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<p class="MsoNormal"><span style="mso-fareast-language:EN-US">Hi Shaun,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">If someone has a solution, that would be really fine.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">Currently I have helped myself by re-enabling the following in the code:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">diff --git a/src/switch_rtp.c b/src/switch_rtp.c<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  index 40d8978..aada64a 100644<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  --- a/src/switch_rtp.c<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  +++ b/src/switch_rtp.c<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">                  if (elapsed > 30000) {<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">                          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s                          rtp_session->last_stun = switch_micro_time_now();<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  -                       //status = SWITCH_STATUS_GENERR;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  -                       //goto end;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  +                       status = SWITCH_STATUS_GENERR;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  +                       goto end;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">                  }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">          }<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">          ice->sending = 3;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  -       // end:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  +       end:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">          READ_DEC(rtp_session);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">          return status;<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">  (END)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">The code has the consequence that the session is cleared if no more media comes for 30 seconds.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">With kind regards<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US">Alex<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b>Von:</b> FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org>
<b>Im Auftrag von </b>Shaun Stokes<br>
<b>Gesendet:</b> Donnerstag, 10. Februar 2022 14:39<br>
<b>An:</b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br>
<b>Betreff:</b> [Freeswitch-users] Media timeout<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black">Hi All,<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black">I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected.<o:p></o:p></span></p>
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<li class="MsoNormal" style="color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
<span style="font-size:12.0pt">media_timeout on a call that supports video but with-out video will fail.<o:p></o:p></span></li><li class="MsoNormal" style="color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
<span style="font-size:12.0pt">media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops.<o:p></o:p></span></li><li class="MsoNormal" style="color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
<span style="font-size:12.0pt">media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout.<o:p></o:p></span></li></ul>
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<p class="MsoNormal"><span style="font-size:12.0pt;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:12.0pt;color:black"><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black;background:white">Thanks,</span><span style="font-size:12.0pt;color:black"><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;color:black;background:white">Shaun</span><span style="font-size:12.0pt;color:black"><o:p></o:p></span></p>
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