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

[asterisk lists]

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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