Hi Shaun,

I have noticed the same thing. "media_timout" in conjunction with WebRTC does not work at all.

If someone has a solution, that would be really fine.

Currently I have helped myself by re-enabling the following in the code:

diff --git a/src/switch_rtp.c b/src/switch_rtp.c

  index 40d8978..aada64a 100644

  --- a/src/switch_rtp.c

  +++ b/src/switch_rtp.c

  @@ -854,8 +854,8 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)

                  if (elapsed > 30000) {

                          switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING, "No %s                          rtp_session->last_stun = switch_micro_time_now();

  -                       //status = SWITCH_STATUS_GENERR;

  -                       //goto end;

  +                       status = SWITCH_STATUS_GENERR;

  +                       goto end;

                  }

          }

  @@ -897,7 +897,7 @@ static switch_status_t ice_out(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice)

          ice->sending = 3;

  -       // end:

  +       end:

          READ_DEC(rtp_session);

          return status;

  (END)

The code has the consequence that the session is cleared if no more media comes for 30 seconds.

With kind regards

Alex

Von: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> Im Auftrag von Shaun Stokes
Gesendet: Donnerstag, 10. Februar 2022 14:39
An: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Betreff: [Freeswitch-users] Media timeout

Hi All,

I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected.

• media_timeout on a call that supports video but with-out video will fail.
• media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops.
• media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout.

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.

Thanks,

Shaun