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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