[Freeswitch-users] Dead channels in freeswitch

Steven Ayre steveayre at gmail.com
Mon May 6 17:47:04 MSD 2013


<profile>
  <settings>
    <param name="rtp-timeout-sec" value="30"/>
  </settings>
</profile>

Note the settings tag within profile.

The same goes for the other parameters previously mentioned.

-Steve



On 6 May 2013 12:41, Farrukh Ali <smfarrukh at live.com> wrote:

> Hi Steve,
>
> So, should I put it in profile tag?
>
> ------------------------------
> From: steveayre at gmail.com
> Date: Mon, 6 May 2013 11:08:17 +0100
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Dead channels in freeswitch
>
> Did you ensure that the parameter was set on the sofia profile? By default
> they're not contained in the sofia.conf.xml file but in the files it pulls
> in via the X-PRE-PROCESS include.
>
> It will not work if you set it in the <global_settings> section.
>
> Also did you try restarting FS after setting the parameter (restarting the
> profile should also work, but restarting the process if you can is a simple
> way to be 100% sure it's reloaded the entire configuration).
>
> -Steve
>
>
>
> On 6 May 2013 09:19, Farrukh Ali <smfarrukh at live.com> wrote:
>
> I tried to configure rtp-timeout-sec in sofia.conf.xml as
>
> <param name="rtp-timeout-sec" value="30"/>
>
> but did not get any change, dead channels don't get hung up!
> Please reply
>
> Regards,
> Muhammad Farrukh
>
> ------------------------------
> Date: Fri, 3 May 2013 09:39:14 -0500
> From: anthony.minessale at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Dead channels in freeswitch
>
> Asterisk uses the same 2 methods.  The difference is probably that one or
> both are set by default or are implicit from some other setting.
> On May 3, 2013 3:01 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> If you're on VoIP and an endpoint completely disappears (eg wifi down)
> then the is absolutely no way for the other endpoint to detect that. That
> goes whatever software you're using.
>
> It's impossible for the down endpoint to tell the other that it's down,
> since it's down. Lack of messages also isn't enough to know it's down -
> it's normal during most of the SIP call for there to be no SIP packets, and
> RTP is designed to cope with packet loss.
>
> Absolutely the only way to detect it is to spot when you haven't heard
> from them for a while and timeout. Using SIP/UDP that's either in
> signalling with session timers (send a SIP packet, wait for a reply,
> timeout if no response within N seconds), or in media with RTP timers
> (hangup call if no RTP received for N seconds).
>
> To use session timers:
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
>  http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
>
> Or to use RTP timers:
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec<http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec>
>
> -Steve
>
>
>
>
> On 3 May 2013 07:12, Farrukh Ali <smfarrukh at live.com> wrote:
>
> Yes BDF, you are right, but I have been using Asterisk for a while and
> never faced such problem, there has to be some mechanism Freeswitch is
> lagging, beside this Freeswitch is quiet flexible to use due to its unique
> XML configuration style,
> And thanks for response
>
> Regards
> Muhammad Farrukh
>
> ------------------------------
> Date: Thu, 2 May 2013 08:51:42 -0400
> From: bdfoster at davri.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Dead channels in freeswitch
>
> Just a note:
>
> In your setup, a timeout if some sort is your only option. It may not be
> efficient, it may not be perfect, but neither are your WiFi/Mobile clients
> apparently.
>
> -BDF
> On May 2, 2013 7:09 AM, "Farrukh Ali" <smfarrukh at live.com> wrote:
>
> Thanks everyone, and Steve timers might not be efficient solution but for
> now it will be good for testing purpose, could you please tell me exactly
> which file and parameter should I look for to change, is this configuration
> located in sofia.conf.xml ? kindly reply!
> and thanks to all again!
>
> Regards,
> Muhammad Farrukh
>
> ------------------------------
> From: steveayre at gmail.com
> Date: Tue, 30 Apr 2013 17:20:36 +0100
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Dead channels in freeswitch
>
> SIP over UDP has no way of knowing the other side has lost connectivity
> (eg wifi down). Sip droid will know the local network is down and be able
> to hang up the call because of that. FreeSWITCH has no such knowledge, so
> won't know the call is hung up and will continue sending RTP.
>
> 3 possibilities are:
> 1) Use session timers
> 2) There is a parameter to hang up a call is no RTP has been received for
> a certain period of time
> 3) Use SIP/TCP, keepalives might help (but the default is measured in
> hours)
>
> Note that all of these are timeouts - you simply won't be able to know the
> call has ended immediately.
>
> Steve
>
>
>
> On 30 Apr 2013, at 08:33, Farrukh Ali <smfarrukh at live.com> wrote:
>
> Hi,
>
> I need some help, there is an issue in Freeswitch with SIP clients, when a
> call is not hung up properly Freeswitch does not close its RTP channels,
> and it starts to get load on bandwidth after too many dead channels are
> generated, Improper hang up means if during call sip client went out of the
> range of WiFi, the client i.e SipDroid shows call hang up but freeswitch
> continues to send voice packets. Kindly reply
>
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