[Freeswitch-users] suggest best codecs to use with freeswitch for good voice quality ??

[Shabbir abbasi]

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 at 32000h,G7221 at 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 at 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 at 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 ??
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at 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
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at 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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150404/3d869b52/attachment-0001.html