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

I put the Who? in Mishehu mishehu at freeswitch.org
Tue Apr 7 20:42:27 MSD 2015


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 
> <mailto: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 <mailto: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 <mailto:shabbirabbasi92 at gmail.com>>
>         wrote:
>
>             thank you for reply
>             it is freeswitch.conf
>              <X-PRE-PROCESS cmd="set"
>             data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>             <X-PRE-PROCESS cmd="set" data="default_country=US"/>
>             <X-PRE-PROCESS cmd="set"
>             data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>
>              <profile name="freeswitch-sip">
>                       <gateways>
>                         <gateway name="asterisk-local">
>                           <param name="proxy" value="127.0.0.1:5060
>             <http://127.0.0.1:5060>"/>
>                           <param name="retry-seconds" value="30"/>
>                           <param name="caller-id-in-from" value="true"/>
>                         </gateway>
>                       </gateways>
>
>                       <domains>
>                         <domain name="all" alias="true" parse="false"/>
>                       </domains>
>
>                       <settings>
>                         <param name="debug" value="1"/>
>                         <param name="sip-trace" value="no"/>
>                         <param name="log-auth-failures" value="false"/>
>                         <param name="forward-unsolicited-mwi-notify"
>             value="false"/>
>                         <param name="context" value="asterisk"/>
>                         <param name="rfc2833-pt" value="101"/>
>                         <param name="sip-port" value="5050"/>
>                         <param name="dialplan" value="XML"/>
>                         <param name="dtmf-type" value="info"/>
>                         <param name="inbound-codec-prefs"
>             value="$${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 <mailto: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
>                 <mailto: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 ??
>
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>
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>
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