[Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk

[Kim Lux]

Thanks for the tips.

The Grandstream doesn't have a G711 or uLaw option for codecs.  It has
PCMU, PCMA and iLBC. Are any of these related to G711 ?

Grandstreams have echo cancellation and it appears to be working after a
few seconds of conversation. 

What is VAD ?


On Thu, 2005-01-27 at 20:22 -0500, asterisk lists wrote:
> Try using g711 (ulaw) and make sure to turn Silence Suppression OFF as
> asterisk needs the full audio stream for assembling the audio streams.
>  Once you get the call quality good using g711 (ulaw), then you can
> play around with the other codecs (g729, etc).  Also, try a 20ms frame
> size.
> 
> Unfortunately, echo is usally introduced at the central office due to
> an impedence imbalance.  Some SIP phones have echo cancellation
> options built-in to compensate (not sure the Grandstream has that
> feature).
> 
> You may also see if you have VAD enabled in the phone.  If you do, turn it OFF.
> 
> Hope that helps!
> 
> - Pedro
> VoIP by TRACI.net
> 
> 
> On Thu, 27 Jan 2005 08:53:02 -0700, Kim Lux <lux at diesel-research.com> wrote:
> > I'm testing a bunch of stuff before we implement our system.
> > 
> > I've got a SIP account and Grandstream phones.  We haven't started using
> > asterisk yet.  Generally we've got good voice quality from all the
> > offices except:
> > 
> > a) We get a lot of echo in the first 10 seconds or so of the call, only
> > on the VOIP calling end.  The callee says the speech sounds normal.  To
> > the caller, the first Hello is almost intelligible with echo.
> > 
> > b) The first part of an abrupt statement from one party gets "clipped".
> > In conversation, when talking switches from one party to the other, a
> > tiny bit of speach gets clipped.
> > 
> > c) If both parties talk at once there is a bit of dropout.
> > 
> > We'd like to improve the voice quality in these respects.  Otherwise the
> > voice quality is excellent.  I've been told it is better than the
> > traditional system several times.
> > 
> > Questions:
> > 
> > a) Are certain codecs better than others at quickly getting the echo
> > cancellation setup ?  Is there a way to get the echo out of the call
> > immediately ?  (Is there a document explaining the features and pitfalls
> > of all the codecs somewhere ?)
> > 
> > b) Is there a way to eliminate the speech clipping when speakers change
> > or both talk at once ?  I've read about asterisk injecting noise and/or
> > sending packets in the absence of speech.  Would that help ?  Is this
> > what the Grandstream "Silence Suppression" is about ?
> > 
> > c) How does one know where to set the following:
> > 
> > iLBC frame size: 20ms 30ms
> > iLBC payload type: (between 96 and 127, default is 98)
> > Silence Suppression: No Yes
> > Voice Frames per TX: (up to 10/20/32/64 for G711/G726/G723/other codecs
> > respectively)
> > Layer 3 QoS: (Diff-Serv or Precedence value)
> > Layer 2 QoS: 802.1Q/VLAN Tag      802.1p priority value  (0-7)
> > 
> > d) One place we've really got a problem is when we use a Grandstream in
> > a big echoy (sp!) room.  We seem to get echo from the room into the call
> > which seems to fool the echo cancellation.  Any ideas on how to get
> > around this problem ?
> > 
> > d) How is asterisk going to change our sound quality  when it is added
> > between the phones and the SIP provider ?  Does it have features that
> > will help with the echo and clipping and if so, how much improvement
> > should we expect ?
> > 
> > Thanks.
> > 
> > --
> > Kim Lux,  Diesel Research Inc.
> > 
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-- 
Kim Lux,  Diesel Research Inc.