please open a github issue. https://github.com/signalwire/freeswitch/issues 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. On Tue, Feb 15, 2022 at 11:33 PM Alexander Haugg <Alexander.Haugg@c4b.de> wrote:
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 _________________________________________________________________________
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