On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.

On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

for the freeswitch side
<action application="set" data="absolute_codec_string=L16@16000h"/>

for the asterisk side

you will have to figure that out yourself.

On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:

thank you for reply

it is freeswitch.conf
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
<X-PRE-PROCESS cmd="set" data="default_country=US"/>
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>

<profile name="freeswitch-sip">
          <gateways>
            <gateway name="asterisk-local">
              <param name="proxy" value="127.0.0.1:5060"/>
              <param name="retry-seconds" value="30"/>
              <param name="caller-id-in-from" value="true"/>
            </gateway>
          </gateways>

          <domains>
            <domain name="all" alias="true" parse="false"/>
          </domains>

          <settings>
            <param name="debug" value="1"/>
            <param name="sip-trace" value="no"/>
            <param name="log-auth-failures" value="false"/>
            <param name="forward-unsolicited-mwi-notify" value="false"/>
            <param name="context" value="asterisk"/>
            <param name="rfc2833-pt" value="101"/>
            <param name="sip-port" value="5050"/>
            <param name="dialplan" value="XML"/>
            <param name="dtmf-type" value="info"/>
            <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
            <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
            <param name="use-rtp-timer" value="true"/>
            <param name="rtp-timer-name" value="soft"/>
            <param name="rtp-timeout-sec" value="300"/>
            <param name="rtp-hold-timeout-sec" value="1800"/>
            <param name="vad" value="none"/>
            <param name="rtp-ip" value="127.0.0.1"/>
            <param name="sip-ip" value="127.0.0.1"/>
            <param name="ext-rtp-ip" value="127.0.0.1"/>
            <param name="ext-sip-ip" value="127.0.0.1"/>
            <param name="inbound-codec-negotiation" value="generous"/>
            <param name="tls" value="false"/>
            <param name="nonce-ttl" value="60"/>
            <param name="auth-calls" value="false"/>
            <param name="auth-all-packets" value="false"/>
            <param name="challenge-realm" value="auto_from"/>
          </settings>

and here is asterisk sip.con
disallow=all
allow=ulaw
allow=alaw

what i need to change ??

On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:

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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_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

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http://confluence.freeswitch.org
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