[Freeswitch-users] Verto: no inbound audio when verto client called from conference

[Mark Melling]

Thanks Giovanni for the suggestion.

I tried some more experiments and basically if I call a verto client and
add them to a conference then they don't hear the audio (although the
conference is detecting audio when they speak).

But if they dial into the conference then everything appears fine and they
do hear audio.

Specifically from fs_cli I entered:

originate <call-url> &conference(<conf-name>@default)

If call-url is a sip client then you hear the conference music, but if
call-url is a verto client you don't hear any conference music. But the
conference does detect when the verto client is speaking (at least the
status in the verto web page indicates the user is talking).

Whereas if you dialled into a conference from a verto client then you would
hear the conference music.

So I'm not sure how I can work around this.







On Wed, 5 Jul 2017 at 08:57 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On 4 July 2017 at 23:55, Mark Melling <mark.melling at savageminds.com>
> wrote:
>
> Hi,
>>
>> I have a dialplan where a user calls a number which results in them being
>> placed in a conference room, and an outbound called made to a verto client
>> when the conference is created, by defining in the dialplan:
>>
>>     <action application="conference_set_auto_outcall" data="user/1000@
>> $${domain}"/>
>>
>> The verto client (which is basically taken from the verto demo) rings and
>> is answered.
>>
>> The original inbound caller can hear the verto client speaking, but the
>> verto client can't hear the the inbound caller (although the audio from the
>> inbound caller is getting to the conference room).
>>
>> If I change the dialling order - so the verto client makes the inbound
>> call and the SIP client is dialled from the conference using the same
>> technique as before (conference_set_auto_outcall). Then everything appears
>> to work fine, i.e. each caller can here the other.
>>
>> Any suggestions as to what might be the issue or steps I could take to
>> help identify the problem.
>>
>
>
> Maybe this is because verto (webrtc) takes time to establish audio because
> of stun, etc etc
>
> Try this: instead of generating autocall from inside conference (eg
> instead of using autocall),originate call to user, wait for her to answer,
> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>
> -giovanni
>
>
>
>>
>> Thanks
>>
>> Mark
>>
>>
>>
>> _________________________________________________________________________
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>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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