[Freeswitch-users] rtp-timeout-sec VS media_timeout

Dragos Oancea dragos at freeswitch.org
Thu Mar 4 09:04:26 UTC 2021


For media timeout there are the following chan vars:
media_timeout,
media_hold_timeout, media_timeout, media_hold_timeout_video,
media_hold_timeout_audio, media_timeout_audio
.

They are in milliseconds, not seconds like rtp-timeout-sec .

<param name="media_timeout" value="300000"/>

On Thu, Mar 4, 2021 at 4:47 AM David P <davidswalkabout at gmail.com> wrote:

> Hi Allan,
>
> I don't know if the media_timeout=300 behavior you saw is a bug or not,
> but I wanted to add my own observation of weirdness about hangup
> cause MEDIA_TIMEOUT...
>
> I just noticed a conference end abruptly after one leg spoke for 5
> minutes. The logs aren't clear why this happened, but it seems that <param
> name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is
> the reason -- it seems that because the *other* leg of the conference
> remained silent, the RTP timeout was reached.
>
> I couldn't find any confluence pages about MEDIA_TIMEOUT by googling.
>
> On Fri, Jan 8, 2021 at 1:08 AM <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>> ---------- Forwarded message ----------
>> From: Allan Kristensen <ak at hejdu.dk>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Wed, 6 Jan 2021 19:37:33 +0100
>> Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
>> We had some problems with "hanging channels" for our webrtc clients (via
>> kamailio).
>> To solve the problem I tried to use "media_timeout" setting but it didn't
>> really work. So I tried the deprecated "rtp-timeout-sec" and this actually
>> works fine?
>>
>> Not working:
>> <param name="media_timeout" value="300"/>
>>
>> Working:
>> <param name="rtp-timeout-sec" value="300"/>
>>
>> How to reproduce: Make a call using webrtc and just close browser window,
>> after some time freeswitch should close the channel because of missing rtp.
>> Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long
>> time!" forever...
>>
>> Anyway, it works now....just curious why...a typo or bug?
>>
>> /Allan
>>
>> Using:
>> FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit
>>
> _________________________________________________________________________
>
> 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 at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210304/645f17a9/attachment.html>


More information about the FreeSWITCH-users mailing list