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
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
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 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Dragos Oancea <dragos@freeswitch.org> Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Media timeout 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<mailto: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<mailto: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<mailto: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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes <shaun@sysconfig.cloud> wrote:
I've figured out our issue with media_timeout_audio, this is using ms not seconds.
Is this a bug, or is the documentation out of date? ------------------------------ *From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Dragos Oancea <dragos@freeswitch.org> *Sent:* 16 February 2022 13:49 *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Media timeout
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 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
I have updated the Confluence page https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Fil... but that variable is mentioned on other pages because it’s a wiki so… We always welcome help with updating the wiki, just ask for edit permission. Thank you. John Boteler Bote Communications From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of Dragos Oancea Sent: Friday, 18 February, 2022 05:48 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Media timeout This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes <shaun@sysconfig.cloud <mailto:shaun@sysconfig.cloud> > wrote: I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? _____ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> > on behalf of Dragos Oancea <dragos@freeswitch.org <mailto:dragos@freeswitch.org> > Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Media timeout 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 <mailto: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 <mailto: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 <mailto: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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
OK, I need to correct my statement. If I configure a timer for the sip profile configuration and specify the value for media_timeout correctly in milliseconds, everything seems to work. Thanks a lot. Von: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> Im Auftrag von Dragos Oancea Gesendet: Freitag, 18. Februar 2022 11:48 An: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Betreff: Re: [Freeswitch-users] Media timeout This has been discussed on this ML a while ago. Documentation out of date. On Fri, Feb 18, 2022 at 11:00 AM Shaun Stokes <shaun@sysconfig.cloud<mailto:shaun@sysconfig.cloud>> wrote: I've figured out our issue with media_timeout_audio, this is using ms not seconds. Is this a bug, or is the documentation out of date? ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> on behalf of Dragos Oancea <dragos@freeswitch.org<mailto:dragos@freeswitch.org>> Sent: 16 February 2022 13:49 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Media timeout 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<mailto: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<mailto: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<mailto: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 _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
participants (4)
-
Alexander Haugg
-
Bote Man
-
Dragos Oancea
-
Shaun Stokes