[Freeswitch-users] sip stack problem in fsw

Tue Sep 21 12:18:09 PDT 2010

What I tried to explain to you was to look at your phone, the phone is
the one with the vad, if you turn it off your problem will be gone and
it will be much easier than hacking FS.

If hacking FS is indeed what you wish, then add this to your dialplan
*before* you call bridge.

<action application="export" data="bridge_generate_comfort_noise=true"/>

This causes FS to generate fake empty RTP packets full of generated
silent comfort noise even when there is no rtp received by the inbound
leg of the bridge.

On Tue, Sep 21, 2010 at 1:46 PM,  <covici at ccs.covici.com> wrote:
> No conference, the only reason I bring this up at all is that it works
> with asterisk, so sofia is doing something different -- maybe not worng
> -- but I would like it to behave so that I don't get hung up on.  I have
> had trouble with more than one sip provider with this terminating
> carrier, them and their comfort noise thing!  So, should I use vad in
> fsw?  Would it generate the noise required?
>
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> 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
>> >
>> >
>> > _______________________________________________
>> > 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
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > _______________________________________________
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>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
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>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900