[Freeswitch-users] Dead channels in freeswitch

Anthony Minessale anthony.minessale at gmail.com
Fri May 3 18:39:14 MSD 2013


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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>>
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
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