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

[Russell Treleaven]

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 at bunnykick.ca>
wrote:

> 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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