<div dir="ltr">please open a github issue.<div><a href="https://github.com/signalwire/freeswitch/issues">https://github.com/signalwire/freeswitch/issues</a><br></div><div><br></div><div>you could also check if switch_rtp_set_max_missed_packets() and/or switch_rtp_set_media_timeout() are called in your particular call setup, with your configs.</div><div> </div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg <<a href="mailto:Alexander.Haugg@c4b.de">Alexander.Haugg@c4b.de</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">





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<p class="MsoNormal"><span>Hi Shaun,<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all.<u></u><u></u></span></p>
<p class="MsoNormal"><span>If someone has a solution, that would be really fine.<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>Currently I have helped myself by re-enabling the following in the code:<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>diff --git a/src/switch_rtp.c b/src/switch_rtp.c<u></u><u></u></span></p>
<p class="MsoNormal"><span>  index 40d8978..aada64a 100644<u></u><u></u></span></p>
<p class="MsoNormal"><span>  --- a/src/switch_rtp.c<u></u><u></u></span></p>
<p class="MsoNormal"><span>  +++ b/src/switch_rtp.c<u></u><u></u></span></p>
<p class="MsoNormal"><span>  @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)<u></u><u></u></span></p>
<p class="MsoNormal"><span>                  if (elapsed > 30000) {<u></u><u></u></span></p>
<p class="MsoNormal"><span>                          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s                          rtp_session->last_stun = switch_micro_time_now();<u></u><u></u></span></p>
<p class="MsoNormal"><span>  -                       //status = SWITCH_STATUS_GENERR;<u></u><u></u></span></p>
<p class="MsoNormal"><span>  -                       //goto end;<u></u><u></u></span></p>
<p class="MsoNormal"><span>  +                       status = SWITCH_STATUS_GENERR;<u></u><u></u></span></p>
<p class="MsoNormal"><span>  +                       goto end;<u></u><u></u></span></p>
<p class="MsoNormal"><span>                  }<u></u><u></u></span></p>
<p class="MsoNormal"><span>          }<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>  @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>          ice->sending = 3;<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>  -       // end:<u></u><u></u></span></p>
<p class="MsoNormal"><span>  +       end:<u></u><u></u></span></p>
<p class="MsoNormal"><span>          READ_DEC(rtp_session);<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>          return status;<u></u><u></u></span></p>
<p class="MsoNormal"><span>  (END)<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>The code has the consequence that the session is cleared if no more media comes for 30 seconds.<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
<p class="MsoNormal"><span>With kind regards<u></u><u></u></span></p>
<p class="MsoNormal"><span>Alex<u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
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<p class="MsoNormal"><b>Von:</b> FreeSWITCH-users <<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>>
<b>Im Auftrag von </b>Shaun Stokes<br>
<b>Gesendet:</b> Donnerstag, 10. Februar 2022 14:39<br>
<b>An:</b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
<b>Betreff:</b> [Freeswitch-users] Media timeout<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal"><span style="font-size:12pt;color:black">Hi All,<u></u><u></u></span></p>
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<div>
<p class="MsoNormal"><span style="font-size:12pt;color:black"><u></u> <u></u></span></p>
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<p class="MsoNormal"><span style="font-size:12pt;color:black">I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected.<u></u><u></u></span></p>
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<ul type="disc">
<li class="MsoNormal" style="color:black">
<span style="font-size:12pt">media_timeout on a call that supports video but with-out video will fail.<u></u><u></u></span></li><li class="MsoNormal" style="color:black">
<span style="font-size:12pt">media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops.<u></u><u></u></span></li><li class="MsoNormal" style="color:black">
<span style="font-size:12pt">media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout.<u></u><u></u></span></li></ul>
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<p class="MsoNormal"><span style="font-size:12pt;color:black;background:white">Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much simpler to apply this per SIP profile than it is per call.</span><span style="font-size:12pt;color:black"><u></u><u></u></span></p>
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<p class="MsoNormal"><span style="font-size:12pt;color:black"><u></u> <u></u></span></p>
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<p class="MsoNormal"><span style="font-size:12pt;color:black;background:white">Thanks,</span><span style="font-size:12pt;color:black"><u></u><u></u></span></p>
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<p class="MsoNormal"><span style="font-size:12pt;color:black;background:white">Shaun</span><span style="font-size:12pt;color:black"><u></u><u></u></span></p>
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