<div dir="ltr">Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.<div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven <span dir="ltr"><<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">for the freeswitch side<div><action application="set" data="absolute_codec_string=L16@16000h"/></div><div><br></div><div>for the asterisk side</div><div>you will have to figure that out yourself.</div><div><br><div><br></div></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi <span dir="ltr"><<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>thank you for reply<br></div>it is freeswitch.conf<br> <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/><br><X-PRE-PROCESS cmd="set" data="default_country=US"/><br><X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br><br> <profile name="freeswitch-sip"><br> <gateways><br> <gateway name="asterisk-local"><br> <param name="proxy" value="<a href="http://127.0.0.1:5060" target="_blank">127.0.0.1:5060</a>"/><br> <param name="retry-seconds" value="30"/><br> <param name="caller-id-in-from" value="true"/><br> </gateway><br> </gateways><br> <br> <domains><br> <domain name="all" alias="true" parse="false"/> <br> </domains><br> <br> <settings><br> <param name="debug" value="1"/><br> <param name="sip-trace" value="no"/><br> <param name="log-auth-failures" value="false"/><br> <param name="forward-unsolicited-mwi-notify" value="false"/><br> <param name="context" value="asterisk"/><br> <param name="rfc2833-pt" value="101"/><br> <param name="sip-port" value="5050"/><br> <param name="dialplan" value="XML"/><br> <param name="dtmf-type" value="info"/><br> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/><br> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/><br> <param name="use-rtp-timer" value="true"/><br> <param name="rtp-timer-name" value="soft"/><br> <param name="rtp-timeout-sec" value="300"/><br> <param name="rtp-hold-timeout-sec" value="1800"/><br> <param name="vad" value="none"/> <br> <param name="rtp-ip" value="127.0.0.1"/><br> <param name="sip-ip" value="127.0.0.1"/><br> <param name="ext-rtp-ip" value="127.0.0.1"/><br> <param name="ext-sip-ip" value="127.0.0.1"/><br> <param name="inbound-codec-negotiation" value="generous"/><br> <param name="tls" value="false"/><br> <param name="nonce-ttl" value="60"/><br> <param name="auth-calls" value="false"/><br> <param name="auth-all-packets" value="false"/><br> <param name="challenge-realm" value="auto_from"/><br> </settings><br><br><br></div>and here is asterisk sip.con<br>disallow=all<br>allow=ulaw<br>allow=alaw<br><br></div>what i need to change ??<br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven <span dir="ltr"><<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>probably the best you can do is limit the number of transcodings and/or resamplings</div><div><br></div><div><div>skype uses silk</div><div><span style="white-space:pre-wrap"> </span>freeswitch core uses L16</div><div><span style="white-space:pre-wrap"> </span>sip session uses <your choice></div><div><span style="white-space:pre-wrap"> </span>asterisk core uses ?</div><div><span style="white-space:pre-wrap"> </span>audio is presented to the dongle as ?</div><div><span style="white-space:pre-wrap"> </span>cellular network uses gsm</div><div><br></div><div>if ? = L16 then make the sip session use L16</div><div>use the loopback interface so that MTU is not an issue</div></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <span dir="ltr"><<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>for this setup<br></div>skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine <br><br>suggest best codecs to use with freeswitch and asterisk for good voice quality ??<br></div>
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