probably the best you can do is limit the number of transcodings and/or resamplings

skype uses silk
freeswitch core uses L16
sip session uses <your choice>
asterisk core uses ?
audio is presented to the dongle as ?
cellular network uses gsm

if ? =  L16 then make the sip session use L16
use the loopback interface so that MTU is not an issue


On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
for this setup
skype   -->    freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] )     on same machine

suggest best codecs to use with freeswitch  and asterisk  for good voice quality ??

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