[Freeswitch-users] two problems with mod_conference

[Anthony Minessale]

Anthony Minessale II

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----- Original Message ----
From: Mark D. Anderson <mda at discerning.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, March 20, 2007 11:55:42 PM
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, *}

You can avoid this by adding a sleep action to your dialplan
first answer the call from the dialplan and then use sleep to run off  1.5 
lead-in seconds to avoid missing some of the audio

<action application="answer"/>
<action application="sleep" data="1500"/>
<action application="conference" data="???"/>


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}.

Like Mike said, we are only getting the DTMF as rfc 2833 and it has been
tested to parse the packets with a duration of as little as 1ms up to several
seconds with accuracy.  As a receiver of rfc2833 there is really nothing to tune
but I know all too well from to much time using rtp that a disagreement in
the protocol could cause some issues so I recommend you take Mike up on his
offer to investigate it.


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

-mda




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