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

Steven Ayre steveayre at gmail.com
Tue Apr 7 13:57:23 MSD 2015 

That's not necessarily the case - GSM is not the only codec used by mobile
networks, there's others such as AMR

On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven at 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 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
>>>>
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>>>
>>>
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>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
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>>>
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>>
>>
>
> _________________________________________________________________________
> 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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