) and
rtp_timeout=10. <br>
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 <br>
<br>
SIP dump is attached. <br>
<br>
According to [1] before called user agent send OK or ACK there
is one way SDP. <br>
In sip dump (attached) i didn't find such SIP packets. Whether
that call was canceled due to RTP inactivity? <br>
<br>
Any help is welcome. <br>
<br>
[1] <a moz-do-not-send="true" class="moz-txt-link-freetext"
href="https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt">https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt</a>
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