[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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>
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