[Freeswitch-users] sip stack problem in fsw

[Anthony Minessale]

are you in a conference running on FreeSWITCH or just bridging a phone
out to a far end?

There is nothing in FreeSWITCH that runs vad by default.
There are vad params in the RTP stack but they are disabled by default

 <!-- VAD choose one (out is a good choice); -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <!-- <param name="vad" value="both"/> -->
    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
    <!--


If these are not set then its the actual phone who is choosing to do
vad not FreeSWITCH so disable it in your phone.


any problems you are having in this regard are 100% to do with who you
are calling being only partially implemented to carry voice over ip.






On Mon, Sep 20, 2010 at 5:34 PM, JRichey <jrichey at itltd.net> wrote:
> The gateway sending calls to the FreeSwitch box ends up dropping the calls.
> There are also some pops and clicks that aren't there when playing a
> continuous audio file.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Brian
> West
> Sent: Monday, September 20, 2010 12:23 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] sip stack problem in fsw
>
>
> You shouldn't see RTP between files when sleeping... its 100% valid to NOT
> send any media when their is NOTHING to be sent.  Does your phone freak out
> or something?
>
> /b
>
> On Sep 20, 2010, at 2:08 PM, JRichey wrote:
>
>> I'm in a similar situation where I need to disable VAD because the other
> end
>> will disconnect the call if the RTP stops.  I've tried the setting Anthony
>> suggested as well as several others I've found on the wiki, but I've not
>> been able to get it to work.  The variables I've been trying are listed
>> below.
>>
>> suppress-cng
>> suppress_cng
>> rtp_disable_vad_out
>> rtp_disable_vad_in
>> bridge_generate_comfort_noise
>> send_silence_when_idle
>>
>>
>> For my test calls I'm just playing back sound files with sleep in-between
>> them and I see no RTP when calling sleep.  I'm using tcpdump and wireshark
>> to analyze the calls.
>>
>>
>> -Justin Richey
>> jrichey at ITLtd.net
>
>
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-- 
Anthony Minessale II

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