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

Russell Treleaven rtreleaven at bunnykick.ca
Tue Apr 7 20:59:57 MSD 2015 

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
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>>>>>>
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>>>>>>
>>>>>
>>>>>
>>>>>
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>>>>>
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>>>>
>>>>
>>>>
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>>>>
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>>>
>>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
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>
>
>
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>
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>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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