[Asterisk-Users] Re: OH323 incoming audio stutter

[Tony Mountifield]

In article <119663dc050425081356bec489 at mail.gmail.com>,
Rafael J. Risco G.V. <rafael.risco at gmail.com> wrote:
> Hi Michael and Tony
> I have the same problem here and I have been able to check that this
> problem can be solved disabling VAD in h323 destination routers, I
> think this is a common problem with h323 and oh323 modules users and
> for me has become a nightmare because my service provider can no
> longer disable VAD support independently for my connection... I will
> appreciate if you can include it in some future releases.

Rafael - thank you for your comments!

I had suspected it was something to do with VAD. When conducting some
tests with a colleague, we noticed that the audio coming into Asterisk
via H.323 had silence suppression on it, and by experimenting with
varying the level of background noise we could hear the squelch cut in
and out.

I don't know why it works for a while and then fails, unless there is
something in chan_oh323 or openh323 that builds up and then reaches
a threshold.

I have demonstrated that two Asterisk boxes talking OH323 to each other
do not experience the problem.

Michael - does H.323 allow a peer to negotiate with the other end to
disable VAD? If so, is there a way in chan_oh323 to invoke that option?

I haven't had the chance yet to capture any RTP traces. Later this week
if it is still an issue with new boxes being installed.

> regards
> rafael risco
> Millicom Peru
> 
> 
> 
> On 4/22/05, Michael Manousos <manousos at inaccessnetworks.com> wrote:
> > 
> > Hi Tony,
> > 
> > Can you get an ethereal trace of the RTP packets on both
> > directions? A short analysis of those streams (from within the
> > ethereal tools) would help us find the problem.
> > 
> > Michael.
> > 
> > Tony Mountifield wrote:
> > > I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of
> > > pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect
> > > to my provider's switch.
> > >
> > > The effect that I am seeing is that a call starts off fine, but suddenly
> > > after a few minutes the audio coming into Asterisk via OH323 gets very
> > > broken up to the point of being about 90% silence with occasional brief
> > > snippets of audio getting through.
> > >
> > > When this happens, the audio going out from Asterisk to the other end
> > > is still fine, with no disturbances.
> > >
> > > I have observed this both when using SIP for the local leg of the call
> > > and when using IAX.
> > >
> > > I'm not really sure where to look to diagnose this, not whether it is
> > > likely to be an Asterisk problem or something in the switch.
> > >
> > > Any advice would be appreciated!

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org