[Freeswitch-users] mod_conference lag

[Clint Popetz]

On Fri, Apr 2, 2010 at 2:17 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> Try calling the public conference at sip:888 at conference.freeswitch.org
> that is a properly setup box and can accept any registrations.
>

Yup, that works, as does calling my existing asterisk pbx.

>
> The sip and the conference all work fine so it's for sure a problem on your end.

Well, this is with a fresh install using:

cd /usr/src ; wget http://www.freeswitch.org/eg/Makefile ; make
make all
cd freeswitch.trunk
make install
make cd-sounds-install
make cd-moh-install

the only thing I'v changed is to add <load module="mod_skypopen"/> and
	<action application="set"
data="skype_add_outband_dtmf_also_when_bridged=true"/>

and to install skype and make it listen to skypopen.   The rest is the
default config.

-Clint


>
>
>
> On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint at 42lines.net> wrote:
>>
>>
>> On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>>>>
>>>> Hi,
>>>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on the machine is nil.  Any ideas?
>>>
>>> Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
>>
>> Correct.
>>
>>>
>>> Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
>>
>> Sip is a whole other can of worms that's not working either.  When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu.  When running xopier on linux, I can't hear anything and DTMF doesn't get through.  Both those softphones work fine with asterisk, including dtmf.  I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
>> (It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)
>> -Clint
>>
>>> -MC
>>>
>>>
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>>
>>
>>
>> --
>> Clint Popetz
>> http://42lines.net
>> Scalable Web Application Development
>>
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>
>
>
> --
> Anthony Minessale II
>
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>
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--
Clint Popetz
http://42lines.net
Scalable Web Application Development