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

[Kim Lux]

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.