[Freeswitch-users] two problems with mod_conference

[Michael Jerris]

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Mark D. Anderson
> Sent: Wednesday, March 21, 2007 12:56 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] two problems with mod_conference
> 
> I'm currently running head (4687) but these problems have
> been unchanged since I've started playing with mod_conference
> about 2 months ago.
> 
> I have tried a cross-product of
>   two ITSPs (vitelity and teliax)
>   either landline caller or cell caller
>   mod_conference "native", or a javascript based on confroom.js
> I have found some differences in behavior in these 8 cases.
> 
> 1. Missing 1 second of initial audio (callers don't hear
> the beginning of the "please enter your pin" message).
> It doesn't matter whether I move the session.answer()
> earlier. This happens except for {teliax, landline, *}
> 

We need to look at the timing of when media is established on this.  I
would start by looking at one variable, and comparing the sip signaling
to the rtp stream.  We need to understand the differences to make more
sense of what is happening here.  There are lots of things that can
affect this including provider doing proper support for early media, and
the end connection providing that support... I am happy to go through
this with you online.

> 2. Doubling (and tripling) of incoming dtmf digits.
> It is quite hard to get it to accept a pin, because
> it keeps thinking I've entered a digit twice, when I
> haven't. As far as I can see, there is no user configuration
> to tune dtmf configuration (like min/max length of a tone,
> or echo cancellation).
> I also don't know if there is a way to force freeswitch/sofia
> to use only a particular dtmf system.
> Note that the problem seems worse when running javascript
> confroom.js, but it happens both ways.
> This happens with both ITSPs.
> I see it except for {*, landline, native mod_conference.c}.

There are no settings for this because we are not doing any inband dtmf
conversion.  We are only receiving what the provider is sending, so the
question is, are we handling what is coming from the provider
incorrectly, or is the provider detecting incorrectly then sending us
double/triple digits.  I would need to see a trace of the rtp traffic,
probably along with FreeSWITCH(tm) debug of the same time to make more
sense of this.  Again, catch me on irc or IM and we can figure out the
best way to hunt down where the problem is.


> 
> Any suggestion for causes, fixes, ways to diagnose further,
> etc., are welcome.
> 
> -mda