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

[Russell Treleaven]

for the freeswitch side
<action application="set" data="absolute_codec_string=L16 at 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 at gmail.com>
wrote:

> thank you for reply
> it is freeswitch.conf
>  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
> <X-PRE-PROCESS cmd="set" data="default_country=US"/>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>
>  <profile name="freeswitch-sip">
>           <gateways>
>             <gateway name="asterisk-local">
>               <param name="proxy" value="127.0.0.1:5060"/>
>               <param name="retry-seconds" value="30"/>
>               <param name="caller-id-in-from" value="true"/>
>             </gateway>
>           </gateways>
>
>           <domains>
>             <domain name="all" alias="true" parse="false"/>
>           </domains>
>
>           <settings>
>             <param name="debug" value="1"/>
>             <param name="sip-trace" value="no"/>
>             <param name="log-auth-failures" value="false"/>
>             <param name="forward-unsolicited-mwi-notify" value="false"/>
>             <param name="context" value="asterisk"/>
>             <param name="rfc2833-pt" value="101"/>
>             <param name="sip-port" value="5050"/>
>             <param name="dialplan" value="XML"/>
>             <param name="dtmf-type" value="info"/>
>             <param name="inbound-codec-prefs"
> value="$${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
>>>
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>>> 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
>>
>
>
> _________________________________________________________________________
> 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
>
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