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

Dmitry Lysenko dvl36.ripe.nick at gmail.com
Tue Apr 7 22:33:36 MSD 2015


Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best
codec, in this case, will be L16.

2015-04-07 19:59 GMT+03:00 Russell Treleaven <rtreleaven at bunnykick.ca>:

> Can the GSM dongle do AMR?
>
> On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>>  I think that AMR is one of the codecs that does not allow for
>> transcoding, at least with the in-tree module.  Might require something
>> additional.
>>
>> --
>> Yossi Neiman
>>
>> On 04/07/2015 04:57 AM, Steven Ayre wrote:
>>
>>  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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>>>>>>> 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
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>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> 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
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>>>
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>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
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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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>>
>
>
> _________________________________________________________________________
> 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
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